pax_global_header00006660000000000000000000000064133577444530014531gustar00rootroot0000000000000052 comment=6761218e51699f46bf25c377e65b3e9ea5e434b9 openal-soft-openal-soft-1.19.1/000077500000000000000000000000001335774445300162765ustar00rootroot00000000000000openal-soft-openal-soft-1.19.1/.gitignore000066400000000000000000000000701335774445300202630ustar00rootroot00000000000000build*/ winbuild/ win64build/ openal-soft.kdev4 .kdev4/ openal-soft-openal-soft-1.19.1/.travis.yml000066400000000000000000000042031335774445300204060ustar00rootroot00000000000000language: c matrix: include: - os: linux dist: trusty - os: linux dist: trusty env: - BUILD_ANDROID=true - os: osx sudo: required install: - > if [[ "${TRAVIS_OS_NAME}" == "linux" && -z "${BUILD_ANDROID}" ]]; then # Install pulseaudio, portaudio, ALSA, JACK dependencies for # corresponding backends. # Install Qt5 dependency for alsoft-config. sudo apt-get install -qq \ libpulse-dev \ portaudio19-dev \ libasound2-dev \ libjack-dev \ qtbase5-dev fi - > if [[ "${TRAVIS_OS_NAME}" == "linux" && "${BUILD_ANDROID}" == "true" ]]; then curl -o ~/android-ndk.zip https://dl.google.com/android/repository/android-ndk-r15-linux-x86_64.zip unzip -q ~/android-ndk.zip -d ~ \ 'android-ndk-r15/build/cmake/*' \ 'android-ndk-r15/build/core/toolchains/arm-linux-androideabi-*/*' \ 'android-ndk-r15/platforms/android-14/arch-arm/*' \ 'android-ndk-r15/source.properties' \ 'android-ndk-r15/sources/cxx-stl/gnu-libstdc++/4.9/libs/armeabi-v7a/*' \ 'android-ndk-r15/sources/cxx-stl/gnu-libstdc++/4.9/include/*' \ 'android-ndk-r15/sysroot/*' \ 'android-ndk-r15/toolchains/arm-linux-androideabi-4.9/prebuilt/linux-x86_64/*' \ 'android-ndk-r15/toolchains/llvm/prebuilt/linux-x86_64/*' fi script: - > if [[ "${TRAVIS_OS_NAME}" == "linux" && -z "${BUILD_ANDROID}" ]]; then cmake \ -DALSOFT_REQUIRE_ALSA=ON \ -DALSOFT_REQUIRE_OSS=ON \ -DALSOFT_REQUIRE_PORTAUDIO=ON \ -DALSOFT_REQUIRE_PULSEAUDIO=ON \ -DALSOFT_REQUIRE_JACK=ON \ -DALSOFT_EMBED_HRTF_DATA=YES \ . fi - > if [[ "${TRAVIS_OS_NAME}" == "linux" && "${BUILD_ANDROID}" == "true" ]]; then cmake \ -DCMAKE_TOOLCHAIN_FILE=~/android-ndk-r15/build/cmake/android.toolchain.cmake \ -DALSOFT_REQUIRE_OPENSL=ON \ -DALSOFT_EMBED_HRTF_DATA=YES \ . fi - > if [[ "${TRAVIS_OS_NAME}" == "osx" ]]; then cmake \ -DALSOFT_REQUIRE_COREAUDIO=ON \ -DALSOFT_EMBED_HRTF_DATA=YES \ . fi - make -j2 openal-soft-openal-soft-1.19.1/Alc/000077500000000000000000000000001335774445300167755ustar00rootroot00000000000000openal-soft-openal-soft-1.19.1/Alc/ALc.c000066400000000000000000004232561335774445300176140ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include "version.h" #include #include #include #include #include #include #include "alMain.h" #include "alSource.h" #include "alListener.h" #include "alSource.h" #include "alBuffer.h" #include "alFilter.h" #include "alEffect.h" #include "alAuxEffectSlot.h" #include "alError.h" #include "mastering.h" #include "bformatdec.h" #include "alu.h" #include "alconfig.h" #include "ringbuffer.h" #include "fpu_modes.h" #include "cpu_caps.h" #include "compat.h" #include "threads.h" #include "alstring.h" #include "almalloc.h" #include "backends/base.h" /************************************************ * Backends ************************************************/ struct BackendInfo { const char *name; ALCbackendFactory* (*getFactory)(void); }; static struct BackendInfo BackendList[] = { #ifdef HAVE_JACK { "jack", ALCjackBackendFactory_getFactory }, #endif #ifdef HAVE_PULSEAUDIO { "pulse", ALCpulseBackendFactory_getFactory }, #endif #ifdef HAVE_ALSA { "alsa", ALCalsaBackendFactory_getFactory }, #endif #ifdef HAVE_COREAUDIO { "core", ALCcoreAudioBackendFactory_getFactory }, #endif #ifdef HAVE_SOLARIS { "solaris", ALCsolarisBackendFactory_getFactory }, #endif #ifdef HAVE_SNDIO { "sndio", SndioBackendFactory_getFactory }, #endif #ifdef HAVE_OSS { "oss", ALCossBackendFactory_getFactory }, #endif #ifdef HAVE_QSA { "qsa", ALCqsaBackendFactory_getFactory }, #endif #ifdef HAVE_WASAPI { "wasapi", ALCwasapiBackendFactory_getFactory }, #endif #ifdef HAVE_DSOUND { "dsound", ALCdsoundBackendFactory_getFactory }, #endif #ifdef HAVE_WINMM { "winmm", ALCwinmmBackendFactory_getFactory }, #endif #ifdef HAVE_PORTAUDIO { "port", ALCportBackendFactory_getFactory }, #endif #ifdef HAVE_OPENSL { "opensl", ALCopenslBackendFactory_getFactory }, #endif #ifdef HAVE_SDL2 { "sdl2", ALCsdl2BackendFactory_getFactory }, #endif { "null", ALCnullBackendFactory_getFactory }, #ifdef HAVE_WAVE { "wave", ALCwaveBackendFactory_getFactory }, #endif }; static ALsizei BackendListSize = COUNTOF(BackendList); #undef EmptyFuncs static struct BackendInfo PlaybackBackend; static struct BackendInfo CaptureBackend; /************************************************ * Functions, enums, and errors ************************************************/ #define DECL(x) { #x, (ALCvoid*)(x) } static const struct { const ALCchar *funcName; ALCvoid *address; } alcFunctions[] = { DECL(alcCreateContext), DECL(alcMakeContextCurrent), DECL(alcProcessContext), DECL(alcSuspendContext), DECL(alcDestroyContext), DECL(alcGetCurrentContext), DECL(alcGetContextsDevice), DECL(alcOpenDevice), DECL(alcCloseDevice), DECL(alcGetError), DECL(alcIsExtensionPresent), DECL(alcGetProcAddress), DECL(alcGetEnumValue), DECL(alcGetString), DECL(alcGetIntegerv), DECL(alcCaptureOpenDevice), DECL(alcCaptureCloseDevice), DECL(alcCaptureStart), DECL(alcCaptureStop), DECL(alcCaptureSamples), DECL(alcSetThreadContext), DECL(alcGetThreadContext), DECL(alcLoopbackOpenDeviceSOFT), DECL(alcIsRenderFormatSupportedSOFT), DECL(alcRenderSamplesSOFT), DECL(alcDevicePauseSOFT), DECL(alcDeviceResumeSOFT), DECL(alcGetStringiSOFT), DECL(alcResetDeviceSOFT), DECL(alcGetInteger64vSOFT), DECL(alEnable), DECL(alDisable), DECL(alIsEnabled), DECL(alGetString), DECL(alGetBooleanv), DECL(alGetIntegerv), DECL(alGetFloatv), DECL(alGetDoublev), DECL(alGetBoolean), DECL(alGetInteger), DECL(alGetFloat), DECL(alGetDouble), DECL(alGetError), DECL(alIsExtensionPresent), DECL(alGetProcAddress), DECL(alGetEnumValue), DECL(alListenerf), DECL(alListener3f), DECL(alListenerfv), DECL(alListeneri), DECL(alListener3i), DECL(alListeneriv), DECL(alGetListenerf), DECL(alGetListener3f), DECL(alGetListenerfv), DECL(alGetListeneri), DECL(alGetListener3i), DECL(alGetListeneriv), DECL(alGenSources), DECL(alDeleteSources), DECL(alIsSource), DECL(alSourcef), DECL(alSource3f), DECL(alSourcefv), DECL(alSourcei), DECL(alSource3i), DECL(alSourceiv), DECL(alGetSourcef), DECL(alGetSource3f), DECL(alGetSourcefv), DECL(alGetSourcei), DECL(alGetSource3i), DECL(alGetSourceiv), DECL(alSourcePlayv), DECL(alSourceStopv), DECL(alSourceRewindv), DECL(alSourcePausev), DECL(alSourcePlay), DECL(alSourceStop), DECL(alSourceRewind), DECL(alSourcePause), DECL(alSourceQueueBuffers), DECL(alSourceUnqueueBuffers), DECL(alGenBuffers), DECL(alDeleteBuffers), DECL(alIsBuffer), DECL(alBufferData), DECL(alBufferf), DECL(alBuffer3f), DECL(alBufferfv), DECL(alBufferi), DECL(alBuffer3i), DECL(alBufferiv), DECL(alGetBufferf), DECL(alGetBuffer3f), DECL(alGetBufferfv), DECL(alGetBufferi), DECL(alGetBuffer3i), DECL(alGetBufferiv), DECL(alDopplerFactor), DECL(alDopplerVelocity), DECL(alSpeedOfSound), DECL(alDistanceModel), DECL(alGenFilters), DECL(alDeleteFilters), DECL(alIsFilter), DECL(alFilteri), DECL(alFilteriv), DECL(alFilterf), DECL(alFilterfv), DECL(alGetFilteri), DECL(alGetFilteriv), DECL(alGetFilterf), DECL(alGetFilterfv), DECL(alGenEffects), DECL(alDeleteEffects), DECL(alIsEffect), DECL(alEffecti), DECL(alEffectiv), DECL(alEffectf), DECL(alEffectfv), DECL(alGetEffecti), DECL(alGetEffectiv), DECL(alGetEffectf), DECL(alGetEffectfv), DECL(alGenAuxiliaryEffectSlots), DECL(alDeleteAuxiliaryEffectSlots), DECL(alIsAuxiliaryEffectSlot), DECL(alAuxiliaryEffectSloti), DECL(alAuxiliaryEffectSlotiv), DECL(alAuxiliaryEffectSlotf), DECL(alAuxiliaryEffectSlotfv), DECL(alGetAuxiliaryEffectSloti), DECL(alGetAuxiliaryEffectSlotiv), DECL(alGetAuxiliaryEffectSlotf), DECL(alGetAuxiliaryEffectSlotfv), DECL(alDeferUpdatesSOFT), DECL(alProcessUpdatesSOFT), DECL(alSourcedSOFT), DECL(alSource3dSOFT), DECL(alSourcedvSOFT), DECL(alGetSourcedSOFT), DECL(alGetSource3dSOFT), DECL(alGetSourcedvSOFT), DECL(alSourcei64SOFT), DECL(alSource3i64SOFT), DECL(alSourcei64vSOFT), DECL(alGetSourcei64SOFT), DECL(alGetSource3i64SOFT), DECL(alGetSourcei64vSOFT), DECL(alGetStringiSOFT), DECL(alBufferStorageSOFT), DECL(alMapBufferSOFT), DECL(alUnmapBufferSOFT), DECL(alFlushMappedBufferSOFT), DECL(alEventControlSOFT), DECL(alEventCallbackSOFT), DECL(alGetPointerSOFT), DECL(alGetPointervSOFT), }; #undef DECL #define DECL(x) { #x, (x) } static const struct { const ALCchar *enumName; ALCenum value; } alcEnumerations[] = { DECL(ALC_INVALID), DECL(ALC_FALSE), DECL(ALC_TRUE), DECL(ALC_MAJOR_VERSION), DECL(ALC_MINOR_VERSION), DECL(ALC_ATTRIBUTES_SIZE), DECL(ALC_ALL_ATTRIBUTES), DECL(ALC_DEFAULT_DEVICE_SPECIFIER), DECL(ALC_DEVICE_SPECIFIER), DECL(ALC_ALL_DEVICES_SPECIFIER), DECL(ALC_DEFAULT_ALL_DEVICES_SPECIFIER), DECL(ALC_EXTENSIONS), DECL(ALC_FREQUENCY), DECL(ALC_REFRESH), DECL(ALC_SYNC), DECL(ALC_MONO_SOURCES), DECL(ALC_STEREO_SOURCES), DECL(ALC_CAPTURE_DEVICE_SPECIFIER), DECL(ALC_CAPTURE_DEFAULT_DEVICE_SPECIFIER), DECL(ALC_CAPTURE_SAMPLES), DECL(ALC_CONNECTED), DECL(ALC_EFX_MAJOR_VERSION), DECL(ALC_EFX_MINOR_VERSION), DECL(ALC_MAX_AUXILIARY_SENDS), DECL(ALC_FORMAT_CHANNELS_SOFT), DECL(ALC_FORMAT_TYPE_SOFT), DECL(ALC_MONO_SOFT), DECL(ALC_STEREO_SOFT), DECL(ALC_QUAD_SOFT), DECL(ALC_5POINT1_SOFT), DECL(ALC_6POINT1_SOFT), DECL(ALC_7POINT1_SOFT), DECL(ALC_BFORMAT3D_SOFT), DECL(ALC_BYTE_SOFT), DECL(ALC_UNSIGNED_BYTE_SOFT), DECL(ALC_SHORT_SOFT), DECL(ALC_UNSIGNED_SHORT_SOFT), DECL(ALC_INT_SOFT), DECL(ALC_UNSIGNED_INT_SOFT), DECL(ALC_FLOAT_SOFT), DECL(ALC_HRTF_SOFT), DECL(ALC_DONT_CARE_SOFT), DECL(ALC_HRTF_STATUS_SOFT), DECL(ALC_HRTF_DISABLED_SOFT), DECL(ALC_HRTF_ENABLED_SOFT), DECL(ALC_HRTF_DENIED_SOFT), DECL(ALC_HRTF_REQUIRED_SOFT), DECL(ALC_HRTF_HEADPHONES_DETECTED_SOFT), DECL(ALC_HRTF_UNSUPPORTED_FORMAT_SOFT), DECL(ALC_NUM_HRTF_SPECIFIERS_SOFT), DECL(ALC_HRTF_SPECIFIER_SOFT), DECL(ALC_HRTF_ID_SOFT), DECL(ALC_AMBISONIC_LAYOUT_SOFT), DECL(ALC_AMBISONIC_SCALING_SOFT), DECL(ALC_AMBISONIC_ORDER_SOFT), DECL(ALC_ACN_SOFT), DECL(ALC_FUMA_SOFT), DECL(ALC_N3D_SOFT), DECL(ALC_SN3D_SOFT), DECL(ALC_OUTPUT_LIMITER_SOFT), DECL(ALC_NO_ERROR), DECL(ALC_INVALID_DEVICE), DECL(ALC_INVALID_CONTEXT), DECL(ALC_INVALID_ENUM), DECL(ALC_INVALID_VALUE), DECL(ALC_OUT_OF_MEMORY), DECL(AL_INVALID), DECL(AL_NONE), DECL(AL_FALSE), DECL(AL_TRUE), DECL(AL_SOURCE_RELATIVE), DECL(AL_CONE_INNER_ANGLE), DECL(AL_CONE_OUTER_ANGLE), DECL(AL_PITCH), DECL(AL_POSITION), DECL(AL_DIRECTION), DECL(AL_VELOCITY), DECL(AL_LOOPING), DECL(AL_BUFFER), DECL(AL_GAIN), DECL(AL_MIN_GAIN), DECL(AL_MAX_GAIN), DECL(AL_ORIENTATION), DECL(AL_REFERENCE_DISTANCE), DECL(AL_ROLLOFF_FACTOR), DECL(AL_CONE_OUTER_GAIN), DECL(AL_MAX_DISTANCE), DECL(AL_SEC_OFFSET), DECL(AL_SAMPLE_OFFSET), DECL(AL_BYTE_OFFSET), DECL(AL_SOURCE_TYPE), DECL(AL_STATIC), DECL(AL_STREAMING), DECL(AL_UNDETERMINED), DECL(AL_METERS_PER_UNIT), DECL(AL_LOOP_POINTS_SOFT), DECL(AL_DIRECT_CHANNELS_SOFT), DECL(AL_DIRECT_FILTER), DECL(AL_AUXILIARY_SEND_FILTER), DECL(AL_AIR_ABSORPTION_FACTOR), DECL(AL_ROOM_ROLLOFF_FACTOR), DECL(AL_CONE_OUTER_GAINHF), DECL(AL_DIRECT_FILTER_GAINHF_AUTO), DECL(AL_AUXILIARY_SEND_FILTER_GAIN_AUTO), DECL(AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO), DECL(AL_SOURCE_STATE), DECL(AL_INITIAL), DECL(AL_PLAYING), DECL(AL_PAUSED), DECL(AL_STOPPED), DECL(AL_BUFFERS_QUEUED), DECL(AL_BUFFERS_PROCESSED), DECL(AL_FORMAT_MONO8), DECL(AL_FORMAT_MONO16), DECL(AL_FORMAT_MONO_FLOAT32), DECL(AL_FORMAT_MONO_DOUBLE_EXT), DECL(AL_FORMAT_STEREO8), DECL(AL_FORMAT_STEREO16), DECL(AL_FORMAT_STEREO_FLOAT32), DECL(AL_FORMAT_STEREO_DOUBLE_EXT), DECL(AL_FORMAT_MONO_IMA4), DECL(AL_FORMAT_STEREO_IMA4), DECL(AL_FORMAT_MONO_MSADPCM_SOFT), DECL(AL_FORMAT_STEREO_MSADPCM_SOFT), DECL(AL_FORMAT_QUAD8_LOKI), DECL(AL_FORMAT_QUAD16_LOKI), DECL(AL_FORMAT_QUAD8), DECL(AL_FORMAT_QUAD16), DECL(AL_FORMAT_QUAD32), DECL(AL_FORMAT_51CHN8), DECL(AL_FORMAT_51CHN16), DECL(AL_FORMAT_51CHN32), DECL(AL_FORMAT_61CHN8), DECL(AL_FORMAT_61CHN16), DECL(AL_FORMAT_61CHN32), DECL(AL_FORMAT_71CHN8), DECL(AL_FORMAT_71CHN16), DECL(AL_FORMAT_71CHN32), DECL(AL_FORMAT_REAR8), DECL(AL_FORMAT_REAR16), DECL(AL_FORMAT_REAR32), DECL(AL_FORMAT_MONO_MULAW), DECL(AL_FORMAT_MONO_MULAW_EXT), DECL(AL_FORMAT_STEREO_MULAW), DECL(AL_FORMAT_STEREO_MULAW_EXT), DECL(AL_FORMAT_QUAD_MULAW), DECL(AL_FORMAT_51CHN_MULAW), DECL(AL_FORMAT_61CHN_MULAW), DECL(AL_FORMAT_71CHN_MULAW), DECL(AL_FORMAT_REAR_MULAW), DECL(AL_FORMAT_MONO_ALAW_EXT), DECL(AL_FORMAT_STEREO_ALAW_EXT), DECL(AL_FORMAT_BFORMAT2D_8), DECL(AL_FORMAT_BFORMAT2D_16), DECL(AL_FORMAT_BFORMAT2D_FLOAT32), DECL(AL_FORMAT_BFORMAT2D_MULAW), DECL(AL_FORMAT_BFORMAT3D_8), DECL(AL_FORMAT_BFORMAT3D_16), DECL(AL_FORMAT_BFORMAT3D_FLOAT32), DECL(AL_FORMAT_BFORMAT3D_MULAW), DECL(AL_FREQUENCY), DECL(AL_BITS), DECL(AL_CHANNELS), DECL(AL_SIZE), DECL(AL_UNPACK_BLOCK_ALIGNMENT_SOFT), DECL(AL_PACK_BLOCK_ALIGNMENT_SOFT), DECL(AL_SOURCE_RADIUS), DECL(AL_STEREO_ANGLES), DECL(AL_UNUSED), DECL(AL_PENDING), DECL(AL_PROCESSED), DECL(AL_NO_ERROR), DECL(AL_INVALID_NAME), DECL(AL_INVALID_ENUM), DECL(AL_INVALID_VALUE), DECL(AL_INVALID_OPERATION), DECL(AL_OUT_OF_MEMORY), DECL(AL_VENDOR), DECL(AL_VERSION), DECL(AL_RENDERER), DECL(AL_EXTENSIONS), DECL(AL_DOPPLER_FACTOR), DECL(AL_DOPPLER_VELOCITY), DECL(AL_DISTANCE_MODEL), DECL(AL_SPEED_OF_SOUND), DECL(AL_SOURCE_DISTANCE_MODEL), DECL(AL_DEFERRED_UPDATES_SOFT), DECL(AL_GAIN_LIMIT_SOFT), DECL(AL_INVERSE_DISTANCE), DECL(AL_INVERSE_DISTANCE_CLAMPED), DECL(AL_LINEAR_DISTANCE), DECL(AL_LINEAR_DISTANCE_CLAMPED), DECL(AL_EXPONENT_DISTANCE), DECL(AL_EXPONENT_DISTANCE_CLAMPED), DECL(AL_FILTER_TYPE), DECL(AL_FILTER_NULL), DECL(AL_FILTER_LOWPASS), DECL(AL_FILTER_HIGHPASS), DECL(AL_FILTER_BANDPASS), DECL(AL_LOWPASS_GAIN), DECL(AL_LOWPASS_GAINHF), DECL(AL_HIGHPASS_GAIN), DECL(AL_HIGHPASS_GAINLF), DECL(AL_BANDPASS_GAIN), DECL(AL_BANDPASS_GAINHF), DECL(AL_BANDPASS_GAINLF), DECL(AL_EFFECT_TYPE), DECL(AL_EFFECT_NULL), DECL(AL_EFFECT_REVERB), DECL(AL_EFFECT_EAXREVERB), DECL(AL_EFFECT_CHORUS), DECL(AL_EFFECT_DISTORTION), DECL(AL_EFFECT_ECHO), DECL(AL_EFFECT_FLANGER), DECL(AL_EFFECT_PITCH_SHIFTER), DECL(AL_EFFECT_FREQUENCY_SHIFTER), #if 0 DECL(AL_EFFECT_VOCAL_MORPHER), #endif DECL(AL_EFFECT_RING_MODULATOR), DECL(AL_EFFECT_AUTOWAH), DECL(AL_EFFECT_COMPRESSOR), DECL(AL_EFFECT_EQUALIZER), DECL(AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT), DECL(AL_EFFECT_DEDICATED_DIALOGUE), DECL(AL_EFFECTSLOT_EFFECT), DECL(AL_EFFECTSLOT_GAIN), DECL(AL_EFFECTSLOT_AUXILIARY_SEND_AUTO), DECL(AL_EFFECTSLOT_NULL), DECL(AL_EAXREVERB_DENSITY), DECL(AL_EAXREVERB_DIFFUSION), DECL(AL_EAXREVERB_GAIN), DECL(AL_EAXREVERB_GAINHF), DECL(AL_EAXREVERB_GAINLF), DECL(AL_EAXREVERB_DECAY_TIME), DECL(AL_EAXREVERB_DECAY_HFRATIO), DECL(AL_EAXREVERB_DECAY_LFRATIO), DECL(AL_EAXREVERB_REFLECTIONS_GAIN), DECL(AL_EAXREVERB_REFLECTIONS_DELAY), DECL(AL_EAXREVERB_REFLECTIONS_PAN), DECL(AL_EAXREVERB_LATE_REVERB_GAIN), DECL(AL_EAXREVERB_LATE_REVERB_DELAY), DECL(AL_EAXREVERB_LATE_REVERB_PAN), DECL(AL_EAXREVERB_ECHO_TIME), DECL(AL_EAXREVERB_ECHO_DEPTH), DECL(AL_EAXREVERB_MODULATION_TIME), DECL(AL_EAXREVERB_MODULATION_DEPTH), DECL(AL_EAXREVERB_AIR_ABSORPTION_GAINHF), DECL(AL_EAXREVERB_HFREFERENCE), DECL(AL_EAXREVERB_LFREFERENCE), DECL(AL_EAXREVERB_ROOM_ROLLOFF_FACTOR), DECL(AL_EAXREVERB_DECAY_HFLIMIT), DECL(AL_REVERB_DENSITY), DECL(AL_REVERB_DIFFUSION), DECL(AL_REVERB_GAIN), DECL(AL_REVERB_GAINHF), DECL(AL_REVERB_DECAY_TIME), DECL(AL_REVERB_DECAY_HFRATIO), DECL(AL_REVERB_REFLECTIONS_GAIN), DECL(AL_REVERB_REFLECTIONS_DELAY), DECL(AL_REVERB_LATE_REVERB_GAIN), DECL(AL_REVERB_LATE_REVERB_DELAY), DECL(AL_REVERB_AIR_ABSORPTION_GAINHF), DECL(AL_REVERB_ROOM_ROLLOFF_FACTOR), DECL(AL_REVERB_DECAY_HFLIMIT), DECL(AL_CHORUS_WAVEFORM), DECL(AL_CHORUS_PHASE), DECL(AL_CHORUS_RATE), DECL(AL_CHORUS_DEPTH), DECL(AL_CHORUS_FEEDBACK), DECL(AL_CHORUS_DELAY), DECL(AL_DISTORTION_EDGE), DECL(AL_DISTORTION_GAIN), DECL(AL_DISTORTION_LOWPASS_CUTOFF), DECL(AL_DISTORTION_EQCENTER), DECL(AL_DISTORTION_EQBANDWIDTH), DECL(AL_ECHO_DELAY), DECL(AL_ECHO_LRDELAY), DECL(AL_ECHO_DAMPING), DECL(AL_ECHO_FEEDBACK), DECL(AL_ECHO_SPREAD), DECL(AL_FLANGER_WAVEFORM), DECL(AL_FLANGER_PHASE), DECL(AL_FLANGER_RATE), DECL(AL_FLANGER_DEPTH), DECL(AL_FLANGER_FEEDBACK), DECL(AL_FLANGER_DELAY), DECL(AL_FREQUENCY_SHIFTER_FREQUENCY), DECL(AL_FREQUENCY_SHIFTER_LEFT_DIRECTION), DECL(AL_FREQUENCY_SHIFTER_RIGHT_DIRECTION), DECL(AL_RING_MODULATOR_FREQUENCY), DECL(AL_RING_MODULATOR_HIGHPASS_CUTOFF), DECL(AL_RING_MODULATOR_WAVEFORM), DECL(AL_PITCH_SHIFTER_COARSE_TUNE), DECL(AL_PITCH_SHIFTER_FINE_TUNE), DECL(AL_COMPRESSOR_ONOFF), DECL(AL_EQUALIZER_LOW_GAIN), DECL(AL_EQUALIZER_LOW_CUTOFF), DECL(AL_EQUALIZER_MID1_GAIN), DECL(AL_EQUALIZER_MID1_CENTER), DECL(AL_EQUALIZER_MID1_WIDTH), DECL(AL_EQUALIZER_MID2_GAIN), DECL(AL_EQUALIZER_MID2_CENTER), DECL(AL_EQUALIZER_MID2_WIDTH), DECL(AL_EQUALIZER_HIGH_GAIN), DECL(AL_EQUALIZER_HIGH_CUTOFF), DECL(AL_DEDICATED_GAIN), DECL(AL_AUTOWAH_ATTACK_TIME), DECL(AL_AUTOWAH_RELEASE_TIME), DECL(AL_AUTOWAH_RESONANCE), DECL(AL_AUTOWAH_PEAK_GAIN), DECL(AL_NUM_RESAMPLERS_SOFT), DECL(AL_DEFAULT_RESAMPLER_SOFT), DECL(AL_SOURCE_RESAMPLER_SOFT), DECL(AL_RESAMPLER_NAME_SOFT), DECL(AL_SOURCE_SPATIALIZE_SOFT), DECL(AL_AUTO_SOFT), DECL(AL_MAP_READ_BIT_SOFT), DECL(AL_MAP_WRITE_BIT_SOFT), DECL(AL_MAP_PERSISTENT_BIT_SOFT), DECL(AL_PRESERVE_DATA_BIT_SOFT), DECL(AL_EVENT_CALLBACK_FUNCTION_SOFT), DECL(AL_EVENT_CALLBACK_USER_PARAM_SOFT), DECL(AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT), DECL(AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT), DECL(AL_EVENT_TYPE_ERROR_SOFT), DECL(AL_EVENT_TYPE_PERFORMANCE_SOFT), DECL(AL_EVENT_TYPE_DEPRECATED_SOFT), }; #undef DECL static const ALCchar alcNoError[] = "No Error"; static const ALCchar alcErrInvalidDevice[] = "Invalid Device"; static const ALCchar alcErrInvalidContext[] = "Invalid Context"; static const ALCchar alcErrInvalidEnum[] = "Invalid Enum"; static const ALCchar alcErrInvalidValue[] = "Invalid Value"; static const ALCchar alcErrOutOfMemory[] = "Out of Memory"; /************************************************ * Global variables ************************************************/ /* Enumerated device names */ static const ALCchar alcDefaultName[] = "OpenAL Soft\0"; static al_string alcAllDevicesList; static al_string alcCaptureDeviceList; /* Default is always the first in the list */ static ALCchar *alcDefaultAllDevicesSpecifier; static ALCchar *alcCaptureDefaultDeviceSpecifier; /* Default context extensions */ static const ALchar alExtList[] = "AL_EXT_ALAW " "AL_EXT_BFORMAT " "AL_EXT_DOUBLE " "AL_EXT_EXPONENT_DISTANCE " "AL_EXT_FLOAT32 " "AL_EXT_IMA4 " "AL_EXT_LINEAR_DISTANCE " "AL_EXT_MCFORMATS " "AL_EXT_MULAW " "AL_EXT_MULAW_BFORMAT " "AL_EXT_MULAW_MCFORMATS " "AL_EXT_OFFSET " "AL_EXT_source_distance_model " "AL_EXT_SOURCE_RADIUS " "AL_EXT_STEREO_ANGLES " "AL_LOKI_quadriphonic " "AL_SOFT_block_alignment " "AL_SOFT_deferred_updates " "AL_SOFT_direct_channels " "AL_SOFTX_events " "AL_SOFTX_filter_gain_ex " "AL_SOFT_gain_clamp_ex " "AL_SOFT_loop_points " "AL_SOFTX_map_buffer " "AL_SOFT_MSADPCM " "AL_SOFT_source_latency " "AL_SOFT_source_length " "AL_SOFT_source_resampler " "AL_SOFT_source_spatialize"; static ATOMIC(ALCenum) LastNullDeviceError = ATOMIC_INIT_STATIC(ALC_NO_ERROR); /* Thread-local current context */ static altss_t LocalContext; /* Process-wide current context */ static ATOMIC(ALCcontext*) GlobalContext = ATOMIC_INIT_STATIC(NULL); /* Mixing thread piority level */ ALint RTPrioLevel; FILE *LogFile; #ifdef _DEBUG enum LogLevel LogLevel = LogWarning; #else enum LogLevel LogLevel = LogError; #endif /* Flag to trap ALC device errors */ static ALCboolean TrapALCError = ALC_FALSE; /* One-time configuration init control */ static alonce_flag alc_config_once = AL_ONCE_FLAG_INIT; /* Default effect that applies to sources that don't have an effect on send 0 */ static ALeffect DefaultEffect; /* Flag to specify if alcSuspendContext/alcProcessContext should defer/process * updates. */ static ALCboolean SuspendDefers = ALC_TRUE; /************************************************ * ALC information ************************************************/ static const ALCchar alcNoDeviceExtList[] = "ALC_ENUMERATE_ALL_EXT ALC_ENUMERATION_EXT ALC_EXT_CAPTURE " "ALC_EXT_thread_local_context ALC_SOFT_loopback"; static const ALCchar alcExtensionList[] = "ALC_ENUMERATE_ALL_EXT ALC_ENUMERATION_EXT ALC_EXT_CAPTURE " "ALC_EXT_DEDICATED ALC_EXT_disconnect ALC_EXT_EFX " "ALC_EXT_thread_local_context ALC_SOFT_device_clock ALC_SOFT_HRTF " "ALC_SOFT_loopback ALC_SOFT_output_limiter ALC_SOFT_pause_device"; static const ALCint alcMajorVersion = 1; static const ALCint alcMinorVersion = 1; static const ALCint alcEFXMajorVersion = 1; static const ALCint alcEFXMinorVersion = 0; /************************************************ * Device lists ************************************************/ static ATOMIC(ALCdevice*) DeviceList = ATOMIC_INIT_STATIC(NULL); static almtx_t ListLock; static inline void LockLists(void) { int ret = almtx_lock(&ListLock); assert(ret == althrd_success); } static inline void UnlockLists(void) { int ret = almtx_unlock(&ListLock); assert(ret == althrd_success); } /************************************************ * Library initialization ************************************************/ #if defined(_WIN32) static void alc_init(void); static void alc_deinit(void); static void alc_deinit_safe(void); #ifndef AL_LIBTYPE_STATIC BOOL APIENTRY DllMain(HINSTANCE hModule, DWORD reason, LPVOID lpReserved) { switch(reason) { case DLL_PROCESS_ATTACH: /* Pin the DLL so we won't get unloaded until the process terminates */ GetModuleHandleExW(GET_MODULE_HANDLE_EX_FLAG_PIN | GET_MODULE_HANDLE_EX_FLAG_FROM_ADDRESS, (WCHAR*)hModule, &hModule); alc_init(); break; case DLL_THREAD_DETACH: althrd_thread_detach(); break; case DLL_PROCESS_DETACH: if(!lpReserved) alc_deinit(); else alc_deinit_safe(); break; } return TRUE; } #elif defined(_MSC_VER) #pragma section(".CRT$XCU",read) static void alc_constructor(void); static void alc_destructor(void); __declspec(allocate(".CRT$XCU")) void (__cdecl* alc_constructor_)(void) = alc_constructor; static void alc_constructor(void) { atexit(alc_destructor); alc_init(); } static void alc_destructor(void) { alc_deinit(); } #elif defined(HAVE_GCC_DESTRUCTOR) static void alc_init(void) __attribute__((constructor)); static void alc_deinit(void) __attribute__((destructor)); #else #error "No static initialization available on this platform!" #endif #elif defined(HAVE_GCC_DESTRUCTOR) static void alc_init(void) __attribute__((constructor)); static void alc_deinit(void) __attribute__((destructor)); #else #error "No global initialization available on this platform!" #endif static void ReleaseThreadCtx(void *ptr); static void alc_init(void) { const char *str; int ret; LogFile = stderr; AL_STRING_INIT(alcAllDevicesList); AL_STRING_INIT(alcCaptureDeviceList); str = getenv("__ALSOFT_HALF_ANGLE_CONES"); if(str && (strcasecmp(str, "true") == 0 || strtol(str, NULL, 0) == 1)) ConeScale *= 0.5f; str = getenv("__ALSOFT_REVERSE_Z"); if(str && (strcasecmp(str, "true") == 0 || strtol(str, NULL, 0) == 1)) ZScale *= -1.0f; str = getenv("__ALSOFT_REVERB_IGNORES_SOUND_SPEED"); if(str && (strcasecmp(str, "true") == 0 || strtol(str, NULL, 0) == 1)) OverrideReverbSpeedOfSound = AL_TRUE; ret = altss_create(&LocalContext, ReleaseThreadCtx); assert(ret == althrd_success); ret = almtx_init(&ListLock, almtx_recursive); assert(ret == althrd_success); } static void alc_initconfig(void) { const char *devs, *str; int capfilter; float valf; int i, n; str = getenv("ALSOFT_LOGLEVEL"); if(str) { long lvl = strtol(str, NULL, 0); if(lvl >= NoLog && lvl <= LogRef) LogLevel = lvl; } str = getenv("ALSOFT_LOGFILE"); if(str && str[0]) { FILE *logfile = al_fopen(str, "wt"); if(logfile) LogFile = logfile; else ERR("Failed to open log file '%s'\n", str); } TRACE("Initializing library v%s-%s %s\n", ALSOFT_VERSION, ALSOFT_GIT_COMMIT_HASH, ALSOFT_GIT_BRANCH); { char buf[1024] = ""; int len = 0; if(BackendListSize > 0) len += snprintf(buf, sizeof(buf), "%s", BackendList[0].name); for(i = 1;i < BackendListSize;i++) len += snprintf(buf+len, sizeof(buf)-len, ", %s", BackendList[i].name); TRACE("Supported backends: %s\n", buf); } ReadALConfig(); str = getenv("__ALSOFT_SUSPEND_CONTEXT"); if(str && *str) { if(strcasecmp(str, "ignore") == 0) { SuspendDefers = ALC_FALSE; TRACE("Selected context suspend behavior, \"ignore\"\n"); } else ERR("Unhandled context suspend behavior setting: \"%s\"\n", str); } capfilter = 0; #if defined(HAVE_SSE4_1) capfilter |= CPU_CAP_SSE | CPU_CAP_SSE2 | CPU_CAP_SSE3 | CPU_CAP_SSE4_1; #elif defined(HAVE_SSE3) capfilter |= CPU_CAP_SSE | CPU_CAP_SSE2 | CPU_CAP_SSE3; #elif defined(HAVE_SSE2) capfilter |= CPU_CAP_SSE | CPU_CAP_SSE2; #elif defined(HAVE_SSE) capfilter |= CPU_CAP_SSE; #endif #ifdef HAVE_NEON capfilter |= CPU_CAP_NEON; #endif if(ConfigValueStr(NULL, NULL, "disable-cpu-exts", &str)) { if(strcasecmp(str, "all") == 0) capfilter = 0; else { size_t len; const char *next = str; do { str = next; while(isspace(str[0])) str++; next = strchr(str, ','); if(!str[0] || str[0] == ',') continue; len = (next ? ((size_t)(next-str)) : strlen(str)); while(len > 0 && isspace(str[len-1])) len--; if(len == 3 && strncasecmp(str, "sse", len) == 0) capfilter &= ~CPU_CAP_SSE; else if(len == 4 && strncasecmp(str, "sse2", len) == 0) capfilter &= ~CPU_CAP_SSE2; else if(len == 4 && strncasecmp(str, "sse3", len) == 0) capfilter &= ~CPU_CAP_SSE3; else if(len == 6 && strncasecmp(str, "sse4.1", len) == 0) capfilter &= ~CPU_CAP_SSE4_1; else if(len == 4 && strncasecmp(str, "neon", len) == 0) capfilter &= ~CPU_CAP_NEON; else WARN("Invalid CPU extension \"%s\"\n", str); } while(next++); } } FillCPUCaps(capfilter); #ifdef _WIN32 RTPrioLevel = 1; #else RTPrioLevel = 0; #endif ConfigValueInt(NULL, NULL, "rt-prio", &RTPrioLevel); aluInit(); aluInitMixer(); str = getenv("ALSOFT_TRAP_ERROR"); if(str && (strcasecmp(str, "true") == 0 || strtol(str, NULL, 0) == 1)) { TrapALError = AL_TRUE; TrapALCError = AL_TRUE; } else { str = getenv("ALSOFT_TRAP_AL_ERROR"); if(str && (strcasecmp(str, "true") == 0 || strtol(str, NULL, 0) == 1)) TrapALError = AL_TRUE; TrapALError = GetConfigValueBool(NULL, NULL, "trap-al-error", TrapALError); str = getenv("ALSOFT_TRAP_ALC_ERROR"); if(str && (strcasecmp(str, "true") == 0 || strtol(str, NULL, 0) == 1)) TrapALCError = ALC_TRUE; TrapALCError = GetConfigValueBool(NULL, NULL, "trap-alc-error", TrapALCError); } if(ConfigValueFloat(NULL, "reverb", "boost", &valf)) ReverbBoost *= powf(10.0f, valf / 20.0f); if(((devs=getenv("ALSOFT_DRIVERS")) && devs[0]) || ConfigValueStr(NULL, NULL, "drivers", &devs)) { int n; size_t len; const char *next = devs; int endlist, delitem; i = 0; do { devs = next; while(isspace(devs[0])) devs++; next = strchr(devs, ','); delitem = (devs[0] == '-'); if(devs[0] == '-') devs++; if(!devs[0] || devs[0] == ',') { endlist = 0; continue; } endlist = 1; len = (next ? ((size_t)(next-devs)) : strlen(devs)); while(len > 0 && isspace(devs[len-1])) len--; #ifdef HAVE_WASAPI /* HACK: For backwards compatibility, convert backend references of * mmdevapi to wasapi. This should eventually be removed. */ if(len == 8 && strncmp(devs, "mmdevapi", len) == 0) { devs = "wasapi"; len = 6; } #endif for(n = i;n < BackendListSize;n++) { if(len == strlen(BackendList[n].name) && strncmp(BackendList[n].name, devs, len) == 0) { if(delitem) { for(;n+1 < BackendListSize;n++) BackendList[n] = BackendList[n+1]; BackendListSize--; } else { struct BackendInfo Bkp = BackendList[n]; for(;n > i;n--) BackendList[n] = BackendList[n-1]; BackendList[n] = Bkp; i++; } break; } } } while(next++); if(endlist) BackendListSize = i; } for(n = i = 0;i < BackendListSize && (!PlaybackBackend.name || !CaptureBackend.name);i++) { ALCbackendFactory *factory; BackendList[n] = BackendList[i]; factory = BackendList[n].getFactory(); if(!V0(factory,init)()) { WARN("Failed to initialize backend \"%s\"\n", BackendList[n].name); continue; } TRACE("Initialized backend \"%s\"\n", BackendList[n].name); if(!PlaybackBackend.name && V(factory,querySupport)(ALCbackend_Playback)) { PlaybackBackend = BackendList[n]; TRACE("Added \"%s\" for playback\n", PlaybackBackend.name); } if(!CaptureBackend.name && V(factory,querySupport)(ALCbackend_Capture)) { CaptureBackend = BackendList[n]; TRACE("Added \"%s\" for capture\n", CaptureBackend.name); } n++; } BackendListSize = n; { ALCbackendFactory *factory = ALCloopbackFactory_getFactory(); V0(factory,init)(); } if(!PlaybackBackend.name) WARN("No playback backend available!\n"); if(!CaptureBackend.name) WARN("No capture backend available!\n"); if(ConfigValueStr(NULL, NULL, "excludefx", &str)) { size_t len; const char *next = str; do { str = next; next = strchr(str, ','); if(!str[0] || next == str) continue; len = (next ? ((size_t)(next-str)) : strlen(str)); for(n = 0;n < EFFECTLIST_SIZE;n++) { if(len == strlen(EffectList[n].name) && strncmp(EffectList[n].name, str, len) == 0) DisabledEffects[EffectList[n].type] = AL_TRUE; } } while(next++); } InitEffect(&DefaultEffect); str = getenv("ALSOFT_DEFAULT_REVERB"); if((str && str[0]) || ConfigValueStr(NULL, NULL, "default-reverb", &str)) LoadReverbPreset(str, &DefaultEffect); } #define DO_INITCONFIG() alcall_once(&alc_config_once, alc_initconfig) /************************************************ * Library deinitialization ************************************************/ static void alc_cleanup(void) { ALCdevice *dev; AL_STRING_DEINIT(alcAllDevicesList); AL_STRING_DEINIT(alcCaptureDeviceList); free(alcDefaultAllDevicesSpecifier); alcDefaultAllDevicesSpecifier = NULL; free(alcCaptureDefaultDeviceSpecifier); alcCaptureDefaultDeviceSpecifier = NULL; if((dev=ATOMIC_EXCHANGE_PTR_SEQ(&DeviceList, NULL)) != NULL) { ALCuint num = 0; do { num++; dev = ATOMIC_LOAD(&dev->next, almemory_order_relaxed); } while(dev != NULL); ERR("%u device%s not closed\n", num, (num>1)?"s":""); } } static void alc_deinit_safe(void) { alc_cleanup(); FreeHrtfs(); FreeALConfig(); almtx_destroy(&ListLock); altss_delete(LocalContext); if(LogFile != stderr) fclose(LogFile); LogFile = NULL; althrd_deinit(); } static void alc_deinit(void) { int i; alc_cleanup(); memset(&PlaybackBackend, 0, sizeof(PlaybackBackend)); memset(&CaptureBackend, 0, sizeof(CaptureBackend)); for(i = 0;i < BackendListSize;i++) { ALCbackendFactory *factory = BackendList[i].getFactory(); V0(factory,deinit)(); } { ALCbackendFactory *factory = ALCloopbackFactory_getFactory(); V0(factory,deinit)(); } alc_deinit_safe(); } /************************************************ * Device enumeration ************************************************/ static void ProbeDevices(al_string *list, struct BackendInfo *backendinfo, enum DevProbe type) { DO_INITCONFIG(); LockLists(); alstr_clear(list); if(backendinfo->getFactory) { ALCbackendFactory *factory = backendinfo->getFactory(); V(factory,probe)(type, list); } UnlockLists(); } static void ProbeAllDevicesList(void) { ProbeDevices(&alcAllDevicesList, &PlaybackBackend, ALL_DEVICE_PROBE); } static void ProbeCaptureDeviceList(void) { ProbeDevices(&alcCaptureDeviceList, &CaptureBackend, CAPTURE_DEVICE_PROBE); } /************************************************ * Device format information ************************************************/ const ALCchar *DevFmtTypeString(enum DevFmtType type) { switch(type) { case DevFmtByte: return "Signed Byte"; case DevFmtUByte: return "Unsigned Byte"; case DevFmtShort: return "Signed Short"; case DevFmtUShort: return "Unsigned Short"; case DevFmtInt: return "Signed Int"; case DevFmtUInt: return "Unsigned Int"; case DevFmtFloat: return "Float"; } return "(unknown type)"; } const ALCchar *DevFmtChannelsString(enum DevFmtChannels chans) { switch(chans) { case DevFmtMono: return "Mono"; case DevFmtStereo: return "Stereo"; case DevFmtQuad: return "Quadraphonic"; case DevFmtX51: return "5.1 Surround"; case DevFmtX51Rear: return "5.1 Surround (Rear)"; case DevFmtX61: return "6.1 Surround"; case DevFmtX71: return "7.1 Surround"; case DevFmtAmbi3D: return "Ambisonic 3D"; } return "(unknown channels)"; } extern inline ALsizei FrameSizeFromDevFmt(enum DevFmtChannels chans, enum DevFmtType type, ALsizei ambiorder); ALsizei BytesFromDevFmt(enum DevFmtType type) { switch(type) { case DevFmtByte: return sizeof(ALbyte); case DevFmtUByte: return sizeof(ALubyte); case DevFmtShort: return sizeof(ALshort); case DevFmtUShort: return sizeof(ALushort); case DevFmtInt: return sizeof(ALint); case DevFmtUInt: return sizeof(ALuint); case DevFmtFloat: return sizeof(ALfloat); } return 0; } ALsizei ChannelsFromDevFmt(enum DevFmtChannels chans, ALsizei ambiorder) { switch(chans) { case DevFmtMono: return 1; case DevFmtStereo: return 2; case DevFmtQuad: return 4; case DevFmtX51: return 6; case DevFmtX51Rear: return 6; case DevFmtX61: return 7; case DevFmtX71: return 8; case DevFmtAmbi3D: return (ambiorder >= 3) ? 16 : (ambiorder == 2) ? 9 : (ambiorder == 1) ? 4 : 1; } return 0; } static ALboolean DecomposeDevFormat(ALenum format, enum DevFmtChannels *chans, enum DevFmtType *type) { static const struct { ALenum format; enum DevFmtChannels channels; enum DevFmtType type; } list[] = { { AL_FORMAT_MONO8, DevFmtMono, DevFmtUByte }, { AL_FORMAT_MONO16, DevFmtMono, DevFmtShort }, { AL_FORMAT_MONO_FLOAT32, DevFmtMono, DevFmtFloat }, { AL_FORMAT_STEREO8, DevFmtStereo, DevFmtUByte }, { AL_FORMAT_STEREO16, DevFmtStereo, DevFmtShort }, { AL_FORMAT_STEREO_FLOAT32, DevFmtStereo, DevFmtFloat }, { AL_FORMAT_QUAD8, DevFmtQuad, DevFmtUByte }, { AL_FORMAT_QUAD16, DevFmtQuad, DevFmtShort }, { AL_FORMAT_QUAD32, DevFmtQuad, DevFmtFloat }, { AL_FORMAT_51CHN8, DevFmtX51, DevFmtUByte }, { AL_FORMAT_51CHN16, DevFmtX51, DevFmtShort }, { AL_FORMAT_51CHN32, DevFmtX51, DevFmtFloat }, { AL_FORMAT_61CHN8, DevFmtX61, DevFmtUByte }, { AL_FORMAT_61CHN16, DevFmtX61, DevFmtShort }, { AL_FORMAT_61CHN32, DevFmtX61, DevFmtFloat }, { AL_FORMAT_71CHN8, DevFmtX71, DevFmtUByte }, { AL_FORMAT_71CHN16, DevFmtX71, DevFmtShort }, { AL_FORMAT_71CHN32, DevFmtX71, DevFmtFloat }, }; ALuint i; for(i = 0;i < COUNTOF(list);i++) { if(list[i].format == format) { *chans = list[i].channels; *type = list[i].type; return AL_TRUE; } } return AL_FALSE; } static ALCboolean IsValidALCType(ALCenum type) { switch(type) { case ALC_BYTE_SOFT: case ALC_UNSIGNED_BYTE_SOFT: case ALC_SHORT_SOFT: case ALC_UNSIGNED_SHORT_SOFT: case ALC_INT_SOFT: case ALC_UNSIGNED_INT_SOFT: case ALC_FLOAT_SOFT: return ALC_TRUE; } return ALC_FALSE; } static ALCboolean IsValidALCChannels(ALCenum channels) { switch(channels) { case ALC_MONO_SOFT: case ALC_STEREO_SOFT: case ALC_QUAD_SOFT: case ALC_5POINT1_SOFT: case ALC_6POINT1_SOFT: case ALC_7POINT1_SOFT: case ALC_BFORMAT3D_SOFT: return ALC_TRUE; } return ALC_FALSE; } static ALCboolean IsValidAmbiLayout(ALCenum layout) { switch(layout) { case ALC_ACN_SOFT: case ALC_FUMA_SOFT: return ALC_TRUE; } return ALC_FALSE; } static ALCboolean IsValidAmbiScaling(ALCenum scaling) { switch(scaling) { case ALC_N3D_SOFT: case ALC_SN3D_SOFT: case ALC_FUMA_SOFT: return ALC_TRUE; } return ALC_FALSE; } /************************************************ * Miscellaneous ALC helpers ************************************************/ /* SetDefaultWFXChannelOrder * * Sets the default channel order used by WaveFormatEx. */ void SetDefaultWFXChannelOrder(ALCdevice *device) { ALsizei i; for(i = 0;i < MAX_OUTPUT_CHANNELS;i++) device->RealOut.ChannelName[i] = InvalidChannel; switch(device->FmtChans) { case DevFmtMono: device->RealOut.ChannelName[0] = FrontCenter; break; case DevFmtStereo: device->RealOut.ChannelName[0] = FrontLeft; device->RealOut.ChannelName[1] = FrontRight; break; case DevFmtQuad: device->RealOut.ChannelName[0] = FrontLeft; device->RealOut.ChannelName[1] = FrontRight; device->RealOut.ChannelName[2] = BackLeft; device->RealOut.ChannelName[3] = BackRight; break; case DevFmtX51: device->RealOut.ChannelName[0] = FrontLeft; device->RealOut.ChannelName[1] = FrontRight; device->RealOut.ChannelName[2] = FrontCenter; device->RealOut.ChannelName[3] = LFE; device->RealOut.ChannelName[4] = SideLeft; device->RealOut.ChannelName[5] = SideRight; break; case DevFmtX51Rear: device->RealOut.ChannelName[0] = FrontLeft; device->RealOut.ChannelName[1] = FrontRight; device->RealOut.ChannelName[2] = FrontCenter; device->RealOut.ChannelName[3] = LFE; device->RealOut.ChannelName[4] = BackLeft; device->RealOut.ChannelName[5] = BackRight; break; case DevFmtX61: device->RealOut.ChannelName[0] = FrontLeft; device->RealOut.ChannelName[1] = FrontRight; device->RealOut.ChannelName[2] = FrontCenter; device->RealOut.ChannelName[3] = LFE; device->RealOut.ChannelName[4] = BackCenter; device->RealOut.ChannelName[5] = SideLeft; device->RealOut.ChannelName[6] = SideRight; break; case DevFmtX71: device->RealOut.ChannelName[0] = FrontLeft; device->RealOut.ChannelName[1] = FrontRight; device->RealOut.ChannelName[2] = FrontCenter; device->RealOut.ChannelName[3] = LFE; device->RealOut.ChannelName[4] = BackLeft; device->RealOut.ChannelName[5] = BackRight; device->RealOut.ChannelName[6] = SideLeft; device->RealOut.ChannelName[7] = SideRight; break; case DevFmtAmbi3D: device->RealOut.ChannelName[0] = Aux0; if(device->AmbiOrder > 0) { device->RealOut.ChannelName[1] = Aux1; device->RealOut.ChannelName[2] = Aux2; device->RealOut.ChannelName[3] = Aux3; } if(device->AmbiOrder > 1) { device->RealOut.ChannelName[4] = Aux4; device->RealOut.ChannelName[5] = Aux5; device->RealOut.ChannelName[6] = Aux6; device->RealOut.ChannelName[7] = Aux7; device->RealOut.ChannelName[8] = Aux8; } if(device->AmbiOrder > 2) { device->RealOut.ChannelName[9] = Aux9; device->RealOut.ChannelName[10] = Aux10; device->RealOut.ChannelName[11] = Aux11; device->RealOut.ChannelName[12] = Aux12; device->RealOut.ChannelName[13] = Aux13; device->RealOut.ChannelName[14] = Aux14; device->RealOut.ChannelName[15] = Aux15; } break; } } /* SetDefaultChannelOrder * * Sets the default channel order used by most non-WaveFormatEx-based APIs. */ void SetDefaultChannelOrder(ALCdevice *device) { ALsizei i; for(i = 0;i < MAX_OUTPUT_CHANNELS;i++) device->RealOut.ChannelName[i] = InvalidChannel; switch(device->FmtChans) { case DevFmtX51Rear: device->RealOut.ChannelName[0] = FrontLeft; device->RealOut.ChannelName[1] = FrontRight; device->RealOut.ChannelName[2] = BackLeft; device->RealOut.ChannelName[3] = BackRight; device->RealOut.ChannelName[4] = FrontCenter; device->RealOut.ChannelName[5] = LFE; return; case DevFmtX71: device->RealOut.ChannelName[0] = FrontLeft; device->RealOut.ChannelName[1] = FrontRight; device->RealOut.ChannelName[2] = BackLeft; device->RealOut.ChannelName[3] = BackRight; device->RealOut.ChannelName[4] = FrontCenter; device->RealOut.ChannelName[5] = LFE; device->RealOut.ChannelName[6] = SideLeft; device->RealOut.ChannelName[7] = SideRight; return; /* Same as WFX order */ case DevFmtMono: case DevFmtStereo: case DevFmtQuad: case DevFmtX51: case DevFmtX61: case DevFmtAmbi3D: SetDefaultWFXChannelOrder(device); break; } } extern inline ALint GetChannelIndex(const enum Channel names[MAX_OUTPUT_CHANNELS], enum Channel chan); extern inline ALint GetChannelIdxByName(const RealMixParams *real, enum Channel chan); /* ALCcontext_DeferUpdates * * Defers/suspends updates for the given context's listener and sources. This * does *NOT* stop mixing, but rather prevents certain property changes from * taking effect. */ void ALCcontext_DeferUpdates(ALCcontext *context) { ATOMIC_STORE_SEQ(&context->DeferUpdates, AL_TRUE); } /* ALCcontext_ProcessUpdates * * Resumes update processing after being deferred. */ void ALCcontext_ProcessUpdates(ALCcontext *context) { almtx_lock(&context->PropLock); if(ATOMIC_EXCHANGE_SEQ(&context->DeferUpdates, AL_FALSE)) { /* Tell the mixer to stop applying updates, then wait for any active * updating to finish, before providing updates. */ ATOMIC_STORE_SEQ(&context->HoldUpdates, AL_TRUE); while((ATOMIC_LOAD(&context->UpdateCount, almemory_order_acquire)&1) != 0) althrd_yield(); if(!ATOMIC_FLAG_TEST_AND_SET(&context->PropsClean, almemory_order_acq_rel)) UpdateContextProps(context); if(!ATOMIC_FLAG_TEST_AND_SET(&context->Listener->PropsClean, almemory_order_acq_rel)) UpdateListenerProps(context); UpdateAllEffectSlotProps(context); UpdateAllSourceProps(context); /* Now with all updates declared, let the mixer continue applying them * so they all happen at once. */ ATOMIC_STORE_SEQ(&context->HoldUpdates, AL_FALSE); } almtx_unlock(&context->PropLock); } /* alcSetError * * Stores the latest ALC device error */ static void alcSetError(ALCdevice *device, ALCenum errorCode) { WARN("Error generated on device %p, code 0x%04x\n", device, errorCode); if(TrapALCError) { #ifdef _WIN32 /* DebugBreak() will cause an exception if there is no debugger */ if(IsDebuggerPresent()) DebugBreak(); #elif defined(SIGTRAP) raise(SIGTRAP); #endif } if(device) ATOMIC_STORE_SEQ(&device->LastError, errorCode); else ATOMIC_STORE_SEQ(&LastNullDeviceError, errorCode); } static struct Compressor *CreateDeviceLimiter(const ALCdevice *device, const ALfloat threshold) { return CompressorInit(device->RealOut.NumChannels, device->Frequency, AL_TRUE, AL_TRUE, AL_TRUE, AL_TRUE, AL_TRUE, 0.001f, 0.002f, 0.0f, 0.0f, threshold, INFINITY, 0.0f, 0.020f, 0.200f); } /* UpdateClockBase * * Updates the device's base clock time with however many samples have been * done. This is used so frequency changes on the device don't cause the time * to jump forward or back. Must not be called while the device is running/ * mixing. */ static inline void UpdateClockBase(ALCdevice *device) { IncrementRef(&device->MixCount); device->ClockBase += device->SamplesDone * DEVICE_CLOCK_RES / device->Frequency; device->SamplesDone = 0; IncrementRef(&device->MixCount); } /* UpdateDeviceParams * * Updates device parameters according to the attribute list (caller is * responsible for holding the list lock). */ static ALCenum UpdateDeviceParams(ALCdevice *device, const ALCint *attrList) { enum HrtfRequestMode hrtf_userreq = Hrtf_Default; enum HrtfRequestMode hrtf_appreq = Hrtf_Default; ALCenum gainLimiter = device->LimiterState; const ALsizei old_sends = device->NumAuxSends; ALsizei new_sends = device->NumAuxSends; enum DevFmtChannels oldChans; enum DevFmtType oldType; ALboolean update_failed; ALCsizei hrtf_id = -1; ALCcontext *context; ALCuint oldFreq; size_t size; ALCsizei i; int val; // Check for attributes if(device->Type == Loopback) { ALCsizei numMono, numStereo, numSends; ALCenum alayout = AL_NONE; ALCenum ascale = AL_NONE; ALCenum schans = AL_NONE; ALCenum stype = AL_NONE; ALCsizei attrIdx = 0; ALCsizei aorder = 0; ALCuint freq = 0; if(!attrList) { WARN("Missing attributes for loopback device\n"); return ALC_INVALID_VALUE; } numMono = device->NumMonoSources; numStereo = device->NumStereoSources; numSends = old_sends; #define TRACE_ATTR(a, v) TRACE("Loopback %s = %d\n", #a, v) while(attrList[attrIdx]) { switch(attrList[attrIdx]) { case ALC_FORMAT_CHANNELS_SOFT: schans = attrList[attrIdx + 1]; TRACE_ATTR(ALC_FORMAT_CHANNELS_SOFT, schans); if(!IsValidALCChannels(schans)) return ALC_INVALID_VALUE; break; case ALC_FORMAT_TYPE_SOFT: stype = attrList[attrIdx + 1]; TRACE_ATTR(ALC_FORMAT_TYPE_SOFT, stype); if(!IsValidALCType(stype)) return ALC_INVALID_VALUE; break; case ALC_FREQUENCY: freq = attrList[attrIdx + 1]; TRACE_ATTR(ALC_FREQUENCY, freq); if(freq < MIN_OUTPUT_RATE) return ALC_INVALID_VALUE; break; case ALC_AMBISONIC_LAYOUT_SOFT: alayout = attrList[attrIdx + 1]; TRACE_ATTR(ALC_AMBISONIC_LAYOUT_SOFT, alayout); if(!IsValidAmbiLayout(alayout)) return ALC_INVALID_VALUE; break; case ALC_AMBISONIC_SCALING_SOFT: ascale = attrList[attrIdx + 1]; TRACE_ATTR(ALC_AMBISONIC_SCALING_SOFT, ascale); if(!IsValidAmbiScaling(ascale)) return ALC_INVALID_VALUE; break; case ALC_AMBISONIC_ORDER_SOFT: aorder = attrList[attrIdx + 1]; TRACE_ATTR(ALC_AMBISONIC_ORDER_SOFT, aorder); if(aorder < 1 || aorder > MAX_AMBI_ORDER) return ALC_INVALID_VALUE; break; case ALC_MONO_SOURCES: numMono = attrList[attrIdx + 1]; TRACE_ATTR(ALC_MONO_SOURCES, numMono); numMono = maxi(numMono, 0); break; case ALC_STEREO_SOURCES: numStereo = attrList[attrIdx + 1]; TRACE_ATTR(ALC_STEREO_SOURCES, numStereo); numStereo = maxi(numStereo, 0); break; case ALC_MAX_AUXILIARY_SENDS: numSends = attrList[attrIdx + 1]; TRACE_ATTR(ALC_MAX_AUXILIARY_SENDS, numSends); numSends = clampi(numSends, 0, MAX_SENDS); break; case ALC_HRTF_SOFT: TRACE_ATTR(ALC_HRTF_SOFT, attrList[attrIdx + 1]); if(attrList[attrIdx + 1] == ALC_FALSE) hrtf_appreq = Hrtf_Disable; else if(attrList[attrIdx + 1] == ALC_TRUE) hrtf_appreq = Hrtf_Enable; else hrtf_appreq = Hrtf_Default; break; case ALC_HRTF_ID_SOFT: hrtf_id = attrList[attrIdx + 1]; TRACE_ATTR(ALC_HRTF_ID_SOFT, hrtf_id); break; case ALC_OUTPUT_LIMITER_SOFT: gainLimiter = attrList[attrIdx + 1]; TRACE_ATTR(ALC_OUTPUT_LIMITER_SOFT, gainLimiter); break; default: TRACE("Loopback 0x%04X = %d (0x%x)\n", attrList[attrIdx], attrList[attrIdx + 1], attrList[attrIdx + 1]); break; } attrIdx += 2; } #undef TRACE_ATTR if(!schans || !stype || !freq) { WARN("Missing format for loopback device\n"); return ALC_INVALID_VALUE; } if(schans == ALC_BFORMAT3D_SOFT && (!alayout || !ascale || !aorder)) { WARN("Missing ambisonic info for loopback device\n"); return ALC_INVALID_VALUE; } if((device->Flags&DEVICE_RUNNING)) V0(device->Backend,stop)(); device->Flags &= ~DEVICE_RUNNING; UpdateClockBase(device); device->Frequency = freq; device->FmtChans = schans; device->FmtType = stype; if(schans == ALC_BFORMAT3D_SOFT) { device->AmbiOrder = aorder; device->AmbiLayout = alayout; device->AmbiScale = ascale; } if(numMono > INT_MAX-numStereo) numMono = INT_MAX-numStereo; numMono += numStereo; if(ConfigValueInt(NULL, NULL, "sources", &numMono)) { if(numMono <= 0) numMono = 256; } else numMono = maxi(numMono, 256); numStereo = mini(numStereo, numMono); numMono -= numStereo; device->SourcesMax = numMono + numStereo; device->NumMonoSources = numMono; device->NumStereoSources = numStereo; if(ConfigValueInt(NULL, NULL, "sends", &new_sends)) new_sends = mini(numSends, clampi(new_sends, 0, MAX_SENDS)); else new_sends = numSends; } else if(attrList && attrList[0]) { ALCsizei numMono, numStereo, numSends; ALCsizei attrIdx = 0; ALCuint freq; /* If a context is already running on the device, stop playback so the * device attributes can be updated. */ if((device->Flags&DEVICE_RUNNING)) V0(device->Backend,stop)(); device->Flags &= ~DEVICE_RUNNING; UpdateClockBase(device); freq = device->Frequency; numMono = device->NumMonoSources; numStereo = device->NumStereoSources; numSends = old_sends; #define TRACE_ATTR(a, v) TRACE("%s = %d\n", #a, v) while(attrList[attrIdx]) { switch(attrList[attrIdx]) { case ALC_FREQUENCY: freq = attrList[attrIdx + 1]; TRACE_ATTR(ALC_FREQUENCY, freq); device->Flags |= DEVICE_FREQUENCY_REQUEST; break; case ALC_MONO_SOURCES: numMono = attrList[attrIdx + 1]; TRACE_ATTR(ALC_MONO_SOURCES, numMono); numMono = maxi(numMono, 0); break; case ALC_STEREO_SOURCES: numStereo = attrList[attrIdx + 1]; TRACE_ATTR(ALC_STEREO_SOURCES, numStereo); numStereo = maxi(numStereo, 0); break; case ALC_MAX_AUXILIARY_SENDS: numSends = attrList[attrIdx + 1]; TRACE_ATTR(ALC_MAX_AUXILIARY_SENDS, numSends); numSends = clampi(numSends, 0, MAX_SENDS); break; case ALC_HRTF_SOFT: TRACE_ATTR(ALC_HRTF_SOFT, attrList[attrIdx + 1]); if(attrList[attrIdx + 1] == ALC_FALSE) hrtf_appreq = Hrtf_Disable; else if(attrList[attrIdx + 1] == ALC_TRUE) hrtf_appreq = Hrtf_Enable; else hrtf_appreq = Hrtf_Default; break; case ALC_HRTF_ID_SOFT: hrtf_id = attrList[attrIdx + 1]; TRACE_ATTR(ALC_HRTF_ID_SOFT, hrtf_id); break; case ALC_OUTPUT_LIMITER_SOFT: gainLimiter = attrList[attrIdx + 1]; TRACE_ATTR(ALC_OUTPUT_LIMITER_SOFT, gainLimiter); break; default: TRACE("0x%04X = %d (0x%x)\n", attrList[attrIdx], attrList[attrIdx + 1], attrList[attrIdx + 1]); break; } attrIdx += 2; } #undef TRACE_ATTR ConfigValueUInt(alstr_get_cstr(device->DeviceName), NULL, "frequency", &freq); freq = maxu(freq, MIN_OUTPUT_RATE); device->UpdateSize = (ALuint64)device->UpdateSize * freq / device->Frequency; /* SSE and Neon do best with the update size being a multiple of 4 */ if((CPUCapFlags&(CPU_CAP_SSE|CPU_CAP_NEON)) != 0) device->UpdateSize = (device->UpdateSize+3)&~3; device->Frequency = freq; if(numMono > INT_MAX-numStereo) numMono = INT_MAX-numStereo; numMono += numStereo; if(ConfigValueInt(alstr_get_cstr(device->DeviceName), NULL, "sources", &numMono)) { if(numMono <= 0) numMono = 256; } else numMono = maxi(numMono, 256); numStereo = mini(numStereo, numMono); numMono -= numStereo; device->SourcesMax = numMono + numStereo; device->NumMonoSources = numMono; device->NumStereoSources = numStereo; if(ConfigValueInt(alstr_get_cstr(device->DeviceName), NULL, "sends", &new_sends)) new_sends = mini(numSends, clampi(new_sends, 0, MAX_SENDS)); else new_sends = numSends; } if((device->Flags&DEVICE_RUNNING)) return ALC_NO_ERROR; al_free(device->Uhj_Encoder); device->Uhj_Encoder = NULL; al_free(device->Bs2b); device->Bs2b = NULL; al_free(device->ChannelDelay[0].Buffer); for(i = 0;i < MAX_OUTPUT_CHANNELS;i++) { device->ChannelDelay[i].Length = 0; device->ChannelDelay[i].Buffer = NULL; } al_free(device->Dry.Buffer); device->Dry.Buffer = NULL; device->Dry.NumChannels = 0; device->FOAOut.Buffer = NULL; device->FOAOut.NumChannels = 0; device->RealOut.Buffer = NULL; device->RealOut.NumChannels = 0; UpdateClockBase(device); device->FixedLatency = 0; device->DitherSeed = DITHER_RNG_SEED; /************************************************************************* * Update device format request if HRTF is requested */ device->HrtfStatus = ALC_HRTF_DISABLED_SOFT; if(device->Type != Loopback) { const char *hrtf; if(ConfigValueStr(alstr_get_cstr(device->DeviceName), NULL, "hrtf", &hrtf)) { if(strcasecmp(hrtf, "true") == 0) hrtf_userreq = Hrtf_Enable; else if(strcasecmp(hrtf, "false") == 0) hrtf_userreq = Hrtf_Disable; else if(strcasecmp(hrtf, "auto") != 0) ERR("Unexpected hrtf value: %s\n", hrtf); } if(hrtf_userreq == Hrtf_Enable || (hrtf_userreq != Hrtf_Disable && hrtf_appreq == Hrtf_Enable)) { struct Hrtf *hrtf = NULL; if(VECTOR_SIZE(device->HrtfList) == 0) { VECTOR_DEINIT(device->HrtfList); device->HrtfList = EnumerateHrtf(device->DeviceName); } if(VECTOR_SIZE(device->HrtfList) > 0) { if(hrtf_id >= 0 && (size_t)hrtf_id < VECTOR_SIZE(device->HrtfList)) hrtf = GetLoadedHrtf(VECTOR_ELEM(device->HrtfList, hrtf_id).hrtf); else hrtf = GetLoadedHrtf(VECTOR_ELEM(device->HrtfList, 0).hrtf); } if(hrtf) { device->FmtChans = DevFmtStereo; device->Frequency = hrtf->sampleRate; device->Flags |= DEVICE_CHANNELS_REQUEST | DEVICE_FREQUENCY_REQUEST; if(device->HrtfHandle) Hrtf_DecRef(device->HrtfHandle); device->HrtfHandle = hrtf; } else { hrtf_userreq = Hrtf_Default; hrtf_appreq = Hrtf_Disable; device->HrtfStatus = ALC_HRTF_UNSUPPORTED_FORMAT_SOFT; } } } oldFreq = device->Frequency; oldChans = device->FmtChans; oldType = device->FmtType; TRACE("Pre-reset: %s%s, %s%s, %s%uhz, %u update size x%d\n", (device->Flags&DEVICE_CHANNELS_REQUEST)?"*":"", DevFmtChannelsString(device->FmtChans), (device->Flags&DEVICE_SAMPLE_TYPE_REQUEST)?"*":"", DevFmtTypeString(device->FmtType), (device->Flags&DEVICE_FREQUENCY_REQUEST)?"*":"", device->Frequency, device->UpdateSize, device->NumUpdates ); if(V0(device->Backend,reset)() == ALC_FALSE) return ALC_INVALID_DEVICE; if(device->FmtChans != oldChans && (device->Flags&DEVICE_CHANNELS_REQUEST)) { ERR("Failed to set %s, got %s instead\n", DevFmtChannelsString(oldChans), DevFmtChannelsString(device->FmtChans)); device->Flags &= ~DEVICE_CHANNELS_REQUEST; } if(device->FmtType != oldType && (device->Flags&DEVICE_SAMPLE_TYPE_REQUEST)) { ERR("Failed to set %s, got %s instead\n", DevFmtTypeString(oldType), DevFmtTypeString(device->FmtType)); device->Flags &= ~DEVICE_SAMPLE_TYPE_REQUEST; } if(device->Frequency != oldFreq && (device->Flags&DEVICE_FREQUENCY_REQUEST)) { ERR("Failed to set %uhz, got %uhz instead\n", oldFreq, device->Frequency); device->Flags &= ~DEVICE_FREQUENCY_REQUEST; } if((device->UpdateSize&3) != 0) { if((CPUCapFlags&CPU_CAP_SSE)) WARN("SSE performs best with multiple of 4 update sizes (%u)\n", device->UpdateSize); if((CPUCapFlags&CPU_CAP_NEON)) WARN("NEON performs best with multiple of 4 update sizes (%u)\n", device->UpdateSize); } TRACE("Post-reset: %s, %s, %uhz, %u update size x%d\n", DevFmtChannelsString(device->FmtChans), DevFmtTypeString(device->FmtType), device->Frequency, device->UpdateSize, device->NumUpdates ); aluInitRenderer(device, hrtf_id, hrtf_appreq, hrtf_userreq); TRACE("Channel config, Dry: %d, FOA: %d, Real: %d\n", device->Dry.NumChannels, device->FOAOut.NumChannels, device->RealOut.NumChannels); /* Allocate extra channels for any post-filter output. */ size = (device->Dry.NumChannels + device->FOAOut.NumChannels + device->RealOut.NumChannels)*sizeof(device->Dry.Buffer[0]); TRACE("Allocating "SZFMT" channels, "SZFMT" bytes\n", size/sizeof(device->Dry.Buffer[0]), size); device->Dry.Buffer = al_calloc(16, size); if(!device->Dry.Buffer) { ERR("Failed to allocate "SZFMT" bytes for mix buffer\n", size); return ALC_INVALID_DEVICE; } if(device->RealOut.NumChannels != 0) device->RealOut.Buffer = device->Dry.Buffer + device->Dry.NumChannels + device->FOAOut.NumChannels; else { device->RealOut.Buffer = device->Dry.Buffer; device->RealOut.NumChannels = device->Dry.NumChannels; } if(device->FOAOut.NumChannels != 0) device->FOAOut.Buffer = device->Dry.Buffer + device->Dry.NumChannels; else { device->FOAOut.Buffer = device->Dry.Buffer; device->FOAOut.NumChannels = device->Dry.NumChannels; } device->NumAuxSends = new_sends; TRACE("Max sources: %d (%d + %d), effect slots: %d, sends: %d\n", device->SourcesMax, device->NumMonoSources, device->NumStereoSources, device->AuxiliaryEffectSlotMax, device->NumAuxSends); device->DitherDepth = 0.0f; if(GetConfigValueBool(alstr_get_cstr(device->DeviceName), NULL, "dither", 1)) { ALint depth = 0; ConfigValueInt(alstr_get_cstr(device->DeviceName), NULL, "dither-depth", &depth); if(depth <= 0) { switch(device->FmtType) { case DevFmtByte: case DevFmtUByte: depth = 8; break; case DevFmtShort: case DevFmtUShort: depth = 16; break; case DevFmtInt: case DevFmtUInt: case DevFmtFloat: break; } } if(depth > 0) { depth = clampi(depth, 2, 24); device->DitherDepth = powf(2.0f, (ALfloat)(depth-1)); } } if(!(device->DitherDepth > 0.0f)) TRACE("Dithering disabled\n"); else TRACE("Dithering enabled (%g-bit, %g)\n", log2f(device->DitherDepth)+1.0f, device->DitherDepth); device->LimiterState = gainLimiter; if(ConfigValueBool(alstr_get_cstr(device->DeviceName), NULL, "output-limiter", &val)) gainLimiter = val ? ALC_TRUE : ALC_FALSE; /* Valid values for gainLimiter are ALC_DONT_CARE_SOFT, ALC_TRUE, and * ALC_FALSE. For ALC_DONT_CARE_SOFT, use the limiter for integer-based * output (where samples must be clamped), and don't for floating-point * (which can take unclamped samples). */ if(gainLimiter == ALC_DONT_CARE_SOFT) { switch(device->FmtType) { case DevFmtByte: case DevFmtUByte: case DevFmtShort: case DevFmtUShort: case DevFmtInt: case DevFmtUInt: gainLimiter = ALC_TRUE; break; case DevFmtFloat: gainLimiter = ALC_FALSE; break; } } if(gainLimiter != ALC_FALSE) { ALfloat thrshld = 1.0f; switch(device->FmtType) { case DevFmtByte: case DevFmtUByte: thrshld = 127.0f / 128.0f; break; case DevFmtShort: case DevFmtUShort: thrshld = 32767.0f / 32768.0f; break; case DevFmtInt: case DevFmtUInt: case DevFmtFloat: break; } if(device->DitherDepth > 0.0f) thrshld -= 1.0f / device->DitherDepth; al_free(device->Limiter); device->Limiter = CreateDeviceLimiter(device, log10f(thrshld) * 20.0f); device->FixedLatency += (ALuint)(GetCompressorLookAhead(device->Limiter) * DEVICE_CLOCK_RES / device->Frequency); } else { al_free(device->Limiter); device->Limiter = NULL; } TRACE("Output limiter %s\n", device->Limiter ? "enabled" : "disabled"); aluSelectPostProcess(device); TRACE("Fixed device latency: %uns\n", device->FixedLatency); /* Need to delay returning failure until replacement Send arrays have been * allocated with the appropriate size. */ update_failed = AL_FALSE; START_MIXER_MODE(); context = ATOMIC_LOAD_SEQ(&device->ContextList); while(context) { SourceSubList *sublist, *subend; struct ALvoiceProps *vprops; ALsizei pos; if(context->DefaultSlot) { ALeffectslot *slot = context->DefaultSlot; ALeffectState *state = slot->Effect.State; state->OutBuffer = device->Dry.Buffer; state->OutChannels = device->Dry.NumChannels; if(V(state,deviceUpdate)(device) == AL_FALSE) update_failed = AL_TRUE; else UpdateEffectSlotProps(slot, context); } almtx_lock(&context->PropLock); almtx_lock(&context->EffectSlotLock); for(pos = 0;pos < (ALsizei)VECTOR_SIZE(context->EffectSlotList);pos++) { ALeffectslot *slot = VECTOR_ELEM(context->EffectSlotList, pos); ALeffectState *state = slot->Effect.State; state->OutBuffer = device->Dry.Buffer; state->OutChannels = device->Dry.NumChannels; if(V(state,deviceUpdate)(device) == AL_FALSE) update_failed = AL_TRUE; else UpdateEffectSlotProps(slot, context); } almtx_unlock(&context->EffectSlotLock); almtx_lock(&context->SourceLock); sublist = VECTOR_BEGIN(context->SourceList); subend = VECTOR_END(context->SourceList); for(;sublist != subend;++sublist) { ALuint64 usemask = ~sublist->FreeMask; while(usemask) { ALsizei idx = CTZ64(usemask); ALsource *source = sublist->Sources + idx; usemask &= ~(U64(1) << idx); if(old_sends != device->NumAuxSends) { ALvoid *sends = al_calloc(16, device->NumAuxSends*sizeof(source->Send[0])); ALsizei s; memcpy(sends, source->Send, mini(device->NumAuxSends, old_sends)*sizeof(source->Send[0]) ); for(s = device->NumAuxSends;s < old_sends;s++) { if(source->Send[s].Slot) DecrementRef(&source->Send[s].Slot->ref); source->Send[s].Slot = NULL; } al_free(source->Send); source->Send = sends; for(s = old_sends;s < device->NumAuxSends;s++) { source->Send[s].Slot = NULL; source->Send[s].Gain = 1.0f; source->Send[s].GainHF = 1.0f; source->Send[s].HFReference = LOWPASSFREQREF; source->Send[s].GainLF = 1.0f; source->Send[s].LFReference = HIGHPASSFREQREF; } } ATOMIC_FLAG_CLEAR(&source->PropsClean, almemory_order_release); } } /* Clear any pre-existing voice property structs, in case the number of * auxiliary sends is changing. Active sources will have updates * respecified in UpdateAllSourceProps. */ vprops = ATOMIC_EXCHANGE_PTR(&context->FreeVoiceProps, NULL, almemory_order_acq_rel); while(vprops) { struct ALvoiceProps *next = ATOMIC_LOAD(&vprops->next, almemory_order_relaxed); al_free(vprops); vprops = next; } AllocateVoices(context, context->MaxVoices, old_sends); for(pos = 0;pos < context->VoiceCount;pos++) { ALvoice *voice = context->Voices[pos]; al_free(ATOMIC_EXCHANGE_PTR(&voice->Update, NULL, almemory_order_acq_rel)); if(ATOMIC_LOAD(&voice->Source, almemory_order_acquire) == NULL) continue; if(device->AvgSpeakerDist > 0.0f) { /* Reinitialize the NFC filters for new parameters. */ ALfloat w1 = SPEEDOFSOUNDMETRESPERSEC / (device->AvgSpeakerDist * device->Frequency); for(i = 0;i < voice->NumChannels;i++) NfcFilterCreate(&voice->Direct.Params[i].NFCtrlFilter, 0.0f, w1); } } almtx_unlock(&context->SourceLock); ATOMIC_FLAG_TEST_AND_SET(&context->PropsClean, almemory_order_release); UpdateContextProps(context); ATOMIC_FLAG_TEST_AND_SET(&context->Listener->PropsClean, almemory_order_release); UpdateListenerProps(context); UpdateAllSourceProps(context); almtx_unlock(&context->PropLock); context = ATOMIC_LOAD(&context->next, almemory_order_relaxed); } END_MIXER_MODE(); if(update_failed) return ALC_INVALID_DEVICE; if(!(device->Flags&DEVICE_PAUSED)) { if(V0(device->Backend,start)() == ALC_FALSE) return ALC_INVALID_DEVICE; device->Flags |= DEVICE_RUNNING; } return ALC_NO_ERROR; } static void InitDevice(ALCdevice *device, enum DeviceType type) { ALsizei i; InitRef(&device->ref, 1); ATOMIC_INIT(&device->Connected, ALC_TRUE); device->Type = type; ATOMIC_INIT(&device->LastError, ALC_NO_ERROR); device->Flags = 0; device->Render_Mode = NormalRender; device->AvgSpeakerDist = 0.0f; device->LimiterState = ALC_DONT_CARE_SOFT; ATOMIC_INIT(&device->ContextList, NULL); device->ClockBase = 0; device->SamplesDone = 0; device->FixedLatency = 0; device->SourcesMax = 0; device->AuxiliaryEffectSlotMax = 0; device->NumAuxSends = 0; device->Dry.Buffer = NULL; device->Dry.NumChannels = 0; device->FOAOut.Buffer = NULL; device->FOAOut.NumChannels = 0; device->RealOut.Buffer = NULL; device->RealOut.NumChannels = 0; AL_STRING_INIT(device->DeviceName); for(i = 0;i < MAX_OUTPUT_CHANNELS;i++) { device->ChannelDelay[i].Gain = 1.0f; device->ChannelDelay[i].Length = 0; device->ChannelDelay[i].Buffer = NULL; } AL_STRING_INIT(device->HrtfName); VECTOR_INIT(device->HrtfList); device->HrtfHandle = NULL; device->Hrtf = NULL; device->Bs2b = NULL; device->Uhj_Encoder = NULL; device->AmbiDecoder = NULL; device->AmbiUp = NULL; device->Stablizer = NULL; device->Limiter = NULL; VECTOR_INIT(device->BufferList); almtx_init(&device->BufferLock, almtx_plain); VECTOR_INIT(device->EffectList); almtx_init(&device->EffectLock, almtx_plain); VECTOR_INIT(device->FilterList); almtx_init(&device->FilterLock, almtx_plain); almtx_init(&device->BackendLock, almtx_plain); device->Backend = NULL; ATOMIC_INIT(&device->next, NULL); } /* FreeDevice * * Frees the device structure, and destroys any objects the app failed to * delete. Called once there's no more references on the device. */ static ALCvoid FreeDevice(ALCdevice *device) { ALsizei i; TRACE("%p\n", device); if(device->Backend) DELETE_OBJ(device->Backend); device->Backend = NULL; almtx_destroy(&device->BackendLock); ReleaseALBuffers(device); #define FREE_BUFFERSUBLIST(x) al_free((x)->Buffers) VECTOR_FOR_EACH(BufferSubList, device->BufferList, FREE_BUFFERSUBLIST); #undef FREE_BUFFERSUBLIST VECTOR_DEINIT(device->BufferList); almtx_destroy(&device->BufferLock); ReleaseALEffects(device); #define FREE_EFFECTSUBLIST(x) al_free((x)->Effects) VECTOR_FOR_EACH(EffectSubList, device->EffectList, FREE_EFFECTSUBLIST); #undef FREE_EFFECTSUBLIST VECTOR_DEINIT(device->EffectList); almtx_destroy(&device->EffectLock); ReleaseALFilters(device); #define FREE_FILTERSUBLIST(x) al_free((x)->Filters) VECTOR_FOR_EACH(FilterSubList, device->FilterList, FREE_FILTERSUBLIST); #undef FREE_FILTERSUBLIST VECTOR_DEINIT(device->FilterList); almtx_destroy(&device->FilterLock); AL_STRING_DEINIT(device->HrtfName); FreeHrtfList(&device->HrtfList); if(device->HrtfHandle) Hrtf_DecRef(device->HrtfHandle); device->HrtfHandle = NULL; al_free(device->Hrtf); device->Hrtf = NULL; al_free(device->Bs2b); device->Bs2b = NULL; al_free(device->Uhj_Encoder); device->Uhj_Encoder = NULL; bformatdec_free(&device->AmbiDecoder); ambiup_free(&device->AmbiUp); al_free(device->Stablizer); device->Stablizer = NULL; al_free(device->Limiter); device->Limiter = NULL; al_free(device->ChannelDelay[0].Buffer); for(i = 0;i < MAX_OUTPUT_CHANNELS;i++) { device->ChannelDelay[i].Gain = 1.0f; device->ChannelDelay[i].Length = 0; device->ChannelDelay[i].Buffer = NULL; } AL_STRING_DEINIT(device->DeviceName); al_free(device->Dry.Buffer); device->Dry.Buffer = NULL; device->Dry.NumChannels = 0; device->FOAOut.Buffer = NULL; device->FOAOut.NumChannels = 0; device->RealOut.Buffer = NULL; device->RealOut.NumChannels = 0; al_free(device); } void ALCdevice_IncRef(ALCdevice *device) { uint ref; ref = IncrementRef(&device->ref); TRACEREF("%p increasing refcount to %u\n", device, ref); } void ALCdevice_DecRef(ALCdevice *device) { uint ref; ref = DecrementRef(&device->ref); TRACEREF("%p decreasing refcount to %u\n", device, ref); if(ref == 0) FreeDevice(device); } /* VerifyDevice * * Checks if the device handle is valid, and increments its ref count if so. */ static ALCboolean VerifyDevice(ALCdevice **device) { ALCdevice *tmpDevice; LockLists(); tmpDevice = ATOMIC_LOAD_SEQ(&DeviceList); while(tmpDevice) { if(tmpDevice == *device) { ALCdevice_IncRef(tmpDevice); UnlockLists(); return ALC_TRUE; } tmpDevice = ATOMIC_LOAD(&tmpDevice->next, almemory_order_relaxed); } UnlockLists(); *device = NULL; return ALC_FALSE; } /* InitContext * * Initializes context fields */ static ALvoid InitContext(ALCcontext *Context) { ALlistener *listener = Context->Listener; struct ALeffectslotArray *auxslots; //Initialise listener listener->Gain = 1.0f; listener->Position[0] = 0.0f; listener->Position[1] = 0.0f; listener->Position[2] = 0.0f; listener->Velocity[0] = 0.0f; listener->Velocity[1] = 0.0f; listener->Velocity[2] = 0.0f; listener->Forward[0] = 0.0f; listener->Forward[1] = 0.0f; listener->Forward[2] = -1.0f; listener->Up[0] = 0.0f; listener->Up[1] = 1.0f; listener->Up[2] = 0.0f; ATOMIC_FLAG_TEST_AND_SET(&listener->PropsClean, almemory_order_relaxed); ATOMIC_INIT(&listener->Update, NULL); //Validate Context InitRef(&Context->UpdateCount, 0); ATOMIC_INIT(&Context->HoldUpdates, AL_FALSE); Context->GainBoost = 1.0f; almtx_init(&Context->PropLock, almtx_plain); ATOMIC_INIT(&Context->LastError, AL_NO_ERROR); VECTOR_INIT(Context->SourceList); Context->NumSources = 0; almtx_init(&Context->SourceLock, almtx_plain); VECTOR_INIT(Context->EffectSlotList); almtx_init(&Context->EffectSlotLock, almtx_plain); if(Context->DefaultSlot) { auxslots = al_calloc(DEF_ALIGN, FAM_SIZE(struct ALeffectslotArray, slot, 1)); auxslots->count = 1; auxslots->slot[0] = Context->DefaultSlot; } else { auxslots = al_calloc(DEF_ALIGN, sizeof(struct ALeffectslotArray)); auxslots->count = 0; } ATOMIC_INIT(&Context->ActiveAuxSlots, auxslots); //Set globals Context->DistanceModel = DefaultDistanceModel; Context->SourceDistanceModel = AL_FALSE; Context->DopplerFactor = 1.0f; Context->DopplerVelocity = 1.0f; Context->SpeedOfSound = SPEEDOFSOUNDMETRESPERSEC; Context->MetersPerUnit = AL_DEFAULT_METERS_PER_UNIT; ATOMIC_FLAG_TEST_AND_SET(&Context->PropsClean, almemory_order_relaxed); ATOMIC_INIT(&Context->DeferUpdates, AL_FALSE); alsem_init(&Context->EventSem, 0); Context->AsyncEvents = NULL; ATOMIC_INIT(&Context->EnabledEvts, 0); almtx_init(&Context->EventCbLock, almtx_plain); Context->EventCb = NULL; Context->EventParam = NULL; ATOMIC_INIT(&Context->Update, NULL); ATOMIC_INIT(&Context->FreeContextProps, NULL); ATOMIC_INIT(&Context->FreeListenerProps, NULL); ATOMIC_INIT(&Context->FreeVoiceProps, NULL); ATOMIC_INIT(&Context->FreeEffectslotProps, NULL); Context->ExtensionList = alExtList; listener->Params.Matrix = IdentityMatrixf; aluVectorSet(&listener->Params.Velocity, 0.0f, 0.0f, 0.0f, 0.0f); listener->Params.Gain = listener->Gain; listener->Params.MetersPerUnit = Context->MetersPerUnit; listener->Params.DopplerFactor = Context->DopplerFactor; listener->Params.SpeedOfSound = Context->SpeedOfSound * Context->DopplerVelocity; listener->Params.ReverbSpeedOfSound = listener->Params.SpeedOfSound * listener->Params.MetersPerUnit; listener->Params.SourceDistanceModel = Context->SourceDistanceModel; listener->Params.DistanceModel = Context->DistanceModel; Context->AsyncEvents = ll_ringbuffer_create(63, sizeof(AsyncEvent), false); if(althrd_create(&Context->EventThread, EventThread, Context) != althrd_success) ERR("Failed to start event thread! Expect problems.\n"); } /* FreeContext * * Cleans up the context, and destroys any remaining objects the app failed to * delete. Called once there's no more references on the context. */ static void FreeContext(ALCcontext *context) { ALlistener *listener = context->Listener; struct ALeffectslotArray *auxslots; struct ALeffectslotProps *eprops; struct ALlistenerProps *lprops; struct ALcontextProps *cprops; struct ALvoiceProps *vprops; size_t count; ALsizei i; TRACE("%p\n", context); if((cprops=ATOMIC_LOAD(&context->Update, almemory_order_acquire)) != NULL) { TRACE("Freed unapplied context update %p\n", cprops); al_free(cprops); } count = 0; cprops = ATOMIC_LOAD(&context->FreeContextProps, almemory_order_acquire); while(cprops) { struct ALcontextProps *next = ATOMIC_LOAD(&cprops->next, almemory_order_acquire); al_free(cprops); cprops = next; ++count; } TRACE("Freed "SZFMT" context property object%s\n", count, (count==1)?"":"s"); if(context->DefaultSlot) { DeinitEffectSlot(context->DefaultSlot); context->DefaultSlot = NULL; } auxslots = ATOMIC_EXCHANGE_PTR(&context->ActiveAuxSlots, NULL, almemory_order_relaxed); al_free(auxslots); ReleaseALSources(context); #define FREE_SOURCESUBLIST(x) al_free((x)->Sources) VECTOR_FOR_EACH(SourceSubList, context->SourceList, FREE_SOURCESUBLIST); #undef FREE_SOURCESUBLIST VECTOR_DEINIT(context->SourceList); context->NumSources = 0; almtx_destroy(&context->SourceLock); count = 0; eprops = ATOMIC_LOAD(&context->FreeEffectslotProps, almemory_order_relaxed); while(eprops) { struct ALeffectslotProps *next = ATOMIC_LOAD(&eprops->next, almemory_order_relaxed); if(eprops->State) ALeffectState_DecRef(eprops->State); al_free(eprops); eprops = next; ++count; } TRACE("Freed "SZFMT" AuxiliaryEffectSlot property object%s\n", count, (count==1)?"":"s"); ReleaseALAuxiliaryEffectSlots(context); #define FREE_EFFECTSLOTPTR(x) al_free(*(x)) VECTOR_FOR_EACH(ALeffectslotPtr, context->EffectSlotList, FREE_EFFECTSLOTPTR); #undef FREE_EFFECTSLOTPTR VECTOR_DEINIT(context->EffectSlotList); almtx_destroy(&context->EffectSlotLock); count = 0; vprops = ATOMIC_LOAD(&context->FreeVoiceProps, almemory_order_relaxed); while(vprops) { struct ALvoiceProps *next = ATOMIC_LOAD(&vprops->next, almemory_order_relaxed); al_free(vprops); vprops = next; ++count; } TRACE("Freed "SZFMT" voice property object%s\n", count, (count==1)?"":"s"); for(i = 0;i < context->VoiceCount;i++) DeinitVoice(context->Voices[i]); al_free(context->Voices); context->Voices = NULL; context->VoiceCount = 0; context->MaxVoices = 0; if((lprops=ATOMIC_LOAD(&listener->Update, almemory_order_acquire)) != NULL) { TRACE("Freed unapplied listener update %p\n", lprops); al_free(lprops); } count = 0; lprops = ATOMIC_LOAD(&context->FreeListenerProps, almemory_order_acquire); while(lprops) { struct ALlistenerProps *next = ATOMIC_LOAD(&lprops->next, almemory_order_acquire); al_free(lprops); lprops = next; ++count; } TRACE("Freed "SZFMT" listener property object%s\n", count, (count==1)?"":"s"); almtx_destroy(&context->EventCbLock); alsem_destroy(&context->EventSem); ll_ringbuffer_free(context->AsyncEvents); context->AsyncEvents = NULL; almtx_destroy(&context->PropLock); ALCdevice_DecRef(context->Device); context->Device = NULL; //Invalidate context memset(context, 0, sizeof(ALCcontext)); al_free(context); } /* ReleaseContext * * Removes the context reference from the given device and removes it from * being current on the running thread or globally. Returns true if other * contexts still exist on the device. */ static bool ReleaseContext(ALCcontext *context, ALCdevice *device) { static const AsyncEvent kill_evt = ASYNC_EVENT(EventType_KillThread); ALCcontext *origctx, *newhead; bool ret = true; if(altss_get(LocalContext) == context) { WARN("%p released while current on thread\n", context); altss_set(LocalContext, NULL); ALCcontext_DecRef(context); } origctx = context; if(ATOMIC_COMPARE_EXCHANGE_PTR_STRONG_SEQ(&GlobalContext, &origctx, NULL)) ALCcontext_DecRef(context); V0(device->Backend,lock)(); origctx = context; newhead = ATOMIC_LOAD(&context->next, almemory_order_relaxed); if(!ATOMIC_COMPARE_EXCHANGE_PTR_STRONG_SEQ(&device->ContextList, &origctx, newhead)) { ALCcontext *list; do { /* origctx is what the desired context failed to match. Try * swapping out the next one in the list. */ list = origctx; origctx = context; } while(!ATOMIC_COMPARE_EXCHANGE_PTR_STRONG_SEQ(&list->next, &origctx, newhead)); } else ret = !!newhead; V0(device->Backend,unlock)(); /* Make sure the context is finished and no longer processing in the mixer * before sending the message queue kill event. The backend's lock does * this, although waiting for a non-odd mix count would work too. */ while(ll_ringbuffer_write(context->AsyncEvents, (const char*)&kill_evt, 1) == 0) althrd_yield(); alsem_post(&context->EventSem); althrd_join(context->EventThread, NULL); ALCcontext_DecRef(context); return ret; } static void ALCcontext_IncRef(ALCcontext *context) { uint ref = IncrementRef(&context->ref); TRACEREF("%p increasing refcount to %u\n", context, ref); } void ALCcontext_DecRef(ALCcontext *context) { uint ref = DecrementRef(&context->ref); TRACEREF("%p decreasing refcount to %u\n", context, ref); if(ref == 0) FreeContext(context); } static void ReleaseThreadCtx(void *ptr) { ALCcontext *context = ptr; uint ref = DecrementRef(&context->ref); TRACEREF("%p decreasing refcount to %u\n", context, ref); ERR("Context %p current for thread being destroyed, possible leak!\n", context); } /* VerifyContext * * Checks that the given context is valid, and increments its reference count. */ static ALCboolean VerifyContext(ALCcontext **context) { ALCdevice *dev; LockLists(); dev = ATOMIC_LOAD_SEQ(&DeviceList); while(dev) { ALCcontext *ctx = ATOMIC_LOAD(&dev->ContextList, almemory_order_acquire); while(ctx) { if(ctx == *context) { ALCcontext_IncRef(ctx); UnlockLists(); return ALC_TRUE; } ctx = ATOMIC_LOAD(&ctx->next, almemory_order_relaxed); } dev = ATOMIC_LOAD(&dev->next, almemory_order_relaxed); } UnlockLists(); *context = NULL; return ALC_FALSE; } /* GetContextRef * * Returns the currently active context for this thread, and adds a reference * without locking it. */ ALCcontext *GetContextRef(void) { ALCcontext *context; context = altss_get(LocalContext); if(context) ALCcontext_IncRef(context); else { LockLists(); context = ATOMIC_LOAD_SEQ(&GlobalContext); if(context) ALCcontext_IncRef(context); UnlockLists(); } return context; } void AllocateVoices(ALCcontext *context, ALsizei num_voices, ALsizei old_sends) { ALCdevice *device = context->Device; ALsizei num_sends = device->NumAuxSends; struct ALvoiceProps *props; size_t sizeof_props; size_t sizeof_voice; ALvoice **voices; ALvoice *voice; ALsizei v = 0; size_t size; if(num_voices == context->MaxVoices && num_sends == old_sends) return; /* Allocate the voice pointers, voices, and the voices' stored source * property set (including the dynamically-sized Send[] array) in one * chunk. */ sizeof_voice = RoundUp(FAM_SIZE(ALvoice, Send, num_sends), 16); sizeof_props = RoundUp(FAM_SIZE(struct ALvoiceProps, Send, num_sends), 16); size = sizeof(ALvoice*) + sizeof_voice + sizeof_props; voices = al_calloc(16, RoundUp(size*num_voices, 16)); /* The voice and property objects are stored interleaved since they're * paired together. */ voice = (ALvoice*)((char*)voices + RoundUp(num_voices*sizeof(ALvoice*), 16)); props = (struct ALvoiceProps*)((char*)voice + sizeof_voice); if(context->Voices) { const ALsizei v_count = mini(context->VoiceCount, num_voices); const ALsizei s_count = mini(old_sends, num_sends); for(;v < v_count;v++) { ALvoice *old_voice = context->Voices[v]; ALsizei i; /* Copy the old voice data and source property set to the new * storage. */ *voice = *old_voice; for(i = 0;i < s_count;i++) voice->Send[i] = old_voice->Send[i]; *props = *(old_voice->Props); for(i = 0;i < s_count;i++) props->Send[i] = old_voice->Props->Send[i]; /* Set this voice's property set pointer and voice reference. */ voice->Props = props; voices[v] = voice; /* Increment pointers to the next storage space. */ voice = (ALvoice*)((char*)props + sizeof_props); props = (struct ALvoiceProps*)((char*)voice + sizeof_voice); } /* Deinit any left over voices that weren't copied over to the new * array. NOTE: If this does anything, v equals num_voices and * num_voices is less than VoiceCount, so the following loop won't do * anything. */ for(;v < context->VoiceCount;v++) DeinitVoice(context->Voices[v]); } /* Finish setting the voices' property set pointers and references. */ for(;v < num_voices;v++) { ATOMIC_INIT(&voice->Update, NULL); voice->Props = props; voices[v] = voice; voice = (ALvoice*)((char*)props + sizeof_props); props = (struct ALvoiceProps*)((char*)voice + sizeof_voice); } al_free(context->Voices); context->Voices = voices; context->MaxVoices = num_voices; context->VoiceCount = mini(context->VoiceCount, num_voices); } /************************************************ * Standard ALC functions ************************************************/ /* alcGetError * * Return last ALC generated error code for the given device */ ALC_API ALCenum ALC_APIENTRY alcGetError(ALCdevice *device) { ALCenum errorCode; if(VerifyDevice(&device)) { errorCode = ATOMIC_EXCHANGE_SEQ(&device->LastError, ALC_NO_ERROR); ALCdevice_DecRef(device); } else errorCode = ATOMIC_EXCHANGE_SEQ(&LastNullDeviceError, ALC_NO_ERROR); return errorCode; } /* alcSuspendContext * * Suspends updates for the given context */ ALC_API ALCvoid ALC_APIENTRY alcSuspendContext(ALCcontext *context) { if(!SuspendDefers) return; if(!VerifyContext(&context)) alcSetError(NULL, ALC_INVALID_CONTEXT); else { ALCcontext_DeferUpdates(context); ALCcontext_DecRef(context); } } /* alcProcessContext * * Resumes processing updates for the given context */ ALC_API ALCvoid ALC_APIENTRY alcProcessContext(ALCcontext *context) { if(!SuspendDefers) return; if(!VerifyContext(&context)) alcSetError(NULL, ALC_INVALID_CONTEXT); else { ALCcontext_ProcessUpdates(context); ALCcontext_DecRef(context); } } /* alcGetString * * Returns information about the device, and error strings */ ALC_API const ALCchar* ALC_APIENTRY alcGetString(ALCdevice *Device, ALCenum param) { const ALCchar *value = NULL; switch(param) { case ALC_NO_ERROR: value = alcNoError; break; case ALC_INVALID_ENUM: value = alcErrInvalidEnum; break; case ALC_INVALID_VALUE: value = alcErrInvalidValue; break; case ALC_INVALID_DEVICE: value = alcErrInvalidDevice; break; case ALC_INVALID_CONTEXT: value = alcErrInvalidContext; break; case ALC_OUT_OF_MEMORY: value = alcErrOutOfMemory; break; case ALC_DEVICE_SPECIFIER: value = alcDefaultName; break; case ALC_ALL_DEVICES_SPECIFIER: if(VerifyDevice(&Device)) { value = alstr_get_cstr(Device->DeviceName); ALCdevice_DecRef(Device); } else { ProbeAllDevicesList(); value = alstr_get_cstr(alcAllDevicesList); } break; case ALC_CAPTURE_DEVICE_SPECIFIER: if(VerifyDevice(&Device)) { value = alstr_get_cstr(Device->DeviceName); ALCdevice_DecRef(Device); } else { ProbeCaptureDeviceList(); value = alstr_get_cstr(alcCaptureDeviceList); } break; /* Default devices are always first in the list */ case ALC_DEFAULT_DEVICE_SPECIFIER: value = alcDefaultName; break; case ALC_DEFAULT_ALL_DEVICES_SPECIFIER: if(alstr_empty(alcAllDevicesList)) ProbeAllDevicesList(); VerifyDevice(&Device); free(alcDefaultAllDevicesSpecifier); alcDefaultAllDevicesSpecifier = strdup(alstr_get_cstr(alcAllDevicesList)); if(!alcDefaultAllDevicesSpecifier) alcSetError(Device, ALC_OUT_OF_MEMORY); value = alcDefaultAllDevicesSpecifier; if(Device) ALCdevice_DecRef(Device); break; case ALC_CAPTURE_DEFAULT_DEVICE_SPECIFIER: if(alstr_empty(alcCaptureDeviceList)) ProbeCaptureDeviceList(); VerifyDevice(&Device); free(alcCaptureDefaultDeviceSpecifier); alcCaptureDefaultDeviceSpecifier = strdup(alstr_get_cstr(alcCaptureDeviceList)); if(!alcCaptureDefaultDeviceSpecifier) alcSetError(Device, ALC_OUT_OF_MEMORY); value = alcCaptureDefaultDeviceSpecifier; if(Device) ALCdevice_DecRef(Device); break; case ALC_EXTENSIONS: if(!VerifyDevice(&Device)) value = alcNoDeviceExtList; else { value = alcExtensionList; ALCdevice_DecRef(Device); } break; case ALC_HRTF_SPECIFIER_SOFT: if(!VerifyDevice(&Device)) alcSetError(NULL, ALC_INVALID_DEVICE); else { almtx_lock(&Device->BackendLock); value = (Device->HrtfHandle ? alstr_get_cstr(Device->HrtfName) : ""); almtx_unlock(&Device->BackendLock); ALCdevice_DecRef(Device); } break; default: VerifyDevice(&Device); alcSetError(Device, ALC_INVALID_ENUM); if(Device) ALCdevice_DecRef(Device); break; } return value; } static inline ALCsizei NumAttrsForDevice(ALCdevice *device) { if(device->Type == Capture) return 9; if(device->Type != Loopback) return 29; if(device->FmtChans == DevFmtAmbi3D) return 35; return 29; } static ALCsizei GetIntegerv(ALCdevice *device, ALCenum param, ALCsizei size, ALCint *values) { ALCsizei i; if(size <= 0 || values == NULL) { alcSetError(device, ALC_INVALID_VALUE); return 0; } if(!device) { switch(param) { case ALC_MAJOR_VERSION: values[0] = alcMajorVersion; return 1; case ALC_MINOR_VERSION: values[0] = alcMinorVersion; return 1; case ALC_ATTRIBUTES_SIZE: case ALC_ALL_ATTRIBUTES: case ALC_FREQUENCY: case ALC_REFRESH: case ALC_SYNC: case ALC_MONO_SOURCES: case ALC_STEREO_SOURCES: case ALC_CAPTURE_SAMPLES: case ALC_FORMAT_CHANNELS_SOFT: case ALC_FORMAT_TYPE_SOFT: case ALC_AMBISONIC_LAYOUT_SOFT: case ALC_AMBISONIC_SCALING_SOFT: case ALC_AMBISONIC_ORDER_SOFT: case ALC_MAX_AMBISONIC_ORDER_SOFT: alcSetError(NULL, ALC_INVALID_DEVICE); return 0; default: alcSetError(NULL, ALC_INVALID_ENUM); return 0; } return 0; } if(device->Type == Capture) { switch(param) { case ALC_ATTRIBUTES_SIZE: values[0] = NumAttrsForDevice(device); return 1; case ALC_ALL_ATTRIBUTES: if(size < NumAttrsForDevice(device)) { alcSetError(device, ALC_INVALID_VALUE); return 0; } i = 0; almtx_lock(&device->BackendLock); values[i++] = ALC_MAJOR_VERSION; values[i++] = alcMajorVersion; values[i++] = ALC_MINOR_VERSION; values[i++] = alcMinorVersion; values[i++] = ALC_CAPTURE_SAMPLES; values[i++] = V0(device->Backend,availableSamples)(); values[i++] = ALC_CONNECTED; values[i++] = ATOMIC_LOAD(&device->Connected, almemory_order_relaxed); almtx_unlock(&device->BackendLock); values[i++] = 0; return i; case ALC_MAJOR_VERSION: values[0] = alcMajorVersion; return 1; case ALC_MINOR_VERSION: values[0] = alcMinorVersion; return 1; case ALC_CAPTURE_SAMPLES: almtx_lock(&device->BackendLock); values[0] = V0(device->Backend,availableSamples)(); almtx_unlock(&device->BackendLock); return 1; case ALC_CONNECTED: values[0] = ATOMIC_LOAD(&device->Connected, almemory_order_acquire); return 1; default: alcSetError(device, ALC_INVALID_ENUM); return 0; } return 0; } /* render device */ switch(param) { case ALC_ATTRIBUTES_SIZE: values[0] = NumAttrsForDevice(device); return 1; case ALC_ALL_ATTRIBUTES: if(size < NumAttrsForDevice(device)) { alcSetError(device, ALC_INVALID_VALUE); return 0; } i = 0; almtx_lock(&device->BackendLock); values[i++] = ALC_MAJOR_VERSION; values[i++] = alcMajorVersion; values[i++] = ALC_MINOR_VERSION; values[i++] = alcMinorVersion; values[i++] = ALC_EFX_MAJOR_VERSION; values[i++] = alcEFXMajorVersion; values[i++] = ALC_EFX_MINOR_VERSION; values[i++] = alcEFXMinorVersion; values[i++] = ALC_FREQUENCY; values[i++] = device->Frequency; if(device->Type != Loopback) { values[i++] = ALC_REFRESH; values[i++] = device->Frequency / device->UpdateSize; values[i++] = ALC_SYNC; values[i++] = ALC_FALSE; } else { if(device->FmtChans == DevFmtAmbi3D) { values[i++] = ALC_AMBISONIC_LAYOUT_SOFT; values[i++] = device->AmbiLayout; values[i++] = ALC_AMBISONIC_SCALING_SOFT; values[i++] = device->AmbiScale; values[i++] = ALC_AMBISONIC_ORDER_SOFT; values[i++] = device->AmbiOrder; } values[i++] = ALC_FORMAT_CHANNELS_SOFT; values[i++] = device->FmtChans; values[i++] = ALC_FORMAT_TYPE_SOFT; values[i++] = device->FmtType; } values[i++] = ALC_MONO_SOURCES; values[i++] = device->NumMonoSources; values[i++] = ALC_STEREO_SOURCES; values[i++] = device->NumStereoSources; values[i++] = ALC_MAX_AUXILIARY_SENDS; values[i++] = device->NumAuxSends; values[i++] = ALC_HRTF_SOFT; values[i++] = (device->HrtfHandle ? ALC_TRUE : ALC_FALSE); values[i++] = ALC_HRTF_STATUS_SOFT; values[i++] = device->HrtfStatus; values[i++] = ALC_OUTPUT_LIMITER_SOFT; values[i++] = device->Limiter ? ALC_TRUE : ALC_FALSE; values[i++] = ALC_MAX_AMBISONIC_ORDER_SOFT; values[i++] = MAX_AMBI_ORDER; almtx_unlock(&device->BackendLock); values[i++] = 0; return i; case ALC_MAJOR_VERSION: values[0] = alcMajorVersion; return 1; case ALC_MINOR_VERSION: values[0] = alcMinorVersion; return 1; case ALC_EFX_MAJOR_VERSION: values[0] = alcEFXMajorVersion; return 1; case ALC_EFX_MINOR_VERSION: values[0] = alcEFXMinorVersion; return 1; case ALC_FREQUENCY: values[0] = device->Frequency; return 1; case ALC_REFRESH: if(device->Type == Loopback) { alcSetError(device, ALC_INVALID_DEVICE); return 0; } almtx_lock(&device->BackendLock); values[0] = device->Frequency / device->UpdateSize; almtx_unlock(&device->BackendLock); return 1; case ALC_SYNC: if(device->Type == Loopback) { alcSetError(device, ALC_INVALID_DEVICE); return 0; } values[0] = ALC_FALSE; return 1; case ALC_FORMAT_CHANNELS_SOFT: if(device->Type != Loopback) { alcSetError(device, ALC_INVALID_DEVICE); return 0; } values[0] = device->FmtChans; return 1; case ALC_FORMAT_TYPE_SOFT: if(device->Type != Loopback) { alcSetError(device, ALC_INVALID_DEVICE); return 0; } values[0] = device->FmtType; return 1; case ALC_AMBISONIC_LAYOUT_SOFT: if(device->Type != Loopback || device->FmtChans != DevFmtAmbi3D) { alcSetError(device, ALC_INVALID_DEVICE); return 0; } values[0] = device->AmbiLayout; return 1; case ALC_AMBISONIC_SCALING_SOFT: if(device->Type != Loopback || device->FmtChans != DevFmtAmbi3D) { alcSetError(device, ALC_INVALID_DEVICE); return 0; } values[0] = device->AmbiScale; return 1; case ALC_AMBISONIC_ORDER_SOFT: if(device->Type != Loopback || device->FmtChans != DevFmtAmbi3D) { alcSetError(device, ALC_INVALID_DEVICE); return 0; } values[0] = device->AmbiOrder; return 1; case ALC_MONO_SOURCES: values[0] = device->NumMonoSources; return 1; case ALC_STEREO_SOURCES: values[0] = device->NumStereoSources; return 1; case ALC_MAX_AUXILIARY_SENDS: values[0] = device->NumAuxSends; return 1; case ALC_CONNECTED: values[0] = ATOMIC_LOAD(&device->Connected, almemory_order_acquire); return 1; case ALC_HRTF_SOFT: values[0] = (device->HrtfHandle ? ALC_TRUE : ALC_FALSE); return 1; case ALC_HRTF_STATUS_SOFT: values[0] = device->HrtfStatus; return 1; case ALC_NUM_HRTF_SPECIFIERS_SOFT: almtx_lock(&device->BackendLock); FreeHrtfList(&device->HrtfList); device->HrtfList = EnumerateHrtf(device->DeviceName); values[0] = (ALCint)VECTOR_SIZE(device->HrtfList); almtx_unlock(&device->BackendLock); return 1; case ALC_OUTPUT_LIMITER_SOFT: values[0] = device->Limiter ? ALC_TRUE : ALC_FALSE; return 1; case ALC_MAX_AMBISONIC_ORDER_SOFT: values[0] = MAX_AMBI_ORDER; return 1; default: alcSetError(device, ALC_INVALID_ENUM); return 0; } return 0; } /* alcGetIntegerv * * Returns information about the device and the version of OpenAL */ ALC_API void ALC_APIENTRY alcGetIntegerv(ALCdevice *device, ALCenum param, ALCsizei size, ALCint *values) { VerifyDevice(&device); if(size <= 0 || values == NULL) alcSetError(device, ALC_INVALID_VALUE); else GetIntegerv(device, param, size, values); if(device) ALCdevice_DecRef(device); } ALC_API void ALC_APIENTRY alcGetInteger64vSOFT(ALCdevice *device, ALCenum pname, ALCsizei size, ALCint64SOFT *values) { ALCint *ivals; ALsizei i; VerifyDevice(&device); if(size <= 0 || values == NULL) alcSetError(device, ALC_INVALID_VALUE); else if(!device || device->Type == Capture) { ivals = malloc(size * sizeof(ALCint)); size = GetIntegerv(device, pname, size, ivals); for(i = 0;i < size;i++) values[i] = ivals[i]; free(ivals); } else /* render device */ { ClockLatency clock; ALuint64 basecount; ALuint samplecount; ALuint refcount; switch(pname) { case ALC_ATTRIBUTES_SIZE: *values = NumAttrsForDevice(device)+4; break; case ALC_ALL_ATTRIBUTES: if(size < NumAttrsForDevice(device)+4) alcSetError(device, ALC_INVALID_VALUE); else { i = 0; almtx_lock(&device->BackendLock); values[i++] = ALC_FREQUENCY; values[i++] = device->Frequency; if(device->Type != Loopback) { values[i++] = ALC_REFRESH; values[i++] = device->Frequency / device->UpdateSize; values[i++] = ALC_SYNC; values[i++] = ALC_FALSE; } else { if(device->FmtChans == DevFmtAmbi3D) { values[i++] = ALC_AMBISONIC_LAYOUT_SOFT; values[i++] = device->AmbiLayout; values[i++] = ALC_AMBISONIC_SCALING_SOFT; values[i++] = device->AmbiScale; values[i++] = ALC_AMBISONIC_ORDER_SOFT; values[i++] = device->AmbiOrder; } values[i++] = ALC_FORMAT_CHANNELS_SOFT; values[i++] = device->FmtChans; values[i++] = ALC_FORMAT_TYPE_SOFT; values[i++] = device->FmtType; } values[i++] = ALC_MONO_SOURCES; values[i++] = device->NumMonoSources; values[i++] = ALC_STEREO_SOURCES; values[i++] = device->NumStereoSources; values[i++] = ALC_MAX_AUXILIARY_SENDS; values[i++] = device->NumAuxSends; values[i++] = ALC_HRTF_SOFT; values[i++] = (device->HrtfHandle ? ALC_TRUE : ALC_FALSE); values[i++] = ALC_HRTF_STATUS_SOFT; values[i++] = device->HrtfStatus; values[i++] = ALC_OUTPUT_LIMITER_SOFT; values[i++] = device->Limiter ? ALC_TRUE : ALC_FALSE; clock = GetClockLatency(device); values[i++] = ALC_DEVICE_CLOCK_SOFT; values[i++] = clock.ClockTime; values[i++] = ALC_DEVICE_LATENCY_SOFT; values[i++] = clock.Latency; almtx_unlock(&device->BackendLock); values[i++] = 0; } break; case ALC_DEVICE_CLOCK_SOFT: almtx_lock(&device->BackendLock); do { while(((refcount=ReadRef(&device->MixCount))&1) != 0) althrd_yield(); basecount = device->ClockBase; samplecount = device->SamplesDone; } while(refcount != ReadRef(&device->MixCount)); *values = basecount + (samplecount*DEVICE_CLOCK_RES/device->Frequency); almtx_unlock(&device->BackendLock); break; case ALC_DEVICE_LATENCY_SOFT: almtx_lock(&device->BackendLock); clock = GetClockLatency(device); almtx_unlock(&device->BackendLock); *values = clock.Latency; break; case ALC_DEVICE_CLOCK_LATENCY_SOFT: if(size < 2) alcSetError(device, ALC_INVALID_VALUE); else { almtx_lock(&device->BackendLock); clock = GetClockLatency(device); almtx_unlock(&device->BackendLock); values[0] = clock.ClockTime; values[1] = clock.Latency; } break; default: ivals = malloc(size * sizeof(ALCint)); size = GetIntegerv(device, pname, size, ivals); for(i = 0;i < size;i++) values[i] = ivals[i]; free(ivals); break; } } if(device) ALCdevice_DecRef(device); } /* alcIsExtensionPresent * * Determines if there is support for a particular extension */ ALC_API ALCboolean ALC_APIENTRY alcIsExtensionPresent(ALCdevice *device, const ALCchar *extName) { ALCboolean bResult = ALC_FALSE; VerifyDevice(&device); if(!extName) alcSetError(device, ALC_INVALID_VALUE); else { size_t len = strlen(extName); const char *ptr = (device ? alcExtensionList : alcNoDeviceExtList); while(ptr && *ptr) { if(strncasecmp(ptr, extName, len) == 0 && (ptr[len] == '\0' || isspace(ptr[len]))) { bResult = ALC_TRUE; break; } if((ptr=strchr(ptr, ' ')) != NULL) { do { ++ptr; } while(isspace(*ptr)); } } } if(device) ALCdevice_DecRef(device); return bResult; } /* alcGetProcAddress * * Retrieves the function address for a particular extension function */ ALC_API ALCvoid* ALC_APIENTRY alcGetProcAddress(ALCdevice *device, const ALCchar *funcName) { ALCvoid *ptr = NULL; if(!funcName) { VerifyDevice(&device); alcSetError(device, ALC_INVALID_VALUE); if(device) ALCdevice_DecRef(device); } else { size_t i = 0; for(i = 0;i < COUNTOF(alcFunctions);i++) { if(strcmp(alcFunctions[i].funcName, funcName) == 0) { ptr = alcFunctions[i].address; break; } } } return ptr; } /* alcGetEnumValue * * Get the value for a particular ALC enumeration name */ ALC_API ALCenum ALC_APIENTRY alcGetEnumValue(ALCdevice *device, const ALCchar *enumName) { ALCenum val = 0; if(!enumName) { VerifyDevice(&device); alcSetError(device, ALC_INVALID_VALUE); if(device) ALCdevice_DecRef(device); } else { size_t i = 0; for(i = 0;i < COUNTOF(alcEnumerations);i++) { if(strcmp(alcEnumerations[i].enumName, enumName) == 0) { val = alcEnumerations[i].value; break; } } } return val; } /* alcCreateContext * * Create and attach a context to the given device. */ ALC_API ALCcontext* ALC_APIENTRY alcCreateContext(ALCdevice *device, const ALCint *attrList) { ALCcontext *ALContext; ALfloat valf; ALCenum err; /* Explicitly hold the list lock while taking the BackendLock in case the * device is asynchronously destropyed, to ensure this new context is * properly cleaned up after being made. */ LockLists(); if(!VerifyDevice(&device) || device->Type == Capture || !ATOMIC_LOAD(&device->Connected, almemory_order_relaxed)) { UnlockLists(); alcSetError(device, ALC_INVALID_DEVICE); if(device) ALCdevice_DecRef(device); return NULL; } almtx_lock(&device->BackendLock); UnlockLists(); ATOMIC_STORE_SEQ(&device->LastError, ALC_NO_ERROR); if(device->Type == Playback && DefaultEffect.type != AL_EFFECT_NULL) ALContext = al_calloc(16, sizeof(ALCcontext)+sizeof(ALlistener)+sizeof(ALeffectslot)); else ALContext = al_calloc(16, sizeof(ALCcontext)+sizeof(ALlistener)); if(!ALContext) { almtx_unlock(&device->BackendLock); alcSetError(device, ALC_OUT_OF_MEMORY); ALCdevice_DecRef(device); return NULL; } InitRef(&ALContext->ref, 1); ALContext->Listener = (ALlistener*)ALContext->_listener_mem; ALContext->DefaultSlot = NULL; ALContext->Voices = NULL; ALContext->VoiceCount = 0; ALContext->MaxVoices = 0; ATOMIC_INIT(&ALContext->ActiveAuxSlots, NULL); ALContext->Device = device; ATOMIC_INIT(&ALContext->next, NULL); if((err=UpdateDeviceParams(device, attrList)) != ALC_NO_ERROR) { almtx_unlock(&device->BackendLock); al_free(ALContext); ALContext = NULL; alcSetError(device, err); if(err == ALC_INVALID_DEVICE) { V0(device->Backend,lock)(); aluHandleDisconnect(device, "Device update failure"); V0(device->Backend,unlock)(); } ALCdevice_DecRef(device); return NULL; } AllocateVoices(ALContext, 256, device->NumAuxSends); if(DefaultEffect.type != AL_EFFECT_NULL && device->Type == Playback) { ALContext->DefaultSlot = (ALeffectslot*)(ALContext->_listener_mem + sizeof(ALlistener)); if(InitEffectSlot(ALContext->DefaultSlot) == AL_NO_ERROR) aluInitEffectPanning(ALContext->DefaultSlot); else { ALContext->DefaultSlot = NULL; ERR("Failed to initialize the default effect slot\n"); } } ALCdevice_IncRef(ALContext->Device); InitContext(ALContext); if(ConfigValueFloat(alstr_get_cstr(device->DeviceName), NULL, "volume-adjust", &valf)) { if(!isfinite(valf)) ERR("volume-adjust must be finite: %f\n", valf); else { ALfloat db = clampf(valf, -24.0f, 24.0f); if(db != valf) WARN("volume-adjust clamped: %f, range: +/-%f\n", valf, 24.0f); ALContext->GainBoost = powf(10.0f, db/20.0f); TRACE("volume-adjust gain: %f\n", ALContext->GainBoost); } } UpdateListenerProps(ALContext); { ALCcontext *head = ATOMIC_LOAD_SEQ(&device->ContextList); do { ATOMIC_STORE(&ALContext->next, head, almemory_order_relaxed); } while(ATOMIC_COMPARE_EXCHANGE_PTR_WEAK_SEQ(&device->ContextList, &head, ALContext) == 0); } almtx_unlock(&device->BackendLock); if(ALContext->DefaultSlot) { if(InitializeEffect(ALContext, ALContext->DefaultSlot, &DefaultEffect) == AL_NO_ERROR) UpdateEffectSlotProps(ALContext->DefaultSlot, ALContext); else ERR("Failed to initialize the default effect\n"); } ALCdevice_DecRef(device); TRACE("Created context %p\n", ALContext); return ALContext; } /* alcDestroyContext * * Remove a context from its device */ ALC_API ALCvoid ALC_APIENTRY alcDestroyContext(ALCcontext *context) { ALCdevice *Device; LockLists(); if(!VerifyContext(&context)) { UnlockLists(); alcSetError(NULL, ALC_INVALID_CONTEXT); return; } Device = context->Device; if(Device) { almtx_lock(&Device->BackendLock); if(!ReleaseContext(context, Device)) { V0(Device->Backend,stop)(); Device->Flags &= ~DEVICE_RUNNING; } almtx_unlock(&Device->BackendLock); } UnlockLists(); ALCcontext_DecRef(context); } /* alcGetCurrentContext * * Returns the currently active context on the calling thread */ ALC_API ALCcontext* ALC_APIENTRY alcGetCurrentContext(void) { ALCcontext *Context = altss_get(LocalContext); if(!Context) Context = ATOMIC_LOAD_SEQ(&GlobalContext); return Context; } /* alcGetThreadContext * * Returns the currently active thread-local context */ ALC_API ALCcontext* ALC_APIENTRY alcGetThreadContext(void) { return altss_get(LocalContext); } /* alcMakeContextCurrent * * Makes the given context the active process-wide context, and removes the * thread-local context for the calling thread. */ ALC_API ALCboolean ALC_APIENTRY alcMakeContextCurrent(ALCcontext *context) { /* context must be valid or NULL */ if(context && !VerifyContext(&context)) { alcSetError(NULL, ALC_INVALID_CONTEXT); return ALC_FALSE; } /* context's reference count is already incremented */ context = ATOMIC_EXCHANGE_PTR_SEQ(&GlobalContext, context); if(context) ALCcontext_DecRef(context); if((context=altss_get(LocalContext)) != NULL) { altss_set(LocalContext, NULL); ALCcontext_DecRef(context); } return ALC_TRUE; } /* alcSetThreadContext * * Makes the given context the active context for the current thread */ ALC_API ALCboolean ALC_APIENTRY alcSetThreadContext(ALCcontext *context) { ALCcontext *old; /* context must be valid or NULL */ if(context && !VerifyContext(&context)) { alcSetError(NULL, ALC_INVALID_CONTEXT); return ALC_FALSE; } /* context's reference count is already incremented */ old = altss_get(LocalContext); altss_set(LocalContext, context); if(old) ALCcontext_DecRef(old); return ALC_TRUE; } /* alcGetContextsDevice * * Returns the device that a particular context is attached to */ ALC_API ALCdevice* ALC_APIENTRY alcGetContextsDevice(ALCcontext *Context) { ALCdevice *Device; if(!VerifyContext(&Context)) { alcSetError(NULL, ALC_INVALID_CONTEXT); return NULL; } Device = Context->Device; ALCcontext_DecRef(Context); return Device; } /* alcOpenDevice * * Opens the named device. */ ALC_API ALCdevice* ALC_APIENTRY alcOpenDevice(const ALCchar *deviceName) { ALCbackendFactory *factory; const ALCchar *fmt; ALCdevice *device; ALCenum err; DO_INITCONFIG(); if(!PlaybackBackend.name) { alcSetError(NULL, ALC_INVALID_VALUE); return NULL; } if(deviceName && (!deviceName[0] || strcasecmp(deviceName, alcDefaultName) == 0 || strcasecmp(deviceName, "openal-soft") == 0 #ifdef _WIN32 /* Some old Windows apps hardcode these expecting OpenAL to use a * specific audio API, even when they're not enumerated. Creative's * router effectively ignores them too. */ || strcasecmp(deviceName, "DirectSound3D") == 0 || strcasecmp(deviceName, "DirectSound") == 0 || strcasecmp(deviceName, "MMSYSTEM") == 0 #endif )) deviceName = NULL; device = al_calloc(16, sizeof(ALCdevice)); if(!device) { alcSetError(NULL, ALC_OUT_OF_MEMORY); return NULL; } //Validate device InitDevice(device, Playback); //Set output format device->FmtChans = DevFmtChannelsDefault; device->FmtType = DevFmtTypeDefault; device->Frequency = DEFAULT_OUTPUT_RATE; device->IsHeadphones = AL_FALSE; device->AmbiLayout = AmbiLayout_Default; device->AmbiScale = AmbiNorm_Default; device->LimiterState = ALC_TRUE; device->NumUpdates = 3; device->UpdateSize = 1024; device->SourcesMax = 256; device->AuxiliaryEffectSlotMax = 64; device->NumAuxSends = DEFAULT_SENDS; if(ConfigValueStr(deviceName, NULL, "channels", &fmt)) { static const struct { const char name[16]; enum DevFmtChannels chans; ALsizei order; } chanlist[] = { { "mono", DevFmtMono, 0 }, { "stereo", DevFmtStereo, 0 }, { "quad", DevFmtQuad, 0 }, { "surround51", DevFmtX51, 0 }, { "surround61", DevFmtX61, 0 }, { "surround71", DevFmtX71, 0 }, { "surround51rear", DevFmtX51Rear, 0 }, { "ambi1", DevFmtAmbi3D, 1 }, { "ambi2", DevFmtAmbi3D, 2 }, { "ambi3", DevFmtAmbi3D, 3 }, }; size_t i; for(i = 0;i < COUNTOF(chanlist);i++) { if(strcasecmp(chanlist[i].name, fmt) == 0) { device->FmtChans = chanlist[i].chans; device->AmbiOrder = chanlist[i].order; device->Flags |= DEVICE_CHANNELS_REQUEST; break; } } if(i == COUNTOF(chanlist)) ERR("Unsupported channels: %s\n", fmt); } if(ConfigValueStr(deviceName, NULL, "sample-type", &fmt)) { static const struct { const char name[16]; enum DevFmtType type; } typelist[] = { { "int8", DevFmtByte }, { "uint8", DevFmtUByte }, { "int16", DevFmtShort }, { "uint16", DevFmtUShort }, { "int32", DevFmtInt }, { "uint32", DevFmtUInt }, { "float32", DevFmtFloat }, }; size_t i; for(i = 0;i < COUNTOF(typelist);i++) { if(strcasecmp(typelist[i].name, fmt) == 0) { device->FmtType = typelist[i].type; device->Flags |= DEVICE_SAMPLE_TYPE_REQUEST; break; } } if(i == COUNTOF(typelist)) ERR("Unsupported sample-type: %s\n", fmt); } if(ConfigValueUInt(deviceName, NULL, "frequency", &device->Frequency)) { device->Flags |= DEVICE_FREQUENCY_REQUEST; if(device->Frequency < MIN_OUTPUT_RATE) ERR("%uhz request clamped to %uhz minimum\n", device->Frequency, MIN_OUTPUT_RATE); device->Frequency = maxu(device->Frequency, MIN_OUTPUT_RATE); } ConfigValueUInt(deviceName, NULL, "periods", &device->NumUpdates); device->NumUpdates = clampu(device->NumUpdates, 2, 16); ConfigValueUInt(deviceName, NULL, "period_size", &device->UpdateSize); device->UpdateSize = clampu(device->UpdateSize, 64, 8192); if((CPUCapFlags&(CPU_CAP_SSE|CPU_CAP_NEON)) != 0) device->UpdateSize = (device->UpdateSize+3)&~3; ConfigValueUInt(deviceName, NULL, "sources", &device->SourcesMax); if(device->SourcesMax == 0) device->SourcesMax = 256; ConfigValueUInt(deviceName, NULL, "slots", &device->AuxiliaryEffectSlotMax); if(device->AuxiliaryEffectSlotMax == 0) device->AuxiliaryEffectSlotMax = 64; else device->AuxiliaryEffectSlotMax = minu(device->AuxiliaryEffectSlotMax, INT_MAX); if(ConfigValueInt(deviceName, NULL, "sends", &device->NumAuxSends)) device->NumAuxSends = clampi( DEFAULT_SENDS, 0, clampi(device->NumAuxSends, 0, MAX_SENDS) ); device->NumStereoSources = 1; device->NumMonoSources = device->SourcesMax - device->NumStereoSources; factory = PlaybackBackend.getFactory(); device->Backend = V(factory,createBackend)(device, ALCbackend_Playback); if(!device->Backend) { FreeDevice(device); alcSetError(NULL, ALC_OUT_OF_MEMORY); return NULL; } // Find a playback device to open if((err=V(device->Backend,open)(deviceName)) != ALC_NO_ERROR) { FreeDevice(device); alcSetError(NULL, err); return NULL; } if(ConfigValueStr(alstr_get_cstr(device->DeviceName), NULL, "ambi-format", &fmt)) { if(strcasecmp(fmt, "fuma") == 0) { device->AmbiLayout = AmbiLayout_FuMa; device->AmbiScale = AmbiNorm_FuMa; } else if(strcasecmp(fmt, "acn+sn3d") == 0) { device->AmbiLayout = AmbiLayout_ACN; device->AmbiScale = AmbiNorm_SN3D; } else if(strcasecmp(fmt, "acn+n3d") == 0) { device->AmbiLayout = AmbiLayout_ACN; device->AmbiScale = AmbiNorm_N3D; } else ERR("Unsupported ambi-format: %s\n", fmt); } { ALCdevice *head = ATOMIC_LOAD_SEQ(&DeviceList); do { ATOMIC_STORE(&device->next, head, almemory_order_relaxed); } while(!ATOMIC_COMPARE_EXCHANGE_PTR_WEAK_SEQ(&DeviceList, &head, device)); } TRACE("Created device %p, \"%s\"\n", device, alstr_get_cstr(device->DeviceName)); return device; } /* alcCloseDevice * * Closes the given device. */ ALC_API ALCboolean ALC_APIENTRY alcCloseDevice(ALCdevice *device) { ALCdevice *iter, *origdev, *nextdev; ALCcontext *ctx; LockLists(); iter = ATOMIC_LOAD_SEQ(&DeviceList); do { if(iter == device) break; iter = ATOMIC_LOAD(&iter->next, almemory_order_relaxed); } while(iter != NULL); if(!iter || iter->Type == Capture) { alcSetError(iter, ALC_INVALID_DEVICE); UnlockLists(); return ALC_FALSE; } almtx_lock(&device->BackendLock); origdev = device; nextdev = ATOMIC_LOAD(&device->next, almemory_order_relaxed); if(!ATOMIC_COMPARE_EXCHANGE_PTR_STRONG_SEQ(&DeviceList, &origdev, nextdev)) { ALCdevice *list; do { list = origdev; origdev = device; } while(!ATOMIC_COMPARE_EXCHANGE_PTR_STRONG_SEQ(&list->next, &origdev, nextdev)); } UnlockLists(); ctx = ATOMIC_LOAD_SEQ(&device->ContextList); while(ctx != NULL) { ALCcontext *next = ATOMIC_LOAD(&ctx->next, almemory_order_relaxed); WARN("Releasing context %p\n", ctx); ReleaseContext(ctx, device); ctx = next; } if((device->Flags&DEVICE_RUNNING)) V0(device->Backend,stop)(); device->Flags &= ~DEVICE_RUNNING; almtx_unlock(&device->BackendLock); ALCdevice_DecRef(device); return ALC_TRUE; } /************************************************ * ALC capture functions ************************************************/ ALC_API ALCdevice* ALC_APIENTRY alcCaptureOpenDevice(const ALCchar *deviceName, ALCuint frequency, ALCenum format, ALCsizei samples) { ALCbackendFactory *factory; ALCdevice *device = NULL; ALCenum err; DO_INITCONFIG(); if(!CaptureBackend.name) { alcSetError(NULL, ALC_INVALID_VALUE); return NULL; } if(samples <= 0) { alcSetError(NULL, ALC_INVALID_VALUE); return NULL; } if(deviceName && (!deviceName[0] || strcasecmp(deviceName, alcDefaultName) == 0 || strcasecmp(deviceName, "openal-soft") == 0)) deviceName = NULL; device = al_calloc(16, sizeof(ALCdevice)); if(!device) { alcSetError(NULL, ALC_OUT_OF_MEMORY); return NULL; } //Validate device InitDevice(device, Capture); device->Frequency = frequency; device->Flags |= DEVICE_FREQUENCY_REQUEST; if(DecomposeDevFormat(format, &device->FmtChans, &device->FmtType) == AL_FALSE) { FreeDevice(device); alcSetError(NULL, ALC_INVALID_ENUM); return NULL; } device->Flags |= DEVICE_CHANNELS_REQUEST | DEVICE_SAMPLE_TYPE_REQUEST; device->IsHeadphones = AL_FALSE; device->AmbiOrder = 0; device->AmbiLayout = AmbiLayout_Default; device->AmbiScale = AmbiNorm_Default; device->UpdateSize = samples; device->NumUpdates = 1; factory = CaptureBackend.getFactory(); device->Backend = V(factory,createBackend)(device, ALCbackend_Capture); if(!device->Backend) { FreeDevice(device); alcSetError(NULL, ALC_OUT_OF_MEMORY); return NULL; } TRACE("Capture format: %s, %s, %uhz, %u update size x%d\n", DevFmtChannelsString(device->FmtChans), DevFmtTypeString(device->FmtType), device->Frequency, device->UpdateSize, device->NumUpdates ); if((err=V(device->Backend,open)(deviceName)) != ALC_NO_ERROR) { FreeDevice(device); alcSetError(NULL, err); return NULL; } { ALCdevice *head = ATOMIC_LOAD_SEQ(&DeviceList); do { ATOMIC_STORE(&device->next, head, almemory_order_relaxed); } while(!ATOMIC_COMPARE_EXCHANGE_PTR_WEAK_SEQ(&DeviceList, &head, device)); } TRACE("Created device %p, \"%s\"\n", device, alstr_get_cstr(device->DeviceName)); return device; } ALC_API ALCboolean ALC_APIENTRY alcCaptureCloseDevice(ALCdevice *device) { ALCdevice *iter, *origdev, *nextdev; LockLists(); iter = ATOMIC_LOAD_SEQ(&DeviceList); do { if(iter == device) break; iter = ATOMIC_LOAD(&iter->next, almemory_order_relaxed); } while(iter != NULL); if(!iter || iter->Type != Capture) { alcSetError(iter, ALC_INVALID_DEVICE); UnlockLists(); return ALC_FALSE; } origdev = device; nextdev = ATOMIC_LOAD(&device->next, almemory_order_relaxed); if(!ATOMIC_COMPARE_EXCHANGE_PTR_STRONG_SEQ(&DeviceList, &origdev, nextdev)) { ALCdevice *list; do { list = origdev; origdev = device; } while(!ATOMIC_COMPARE_EXCHANGE_PTR_STRONG_SEQ(&list->next, &origdev, nextdev)); } UnlockLists(); almtx_lock(&device->BackendLock); if((device->Flags&DEVICE_RUNNING)) V0(device->Backend,stop)(); device->Flags &= ~DEVICE_RUNNING; almtx_unlock(&device->BackendLock); ALCdevice_DecRef(device); return ALC_TRUE; } ALC_API void ALC_APIENTRY alcCaptureStart(ALCdevice *device) { if(!VerifyDevice(&device) || device->Type != Capture) alcSetError(device, ALC_INVALID_DEVICE); else { almtx_lock(&device->BackendLock); if(!ATOMIC_LOAD(&device->Connected, almemory_order_acquire)) alcSetError(device, ALC_INVALID_DEVICE); else if(!(device->Flags&DEVICE_RUNNING)) { if(V0(device->Backend,start)()) device->Flags |= DEVICE_RUNNING; else { aluHandleDisconnect(device, "Device start failure"); alcSetError(device, ALC_INVALID_DEVICE); } } almtx_unlock(&device->BackendLock); } if(device) ALCdevice_DecRef(device); } ALC_API void ALC_APIENTRY alcCaptureStop(ALCdevice *device) { if(!VerifyDevice(&device) || device->Type != Capture) alcSetError(device, ALC_INVALID_DEVICE); else { almtx_lock(&device->BackendLock); if((device->Flags&DEVICE_RUNNING)) V0(device->Backend,stop)(); device->Flags &= ~DEVICE_RUNNING; almtx_unlock(&device->BackendLock); } if(device) ALCdevice_DecRef(device); } ALC_API void ALC_APIENTRY alcCaptureSamples(ALCdevice *device, ALCvoid *buffer, ALCsizei samples) { if(!VerifyDevice(&device) || device->Type != Capture) alcSetError(device, ALC_INVALID_DEVICE); else { ALCenum err = ALC_INVALID_VALUE; almtx_lock(&device->BackendLock); if(samples >= 0 && V0(device->Backend,availableSamples)() >= (ALCuint)samples) err = V(device->Backend,captureSamples)(buffer, samples); almtx_unlock(&device->BackendLock); if(err != ALC_NO_ERROR) alcSetError(device, err); } if(device) ALCdevice_DecRef(device); } /************************************************ * ALC loopback functions ************************************************/ /* alcLoopbackOpenDeviceSOFT * * Open a loopback device, for manual rendering. */ ALC_API ALCdevice* ALC_APIENTRY alcLoopbackOpenDeviceSOFT(const ALCchar *deviceName) { ALCbackendFactory *factory; ALCdevice *device; DO_INITCONFIG(); /* Make sure the device name, if specified, is us. */ if(deviceName && strcmp(deviceName, alcDefaultName) != 0) { alcSetError(NULL, ALC_INVALID_VALUE); return NULL; } device = al_calloc(16, sizeof(ALCdevice)); if(!device) { alcSetError(NULL, ALC_OUT_OF_MEMORY); return NULL; } //Validate device InitDevice(device, Loopback); device->SourcesMax = 256; device->AuxiliaryEffectSlotMax = 64; device->NumAuxSends = DEFAULT_SENDS; //Set output format device->NumUpdates = 0; device->UpdateSize = 0; device->Frequency = DEFAULT_OUTPUT_RATE; device->FmtChans = DevFmtChannelsDefault; device->FmtType = DevFmtTypeDefault; device->IsHeadphones = AL_FALSE; device->AmbiLayout = AmbiLayout_Default; device->AmbiScale = AmbiNorm_Default; ConfigValueUInt(NULL, NULL, "sources", &device->SourcesMax); if(device->SourcesMax == 0) device->SourcesMax = 256; ConfigValueUInt(NULL, NULL, "slots", &device->AuxiliaryEffectSlotMax); if(device->AuxiliaryEffectSlotMax == 0) device->AuxiliaryEffectSlotMax = 64; else device->AuxiliaryEffectSlotMax = minu(device->AuxiliaryEffectSlotMax, INT_MAX); if(ConfigValueInt(NULL, NULL, "sends", &device->NumAuxSends)) device->NumAuxSends = clampi( DEFAULT_SENDS, 0, clampi(device->NumAuxSends, 0, MAX_SENDS) ); device->NumStereoSources = 1; device->NumMonoSources = device->SourcesMax - device->NumStereoSources; factory = ALCloopbackFactory_getFactory(); device->Backend = V(factory,createBackend)(device, ALCbackend_Loopback); if(!device->Backend) { al_free(device); alcSetError(NULL, ALC_OUT_OF_MEMORY); return NULL; } // Open the "backend" V(device->Backend,open)("Loopback"); { ALCdevice *head = ATOMIC_LOAD_SEQ(&DeviceList); do { ATOMIC_STORE(&device->next, head, almemory_order_relaxed); } while(!ATOMIC_COMPARE_EXCHANGE_PTR_WEAK_SEQ(&DeviceList, &head, device)); } TRACE("Created device %p\n", device); return device; } /* alcIsRenderFormatSupportedSOFT * * Determines if the loopback device supports the given format for rendering. */ ALC_API ALCboolean ALC_APIENTRY alcIsRenderFormatSupportedSOFT(ALCdevice *device, ALCsizei freq, ALCenum channels, ALCenum type) { ALCboolean ret = ALC_FALSE; if(!VerifyDevice(&device) || device->Type != Loopback) alcSetError(device, ALC_INVALID_DEVICE); else if(freq <= 0) alcSetError(device, ALC_INVALID_VALUE); else { if(IsValidALCType(type) && IsValidALCChannels(channels) && freq >= MIN_OUTPUT_RATE) ret = ALC_TRUE; } if(device) ALCdevice_DecRef(device); return ret; } /* alcRenderSamplesSOFT * * Renders some samples into a buffer, using the format last set by the * attributes given to alcCreateContext. */ FORCE_ALIGN ALC_API void ALC_APIENTRY alcRenderSamplesSOFT(ALCdevice *device, ALCvoid *buffer, ALCsizei samples) { if(!VerifyDevice(&device) || device->Type != Loopback) alcSetError(device, ALC_INVALID_DEVICE); else if(samples < 0 || (samples > 0 && buffer == NULL)) alcSetError(device, ALC_INVALID_VALUE); else { V0(device->Backend,lock)(); aluMixData(device, buffer, samples); V0(device->Backend,unlock)(); } if(device) ALCdevice_DecRef(device); } /************************************************ * ALC DSP pause/resume functions ************************************************/ /* alcDevicePauseSOFT * * Pause the DSP to stop audio processing. */ ALC_API void ALC_APIENTRY alcDevicePauseSOFT(ALCdevice *device) { if(!VerifyDevice(&device) || device->Type != Playback) alcSetError(device, ALC_INVALID_DEVICE); else { almtx_lock(&device->BackendLock); if((device->Flags&DEVICE_RUNNING)) V0(device->Backend,stop)(); device->Flags &= ~DEVICE_RUNNING; device->Flags |= DEVICE_PAUSED; almtx_unlock(&device->BackendLock); } if(device) ALCdevice_DecRef(device); } /* alcDeviceResumeSOFT * * Resume the DSP to restart audio processing. */ ALC_API void ALC_APIENTRY alcDeviceResumeSOFT(ALCdevice *device) { if(!VerifyDevice(&device) || device->Type != Playback) alcSetError(device, ALC_INVALID_DEVICE); else { almtx_lock(&device->BackendLock); if((device->Flags&DEVICE_PAUSED)) { device->Flags &= ~DEVICE_PAUSED; if(ATOMIC_LOAD_SEQ(&device->ContextList) != NULL) { if(V0(device->Backend,start)() != ALC_FALSE) device->Flags |= DEVICE_RUNNING; else { V0(device->Backend,lock)(); aluHandleDisconnect(device, "Device start failure"); V0(device->Backend,unlock)(); alcSetError(device, ALC_INVALID_DEVICE); } } } almtx_unlock(&device->BackendLock); } if(device) ALCdevice_DecRef(device); } /************************************************ * ALC HRTF functions ************************************************/ /* alcGetStringiSOFT * * Gets a string parameter at the given index. */ ALC_API const ALCchar* ALC_APIENTRY alcGetStringiSOFT(ALCdevice *device, ALCenum paramName, ALCsizei index) { const ALCchar *str = NULL; if(!VerifyDevice(&device) || device->Type == Capture) alcSetError(device, ALC_INVALID_DEVICE); else switch(paramName) { case ALC_HRTF_SPECIFIER_SOFT: if(index >= 0 && (size_t)index < VECTOR_SIZE(device->HrtfList)) str = alstr_get_cstr(VECTOR_ELEM(device->HrtfList, index).name); else alcSetError(device, ALC_INVALID_VALUE); break; default: alcSetError(device, ALC_INVALID_ENUM); break; } if(device) ALCdevice_DecRef(device); return str; } /* alcResetDeviceSOFT * * Resets the given device output, using the specified attribute list. */ ALC_API ALCboolean ALC_APIENTRY alcResetDeviceSOFT(ALCdevice *device, const ALCint *attribs) { ALCenum err; LockLists(); if(!VerifyDevice(&device) || device->Type == Capture || !ATOMIC_LOAD(&device->Connected, almemory_order_relaxed)) { UnlockLists(); alcSetError(device, ALC_INVALID_DEVICE); if(device) ALCdevice_DecRef(device); return ALC_FALSE; } almtx_lock(&device->BackendLock); UnlockLists(); err = UpdateDeviceParams(device, attribs); almtx_unlock(&device->BackendLock); if(err != ALC_NO_ERROR) { alcSetError(device, err); if(err == ALC_INVALID_DEVICE) { V0(device->Backend,lock)(); aluHandleDisconnect(device, "Device start failure"); V0(device->Backend,unlock)(); } ALCdevice_DecRef(device); return ALC_FALSE; } ALCdevice_DecRef(device); return ALC_TRUE; } openal-soft-openal-soft-1.19.1/Alc/ALu.c000066400000000000000000002117731335774445300176350ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include #include #include "alMain.h" #include "alSource.h" #include "alBuffer.h" #include "alListener.h" #include "alAuxEffectSlot.h" #include "alu.h" #include "bs2b.h" #include "hrtf.h" #include "mastering.h" #include "uhjfilter.h" #include "bformatdec.h" #include "static_assert.h" #include "ringbuffer.h" #include "filters/splitter.h" #include "mixer/defs.h" #include "fpu_modes.h" #include "cpu_caps.h" #include "bsinc_inc.h" #include "backends/base.h" extern inline ALfloat minf(ALfloat a, ALfloat b); extern inline ALfloat maxf(ALfloat a, ALfloat b); extern inline ALfloat clampf(ALfloat val, ALfloat min, ALfloat max); extern inline ALdouble mind(ALdouble a, ALdouble b); extern inline ALdouble maxd(ALdouble a, ALdouble b); extern inline ALdouble clampd(ALdouble val, ALdouble min, ALdouble max); extern inline ALuint minu(ALuint a, ALuint b); extern inline ALuint maxu(ALuint a, ALuint b); extern inline ALuint clampu(ALuint val, ALuint min, ALuint max); extern inline ALint mini(ALint a, ALint b); extern inline ALint maxi(ALint a, ALint b); extern inline ALint clampi(ALint val, ALint min, ALint max); extern inline ALint64 mini64(ALint64 a, ALint64 b); extern inline ALint64 maxi64(ALint64 a, ALint64 b); extern inline ALint64 clampi64(ALint64 val, ALint64 min, ALint64 max); extern inline ALuint64 minu64(ALuint64 a, ALuint64 b); extern inline ALuint64 maxu64(ALuint64 a, ALuint64 b); extern inline ALuint64 clampu64(ALuint64 val, ALuint64 min, ALuint64 max); extern inline size_t minz(size_t a, size_t b); extern inline size_t maxz(size_t a, size_t b); extern inline size_t clampz(size_t val, size_t min, size_t max); extern inline ALfloat lerp(ALfloat val1, ALfloat val2, ALfloat mu); extern inline ALfloat cubic(ALfloat val1, ALfloat val2, ALfloat val3, ALfloat val4, ALfloat mu); extern inline void aluVectorSet(aluVector *restrict vector, ALfloat x, ALfloat y, ALfloat z, ALfloat w); extern inline void aluMatrixfSetRow(aluMatrixf *matrix, ALuint row, ALfloat m0, ALfloat m1, ALfloat m2, ALfloat m3); extern inline void aluMatrixfSet(aluMatrixf *matrix, ALfloat m00, ALfloat m01, ALfloat m02, ALfloat m03, ALfloat m10, ALfloat m11, ALfloat m12, ALfloat m13, ALfloat m20, ALfloat m21, ALfloat m22, ALfloat m23, ALfloat m30, ALfloat m31, ALfloat m32, ALfloat m33); /* Cone scalar */ ALfloat ConeScale = 1.0f; /* Localized Z scalar for mono sources */ ALfloat ZScale = 1.0f; /* Force default speed of sound for distance-related reverb decay. */ ALboolean OverrideReverbSpeedOfSound = AL_FALSE; const aluMatrixf IdentityMatrixf = {{ { 1.0f, 0.0f, 0.0f, 0.0f }, { 0.0f, 1.0f, 0.0f, 0.0f }, { 0.0f, 0.0f, 1.0f, 0.0f }, { 0.0f, 0.0f, 0.0f, 1.0f }, }}; static void ClearArray(ALfloat f[MAX_OUTPUT_CHANNELS]) { size_t i; for(i = 0;i < MAX_OUTPUT_CHANNELS;i++) f[i] = 0.0f; } struct ChanMap { enum Channel channel; ALfloat angle; ALfloat elevation; }; static HrtfDirectMixerFunc MixDirectHrtf = MixDirectHrtf_C; void DeinitVoice(ALvoice *voice) { al_free(ATOMIC_EXCHANGE_PTR_SEQ(&voice->Update, NULL)); } static inline HrtfDirectMixerFunc SelectHrtfMixer(void) { #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return MixDirectHrtf_Neon; #endif #ifdef HAVE_SSE if((CPUCapFlags&CPU_CAP_SSE)) return MixDirectHrtf_SSE; #endif return MixDirectHrtf_C; } /* This RNG method was created based on the math found in opusdec. It's quick, * and starting with a seed value of 22222, is suitable for generating * whitenoise. */ static inline ALuint dither_rng(ALuint *seed) { *seed = (*seed * 96314165) + 907633515; return *seed; } static inline void aluCrossproduct(const ALfloat *inVector1, const ALfloat *inVector2, ALfloat *outVector) { outVector[0] = inVector1[1]*inVector2[2] - inVector1[2]*inVector2[1]; outVector[1] = inVector1[2]*inVector2[0] - inVector1[0]*inVector2[2]; outVector[2] = inVector1[0]*inVector2[1] - inVector1[1]*inVector2[0]; } static inline ALfloat aluDotproduct(const aluVector *vec1, const aluVector *vec2) { return vec1->v[0]*vec2->v[0] + vec1->v[1]*vec2->v[1] + vec1->v[2]*vec2->v[2]; } static ALfloat aluNormalize(ALfloat *vec) { ALfloat length = sqrtf(vec[0]*vec[0] + vec[1]*vec[1] + vec[2]*vec[2]); if(length > FLT_EPSILON) { ALfloat inv_length = 1.0f/length; vec[0] *= inv_length; vec[1] *= inv_length; vec[2] *= inv_length; return length; } vec[0] = vec[1] = vec[2] = 0.0f; return 0.0f; } static void aluMatrixfFloat3(ALfloat *vec, ALfloat w, const aluMatrixf *mtx) { ALfloat v[4] = { vec[0], vec[1], vec[2], w }; vec[0] = v[0]*mtx->m[0][0] + v[1]*mtx->m[1][0] + v[2]*mtx->m[2][0] + v[3]*mtx->m[3][0]; vec[1] = v[0]*mtx->m[0][1] + v[1]*mtx->m[1][1] + v[2]*mtx->m[2][1] + v[3]*mtx->m[3][1]; vec[2] = v[0]*mtx->m[0][2] + v[1]*mtx->m[1][2] + v[2]*mtx->m[2][2] + v[3]*mtx->m[3][2]; } static aluVector aluMatrixfVector(const aluMatrixf *mtx, const aluVector *vec) { aluVector v; v.v[0] = vec->v[0]*mtx->m[0][0] + vec->v[1]*mtx->m[1][0] + vec->v[2]*mtx->m[2][0] + vec->v[3]*mtx->m[3][0]; v.v[1] = vec->v[0]*mtx->m[0][1] + vec->v[1]*mtx->m[1][1] + vec->v[2]*mtx->m[2][1] + vec->v[3]*mtx->m[3][1]; v.v[2] = vec->v[0]*mtx->m[0][2] + vec->v[1]*mtx->m[1][2] + vec->v[2]*mtx->m[2][2] + vec->v[3]*mtx->m[3][2]; v.v[3] = vec->v[0]*mtx->m[0][3] + vec->v[1]*mtx->m[1][3] + vec->v[2]*mtx->m[2][3] + vec->v[3]*mtx->m[3][3]; return v; } void aluInit(void) { MixDirectHrtf = SelectHrtfMixer(); } static void SendSourceStoppedEvent(ALCcontext *context, ALuint id) { AsyncEvent evt = ASYNC_EVENT(EventType_SourceStateChange); ALbitfieldSOFT enabledevt; size_t strpos; ALuint scale; enabledevt = ATOMIC_LOAD(&context->EnabledEvts, almemory_order_acquire); if(!(enabledevt&EventType_SourceStateChange)) return; evt.u.user.type = AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT; evt.u.user.id = id; evt.u.user.param = AL_STOPPED; /* Normally snprintf would be used, but this is called from the mixer and * that function's not real-time safe, so we have to construct it manually. */ strcpy(evt.u.user.msg, "Source ID "); strpos = 10; scale = 1000000000; while(scale > 0 && scale > id) scale /= 10; while(scale > 0) { evt.u.user.msg[strpos++] = '0' + ((id/scale)%10); scale /= 10; } strcpy(evt.u.user.msg+strpos, " state changed to AL_STOPPED"); if(ll_ringbuffer_write(context->AsyncEvents, (const char*)&evt, 1) == 1) alsem_post(&context->EventSem); } static void ProcessHrtf(ALCdevice *device, ALsizei SamplesToDo) { DirectHrtfState *state; int lidx, ridx; ALsizei c; if(device->AmbiUp) ambiup_process(device->AmbiUp, device->Dry.Buffer, device->Dry.NumChannels, device->FOAOut.Buffer, SamplesToDo ); lidx = GetChannelIdxByName(&device->RealOut, FrontLeft); ridx = GetChannelIdxByName(&device->RealOut, FrontRight); assert(lidx != -1 && ridx != -1); state = device->Hrtf; for(c = 0;c < device->Dry.NumChannels;c++) { MixDirectHrtf(device->RealOut.Buffer[lidx], device->RealOut.Buffer[ridx], device->Dry.Buffer[c], state->Offset, state->IrSize, state->Chan[c].Coeffs, state->Chan[c].Values, SamplesToDo ); } state->Offset += SamplesToDo; } static void ProcessAmbiDec(ALCdevice *device, ALsizei SamplesToDo) { if(device->Dry.Buffer != device->FOAOut.Buffer) bformatdec_upSample(device->AmbiDecoder, device->Dry.Buffer, device->FOAOut.Buffer, device->FOAOut.NumChannels, SamplesToDo ); bformatdec_process(device->AmbiDecoder, device->RealOut.Buffer, device->RealOut.NumChannels, device->Dry.Buffer, SamplesToDo ); } static void ProcessAmbiUp(ALCdevice *device, ALsizei SamplesToDo) { ambiup_process(device->AmbiUp, device->RealOut.Buffer, device->RealOut.NumChannels, device->FOAOut.Buffer, SamplesToDo ); } static void ProcessUhj(ALCdevice *device, ALsizei SamplesToDo) { int lidx = GetChannelIdxByName(&device->RealOut, FrontLeft); int ridx = GetChannelIdxByName(&device->RealOut, FrontRight); assert(lidx != -1 && ridx != -1); /* Encode to stereo-compatible 2-channel UHJ output. */ EncodeUhj2(device->Uhj_Encoder, device->RealOut.Buffer[lidx], device->RealOut.Buffer[ridx], device->Dry.Buffer, SamplesToDo ); } static void ProcessBs2b(ALCdevice *device, ALsizei SamplesToDo) { int lidx = GetChannelIdxByName(&device->RealOut, FrontLeft); int ridx = GetChannelIdxByName(&device->RealOut, FrontRight); assert(lidx != -1 && ridx != -1); /* Apply binaural/crossfeed filter */ bs2b_cross_feed(device->Bs2b, device->RealOut.Buffer[lidx], device->RealOut.Buffer[ridx], SamplesToDo); } void aluSelectPostProcess(ALCdevice *device) { if(device->HrtfHandle) device->PostProcess = ProcessHrtf; else if(device->AmbiDecoder) device->PostProcess = ProcessAmbiDec; else if(device->AmbiUp) device->PostProcess = ProcessAmbiUp; else if(device->Uhj_Encoder) device->PostProcess = ProcessUhj; else if(device->Bs2b) device->PostProcess = ProcessBs2b; else device->PostProcess = NULL; } /* Prepares the interpolator for a given rate (determined by increment). * * With a bit of work, and a trade of memory for CPU cost, this could be * modified for use with an interpolated increment for buttery-smooth pitch * changes. */ void BsincPrepare(const ALuint increment, BsincState *state, const BSincTable *table) { ALfloat sf = 0.0f; ALsizei si = BSINC_SCALE_COUNT-1; if(increment > FRACTIONONE) { sf = (ALfloat)FRACTIONONE / increment; sf = maxf(0.0f, (BSINC_SCALE_COUNT-1) * (sf-table->scaleBase) * table->scaleRange); si = float2int(sf); /* The interpolation factor is fit to this diagonally-symmetric curve * to reduce the transition ripple caused by interpolating different * scales of the sinc function. */ sf = 1.0f - cosf(asinf(sf - si)); } state->sf = sf; state->m = table->m[si]; state->l = (state->m/2) - 1; state->filter = table->Tab + table->filterOffset[si]; } static bool CalcContextParams(ALCcontext *Context) { ALlistener *Listener = Context->Listener; struct ALcontextProps *props; props = ATOMIC_EXCHANGE_PTR(&Context->Update, NULL, almemory_order_acq_rel); if(!props) return false; Listener->Params.MetersPerUnit = props->MetersPerUnit; Listener->Params.DopplerFactor = props->DopplerFactor; Listener->Params.SpeedOfSound = props->SpeedOfSound * props->DopplerVelocity; if(!OverrideReverbSpeedOfSound) Listener->Params.ReverbSpeedOfSound = Listener->Params.SpeedOfSound * Listener->Params.MetersPerUnit; Listener->Params.SourceDistanceModel = props->SourceDistanceModel; Listener->Params.DistanceModel = props->DistanceModel; ATOMIC_REPLACE_HEAD(struct ALcontextProps*, &Context->FreeContextProps, props); return true; } static bool CalcListenerParams(ALCcontext *Context) { ALlistener *Listener = Context->Listener; ALfloat N[3], V[3], U[3], P[3]; struct ALlistenerProps *props; aluVector vel; props = ATOMIC_EXCHANGE_PTR(&Listener->Update, NULL, almemory_order_acq_rel); if(!props) return false; /* AT then UP */ N[0] = props->Forward[0]; N[1] = props->Forward[1]; N[2] = props->Forward[2]; aluNormalize(N); V[0] = props->Up[0]; V[1] = props->Up[1]; V[2] = props->Up[2]; aluNormalize(V); /* Build and normalize right-vector */ aluCrossproduct(N, V, U); aluNormalize(U); aluMatrixfSet(&Listener->Params.Matrix, U[0], V[0], -N[0], 0.0, U[1], V[1], -N[1], 0.0, U[2], V[2], -N[2], 0.0, 0.0, 0.0, 0.0, 1.0 ); P[0] = props->Position[0]; P[1] = props->Position[1]; P[2] = props->Position[2]; aluMatrixfFloat3(P, 1.0, &Listener->Params.Matrix); aluMatrixfSetRow(&Listener->Params.Matrix, 3, -P[0], -P[1], -P[2], 1.0f); aluVectorSet(&vel, props->Velocity[0], props->Velocity[1], props->Velocity[2], 0.0f); Listener->Params.Velocity = aluMatrixfVector(&Listener->Params.Matrix, &vel); Listener->Params.Gain = props->Gain * Context->GainBoost; ATOMIC_REPLACE_HEAD(struct ALlistenerProps*, &Context->FreeListenerProps, props); return true; } static bool CalcEffectSlotParams(ALeffectslot *slot, ALCcontext *context, bool force) { struct ALeffectslotProps *props; ALeffectState *state; props = ATOMIC_EXCHANGE_PTR(&slot->Update, NULL, almemory_order_acq_rel); if(!props && !force) return false; if(props) { slot->Params.Gain = props->Gain; slot->Params.AuxSendAuto = props->AuxSendAuto; slot->Params.EffectType = props->Type; slot->Params.EffectProps = props->Props; if(IsReverbEffect(props->Type)) { slot->Params.RoomRolloff = props->Props.Reverb.RoomRolloffFactor; slot->Params.DecayTime = props->Props.Reverb.DecayTime; slot->Params.DecayLFRatio = props->Props.Reverb.DecayLFRatio; slot->Params.DecayHFRatio = props->Props.Reverb.DecayHFRatio; slot->Params.DecayHFLimit = props->Props.Reverb.DecayHFLimit; slot->Params.AirAbsorptionGainHF = props->Props.Reverb.AirAbsorptionGainHF; } else { slot->Params.RoomRolloff = 0.0f; slot->Params.DecayTime = 0.0f; slot->Params.DecayLFRatio = 0.0f; slot->Params.DecayHFRatio = 0.0f; slot->Params.DecayHFLimit = AL_FALSE; slot->Params.AirAbsorptionGainHF = 1.0f; } state = props->State; if(state == slot->Params.EffectState) { /* If the effect state is the same as current, we can decrement its * count safely to remove it from the update object (it can't reach * 0 refs since the current params also hold a reference). */ DecrementRef(&state->Ref); props->State = NULL; } else { /* Otherwise, replace it and send off the old one with a release * event. */ AsyncEvent evt = ASYNC_EVENT(EventType_ReleaseEffectState); evt.u.EffectState = slot->Params.EffectState; slot->Params.EffectState = state; props->State = NULL; if(LIKELY(ll_ringbuffer_write(context->AsyncEvents, (const char*)&evt, 1) != 0)) alsem_post(&context->EventSem); else { /* If writing the event failed, the queue was probably full. * Store the old state in the property object where it can * eventually be cleaned up sometime later (not ideal, but * better than blocking or leaking). */ props->State = evt.u.EffectState; } } ATOMIC_REPLACE_HEAD(struct ALeffectslotProps*, &context->FreeEffectslotProps, props); } else state = slot->Params.EffectState; V(state,update)(context, slot, &slot->Params.EffectProps); return true; } static const struct ChanMap MonoMap[1] = { { FrontCenter, 0.0f, 0.0f } }, RearMap[2] = { { BackLeft, DEG2RAD(-150.0f), DEG2RAD(0.0f) }, { BackRight, DEG2RAD( 150.0f), DEG2RAD(0.0f) } }, QuadMap[4] = { { FrontLeft, DEG2RAD( -45.0f), DEG2RAD(0.0f) }, { FrontRight, DEG2RAD( 45.0f), DEG2RAD(0.0f) }, { BackLeft, DEG2RAD(-135.0f), DEG2RAD(0.0f) }, { BackRight, DEG2RAD( 135.0f), DEG2RAD(0.0f) } }, X51Map[6] = { { FrontLeft, DEG2RAD( -30.0f), DEG2RAD(0.0f) }, { FrontRight, DEG2RAD( 30.0f), DEG2RAD(0.0f) }, { FrontCenter, DEG2RAD( 0.0f), DEG2RAD(0.0f) }, { LFE, 0.0f, 0.0f }, { SideLeft, DEG2RAD(-110.0f), DEG2RAD(0.0f) }, { SideRight, DEG2RAD( 110.0f), DEG2RAD(0.0f) } }, X61Map[7] = { { FrontLeft, DEG2RAD(-30.0f), DEG2RAD(0.0f) }, { FrontRight, DEG2RAD( 30.0f), DEG2RAD(0.0f) }, { FrontCenter, DEG2RAD( 0.0f), DEG2RAD(0.0f) }, { LFE, 0.0f, 0.0f }, { BackCenter, DEG2RAD(180.0f), DEG2RAD(0.0f) }, { SideLeft, DEG2RAD(-90.0f), DEG2RAD(0.0f) }, { SideRight, DEG2RAD( 90.0f), DEG2RAD(0.0f) } }, X71Map[8] = { { FrontLeft, DEG2RAD( -30.0f), DEG2RAD(0.0f) }, { FrontRight, DEG2RAD( 30.0f), DEG2RAD(0.0f) }, { FrontCenter, DEG2RAD( 0.0f), DEG2RAD(0.0f) }, { LFE, 0.0f, 0.0f }, { BackLeft, DEG2RAD(-150.0f), DEG2RAD(0.0f) }, { BackRight, DEG2RAD( 150.0f), DEG2RAD(0.0f) }, { SideLeft, DEG2RAD( -90.0f), DEG2RAD(0.0f) }, { SideRight, DEG2RAD( 90.0f), DEG2RAD(0.0f) } }; static void CalcPanningAndFilters(ALvoice *voice, const ALfloat Azi, const ALfloat Elev, const ALfloat Distance, const ALfloat Spread, const ALfloat DryGain, const ALfloat DryGainHF, const ALfloat DryGainLF, const ALfloat *WetGain, const ALfloat *WetGainLF, const ALfloat *WetGainHF, ALeffectslot **SendSlots, const ALbuffer *Buffer, const struct ALvoiceProps *props, const ALlistener *Listener, const ALCdevice *Device) { struct ChanMap StereoMap[2] = { { FrontLeft, DEG2RAD(-30.0f), DEG2RAD(0.0f) }, { FrontRight, DEG2RAD( 30.0f), DEG2RAD(0.0f) } }; bool DirectChannels = props->DirectChannels; const ALsizei NumSends = Device->NumAuxSends; const ALuint Frequency = Device->Frequency; const struct ChanMap *chans = NULL; ALsizei num_channels = 0; bool isbformat = false; ALfloat downmix_gain = 1.0f; ALsizei c, i; switch(Buffer->FmtChannels) { case FmtMono: chans = MonoMap; num_channels = 1; /* Mono buffers are never played direct. */ DirectChannels = false; break; case FmtStereo: /* Convert counter-clockwise to clockwise. */ StereoMap[0].angle = -props->StereoPan[0]; StereoMap[1].angle = -props->StereoPan[1]; chans = StereoMap; num_channels = 2; downmix_gain = 1.0f / 2.0f; break; case FmtRear: chans = RearMap; num_channels = 2; downmix_gain = 1.0f / 2.0f; break; case FmtQuad: chans = QuadMap; num_channels = 4; downmix_gain = 1.0f / 4.0f; break; case FmtX51: chans = X51Map; num_channels = 6; /* NOTE: Excludes LFE. */ downmix_gain = 1.0f / 5.0f; break; case FmtX61: chans = X61Map; num_channels = 7; /* NOTE: Excludes LFE. */ downmix_gain = 1.0f / 6.0f; break; case FmtX71: chans = X71Map; num_channels = 8; /* NOTE: Excludes LFE. */ downmix_gain = 1.0f / 7.0f; break; case FmtBFormat2D: num_channels = 3; isbformat = true; DirectChannels = false; break; case FmtBFormat3D: num_channels = 4; isbformat = true; DirectChannels = false; break; } for(c = 0;c < num_channels;c++) { memset(&voice->Direct.Params[c].Hrtf.Target, 0, sizeof(voice->Direct.Params[c].Hrtf.Target)); ClearArray(voice->Direct.Params[c].Gains.Target); } for(i = 0;i < NumSends;i++) { for(c = 0;c < num_channels;c++) ClearArray(voice->Send[i].Params[c].Gains.Target); } voice->Flags &= ~(VOICE_HAS_HRTF | VOICE_HAS_NFC); if(isbformat) { /* Special handling for B-Format sources. */ if(Distance > FLT_EPSILON) { /* Panning a B-Format sound toward some direction is easy. Just pan * the first (W) channel as a normal mono sound and silence the * others. */ ALfloat coeffs[MAX_AMBI_COEFFS]; if(Device->AvgSpeakerDist > 0.0f) { ALfloat mdist = Distance * Listener->Params.MetersPerUnit; ALfloat w0 = SPEEDOFSOUNDMETRESPERSEC / (mdist * (ALfloat)Device->Frequency); ALfloat w1 = SPEEDOFSOUNDMETRESPERSEC / (Device->AvgSpeakerDist * (ALfloat)Device->Frequency); /* Clamp w0 for really close distances, to prevent excessive * bass. */ w0 = minf(w0, w1*4.0f); /* Only need to adjust the first channel of a B-Format source. */ NfcFilterAdjust(&voice->Direct.Params[0].NFCtrlFilter, w0); for(i = 0;i < MAX_AMBI_ORDER+1;i++) voice->Direct.ChannelsPerOrder[i] = Device->NumChannelsPerOrder[i]; voice->Flags |= VOICE_HAS_NFC; } /* A scalar of 1.5 for plain stereo results in +/-60 degrees being * moved to +/-90 degrees for direct right and left speaker * responses. */ CalcAngleCoeffs((Device->Render_Mode==StereoPair) ? ScaleAzimuthFront(Azi, 1.5f) : Azi, Elev, Spread, coeffs); /* NOTE: W needs to be scaled by sqrt(2) due to FuMa normalization. */ ComputePanGains(&Device->Dry, coeffs, DryGain*SQRTF_2, voice->Direct.Params[0].Gains.Target); for(i = 0;i < NumSends;i++) { const ALeffectslot *Slot = SendSlots[i]; if(Slot) ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels, coeffs, WetGain[i]*SQRTF_2, voice->Send[i].Params[0].Gains.Target ); } } else { /* Local B-Format sources have their XYZ channels rotated according * to the orientation. */ ALfloat N[3], V[3], U[3]; aluMatrixf matrix; if(Device->AvgSpeakerDist > 0.0f) { /* NOTE: The NFCtrlFilters were created with a w0 of 0, which * is what we want for FOA input. The first channel may have * been previously re-adjusted if panned, so reset it. */ NfcFilterAdjust(&voice->Direct.Params[0].NFCtrlFilter, 0.0f); voice->Direct.ChannelsPerOrder[0] = 1; voice->Direct.ChannelsPerOrder[1] = mini(voice->Direct.Channels-1, 3); for(i = 2;i < MAX_AMBI_ORDER+1;i++) voice->Direct.ChannelsPerOrder[i] = 0; voice->Flags |= VOICE_HAS_NFC; } /* AT then UP */ N[0] = props->Orientation[0][0]; N[1] = props->Orientation[0][1]; N[2] = props->Orientation[0][2]; aluNormalize(N); V[0] = props->Orientation[1][0]; V[1] = props->Orientation[1][1]; V[2] = props->Orientation[1][2]; aluNormalize(V); if(!props->HeadRelative) { const aluMatrixf *lmatrix = &Listener->Params.Matrix; aluMatrixfFloat3(N, 0.0f, lmatrix); aluMatrixfFloat3(V, 0.0f, lmatrix); } /* Build and normalize right-vector */ aluCrossproduct(N, V, U); aluNormalize(U); /* Build a rotate + conversion matrix (FuMa -> ACN+N3D). NOTE: This * matrix is transposed, for the inputs to align on the rows and * outputs on the columns. */ aluMatrixfSet(&matrix, // ACN0 ACN1 ACN2 ACN3 SQRTF_2, 0.0f, 0.0f, 0.0f, // Ambi W 0.0f, -N[0]*SQRTF_3, N[1]*SQRTF_3, -N[2]*SQRTF_3, // Ambi X 0.0f, U[0]*SQRTF_3, -U[1]*SQRTF_3, U[2]*SQRTF_3, // Ambi Y 0.0f, -V[0]*SQRTF_3, V[1]*SQRTF_3, -V[2]*SQRTF_3 // Ambi Z ); voice->Direct.Buffer = Device->FOAOut.Buffer; voice->Direct.Channels = Device->FOAOut.NumChannels; for(c = 0;c < num_channels;c++) ComputePanGains(&Device->FOAOut, matrix.m[c], DryGain, voice->Direct.Params[c].Gains.Target); for(i = 0;i < NumSends;i++) { const ALeffectslot *Slot = SendSlots[i]; if(Slot) { for(c = 0;c < num_channels;c++) ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels, matrix.m[c], WetGain[i], voice->Send[i].Params[c].Gains.Target ); } } } } else if(DirectChannels) { /* Direct source channels always play local. Skip the virtual channels * and write inputs to the matching real outputs. */ voice->Direct.Buffer = Device->RealOut.Buffer; voice->Direct.Channels = Device->RealOut.NumChannels; for(c = 0;c < num_channels;c++) { int idx = GetChannelIdxByName(&Device->RealOut, chans[c].channel); if(idx != -1) voice->Direct.Params[c].Gains.Target[idx] = DryGain; } /* Auxiliary sends still use normal channel panning since they mix to * B-Format, which can't channel-match. */ for(c = 0;c < num_channels;c++) { ALfloat coeffs[MAX_AMBI_COEFFS]; CalcAngleCoeffs(chans[c].angle, chans[c].elevation, 0.0f, coeffs); for(i = 0;i < NumSends;i++) { const ALeffectslot *Slot = SendSlots[i]; if(Slot) ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels, coeffs, WetGain[i], voice->Send[i].Params[c].Gains.Target ); } } } else if(Device->Render_Mode == HrtfRender) { /* Full HRTF rendering. Skip the virtual channels and render to the * real outputs. */ voice->Direct.Buffer = Device->RealOut.Buffer; voice->Direct.Channels = Device->RealOut.NumChannels; if(Distance > FLT_EPSILON) { ALfloat coeffs[MAX_AMBI_COEFFS]; /* Get the HRIR coefficients and delays just once, for the given * source direction. */ GetHrtfCoeffs(Device->HrtfHandle, Elev, Azi, Spread, voice->Direct.Params[0].Hrtf.Target.Coeffs, voice->Direct.Params[0].Hrtf.Target.Delay); voice->Direct.Params[0].Hrtf.Target.Gain = DryGain * downmix_gain; /* Remaining channels use the same results as the first. */ for(c = 1;c < num_channels;c++) { /* Skip LFE */ if(chans[c].channel != LFE) voice->Direct.Params[c].Hrtf.Target = voice->Direct.Params[0].Hrtf.Target; } /* Calculate the directional coefficients once, which apply to all * input channels of the source sends. */ CalcAngleCoeffs(Azi, Elev, Spread, coeffs); for(i = 0;i < NumSends;i++) { const ALeffectslot *Slot = SendSlots[i]; if(Slot) for(c = 0;c < num_channels;c++) { /* Skip LFE */ if(chans[c].channel != LFE) ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels, coeffs, WetGain[i] * downmix_gain, voice->Send[i].Params[c].Gains.Target ); } } } else { /* Local sources on HRTF play with each channel panned to its * relative location around the listener, providing "virtual * speaker" responses. */ for(c = 0;c < num_channels;c++) { ALfloat coeffs[MAX_AMBI_COEFFS]; if(chans[c].channel == LFE) { /* Skip LFE */ continue; } /* Get the HRIR coefficients and delays for this channel * position. */ GetHrtfCoeffs(Device->HrtfHandle, chans[c].elevation, chans[c].angle, Spread, voice->Direct.Params[c].Hrtf.Target.Coeffs, voice->Direct.Params[c].Hrtf.Target.Delay ); voice->Direct.Params[c].Hrtf.Target.Gain = DryGain; /* Normal panning for auxiliary sends. */ CalcAngleCoeffs(chans[c].angle, chans[c].elevation, Spread, coeffs); for(i = 0;i < NumSends;i++) { const ALeffectslot *Slot = SendSlots[i]; if(Slot) ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels, coeffs, WetGain[i], voice->Send[i].Params[c].Gains.Target ); } } } voice->Flags |= VOICE_HAS_HRTF; } else { /* Non-HRTF rendering. Use normal panning to the output. */ if(Distance > FLT_EPSILON) { ALfloat coeffs[MAX_AMBI_COEFFS]; ALfloat w0 = 0.0f; /* Calculate NFC filter coefficient if needed. */ if(Device->AvgSpeakerDist > 0.0f) { ALfloat mdist = Distance * Listener->Params.MetersPerUnit; ALfloat w1 = SPEEDOFSOUNDMETRESPERSEC / (Device->AvgSpeakerDist * (ALfloat)Device->Frequency); w0 = SPEEDOFSOUNDMETRESPERSEC / (mdist * (ALfloat)Device->Frequency); /* Clamp w0 for really close distances, to prevent excessive * bass. */ w0 = minf(w0, w1*4.0f); /* Adjust NFC filters. */ for(c = 0;c < num_channels;c++) NfcFilterAdjust(&voice->Direct.Params[c].NFCtrlFilter, w0); for(i = 0;i < MAX_AMBI_ORDER+1;i++) voice->Direct.ChannelsPerOrder[i] = Device->NumChannelsPerOrder[i]; voice->Flags |= VOICE_HAS_NFC; } /* Calculate the directional coefficients once, which apply to all * input channels. */ CalcAngleCoeffs((Device->Render_Mode==StereoPair) ? ScaleAzimuthFront(Azi, 1.5f) : Azi, Elev, Spread, coeffs); for(c = 0;c < num_channels;c++) { /* Special-case LFE */ if(chans[c].channel == LFE) { if(Device->Dry.Buffer == Device->RealOut.Buffer) { int idx = GetChannelIdxByName(&Device->RealOut, chans[c].channel); if(idx != -1) voice->Direct.Params[c].Gains.Target[idx] = DryGain; } continue; } ComputePanGains(&Device->Dry, coeffs, DryGain * downmix_gain, voice->Direct.Params[c].Gains.Target); } for(i = 0;i < NumSends;i++) { const ALeffectslot *Slot = SendSlots[i]; if(Slot) for(c = 0;c < num_channels;c++) { /* Skip LFE */ if(chans[c].channel != LFE) ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels, coeffs, WetGain[i] * downmix_gain, voice->Send[i].Params[c].Gains.Target ); } } } else { ALfloat w0 = 0.0f; if(Device->AvgSpeakerDist > 0.0f) { /* If the source distance is 0, set w0 to w1 to act as a pass- * through. We still want to pass the signal through the * filters so they keep an appropriate history, in case the * source moves away from the listener. */ w0 = SPEEDOFSOUNDMETRESPERSEC / (Device->AvgSpeakerDist * (ALfloat)Device->Frequency); for(c = 0;c < num_channels;c++) NfcFilterAdjust(&voice->Direct.Params[c].NFCtrlFilter, w0); for(i = 0;i < MAX_AMBI_ORDER+1;i++) voice->Direct.ChannelsPerOrder[i] = Device->NumChannelsPerOrder[i]; voice->Flags |= VOICE_HAS_NFC; } for(c = 0;c < num_channels;c++) { ALfloat coeffs[MAX_AMBI_COEFFS]; /* Special-case LFE */ if(chans[c].channel == LFE) { if(Device->Dry.Buffer == Device->RealOut.Buffer) { int idx = GetChannelIdxByName(&Device->RealOut, chans[c].channel); if(idx != -1) voice->Direct.Params[c].Gains.Target[idx] = DryGain; } continue; } CalcAngleCoeffs( (Device->Render_Mode==StereoPair) ? ScaleAzimuthFront(chans[c].angle, 3.0f) : chans[c].angle, chans[c].elevation, Spread, coeffs ); ComputePanGains(&Device->Dry, coeffs, DryGain, voice->Direct.Params[c].Gains.Target); for(i = 0;i < NumSends;i++) { const ALeffectslot *Slot = SendSlots[i]; if(Slot) ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels, coeffs, WetGain[i], voice->Send[i].Params[c].Gains.Target ); } } } } { ALfloat hfScale = props->Direct.HFReference / Frequency; ALfloat lfScale = props->Direct.LFReference / Frequency; ALfloat gainHF = maxf(DryGainHF, 0.001f); /* Limit -60dB */ ALfloat gainLF = maxf(DryGainLF, 0.001f); voice->Direct.FilterType = AF_None; if(gainHF != 1.0f) voice->Direct.FilterType |= AF_LowPass; if(gainLF != 1.0f) voice->Direct.FilterType |= AF_HighPass; BiquadFilter_setParams( &voice->Direct.Params[0].LowPass, BiquadType_HighShelf, gainHF, hfScale, calc_rcpQ_from_slope(gainHF, 1.0f) ); BiquadFilter_setParams( &voice->Direct.Params[0].HighPass, BiquadType_LowShelf, gainLF, lfScale, calc_rcpQ_from_slope(gainLF, 1.0f) ); for(c = 1;c < num_channels;c++) { BiquadFilter_copyParams(&voice->Direct.Params[c].LowPass, &voice->Direct.Params[0].LowPass); BiquadFilter_copyParams(&voice->Direct.Params[c].HighPass, &voice->Direct.Params[0].HighPass); } } for(i = 0;i < NumSends;i++) { ALfloat hfScale = props->Send[i].HFReference / Frequency; ALfloat lfScale = props->Send[i].LFReference / Frequency; ALfloat gainHF = maxf(WetGainHF[i], 0.001f); ALfloat gainLF = maxf(WetGainLF[i], 0.001f); voice->Send[i].FilterType = AF_None; if(gainHF != 1.0f) voice->Send[i].FilterType |= AF_LowPass; if(gainLF != 1.0f) voice->Send[i].FilterType |= AF_HighPass; BiquadFilter_setParams( &voice->Send[i].Params[0].LowPass, BiquadType_HighShelf, gainHF, hfScale, calc_rcpQ_from_slope(gainHF, 1.0f) ); BiquadFilter_setParams( &voice->Send[i].Params[0].HighPass, BiquadType_LowShelf, gainLF, lfScale, calc_rcpQ_from_slope(gainLF, 1.0f) ); for(c = 1;c < num_channels;c++) { BiquadFilter_copyParams(&voice->Send[i].Params[c].LowPass, &voice->Send[i].Params[0].LowPass); BiquadFilter_copyParams(&voice->Send[i].Params[c].HighPass, &voice->Send[i].Params[0].HighPass); } } } static void CalcNonAttnSourceParams(ALvoice *voice, const struct ALvoiceProps *props, const ALbuffer *ALBuffer, const ALCcontext *ALContext) { const ALCdevice *Device = ALContext->Device; const ALlistener *Listener = ALContext->Listener; ALfloat DryGain, DryGainHF, DryGainLF; ALfloat WetGain[MAX_SENDS]; ALfloat WetGainHF[MAX_SENDS]; ALfloat WetGainLF[MAX_SENDS]; ALeffectslot *SendSlots[MAX_SENDS]; ALfloat Pitch; ALsizei i; voice->Direct.Buffer = Device->Dry.Buffer; voice->Direct.Channels = Device->Dry.NumChannels; for(i = 0;i < Device->NumAuxSends;i++) { SendSlots[i] = props->Send[i].Slot; if(!SendSlots[i] && i == 0) SendSlots[i] = ALContext->DefaultSlot; if(!SendSlots[i] || SendSlots[i]->Params.EffectType == AL_EFFECT_NULL) { SendSlots[i] = NULL; voice->Send[i].Buffer = NULL; voice->Send[i].Channels = 0; } else { voice->Send[i].Buffer = SendSlots[i]->WetBuffer; voice->Send[i].Channels = SendSlots[i]->NumChannels; } } /* Calculate the stepping value */ Pitch = (ALfloat)ALBuffer->Frequency/(ALfloat)Device->Frequency * props->Pitch; if(Pitch > (ALfloat)MAX_PITCH) voice->Step = MAX_PITCH<Step = maxi(fastf2i(Pitch * FRACTIONONE), 1); if(props->Resampler == BSinc24Resampler) BsincPrepare(voice->Step, &voice->ResampleState.bsinc, &bsinc24); else if(props->Resampler == BSinc12Resampler) BsincPrepare(voice->Step, &voice->ResampleState.bsinc, &bsinc12); voice->Resampler = SelectResampler(props->Resampler); /* Calculate gains */ DryGain = clampf(props->Gain, props->MinGain, props->MaxGain); DryGain *= props->Direct.Gain * Listener->Params.Gain; DryGain = minf(DryGain, GAIN_MIX_MAX); DryGainHF = props->Direct.GainHF; DryGainLF = props->Direct.GainLF; for(i = 0;i < Device->NumAuxSends;i++) { WetGain[i] = clampf(props->Gain, props->MinGain, props->MaxGain); WetGain[i] *= props->Send[i].Gain * Listener->Params.Gain; WetGain[i] = minf(WetGain[i], GAIN_MIX_MAX); WetGainHF[i] = props->Send[i].GainHF; WetGainLF[i] = props->Send[i].GainLF; } CalcPanningAndFilters(voice, 0.0f, 0.0f, 0.0f, 0.0f, DryGain, DryGainHF, DryGainLF, WetGain, WetGainLF, WetGainHF, SendSlots, ALBuffer, props, Listener, Device); } static void CalcAttnSourceParams(ALvoice *voice, const struct ALvoiceProps *props, const ALbuffer *ALBuffer, const ALCcontext *ALContext) { const ALCdevice *Device = ALContext->Device; const ALlistener *Listener = ALContext->Listener; const ALsizei NumSends = Device->NumAuxSends; aluVector Position, Velocity, Direction, SourceToListener; ALfloat Distance, ClampedDist, DopplerFactor; ALeffectslot *SendSlots[MAX_SENDS]; ALfloat RoomRolloff[MAX_SENDS]; ALfloat DecayDistance[MAX_SENDS]; ALfloat DecayLFDistance[MAX_SENDS]; ALfloat DecayHFDistance[MAX_SENDS]; ALfloat DryGain, DryGainHF, DryGainLF; ALfloat WetGain[MAX_SENDS]; ALfloat WetGainHF[MAX_SENDS]; ALfloat WetGainLF[MAX_SENDS]; bool directional; ALfloat ev, az; ALfloat spread; ALfloat Pitch; ALint i; /* Set mixing buffers and get send parameters. */ voice->Direct.Buffer = Device->Dry.Buffer; voice->Direct.Channels = Device->Dry.NumChannels; for(i = 0;i < NumSends;i++) { SendSlots[i] = props->Send[i].Slot; if(!SendSlots[i] && i == 0) SendSlots[i] = ALContext->DefaultSlot; if(!SendSlots[i] || SendSlots[i]->Params.EffectType == AL_EFFECT_NULL) { SendSlots[i] = NULL; RoomRolloff[i] = 0.0f; DecayDistance[i] = 0.0f; DecayLFDistance[i] = 0.0f; DecayHFDistance[i] = 0.0f; } else if(SendSlots[i]->Params.AuxSendAuto) { RoomRolloff[i] = SendSlots[i]->Params.RoomRolloff + props->RoomRolloffFactor; /* Calculate the distances to where this effect's decay reaches * -60dB. */ DecayDistance[i] = SendSlots[i]->Params.DecayTime * Listener->Params.ReverbSpeedOfSound; DecayLFDistance[i] = DecayDistance[i] * SendSlots[i]->Params.DecayLFRatio; DecayHFDistance[i] = DecayDistance[i] * SendSlots[i]->Params.DecayHFRatio; if(SendSlots[i]->Params.DecayHFLimit) { ALfloat airAbsorption = SendSlots[i]->Params.AirAbsorptionGainHF; if(airAbsorption < 1.0f) { /* Calculate the distance to where this effect's air * absorption reaches -60dB, and limit the effect's HF * decay distance (so it doesn't take any longer to decay * than the air would allow). */ ALfloat absorb_dist = log10f(REVERB_DECAY_GAIN) / log10f(airAbsorption); DecayHFDistance[i] = minf(absorb_dist, DecayHFDistance[i]); } } } else { /* If the slot's auxiliary send auto is off, the data sent to the * effect slot is the same as the dry path, sans filter effects */ RoomRolloff[i] = props->RolloffFactor; DecayDistance[i] = 0.0f; DecayLFDistance[i] = 0.0f; DecayHFDistance[i] = 0.0f; } if(!SendSlots[i]) { voice->Send[i].Buffer = NULL; voice->Send[i].Channels = 0; } else { voice->Send[i].Buffer = SendSlots[i]->WetBuffer; voice->Send[i].Channels = SendSlots[i]->NumChannels; } } /* Transform source to listener space (convert to head relative) */ aluVectorSet(&Position, props->Position[0], props->Position[1], props->Position[2], 1.0f); aluVectorSet(&Direction, props->Direction[0], props->Direction[1], props->Direction[2], 0.0f); aluVectorSet(&Velocity, props->Velocity[0], props->Velocity[1], props->Velocity[2], 0.0f); if(props->HeadRelative == AL_FALSE) { const aluMatrixf *Matrix = &Listener->Params.Matrix; /* Transform source vectors */ Position = aluMatrixfVector(Matrix, &Position); Velocity = aluMatrixfVector(Matrix, &Velocity); Direction = aluMatrixfVector(Matrix, &Direction); } else { const aluVector *lvelocity = &Listener->Params.Velocity; /* Offset the source velocity to be relative of the listener velocity */ Velocity.v[0] += lvelocity->v[0]; Velocity.v[1] += lvelocity->v[1]; Velocity.v[2] += lvelocity->v[2]; } directional = aluNormalize(Direction.v) > 0.0f; SourceToListener.v[0] = -Position.v[0]; SourceToListener.v[1] = -Position.v[1]; SourceToListener.v[2] = -Position.v[2]; SourceToListener.v[3] = 0.0f; Distance = aluNormalize(SourceToListener.v); /* Initial source gain */ DryGain = props->Gain; DryGainHF = 1.0f; DryGainLF = 1.0f; for(i = 0;i < NumSends;i++) { WetGain[i] = props->Gain; WetGainHF[i] = 1.0f; WetGainLF[i] = 1.0f; } /* Calculate distance attenuation */ ClampedDist = Distance; switch(Listener->Params.SourceDistanceModel ? props->DistanceModel : Listener->Params.DistanceModel) { case InverseDistanceClamped: ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance); if(props->MaxDistance < props->RefDistance) break; /*fall-through*/ case InverseDistance: if(!(props->RefDistance > 0.0f)) ClampedDist = props->RefDistance; else { ALfloat dist = lerp(props->RefDistance, ClampedDist, props->RolloffFactor); if(dist > 0.0f) DryGain *= props->RefDistance / dist; for(i = 0;i < NumSends;i++) { dist = lerp(props->RefDistance, ClampedDist, RoomRolloff[i]); if(dist > 0.0f) WetGain[i] *= props->RefDistance / dist; } } break; case LinearDistanceClamped: ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance); if(props->MaxDistance < props->RefDistance) break; /*fall-through*/ case LinearDistance: if(!(props->MaxDistance != props->RefDistance)) ClampedDist = props->RefDistance; else { ALfloat attn = props->RolloffFactor * (ClampedDist-props->RefDistance) / (props->MaxDistance-props->RefDistance); DryGain *= maxf(1.0f - attn, 0.0f); for(i = 0;i < NumSends;i++) { attn = RoomRolloff[i] * (ClampedDist-props->RefDistance) / (props->MaxDistance-props->RefDistance); WetGain[i] *= maxf(1.0f - attn, 0.0f); } } break; case ExponentDistanceClamped: ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance); if(props->MaxDistance < props->RefDistance) break; /*fall-through*/ case ExponentDistance: if(!(ClampedDist > 0.0f && props->RefDistance > 0.0f)) ClampedDist = props->RefDistance; else { DryGain *= powf(ClampedDist/props->RefDistance, -props->RolloffFactor); for(i = 0;i < NumSends;i++) WetGain[i] *= powf(ClampedDist/props->RefDistance, -RoomRolloff[i]); } break; case DisableDistance: ClampedDist = props->RefDistance; break; } /* Calculate directional soundcones */ if(directional && props->InnerAngle < 360.0f) { ALfloat ConeVolume; ALfloat ConeHF; ALfloat Angle; Angle = acosf(aluDotproduct(&Direction, &SourceToListener)); Angle = RAD2DEG(Angle * ConeScale * 2.0f); if(!(Angle > props->InnerAngle)) { ConeVolume = 1.0f; ConeHF = 1.0f; } else if(Angle < props->OuterAngle) { ALfloat scale = ( Angle-props->InnerAngle) / (props->OuterAngle-props->InnerAngle); ConeVolume = lerp(1.0f, props->OuterGain, scale); ConeHF = lerp(1.0f, props->OuterGainHF, scale); } else { ConeVolume = props->OuterGain; ConeHF = props->OuterGainHF; } DryGain *= ConeVolume; if(props->DryGainHFAuto) DryGainHF *= ConeHF; if(props->WetGainAuto) { for(i = 0;i < NumSends;i++) WetGain[i] *= ConeVolume; } if(props->WetGainHFAuto) { for(i = 0;i < NumSends;i++) WetGainHF[i] *= ConeHF; } } /* Apply gain and frequency filters */ DryGain = clampf(DryGain, props->MinGain, props->MaxGain); DryGain = minf(DryGain*props->Direct.Gain*Listener->Params.Gain, GAIN_MIX_MAX); DryGainHF *= props->Direct.GainHF; DryGainLF *= props->Direct.GainLF; for(i = 0;i < NumSends;i++) { WetGain[i] = clampf(WetGain[i], props->MinGain, props->MaxGain); WetGain[i] = minf(WetGain[i]*props->Send[i].Gain*Listener->Params.Gain, GAIN_MIX_MAX); WetGainHF[i] *= props->Send[i].GainHF; WetGainLF[i] *= props->Send[i].GainLF; } /* Distance-based air absorption and initial send decay. */ if(ClampedDist > props->RefDistance && props->RolloffFactor > 0.0f) { ALfloat meters_base = (ClampedDist-props->RefDistance) * props->RolloffFactor * Listener->Params.MetersPerUnit; if(props->AirAbsorptionFactor > 0.0f) { ALfloat hfattn = powf(AIRABSORBGAINHF, meters_base * props->AirAbsorptionFactor); DryGainHF *= hfattn; for(i = 0;i < NumSends;i++) WetGainHF[i] *= hfattn; } if(props->WetGainAuto) { /* Apply a decay-time transformation to the wet path, based on the * source distance in meters. The initial decay of the reverb * effect is calculated and applied to the wet path. */ for(i = 0;i < NumSends;i++) { ALfloat gain, gainhf, gainlf; if(!(DecayDistance[i] > 0.0f)) continue; gain = powf(REVERB_DECAY_GAIN, meters_base/DecayDistance[i]); WetGain[i] *= gain; /* Yes, the wet path's air absorption is applied with * WetGainAuto on, rather than WetGainHFAuto. */ if(gain > 0.0f) { gainhf = powf(REVERB_DECAY_GAIN, meters_base/DecayHFDistance[i]); WetGainHF[i] *= minf(gainhf / gain, 1.0f); gainlf = powf(REVERB_DECAY_GAIN, meters_base/DecayLFDistance[i]); WetGainLF[i] *= minf(gainlf / gain, 1.0f); } } } } /* Initial source pitch */ Pitch = props->Pitch; /* Calculate velocity-based doppler effect */ DopplerFactor = props->DopplerFactor * Listener->Params.DopplerFactor; if(DopplerFactor > 0.0f) { const aluVector *lvelocity = &Listener->Params.Velocity; const ALfloat SpeedOfSound = Listener->Params.SpeedOfSound; ALfloat vss, vls; vss = aluDotproduct(&Velocity, &SourceToListener) * DopplerFactor; vls = aluDotproduct(lvelocity, &SourceToListener) * DopplerFactor; if(!(vls < SpeedOfSound)) { /* Listener moving away from the source at the speed of sound. * Sound waves can't catch it. */ Pitch = 0.0f; } else if(!(vss < SpeedOfSound)) { /* Source moving toward the listener at the speed of sound. Sound * waves bunch up to extreme frequencies. */ Pitch = HUGE_VALF; } else { /* Source and listener movement is nominal. Calculate the proper * doppler shift. */ Pitch *= (SpeedOfSound-vls) / (SpeedOfSound-vss); } } /* Adjust pitch based on the buffer and output frequencies, and calculate * fixed-point stepping value. */ Pitch *= (ALfloat)ALBuffer->Frequency/(ALfloat)Device->Frequency; if(Pitch > (ALfloat)MAX_PITCH) voice->Step = MAX_PITCH<Step = maxi(fastf2i(Pitch * FRACTIONONE), 1); if(props->Resampler == BSinc24Resampler) BsincPrepare(voice->Step, &voice->ResampleState.bsinc, &bsinc24); else if(props->Resampler == BSinc12Resampler) BsincPrepare(voice->Step, &voice->ResampleState.bsinc, &bsinc12); voice->Resampler = SelectResampler(props->Resampler); if(Distance > 0.0f) { /* Clamp Y, in case rounding errors caused it to end up outside of * -1...+1. */ ev = asinf(clampf(-SourceToListener.v[1], -1.0f, 1.0f)); /* Double negation on Z cancels out; negate once for changing source- * to-listener to listener-to-source, and again for right-handed coords * with -Z in front. */ az = atan2f(-SourceToListener.v[0], SourceToListener.v[2]*ZScale); } else ev = az = 0.0f; if(props->Radius > Distance) spread = F_TAU - Distance/props->Radius*F_PI; else if(Distance > 0.0f) spread = asinf(props->Radius / Distance) * 2.0f; else spread = 0.0f; CalcPanningAndFilters(voice, az, ev, Distance, spread, DryGain, DryGainHF, DryGainLF, WetGain, WetGainLF, WetGainHF, SendSlots, ALBuffer, props, Listener, Device); } static void CalcSourceParams(ALvoice *voice, ALCcontext *context, bool force) { ALbufferlistitem *BufferListItem; struct ALvoiceProps *props; props = ATOMIC_EXCHANGE_PTR(&voice->Update, NULL, almemory_order_acq_rel); if(!props && !force) return; if(props) { memcpy(voice->Props, props, FAM_SIZE(struct ALvoiceProps, Send, context->Device->NumAuxSends) ); ATOMIC_REPLACE_HEAD(struct ALvoiceProps*, &context->FreeVoiceProps, props); } props = voice->Props; BufferListItem = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed); while(BufferListItem != NULL) { const ALbuffer *buffer = NULL; ALsizei i = 0; while(!buffer && i < BufferListItem->num_buffers) buffer = BufferListItem->buffers[i]; if(LIKELY(buffer)) { if(props->SpatializeMode == SpatializeOn || (props->SpatializeMode == SpatializeAuto && buffer->FmtChannels == FmtMono)) CalcAttnSourceParams(voice, props, buffer, context); else CalcNonAttnSourceParams(voice, props, buffer, context); break; } BufferListItem = ATOMIC_LOAD(&BufferListItem->next, almemory_order_acquire); } } static void ProcessParamUpdates(ALCcontext *ctx, const struct ALeffectslotArray *slots) { ALvoice **voice, **voice_end; ALsource *source; ALsizei i; IncrementRef(&ctx->UpdateCount); if(!ATOMIC_LOAD(&ctx->HoldUpdates, almemory_order_acquire)) { bool cforce = CalcContextParams(ctx); bool force = CalcListenerParams(ctx) | cforce; for(i = 0;i < slots->count;i++) force |= CalcEffectSlotParams(slots->slot[i], ctx, cforce); voice = ctx->Voices; voice_end = voice + ctx->VoiceCount; for(;voice != voice_end;++voice) { source = ATOMIC_LOAD(&(*voice)->Source, almemory_order_acquire); if(source) CalcSourceParams(*voice, ctx, force); } } IncrementRef(&ctx->UpdateCount); } static void ApplyStablizer(FrontStablizer *Stablizer, ALfloat (*restrict Buffer)[BUFFERSIZE], int lidx, int ridx, int cidx, ALsizei SamplesToDo, ALsizei NumChannels) { ALfloat (*restrict lsplit)[BUFFERSIZE] = ASSUME_ALIGNED(Stablizer->LSplit, 16); ALfloat (*restrict rsplit)[BUFFERSIZE] = ASSUME_ALIGNED(Stablizer->RSplit, 16); ALsizei i; /* Apply an all-pass to all channels, except the front-left and front- * right, so they maintain the same relative phase. */ for(i = 0;i < NumChannels;i++) { if(i == lidx || i == ridx) continue; splitterap_process(&Stablizer->APFilter[i], Buffer[i], SamplesToDo); } bandsplit_process(&Stablizer->LFilter, lsplit[1], lsplit[0], Buffer[lidx], SamplesToDo); bandsplit_process(&Stablizer->RFilter, rsplit[1], rsplit[0], Buffer[ridx], SamplesToDo); for(i = 0;i < SamplesToDo;i++) { ALfloat lfsum, hfsum; ALfloat m, s, c; lfsum = lsplit[0][i] + rsplit[0][i]; hfsum = lsplit[1][i] + rsplit[1][i]; s = lsplit[0][i] + lsplit[1][i] - rsplit[0][i] - rsplit[1][i]; /* This pans the separate low- and high-frequency sums between being on * the center channel and the left/right channels. The low-frequency * sum is 1/3rd toward center (2/3rds on left/right) and the high- * frequency sum is 1/4th toward center (3/4ths on left/right). These * values can be tweaked. */ m = lfsum*cosf(1.0f/3.0f * F_PI_2) + hfsum*cosf(1.0f/4.0f * F_PI_2); c = lfsum*sinf(1.0f/3.0f * F_PI_2) + hfsum*sinf(1.0f/4.0f * F_PI_2); /* The generated center channel signal adds to the existing signal, * while the modified left and right channels replace. */ Buffer[lidx][i] = (m + s) * 0.5f; Buffer[ridx][i] = (m - s) * 0.5f; Buffer[cidx][i] += c * 0.5f; } } static void ApplyDistanceComp(ALfloat (*restrict Samples)[BUFFERSIZE], DistanceComp *distcomp, ALfloat *restrict Values, ALsizei SamplesToDo, ALsizei numchans) { ALsizei i, c; Values = ASSUME_ALIGNED(Values, 16); for(c = 0;c < numchans;c++) { ALfloat *restrict inout = ASSUME_ALIGNED(Samples[c], 16); const ALfloat gain = distcomp[c].Gain; const ALsizei base = distcomp[c].Length; ALfloat *restrict distbuf = ASSUME_ALIGNED(distcomp[c].Buffer, 16); if(base == 0) { if(gain < 1.0f) { for(i = 0;i < SamplesToDo;i++) inout[i] *= gain; } continue; } if(LIKELY(SamplesToDo >= base)) { for(i = 0;i < base;i++) Values[i] = distbuf[i]; for(;i < SamplesToDo;i++) Values[i] = inout[i-base]; memcpy(distbuf, &inout[SamplesToDo-base], base*sizeof(ALfloat)); } else { for(i = 0;i < SamplesToDo;i++) Values[i] = distbuf[i]; memmove(distbuf, distbuf+SamplesToDo, (base-SamplesToDo)*sizeof(ALfloat)); memcpy(distbuf+base-SamplesToDo, inout, SamplesToDo*sizeof(ALfloat)); } for(i = 0;i < SamplesToDo;i++) inout[i] = Values[i]*gain; } } static void ApplyDither(ALfloat (*restrict Samples)[BUFFERSIZE], ALuint *dither_seed, const ALfloat quant_scale, const ALsizei SamplesToDo, const ALsizei numchans) { const ALfloat invscale = 1.0f / quant_scale; ALuint seed = *dither_seed; ALsizei c, i; ASSUME(numchans > 0); ASSUME(SamplesToDo > 0); /* Dithering. Step 1, generate whitenoise (uniform distribution of random * values between -1 and +1). Step 2 is to add the noise to the samples, * before rounding and after scaling up to the desired quantization depth. */ for(c = 0;c < numchans;c++) { ALfloat *restrict samples = Samples[c]; for(i = 0;i < SamplesToDo;i++) { ALfloat val = samples[i] * quant_scale; ALuint rng0 = dither_rng(&seed); ALuint rng1 = dither_rng(&seed); val += (ALfloat)(rng0*(1.0/UINT_MAX) - rng1*(1.0/UINT_MAX)); samples[i] = fast_roundf(val) * invscale; } } *dither_seed = seed; } static inline ALfloat Conv_ALfloat(ALfloat val) { return val; } static inline ALint Conv_ALint(ALfloat val) { /* Floats have a 23-bit mantissa. There is an implied 1 bit in the mantissa * along with the sign bit, giving 25 bits total, so [-16777216, +16777216] * is the max value a normalized float can be scaled to before losing * precision. */ return fastf2i(clampf(val*16777216.0f, -16777216.0f, 16777215.0f))<<7; } static inline ALshort Conv_ALshort(ALfloat val) { return fastf2i(clampf(val*32768.0f, -32768.0f, 32767.0f)); } static inline ALbyte Conv_ALbyte(ALfloat val) { return fastf2i(clampf(val*128.0f, -128.0f, 127.0f)); } /* Define unsigned output variations. */ #define DECL_TEMPLATE(T, func, O) \ static inline T Conv_##T(ALfloat val) { return func(val)+O; } DECL_TEMPLATE(ALubyte, Conv_ALbyte, 128) DECL_TEMPLATE(ALushort, Conv_ALshort, 32768) DECL_TEMPLATE(ALuint, Conv_ALint, 2147483648u) #undef DECL_TEMPLATE #define DECL_TEMPLATE(T, A) \ static void Write##A(const ALfloat (*restrict InBuffer)[BUFFERSIZE], \ ALvoid *OutBuffer, ALsizei Offset, ALsizei SamplesToDo, \ ALsizei numchans) \ { \ ALsizei i, j; \ \ ASSUME(numchans > 0); \ ASSUME(SamplesToDo > 0); \ \ for(j = 0;j < numchans;j++) \ { \ const ALfloat *restrict in = ASSUME_ALIGNED(InBuffer[j], 16); \ T *restrict out = (T*)OutBuffer + Offset*numchans + j; \ \ for(i = 0;i < SamplesToDo;i++) \ out[i*numchans] = Conv_##T(in[i]); \ } \ } DECL_TEMPLATE(ALfloat, F32) DECL_TEMPLATE(ALuint, UI32) DECL_TEMPLATE(ALint, I32) DECL_TEMPLATE(ALushort, UI16) DECL_TEMPLATE(ALshort, I16) DECL_TEMPLATE(ALubyte, UI8) DECL_TEMPLATE(ALbyte, I8) #undef DECL_TEMPLATE void aluMixData(ALCdevice *device, ALvoid *OutBuffer, ALsizei NumSamples) { ALsizei SamplesToDo; ALsizei SamplesDone; ALCcontext *ctx; ALsizei i, c; START_MIXER_MODE(); for(SamplesDone = 0;SamplesDone < NumSamples;) { SamplesToDo = mini(NumSamples-SamplesDone, BUFFERSIZE); for(c = 0;c < device->Dry.NumChannels;c++) memset(device->Dry.Buffer[c], 0, SamplesToDo*sizeof(ALfloat)); if(device->Dry.Buffer != device->FOAOut.Buffer) for(c = 0;c < device->FOAOut.NumChannels;c++) memset(device->FOAOut.Buffer[c], 0, SamplesToDo*sizeof(ALfloat)); if(device->Dry.Buffer != device->RealOut.Buffer) for(c = 0;c < device->RealOut.NumChannels;c++) memset(device->RealOut.Buffer[c], 0, SamplesToDo*sizeof(ALfloat)); IncrementRef(&device->MixCount); ctx = ATOMIC_LOAD(&device->ContextList, almemory_order_acquire); while(ctx) { const struct ALeffectslotArray *auxslots; auxslots = ATOMIC_LOAD(&ctx->ActiveAuxSlots, almemory_order_acquire); ProcessParamUpdates(ctx, auxslots); for(i = 0;i < auxslots->count;i++) { ALeffectslot *slot = auxslots->slot[i]; for(c = 0;c < slot->NumChannels;c++) memset(slot->WetBuffer[c], 0, SamplesToDo*sizeof(ALfloat)); } /* source processing */ for(i = 0;i < ctx->VoiceCount;i++) { ALvoice *voice = ctx->Voices[i]; ALsource *source = ATOMIC_LOAD(&voice->Source, almemory_order_acquire); if(source && ATOMIC_LOAD(&voice->Playing, almemory_order_relaxed) && voice->Step > 0) { if(!MixSource(voice, source->id, ctx, SamplesToDo)) { ATOMIC_STORE(&voice->Source, NULL, almemory_order_relaxed); ATOMIC_STORE(&voice->Playing, false, almemory_order_release); SendSourceStoppedEvent(ctx, source->id); } } } /* effect slot processing */ for(i = 0;i < auxslots->count;i++) { const ALeffectslot *slot = auxslots->slot[i]; ALeffectState *state = slot->Params.EffectState; V(state,process)(SamplesToDo, slot->WetBuffer, state->OutBuffer, state->OutChannels); } ctx = ATOMIC_LOAD(&ctx->next, almemory_order_relaxed); } /* Increment the clock time. Every second's worth of samples is * converted and added to clock base so that large sample counts don't * overflow during conversion. This also guarantees an exact, stable * conversion. */ device->SamplesDone += SamplesToDo; device->ClockBase += (device->SamplesDone/device->Frequency) * DEVICE_CLOCK_RES; device->SamplesDone %= device->Frequency; IncrementRef(&device->MixCount); /* Apply post-process for finalizing the Dry mix to the RealOut * (Ambisonic decode, UHJ encode, etc). */ if(LIKELY(device->PostProcess)) device->PostProcess(device, SamplesToDo); if(device->Stablizer) { int lidx = GetChannelIdxByName(&device->RealOut, FrontLeft); int ridx = GetChannelIdxByName(&device->RealOut, FrontRight); int cidx = GetChannelIdxByName(&device->RealOut, FrontCenter); assert(lidx >= 0 && ridx >= 0 && cidx >= 0); ApplyStablizer(device->Stablizer, device->RealOut.Buffer, lidx, ridx, cidx, SamplesToDo, device->RealOut.NumChannels); } ApplyDistanceComp(device->RealOut.Buffer, device->ChannelDelay, device->TempBuffer[0], SamplesToDo, device->RealOut.NumChannels); if(device->Limiter) ApplyCompression(device->Limiter, SamplesToDo, device->RealOut.Buffer); if(device->DitherDepth > 0.0f) ApplyDither(device->RealOut.Buffer, &device->DitherSeed, device->DitherDepth, SamplesToDo, device->RealOut.NumChannels); if(LIKELY(OutBuffer)) { ALfloat (*Buffer)[BUFFERSIZE] = device->RealOut.Buffer; ALsizei Channels = device->RealOut.NumChannels; switch(device->FmtType) { #define HANDLE_WRITE(T, S) case T: \ Write##S(Buffer, OutBuffer, SamplesDone, SamplesToDo, Channels); break; HANDLE_WRITE(DevFmtByte, I8) HANDLE_WRITE(DevFmtUByte, UI8) HANDLE_WRITE(DevFmtShort, I16) HANDLE_WRITE(DevFmtUShort, UI16) HANDLE_WRITE(DevFmtInt, I32) HANDLE_WRITE(DevFmtUInt, UI32) HANDLE_WRITE(DevFmtFloat, F32) #undef HANDLE_WRITE } } SamplesDone += SamplesToDo; } END_MIXER_MODE(); } void aluHandleDisconnect(ALCdevice *device, const char *msg, ...) { AsyncEvent evt = ASYNC_EVENT(EventType_Disconnected); ALCcontext *ctx; va_list args; int msglen; if(!ATOMIC_EXCHANGE(&device->Connected, AL_FALSE, almemory_order_acq_rel)) return; evt.u.user.type = AL_EVENT_TYPE_DISCONNECTED_SOFT; evt.u.user.id = 0; evt.u.user.param = 0; va_start(args, msg); msglen = vsnprintf(evt.u.user.msg, sizeof(evt.u.user.msg), msg, args); va_end(args); if(msglen < 0 || (size_t)msglen >= sizeof(evt.u.user.msg)) evt.u.user.msg[sizeof(evt.u.user.msg)-1] = 0; ctx = ATOMIC_LOAD_SEQ(&device->ContextList); while(ctx) { ALbitfieldSOFT enabledevt = ATOMIC_LOAD(&ctx->EnabledEvts, almemory_order_acquire); ALsizei i; if((enabledevt&EventType_Disconnected) && ll_ringbuffer_write(ctx->AsyncEvents, (const char*)&evt, 1) == 1) alsem_post(&ctx->EventSem); for(i = 0;i < ctx->VoiceCount;i++) { ALvoice *voice = ctx->Voices[i]; ALsource *source; source = ATOMIC_EXCHANGE_PTR(&voice->Source, NULL, almemory_order_relaxed); if(source && ATOMIC_LOAD(&voice->Playing, almemory_order_relaxed)) { /* If the source's voice was playing, it's now effectively * stopped (the source state will be updated the next time it's * checked). */ SendSourceStoppedEvent(ctx, source->id); } ATOMIC_STORE(&voice->Playing, false, almemory_order_release); } ctx = ATOMIC_LOAD(&ctx->next, almemory_order_relaxed); } } openal-soft-openal-soft-1.19.1/Alc/alconfig.c000066400000000000000000000462221335774445300207310ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #ifdef _WIN32 #ifdef __MINGW32__ #define _WIN32_IE 0x501 #else #define _WIN32_IE 0x400 #endif #endif #include "config.h" #include #include #include #include #ifdef _WIN32_IE #include #include #endif #ifdef __APPLE__ #include #endif #include "alMain.h" #include "alconfig.h" #include "compat.h" #include "bool.h" typedef struct ConfigEntry { char *key; char *value; } ConfigEntry; typedef struct ConfigBlock { ConfigEntry *entries; unsigned int entryCount; } ConfigBlock; static ConfigBlock cfgBlock; static char *lstrip(char *line) { while(isspace(line[0])) line++; return line; } static char *rstrip(char *line) { size_t len = strlen(line); while(len > 0 && isspace(line[len-1])) len--; line[len] = 0; return line; } static int readline(FILE *f, char **output, size_t *maxlen) { size_t len = 0; int c; while((c=fgetc(f)) != EOF && (c == '\r' || c == '\n')) ; if(c == EOF) return 0; do { if(len+1 >= *maxlen) { void *temp = NULL; size_t newmax; newmax = (*maxlen ? (*maxlen)<<1 : 32); if(newmax > *maxlen) temp = realloc(*output, newmax); if(!temp) { ERR("Failed to realloc "SZFMT" bytes from "SZFMT"!\n", newmax, *maxlen); return 0; } *output = temp; *maxlen = newmax; } (*output)[len++] = c; (*output)[len] = '\0'; } while((c=fgetc(f)) != EOF && c != '\r' && c != '\n'); return 1; } static char *expdup(const char *str) { char *output = NULL; size_t maxlen = 0; size_t len = 0; while(*str != '\0') { const char *addstr; size_t addstrlen; size_t i; if(str[0] != '$') { const char *next = strchr(str, '$'); addstr = str; addstrlen = next ? (size_t)(next-str) : strlen(str); str += addstrlen; } else { str++; if(*str == '$') { const char *next = strchr(str+1, '$'); addstr = str; addstrlen = next ? (size_t)(next-str) : strlen(str); str += addstrlen; } else { bool hasbraces; char envname[1024]; size_t k = 0; hasbraces = (*str == '{'); if(hasbraces) str++; while((isalnum(*str) || *str == '_') && k < sizeof(envname)-1) envname[k++] = *(str++); envname[k++] = '\0'; if(hasbraces && *str != '}') continue; if(hasbraces) str++; if((addstr=getenv(envname)) == NULL) continue; addstrlen = strlen(addstr); } } if(addstrlen == 0) continue; if(addstrlen >= maxlen-len) { void *temp = NULL; size_t newmax; newmax = len+addstrlen+1; if(newmax > maxlen) temp = realloc(output, newmax); if(!temp) { ERR("Failed to realloc "SZFMT" bytes from "SZFMT"!\n", newmax, maxlen); return output; } output = temp; maxlen = newmax; } for(i = 0;i < addstrlen;i++) output[len++] = addstr[i]; output[len] = '\0'; } return output ? output : calloc(1, 1); } static void LoadConfigFromFile(FILE *f) { char curSection[128] = ""; char *buffer = NULL; size_t maxlen = 0; ConfigEntry *ent; while(readline(f, &buffer, &maxlen)) { char *line, *comment; char key[256] = ""; char value[256] = ""; line = rstrip(lstrip(buffer)); if(!line[0]) continue; if(line[0] == '[') { char *section = line+1; char *endsection; endsection = strchr(section, ']'); if(!endsection || section == endsection) { ERR("config parse error: bad line \"%s\"\n", line); continue; } if(endsection[1] != 0) { char *end = endsection+1; while(isspace(*end)) ++end; if(*end != 0 && *end != '#') { ERR("config parse error: bad line \"%s\"\n", line); continue; } } *endsection = 0; if(strcasecmp(section, "general") == 0) curSection[0] = 0; else { size_t len, p = 0; do { char *nextp = strchr(section, '%'); if(!nextp) { strncpy(curSection+p, section, sizeof(curSection)-1-p); break; } len = nextp - section; if(len > sizeof(curSection)-1-p) len = sizeof(curSection)-1-p; strncpy(curSection+p, section, len); p += len; section = nextp; if(((section[1] >= '0' && section[1] <= '9') || (section[1] >= 'a' && section[1] <= 'f') || (section[1] >= 'A' && section[1] <= 'F')) && ((section[2] >= '0' && section[2] <= '9') || (section[2] >= 'a' && section[2] <= 'f') || (section[2] >= 'A' && section[2] <= 'F'))) { unsigned char b = 0; if(section[1] >= '0' && section[1] <= '9') b = (section[1]-'0') << 4; else if(section[1] >= 'a' && section[1] <= 'f') b = (section[1]-'a'+0xa) << 4; else if(section[1] >= 'A' && section[1] <= 'F') b = (section[1]-'A'+0x0a) << 4; if(section[2] >= '0' && section[2] <= '9') b |= (section[2]-'0'); else if(section[2] >= 'a' && section[2] <= 'f') b |= (section[2]-'a'+0xa); else if(section[2] >= 'A' && section[2] <= 'F') b |= (section[2]-'A'+0x0a); if(p < sizeof(curSection)-1) curSection[p++] = b; section += 3; } else if(section[1] == '%') { if(p < sizeof(curSection)-1) curSection[p++] = '%'; section += 2; } else { if(p < sizeof(curSection)-1) curSection[p++] = '%'; section += 1; } if(p < sizeof(curSection)-1) curSection[p] = 0; } while(p < sizeof(curSection)-1 && *section != 0); curSection[sizeof(curSection)-1] = 0; } continue; } comment = strchr(line, '#'); if(comment) *(comment++) = 0; if(!line[0]) continue; if(sscanf(line, "%255[^=] = \"%255[^\"]\"", key, value) == 2 || sscanf(line, "%255[^=] = '%255[^\']'", key, value) == 2 || sscanf(line, "%255[^=] = %255[^\n]", key, value) == 2) { /* sscanf doesn't handle '' or "" as empty values, so clip it * manually. */ if(strcmp(value, "\"\"") == 0 || strcmp(value, "''") == 0) value[0] = 0; } else if(sscanf(line, "%255[^=] %255[=]", key, value) == 2) { /* Special case for 'key =' */ value[0] = 0; } else { ERR("config parse error: malformed option line: \"%s\"\n\n", line); continue; } rstrip(key); if(curSection[0] != 0) { size_t len = strlen(curSection); memmove(&key[len+1], key, sizeof(key)-1-len); key[len] = '/'; memcpy(key, curSection, len); } /* Check if we already have this option set */ ent = cfgBlock.entries; while((unsigned int)(ent-cfgBlock.entries) < cfgBlock.entryCount) { if(strcasecmp(ent->key, key) == 0) break; ent++; } if((unsigned int)(ent-cfgBlock.entries) >= cfgBlock.entryCount) { /* Allocate a new option entry */ ent = realloc(cfgBlock.entries, (cfgBlock.entryCount+1)*sizeof(ConfigEntry)); if(!ent) { ERR("config parse error: error reallocating config entries\n"); continue; } cfgBlock.entries = ent; ent = cfgBlock.entries + cfgBlock.entryCount; cfgBlock.entryCount++; ent->key = strdup(key); ent->value = NULL; } free(ent->value); ent->value = expdup(value); TRACE("found '%s' = '%s'\n", ent->key, ent->value); } free(buffer); } #ifdef _WIN32 void ReadALConfig(void) { al_string ppath = AL_STRING_INIT_STATIC(); WCHAR buffer[MAX_PATH]; const WCHAR *str; FILE *f; if(SHGetSpecialFolderPathW(NULL, buffer, CSIDL_APPDATA, FALSE) != FALSE) { al_string filepath = AL_STRING_INIT_STATIC(); alstr_copy_wcstr(&filepath, buffer); alstr_append_cstr(&filepath, "\\alsoft.ini"); TRACE("Loading config %s...\n", alstr_get_cstr(filepath)); f = al_fopen(alstr_get_cstr(filepath), "rt"); if(f) { LoadConfigFromFile(f); fclose(f); } alstr_reset(&filepath); } GetProcBinary(&ppath, NULL); if(!alstr_empty(ppath)) { alstr_append_cstr(&ppath, "\\alsoft.ini"); TRACE("Loading config %s...\n", alstr_get_cstr(ppath)); f = al_fopen(alstr_get_cstr(ppath), "r"); if(f) { LoadConfigFromFile(f); fclose(f); } } if((str=_wgetenv(L"ALSOFT_CONF")) != NULL && *str) { al_string filepath = AL_STRING_INIT_STATIC(); alstr_copy_wcstr(&filepath, str); TRACE("Loading config %s...\n", alstr_get_cstr(filepath)); f = al_fopen(alstr_get_cstr(filepath), "rt"); if(f) { LoadConfigFromFile(f); fclose(f); } alstr_reset(&filepath); } alstr_reset(&ppath); } #else void ReadALConfig(void) { al_string confpaths = AL_STRING_INIT_STATIC(); al_string fname = AL_STRING_INIT_STATIC(); const char *str; FILE *f; str = "/etc/openal/alsoft.conf"; TRACE("Loading config %s...\n", str); f = al_fopen(str, "r"); if(f) { LoadConfigFromFile(f); fclose(f); } if(!(str=getenv("XDG_CONFIG_DIRS")) || str[0] == 0) str = "/etc/xdg"; alstr_copy_cstr(&confpaths, str); /* Go through the list in reverse, since "the order of base directories * denotes their importance; the first directory listed is the most * important". Ergo, we need to load the settings from the later dirs * first so that the settings in the earlier dirs override them. */ while(!alstr_empty(confpaths)) { char *next = strrchr(alstr_get_cstr(confpaths), ':'); if(next) { size_t len = next - alstr_get_cstr(confpaths); alstr_copy_cstr(&fname, next+1); VECTOR_RESIZE(confpaths, len, len+1); VECTOR_ELEM(confpaths, len) = 0; } else { alstr_reset(&fname); fname = confpaths; AL_STRING_INIT(confpaths); } if(alstr_empty(fname) || VECTOR_FRONT(fname) != '/') WARN("Ignoring XDG config dir: %s\n", alstr_get_cstr(fname)); else { if(VECTOR_BACK(fname) != '/') alstr_append_cstr(&fname, "/alsoft.conf"); else alstr_append_cstr(&fname, "alsoft.conf"); TRACE("Loading config %s...\n", alstr_get_cstr(fname)); f = al_fopen(alstr_get_cstr(fname), "r"); if(f) { LoadConfigFromFile(f); fclose(f); } } alstr_clear(&fname); } #ifdef __APPLE__ CFBundleRef mainBundle = CFBundleGetMainBundle(); if(mainBundle) { unsigned char fileName[PATH_MAX]; CFURLRef configURL; if((configURL=CFBundleCopyResourceURL(mainBundle, CFSTR(".alsoftrc"), CFSTR(""), NULL)) && CFURLGetFileSystemRepresentation(configURL, true, fileName, sizeof(fileName))) { f = al_fopen((const char*)fileName, "r"); if(f) { LoadConfigFromFile(f); fclose(f); } } } #endif if((str=getenv("HOME")) != NULL && *str) { alstr_copy_cstr(&fname, str); if(VECTOR_BACK(fname) != '/') alstr_append_cstr(&fname, "/.alsoftrc"); else alstr_append_cstr(&fname, ".alsoftrc"); TRACE("Loading config %s...\n", alstr_get_cstr(fname)); f = al_fopen(alstr_get_cstr(fname), "r"); if(f) { LoadConfigFromFile(f); fclose(f); } } if((str=getenv("XDG_CONFIG_HOME")) != NULL && str[0] != 0) { alstr_copy_cstr(&fname, str); if(VECTOR_BACK(fname) != '/') alstr_append_cstr(&fname, "/alsoft.conf"); else alstr_append_cstr(&fname, "alsoft.conf"); } else { alstr_clear(&fname); if((str=getenv("HOME")) != NULL && str[0] != 0) { alstr_copy_cstr(&fname, str); if(VECTOR_BACK(fname) != '/') alstr_append_cstr(&fname, "/.config/alsoft.conf"); else alstr_append_cstr(&fname, ".config/alsoft.conf"); } } if(!alstr_empty(fname)) { TRACE("Loading config %s...\n", alstr_get_cstr(fname)); f = al_fopen(alstr_get_cstr(fname), "r"); if(f) { LoadConfigFromFile(f); fclose(f); } } alstr_clear(&fname); GetProcBinary(&fname, NULL); if(!alstr_empty(fname)) { if(VECTOR_BACK(fname) != '/') alstr_append_cstr(&fname, "/alsoft.conf"); else alstr_append_cstr(&fname, "alsoft.conf"); TRACE("Loading config %s...\n", alstr_get_cstr(fname)); f = al_fopen(alstr_get_cstr(fname), "r"); if(f) { LoadConfigFromFile(f); fclose(f); } } if((str=getenv("ALSOFT_CONF")) != NULL && *str) { TRACE("Loading config %s...\n", str); f = al_fopen(str, "r"); if(f) { LoadConfigFromFile(f); fclose(f); } } alstr_reset(&fname); alstr_reset(&confpaths); } #endif void FreeALConfig(void) { unsigned int i; for(i = 0;i < cfgBlock.entryCount;i++) { free(cfgBlock.entries[i].key); free(cfgBlock.entries[i].value); } free(cfgBlock.entries); } const char *GetConfigValue(const char *devName, const char *blockName, const char *keyName, const char *def) { unsigned int i; char key[256]; if(!keyName) return def; if(blockName && strcasecmp(blockName, "general") != 0) { if(devName) snprintf(key, sizeof(key), "%s/%s/%s", blockName, devName, keyName); else snprintf(key, sizeof(key), "%s/%s", blockName, keyName); } else { if(devName) snprintf(key, sizeof(key), "%s/%s", devName, keyName); else { strncpy(key, keyName, sizeof(key)-1); key[sizeof(key)-1] = 0; } } for(i = 0;i < cfgBlock.entryCount;i++) { if(strcmp(cfgBlock.entries[i].key, key) == 0) { TRACE("Found %s = \"%s\"\n", key, cfgBlock.entries[i].value); if(cfgBlock.entries[i].value[0]) return cfgBlock.entries[i].value; return def; } } if(!devName) { TRACE("Key %s not found\n", key); return def; } return GetConfigValue(NULL, blockName, keyName, def); } int ConfigValueExists(const char *devName, const char *blockName, const char *keyName) { const char *val = GetConfigValue(devName, blockName, keyName, ""); return !!val[0]; } int ConfigValueStr(const char *devName, const char *blockName, const char *keyName, const char **ret) { const char *val = GetConfigValue(devName, blockName, keyName, ""); if(!val[0]) return 0; *ret = val; return 1; } int ConfigValueInt(const char *devName, const char *blockName, const char *keyName, int *ret) { const char *val = GetConfigValue(devName, blockName, keyName, ""); if(!val[0]) return 0; *ret = strtol(val, NULL, 0); return 1; } int ConfigValueUInt(const char *devName, const char *blockName, const char *keyName, unsigned int *ret) { const char *val = GetConfigValue(devName, blockName, keyName, ""); if(!val[0]) return 0; *ret = strtoul(val, NULL, 0); return 1; } int ConfigValueFloat(const char *devName, const char *blockName, const char *keyName, float *ret) { const char *val = GetConfigValue(devName, blockName, keyName, ""); if(!val[0]) return 0; #ifdef HAVE_STRTOF *ret = strtof(val, NULL); #else *ret = (float)strtod(val, NULL); #endif return 1; } int ConfigValueBool(const char *devName, const char *blockName, const char *keyName, int *ret) { const char *val = GetConfigValue(devName, blockName, keyName, ""); if(!val[0]) return 0; *ret = (strcasecmp(val, "true") == 0 || strcasecmp(val, "yes") == 0 || strcasecmp(val, "on") == 0 || atoi(val) != 0); return 1; } int GetConfigValueBool(const char *devName, const char *blockName, const char *keyName, int def) { const char *val = GetConfigValue(devName, blockName, keyName, ""); if(!val[0]) return !!def; return (strcasecmp(val, "true") == 0 || strcasecmp(val, "yes") == 0 || strcasecmp(val, "on") == 0 || atoi(val) != 0); } openal-soft-openal-soft-1.19.1/Alc/alconfig.h000066400000000000000000000016161335774445300207340ustar00rootroot00000000000000#ifndef ALCONFIG_H #define ALCONFIG_H void ReadALConfig(void); void FreeALConfig(void); int ConfigValueExists(const char *devName, const char *blockName, const char *keyName); const char *GetConfigValue(const char *devName, const char *blockName, const char *keyName, const char *def); int GetConfigValueBool(const char *devName, const char *blockName, const char *keyName, int def); int ConfigValueStr(const char *devName, const char *blockName, const char *keyName, const char **ret); int ConfigValueInt(const char *devName, const char *blockName, const char *keyName, int *ret); int ConfigValueUInt(const char *devName, const char *blockName, const char *keyName, unsigned int *ret); int ConfigValueFloat(const char *devName, const char *blockName, const char *keyName, float *ret); int ConfigValueBool(const char *devName, const char *blockName, const char *keyName, int *ret); #endif /* ALCONFIG_H */ openal-soft-openal-soft-1.19.1/Alc/alstring.h000066400000000000000000000035141335774445300207740ustar00rootroot00000000000000#ifndef ALSTRING_H #define ALSTRING_H #include #include "vector.h" #ifdef __cplusplus extern "C" { #endif typedef char al_string_char_type; TYPEDEF_VECTOR(al_string_char_type, al_string) TYPEDEF_VECTOR(al_string, vector_al_string) inline void alstr_reset(al_string *str) { VECTOR_DEINIT(*str); } #define AL_STRING_INIT(_x) do { (_x) = (al_string)NULL; } while(0) #define AL_STRING_INIT_STATIC() ((al_string)NULL) #define AL_STRING_DEINIT(_x) alstr_reset(&(_x)) inline size_t alstr_length(const_al_string str) { return VECTOR_SIZE(str); } inline ALboolean alstr_empty(const_al_string str) { return alstr_length(str) == 0; } inline const al_string_char_type *alstr_get_cstr(const_al_string str) { return str ? &VECTOR_FRONT(str) : ""; } void alstr_clear(al_string *str); int alstr_cmp(const_al_string str1, const_al_string str2); int alstr_cmp_cstr(const_al_string str1, const al_string_char_type *str2); void alstr_copy(al_string *str, const_al_string from); void alstr_copy_cstr(al_string *str, const al_string_char_type *from); void alstr_copy_range(al_string *str, const al_string_char_type *from, const al_string_char_type *to); void alstr_append_char(al_string *str, const al_string_char_type c); void alstr_append_cstr(al_string *str, const al_string_char_type *from); void alstr_append_range(al_string *str, const al_string_char_type *from, const al_string_char_type *to); #ifdef _WIN32 #include /* Windows-only methods to deal with WideChar strings. */ void alstr_copy_wcstr(al_string *str, const wchar_t *from); void alstr_append_wcstr(al_string *str, const wchar_t *from); void alstr_copy_wrange(al_string *str, const wchar_t *from, const wchar_t *to); void alstr_append_wrange(al_string *str, const wchar_t *from, const wchar_t *to); #endif #ifdef __cplusplus } /* extern "C" */ #endif #endif /* ALSTRING_H */ openal-soft-openal-soft-1.19.1/Alc/ambdec.c000066400000000000000000000402731335774445300203620ustar00rootroot00000000000000 #include "config.h" #include "ambdec.h" #include #include #include #include "compat.h" static char *lstrip(char *line) { while(isspace(line[0])) line++; return line; } static char *rstrip(char *line) { size_t len = strlen(line); while(len > 0 && isspace(line[len-1])) len--; line[len] = 0; return line; } static int readline(FILE *f, char **output, size_t *maxlen) { size_t len = 0; int c; while((c=fgetc(f)) != EOF && (c == '\r' || c == '\n')) ; if(c == EOF) return 0; do { if(len+1 >= *maxlen) { void *temp = NULL; size_t newmax; newmax = (*maxlen ? (*maxlen)<<1 : 32); if(newmax > *maxlen) temp = realloc(*output, newmax); if(!temp) { ERR("Failed to realloc "SZFMT" bytes from "SZFMT"!\n", newmax, *maxlen); return 0; } *output = temp; *maxlen = newmax; } (*output)[len++] = c; (*output)[len] = '\0'; } while((c=fgetc(f)) != EOF && c != '\r' && c != '\n'); return 1; } /* Custom strtok_r, since we can't rely on it existing. */ static char *my_strtok_r(char *str, const char *delim, char **saveptr) { /* Sanity check and update internal pointer. */ if(!saveptr || !delim) return NULL; if(str) *saveptr = str; str = *saveptr; /* Nothing more to do with this string. */ if(!str) return NULL; /* Find the first non-delimiter character. */ while(*str != '\0' && strchr(delim, *str) != NULL) str++; if(*str == '\0') { /* End of string. */ *saveptr = NULL; return NULL; } /* Find the next delimiter character. */ *saveptr = strpbrk(str, delim); if(*saveptr) *((*saveptr)++) = '\0'; return str; } static char *read_int(ALint *num, const char *line, int base) { char *end; *num = strtol(line, &end, base); if(end && *end != '\0') end = lstrip(end); return end; } static char *read_uint(ALuint *num, const char *line, int base) { char *end; *num = strtoul(line, &end, base); if(end && *end != '\0') end = lstrip(end); return end; } static char *read_float(ALfloat *num, const char *line) { char *end; #ifdef HAVE_STRTOF *num = strtof(line, &end); #else *num = (ALfloat)strtod(line, &end); #endif if(end && *end != '\0') end = lstrip(end); return end; } char *read_clipped_line(FILE *f, char **buffer, size_t *maxlen) { while(readline(f, buffer, maxlen)) { char *line, *comment; line = lstrip(*buffer); comment = strchr(line, '#'); if(comment) *(comment++) = 0; line = rstrip(line); if(line[0]) return line; } return NULL; } static int load_ambdec_speakers(AmbDecConf *conf, FILE *f, char **buffer, size_t *maxlen, char **saveptr) { ALsizei cur = 0; while(cur < conf->NumSpeakers) { const char *cmd = my_strtok_r(NULL, " \t", saveptr); if(!cmd) { char *line = read_clipped_line(f, buffer, maxlen); if(!line) { ERR("Unexpected end of file\n"); return 0; } cmd = my_strtok_r(line, " \t", saveptr); } if(strcmp(cmd, "add_spkr") == 0) { const char *name = my_strtok_r(NULL, " \t", saveptr); const char *dist = my_strtok_r(NULL, " \t", saveptr); const char *az = my_strtok_r(NULL, " \t", saveptr); const char *elev = my_strtok_r(NULL, " \t", saveptr); const char *conn = my_strtok_r(NULL, " \t", saveptr); if(!name) WARN("Name not specified for speaker %u\n", cur+1); else alstr_copy_cstr(&conf->Speakers[cur].Name, name); if(!dist) WARN("Distance not specified for speaker %u\n", cur+1); else read_float(&conf->Speakers[cur].Distance, dist); if(!az) WARN("Azimuth not specified for speaker %u\n", cur+1); else read_float(&conf->Speakers[cur].Azimuth, az); if(!elev) WARN("Elevation not specified for speaker %u\n", cur+1); else read_float(&conf->Speakers[cur].Elevation, elev); if(!conn) TRACE("Connection not specified for speaker %u\n", cur+1); else alstr_copy_cstr(&conf->Speakers[cur].Connection, conn); cur++; } else { ERR("Unexpected speakers command: %s\n", cmd); return 0; } cmd = my_strtok_r(NULL, " \t", saveptr); if(cmd) { ERR("Unexpected junk on line: %s\n", cmd); return 0; } } return 1; } static int load_ambdec_matrix(ALfloat *gains, ALfloat (*matrix)[MAX_AMBI_COEFFS], ALsizei maxrow, FILE *f, char **buffer, size_t *maxlen, char **saveptr) { int gotgains = 0; ALsizei cur = 0; while(cur < maxrow) { const char *cmd = my_strtok_r(NULL, " \t", saveptr); if(!cmd) { char *line = read_clipped_line(f, buffer, maxlen); if(!line) { ERR("Unexpected end of file\n"); return 0; } cmd = my_strtok_r(line, " \t", saveptr); } if(strcmp(cmd, "order_gain") == 0) { ALuint curgain = 0; char *line; while((line=my_strtok_r(NULL, " \t", saveptr)) != NULL) { ALfloat value; line = read_float(&value, line); if(line && *line != '\0') { ERR("Extra junk on gain %u: %s\n", curgain+1, line); return 0; } if(curgain < MAX_AMBI_ORDER+1) gains[curgain] = value; curgain++; } while(curgain < MAX_AMBI_ORDER+1) gains[curgain++] = 0.0f; gotgains = 1; } else if(strcmp(cmd, "add_row") == 0) { ALuint curidx = 0; char *line; while((line=my_strtok_r(NULL, " \t", saveptr)) != NULL) { ALfloat value; line = read_float(&value, line); if(line && *line != '\0') { ERR("Extra junk on matrix element %ux%u: %s\n", cur, curidx, line); return 0; } if(curidx < MAX_AMBI_COEFFS) matrix[cur][curidx] = value; curidx++; } while(curidx < MAX_AMBI_COEFFS) matrix[cur][curidx++] = 0.0f; cur++; } else { ERR("Unexpected speakers command: %s\n", cmd); return 0; } cmd = my_strtok_r(NULL, " \t", saveptr); if(cmd) { ERR("Unexpected junk on line: %s\n", cmd); return 0; } } if(!gotgains) { ERR("Matrix order_gain not specified\n"); return 0; } return 1; } void ambdec_init(AmbDecConf *conf) { ALsizei i; memset(conf, 0, sizeof(*conf)); AL_STRING_INIT(conf->Description); for(i = 0;i < MAX_OUTPUT_CHANNELS;i++) { AL_STRING_INIT(conf->Speakers[i].Name); AL_STRING_INIT(conf->Speakers[i].Connection); } } void ambdec_deinit(AmbDecConf *conf) { ALsizei i; alstr_reset(&conf->Description); for(i = 0;i < MAX_OUTPUT_CHANNELS;i++) { alstr_reset(&conf->Speakers[i].Name); alstr_reset(&conf->Speakers[i].Connection); } memset(conf, 0, sizeof(*conf)); } int ambdec_load(AmbDecConf *conf, const char *fname) { char *buffer = NULL; size_t maxlen = 0; char *line; FILE *f; f = al_fopen(fname, "r"); if(!f) { ERR("Failed to open: %s\n", fname); return 0; } while((line=read_clipped_line(f, &buffer, &maxlen)) != NULL) { char *saveptr; char *command; command = my_strtok_r(line, "/ \t", &saveptr); if(!command) { ERR("Malformed line: %s\n", line); goto fail; } if(strcmp(command, "description") == 0) { char *value = my_strtok_r(NULL, "", &saveptr); alstr_copy_cstr(&conf->Description, lstrip(value)); } else if(strcmp(command, "version") == 0) { line = my_strtok_r(NULL, "", &saveptr); line = read_uint(&conf->Version, line, 10); if(line && *line != '\0') { ERR("Extra junk after version: %s\n", line); goto fail; } if(conf->Version != 3) { ERR("Unsupported version: %u\n", conf->Version); goto fail; } } else if(strcmp(command, "dec") == 0) { const char *dec = my_strtok_r(NULL, "/ \t", &saveptr); if(strcmp(dec, "chan_mask") == 0) { line = my_strtok_r(NULL, "", &saveptr); line = read_uint(&conf->ChanMask, line, 16); if(line && *line != '\0') { ERR("Extra junk after mask: %s\n", line); goto fail; } } else if(strcmp(dec, "freq_bands") == 0) { line = my_strtok_r(NULL, "", &saveptr); line = read_uint(&conf->FreqBands, line, 10); if(line && *line != '\0') { ERR("Extra junk after freq_bands: %s\n", line); goto fail; } if(conf->FreqBands != 1 && conf->FreqBands != 2) { ERR("Invalid freq_bands value: %u\n", conf->FreqBands); goto fail; } } else if(strcmp(dec, "speakers") == 0) { line = my_strtok_r(NULL, "", &saveptr); line = read_int(&conf->NumSpeakers, line, 10); if(line && *line != '\0') { ERR("Extra junk after speakers: %s\n", line); goto fail; } if(conf->NumSpeakers > MAX_OUTPUT_CHANNELS) { ERR("Unsupported speaker count: %u\n", conf->NumSpeakers); goto fail; } } else if(strcmp(dec, "coeff_scale") == 0) { line = my_strtok_r(NULL, " \t", &saveptr); if(strcmp(line, "n3d") == 0) conf->CoeffScale = ADS_N3D; else if(strcmp(line, "sn3d") == 0) conf->CoeffScale = ADS_SN3D; else if(strcmp(line, "fuma") == 0) conf->CoeffScale = ADS_FuMa; else { ERR("Unsupported coeff scale: %s\n", line); goto fail; } } else { ERR("Unexpected /dec option: %s\n", dec); goto fail; } } else if(strcmp(command, "opt") == 0) { const char *opt = my_strtok_r(NULL, "/ \t", &saveptr); if(strcmp(opt, "xover_freq") == 0) { line = my_strtok_r(NULL, "", &saveptr); line = read_float(&conf->XOverFreq, line); if(line && *line != '\0') { ERR("Extra junk after xover_freq: %s\n", line); goto fail; } } else if(strcmp(opt, "xover_ratio") == 0) { line = my_strtok_r(NULL, "", &saveptr); line = read_float(&conf->XOverRatio, line); if(line && *line != '\0') { ERR("Extra junk after xover_ratio: %s\n", line); goto fail; } } else if(strcmp(opt, "input_scale") == 0 || strcmp(opt, "nfeff_comp") == 0 || strcmp(opt, "delay_comp") == 0 || strcmp(opt, "level_comp") == 0) { /* Unused */ my_strtok_r(NULL, " \t", &saveptr); } else { ERR("Unexpected /opt option: %s\n", opt); goto fail; } } else if(strcmp(command, "speakers") == 0) { const char *value = my_strtok_r(NULL, "/ \t", &saveptr); if(strcmp(value, "{") != 0) { ERR("Expected { after %s command, got %s\n", command, value); goto fail; } if(!load_ambdec_speakers(conf, f, &buffer, &maxlen, &saveptr)) goto fail; value = my_strtok_r(NULL, "/ \t", &saveptr); if(!value) { line = read_clipped_line(f, &buffer, &maxlen); if(!line) { ERR("Unexpected end of file\n"); goto fail; } value = my_strtok_r(line, "/ \t", &saveptr); } if(strcmp(value, "}") != 0) { ERR("Expected } after speaker definitions, got %s\n", value); goto fail; } } else if(strcmp(command, "lfmatrix") == 0 || strcmp(command, "hfmatrix") == 0 || strcmp(command, "matrix") == 0) { const char *value = my_strtok_r(NULL, "/ \t", &saveptr); if(strcmp(value, "{") != 0) { ERR("Expected { after %s command, got %s\n", command, value); goto fail; } if(conf->FreqBands == 1) { if(strcmp(command, "matrix") != 0) { ERR("Unexpected \"%s\" type for a single-band decoder\n", command); goto fail; } if(!load_ambdec_matrix(conf->HFOrderGain, conf->HFMatrix, conf->NumSpeakers, f, &buffer, &maxlen, &saveptr)) goto fail; } else { if(strcmp(command, "lfmatrix") == 0) { if(!load_ambdec_matrix(conf->LFOrderGain, conf->LFMatrix, conf->NumSpeakers, f, &buffer, &maxlen, &saveptr)) goto fail; } else if(strcmp(command, "hfmatrix") == 0) { if(!load_ambdec_matrix(conf->HFOrderGain, conf->HFMatrix, conf->NumSpeakers, f, &buffer, &maxlen, &saveptr)) goto fail; } else { ERR("Unexpected \"%s\" type for a dual-band decoder\n", command); goto fail; } } value = my_strtok_r(NULL, "/ \t", &saveptr); if(!value) { line = read_clipped_line(f, &buffer, &maxlen); if(!line) { ERR("Unexpected end of file\n"); goto fail; } value = my_strtok_r(line, "/ \t", &saveptr); } if(strcmp(value, "}") != 0) { ERR("Expected } after matrix definitions, got %s\n", value); goto fail; } } else if(strcmp(command, "end") == 0) { line = my_strtok_r(NULL, "/ \t", &saveptr); if(line) { ERR("Unexpected junk on end: %s\n", line); goto fail; } fclose(f); free(buffer); return 1; } else { ERR("Unexpected command: %s\n", command); goto fail; } line = my_strtok_r(NULL, "/ \t", &saveptr); if(line) { ERR("Unexpected junk on line: %s\n", line); goto fail; } } ERR("Unexpected end of file\n"); fail: fclose(f); free(buffer); return 0; } openal-soft-openal-soft-1.19.1/Alc/ambdec.h000066400000000000000000000020411335774445300203560ustar00rootroot00000000000000#ifndef AMBDEC_H #define AMBDEC_H #include "alstring.h" #include "alMain.h" /* Helpers to read .ambdec configuration files. */ enum AmbDecScaleType { ADS_N3D, ADS_SN3D, ADS_FuMa, }; typedef struct AmbDecConf { al_string Description; ALuint Version; /* Must be 3 */ ALuint ChanMask; ALuint FreqBands; /* Must be 1 or 2 */ ALsizei NumSpeakers; enum AmbDecScaleType CoeffScale; ALfloat XOverFreq; ALfloat XOverRatio; struct { al_string Name; ALfloat Distance; ALfloat Azimuth; ALfloat Elevation; al_string Connection; } Speakers[MAX_OUTPUT_CHANNELS]; /* Unused when FreqBands == 1 */ ALfloat LFOrderGain[MAX_AMBI_ORDER+1]; ALfloat LFMatrix[MAX_OUTPUT_CHANNELS][MAX_AMBI_COEFFS]; ALfloat HFOrderGain[MAX_AMBI_ORDER+1]; ALfloat HFMatrix[MAX_OUTPUT_CHANNELS][MAX_AMBI_COEFFS]; } AmbDecConf; void ambdec_init(AmbDecConf *conf); void ambdec_deinit(AmbDecConf *conf); int ambdec_load(AmbDecConf *conf, const char *fname); #endif /* AMBDEC_H */ openal-soft-openal-soft-1.19.1/Alc/backends/000077500000000000000000000000001335774445300205475ustar00rootroot00000000000000openal-soft-openal-soft-1.19.1/Alc/backends/alsa.c000066400000000000000000001451531335774445300216440ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include "alMain.h" #include "alu.h" #include "alconfig.h" #include "ringbuffer.h" #include "threads.h" #include "compat.h" #include "backends/base.h" #include static const ALCchar alsaDevice[] = "ALSA Default"; #ifdef HAVE_DYNLOAD #define ALSA_FUNCS(MAGIC) \ MAGIC(snd_strerror); \ MAGIC(snd_pcm_open); \ MAGIC(snd_pcm_close); \ MAGIC(snd_pcm_nonblock); \ MAGIC(snd_pcm_frames_to_bytes); \ MAGIC(snd_pcm_bytes_to_frames); \ MAGIC(snd_pcm_hw_params_malloc); \ MAGIC(snd_pcm_hw_params_free); \ MAGIC(snd_pcm_hw_params_any); \ MAGIC(snd_pcm_hw_params_current); \ MAGIC(snd_pcm_hw_params_set_access); \ MAGIC(snd_pcm_hw_params_set_format); \ MAGIC(snd_pcm_hw_params_set_channels); \ MAGIC(snd_pcm_hw_params_set_periods_near); \ MAGIC(snd_pcm_hw_params_set_rate_near); \ MAGIC(snd_pcm_hw_params_set_rate); \ MAGIC(snd_pcm_hw_params_set_rate_resample); \ MAGIC(snd_pcm_hw_params_set_buffer_time_near); \ MAGIC(snd_pcm_hw_params_set_period_time_near); \ MAGIC(snd_pcm_hw_params_set_buffer_size_near); \ MAGIC(snd_pcm_hw_params_set_period_size_near); \ MAGIC(snd_pcm_hw_params_set_buffer_size_min); \ MAGIC(snd_pcm_hw_params_get_buffer_time_min); \ MAGIC(snd_pcm_hw_params_get_buffer_time_max); \ MAGIC(snd_pcm_hw_params_get_period_time_min); \ MAGIC(snd_pcm_hw_params_get_period_time_max); \ MAGIC(snd_pcm_hw_params_get_buffer_size); \ MAGIC(snd_pcm_hw_params_get_period_size); \ MAGIC(snd_pcm_hw_params_get_access); \ MAGIC(snd_pcm_hw_params_get_periods); \ MAGIC(snd_pcm_hw_params_test_format); \ MAGIC(snd_pcm_hw_params_test_channels); \ MAGIC(snd_pcm_hw_params); \ MAGIC(snd_pcm_sw_params_malloc); \ MAGIC(snd_pcm_sw_params_current); \ MAGIC(snd_pcm_sw_params_set_avail_min); \ MAGIC(snd_pcm_sw_params_set_stop_threshold); \ MAGIC(snd_pcm_sw_params); \ MAGIC(snd_pcm_sw_params_free); \ MAGIC(snd_pcm_prepare); \ MAGIC(snd_pcm_start); \ MAGIC(snd_pcm_resume); \ MAGIC(snd_pcm_reset); \ MAGIC(snd_pcm_wait); \ MAGIC(snd_pcm_delay); \ MAGIC(snd_pcm_state); \ MAGIC(snd_pcm_avail_update); \ MAGIC(snd_pcm_areas_silence); \ MAGIC(snd_pcm_mmap_begin); \ MAGIC(snd_pcm_mmap_commit); \ MAGIC(snd_pcm_readi); \ MAGIC(snd_pcm_writei); \ MAGIC(snd_pcm_drain); \ MAGIC(snd_pcm_drop); \ MAGIC(snd_pcm_recover); \ MAGIC(snd_pcm_info_malloc); \ MAGIC(snd_pcm_info_free); \ MAGIC(snd_pcm_info_set_device); \ MAGIC(snd_pcm_info_set_subdevice); \ MAGIC(snd_pcm_info_set_stream); \ MAGIC(snd_pcm_info_get_name); \ MAGIC(snd_ctl_pcm_next_device); \ MAGIC(snd_ctl_pcm_info); \ MAGIC(snd_ctl_open); \ MAGIC(snd_ctl_close); \ MAGIC(snd_ctl_card_info_malloc); \ MAGIC(snd_ctl_card_info_free); \ MAGIC(snd_ctl_card_info); \ MAGIC(snd_ctl_card_info_get_name); \ MAGIC(snd_ctl_card_info_get_id); \ MAGIC(snd_card_next); \ MAGIC(snd_config_update_free_global) static void *alsa_handle; #define MAKE_FUNC(f) static __typeof(f) * p##f ALSA_FUNCS(MAKE_FUNC); #undef MAKE_FUNC #define snd_strerror psnd_strerror #define snd_pcm_open psnd_pcm_open #define snd_pcm_close psnd_pcm_close #define snd_pcm_nonblock psnd_pcm_nonblock #define snd_pcm_frames_to_bytes psnd_pcm_frames_to_bytes #define snd_pcm_bytes_to_frames psnd_pcm_bytes_to_frames #define snd_pcm_hw_params_malloc psnd_pcm_hw_params_malloc #define snd_pcm_hw_params_free psnd_pcm_hw_params_free #define snd_pcm_hw_params_any psnd_pcm_hw_params_any #define snd_pcm_hw_params_current psnd_pcm_hw_params_current #define snd_pcm_hw_params_set_access psnd_pcm_hw_params_set_access #define snd_pcm_hw_params_set_format psnd_pcm_hw_params_set_format #define snd_pcm_hw_params_set_channels psnd_pcm_hw_params_set_channels #define snd_pcm_hw_params_set_periods_near psnd_pcm_hw_params_set_periods_near #define snd_pcm_hw_params_set_rate_near psnd_pcm_hw_params_set_rate_near #define snd_pcm_hw_params_set_rate psnd_pcm_hw_params_set_rate #define snd_pcm_hw_params_set_rate_resample psnd_pcm_hw_params_set_rate_resample #define snd_pcm_hw_params_set_buffer_time_near psnd_pcm_hw_params_set_buffer_time_near #define snd_pcm_hw_params_set_period_time_near psnd_pcm_hw_params_set_period_time_near #define snd_pcm_hw_params_set_buffer_size_near psnd_pcm_hw_params_set_buffer_size_near #define snd_pcm_hw_params_set_period_size_near psnd_pcm_hw_params_set_period_size_near #define snd_pcm_hw_params_set_buffer_size_min psnd_pcm_hw_params_set_buffer_size_min #define snd_pcm_hw_params_get_buffer_time_min psnd_pcm_hw_params_get_buffer_time_min #define snd_pcm_hw_params_get_buffer_time_max psnd_pcm_hw_params_get_buffer_time_max #define snd_pcm_hw_params_get_period_time_min psnd_pcm_hw_params_get_period_time_min #define snd_pcm_hw_params_get_period_time_max psnd_pcm_hw_params_get_period_time_max #define snd_pcm_hw_params_get_buffer_size psnd_pcm_hw_params_get_buffer_size #define snd_pcm_hw_params_get_period_size psnd_pcm_hw_params_get_period_size #define snd_pcm_hw_params_get_access psnd_pcm_hw_params_get_access #define snd_pcm_hw_params_get_periods psnd_pcm_hw_params_get_periods #define snd_pcm_hw_params_test_format psnd_pcm_hw_params_test_format #define snd_pcm_hw_params_test_channels psnd_pcm_hw_params_test_channels #define snd_pcm_hw_params psnd_pcm_hw_params #define snd_pcm_sw_params_malloc psnd_pcm_sw_params_malloc #define snd_pcm_sw_params_current psnd_pcm_sw_params_current #define snd_pcm_sw_params_set_avail_min psnd_pcm_sw_params_set_avail_min #define snd_pcm_sw_params_set_stop_threshold psnd_pcm_sw_params_set_stop_threshold #define snd_pcm_sw_params psnd_pcm_sw_params #define snd_pcm_sw_params_free psnd_pcm_sw_params_free #define snd_pcm_prepare psnd_pcm_prepare #define snd_pcm_start psnd_pcm_start #define snd_pcm_resume psnd_pcm_resume #define snd_pcm_reset psnd_pcm_reset #define snd_pcm_wait psnd_pcm_wait #define snd_pcm_delay psnd_pcm_delay #define snd_pcm_state psnd_pcm_state #define snd_pcm_avail_update psnd_pcm_avail_update #define snd_pcm_areas_silence psnd_pcm_areas_silence #define snd_pcm_mmap_begin psnd_pcm_mmap_begin #define snd_pcm_mmap_commit psnd_pcm_mmap_commit #define snd_pcm_readi psnd_pcm_readi #define snd_pcm_writei psnd_pcm_writei #define snd_pcm_drain psnd_pcm_drain #define snd_pcm_drop psnd_pcm_drop #define snd_pcm_recover psnd_pcm_recover #define snd_pcm_info_malloc psnd_pcm_info_malloc #define snd_pcm_info_free psnd_pcm_info_free #define snd_pcm_info_set_device psnd_pcm_info_set_device #define snd_pcm_info_set_subdevice psnd_pcm_info_set_subdevice #define snd_pcm_info_set_stream psnd_pcm_info_set_stream #define snd_pcm_info_get_name psnd_pcm_info_get_name #define snd_ctl_pcm_next_device psnd_ctl_pcm_next_device #define snd_ctl_pcm_info psnd_ctl_pcm_info #define snd_ctl_open psnd_ctl_open #define snd_ctl_close psnd_ctl_close #define snd_ctl_card_info_malloc psnd_ctl_card_info_malloc #define snd_ctl_card_info_free psnd_ctl_card_info_free #define snd_ctl_card_info psnd_ctl_card_info #define snd_ctl_card_info_get_name psnd_ctl_card_info_get_name #define snd_ctl_card_info_get_id psnd_ctl_card_info_get_id #define snd_card_next psnd_card_next #define snd_config_update_free_global psnd_config_update_free_global #endif static ALCboolean alsa_load(void) { ALCboolean error = ALC_FALSE; #ifdef HAVE_DYNLOAD if(!alsa_handle) { al_string missing_funcs = AL_STRING_INIT_STATIC(); alsa_handle = LoadLib("libasound.so.2"); if(!alsa_handle) { WARN("Failed to load %s\n", "libasound.so.2"); return ALC_FALSE; } error = ALC_FALSE; #define LOAD_FUNC(f) do { \ p##f = GetSymbol(alsa_handle, #f); \ if(p##f == NULL) { \ error = ALC_TRUE; \ alstr_append_cstr(&missing_funcs, "\n" #f); \ } \ } while(0) ALSA_FUNCS(LOAD_FUNC); #undef LOAD_FUNC if(error) { WARN("Missing expected functions:%s\n", alstr_get_cstr(missing_funcs)); CloseLib(alsa_handle); alsa_handle = NULL; } alstr_reset(&missing_funcs); } #endif return !error; } typedef struct { al_string name; al_string device_name; } DevMap; TYPEDEF_VECTOR(DevMap, vector_DevMap) static vector_DevMap PlaybackDevices; static vector_DevMap CaptureDevices; static void clear_devlist(vector_DevMap *devlist) { #define FREE_DEV(i) do { \ AL_STRING_DEINIT((i)->name); \ AL_STRING_DEINIT((i)->device_name); \ } while(0) VECTOR_FOR_EACH(DevMap, *devlist, FREE_DEV); VECTOR_RESIZE(*devlist, 0, 0); #undef FREE_DEV } static const char *prefix_name(snd_pcm_stream_t stream) { assert(stream == SND_PCM_STREAM_PLAYBACK || stream == SND_PCM_STREAM_CAPTURE); return (stream==SND_PCM_STREAM_PLAYBACK) ? "device-prefix" : "capture-prefix"; } static void probe_devices(snd_pcm_stream_t stream, vector_DevMap *DeviceList) { const char *main_prefix = "plughw:"; snd_ctl_t *handle; snd_ctl_card_info_t *info; snd_pcm_info_t *pcminfo; int card, err, dev; DevMap entry; clear_devlist(DeviceList); snd_ctl_card_info_malloc(&info); snd_pcm_info_malloc(&pcminfo); AL_STRING_INIT(entry.name); AL_STRING_INIT(entry.device_name); alstr_copy_cstr(&entry.name, alsaDevice); alstr_copy_cstr(&entry.device_name, GetConfigValue( NULL, "alsa", (stream==SND_PCM_STREAM_PLAYBACK) ? "device" : "capture", "default" )); VECTOR_PUSH_BACK(*DeviceList, entry); if(stream == SND_PCM_STREAM_PLAYBACK) { const char *customdevs, *sep, *next; next = GetConfigValue(NULL, "alsa", "custom-devices", ""); while((customdevs=next) != NULL && customdevs[0]) { next = strchr(customdevs, ';'); sep = strchr(customdevs, '='); if(!sep) { al_string spec = AL_STRING_INIT_STATIC(); if(next) alstr_copy_range(&spec, customdevs, next++); else alstr_copy_cstr(&spec, customdevs); ERR("Invalid ALSA device specification \"%s\"\n", alstr_get_cstr(spec)); alstr_reset(&spec); continue; } AL_STRING_INIT(entry.name); AL_STRING_INIT(entry.device_name); alstr_copy_range(&entry.name, customdevs, sep++); if(next) alstr_copy_range(&entry.device_name, sep, next++); else alstr_copy_cstr(&entry.device_name, sep); TRACE("Got device \"%s\", \"%s\"\n", alstr_get_cstr(entry.name), alstr_get_cstr(entry.device_name)); VECTOR_PUSH_BACK(*DeviceList, entry); } } card = -1; if((err=snd_card_next(&card)) < 0) ERR("Failed to find a card: %s\n", snd_strerror(err)); ConfigValueStr(NULL, "alsa", prefix_name(stream), &main_prefix); while(card >= 0) { const char *card_prefix = main_prefix; const char *cardname, *cardid; char name[256]; snprintf(name, sizeof(name), "hw:%d", card); if((err = snd_ctl_open(&handle, name, 0)) < 0) { ERR("control open (hw:%d): %s\n", card, snd_strerror(err)); goto next_card; } if((err = snd_ctl_card_info(handle, info)) < 0) { ERR("control hardware info (hw:%d): %s\n", card, snd_strerror(err)); snd_ctl_close(handle); goto next_card; } cardname = snd_ctl_card_info_get_name(info); cardid = snd_ctl_card_info_get_id(info); snprintf(name, sizeof(name), "%s-%s", prefix_name(stream), cardid); ConfigValueStr(NULL, "alsa", name, &card_prefix); dev = -1; while(1) { const char *device_prefix = card_prefix; const char *devname; char device[128]; if(snd_ctl_pcm_next_device(handle, &dev) < 0) ERR("snd_ctl_pcm_next_device failed\n"); if(dev < 0) break; snd_pcm_info_set_device(pcminfo, dev); snd_pcm_info_set_subdevice(pcminfo, 0); snd_pcm_info_set_stream(pcminfo, stream); if((err = snd_ctl_pcm_info(handle, pcminfo)) < 0) { if(err != -ENOENT) ERR("control digital audio info (hw:%d): %s\n", card, snd_strerror(err)); continue; } devname = snd_pcm_info_get_name(pcminfo); snprintf(name, sizeof(name), "%s-%s-%d", prefix_name(stream), cardid, dev); ConfigValueStr(NULL, "alsa", name, &device_prefix); snprintf(name, sizeof(name), "%s, %s (CARD=%s,DEV=%d)", cardname, devname, cardid, dev); snprintf(device, sizeof(device), "%sCARD=%s,DEV=%d", device_prefix, cardid, dev); TRACE("Got device \"%s\", \"%s\"\n", name, device); AL_STRING_INIT(entry.name); AL_STRING_INIT(entry.device_name); alstr_copy_cstr(&entry.name, name); alstr_copy_cstr(&entry.device_name, device); VECTOR_PUSH_BACK(*DeviceList, entry); } snd_ctl_close(handle); next_card: if(snd_card_next(&card) < 0) { ERR("snd_card_next failed\n"); break; } } snd_pcm_info_free(pcminfo); snd_ctl_card_info_free(info); } static int verify_state(snd_pcm_t *handle) { snd_pcm_state_t state = snd_pcm_state(handle); int err; switch(state) { case SND_PCM_STATE_OPEN: case SND_PCM_STATE_SETUP: case SND_PCM_STATE_PREPARED: case SND_PCM_STATE_RUNNING: case SND_PCM_STATE_DRAINING: case SND_PCM_STATE_PAUSED: /* All Okay */ break; case SND_PCM_STATE_XRUN: if((err=snd_pcm_recover(handle, -EPIPE, 1)) < 0) return err; break; case SND_PCM_STATE_SUSPENDED: if((err=snd_pcm_recover(handle, -ESTRPIPE, 1)) < 0) return err; break; case SND_PCM_STATE_DISCONNECTED: return -ENODEV; } return state; } typedef struct ALCplaybackAlsa { DERIVE_FROM_TYPE(ALCbackend); snd_pcm_t *pcmHandle; ALvoid *buffer; ALsizei size; ATOMIC(ALenum) killNow; althrd_t thread; } ALCplaybackAlsa; static int ALCplaybackAlsa_mixerProc(void *ptr); static int ALCplaybackAlsa_mixerNoMMapProc(void *ptr); static void ALCplaybackAlsa_Construct(ALCplaybackAlsa *self, ALCdevice *device); static void ALCplaybackAlsa_Destruct(ALCplaybackAlsa *self); static ALCenum ALCplaybackAlsa_open(ALCplaybackAlsa *self, const ALCchar *name); static ALCboolean ALCplaybackAlsa_reset(ALCplaybackAlsa *self); static ALCboolean ALCplaybackAlsa_start(ALCplaybackAlsa *self); static void ALCplaybackAlsa_stop(ALCplaybackAlsa *self); static DECLARE_FORWARD2(ALCplaybackAlsa, ALCbackend, ALCenum, captureSamples, void*, ALCuint) static DECLARE_FORWARD(ALCplaybackAlsa, ALCbackend, ALCuint, availableSamples) static ClockLatency ALCplaybackAlsa_getClockLatency(ALCplaybackAlsa *self); static DECLARE_FORWARD(ALCplaybackAlsa, ALCbackend, void, lock) static DECLARE_FORWARD(ALCplaybackAlsa, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCplaybackAlsa) DEFINE_ALCBACKEND_VTABLE(ALCplaybackAlsa); static void ALCplaybackAlsa_Construct(ALCplaybackAlsa *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCplaybackAlsa, ALCbackend, self); self->pcmHandle = NULL; self->buffer = NULL; ATOMIC_INIT(&self->killNow, AL_TRUE); } void ALCplaybackAlsa_Destruct(ALCplaybackAlsa *self) { if(self->pcmHandle) snd_pcm_close(self->pcmHandle); self->pcmHandle = NULL; ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static int ALCplaybackAlsa_mixerProc(void *ptr) { ALCplaybackAlsa *self = (ALCplaybackAlsa*)ptr; ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; const snd_pcm_channel_area_t *areas = NULL; snd_pcm_uframes_t update_size, num_updates; snd_pcm_sframes_t avail, commitres; snd_pcm_uframes_t offset, frames; char *WritePtr; int err; SetRTPriority(); althrd_setname(althrd_current(), MIXER_THREAD_NAME); update_size = device->UpdateSize; num_updates = device->NumUpdates; while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire)) { int state = verify_state(self->pcmHandle); if(state < 0) { ERR("Invalid state detected: %s\n", snd_strerror(state)); ALCplaybackAlsa_lock(self); aluHandleDisconnect(device, "Bad state: %s", snd_strerror(state)); ALCplaybackAlsa_unlock(self); break; } avail = snd_pcm_avail_update(self->pcmHandle); if(avail < 0) { ERR("available update failed: %s\n", snd_strerror(avail)); continue; } if((snd_pcm_uframes_t)avail > update_size*(num_updates+1)) { WARN("available samples exceeds the buffer size\n"); snd_pcm_reset(self->pcmHandle); continue; } // make sure there's frames to process if((snd_pcm_uframes_t)avail < update_size) { if(state != SND_PCM_STATE_RUNNING) { err = snd_pcm_start(self->pcmHandle); if(err < 0) { ERR("start failed: %s\n", snd_strerror(err)); continue; } } if(snd_pcm_wait(self->pcmHandle, 1000) == 0) ERR("Wait timeout... buffer size too low?\n"); continue; } avail -= avail%update_size; // it is possible that contiguous areas are smaller, thus we use a loop ALCplaybackAlsa_lock(self); while(avail > 0) { frames = avail; err = snd_pcm_mmap_begin(self->pcmHandle, &areas, &offset, &frames); if(err < 0) { ERR("mmap begin error: %s\n", snd_strerror(err)); break; } WritePtr = (char*)areas->addr + (offset * areas->step / 8); aluMixData(device, WritePtr, frames); commitres = snd_pcm_mmap_commit(self->pcmHandle, offset, frames); if(commitres < 0 || (commitres-frames) != 0) { ERR("mmap commit error: %s\n", snd_strerror(commitres >= 0 ? -EPIPE : commitres)); break; } avail -= frames; } ALCplaybackAlsa_unlock(self); } return 0; } static int ALCplaybackAlsa_mixerNoMMapProc(void *ptr) { ALCplaybackAlsa *self = (ALCplaybackAlsa*)ptr; ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; snd_pcm_uframes_t update_size, num_updates; snd_pcm_sframes_t avail; char *WritePtr; int err; SetRTPriority(); althrd_setname(althrd_current(), MIXER_THREAD_NAME); update_size = device->UpdateSize; num_updates = device->NumUpdates; while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire)) { int state = verify_state(self->pcmHandle); if(state < 0) { ERR("Invalid state detected: %s\n", snd_strerror(state)); ALCplaybackAlsa_lock(self); aluHandleDisconnect(device, "Bad state: %s", snd_strerror(state)); ALCplaybackAlsa_unlock(self); break; } avail = snd_pcm_avail_update(self->pcmHandle); if(avail < 0) { ERR("available update failed: %s\n", snd_strerror(avail)); continue; } if((snd_pcm_uframes_t)avail > update_size*num_updates) { WARN("available samples exceeds the buffer size\n"); snd_pcm_reset(self->pcmHandle); continue; } if((snd_pcm_uframes_t)avail < update_size) { if(state != SND_PCM_STATE_RUNNING) { err = snd_pcm_start(self->pcmHandle); if(err < 0) { ERR("start failed: %s\n", snd_strerror(err)); continue; } } if(snd_pcm_wait(self->pcmHandle, 1000) == 0) ERR("Wait timeout... buffer size too low?\n"); continue; } ALCplaybackAlsa_lock(self); WritePtr = self->buffer; avail = snd_pcm_bytes_to_frames(self->pcmHandle, self->size); aluMixData(device, WritePtr, avail); while(avail > 0) { int ret = snd_pcm_writei(self->pcmHandle, WritePtr, avail); switch (ret) { case -EAGAIN: continue; #if ESTRPIPE != EPIPE case -ESTRPIPE: #endif case -EPIPE: case -EINTR: ret = snd_pcm_recover(self->pcmHandle, ret, 1); if(ret < 0) avail = 0; break; default: if (ret >= 0) { WritePtr += snd_pcm_frames_to_bytes(self->pcmHandle, ret); avail -= ret; } break; } if (ret < 0) { ret = snd_pcm_prepare(self->pcmHandle); if(ret < 0) break; } } ALCplaybackAlsa_unlock(self); } return 0; } static ALCenum ALCplaybackAlsa_open(ALCplaybackAlsa *self, const ALCchar *name) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; const char *driver = NULL; int err; if(name) { const DevMap *iter; if(VECTOR_SIZE(PlaybackDevices) == 0) probe_devices(SND_PCM_STREAM_PLAYBACK, &PlaybackDevices); #define MATCH_NAME(i) (alstr_cmp_cstr((i)->name, name) == 0) VECTOR_FIND_IF(iter, const DevMap, PlaybackDevices, MATCH_NAME); #undef MATCH_NAME if(iter == VECTOR_END(PlaybackDevices)) return ALC_INVALID_VALUE; driver = alstr_get_cstr(iter->device_name); } else { name = alsaDevice; driver = GetConfigValue(NULL, "alsa", "device", "default"); } TRACE("Opening device \"%s\"\n", driver); err = snd_pcm_open(&self->pcmHandle, driver, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); if(err < 0) { ERR("Could not open playback device '%s': %s\n", driver, snd_strerror(err)); return ALC_OUT_OF_MEMORY; } /* Free alsa's global config tree. Otherwise valgrind reports a ton of leaks. */ snd_config_update_free_global(); alstr_copy_cstr(&device->DeviceName, name); return ALC_NO_ERROR; } static ALCboolean ALCplaybackAlsa_reset(ALCplaybackAlsa *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; snd_pcm_uframes_t periodSizeInFrames; unsigned int periodLen, bufferLen; snd_pcm_sw_params_t *sp = NULL; snd_pcm_hw_params_t *hp = NULL; snd_pcm_format_t format = -1; snd_pcm_access_t access; unsigned int periods; unsigned int rate; const char *funcerr; int allowmmap; int dir; int err; switch(device->FmtType) { case DevFmtByte: format = SND_PCM_FORMAT_S8; break; case DevFmtUByte: format = SND_PCM_FORMAT_U8; break; case DevFmtShort: format = SND_PCM_FORMAT_S16; break; case DevFmtUShort: format = SND_PCM_FORMAT_U16; break; case DevFmtInt: format = SND_PCM_FORMAT_S32; break; case DevFmtUInt: format = SND_PCM_FORMAT_U32; break; case DevFmtFloat: format = SND_PCM_FORMAT_FLOAT; break; } allowmmap = GetConfigValueBool(alstr_get_cstr(device->DeviceName), "alsa", "mmap", 1); periods = device->NumUpdates; periodLen = (ALuint64)device->UpdateSize * 1000000 / device->Frequency; bufferLen = periodLen * periods; rate = device->Frequency; snd_pcm_hw_params_malloc(&hp); #define CHECK(x) if((funcerr=#x),(err=(x)) < 0) goto error CHECK(snd_pcm_hw_params_any(self->pcmHandle, hp)); /* set interleaved access */ if(!allowmmap || snd_pcm_hw_params_set_access(self->pcmHandle, hp, SND_PCM_ACCESS_MMAP_INTERLEAVED) < 0) { /* No mmap */ CHECK(snd_pcm_hw_params_set_access(self->pcmHandle, hp, SND_PCM_ACCESS_RW_INTERLEAVED)); } /* test and set format (implicitly sets sample bits) */ if(snd_pcm_hw_params_test_format(self->pcmHandle, hp, format) < 0) { static const struct { snd_pcm_format_t format; enum DevFmtType fmttype; } formatlist[] = { { SND_PCM_FORMAT_FLOAT, DevFmtFloat }, { SND_PCM_FORMAT_S32, DevFmtInt }, { SND_PCM_FORMAT_U32, DevFmtUInt }, { SND_PCM_FORMAT_S16, DevFmtShort }, { SND_PCM_FORMAT_U16, DevFmtUShort }, { SND_PCM_FORMAT_S8, DevFmtByte }, { SND_PCM_FORMAT_U8, DevFmtUByte }, }; size_t k; for(k = 0;k < COUNTOF(formatlist);k++) { format = formatlist[k].format; if(snd_pcm_hw_params_test_format(self->pcmHandle, hp, format) >= 0) { device->FmtType = formatlist[k].fmttype; break; } } } CHECK(snd_pcm_hw_params_set_format(self->pcmHandle, hp, format)); /* test and set channels (implicitly sets frame bits) */ if(snd_pcm_hw_params_test_channels(self->pcmHandle, hp, ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder)) < 0) { static const enum DevFmtChannels channellist[] = { DevFmtStereo, DevFmtQuad, DevFmtX51, DevFmtX71, DevFmtMono, }; size_t k; for(k = 0;k < COUNTOF(channellist);k++) { if(snd_pcm_hw_params_test_channels(self->pcmHandle, hp, ChannelsFromDevFmt(channellist[k], 0)) >= 0) { device->FmtChans = channellist[k]; device->AmbiOrder = 0; break; } } } CHECK(snd_pcm_hw_params_set_channels(self->pcmHandle, hp, ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder))); /* set rate (implicitly constrains period/buffer parameters) */ if(!GetConfigValueBool(alstr_get_cstr(device->DeviceName), "alsa", "allow-resampler", 0) || !(device->Flags&DEVICE_FREQUENCY_REQUEST)) { if(snd_pcm_hw_params_set_rate_resample(self->pcmHandle, hp, 0) < 0) ERR("Failed to disable ALSA resampler\n"); } else if(snd_pcm_hw_params_set_rate_resample(self->pcmHandle, hp, 1) < 0) ERR("Failed to enable ALSA resampler\n"); CHECK(snd_pcm_hw_params_set_rate_near(self->pcmHandle, hp, &rate, NULL)); /* set buffer time (implicitly constrains period/buffer parameters) */ if((err=snd_pcm_hw_params_set_buffer_time_near(self->pcmHandle, hp, &bufferLen, NULL)) < 0) ERR("snd_pcm_hw_params_set_buffer_time_near failed: %s\n", snd_strerror(err)); /* set period time (implicitly sets buffer size/bytes/time and period size/bytes) */ if((err=snd_pcm_hw_params_set_period_time_near(self->pcmHandle, hp, &periodLen, NULL)) < 0) ERR("snd_pcm_hw_params_set_period_time_near failed: %s\n", snd_strerror(err)); /* install and prepare hardware configuration */ CHECK(snd_pcm_hw_params(self->pcmHandle, hp)); /* retrieve configuration info */ CHECK(snd_pcm_hw_params_get_access(hp, &access)); CHECK(snd_pcm_hw_params_get_period_size(hp, &periodSizeInFrames, NULL)); CHECK(snd_pcm_hw_params_get_periods(hp, &periods, &dir)); if(dir != 0) WARN("Inexact period count: %u (%d)\n", periods, dir); snd_pcm_hw_params_free(hp); hp = NULL; snd_pcm_sw_params_malloc(&sp); CHECK(snd_pcm_sw_params_current(self->pcmHandle, sp)); CHECK(snd_pcm_sw_params_set_avail_min(self->pcmHandle, sp, periodSizeInFrames)); CHECK(snd_pcm_sw_params_set_stop_threshold(self->pcmHandle, sp, periodSizeInFrames*periods)); CHECK(snd_pcm_sw_params(self->pcmHandle, sp)); #undef CHECK snd_pcm_sw_params_free(sp); sp = NULL; device->NumUpdates = periods; device->UpdateSize = periodSizeInFrames; device->Frequency = rate; SetDefaultChannelOrder(device); return ALC_TRUE; error: ERR("%s failed: %s\n", funcerr, snd_strerror(err)); if(hp) snd_pcm_hw_params_free(hp); if(sp) snd_pcm_sw_params_free(sp); return ALC_FALSE; } static ALCboolean ALCplaybackAlsa_start(ALCplaybackAlsa *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; int (*thread_func)(void*) = NULL; snd_pcm_hw_params_t *hp = NULL; snd_pcm_access_t access; const char *funcerr; int err; snd_pcm_hw_params_malloc(&hp); #define CHECK(x) if((funcerr=#x),(err=(x)) < 0) goto error CHECK(snd_pcm_hw_params_current(self->pcmHandle, hp)); /* retrieve configuration info */ CHECK(snd_pcm_hw_params_get_access(hp, &access)); #undef CHECK snd_pcm_hw_params_free(hp); hp = NULL; self->size = snd_pcm_frames_to_bytes(self->pcmHandle, device->UpdateSize); if(access == SND_PCM_ACCESS_RW_INTERLEAVED) { self->buffer = al_malloc(16, self->size); if(!self->buffer) { ERR("buffer malloc failed\n"); return ALC_FALSE; } thread_func = ALCplaybackAlsa_mixerNoMMapProc; } else { err = snd_pcm_prepare(self->pcmHandle); if(err < 0) { ERR("snd_pcm_prepare(data->pcmHandle) failed: %s\n", snd_strerror(err)); return ALC_FALSE; } thread_func = ALCplaybackAlsa_mixerProc; } ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release); if(althrd_create(&self->thread, thread_func, self) != althrd_success) { ERR("Could not create playback thread\n"); al_free(self->buffer); self->buffer = NULL; return ALC_FALSE; } return ALC_TRUE; error: ERR("%s failed: %s\n", funcerr, snd_strerror(err)); if(hp) snd_pcm_hw_params_free(hp); return ALC_FALSE; } static void ALCplaybackAlsa_stop(ALCplaybackAlsa *self) { int res; if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel)) return; althrd_join(self->thread, &res); al_free(self->buffer); self->buffer = NULL; } static ClockLatency ALCplaybackAlsa_getClockLatency(ALCplaybackAlsa *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; snd_pcm_sframes_t delay = 0; ClockLatency ret; int err; ALCplaybackAlsa_lock(self); ret.ClockTime = GetDeviceClockTime(device); if((err=snd_pcm_delay(self->pcmHandle, &delay)) < 0) { ERR("Failed to get pcm delay: %s\n", snd_strerror(err)); delay = 0; } if(delay < 0) delay = 0; ret.Latency = delay * DEVICE_CLOCK_RES / device->Frequency; ALCplaybackAlsa_unlock(self); return ret; } typedef struct ALCcaptureAlsa { DERIVE_FROM_TYPE(ALCbackend); snd_pcm_t *pcmHandle; ALvoid *buffer; ALsizei size; ALboolean doCapture; ll_ringbuffer_t *ring; snd_pcm_sframes_t last_avail; } ALCcaptureAlsa; static void ALCcaptureAlsa_Construct(ALCcaptureAlsa *self, ALCdevice *device); static void ALCcaptureAlsa_Destruct(ALCcaptureAlsa *self); static ALCenum ALCcaptureAlsa_open(ALCcaptureAlsa *self, const ALCchar *name); static DECLARE_FORWARD(ALCcaptureAlsa, ALCbackend, ALCboolean, reset) static ALCboolean ALCcaptureAlsa_start(ALCcaptureAlsa *self); static void ALCcaptureAlsa_stop(ALCcaptureAlsa *self); static ALCenum ALCcaptureAlsa_captureSamples(ALCcaptureAlsa *self, ALCvoid *buffer, ALCuint samples); static ALCuint ALCcaptureAlsa_availableSamples(ALCcaptureAlsa *self); static ClockLatency ALCcaptureAlsa_getClockLatency(ALCcaptureAlsa *self); static DECLARE_FORWARD(ALCcaptureAlsa, ALCbackend, void, lock) static DECLARE_FORWARD(ALCcaptureAlsa, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCcaptureAlsa) DEFINE_ALCBACKEND_VTABLE(ALCcaptureAlsa); static void ALCcaptureAlsa_Construct(ALCcaptureAlsa *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCcaptureAlsa, ALCbackend, self); self->pcmHandle = NULL; self->buffer = NULL; self->ring = NULL; } void ALCcaptureAlsa_Destruct(ALCcaptureAlsa *self) { if(self->pcmHandle) snd_pcm_close(self->pcmHandle); self->pcmHandle = NULL; al_free(self->buffer); self->buffer = NULL; ll_ringbuffer_free(self->ring); self->ring = NULL; ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static ALCenum ALCcaptureAlsa_open(ALCcaptureAlsa *self, const ALCchar *name) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; const char *driver = NULL; snd_pcm_hw_params_t *hp; snd_pcm_uframes_t bufferSizeInFrames; snd_pcm_uframes_t periodSizeInFrames; ALboolean needring = AL_FALSE; snd_pcm_format_t format = -1; const char *funcerr; int err; if(name) { const DevMap *iter; if(VECTOR_SIZE(CaptureDevices) == 0) probe_devices(SND_PCM_STREAM_CAPTURE, &CaptureDevices); #define MATCH_NAME(i) (alstr_cmp_cstr((i)->name, name) == 0) VECTOR_FIND_IF(iter, const DevMap, CaptureDevices, MATCH_NAME); #undef MATCH_NAME if(iter == VECTOR_END(CaptureDevices)) return ALC_INVALID_VALUE; driver = alstr_get_cstr(iter->device_name); } else { name = alsaDevice; driver = GetConfigValue(NULL, "alsa", "capture", "default"); } TRACE("Opening device \"%s\"\n", driver); err = snd_pcm_open(&self->pcmHandle, driver, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK); if(err < 0) { ERR("Could not open capture device '%s': %s\n", driver, snd_strerror(err)); return ALC_INVALID_VALUE; } /* Free alsa's global config tree. Otherwise valgrind reports a ton of leaks. */ snd_config_update_free_global(); switch(device->FmtType) { case DevFmtByte: format = SND_PCM_FORMAT_S8; break; case DevFmtUByte: format = SND_PCM_FORMAT_U8; break; case DevFmtShort: format = SND_PCM_FORMAT_S16; break; case DevFmtUShort: format = SND_PCM_FORMAT_U16; break; case DevFmtInt: format = SND_PCM_FORMAT_S32; break; case DevFmtUInt: format = SND_PCM_FORMAT_U32; break; case DevFmtFloat: format = SND_PCM_FORMAT_FLOAT; break; } funcerr = NULL; bufferSizeInFrames = maxu(device->UpdateSize*device->NumUpdates, 100*device->Frequency/1000); periodSizeInFrames = minu(bufferSizeInFrames, 25*device->Frequency/1000); snd_pcm_hw_params_malloc(&hp); #define CHECK(x) if((funcerr=#x),(err=(x)) < 0) goto error CHECK(snd_pcm_hw_params_any(self->pcmHandle, hp)); /* set interleaved access */ CHECK(snd_pcm_hw_params_set_access(self->pcmHandle, hp, SND_PCM_ACCESS_RW_INTERLEAVED)); /* set format (implicitly sets sample bits) */ CHECK(snd_pcm_hw_params_set_format(self->pcmHandle, hp, format)); /* set channels (implicitly sets frame bits) */ CHECK(snd_pcm_hw_params_set_channels(self->pcmHandle, hp, ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder))); /* set rate (implicitly constrains period/buffer parameters) */ CHECK(snd_pcm_hw_params_set_rate(self->pcmHandle, hp, device->Frequency, 0)); /* set buffer size in frame units (implicitly sets period size/bytes/time and buffer time/bytes) */ if(snd_pcm_hw_params_set_buffer_size_min(self->pcmHandle, hp, &bufferSizeInFrames) < 0) { TRACE("Buffer too large, using intermediate ring buffer\n"); needring = AL_TRUE; CHECK(snd_pcm_hw_params_set_buffer_size_near(self->pcmHandle, hp, &bufferSizeInFrames)); } /* set buffer size in frame units (implicitly sets period size/bytes/time and buffer time/bytes) */ CHECK(snd_pcm_hw_params_set_period_size_near(self->pcmHandle, hp, &periodSizeInFrames, NULL)); /* install and prepare hardware configuration */ CHECK(snd_pcm_hw_params(self->pcmHandle, hp)); /* retrieve configuration info */ CHECK(snd_pcm_hw_params_get_period_size(hp, &periodSizeInFrames, NULL)); #undef CHECK snd_pcm_hw_params_free(hp); hp = NULL; if(needring) { self->ring = ll_ringbuffer_create( device->UpdateSize*device->NumUpdates, FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder), false ); if(!self->ring) { ERR("ring buffer create failed\n"); goto error2; } } alstr_copy_cstr(&device->DeviceName, name); return ALC_NO_ERROR; error: ERR("%s failed: %s\n", funcerr, snd_strerror(err)); if(hp) snd_pcm_hw_params_free(hp); error2: ll_ringbuffer_free(self->ring); self->ring = NULL; snd_pcm_close(self->pcmHandle); self->pcmHandle = NULL; return ALC_INVALID_VALUE; } static ALCboolean ALCcaptureAlsa_start(ALCcaptureAlsa *self) { int err = snd_pcm_prepare(self->pcmHandle); if(err < 0) ERR("prepare failed: %s\n", snd_strerror(err)); else { err = snd_pcm_start(self->pcmHandle); if(err < 0) ERR("start failed: %s\n", snd_strerror(err)); } if(err < 0) { aluHandleDisconnect(STATIC_CAST(ALCbackend, self)->mDevice, "Capture state failure: %s", snd_strerror(err)); return ALC_FALSE; } self->doCapture = AL_TRUE; return ALC_TRUE; } static void ALCcaptureAlsa_stop(ALCcaptureAlsa *self) { ALCuint avail; int err; /* OpenAL requires access to unread audio after stopping, but ALSA's * snd_pcm_drain is unreliable and snd_pcm_drop drops it. Capture what's * available now so it'll be available later after the drop. */ avail = ALCcaptureAlsa_availableSamples(self); if(!self->ring && avail > 0) { /* The ring buffer implicitly captures when checking availability. * Direct access needs to explicitly capture it into temp storage. */ ALsizei size; void *ptr; size = snd_pcm_frames_to_bytes(self->pcmHandle, avail); ptr = al_malloc(16, size); if(ptr) { ALCcaptureAlsa_captureSamples(self, ptr, avail); al_free(self->buffer); self->buffer = ptr; self->size = size; } } err = snd_pcm_drop(self->pcmHandle); if(err < 0) ERR("drop failed: %s\n", snd_strerror(err)); self->doCapture = AL_FALSE; } static ALCenum ALCcaptureAlsa_captureSamples(ALCcaptureAlsa *self, ALCvoid *buffer, ALCuint samples) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; if(self->ring) { ll_ringbuffer_read(self->ring, buffer, samples); return ALC_NO_ERROR; } self->last_avail -= samples; while(ATOMIC_LOAD(&device->Connected, almemory_order_acquire) && samples > 0) { snd_pcm_sframes_t amt = 0; if(self->size > 0) { /* First get any data stored from the last stop */ amt = snd_pcm_bytes_to_frames(self->pcmHandle, self->size); if((snd_pcm_uframes_t)amt > samples) amt = samples; amt = snd_pcm_frames_to_bytes(self->pcmHandle, amt); memcpy(buffer, self->buffer, amt); if(self->size > amt) { memmove(self->buffer, self->buffer+amt, self->size - amt); self->size -= amt; } else { al_free(self->buffer); self->buffer = NULL; self->size = 0; } amt = snd_pcm_bytes_to_frames(self->pcmHandle, amt); } else if(self->doCapture) amt = snd_pcm_readi(self->pcmHandle, buffer, samples); if(amt < 0) { ERR("read error: %s\n", snd_strerror(amt)); if(amt == -EAGAIN) continue; if((amt=snd_pcm_recover(self->pcmHandle, amt, 1)) >= 0) { amt = snd_pcm_start(self->pcmHandle); if(amt >= 0) amt = snd_pcm_avail_update(self->pcmHandle); } if(amt < 0) { ERR("restore error: %s\n", snd_strerror(amt)); aluHandleDisconnect(device, "Capture recovery failure: %s", snd_strerror(amt)); break; } /* If the amount available is less than what's asked, we lost it * during recovery. So just give silence instead. */ if((snd_pcm_uframes_t)amt < samples) break; continue; } buffer = (ALbyte*)buffer + amt; samples -= amt; } if(samples > 0) memset(buffer, ((device->FmtType == DevFmtUByte) ? 0x80 : 0), snd_pcm_frames_to_bytes(self->pcmHandle, samples)); return ALC_NO_ERROR; } static ALCuint ALCcaptureAlsa_availableSamples(ALCcaptureAlsa *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; snd_pcm_sframes_t avail = 0; if(ATOMIC_LOAD(&device->Connected, almemory_order_acquire) && self->doCapture) avail = snd_pcm_avail_update(self->pcmHandle); if(avail < 0) { ERR("avail update failed: %s\n", snd_strerror(avail)); if((avail=snd_pcm_recover(self->pcmHandle, avail, 1)) >= 0) { if(self->doCapture) avail = snd_pcm_start(self->pcmHandle); if(avail >= 0) avail = snd_pcm_avail_update(self->pcmHandle); } if(avail < 0) { ERR("restore error: %s\n", snd_strerror(avail)); aluHandleDisconnect(device, "Capture recovery failure: %s", snd_strerror(avail)); } } if(!self->ring) { if(avail < 0) avail = 0; avail += snd_pcm_bytes_to_frames(self->pcmHandle, self->size); if(avail > self->last_avail) self->last_avail = avail; return self->last_avail; } while(avail > 0) { ll_ringbuffer_data_t vec[2]; snd_pcm_sframes_t amt; ll_ringbuffer_get_write_vector(self->ring, vec); if(vec[0].len == 0) break; amt = (vec[0].len < (snd_pcm_uframes_t)avail) ? vec[0].len : (snd_pcm_uframes_t)avail; amt = snd_pcm_readi(self->pcmHandle, vec[0].buf, amt); if(amt < 0) { ERR("read error: %s\n", snd_strerror(amt)); if(amt == -EAGAIN) continue; if((amt=snd_pcm_recover(self->pcmHandle, amt, 1)) >= 0) { if(self->doCapture) amt = snd_pcm_start(self->pcmHandle); if(amt >= 0) amt = snd_pcm_avail_update(self->pcmHandle); } if(amt < 0) { ERR("restore error: %s\n", snd_strerror(amt)); aluHandleDisconnect(device, "Capture recovery failure: %s", snd_strerror(amt)); break; } avail = amt; continue; } ll_ringbuffer_write_advance(self->ring, amt); avail -= amt; } return ll_ringbuffer_read_space(self->ring); } static ClockLatency ALCcaptureAlsa_getClockLatency(ALCcaptureAlsa *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; snd_pcm_sframes_t delay = 0; ClockLatency ret; int err; ALCcaptureAlsa_lock(self); ret.ClockTime = GetDeviceClockTime(device); if((err=snd_pcm_delay(self->pcmHandle, &delay)) < 0) { ERR("Failed to get pcm delay: %s\n", snd_strerror(err)); delay = 0; } if(delay < 0) delay = 0; ret.Latency = delay * DEVICE_CLOCK_RES / device->Frequency; ALCcaptureAlsa_unlock(self); return ret; } typedef struct ALCalsaBackendFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } ALCalsaBackendFactory; #define ALCALSABACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCalsaBackendFactory, ALCbackendFactory) } } static ALCboolean ALCalsaBackendFactory_init(ALCalsaBackendFactory* UNUSED(self)) { VECTOR_INIT(PlaybackDevices); VECTOR_INIT(CaptureDevices); if(!alsa_load()) return ALC_FALSE; return ALC_TRUE; } static void ALCalsaBackendFactory_deinit(ALCalsaBackendFactory* UNUSED(self)) { clear_devlist(&PlaybackDevices); VECTOR_DEINIT(PlaybackDevices); clear_devlist(&CaptureDevices); VECTOR_DEINIT(CaptureDevices); #ifdef HAVE_DYNLOAD if(alsa_handle) CloseLib(alsa_handle); alsa_handle = NULL; #endif } static ALCboolean ALCalsaBackendFactory_querySupport(ALCalsaBackendFactory* UNUSED(self), ALCbackend_Type type) { if(type == ALCbackend_Playback || type == ALCbackend_Capture) return ALC_TRUE; return ALC_FALSE; } static void ALCalsaBackendFactory_probe(ALCalsaBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames) { switch(type) { #define APPEND_OUTNAME(i) do { \ if(!alstr_empty((i)->name)) \ alstr_append_range(outnames, VECTOR_BEGIN((i)->name), \ VECTOR_END((i)->name)+1); \ } while(0) case ALL_DEVICE_PROBE: probe_devices(SND_PCM_STREAM_PLAYBACK, &PlaybackDevices); VECTOR_FOR_EACH(const DevMap, PlaybackDevices, APPEND_OUTNAME); break; case CAPTURE_DEVICE_PROBE: probe_devices(SND_PCM_STREAM_CAPTURE, &CaptureDevices); VECTOR_FOR_EACH(const DevMap, CaptureDevices, APPEND_OUTNAME); break; #undef APPEND_OUTNAME } } static ALCbackend* ALCalsaBackendFactory_createBackend(ALCalsaBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) { if(type == ALCbackend_Playback) { ALCplaybackAlsa *backend; NEW_OBJ(backend, ALCplaybackAlsa)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } if(type == ALCbackend_Capture) { ALCcaptureAlsa *backend; NEW_OBJ(backend, ALCcaptureAlsa)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } return NULL; } DEFINE_ALCBACKENDFACTORY_VTABLE(ALCalsaBackendFactory); ALCbackendFactory *ALCalsaBackendFactory_getFactory(void) { static ALCalsaBackendFactory factory = ALCALSABACKENDFACTORY_INITIALIZER; return STATIC_CAST(ALCbackendFactory, &factory); } openal-soft-openal-soft-1.19.1/Alc/backends/base.c000066400000000000000000000041461335774445300216320ustar00rootroot00000000000000 #include "config.h" #include #include "alMain.h" #include "alu.h" #include "backends/base.h" extern inline ALuint64 GetDeviceClockTime(ALCdevice *device); extern inline void ALCdevice_Lock(ALCdevice *device); extern inline void ALCdevice_Unlock(ALCdevice *device); extern inline ClockLatency GetClockLatency(ALCdevice *device); /* Base ALCbackend method implementations. */ void ALCbackend_Construct(ALCbackend *self, ALCdevice *device) { int ret = almtx_init(&self->mMutex, almtx_recursive); assert(ret == althrd_success); self->mDevice = device; } void ALCbackend_Destruct(ALCbackend *self) { almtx_destroy(&self->mMutex); } ALCboolean ALCbackend_reset(ALCbackend* UNUSED(self)) { return ALC_FALSE; } ALCenum ALCbackend_captureSamples(ALCbackend* UNUSED(self), void* UNUSED(buffer), ALCuint UNUSED(samples)) { return ALC_INVALID_DEVICE; } ALCuint ALCbackend_availableSamples(ALCbackend* UNUSED(self)) { return 0; } ClockLatency ALCbackend_getClockLatency(ALCbackend *self) { ALCdevice *device = self->mDevice; ALuint refcount; ClockLatency ret; do { while(((refcount=ATOMIC_LOAD(&device->MixCount, almemory_order_acquire))&1)) althrd_yield(); ret.ClockTime = GetDeviceClockTime(device); ATOMIC_THREAD_FENCE(almemory_order_acquire); } while(refcount != ATOMIC_LOAD(&device->MixCount, almemory_order_relaxed)); /* NOTE: The device will generally have about all but one periods filled at * any given time during playback. Without a more accurate measurement from * the output, this is an okay approximation. */ ret.Latency = device->UpdateSize * DEVICE_CLOCK_RES / device->Frequency * maxu(device->NumUpdates-1, 1); return ret; } void ALCbackend_lock(ALCbackend *self) { int ret = almtx_lock(&self->mMutex); assert(ret == althrd_success); } void ALCbackend_unlock(ALCbackend *self) { int ret = almtx_unlock(&self->mMutex); assert(ret == althrd_success); } /* Base ALCbackendFactory method implementations. */ void ALCbackendFactory_deinit(ALCbackendFactory* UNUSED(self)) { } openal-soft-openal-soft-1.19.1/Alc/backends/base.h000066400000000000000000000160161335774445300216360ustar00rootroot00000000000000#ifndef AL_BACKENDS_BASE_H #define AL_BACKENDS_BASE_H #include "alMain.h" #include "threads.h" #include "alstring.h" #ifdef __cplusplus extern "C" { #endif typedef struct ClockLatency { ALint64 ClockTime; ALint64 Latency; } ClockLatency; /* Helper to get the current clock time from the device's ClockBase, and * SamplesDone converted from the sample rate. */ inline ALuint64 GetDeviceClockTime(ALCdevice *device) { return device->ClockBase + (device->SamplesDone * DEVICE_CLOCK_RES / device->Frequency); } struct ALCbackendVtable; typedef struct ALCbackend { const struct ALCbackendVtable *vtbl; ALCdevice *mDevice; almtx_t mMutex; } ALCbackend; void ALCbackend_Construct(ALCbackend *self, ALCdevice *device); void ALCbackend_Destruct(ALCbackend *self); ALCboolean ALCbackend_reset(ALCbackend *self); ALCenum ALCbackend_captureSamples(ALCbackend *self, void *buffer, ALCuint samples); ALCuint ALCbackend_availableSamples(ALCbackend *self); ClockLatency ALCbackend_getClockLatency(ALCbackend *self); void ALCbackend_lock(ALCbackend *self); void ALCbackend_unlock(ALCbackend *self); struct ALCbackendVtable { void (*const Destruct)(ALCbackend*); ALCenum (*const open)(ALCbackend*, const ALCchar*); ALCboolean (*const reset)(ALCbackend*); ALCboolean (*const start)(ALCbackend*); void (*const stop)(ALCbackend*); ALCenum (*const captureSamples)(ALCbackend*, void*, ALCuint); ALCuint (*const availableSamples)(ALCbackend*); ClockLatency (*const getClockLatency)(ALCbackend*); void (*const lock)(ALCbackend*); void (*const unlock)(ALCbackend*); void (*const Delete)(void*); }; #define DEFINE_ALCBACKEND_VTABLE(T) \ DECLARE_THUNK(T, ALCbackend, void, Destruct) \ DECLARE_THUNK1(T, ALCbackend, ALCenum, open, const ALCchar*) \ DECLARE_THUNK(T, ALCbackend, ALCboolean, reset) \ DECLARE_THUNK(T, ALCbackend, ALCboolean, start) \ DECLARE_THUNK(T, ALCbackend, void, stop) \ DECLARE_THUNK2(T, ALCbackend, ALCenum, captureSamples, void*, ALCuint) \ DECLARE_THUNK(T, ALCbackend, ALCuint, availableSamples) \ DECLARE_THUNK(T, ALCbackend, ClockLatency, getClockLatency) \ DECLARE_THUNK(T, ALCbackend, void, lock) \ DECLARE_THUNK(T, ALCbackend, void, unlock) \ static void T##_ALCbackend_Delete(void *ptr) \ { T##_Delete(STATIC_UPCAST(T, ALCbackend, (ALCbackend*)ptr)); } \ \ static const struct ALCbackendVtable T##_ALCbackend_vtable = { \ T##_ALCbackend_Destruct, \ \ T##_ALCbackend_open, \ T##_ALCbackend_reset, \ T##_ALCbackend_start, \ T##_ALCbackend_stop, \ T##_ALCbackend_captureSamples, \ T##_ALCbackend_availableSamples, \ T##_ALCbackend_getClockLatency, \ T##_ALCbackend_lock, \ T##_ALCbackend_unlock, \ \ T##_ALCbackend_Delete, \ } typedef enum ALCbackend_Type { ALCbackend_Playback, ALCbackend_Capture, ALCbackend_Loopback } ALCbackend_Type; struct ALCbackendFactoryVtable; typedef struct ALCbackendFactory { const struct ALCbackendFactoryVtable *vtbl; } ALCbackendFactory; void ALCbackendFactory_deinit(ALCbackendFactory *self); struct ALCbackendFactoryVtable { ALCboolean (*const init)(ALCbackendFactory *self); void (*const deinit)(ALCbackendFactory *self); ALCboolean (*const querySupport)(ALCbackendFactory *self, ALCbackend_Type type); void (*const probe)(ALCbackendFactory *self, enum DevProbe type, al_string *outnames); ALCbackend* (*const createBackend)(ALCbackendFactory *self, ALCdevice *device, ALCbackend_Type type); }; #define DEFINE_ALCBACKENDFACTORY_VTABLE(T) \ DECLARE_THUNK(T, ALCbackendFactory, ALCboolean, init) \ DECLARE_THUNK(T, ALCbackendFactory, void, deinit) \ DECLARE_THUNK1(T, ALCbackendFactory, ALCboolean, querySupport, ALCbackend_Type) \ DECLARE_THUNK2(T, ALCbackendFactory, void, probe, enum DevProbe, al_string*) \ DECLARE_THUNK2(T, ALCbackendFactory, ALCbackend*, createBackend, ALCdevice*, ALCbackend_Type) \ \ static const struct ALCbackendFactoryVtable T##_ALCbackendFactory_vtable = { \ T##_ALCbackendFactory_init, \ T##_ALCbackendFactory_deinit, \ T##_ALCbackendFactory_querySupport, \ T##_ALCbackendFactory_probe, \ T##_ALCbackendFactory_createBackend, \ } ALCbackendFactory *ALCpulseBackendFactory_getFactory(void); ALCbackendFactory *ALCalsaBackendFactory_getFactory(void); ALCbackendFactory *ALCcoreAudioBackendFactory_getFactory(void); ALCbackendFactory *ALCossBackendFactory_getFactory(void); ALCbackendFactory *ALCjackBackendFactory_getFactory(void); ALCbackendFactory *ALCsolarisBackendFactory_getFactory(void); ALCbackendFactory *SndioBackendFactory_getFactory(void); ALCbackendFactory *ALCqsaBackendFactory_getFactory(void); ALCbackendFactory *ALCwasapiBackendFactory_getFactory(void); ALCbackendFactory *ALCdsoundBackendFactory_getFactory(void); ALCbackendFactory *ALCwinmmBackendFactory_getFactory(void); ALCbackendFactory *ALCportBackendFactory_getFactory(void); ALCbackendFactory *ALCopenslBackendFactory_getFactory(void); ALCbackendFactory *ALCnullBackendFactory_getFactory(void); ALCbackendFactory *ALCwaveBackendFactory_getFactory(void); ALCbackendFactory *ALCsdl2BackendFactory_getFactory(void); ALCbackendFactory *ALCloopbackFactory_getFactory(void); inline void ALCdevice_Lock(ALCdevice *device) { V0(device->Backend,lock)(); } inline void ALCdevice_Unlock(ALCdevice *device) { V0(device->Backend,unlock)(); } inline ClockLatency GetClockLatency(ALCdevice *device) { ClockLatency ret = V0(device->Backend,getClockLatency)(); ret.Latency += device->FixedLatency; return ret; } #ifdef __cplusplus } /* extern "C" */ #endif #endif /* AL_BACKENDS_BASE_H */ openal-soft-openal-soft-1.19.1/Alc/backends/coreaudio.c000066400000000000000000000665101335774445300226750ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include "alMain.h" #include "alu.h" #include "ringbuffer.h" #include #include #include #include "backends/base.h" static const ALCchar ca_device[] = "CoreAudio Default"; typedef struct ALCcoreAudioPlayback { DERIVE_FROM_TYPE(ALCbackend); AudioUnit audioUnit; ALuint frameSize; AudioStreamBasicDescription format; // This is the OpenAL format as a CoreAudio ASBD } ALCcoreAudioPlayback; static void ALCcoreAudioPlayback_Construct(ALCcoreAudioPlayback *self, ALCdevice *device); static void ALCcoreAudioPlayback_Destruct(ALCcoreAudioPlayback *self); static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCchar *name); static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self); static ALCboolean ALCcoreAudioPlayback_start(ALCcoreAudioPlayback *self); static void ALCcoreAudioPlayback_stop(ALCcoreAudioPlayback *self); static DECLARE_FORWARD2(ALCcoreAudioPlayback, ALCbackend, ALCenum, captureSamples, void*, ALCuint) static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, ALCuint, availableSamples) static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, void, lock) static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCcoreAudioPlayback) DEFINE_ALCBACKEND_VTABLE(ALCcoreAudioPlayback); static void ALCcoreAudioPlayback_Construct(ALCcoreAudioPlayback *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCcoreAudioPlayback, ALCbackend, self); self->frameSize = 0; memset(&self->format, 0, sizeof(self->format)); } static void ALCcoreAudioPlayback_Destruct(ALCcoreAudioPlayback *self) { AudioUnitUninitialize(self->audioUnit); AudioComponentInstanceDispose(self->audioUnit); ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static OSStatus ALCcoreAudioPlayback_MixerProc(void *inRefCon, AudioUnitRenderActionFlags* UNUSED(ioActionFlags), const AudioTimeStamp* UNUSED(inTimeStamp), UInt32 UNUSED(inBusNumber), UInt32 UNUSED(inNumberFrames), AudioBufferList *ioData) { ALCcoreAudioPlayback *self = inRefCon; ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; ALCcoreAudioPlayback_lock(self); aluMixData(device, ioData->mBuffers[0].mData, ioData->mBuffers[0].mDataByteSize / self->frameSize); ALCcoreAudioPlayback_unlock(self); return noErr; } static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCchar *name) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; AudioComponentDescription desc; AudioComponent comp; OSStatus err; if(!name) name = ca_device; else if(strcmp(name, ca_device) != 0) return ALC_INVALID_VALUE; /* open the default output unit */ desc.componentType = kAudioUnitType_Output; #if TARGET_OS_IOS desc.componentSubType = kAudioUnitSubType_RemoteIO; #else desc.componentSubType = kAudioUnitSubType_DefaultOutput; #endif desc.componentManufacturer = kAudioUnitManufacturer_Apple; desc.componentFlags = 0; desc.componentFlagsMask = 0; comp = AudioComponentFindNext(NULL, &desc); if(comp == NULL) { ERR("AudioComponentFindNext failed\n"); return ALC_INVALID_VALUE; } err = AudioComponentInstanceNew(comp, &self->audioUnit); if(err != noErr) { ERR("AudioComponentInstanceNew failed\n"); return ALC_INVALID_VALUE; } /* init and start the default audio unit... */ err = AudioUnitInitialize(self->audioUnit); if(err != noErr) { ERR("AudioUnitInitialize failed\n"); AudioComponentInstanceDispose(self->audioUnit); return ALC_INVALID_VALUE; } alstr_copy_cstr(&device->DeviceName, name); return ALC_NO_ERROR; } static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; AudioStreamBasicDescription streamFormat; AURenderCallbackStruct input; OSStatus err; UInt32 size; err = AudioUnitUninitialize(self->audioUnit); if(err != noErr) ERR("-- AudioUnitUninitialize failed.\n"); /* retrieve default output unit's properties (output side) */ size = sizeof(AudioStreamBasicDescription); err = AudioUnitGetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size); if(err != noErr || size != sizeof(AudioStreamBasicDescription)) { ERR("AudioUnitGetProperty failed\n"); return ALC_FALSE; } #if 0 TRACE("Output streamFormat of default output unit -\n"); TRACE(" streamFormat.mFramesPerPacket = %d\n", streamFormat.mFramesPerPacket); TRACE(" streamFormat.mChannelsPerFrame = %d\n", streamFormat.mChannelsPerFrame); TRACE(" streamFormat.mBitsPerChannel = %d\n", streamFormat.mBitsPerChannel); TRACE(" streamFormat.mBytesPerPacket = %d\n", streamFormat.mBytesPerPacket); TRACE(" streamFormat.mBytesPerFrame = %d\n", streamFormat.mBytesPerFrame); TRACE(" streamFormat.mSampleRate = %5.0f\n", streamFormat.mSampleRate); #endif /* set default output unit's input side to match output side */ err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); return ALC_FALSE; } if(device->Frequency != streamFormat.mSampleRate) { device->NumUpdates = (ALuint)((ALuint64)device->NumUpdates * streamFormat.mSampleRate / device->Frequency); device->Frequency = streamFormat.mSampleRate; } /* FIXME: How to tell what channels are what in the output device, and how * to specify what we're giving? eg, 6.0 vs 5.1 */ switch(streamFormat.mChannelsPerFrame) { case 1: device->FmtChans = DevFmtMono; break; case 2: device->FmtChans = DevFmtStereo; break; case 4: device->FmtChans = DevFmtQuad; break; case 6: device->FmtChans = DevFmtX51; break; case 7: device->FmtChans = DevFmtX61; break; case 8: device->FmtChans = DevFmtX71; break; default: ERR("Unhandled channel count (%d), using Stereo\n", streamFormat.mChannelsPerFrame); device->FmtChans = DevFmtStereo; streamFormat.mChannelsPerFrame = 2; break; } SetDefaultWFXChannelOrder(device); /* use channel count and sample rate from the default output unit's current * parameters, but reset everything else */ streamFormat.mFramesPerPacket = 1; streamFormat.mFormatFlags = 0; switch(device->FmtType) { case DevFmtUByte: device->FmtType = DevFmtByte; /* fall-through */ case DevFmtByte: streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger; streamFormat.mBitsPerChannel = 8; break; case DevFmtUShort: device->FmtType = DevFmtShort; /* fall-through */ case DevFmtShort: streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger; streamFormat.mBitsPerChannel = 16; break; case DevFmtUInt: device->FmtType = DevFmtInt; /* fall-through */ case DevFmtInt: streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger; streamFormat.mBitsPerChannel = 32; break; case DevFmtFloat: streamFormat.mFormatFlags = kLinearPCMFormatFlagIsFloat; streamFormat.mBitsPerChannel = 32; break; } streamFormat.mBytesPerFrame = streamFormat.mChannelsPerFrame * streamFormat.mBitsPerChannel / 8; streamFormat.mBytesPerPacket = streamFormat.mBytesPerFrame; streamFormat.mFormatID = kAudioFormatLinearPCM; streamFormat.mFormatFlags |= kAudioFormatFlagsNativeEndian | kLinearPCMFormatFlagIsPacked; err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription)); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); return ALC_FALSE; } /* setup callback */ self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); input.inputProc = ALCcoreAudioPlayback_MixerProc; input.inputProcRefCon = self; err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct)); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); return ALC_FALSE; } /* init the default audio unit... */ err = AudioUnitInitialize(self->audioUnit); if(err != noErr) { ERR("AudioUnitInitialize failed\n"); return ALC_FALSE; } return ALC_TRUE; } static ALCboolean ALCcoreAudioPlayback_start(ALCcoreAudioPlayback *self) { OSStatus err = AudioOutputUnitStart(self->audioUnit); if(err != noErr) { ERR("AudioOutputUnitStart failed\n"); return ALC_FALSE; } return ALC_TRUE; } static void ALCcoreAudioPlayback_stop(ALCcoreAudioPlayback *self) { OSStatus err = AudioOutputUnitStop(self->audioUnit); if(err != noErr) ERR("AudioOutputUnitStop failed\n"); } typedef struct ALCcoreAudioCapture { DERIVE_FROM_TYPE(ALCbackend); AudioUnit audioUnit; ALuint frameSize; ALdouble sampleRateRatio; // Ratio of hardware sample rate / requested sample rate AudioStreamBasicDescription format; // This is the OpenAL format as a CoreAudio ASBD AudioConverterRef audioConverter; // Sample rate converter if needed AudioBufferList *bufferList; // Buffer for data coming from the input device ALCvoid *resampleBuffer; // Buffer for returned RingBuffer data when resampling ll_ringbuffer_t *ring; } ALCcoreAudioCapture; static void ALCcoreAudioCapture_Construct(ALCcoreAudioCapture *self, ALCdevice *device); static void ALCcoreAudioCapture_Destruct(ALCcoreAudioCapture *self); static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar *name); static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, ALCboolean, reset) static ALCboolean ALCcoreAudioCapture_start(ALCcoreAudioCapture *self); static void ALCcoreAudioCapture_stop(ALCcoreAudioCapture *self); static ALCenum ALCcoreAudioCapture_captureSamples(ALCcoreAudioCapture *self, ALCvoid *buffer, ALCuint samples); static ALCuint ALCcoreAudioCapture_availableSamples(ALCcoreAudioCapture *self); static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, void, lock) static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCcoreAudioCapture) DEFINE_ALCBACKEND_VTABLE(ALCcoreAudioCapture); static AudioBufferList *allocate_buffer_list(UInt32 channelCount, UInt32 byteSize) { AudioBufferList *list; list = calloc(1, FAM_SIZE(AudioBufferList, mBuffers, 1) + byteSize); if(list) { list->mNumberBuffers = 1; list->mBuffers[0].mNumberChannels = channelCount; list->mBuffers[0].mDataByteSize = byteSize; list->mBuffers[0].mData = &list->mBuffers[1]; } return list; } static void destroy_buffer_list(AudioBufferList *list) { free(list); } static void ALCcoreAudioCapture_Construct(ALCcoreAudioCapture *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCcoreAudioCapture, ALCbackend, self); self->audioUnit = 0; self->audioConverter = NULL; self->bufferList = NULL; self->resampleBuffer = NULL; self->ring = NULL; } static void ALCcoreAudioCapture_Destruct(ALCcoreAudioCapture *self) { ll_ringbuffer_free(self->ring); self->ring = NULL; free(self->resampleBuffer); self->resampleBuffer = NULL; destroy_buffer_list(self->bufferList); self->bufferList = NULL; if(self->audioConverter) AudioConverterDispose(self->audioConverter); self->audioConverter = NULL; if(self->audioUnit) AudioComponentInstanceDispose(self->audioUnit); self->audioUnit = 0; ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static OSStatus ALCcoreAudioCapture_RecordProc(void *inRefCon, AudioUnitRenderActionFlags* UNUSED(ioActionFlags), const AudioTimeStamp *inTimeStamp, UInt32 UNUSED(inBusNumber), UInt32 inNumberFrames, AudioBufferList* UNUSED(ioData)) { ALCcoreAudioCapture *self = inRefCon; AudioUnitRenderActionFlags flags = 0; OSStatus err; // fill the bufferList with data from the input device err = AudioUnitRender(self->audioUnit, &flags, inTimeStamp, 1, inNumberFrames, self->bufferList); if(err != noErr) { ERR("AudioUnitRender error: %d\n", err); return err; } ll_ringbuffer_write(self->ring, self->bufferList->mBuffers[0].mData, inNumberFrames); return noErr; } static OSStatus ALCcoreAudioCapture_ConvertCallback(AudioConverterRef UNUSED(inAudioConverter), UInt32 *ioNumberDataPackets, AudioBufferList *ioData, AudioStreamPacketDescription** UNUSED(outDataPacketDescription), void *inUserData) { ALCcoreAudioCapture *self = inUserData; // Read from the ring buffer and store temporarily in a large buffer ll_ringbuffer_read(self->ring, self->resampleBuffer, *ioNumberDataPackets); // Set the input data ioData->mNumberBuffers = 1; ioData->mBuffers[0].mNumberChannels = self->format.mChannelsPerFrame; ioData->mBuffers[0].mData = self->resampleBuffer; ioData->mBuffers[0].mDataByteSize = (*ioNumberDataPackets) * self->format.mBytesPerFrame; return noErr; } static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar *name) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; AudioStreamBasicDescription requestedFormat; // The application requested format AudioStreamBasicDescription hardwareFormat; // The hardware format AudioStreamBasicDescription outputFormat; // The AudioUnit output format AURenderCallbackStruct input; AudioComponentDescription desc; UInt32 outputFrameCount; UInt32 propertySize; AudioObjectPropertyAddress propertyAddress; UInt32 enableIO; AudioComponent comp; OSStatus err; if(!name) name = ca_device; else if(strcmp(name, ca_device) != 0) return ALC_INVALID_VALUE; desc.componentType = kAudioUnitType_Output; #if TARGET_OS_IOS desc.componentSubType = kAudioUnitSubType_RemoteIO; #else desc.componentSubType = kAudioUnitSubType_HALOutput; #endif desc.componentManufacturer = kAudioUnitManufacturer_Apple; desc.componentFlags = 0; desc.componentFlagsMask = 0; // Search for component with given description comp = AudioComponentFindNext(NULL, &desc); if(comp == NULL) { ERR("AudioComponentFindNext failed\n"); return ALC_INVALID_VALUE; } // Open the component err = AudioComponentInstanceNew(comp, &self->audioUnit); if(err != noErr) { ERR("AudioComponentInstanceNew failed\n"); goto error; } // Turn off AudioUnit output enableIO = 0; err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint)); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); goto error; } // Turn on AudioUnit input enableIO = 1; err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint)); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); goto error; } #if !TARGET_OS_IOS // Get the default input device AudioDeviceID inputDevice = kAudioDeviceUnknown; propertySize = sizeof(AudioDeviceID); propertyAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice; propertyAddress.mScope = kAudioObjectPropertyScopeGlobal; propertyAddress.mElement = kAudioObjectPropertyElementMaster; err = AudioObjectGetPropertyData(kAudioObjectSystemObject, &propertyAddress, 0, NULL, &propertySize, &inputDevice); if(err != noErr) { ERR("AudioObjectGetPropertyData failed\n"); goto error; } if(inputDevice == kAudioDeviceUnknown) { ERR("No input device found\n"); goto error; } // Track the input device err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID)); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); goto error; } #endif // set capture callback input.inputProc = ALCcoreAudioCapture_RecordProc; input.inputProcRefCon = self; err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct)); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); goto error; } // Initialize the device err = AudioUnitInitialize(self->audioUnit); if(err != noErr) { ERR("AudioUnitInitialize failed\n"); goto error; } // Get the hardware format propertySize = sizeof(AudioStreamBasicDescription); err = AudioUnitGetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize); if(err != noErr || propertySize != sizeof(AudioStreamBasicDescription)) { ERR("AudioUnitGetProperty failed\n"); goto error; } // Set up the requested format description switch(device->FmtType) { case DevFmtUByte: requestedFormat.mBitsPerChannel = 8; requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked; break; case DevFmtShort: requestedFormat.mBitsPerChannel = 16; requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked; break; case DevFmtInt: requestedFormat.mBitsPerChannel = 32; requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked; break; case DevFmtFloat: requestedFormat.mBitsPerChannel = 32; requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked; break; case DevFmtByte: case DevFmtUShort: case DevFmtUInt: ERR("%s samples not supported\n", DevFmtTypeString(device->FmtType)); goto error; } switch(device->FmtChans) { case DevFmtMono: requestedFormat.mChannelsPerFrame = 1; break; case DevFmtStereo: requestedFormat.mChannelsPerFrame = 2; break; case DevFmtQuad: case DevFmtX51: case DevFmtX51Rear: case DevFmtX61: case DevFmtX71: case DevFmtAmbi3D: ERR("%s not supported\n", DevFmtChannelsString(device->FmtChans)); goto error; } requestedFormat.mBytesPerFrame = requestedFormat.mChannelsPerFrame * requestedFormat.mBitsPerChannel / 8; requestedFormat.mBytesPerPacket = requestedFormat.mBytesPerFrame; requestedFormat.mSampleRate = device->Frequency; requestedFormat.mFormatID = kAudioFormatLinearPCM; requestedFormat.mReserved = 0; requestedFormat.mFramesPerPacket = 1; // save requested format description for later use self->format = requestedFormat; self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); // Use intermediate format for sample rate conversion (outputFormat) // Set sample rate to the same as hardware for resampling later outputFormat = requestedFormat; outputFormat.mSampleRate = hardwareFormat.mSampleRate; // Determine sample rate ratio for resampling self->sampleRateRatio = outputFormat.mSampleRate / device->Frequency; // The output format should be the requested format, but using the hardware sample rate // This is because the AudioUnit will automatically scale other properties, except for sample rate err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat)); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); goto error; } // Set the AudioUnit output format frame count outputFrameCount = device->UpdateSize * self->sampleRateRatio; err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Output, 0, &outputFrameCount, sizeof(outputFrameCount)); if(err != noErr) { ERR("AudioUnitSetProperty failed: %d\n", err); goto error; } // Set up sample converter err = AudioConverterNew(&outputFormat, &requestedFormat, &self->audioConverter); if(err != noErr) { ERR("AudioConverterNew failed: %d\n", err); goto error; } // Create a buffer for use in the resample callback self->resampleBuffer = malloc(device->UpdateSize * self->frameSize * self->sampleRateRatio); // Allocate buffer for the AudioUnit output self->bufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * self->frameSize * self->sampleRateRatio); if(self->bufferList == NULL) goto error; self->ring = ll_ringbuffer_create( (size_t)ceil(device->UpdateSize*self->sampleRateRatio*device->NumUpdates), self->frameSize, false ); if(!self->ring) goto error; alstr_copy_cstr(&device->DeviceName, name); return ALC_NO_ERROR; error: ll_ringbuffer_free(self->ring); self->ring = NULL; free(self->resampleBuffer); self->resampleBuffer = NULL; destroy_buffer_list(self->bufferList); self->bufferList = NULL; if(self->audioConverter) AudioConverterDispose(self->audioConverter); self->audioConverter = NULL; if(self->audioUnit) AudioComponentInstanceDispose(self->audioUnit); self->audioUnit = 0; return ALC_INVALID_VALUE; } static ALCboolean ALCcoreAudioCapture_start(ALCcoreAudioCapture *self) { OSStatus err = AudioOutputUnitStart(self->audioUnit); if(err != noErr) { ERR("AudioOutputUnitStart failed\n"); return ALC_FALSE; } return ALC_TRUE; } static void ALCcoreAudioCapture_stop(ALCcoreAudioCapture *self) { OSStatus err = AudioOutputUnitStop(self->audioUnit); if(err != noErr) ERR("AudioOutputUnitStop failed\n"); } static ALCenum ALCcoreAudioCapture_captureSamples(ALCcoreAudioCapture *self, ALCvoid *buffer, ALCuint samples) { union { ALbyte _[sizeof(AudioBufferList) + sizeof(AudioBuffer)]; AudioBufferList list; } audiobuf = { { 0 } }; UInt32 frameCount; OSStatus err; // If no samples are requested, just return if(samples == 0) return ALC_NO_ERROR; // Point the resampling buffer to the capture buffer audiobuf.list.mNumberBuffers = 1; audiobuf.list.mBuffers[0].mNumberChannels = self->format.mChannelsPerFrame; audiobuf.list.mBuffers[0].mDataByteSize = samples * self->frameSize; audiobuf.list.mBuffers[0].mData = buffer; // Resample into another AudioBufferList frameCount = samples; err = AudioConverterFillComplexBuffer(self->audioConverter, ALCcoreAudioCapture_ConvertCallback, self, &frameCount, &audiobuf.list, NULL ); if(err != noErr) { ERR("AudioConverterFillComplexBuffer error: %d\n", err); return ALC_INVALID_VALUE; } return ALC_NO_ERROR; } static ALCuint ALCcoreAudioCapture_availableSamples(ALCcoreAudioCapture *self) { return ll_ringbuffer_read_space(self->ring) / self->sampleRateRatio; } typedef struct ALCcoreAudioBackendFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } ALCcoreAudioBackendFactory; #define ALCCOREAUDIOBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCcoreAudioBackendFactory, ALCbackendFactory) } } ALCbackendFactory *ALCcoreAudioBackendFactory_getFactory(void); static ALCboolean ALCcoreAudioBackendFactory_init(ALCcoreAudioBackendFactory *self); static DECLARE_FORWARD(ALCcoreAudioBackendFactory, ALCbackendFactory, void, deinit) static ALCboolean ALCcoreAudioBackendFactory_querySupport(ALCcoreAudioBackendFactory *self, ALCbackend_Type type); static void ALCcoreAudioBackendFactory_probe(ALCcoreAudioBackendFactory *self, enum DevProbe type, al_string *outnames); static ALCbackend* ALCcoreAudioBackendFactory_createBackend(ALCcoreAudioBackendFactory *self, ALCdevice *device, ALCbackend_Type type); DEFINE_ALCBACKENDFACTORY_VTABLE(ALCcoreAudioBackendFactory); ALCbackendFactory *ALCcoreAudioBackendFactory_getFactory(void) { static ALCcoreAudioBackendFactory factory = ALCCOREAUDIOBACKENDFACTORY_INITIALIZER; return STATIC_CAST(ALCbackendFactory, &factory); } static ALCboolean ALCcoreAudioBackendFactory_init(ALCcoreAudioBackendFactory* UNUSED(self)) { return ALC_TRUE; } static ALCboolean ALCcoreAudioBackendFactory_querySupport(ALCcoreAudioBackendFactory* UNUSED(self), ALCbackend_Type type) { if(type == ALCbackend_Playback || ALCbackend_Capture) return ALC_TRUE; return ALC_FALSE; } static void ALCcoreAudioBackendFactory_probe(ALCcoreAudioBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames) { switch(type) { case ALL_DEVICE_PROBE: case CAPTURE_DEVICE_PROBE: alstr_append_range(outnames, ca_device, ca_device+sizeof(ca_device)); break; } } static ALCbackend* ALCcoreAudioBackendFactory_createBackend(ALCcoreAudioBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) { if(type == ALCbackend_Playback) { ALCcoreAudioPlayback *backend; NEW_OBJ(backend, ALCcoreAudioPlayback)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } if(type == ALCbackend_Capture) { ALCcoreAudioCapture *backend; NEW_OBJ(backend, ALCcoreAudioCapture)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } return NULL; } openal-soft-openal-soft-1.19.1/Alc/backends/dsound.c000066400000000000000000001115531335774445300222150ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include #include #include #ifndef _WAVEFORMATEXTENSIBLE_ #include #include #endif #include "alMain.h" #include "alu.h" #include "ringbuffer.h" #include "threads.h" #include "compat.h" #include "alstring.h" #include "backends/base.h" #ifndef DSSPEAKER_5POINT1 # define DSSPEAKER_5POINT1 0x00000006 #endif #ifndef DSSPEAKER_5POINT1_BACK # define DSSPEAKER_5POINT1_BACK 0x00000006 #endif #ifndef DSSPEAKER_7POINT1 # define DSSPEAKER_7POINT1 0x00000007 #endif #ifndef DSSPEAKER_7POINT1_SURROUND # define DSSPEAKER_7POINT1_SURROUND 0x00000008 #endif #ifndef DSSPEAKER_5POINT1_SURROUND # define DSSPEAKER_5POINT1_SURROUND 0x00000009 #endif DEFINE_GUID(KSDATAFORMAT_SUBTYPE_PCM, 0x00000001, 0x0000, 0x0010, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71); DEFINE_GUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, 0x00000003, 0x0000, 0x0010, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71); #define DEVNAME_HEAD "OpenAL Soft on " #ifdef HAVE_DYNLOAD static void *ds_handle; static HRESULT (WINAPI *pDirectSoundCreate)(const GUID *pcGuidDevice, IDirectSound **ppDS, IUnknown *pUnkOuter); static HRESULT (WINAPI *pDirectSoundEnumerateW)(LPDSENUMCALLBACKW pDSEnumCallback, void *pContext); static HRESULT (WINAPI *pDirectSoundCaptureCreate)(const GUID *pcGuidDevice, IDirectSoundCapture **ppDSC, IUnknown *pUnkOuter); static HRESULT (WINAPI *pDirectSoundCaptureEnumerateW)(LPDSENUMCALLBACKW pDSEnumCallback, void *pContext); #define DirectSoundCreate pDirectSoundCreate #define DirectSoundEnumerateW pDirectSoundEnumerateW #define DirectSoundCaptureCreate pDirectSoundCaptureCreate #define DirectSoundCaptureEnumerateW pDirectSoundCaptureEnumerateW #endif static ALCboolean DSoundLoad(void) { #ifdef HAVE_DYNLOAD if(!ds_handle) { ds_handle = LoadLib("dsound.dll"); if(ds_handle == NULL) { ERR("Failed to load dsound.dll\n"); return ALC_FALSE; } #define LOAD_FUNC(f) do { \ p##f = GetSymbol(ds_handle, #f); \ if(p##f == NULL) { \ CloseLib(ds_handle); \ ds_handle = NULL; \ return ALC_FALSE; \ } \ } while(0) LOAD_FUNC(DirectSoundCreate); LOAD_FUNC(DirectSoundEnumerateW); LOAD_FUNC(DirectSoundCaptureCreate); LOAD_FUNC(DirectSoundCaptureEnumerateW); #undef LOAD_FUNC } #endif return ALC_TRUE; } #define MAX_UPDATES 128 typedef struct { al_string name; GUID guid; } DevMap; TYPEDEF_VECTOR(DevMap, vector_DevMap) static vector_DevMap PlaybackDevices; static vector_DevMap CaptureDevices; static void clear_devlist(vector_DevMap *list) { #define DEINIT_STR(i) AL_STRING_DEINIT((i)->name) VECTOR_FOR_EACH(DevMap, *list, DEINIT_STR); VECTOR_RESIZE(*list, 0, 0); #undef DEINIT_STR } static BOOL CALLBACK DSoundEnumDevices(GUID *guid, const WCHAR *desc, const WCHAR* UNUSED(drvname), void *data) { vector_DevMap *devices = data; OLECHAR *guidstr = NULL; DevMap entry; HRESULT hr; int count; if(!guid) return TRUE; AL_STRING_INIT(entry.name); count = 0; while(1) { const DevMap *iter; alstr_copy_cstr(&entry.name, DEVNAME_HEAD); alstr_append_wcstr(&entry.name, desc); if(count != 0) { char str[64]; snprintf(str, sizeof(str), " #%d", count+1); alstr_append_cstr(&entry.name, str); } #define MATCH_ENTRY(i) (alstr_cmp(entry.name, (i)->name) == 0) VECTOR_FIND_IF(iter, const DevMap, *devices, MATCH_ENTRY); if(iter == VECTOR_END(*devices)) break; #undef MATCH_ENTRY count++; } entry.guid = *guid; hr = StringFromCLSID(guid, &guidstr); if(SUCCEEDED(hr)) { TRACE("Got device \"%s\", GUID \"%ls\"\n", alstr_get_cstr(entry.name), guidstr); CoTaskMemFree(guidstr); } VECTOR_PUSH_BACK(*devices, entry); return TRUE; } typedef struct ALCdsoundPlayback { DERIVE_FROM_TYPE(ALCbackend); IDirectSound *DS; IDirectSoundBuffer *PrimaryBuffer; IDirectSoundBuffer *Buffer; IDirectSoundNotify *Notifies; HANDLE NotifyEvent; ATOMIC(ALenum) killNow; althrd_t thread; } ALCdsoundPlayback; static int ALCdsoundPlayback_mixerProc(void *ptr); static void ALCdsoundPlayback_Construct(ALCdsoundPlayback *self, ALCdevice *device); static void ALCdsoundPlayback_Destruct(ALCdsoundPlayback *self); static ALCenum ALCdsoundPlayback_open(ALCdsoundPlayback *self, const ALCchar *name); static ALCboolean ALCdsoundPlayback_reset(ALCdsoundPlayback *self); static ALCboolean ALCdsoundPlayback_start(ALCdsoundPlayback *self); static void ALCdsoundPlayback_stop(ALCdsoundPlayback *self); static DECLARE_FORWARD2(ALCdsoundPlayback, ALCbackend, ALCenum, captureSamples, void*, ALCuint) static DECLARE_FORWARD(ALCdsoundPlayback, ALCbackend, ALCuint, availableSamples) static DECLARE_FORWARD(ALCdsoundPlayback, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(ALCdsoundPlayback, ALCbackend, void, lock) static DECLARE_FORWARD(ALCdsoundPlayback, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCdsoundPlayback) DEFINE_ALCBACKEND_VTABLE(ALCdsoundPlayback); static void ALCdsoundPlayback_Construct(ALCdsoundPlayback *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCdsoundPlayback, ALCbackend, self); self->DS = NULL; self->PrimaryBuffer = NULL; self->Buffer = NULL; self->Notifies = NULL; self->NotifyEvent = NULL; ATOMIC_INIT(&self->killNow, AL_TRUE); } static void ALCdsoundPlayback_Destruct(ALCdsoundPlayback *self) { if(self->Notifies) IDirectSoundNotify_Release(self->Notifies); self->Notifies = NULL; if(self->Buffer) IDirectSoundBuffer_Release(self->Buffer); self->Buffer = NULL; if(self->PrimaryBuffer != NULL) IDirectSoundBuffer_Release(self->PrimaryBuffer); self->PrimaryBuffer = NULL; if(self->DS) IDirectSound_Release(self->DS); self->DS = NULL; if(self->NotifyEvent) CloseHandle(self->NotifyEvent); self->NotifyEvent = NULL; ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } FORCE_ALIGN static int ALCdsoundPlayback_mixerProc(void *ptr) { ALCdsoundPlayback *self = ptr; ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; DSBCAPS DSBCaps; DWORD LastCursor = 0; DWORD PlayCursor; void *WritePtr1, *WritePtr2; DWORD WriteCnt1, WriteCnt2; BOOL Playing = FALSE; DWORD FrameSize; DWORD FragSize; DWORD avail; HRESULT err; SetRTPriority(); althrd_setname(althrd_current(), MIXER_THREAD_NAME); memset(&DSBCaps, 0, sizeof(DSBCaps)); DSBCaps.dwSize = sizeof(DSBCaps); err = IDirectSoundBuffer_GetCaps(self->Buffer, &DSBCaps); if(FAILED(err)) { ERR("Failed to get buffer caps: 0x%lx\n", err); ALCdevice_Lock(device); aluHandleDisconnect(device, "Failure retrieving playback buffer info: 0x%lx", err); ALCdevice_Unlock(device); return 1; } FrameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); FragSize = device->UpdateSize * FrameSize; IDirectSoundBuffer_GetCurrentPosition(self->Buffer, &LastCursor, NULL); while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) && ATOMIC_LOAD(&device->Connected, almemory_order_acquire)) { // Get current play cursor IDirectSoundBuffer_GetCurrentPosition(self->Buffer, &PlayCursor, NULL); avail = (PlayCursor-LastCursor+DSBCaps.dwBufferBytes) % DSBCaps.dwBufferBytes; if(avail < FragSize) { if(!Playing) { err = IDirectSoundBuffer_Play(self->Buffer, 0, 0, DSBPLAY_LOOPING); if(FAILED(err)) { ERR("Failed to play buffer: 0x%lx\n", err); ALCdevice_Lock(device); aluHandleDisconnect(device, "Failure starting playback: 0x%lx", err); ALCdevice_Unlock(device); return 1; } Playing = TRUE; } avail = WaitForSingleObjectEx(self->NotifyEvent, 2000, FALSE); if(avail != WAIT_OBJECT_0) ERR("WaitForSingleObjectEx error: 0x%lx\n", avail); continue; } avail -= avail%FragSize; // Lock output buffer WriteCnt1 = 0; WriteCnt2 = 0; err = IDirectSoundBuffer_Lock(self->Buffer, LastCursor, avail, &WritePtr1, &WriteCnt1, &WritePtr2, &WriteCnt2, 0); // If the buffer is lost, restore it and lock if(err == DSERR_BUFFERLOST) { WARN("Buffer lost, restoring...\n"); err = IDirectSoundBuffer_Restore(self->Buffer); if(SUCCEEDED(err)) { Playing = FALSE; LastCursor = 0; err = IDirectSoundBuffer_Lock(self->Buffer, 0, DSBCaps.dwBufferBytes, &WritePtr1, &WriteCnt1, &WritePtr2, &WriteCnt2, 0); } } // Successfully locked the output buffer if(SUCCEEDED(err)) { // If we have an active context, mix data directly into output buffer otherwise fill with silence ALCdevice_Lock(device); aluMixData(device, WritePtr1, WriteCnt1/FrameSize); aluMixData(device, WritePtr2, WriteCnt2/FrameSize); ALCdevice_Unlock(device); // Unlock output buffer only when successfully locked IDirectSoundBuffer_Unlock(self->Buffer, WritePtr1, WriteCnt1, WritePtr2, WriteCnt2); } else { ERR("Buffer lock error: %#lx\n", err); ALCdevice_Lock(device); aluHandleDisconnect(device, "Failed to lock output buffer: 0x%lx", err); ALCdevice_Unlock(device); return 1; } // Update old write cursor location LastCursor += WriteCnt1+WriteCnt2; LastCursor %= DSBCaps.dwBufferBytes; } return 0; } static ALCenum ALCdsoundPlayback_open(ALCdsoundPlayback *self, const ALCchar *deviceName) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; const GUID *guid = NULL; HRESULT hr, hrcom; if(VECTOR_SIZE(PlaybackDevices) == 0) { /* Initialize COM to prevent name truncation */ hrcom = CoInitialize(NULL); hr = DirectSoundEnumerateW(DSoundEnumDevices, &PlaybackDevices); if(FAILED(hr)) ERR("Error enumerating DirectSound devices (0x%lx)!\n", hr); if(SUCCEEDED(hrcom)) CoUninitialize(); } if(!deviceName && VECTOR_SIZE(PlaybackDevices) > 0) { deviceName = alstr_get_cstr(VECTOR_FRONT(PlaybackDevices).name); guid = &VECTOR_FRONT(PlaybackDevices).guid; } else { const DevMap *iter; #define MATCH_NAME(i) (alstr_cmp_cstr((i)->name, deviceName) == 0) VECTOR_FIND_IF(iter, const DevMap, PlaybackDevices, MATCH_NAME); #undef MATCH_NAME if(iter == VECTOR_END(PlaybackDevices)) return ALC_INVALID_VALUE; guid = &iter->guid; } hr = DS_OK; self->NotifyEvent = CreateEventW(NULL, FALSE, FALSE, NULL); if(self->NotifyEvent == NULL) hr = E_FAIL; //DirectSound Init code if(SUCCEEDED(hr)) hr = DirectSoundCreate(guid, &self->DS, NULL); if(SUCCEEDED(hr)) hr = IDirectSound_SetCooperativeLevel(self->DS, GetForegroundWindow(), DSSCL_PRIORITY); if(FAILED(hr)) { if(self->DS) IDirectSound_Release(self->DS); self->DS = NULL; if(self->NotifyEvent) CloseHandle(self->NotifyEvent); self->NotifyEvent = NULL; ERR("Device init failed: 0x%08lx\n", hr); return ALC_INVALID_VALUE; } alstr_copy_cstr(&device->DeviceName, deviceName); return ALC_NO_ERROR; } static ALCboolean ALCdsoundPlayback_reset(ALCdsoundPlayback *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; DSBUFFERDESC DSBDescription; WAVEFORMATEXTENSIBLE OutputType; DWORD speakers; HRESULT hr; memset(&OutputType, 0, sizeof(OutputType)); if(self->Notifies) IDirectSoundNotify_Release(self->Notifies); self->Notifies = NULL; if(self->Buffer) IDirectSoundBuffer_Release(self->Buffer); self->Buffer = NULL; if(self->PrimaryBuffer != NULL) IDirectSoundBuffer_Release(self->PrimaryBuffer); self->PrimaryBuffer = NULL; switch(device->FmtType) { case DevFmtByte: device->FmtType = DevFmtUByte; break; case DevFmtFloat: if((device->Flags&DEVICE_SAMPLE_TYPE_REQUEST)) break; /* fall-through */ case DevFmtUShort: device->FmtType = DevFmtShort; break; case DevFmtUInt: device->FmtType = DevFmtInt; break; case DevFmtUByte: case DevFmtShort: case DevFmtInt: break; } hr = IDirectSound_GetSpeakerConfig(self->DS, &speakers); if(SUCCEEDED(hr)) { speakers = DSSPEAKER_CONFIG(speakers); if(!(device->Flags&DEVICE_CHANNELS_REQUEST)) { if(speakers == DSSPEAKER_MONO) device->FmtChans = DevFmtMono; else if(speakers == DSSPEAKER_STEREO || speakers == DSSPEAKER_HEADPHONE) device->FmtChans = DevFmtStereo; else if(speakers == DSSPEAKER_QUAD) device->FmtChans = DevFmtQuad; else if(speakers == DSSPEAKER_5POINT1_SURROUND) device->FmtChans = DevFmtX51; else if(speakers == DSSPEAKER_5POINT1_BACK) device->FmtChans = DevFmtX51Rear; else if(speakers == DSSPEAKER_7POINT1 || speakers == DSSPEAKER_7POINT1_SURROUND) device->FmtChans = DevFmtX71; else ERR("Unknown system speaker config: 0x%lx\n", speakers); } device->IsHeadphones = (device->FmtChans == DevFmtStereo && speakers == DSSPEAKER_HEADPHONE); switch(device->FmtChans) { case DevFmtMono: OutputType.dwChannelMask = SPEAKER_FRONT_CENTER; break; case DevFmtAmbi3D: device->FmtChans = DevFmtStereo; /*fall-through*/ case DevFmtStereo: OutputType.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT; break; case DevFmtQuad: OutputType.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT; break; case DevFmtX51: OutputType.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT; break; case DevFmtX51Rear: OutputType.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT; break; case DevFmtX61: OutputType.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_CENTER | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT; break; case DevFmtX71: OutputType.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT; break; } retry_open: hr = S_OK; OutputType.Format.wFormatTag = WAVE_FORMAT_PCM; OutputType.Format.nChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); OutputType.Format.wBitsPerSample = BytesFromDevFmt(device->FmtType) * 8; OutputType.Format.nBlockAlign = OutputType.Format.nChannels*OutputType.Format.wBitsPerSample/8; OutputType.Format.nSamplesPerSec = device->Frequency; OutputType.Format.nAvgBytesPerSec = OutputType.Format.nSamplesPerSec*OutputType.Format.nBlockAlign; OutputType.Format.cbSize = 0; } if(OutputType.Format.nChannels > 2 || device->FmtType == DevFmtFloat) { OutputType.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE; OutputType.Samples.wValidBitsPerSample = OutputType.Format.wBitsPerSample; OutputType.Format.cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX); if(device->FmtType == DevFmtFloat) OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT; else OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; if(self->PrimaryBuffer) IDirectSoundBuffer_Release(self->PrimaryBuffer); self->PrimaryBuffer = NULL; } else { if(SUCCEEDED(hr) && !self->PrimaryBuffer) { memset(&DSBDescription,0,sizeof(DSBUFFERDESC)); DSBDescription.dwSize=sizeof(DSBUFFERDESC); DSBDescription.dwFlags=DSBCAPS_PRIMARYBUFFER; hr = IDirectSound_CreateSoundBuffer(self->DS, &DSBDescription, &self->PrimaryBuffer, NULL); } if(SUCCEEDED(hr)) hr = IDirectSoundBuffer_SetFormat(self->PrimaryBuffer,&OutputType.Format); } if(SUCCEEDED(hr)) { if(device->NumUpdates > MAX_UPDATES) { device->UpdateSize = (device->UpdateSize*device->NumUpdates + MAX_UPDATES-1) / MAX_UPDATES; device->NumUpdates = MAX_UPDATES; } memset(&DSBDescription,0,sizeof(DSBUFFERDESC)); DSBDescription.dwSize=sizeof(DSBUFFERDESC); DSBDescription.dwFlags=DSBCAPS_CTRLPOSITIONNOTIFY|DSBCAPS_GETCURRENTPOSITION2|DSBCAPS_GLOBALFOCUS; DSBDescription.dwBufferBytes=device->UpdateSize * device->NumUpdates * OutputType.Format.nBlockAlign; DSBDescription.lpwfxFormat=&OutputType.Format; hr = IDirectSound_CreateSoundBuffer(self->DS, &DSBDescription, &self->Buffer, NULL); if(FAILED(hr) && device->FmtType == DevFmtFloat) { device->FmtType = DevFmtShort; goto retry_open; } } if(SUCCEEDED(hr)) { hr = IDirectSoundBuffer_QueryInterface(self->Buffer, &IID_IDirectSoundNotify, (void**)&self->Notifies); if(SUCCEEDED(hr)) { DSBPOSITIONNOTIFY notifies[MAX_UPDATES]; ALuint i; for(i = 0;i < device->NumUpdates;++i) { notifies[i].dwOffset = i * device->UpdateSize * OutputType.Format.nBlockAlign; notifies[i].hEventNotify = self->NotifyEvent; } if(IDirectSoundNotify_SetNotificationPositions(self->Notifies, device->NumUpdates, notifies) != DS_OK) hr = E_FAIL; } } if(FAILED(hr)) { if(self->Notifies != NULL) IDirectSoundNotify_Release(self->Notifies); self->Notifies = NULL; if(self->Buffer != NULL) IDirectSoundBuffer_Release(self->Buffer); self->Buffer = NULL; if(self->PrimaryBuffer != NULL) IDirectSoundBuffer_Release(self->PrimaryBuffer); self->PrimaryBuffer = NULL; return ALC_FALSE; } ResetEvent(self->NotifyEvent); SetDefaultWFXChannelOrder(device); return ALC_TRUE; } static ALCboolean ALCdsoundPlayback_start(ALCdsoundPlayback *self) { ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release); if(althrd_create(&self->thread, ALCdsoundPlayback_mixerProc, self) != althrd_success) return ALC_FALSE; return ALC_TRUE; } static void ALCdsoundPlayback_stop(ALCdsoundPlayback *self) { int res; if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel)) return; althrd_join(self->thread, &res); IDirectSoundBuffer_Stop(self->Buffer); } typedef struct ALCdsoundCapture { DERIVE_FROM_TYPE(ALCbackend); IDirectSoundCapture *DSC; IDirectSoundCaptureBuffer *DSCbuffer; DWORD BufferBytes; DWORD Cursor; ll_ringbuffer_t *Ring; } ALCdsoundCapture; static void ALCdsoundCapture_Construct(ALCdsoundCapture *self, ALCdevice *device); static void ALCdsoundCapture_Destruct(ALCdsoundCapture *self); static ALCenum ALCdsoundCapture_open(ALCdsoundCapture *self, const ALCchar *name); static DECLARE_FORWARD(ALCdsoundCapture, ALCbackend, ALCboolean, reset) static ALCboolean ALCdsoundCapture_start(ALCdsoundCapture *self); static void ALCdsoundCapture_stop(ALCdsoundCapture *self); static ALCenum ALCdsoundCapture_captureSamples(ALCdsoundCapture *self, ALCvoid *buffer, ALCuint samples); static ALCuint ALCdsoundCapture_availableSamples(ALCdsoundCapture *self); static DECLARE_FORWARD(ALCdsoundCapture, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(ALCdsoundCapture, ALCbackend, void, lock) static DECLARE_FORWARD(ALCdsoundCapture, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCdsoundCapture) DEFINE_ALCBACKEND_VTABLE(ALCdsoundCapture); static void ALCdsoundCapture_Construct(ALCdsoundCapture *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCdsoundCapture, ALCbackend, self); self->DSC = NULL; self->DSCbuffer = NULL; self->Ring = NULL; } static void ALCdsoundCapture_Destruct(ALCdsoundCapture *self) { ll_ringbuffer_free(self->Ring); self->Ring = NULL; if(self->DSCbuffer != NULL) { IDirectSoundCaptureBuffer_Stop(self->DSCbuffer); IDirectSoundCaptureBuffer_Release(self->DSCbuffer); self->DSCbuffer = NULL; } if(self->DSC) IDirectSoundCapture_Release(self->DSC); self->DSC = NULL; ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static ALCenum ALCdsoundCapture_open(ALCdsoundCapture *self, const ALCchar *deviceName) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; WAVEFORMATEXTENSIBLE InputType; DSCBUFFERDESC DSCBDescription; const GUID *guid = NULL; HRESULT hr, hrcom; ALuint samples; if(VECTOR_SIZE(CaptureDevices) == 0) { /* Initialize COM to prevent name truncation */ hrcom = CoInitialize(NULL); hr = DirectSoundCaptureEnumerateW(DSoundEnumDevices, &CaptureDevices); if(FAILED(hr)) ERR("Error enumerating DirectSound devices (0x%lx)!\n", hr); if(SUCCEEDED(hrcom)) CoUninitialize(); } if(!deviceName && VECTOR_SIZE(CaptureDevices) > 0) { deviceName = alstr_get_cstr(VECTOR_FRONT(CaptureDevices).name); guid = &VECTOR_FRONT(CaptureDevices).guid; } else { const DevMap *iter; #define MATCH_NAME(i) (alstr_cmp_cstr((i)->name, deviceName) == 0) VECTOR_FIND_IF(iter, const DevMap, CaptureDevices, MATCH_NAME); #undef MATCH_NAME if(iter == VECTOR_END(CaptureDevices)) return ALC_INVALID_VALUE; guid = &iter->guid; } switch(device->FmtType) { case DevFmtByte: case DevFmtUShort: case DevFmtUInt: WARN("%s capture samples not supported\n", DevFmtTypeString(device->FmtType)); return ALC_INVALID_ENUM; case DevFmtUByte: case DevFmtShort: case DevFmtInt: case DevFmtFloat: break; } memset(&InputType, 0, sizeof(InputType)); switch(device->FmtChans) { case DevFmtMono: InputType.dwChannelMask = SPEAKER_FRONT_CENTER; break; case DevFmtStereo: InputType.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT; break; case DevFmtQuad: InputType.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT; break; case DevFmtX51: InputType.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT; break; case DevFmtX51Rear: InputType.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT; break; case DevFmtX61: InputType.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_CENTER | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT; break; case DevFmtX71: InputType.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT; break; case DevFmtAmbi3D: WARN("%s capture not supported\n", DevFmtChannelsString(device->FmtChans)); return ALC_INVALID_ENUM; } InputType.Format.wFormatTag = WAVE_FORMAT_PCM; InputType.Format.nChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); InputType.Format.wBitsPerSample = BytesFromDevFmt(device->FmtType) * 8; InputType.Format.nBlockAlign = InputType.Format.nChannels*InputType.Format.wBitsPerSample/8; InputType.Format.nSamplesPerSec = device->Frequency; InputType.Format.nAvgBytesPerSec = InputType.Format.nSamplesPerSec*InputType.Format.nBlockAlign; InputType.Format.cbSize = 0; InputType.Samples.wValidBitsPerSample = InputType.Format.wBitsPerSample; if(device->FmtType == DevFmtFloat) InputType.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT; else InputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; if(InputType.Format.nChannels > 2 || device->FmtType == DevFmtFloat) { InputType.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE; InputType.Format.cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX); } samples = device->UpdateSize * device->NumUpdates; samples = maxu(samples, 100 * device->Frequency / 1000); memset(&DSCBDescription, 0, sizeof(DSCBUFFERDESC)); DSCBDescription.dwSize = sizeof(DSCBUFFERDESC); DSCBDescription.dwFlags = 0; DSCBDescription.dwBufferBytes = samples * InputType.Format.nBlockAlign; DSCBDescription.lpwfxFormat = &InputType.Format; //DirectSoundCapture Init code hr = DirectSoundCaptureCreate(guid, &self->DSC, NULL); if(SUCCEEDED(hr)) hr = IDirectSoundCapture_CreateCaptureBuffer(self->DSC, &DSCBDescription, &self->DSCbuffer, NULL); if(SUCCEEDED(hr)) { self->Ring = ll_ringbuffer_create(device->UpdateSize*device->NumUpdates, InputType.Format.nBlockAlign, false); if(self->Ring == NULL) hr = DSERR_OUTOFMEMORY; } if(FAILED(hr)) { ERR("Device init failed: 0x%08lx\n", hr); ll_ringbuffer_free(self->Ring); self->Ring = NULL; if(self->DSCbuffer != NULL) IDirectSoundCaptureBuffer_Release(self->DSCbuffer); self->DSCbuffer = NULL; if(self->DSC) IDirectSoundCapture_Release(self->DSC); self->DSC = NULL; return ALC_INVALID_VALUE; } self->BufferBytes = DSCBDescription.dwBufferBytes; SetDefaultWFXChannelOrder(device); alstr_copy_cstr(&device->DeviceName, deviceName); return ALC_NO_ERROR; } static ALCboolean ALCdsoundCapture_start(ALCdsoundCapture *self) { HRESULT hr; hr = IDirectSoundCaptureBuffer_Start(self->DSCbuffer, DSCBSTART_LOOPING); if(FAILED(hr)) { ERR("start failed: 0x%08lx\n", hr); aluHandleDisconnect(STATIC_CAST(ALCbackend, self)->mDevice, "Failure starting capture: 0x%lx", hr); return ALC_FALSE; } return ALC_TRUE; } static void ALCdsoundCapture_stop(ALCdsoundCapture *self) { HRESULT hr; hr = IDirectSoundCaptureBuffer_Stop(self->DSCbuffer); if(FAILED(hr)) { ERR("stop failed: 0x%08lx\n", hr); aluHandleDisconnect(STATIC_CAST(ALCbackend, self)->mDevice, "Failure stopping capture: 0x%lx", hr); } } static ALCenum ALCdsoundCapture_captureSamples(ALCdsoundCapture *self, ALCvoid *buffer, ALCuint samples) { ll_ringbuffer_read(self->Ring, buffer, samples); return ALC_NO_ERROR; } static ALCuint ALCdsoundCapture_availableSamples(ALCdsoundCapture *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; DWORD ReadCursor, LastCursor, BufferBytes, NumBytes; void *ReadPtr1, *ReadPtr2; DWORD ReadCnt1, ReadCnt2; DWORD FrameSize; HRESULT hr; if(!ATOMIC_LOAD(&device->Connected, almemory_order_acquire)) goto done; FrameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); BufferBytes = self->BufferBytes; LastCursor = self->Cursor; hr = IDirectSoundCaptureBuffer_GetCurrentPosition(self->DSCbuffer, NULL, &ReadCursor); if(SUCCEEDED(hr)) { NumBytes = (ReadCursor-LastCursor + BufferBytes) % BufferBytes; if(NumBytes == 0) goto done; hr = IDirectSoundCaptureBuffer_Lock(self->DSCbuffer, LastCursor, NumBytes, &ReadPtr1, &ReadCnt1, &ReadPtr2, &ReadCnt2, 0); } if(SUCCEEDED(hr)) { ll_ringbuffer_write(self->Ring, ReadPtr1, ReadCnt1/FrameSize); if(ReadPtr2 != NULL) ll_ringbuffer_write(self->Ring, ReadPtr2, ReadCnt2/FrameSize); hr = IDirectSoundCaptureBuffer_Unlock(self->DSCbuffer, ReadPtr1, ReadCnt1, ReadPtr2, ReadCnt2); self->Cursor = (LastCursor+ReadCnt1+ReadCnt2) % BufferBytes; } if(FAILED(hr)) { ERR("update failed: 0x%08lx\n", hr); aluHandleDisconnect(device, "Failure retrieving capture data: 0x%lx", hr); } done: return (ALCuint)ll_ringbuffer_read_space(self->Ring); } typedef struct ALCdsoundBackendFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } ALCdsoundBackendFactory; #define ALCDSOUNDBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCdsoundBackendFactory, ALCbackendFactory) } } ALCbackendFactory *ALCdsoundBackendFactory_getFactory(void); static ALCboolean ALCdsoundBackendFactory_init(ALCdsoundBackendFactory *self); static void ALCdsoundBackendFactory_deinit(ALCdsoundBackendFactory *self); static ALCboolean ALCdsoundBackendFactory_querySupport(ALCdsoundBackendFactory *self, ALCbackend_Type type); static void ALCdsoundBackendFactory_probe(ALCdsoundBackendFactory *self, enum DevProbe type, al_string *outnames); static ALCbackend* ALCdsoundBackendFactory_createBackend(ALCdsoundBackendFactory *self, ALCdevice *device, ALCbackend_Type type); DEFINE_ALCBACKENDFACTORY_VTABLE(ALCdsoundBackendFactory); ALCbackendFactory *ALCdsoundBackendFactory_getFactory(void) { static ALCdsoundBackendFactory factory = ALCDSOUNDBACKENDFACTORY_INITIALIZER; return STATIC_CAST(ALCbackendFactory, &factory); } static ALCboolean ALCdsoundBackendFactory_init(ALCdsoundBackendFactory* UNUSED(self)) { VECTOR_INIT(PlaybackDevices); VECTOR_INIT(CaptureDevices); if(!DSoundLoad()) return ALC_FALSE; return ALC_TRUE; } static void ALCdsoundBackendFactory_deinit(ALCdsoundBackendFactory* UNUSED(self)) { clear_devlist(&PlaybackDevices); VECTOR_DEINIT(PlaybackDevices); clear_devlist(&CaptureDevices); VECTOR_DEINIT(CaptureDevices); #ifdef HAVE_DYNLOAD if(ds_handle) CloseLib(ds_handle); ds_handle = NULL; #endif } static ALCboolean ALCdsoundBackendFactory_querySupport(ALCdsoundBackendFactory* UNUSED(self), ALCbackend_Type type) { if(type == ALCbackend_Playback || type == ALCbackend_Capture) return ALC_TRUE; return ALC_FALSE; } static void ALCdsoundBackendFactory_probe(ALCdsoundBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames) { HRESULT hr, hrcom; /* Initialize COM to prevent name truncation */ hrcom = CoInitialize(NULL); switch(type) { #define APPEND_OUTNAME(e) do { \ if(!alstr_empty((e)->name)) \ alstr_append_range(outnames, VECTOR_BEGIN((e)->name), \ VECTOR_END((e)->name)+1); \ } while(0) case ALL_DEVICE_PROBE: clear_devlist(&PlaybackDevices); hr = DirectSoundEnumerateW(DSoundEnumDevices, &PlaybackDevices); if(FAILED(hr)) ERR("Error enumerating DirectSound playback devices (0x%lx)!\n", hr); VECTOR_FOR_EACH(const DevMap, PlaybackDevices, APPEND_OUTNAME); break; case CAPTURE_DEVICE_PROBE: clear_devlist(&CaptureDevices); hr = DirectSoundCaptureEnumerateW(DSoundEnumDevices, &CaptureDevices); if(FAILED(hr)) ERR("Error enumerating DirectSound capture devices (0x%lx)!\n", hr); VECTOR_FOR_EACH(const DevMap, CaptureDevices, APPEND_OUTNAME); break; #undef APPEND_OUTNAME } if(SUCCEEDED(hrcom)) CoUninitialize(); } static ALCbackend* ALCdsoundBackendFactory_createBackend(ALCdsoundBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) { if(type == ALCbackend_Playback) { ALCdsoundPlayback *backend; NEW_OBJ(backend, ALCdsoundPlayback)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } if(type == ALCbackend_Capture) { ALCdsoundCapture *backend; NEW_OBJ(backend, ALCdsoundCapture)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } return NULL; } openal-soft-openal-soft-1.19.1/Alc/backends/jack.c000066400000000000000000000432461335774445300216340ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include "alMain.h" #include "alu.h" #include "alconfig.h" #include "ringbuffer.h" #include "threads.h" #include "compat.h" #include "backends/base.h" #include #include static const ALCchar jackDevice[] = "JACK Default"; #ifdef HAVE_DYNLOAD #define JACK_FUNCS(MAGIC) \ MAGIC(jack_client_open); \ MAGIC(jack_client_close); \ MAGIC(jack_client_name_size); \ MAGIC(jack_get_client_name); \ MAGIC(jack_connect); \ MAGIC(jack_activate); \ MAGIC(jack_deactivate); \ MAGIC(jack_port_register); \ MAGIC(jack_port_unregister); \ MAGIC(jack_port_get_buffer); \ MAGIC(jack_port_name); \ MAGIC(jack_get_ports); \ MAGIC(jack_free); \ MAGIC(jack_get_sample_rate); \ MAGIC(jack_set_error_function); \ MAGIC(jack_set_process_callback); \ MAGIC(jack_set_buffer_size_callback); \ MAGIC(jack_set_buffer_size); \ MAGIC(jack_get_buffer_size); static void *jack_handle; #define MAKE_FUNC(f) static __typeof(f) * p##f JACK_FUNCS(MAKE_FUNC); static __typeof(jack_error_callback) * pjack_error_callback; #undef MAKE_FUNC #define jack_client_open pjack_client_open #define jack_client_close pjack_client_close #define jack_client_name_size pjack_client_name_size #define jack_get_client_name pjack_get_client_name #define jack_connect pjack_connect #define jack_activate pjack_activate #define jack_deactivate pjack_deactivate #define jack_port_register pjack_port_register #define jack_port_unregister pjack_port_unregister #define jack_port_get_buffer pjack_port_get_buffer #define jack_port_name pjack_port_name #define jack_get_ports pjack_get_ports #define jack_free pjack_free #define jack_get_sample_rate pjack_get_sample_rate #define jack_set_error_function pjack_set_error_function #define jack_set_process_callback pjack_set_process_callback #define jack_set_buffer_size_callback pjack_set_buffer_size_callback #define jack_set_buffer_size pjack_set_buffer_size #define jack_get_buffer_size pjack_get_buffer_size #define jack_error_callback (*pjack_error_callback) #endif static jack_options_t ClientOptions = JackNullOption; static ALCboolean jack_load(void) { ALCboolean error = ALC_FALSE; #ifdef HAVE_DYNLOAD if(!jack_handle) { al_string missing_funcs = AL_STRING_INIT_STATIC(); #ifdef _WIN32 #define JACKLIB "libjack.dll" #else #define JACKLIB "libjack.so.0" #endif jack_handle = LoadLib(JACKLIB); if(!jack_handle) { WARN("Failed to load %s\n", JACKLIB); return ALC_FALSE; } error = ALC_FALSE; #define LOAD_FUNC(f) do { \ p##f = GetSymbol(jack_handle, #f); \ if(p##f == NULL) { \ error = ALC_TRUE; \ alstr_append_cstr(&missing_funcs, "\n" #f); \ } \ } while(0) JACK_FUNCS(LOAD_FUNC); #undef LOAD_FUNC /* Optional symbols. These don't exist in all versions of JACK. */ #define LOAD_SYM(f) p##f = GetSymbol(jack_handle, #f) LOAD_SYM(jack_error_callback); #undef LOAD_SYM if(error) { WARN("Missing expected functions:%s\n", alstr_get_cstr(missing_funcs)); CloseLib(jack_handle); jack_handle = NULL; } alstr_reset(&missing_funcs); } #endif return !error; } typedef struct ALCjackPlayback { DERIVE_FROM_TYPE(ALCbackend); jack_client_t *Client; jack_port_t *Port[MAX_OUTPUT_CHANNELS]; ll_ringbuffer_t *Ring; alsem_t Sem; ATOMIC(ALenum) killNow; althrd_t thread; } ALCjackPlayback; static int ALCjackPlayback_bufferSizeNotify(jack_nframes_t numframes, void *arg); static int ALCjackPlayback_process(jack_nframes_t numframes, void *arg); static int ALCjackPlayback_mixerProc(void *arg); static void ALCjackPlayback_Construct(ALCjackPlayback *self, ALCdevice *device); static void ALCjackPlayback_Destruct(ALCjackPlayback *self); static ALCenum ALCjackPlayback_open(ALCjackPlayback *self, const ALCchar *name); static ALCboolean ALCjackPlayback_reset(ALCjackPlayback *self); static ALCboolean ALCjackPlayback_start(ALCjackPlayback *self); static void ALCjackPlayback_stop(ALCjackPlayback *self); static DECLARE_FORWARD2(ALCjackPlayback, ALCbackend, ALCenum, captureSamples, void*, ALCuint) static DECLARE_FORWARD(ALCjackPlayback, ALCbackend, ALCuint, availableSamples) static ClockLatency ALCjackPlayback_getClockLatency(ALCjackPlayback *self); static DECLARE_FORWARD(ALCjackPlayback, ALCbackend, void, lock) static DECLARE_FORWARD(ALCjackPlayback, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCjackPlayback) DEFINE_ALCBACKEND_VTABLE(ALCjackPlayback); static void ALCjackPlayback_Construct(ALCjackPlayback *self, ALCdevice *device) { ALuint i; ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCjackPlayback, ALCbackend, self); alsem_init(&self->Sem, 0); self->Client = NULL; for(i = 0;i < MAX_OUTPUT_CHANNELS;i++) self->Port[i] = NULL; self->Ring = NULL; ATOMIC_INIT(&self->killNow, AL_TRUE); } static void ALCjackPlayback_Destruct(ALCjackPlayback *self) { ALuint i; if(self->Client) { for(i = 0;i < MAX_OUTPUT_CHANNELS;i++) { if(self->Port[i]) jack_port_unregister(self->Client, self->Port[i]); self->Port[i] = NULL; } jack_client_close(self->Client); self->Client = NULL; } alsem_destroy(&self->Sem); ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static int ALCjackPlayback_bufferSizeNotify(jack_nframes_t numframes, void *arg) { ALCjackPlayback *self = arg; ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; ALuint bufsize; ALCjackPlayback_lock(self); device->UpdateSize = numframes; device->NumUpdates = 2; bufsize = device->UpdateSize; if(ConfigValueUInt(alstr_get_cstr(device->DeviceName), "jack", "buffer-size", &bufsize)) bufsize = maxu(NextPowerOf2(bufsize), device->UpdateSize); device->NumUpdates = (bufsize+device->UpdateSize) / device->UpdateSize; TRACE("%u update size x%u\n", device->UpdateSize, device->NumUpdates); ll_ringbuffer_free(self->Ring); self->Ring = ll_ringbuffer_create(bufsize, FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder), true ); if(!self->Ring) { ERR("Failed to reallocate ringbuffer\n"); aluHandleDisconnect(device, "Failed to reallocate %u-sample buffer", bufsize); } ALCjackPlayback_unlock(self); return 0; } static int ALCjackPlayback_process(jack_nframes_t numframes, void *arg) { ALCjackPlayback *self = arg; jack_default_audio_sample_t *out[MAX_OUTPUT_CHANNELS]; ll_ringbuffer_data_t data[2]; jack_nframes_t total = 0; jack_nframes_t todo; ALsizei i, c, numchans; ll_ringbuffer_get_read_vector(self->Ring, data); for(c = 0;c < MAX_OUTPUT_CHANNELS && self->Port[c];c++) out[c] = jack_port_get_buffer(self->Port[c], numframes); numchans = c; todo = minu(numframes, data[0].len); for(c = 0;c < numchans;c++) { const ALfloat *restrict in = ((ALfloat*)data[0].buf) + c; for(i = 0;(jack_nframes_t)i < todo;i++) out[c][i] = in[i*numchans]; out[c] += todo; } total += todo; todo = minu(numframes-total, data[1].len); if(todo > 0) { for(c = 0;c < numchans;c++) { const ALfloat *restrict in = ((ALfloat*)data[1].buf) + c; for(i = 0;(jack_nframes_t)i < todo;i++) out[c][i] = in[i*numchans]; out[c] += todo; } total += todo; } ll_ringbuffer_read_advance(self->Ring, total); alsem_post(&self->Sem); if(numframes > total) { todo = numframes-total; for(c = 0;c < numchans;c++) { for(i = 0;(jack_nframes_t)i < todo;i++) out[c][i] = 0.0f; } } return 0; } static int ALCjackPlayback_mixerProc(void *arg) { ALCjackPlayback *self = arg; ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; ll_ringbuffer_data_t data[2]; SetRTPriority(); althrd_setname(althrd_current(), MIXER_THREAD_NAME); ALCjackPlayback_lock(self); while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) && ATOMIC_LOAD(&device->Connected, almemory_order_acquire)) { ALuint todo, len1, len2; if(ll_ringbuffer_write_space(self->Ring) < device->UpdateSize) { ALCjackPlayback_unlock(self); alsem_wait(&self->Sem); ALCjackPlayback_lock(self); continue; } ll_ringbuffer_get_write_vector(self->Ring, data); todo = data[0].len + data[1].len; todo -= todo%device->UpdateSize; len1 = minu(data[0].len, todo); len2 = minu(data[1].len, todo-len1); aluMixData(device, data[0].buf, len1); if(len2 > 0) aluMixData(device, data[1].buf, len2); ll_ringbuffer_write_advance(self->Ring, todo); } ALCjackPlayback_unlock(self); return 0; } static ALCenum ALCjackPlayback_open(ALCjackPlayback *self, const ALCchar *name) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; const char *client_name = "alsoft"; jack_status_t status; if(!name) name = jackDevice; else if(strcmp(name, jackDevice) != 0) return ALC_INVALID_VALUE; self->Client = jack_client_open(client_name, ClientOptions, &status, NULL); if(self->Client == NULL) { ERR("jack_client_open() failed, status = 0x%02x\n", status); return ALC_INVALID_VALUE; } if((status&JackServerStarted)) TRACE("JACK server started\n"); if((status&JackNameNotUnique)) { client_name = jack_get_client_name(self->Client); TRACE("Client name not unique, got `%s' instead\n", client_name); } jack_set_process_callback(self->Client, ALCjackPlayback_process, self); jack_set_buffer_size_callback(self->Client, ALCjackPlayback_bufferSizeNotify, self); alstr_copy_cstr(&device->DeviceName, name); return ALC_NO_ERROR; } static ALCboolean ALCjackPlayback_reset(ALCjackPlayback *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; ALsizei numchans, i; ALuint bufsize; for(i = 0;i < MAX_OUTPUT_CHANNELS;i++) { if(self->Port[i]) jack_port_unregister(self->Client, self->Port[i]); self->Port[i] = NULL; } /* Ignore the requested buffer metrics and just keep one JACK-sized buffer * ready for when requested. */ device->Frequency = jack_get_sample_rate(self->Client); device->UpdateSize = jack_get_buffer_size(self->Client); device->NumUpdates = 2; bufsize = device->UpdateSize; if(ConfigValueUInt(alstr_get_cstr(device->DeviceName), "jack", "buffer-size", &bufsize)) bufsize = maxu(NextPowerOf2(bufsize), device->UpdateSize); device->NumUpdates = (bufsize+device->UpdateSize) / device->UpdateSize; /* Force 32-bit float output. */ device->FmtType = DevFmtFloat; numchans = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); for(i = 0;i < numchans;i++) { char name[64]; snprintf(name, sizeof(name), "channel_%d", i+1); self->Port[i] = jack_port_register(self->Client, name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0); if(self->Port[i] == NULL) { ERR("Not enough JACK ports available for %s output\n", DevFmtChannelsString(device->FmtChans)); if(i == 0) return ALC_FALSE; break; } } if(i < numchans) { if(i == 1) device->FmtChans = DevFmtMono; else { for(--i;i >= 2;i--) { jack_port_unregister(self->Client, self->Port[i]); self->Port[i] = NULL; } device->FmtChans = DevFmtStereo; } } ll_ringbuffer_free(self->Ring); self->Ring = ll_ringbuffer_create(bufsize, FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder), true ); if(!self->Ring) { ERR("Failed to allocate ringbuffer\n"); return ALC_FALSE; } SetDefaultChannelOrder(device); return ALC_TRUE; } static ALCboolean ALCjackPlayback_start(ALCjackPlayback *self) { const char **ports; ALsizei i; if(jack_activate(self->Client)) { ERR("Failed to activate client\n"); return ALC_FALSE; } ports = jack_get_ports(self->Client, NULL, NULL, JackPortIsPhysical|JackPortIsInput); if(ports == NULL) { ERR("No physical playback ports found\n"); jack_deactivate(self->Client); return ALC_FALSE; } for(i = 0;i < MAX_OUTPUT_CHANNELS && self->Port[i];i++) { if(!ports[i]) { ERR("No physical playback port for \"%s\"\n", jack_port_name(self->Port[i])); break; } if(jack_connect(self->Client, jack_port_name(self->Port[i]), ports[i])) ERR("Failed to connect output port \"%s\" to \"%s\"\n", jack_port_name(self->Port[i]), ports[i]); } jack_free(ports); ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release); if(althrd_create(&self->thread, ALCjackPlayback_mixerProc, self) != althrd_success) { jack_deactivate(self->Client); return ALC_FALSE; } return ALC_TRUE; } static void ALCjackPlayback_stop(ALCjackPlayback *self) { int res; if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel)) return; alsem_post(&self->Sem); althrd_join(self->thread, &res); jack_deactivate(self->Client); } static ClockLatency ALCjackPlayback_getClockLatency(ALCjackPlayback *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; ClockLatency ret; ALCjackPlayback_lock(self); ret.ClockTime = GetDeviceClockTime(device); ret.Latency = ll_ringbuffer_read_space(self->Ring) * DEVICE_CLOCK_RES / device->Frequency; ALCjackPlayback_unlock(self); return ret; } static void jack_msg_handler(const char *message) { WARN("%s\n", message); } typedef struct ALCjackBackendFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } ALCjackBackendFactory; #define ALCJACKBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCjackBackendFactory, ALCbackendFactory) } } static ALCboolean ALCjackBackendFactory_init(ALCjackBackendFactory* UNUSED(self)) { void (*old_error_cb)(const char*); jack_client_t *client; jack_status_t status; if(!jack_load()) return ALC_FALSE; if(!GetConfigValueBool(NULL, "jack", "spawn-server", 0)) ClientOptions |= JackNoStartServer; old_error_cb = (&jack_error_callback ? jack_error_callback : NULL); jack_set_error_function(jack_msg_handler); client = jack_client_open("alsoft", ClientOptions, &status, NULL); jack_set_error_function(old_error_cb); if(client == NULL) { WARN("jack_client_open() failed, 0x%02x\n", status); if((status&JackServerFailed) && !(ClientOptions&JackNoStartServer)) ERR("Unable to connect to JACK server\n"); return ALC_FALSE; } jack_client_close(client); return ALC_TRUE; } static void ALCjackBackendFactory_deinit(ALCjackBackendFactory* UNUSED(self)) { #ifdef HAVE_DYNLOAD if(jack_handle) CloseLib(jack_handle); jack_handle = NULL; #endif } static ALCboolean ALCjackBackendFactory_querySupport(ALCjackBackendFactory* UNUSED(self), ALCbackend_Type type) { if(type == ALCbackend_Playback) return ALC_TRUE; return ALC_FALSE; } static void ALCjackBackendFactory_probe(ALCjackBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames) { switch(type) { case ALL_DEVICE_PROBE: alstr_append_range(outnames, jackDevice, jackDevice+sizeof(jackDevice)); break; case CAPTURE_DEVICE_PROBE: break; } } static ALCbackend* ALCjackBackendFactory_createBackend(ALCjackBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) { if(type == ALCbackend_Playback) { ALCjackPlayback *backend; NEW_OBJ(backend, ALCjackPlayback)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } return NULL; } DEFINE_ALCBACKENDFACTORY_VTABLE(ALCjackBackendFactory); ALCbackendFactory *ALCjackBackendFactory_getFactory(void) { static ALCjackBackendFactory factory = ALCJACKBACKENDFACTORY_INITIALIZER; return STATIC_CAST(ALCbackendFactory, &factory); } openal-soft-openal-soft-1.19.1/Alc/backends/loopback.c000066400000000000000000000104111335774445300225020ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2011 by Chris Robinson * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include "alMain.h" #include "alu.h" #include "backends/base.h" typedef struct ALCloopback { DERIVE_FROM_TYPE(ALCbackend); } ALCloopback; static void ALCloopback_Construct(ALCloopback *self, ALCdevice *device); static DECLARE_FORWARD(ALCloopback, ALCbackend, void, Destruct) static ALCenum ALCloopback_open(ALCloopback *self, const ALCchar *name); static ALCboolean ALCloopback_reset(ALCloopback *self); static ALCboolean ALCloopback_start(ALCloopback *self); static void ALCloopback_stop(ALCloopback *self); static DECLARE_FORWARD2(ALCloopback, ALCbackend, ALCenum, captureSamples, void*, ALCuint) static DECLARE_FORWARD(ALCloopback, ALCbackend, ALCuint, availableSamples) static DECLARE_FORWARD(ALCloopback, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(ALCloopback, ALCbackend, void, lock) static DECLARE_FORWARD(ALCloopback, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCloopback) DEFINE_ALCBACKEND_VTABLE(ALCloopback); static void ALCloopback_Construct(ALCloopback *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCloopback, ALCbackend, self); } static ALCenum ALCloopback_open(ALCloopback *self, const ALCchar *name) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; alstr_copy_cstr(&device->DeviceName, name); return ALC_NO_ERROR; } static ALCboolean ALCloopback_reset(ALCloopback *self) { SetDefaultWFXChannelOrder(STATIC_CAST(ALCbackend, self)->mDevice); return ALC_TRUE; } static ALCboolean ALCloopback_start(ALCloopback* UNUSED(self)) { return ALC_TRUE; } static void ALCloopback_stop(ALCloopback* UNUSED(self)) { } typedef struct ALCloopbackFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } ALCloopbackFactory; #define ALCNULLBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCloopbackFactory, ALCbackendFactory) } } ALCbackendFactory *ALCloopbackFactory_getFactory(void); static ALCboolean ALCloopbackFactory_init(ALCloopbackFactory *self); static DECLARE_FORWARD(ALCloopbackFactory, ALCbackendFactory, void, deinit) static ALCboolean ALCloopbackFactory_querySupport(ALCloopbackFactory *self, ALCbackend_Type type); static void ALCloopbackFactory_probe(ALCloopbackFactory *self, enum DevProbe type, al_string *outnames); static ALCbackend* ALCloopbackFactory_createBackend(ALCloopbackFactory *self, ALCdevice *device, ALCbackend_Type type); DEFINE_ALCBACKENDFACTORY_VTABLE(ALCloopbackFactory); ALCbackendFactory *ALCloopbackFactory_getFactory(void) { static ALCloopbackFactory factory = ALCNULLBACKENDFACTORY_INITIALIZER; return STATIC_CAST(ALCbackendFactory, &factory); } static ALCboolean ALCloopbackFactory_init(ALCloopbackFactory* UNUSED(self)) { return ALC_TRUE; } static ALCboolean ALCloopbackFactory_querySupport(ALCloopbackFactory* UNUSED(self), ALCbackend_Type type) { if(type == ALCbackend_Loopback) return ALC_TRUE; return ALC_FALSE; } static void ALCloopbackFactory_probe(ALCloopbackFactory* UNUSED(self), enum DevProbe UNUSED(type), al_string* UNUSED(outnames)) { } static ALCbackend* ALCloopbackFactory_createBackend(ALCloopbackFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) { if(type == ALCbackend_Loopback) { ALCloopback *backend; NEW_OBJ(backend, ALCloopback)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } return NULL; } openal-soft-openal-soft-1.19.1/Alc/backends/null.c000066400000000000000000000156731335774445300217010ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2010 by Chris Robinson * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #ifdef HAVE_WINDOWS_H #include #endif #include "alMain.h" #include "alu.h" #include "threads.h" #include "compat.h" #include "backends/base.h" typedef struct ALCnullBackend { DERIVE_FROM_TYPE(ALCbackend); ATOMIC(int) killNow; althrd_t thread; } ALCnullBackend; static int ALCnullBackend_mixerProc(void *ptr); static void ALCnullBackend_Construct(ALCnullBackend *self, ALCdevice *device); static DECLARE_FORWARD(ALCnullBackend, ALCbackend, void, Destruct) static ALCenum ALCnullBackend_open(ALCnullBackend *self, const ALCchar *name); static ALCboolean ALCnullBackend_reset(ALCnullBackend *self); static ALCboolean ALCnullBackend_start(ALCnullBackend *self); static void ALCnullBackend_stop(ALCnullBackend *self); static DECLARE_FORWARD2(ALCnullBackend, ALCbackend, ALCenum, captureSamples, void*, ALCuint) static DECLARE_FORWARD(ALCnullBackend, ALCbackend, ALCuint, availableSamples) static DECLARE_FORWARD(ALCnullBackend, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(ALCnullBackend, ALCbackend, void, lock) static DECLARE_FORWARD(ALCnullBackend, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCnullBackend) DEFINE_ALCBACKEND_VTABLE(ALCnullBackend); static const ALCchar nullDevice[] = "No Output"; static void ALCnullBackend_Construct(ALCnullBackend *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCnullBackend, ALCbackend, self); ATOMIC_INIT(&self->killNow, AL_TRUE); } static int ALCnullBackend_mixerProc(void *ptr) { ALCnullBackend *self = (ALCnullBackend*)ptr; ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; struct timespec now, start; ALuint64 avail, done; const long restTime = (long)((ALuint64)device->UpdateSize * 1000000000 / device->Frequency / 2); SetRTPriority(); althrd_setname(althrd_current(), MIXER_THREAD_NAME); done = 0; if(altimespec_get(&start, AL_TIME_UTC) != AL_TIME_UTC) { ERR("Failed to get starting time\n"); return 1; } while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) && ATOMIC_LOAD(&device->Connected, almemory_order_acquire)) { if(altimespec_get(&now, AL_TIME_UTC) != AL_TIME_UTC) { ERR("Failed to get current time\n"); return 1; } avail = (now.tv_sec - start.tv_sec) * device->Frequency; avail += (ALint64)(now.tv_nsec - start.tv_nsec) * device->Frequency / 1000000000; if(avail < done) { /* Oops, time skipped backwards. Reset the number of samples done * with one update available since we (likely) just came back from * sleeping. */ done = avail - device->UpdateSize; } if(avail-done < device->UpdateSize) al_nssleep(restTime); else while(avail-done >= device->UpdateSize) { ALCnullBackend_lock(self); aluMixData(device, NULL, device->UpdateSize); ALCnullBackend_unlock(self); done += device->UpdateSize; } } return 0; } static ALCenum ALCnullBackend_open(ALCnullBackend *self, const ALCchar *name) { ALCdevice *device; if(!name) name = nullDevice; else if(strcmp(name, nullDevice) != 0) return ALC_INVALID_VALUE; device = STATIC_CAST(ALCbackend, self)->mDevice; alstr_copy_cstr(&device->DeviceName, name); return ALC_NO_ERROR; } static ALCboolean ALCnullBackend_reset(ALCnullBackend *self) { SetDefaultWFXChannelOrder(STATIC_CAST(ALCbackend, self)->mDevice); return ALC_TRUE; } static ALCboolean ALCnullBackend_start(ALCnullBackend *self) { ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release); if(althrd_create(&self->thread, ALCnullBackend_mixerProc, self) != althrd_success) return ALC_FALSE; return ALC_TRUE; } static void ALCnullBackend_stop(ALCnullBackend *self) { int res; if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel)) return; althrd_join(self->thread, &res); } typedef struct ALCnullBackendFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } ALCnullBackendFactory; #define ALCNULLBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCnullBackendFactory, ALCbackendFactory) } } ALCbackendFactory *ALCnullBackendFactory_getFactory(void); static ALCboolean ALCnullBackendFactory_init(ALCnullBackendFactory *self); static DECLARE_FORWARD(ALCnullBackendFactory, ALCbackendFactory, void, deinit) static ALCboolean ALCnullBackendFactory_querySupport(ALCnullBackendFactory *self, ALCbackend_Type type); static void ALCnullBackendFactory_probe(ALCnullBackendFactory *self, enum DevProbe type, al_string *outnames); static ALCbackend* ALCnullBackendFactory_createBackend(ALCnullBackendFactory *self, ALCdevice *device, ALCbackend_Type type); DEFINE_ALCBACKENDFACTORY_VTABLE(ALCnullBackendFactory); ALCbackendFactory *ALCnullBackendFactory_getFactory(void) { static ALCnullBackendFactory factory = ALCNULLBACKENDFACTORY_INITIALIZER; return STATIC_CAST(ALCbackendFactory, &factory); } static ALCboolean ALCnullBackendFactory_init(ALCnullBackendFactory* UNUSED(self)) { return ALC_TRUE; } static ALCboolean ALCnullBackendFactory_querySupport(ALCnullBackendFactory* UNUSED(self), ALCbackend_Type type) { if(type == ALCbackend_Playback) return ALC_TRUE; return ALC_FALSE; } static void ALCnullBackendFactory_probe(ALCnullBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames) { switch(type) { case ALL_DEVICE_PROBE: case CAPTURE_DEVICE_PROBE: alstr_append_range(outnames, nullDevice, nullDevice+sizeof(nullDevice)); break; } } static ALCbackend* ALCnullBackendFactory_createBackend(ALCnullBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) { if(type == ALCbackend_Playback) { ALCnullBackend *backend; NEW_OBJ(backend, ALCnullBackend)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } return NULL; } openal-soft-openal-soft-1.19.1/Alc/backends/opensl.c000066400000000000000000001111521335774445300222140ustar00rootroot00000000000000/* * Copyright (C) 2011 The Android Open Source Project * * Licensed under the Apache License, Version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. * See the License for the specific language governing permissions and * limitations under the License. */ /* This is an OpenAL backend for Android using the native audio APIs based on * OpenSL ES 1.0.1. It is based on source code for the native-audio sample app * bundled with NDK. */ #include "config.h" #include #include #include "alMain.h" #include "alu.h" #include "ringbuffer.h" #include "threads.h" #include "compat.h" #include "backends/base.h" #include #include #include /* Helper macros */ #define VCALL(obj, func) ((*(obj))->func((obj), EXTRACT_VCALL_ARGS #define VCALL0(obj, func) ((*(obj))->func((obj) EXTRACT_VCALL_ARGS static const ALCchar opensl_device[] = "OpenSL"; static SLuint32 GetChannelMask(enum DevFmtChannels chans) { switch(chans) { case DevFmtMono: return SL_SPEAKER_FRONT_CENTER; case DevFmtStereo: return SL_SPEAKER_FRONT_LEFT|SL_SPEAKER_FRONT_RIGHT; case DevFmtQuad: return SL_SPEAKER_FRONT_LEFT|SL_SPEAKER_FRONT_RIGHT| SL_SPEAKER_BACK_LEFT|SL_SPEAKER_BACK_RIGHT; case DevFmtX51: return SL_SPEAKER_FRONT_LEFT|SL_SPEAKER_FRONT_RIGHT| SL_SPEAKER_FRONT_CENTER|SL_SPEAKER_LOW_FREQUENCY| SL_SPEAKER_SIDE_LEFT|SL_SPEAKER_SIDE_RIGHT; case DevFmtX51Rear: return SL_SPEAKER_FRONT_LEFT|SL_SPEAKER_FRONT_RIGHT| SL_SPEAKER_FRONT_CENTER|SL_SPEAKER_LOW_FREQUENCY| SL_SPEAKER_BACK_LEFT|SL_SPEAKER_BACK_RIGHT; case DevFmtX61: return SL_SPEAKER_FRONT_LEFT|SL_SPEAKER_FRONT_RIGHT| SL_SPEAKER_FRONT_CENTER|SL_SPEAKER_LOW_FREQUENCY| SL_SPEAKER_BACK_CENTER| SL_SPEAKER_SIDE_LEFT|SL_SPEAKER_SIDE_RIGHT; case DevFmtX71: return SL_SPEAKER_FRONT_LEFT|SL_SPEAKER_FRONT_RIGHT| SL_SPEAKER_FRONT_CENTER|SL_SPEAKER_LOW_FREQUENCY| SL_SPEAKER_BACK_LEFT|SL_SPEAKER_BACK_RIGHT| SL_SPEAKER_SIDE_LEFT|SL_SPEAKER_SIDE_RIGHT; case DevFmtAmbi3D: break; } return 0; } #ifdef SL_DATAFORMAT_PCM_EX static SLuint32 GetTypeRepresentation(enum DevFmtType type) { switch(type) { case DevFmtUByte: case DevFmtUShort: case DevFmtUInt: return SL_PCM_REPRESENTATION_UNSIGNED_INT; case DevFmtByte: case DevFmtShort: case DevFmtInt: return SL_PCM_REPRESENTATION_SIGNED_INT; case DevFmtFloat: return SL_PCM_REPRESENTATION_FLOAT; } return 0; } #endif static const char *res_str(SLresult result) { switch(result) { case SL_RESULT_SUCCESS: return "Success"; case SL_RESULT_PRECONDITIONS_VIOLATED: return "Preconditions violated"; case SL_RESULT_PARAMETER_INVALID: return "Parameter invalid"; case SL_RESULT_MEMORY_FAILURE: return "Memory failure"; case SL_RESULT_RESOURCE_ERROR: return "Resource error"; case SL_RESULT_RESOURCE_LOST: return "Resource lost"; case SL_RESULT_IO_ERROR: return "I/O error"; case SL_RESULT_BUFFER_INSUFFICIENT: return "Buffer insufficient"; case SL_RESULT_CONTENT_CORRUPTED: return "Content corrupted"; case SL_RESULT_CONTENT_UNSUPPORTED: return "Content unsupported"; case SL_RESULT_CONTENT_NOT_FOUND: return "Content not found"; case SL_RESULT_PERMISSION_DENIED: return "Permission denied"; case SL_RESULT_FEATURE_UNSUPPORTED: return "Feature unsupported"; case SL_RESULT_INTERNAL_ERROR: return "Internal error"; case SL_RESULT_UNKNOWN_ERROR: return "Unknown error"; case SL_RESULT_OPERATION_ABORTED: return "Operation aborted"; case SL_RESULT_CONTROL_LOST: return "Control lost"; #ifdef SL_RESULT_READONLY case SL_RESULT_READONLY: return "ReadOnly"; #endif #ifdef SL_RESULT_ENGINEOPTION_UNSUPPORTED case SL_RESULT_ENGINEOPTION_UNSUPPORTED: return "Engine option unsupported"; #endif #ifdef SL_RESULT_SOURCE_SINK_INCOMPATIBLE case SL_RESULT_SOURCE_SINK_INCOMPATIBLE: return "Source/Sink incompatible"; #endif } return "Unknown error code"; } #define PRINTERR(x, s) do { \ if((x) != SL_RESULT_SUCCESS) \ ERR("%s: %s\n", (s), res_str((x))); \ } while(0) typedef struct ALCopenslPlayback { DERIVE_FROM_TYPE(ALCbackend); /* engine interfaces */ SLObjectItf mEngineObj; SLEngineItf mEngine; /* output mix interfaces */ SLObjectItf mOutputMix; /* buffer queue player interfaces */ SLObjectItf mBufferQueueObj; ll_ringbuffer_t *mRing; alsem_t mSem; ALsizei mFrameSize; ATOMIC(ALenum) mKillNow; althrd_t mThread; } ALCopenslPlayback; static void ALCopenslPlayback_process(SLAndroidSimpleBufferQueueItf bq, void *context); static int ALCopenslPlayback_mixerProc(void *arg); static void ALCopenslPlayback_Construct(ALCopenslPlayback *self, ALCdevice *device); static void ALCopenslPlayback_Destruct(ALCopenslPlayback *self); static ALCenum ALCopenslPlayback_open(ALCopenslPlayback *self, const ALCchar *name); static ALCboolean ALCopenslPlayback_reset(ALCopenslPlayback *self); static ALCboolean ALCopenslPlayback_start(ALCopenslPlayback *self); static void ALCopenslPlayback_stop(ALCopenslPlayback *self); static DECLARE_FORWARD2(ALCopenslPlayback, ALCbackend, ALCenum, captureSamples, void*, ALCuint) static DECLARE_FORWARD(ALCopenslPlayback, ALCbackend, ALCuint, availableSamples) static ClockLatency ALCopenslPlayback_getClockLatency(ALCopenslPlayback *self); static DECLARE_FORWARD(ALCopenslPlayback, ALCbackend, void, lock) static DECLARE_FORWARD(ALCopenslPlayback, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCopenslPlayback) DEFINE_ALCBACKEND_VTABLE(ALCopenslPlayback); static void ALCopenslPlayback_Construct(ALCopenslPlayback *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCopenslPlayback, ALCbackend, self); self->mEngineObj = NULL; self->mEngine = NULL; self->mOutputMix = NULL; self->mBufferQueueObj = NULL; self->mRing = NULL; alsem_init(&self->mSem, 0); self->mFrameSize = 0; ATOMIC_INIT(&self->mKillNow, AL_FALSE); } static void ALCopenslPlayback_Destruct(ALCopenslPlayback* self) { if(self->mBufferQueueObj != NULL) VCALL0(self->mBufferQueueObj,Destroy)(); self->mBufferQueueObj = NULL; if(self->mOutputMix) VCALL0(self->mOutputMix,Destroy)(); self->mOutputMix = NULL; if(self->mEngineObj) VCALL0(self->mEngineObj,Destroy)(); self->mEngineObj = NULL; self->mEngine = NULL; ll_ringbuffer_free(self->mRing); self->mRing = NULL; alsem_destroy(&self->mSem); ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } /* this callback handler is called every time a buffer finishes playing */ static void ALCopenslPlayback_process(SLAndroidSimpleBufferQueueItf UNUSED(bq), void *context) { ALCopenslPlayback *self = context; /* A note on the ringbuffer usage: The buffer queue seems to hold on to the * pointer passed to the Enqueue method, rather than copying the audio. * Consequently, the ringbuffer contains the audio that is currently queued * and waiting to play. This process() callback is called when a buffer is * finished, so we simply move the read pointer up to indicate the space is * available for writing again, and wake up the mixer thread to mix and * queue more audio. */ ll_ringbuffer_read_advance(self->mRing, 1); alsem_post(&self->mSem); } static int ALCopenslPlayback_mixerProc(void *arg) { ALCopenslPlayback *self = arg; ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; SLAndroidSimpleBufferQueueItf bufferQueue; ll_ringbuffer_data_t data[2]; SLPlayItf player; SLresult result; SetRTPriority(); althrd_setname(althrd_current(), MIXER_THREAD_NAME); result = VCALL(self->mBufferQueueObj,GetInterface)(SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &bufferQueue); PRINTERR(result, "bufferQueue->GetInterface SL_IID_ANDROIDSIMPLEBUFFERQUEUE"); if(SL_RESULT_SUCCESS == result) { result = VCALL(self->mBufferQueueObj,GetInterface)(SL_IID_PLAY, &player); PRINTERR(result, "bufferQueue->GetInterface SL_IID_PLAY"); } ALCopenslPlayback_lock(self); if(SL_RESULT_SUCCESS != result) aluHandleDisconnect(device, "Failed to get playback buffer: 0x%08x", result); while(SL_RESULT_SUCCESS == result && !ATOMIC_LOAD(&self->mKillNow, almemory_order_acquire) && ATOMIC_LOAD(&device->Connected, almemory_order_acquire)) { size_t todo; if(ll_ringbuffer_write_space(self->mRing) == 0) { SLuint32 state = 0; result = VCALL(player,GetPlayState)(&state); PRINTERR(result, "player->GetPlayState"); if(SL_RESULT_SUCCESS == result && state != SL_PLAYSTATE_PLAYING) { result = VCALL(player,SetPlayState)(SL_PLAYSTATE_PLAYING); PRINTERR(result, "player->SetPlayState"); } if(SL_RESULT_SUCCESS != result) { aluHandleDisconnect(device, "Failed to start platback: 0x%08x", result); break; } if(ll_ringbuffer_write_space(self->mRing) == 0) { ALCopenslPlayback_unlock(self); alsem_wait(&self->mSem); ALCopenslPlayback_lock(self); continue; } } ll_ringbuffer_get_write_vector(self->mRing, data); aluMixData(device, data[0].buf, data[0].len*device->UpdateSize); if(data[1].len > 0) aluMixData(device, data[1].buf, data[1].len*device->UpdateSize); todo = data[0].len+data[1].len; ll_ringbuffer_write_advance(self->mRing, todo); for(size_t i = 0;i < todo;i++) { if(!data[0].len) { data[0] = data[1]; data[1].buf = NULL; data[1].len = 0; } result = VCALL(bufferQueue,Enqueue)(data[0].buf, device->UpdateSize*self->mFrameSize); PRINTERR(result, "bufferQueue->Enqueue"); if(SL_RESULT_SUCCESS != result) { aluHandleDisconnect(device, "Failed to queue audio: 0x%08x", result); break; } data[0].len--; data[0].buf += device->UpdateSize*self->mFrameSize; } } ALCopenslPlayback_unlock(self); return 0; } static ALCenum ALCopenslPlayback_open(ALCopenslPlayback *self, const ALCchar *name) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; SLresult result; if(!name) name = opensl_device; else if(strcmp(name, opensl_device) != 0) return ALC_INVALID_VALUE; // create engine result = slCreateEngine(&self->mEngineObj, 0, NULL, 0, NULL, NULL); PRINTERR(result, "slCreateEngine"); if(SL_RESULT_SUCCESS == result) { result = VCALL(self->mEngineObj,Realize)(SL_BOOLEAN_FALSE); PRINTERR(result, "engine->Realize"); } if(SL_RESULT_SUCCESS == result) { result = VCALL(self->mEngineObj,GetInterface)(SL_IID_ENGINE, &self->mEngine); PRINTERR(result, "engine->GetInterface"); } if(SL_RESULT_SUCCESS == result) { result = VCALL(self->mEngine,CreateOutputMix)(&self->mOutputMix, 0, NULL, NULL); PRINTERR(result, "engine->CreateOutputMix"); } if(SL_RESULT_SUCCESS == result) { result = VCALL(self->mOutputMix,Realize)(SL_BOOLEAN_FALSE); PRINTERR(result, "outputMix->Realize"); } if(SL_RESULT_SUCCESS != result) { if(self->mOutputMix != NULL) VCALL0(self->mOutputMix,Destroy)(); self->mOutputMix = NULL; if(self->mEngineObj != NULL) VCALL0(self->mEngineObj,Destroy)(); self->mEngineObj = NULL; self->mEngine = NULL; return ALC_INVALID_VALUE; } alstr_copy_cstr(&device->DeviceName, name); return ALC_NO_ERROR; } static ALCboolean ALCopenslPlayback_reset(ALCopenslPlayback *self) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; SLDataLocator_AndroidSimpleBufferQueue loc_bufq; SLDataLocator_OutputMix loc_outmix; SLDataSource audioSrc; SLDataSink audioSnk; ALuint sampleRate; SLInterfaceID ids[2]; SLboolean reqs[2]; SLresult result; if(self->mBufferQueueObj != NULL) VCALL0(self->mBufferQueueObj,Destroy)(); self->mBufferQueueObj = NULL; ll_ringbuffer_free(self->mRing); self->mRing = NULL; sampleRate = device->Frequency; #if 0 if(!(device->Flags&DEVICE_FREQUENCY_REQUEST)) { /* FIXME: Disabled until I figure out how to get the Context needed for * the getSystemService call. */ JNIEnv *env = Android_GetJNIEnv(); jobject jctx = Android_GetContext(); /* Get necessary stuff for using java.lang.Integer, * android.content.Context, and android.media.AudioManager. */ jclass int_cls = JCALL(env,FindClass)("java/lang/Integer"); jmethodID int_parseint = JCALL(env,GetStaticMethodID)(int_cls, "parseInt", "(Ljava/lang/String;)I" ); TRACE("Integer: %p, parseInt: %p\n", int_cls, int_parseint); jclass ctx_cls = JCALL(env,FindClass)("android/content/Context"); jfieldID ctx_audsvc = JCALL(env,GetStaticFieldID)(ctx_cls, "AUDIO_SERVICE", "Ljava/lang/String;" ); jmethodID ctx_getSysSvc = JCALL(env,GetMethodID)(ctx_cls, "getSystemService", "(Ljava/lang/String;)Ljava/lang/Object;" ); TRACE("Context: %p, AUDIO_SERVICE: %p, getSystemService: %p\n", ctx_cls, ctx_audsvc, ctx_getSysSvc); jclass audmgr_cls = JCALL(env,FindClass)("android/media/AudioManager"); jfieldID audmgr_prop_out_srate = JCALL(env,GetStaticFieldID)(audmgr_cls, "PROPERTY_OUTPUT_SAMPLE_RATE", "Ljava/lang/String;" ); jmethodID audmgr_getproperty = JCALL(env,GetMethodID)(audmgr_cls, "getProperty", "(Ljava/lang/String;)Ljava/lang/String;" ); TRACE("AudioManager: %p, PROPERTY_OUTPUT_SAMPLE_RATE: %p, getProperty: %p\n", audmgr_cls, audmgr_prop_out_srate, audmgr_getproperty); const char *strchars; jstring strobj; /* Now make the calls. */ //AudioManager audMgr = (AudioManager)getSystemService(Context.AUDIO_SERVICE); strobj = JCALL(env,GetStaticObjectField)(ctx_cls, ctx_audsvc); jobject audMgr = JCALL(env,CallObjectMethod)(jctx, ctx_getSysSvc, strobj); strchars = JCALL(env,GetStringUTFChars)(strobj, NULL); TRACE("Context.getSystemService(%s) = %p\n", strchars, audMgr); JCALL(env,ReleaseStringUTFChars)(strobj, strchars); //String srateStr = audMgr.getProperty(AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE); strobj = JCALL(env,GetStaticObjectField)(audmgr_cls, audmgr_prop_out_srate); jstring srateStr = JCALL(env,CallObjectMethod)(audMgr, audmgr_getproperty, strobj); strchars = JCALL(env,GetStringUTFChars)(strobj, NULL); TRACE("audMgr.getProperty(%s) = %p\n", strchars, srateStr); JCALL(env,ReleaseStringUTFChars)(strobj, strchars); //int sampleRate = Integer.parseInt(srateStr); sampleRate = JCALL(env,CallStaticIntMethod)(int_cls, int_parseint, srateStr); strchars = JCALL(env,GetStringUTFChars)(srateStr, NULL); TRACE("Got system sample rate %uhz (%s)\n", sampleRate, strchars); JCALL(env,ReleaseStringUTFChars)(srateStr, strchars); if(!sampleRate) sampleRate = device->Frequency; else sampleRate = maxu(sampleRate, MIN_OUTPUT_RATE); } #endif if(sampleRate != device->Frequency) { device->NumUpdates = (device->NumUpdates*sampleRate + (device->Frequency>>1)) / device->Frequency; device->NumUpdates = maxu(device->NumUpdates, 2); device->Frequency = sampleRate; } device->FmtChans = DevFmtStereo; device->FmtType = DevFmtShort; SetDefaultWFXChannelOrder(device); self->mFrameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); loc_bufq.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE; loc_bufq.numBuffers = device->NumUpdates; #ifdef SL_DATAFORMAT_PCM_EX SLDataFormat_PCM_EX format_pcm; format_pcm.formatType = SL_DATAFORMAT_PCM_EX; format_pcm.numChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); format_pcm.sampleRate = device->Frequency * 1000; format_pcm.bitsPerSample = BytesFromDevFmt(device->FmtType) * 8; format_pcm.containerSize = format_pcm.bitsPerSample; format_pcm.channelMask = GetChannelMask(device->FmtChans); format_pcm.endianness = IS_LITTLE_ENDIAN ? SL_BYTEORDER_LITTLEENDIAN : SL_BYTEORDER_BIGENDIAN; format_pcm.representation = GetTypeRepresentation(device->FmtType); #else SLDataFormat_PCM format_pcm; format_pcm.formatType = SL_DATAFORMAT_PCM; format_pcm.numChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); format_pcm.samplesPerSec = device->Frequency * 1000; format_pcm.bitsPerSample = BytesFromDevFmt(device->FmtType) * 8; format_pcm.containerSize = format_pcm.bitsPerSample; format_pcm.channelMask = GetChannelMask(device->FmtChans); format_pcm.endianness = IS_LITTLE_ENDIAN ? SL_BYTEORDER_LITTLEENDIAN : SL_BYTEORDER_BIGENDIAN; #endif audioSrc.pLocator = &loc_bufq; audioSrc.pFormat = &format_pcm; loc_outmix.locatorType = SL_DATALOCATOR_OUTPUTMIX; loc_outmix.outputMix = self->mOutputMix; audioSnk.pLocator = &loc_outmix; audioSnk.pFormat = NULL; ids[0] = SL_IID_ANDROIDSIMPLEBUFFERQUEUE; reqs[0] = SL_BOOLEAN_TRUE; ids[1] = SL_IID_ANDROIDCONFIGURATION; reqs[1] = SL_BOOLEAN_FALSE; result = VCALL(self->mEngine,CreateAudioPlayer)(&self->mBufferQueueObj, &audioSrc, &audioSnk, COUNTOF(ids), ids, reqs ); PRINTERR(result, "engine->CreateAudioPlayer"); if(SL_RESULT_SUCCESS == result) { /* Set the stream type to "media" (games, music, etc), if possible. */ SLAndroidConfigurationItf config; result = VCALL(self->mBufferQueueObj,GetInterface)(SL_IID_ANDROIDCONFIGURATION, &config); PRINTERR(result, "bufferQueue->GetInterface SL_IID_ANDROIDCONFIGURATION"); if(SL_RESULT_SUCCESS == result) { SLint32 streamType = SL_ANDROID_STREAM_MEDIA; result = VCALL(config,SetConfiguration)(SL_ANDROID_KEY_STREAM_TYPE, &streamType, sizeof(streamType) ); PRINTERR(result, "config->SetConfiguration"); } /* Clear any error since this was optional. */ result = SL_RESULT_SUCCESS; } if(SL_RESULT_SUCCESS == result) { result = VCALL(self->mBufferQueueObj,Realize)(SL_BOOLEAN_FALSE); PRINTERR(result, "bufferQueue->Realize"); } if(SL_RESULT_SUCCESS == result) { self->mRing = ll_ringbuffer_create(device->NumUpdates, self->mFrameSize*device->UpdateSize, true ); if(!self->mRing) { ERR("Out of memory allocating ring buffer %ux%u %u\n", device->UpdateSize, device->NumUpdates, self->mFrameSize); result = SL_RESULT_MEMORY_FAILURE; } } if(SL_RESULT_SUCCESS != result) { if(self->mBufferQueueObj != NULL) VCALL0(self->mBufferQueueObj,Destroy)(); self->mBufferQueueObj = NULL; return ALC_FALSE; } return ALC_TRUE; } static ALCboolean ALCopenslPlayback_start(ALCopenslPlayback *self) { SLAndroidSimpleBufferQueueItf bufferQueue; SLresult result; ll_ringbuffer_reset(self->mRing); result = VCALL(self->mBufferQueueObj,GetInterface)(SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &bufferQueue); PRINTERR(result, "bufferQueue->GetInterface"); if(SL_RESULT_SUCCESS != result) return ALC_FALSE; result = VCALL(bufferQueue,RegisterCallback)(ALCopenslPlayback_process, self); PRINTERR(result, "bufferQueue->RegisterCallback"); if(SL_RESULT_SUCCESS != result) return ALC_FALSE; ATOMIC_STORE_SEQ(&self->mKillNow, AL_FALSE); if(althrd_create(&self->mThread, ALCopenslPlayback_mixerProc, self) != althrd_success) { ERR("Failed to start mixer thread\n"); return ALC_FALSE; } return ALC_TRUE; } static void ALCopenslPlayback_stop(ALCopenslPlayback *self) { SLAndroidSimpleBufferQueueItf bufferQueue; SLPlayItf player; SLresult result; int res; if(ATOMIC_EXCHANGE_SEQ(&self->mKillNow, AL_TRUE)) return; alsem_post(&self->mSem); althrd_join(self->mThread, &res); result = VCALL(self->mBufferQueueObj,GetInterface)(SL_IID_PLAY, &player); PRINTERR(result, "bufferQueue->GetInterface"); if(SL_RESULT_SUCCESS == result) { result = VCALL(player,SetPlayState)(SL_PLAYSTATE_STOPPED); PRINTERR(result, "player->SetPlayState"); } result = VCALL(self->mBufferQueueObj,GetInterface)(SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &bufferQueue); PRINTERR(result, "bufferQueue->GetInterface"); if(SL_RESULT_SUCCESS == result) { result = VCALL0(bufferQueue,Clear)(); PRINTERR(result, "bufferQueue->Clear"); } if(SL_RESULT_SUCCESS == result) { result = VCALL(bufferQueue,RegisterCallback)(NULL, NULL); PRINTERR(result, "bufferQueue->RegisterCallback"); } if(SL_RESULT_SUCCESS == result) { SLAndroidSimpleBufferQueueState state; do { althrd_yield(); result = VCALL(bufferQueue,GetState)(&state); } while(SL_RESULT_SUCCESS == result && state.count > 0); PRINTERR(result, "bufferQueue->GetState"); } } static ClockLatency ALCopenslPlayback_getClockLatency(ALCopenslPlayback *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; ClockLatency ret; ALCopenslPlayback_lock(self); ret.ClockTime = GetDeviceClockTime(device); ret.Latency = ll_ringbuffer_read_space(self->mRing)*device->UpdateSize * DEVICE_CLOCK_RES / device->Frequency; ALCopenslPlayback_unlock(self); return ret; } typedef struct ALCopenslCapture { DERIVE_FROM_TYPE(ALCbackend); /* engine interfaces */ SLObjectItf mEngineObj; SLEngineItf mEngine; /* recording interfaces */ SLObjectItf mRecordObj; ll_ringbuffer_t *mRing; ALCuint mSplOffset; ALsizei mFrameSize; } ALCopenslCapture; static void ALCopenslCapture_process(SLAndroidSimpleBufferQueueItf bq, void *context); static void ALCopenslCapture_Construct(ALCopenslCapture *self, ALCdevice *device); static void ALCopenslCapture_Destruct(ALCopenslCapture *self); static ALCenum ALCopenslCapture_open(ALCopenslCapture *self, const ALCchar *name); static DECLARE_FORWARD(ALCopenslCapture, ALCbackend, ALCboolean, reset) static ALCboolean ALCopenslCapture_start(ALCopenslCapture *self); static void ALCopenslCapture_stop(ALCopenslCapture *self); static ALCenum ALCopenslCapture_captureSamples(ALCopenslCapture *self, ALCvoid *buffer, ALCuint samples); static ALCuint ALCopenslCapture_availableSamples(ALCopenslCapture *self); static DECLARE_FORWARD(ALCopenslCapture, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(ALCopenslCapture, ALCbackend, void, lock) static DECLARE_FORWARD(ALCopenslCapture, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCopenslCapture) DEFINE_ALCBACKEND_VTABLE(ALCopenslCapture); static void ALCopenslCapture_process(SLAndroidSimpleBufferQueueItf UNUSED(bq), void *context) { ALCopenslCapture *self = context; /* A new chunk has been written into the ring buffer, advance it. */ ll_ringbuffer_write_advance(self->mRing, 1); } static void ALCopenslCapture_Construct(ALCopenslCapture *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCopenslCapture, ALCbackend, self); self->mEngineObj = NULL; self->mEngine = NULL; self->mRecordObj = NULL; self->mRing = NULL; self->mSplOffset = 0; self->mFrameSize = 0; } static void ALCopenslCapture_Destruct(ALCopenslCapture *self) { if(self->mRecordObj != NULL) VCALL0(self->mRecordObj,Destroy)(); self->mRecordObj = NULL; if(self->mEngineObj != NULL) VCALL0(self->mEngineObj,Destroy)(); self->mEngineObj = NULL; self->mEngine = NULL; ll_ringbuffer_free(self->mRing); self->mRing = NULL; ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static ALCenum ALCopenslCapture_open(ALCopenslCapture *self, const ALCchar *name) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; SLDataLocator_AndroidSimpleBufferQueue loc_bq; SLAndroidSimpleBufferQueueItf bufferQueue; SLDataLocator_IODevice loc_dev; SLDataSource audioSrc; SLDataSink audioSnk; SLresult result; if(!name) name = opensl_device; else if(strcmp(name, opensl_device) != 0) return ALC_INVALID_VALUE; result = slCreateEngine(&self->mEngineObj, 0, NULL, 0, NULL, NULL); PRINTERR(result, "slCreateEngine"); if(SL_RESULT_SUCCESS == result) { result = VCALL(self->mEngineObj,Realize)(SL_BOOLEAN_FALSE); PRINTERR(result, "engine->Realize"); } if(SL_RESULT_SUCCESS == result) { result = VCALL(self->mEngineObj,GetInterface)(SL_IID_ENGINE, &self->mEngine); PRINTERR(result, "engine->GetInterface"); } if(SL_RESULT_SUCCESS == result) { /* Ensure the total length is at least 100ms */ ALsizei length = maxi(device->NumUpdates * device->UpdateSize, device->Frequency / 10); /* Ensure the per-chunk length is at least 10ms, and no more than 50ms. */ ALsizei update_len = clampi(device->NumUpdates*device->UpdateSize / 3, device->Frequency / 100, device->Frequency / 100 * 5); device->UpdateSize = update_len; device->NumUpdates = (length+update_len-1) / update_len; self->mFrameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); } loc_dev.locatorType = SL_DATALOCATOR_IODEVICE; loc_dev.deviceType = SL_IODEVICE_AUDIOINPUT; loc_dev.deviceID = SL_DEFAULTDEVICEID_AUDIOINPUT; loc_dev.device = NULL; audioSrc.pLocator = &loc_dev; audioSrc.pFormat = NULL; loc_bq.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE; loc_bq.numBuffers = device->NumUpdates; #ifdef SL_DATAFORMAT_PCM_EX SLDataFormat_PCM_EX format_pcm; format_pcm.formatType = SL_DATAFORMAT_PCM_EX; format_pcm.numChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); format_pcm.sampleRate = device->Frequency * 1000; format_pcm.bitsPerSample = BytesFromDevFmt(device->FmtType) * 8; format_pcm.containerSize = format_pcm.bitsPerSample; format_pcm.channelMask = GetChannelMask(device->FmtChans); format_pcm.endianness = IS_LITTLE_ENDIAN ? SL_BYTEORDER_LITTLEENDIAN : SL_BYTEORDER_BIGENDIAN; format_pcm.representation = GetTypeRepresentation(device->FmtType); #else SLDataFormat_PCM format_pcm; format_pcm.formatType = SL_DATAFORMAT_PCM; format_pcm.numChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); format_pcm.samplesPerSec = device->Frequency * 1000; format_pcm.bitsPerSample = BytesFromDevFmt(device->FmtType) * 8; format_pcm.containerSize = format_pcm.bitsPerSample; format_pcm.channelMask = GetChannelMask(device->FmtChans); format_pcm.endianness = IS_LITTLE_ENDIAN ? SL_BYTEORDER_LITTLEENDIAN : SL_BYTEORDER_BIGENDIAN; #endif audioSnk.pLocator = &loc_bq; audioSnk.pFormat = &format_pcm; if(SL_RESULT_SUCCESS == result) { const SLInterfaceID ids[2] = { SL_IID_ANDROIDSIMPLEBUFFERQUEUE, SL_IID_ANDROIDCONFIGURATION }; const SLboolean reqs[2] = { SL_BOOLEAN_TRUE, SL_BOOLEAN_FALSE }; result = VCALL(self->mEngine,CreateAudioRecorder)(&self->mRecordObj, &audioSrc, &audioSnk, COUNTOF(ids), ids, reqs ); PRINTERR(result, "engine->CreateAudioRecorder"); } if(SL_RESULT_SUCCESS == result) { /* Set the record preset to "generic", if possible. */ SLAndroidConfigurationItf config; result = VCALL(self->mRecordObj,GetInterface)(SL_IID_ANDROIDCONFIGURATION, &config); PRINTERR(result, "recordObj->GetInterface SL_IID_ANDROIDCONFIGURATION"); if(SL_RESULT_SUCCESS == result) { SLuint32 preset = SL_ANDROID_RECORDING_PRESET_GENERIC; result = VCALL(config,SetConfiguration)(SL_ANDROID_KEY_RECORDING_PRESET, &preset, sizeof(preset) ); PRINTERR(result, "config->SetConfiguration"); } /* Clear any error since this was optional. */ result = SL_RESULT_SUCCESS; } if(SL_RESULT_SUCCESS == result) { result = VCALL(self->mRecordObj,Realize)(SL_BOOLEAN_FALSE); PRINTERR(result, "recordObj->Realize"); } if(SL_RESULT_SUCCESS == result) { self->mRing = ll_ringbuffer_create(device->NumUpdates, device->UpdateSize*self->mFrameSize, false ); result = VCALL(self->mRecordObj,GetInterface)(SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &bufferQueue); PRINTERR(result, "recordObj->GetInterface"); } if(SL_RESULT_SUCCESS == result) { result = VCALL(bufferQueue,RegisterCallback)(ALCopenslCapture_process, self); PRINTERR(result, "bufferQueue->RegisterCallback"); } if(SL_RESULT_SUCCESS == result) { ALsizei chunk_size = device->UpdateSize * self->mFrameSize; ll_ringbuffer_data_t data[2]; size_t i; ll_ringbuffer_get_write_vector(self->mRing, data); for(i = 0;i < data[0].len && SL_RESULT_SUCCESS == result;i++) { result = VCALL(bufferQueue,Enqueue)(data[0].buf + chunk_size*i, chunk_size); PRINTERR(result, "bufferQueue->Enqueue"); } for(i = 0;i < data[1].len && SL_RESULT_SUCCESS == result;i++) { result = VCALL(bufferQueue,Enqueue)(data[1].buf + chunk_size*i, chunk_size); PRINTERR(result, "bufferQueue->Enqueue"); } } if(SL_RESULT_SUCCESS != result) { if(self->mRecordObj != NULL) VCALL0(self->mRecordObj,Destroy)(); self->mRecordObj = NULL; if(self->mEngineObj != NULL) VCALL0(self->mEngineObj,Destroy)(); self->mEngineObj = NULL; self->mEngine = NULL; return ALC_INVALID_VALUE; } alstr_copy_cstr(&device->DeviceName, name); return ALC_NO_ERROR; } static ALCboolean ALCopenslCapture_start(ALCopenslCapture *self) { SLRecordItf record; SLresult result; result = VCALL(self->mRecordObj,GetInterface)(SL_IID_RECORD, &record); PRINTERR(result, "recordObj->GetInterface"); if(SL_RESULT_SUCCESS == result) { result = VCALL(record,SetRecordState)(SL_RECORDSTATE_RECORDING); PRINTERR(result, "record->SetRecordState"); } if(SL_RESULT_SUCCESS != result) { ALCopenslCapture_lock(self); aluHandleDisconnect(STATIC_CAST(ALCbackend, self)->mDevice, "Failed to start capture: 0x%08x", result); ALCopenslCapture_unlock(self); return ALC_FALSE; } return ALC_TRUE; } static void ALCopenslCapture_stop(ALCopenslCapture *self) { SLRecordItf record; SLresult result; result = VCALL(self->mRecordObj,GetInterface)(SL_IID_RECORD, &record); PRINTERR(result, "recordObj->GetInterface"); if(SL_RESULT_SUCCESS == result) { result = VCALL(record,SetRecordState)(SL_RECORDSTATE_PAUSED); PRINTERR(result, "record->SetRecordState"); } } static ALCenum ALCopenslCapture_captureSamples(ALCopenslCapture *self, ALCvoid *buffer, ALCuint samples) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; ALsizei chunk_size = device->UpdateSize * self->mFrameSize; SLAndroidSimpleBufferQueueItf bufferQueue; ll_ringbuffer_data_t data[2]; SLresult result; ALCuint i; result = VCALL(self->mRecordObj,GetInterface)(SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &bufferQueue); PRINTERR(result, "recordObj->GetInterface"); /* Read the desired samples from the ring buffer then advance its read * pointer. */ ll_ringbuffer_get_read_vector(self->mRing, data); for(i = 0;i < samples;) { ALCuint rem = minu(samples - i, device->UpdateSize - self->mSplOffset); memcpy((ALCbyte*)buffer + i*self->mFrameSize, data[0].buf + self->mSplOffset*self->mFrameSize, rem * self->mFrameSize); self->mSplOffset += rem; if(self->mSplOffset == device->UpdateSize) { /* Finished a chunk, reset the offset and advance the read pointer. */ self->mSplOffset = 0; ll_ringbuffer_read_advance(self->mRing, 1); result = VCALL(bufferQueue,Enqueue)(data[0].buf, chunk_size); PRINTERR(result, "bufferQueue->Enqueue"); if(SL_RESULT_SUCCESS != result) break; data[0].len--; if(!data[0].len) data[0] = data[1]; else data[0].buf += chunk_size; } i += rem; } if(SL_RESULT_SUCCESS != result) { ALCopenslCapture_lock(self); aluHandleDisconnect(device, "Failed to update capture buffer: 0x%08x", result); ALCopenslCapture_unlock(self); return ALC_INVALID_DEVICE; } return ALC_NO_ERROR; } static ALCuint ALCopenslCapture_availableSamples(ALCopenslCapture *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; return ll_ringbuffer_read_space(self->mRing) * device->UpdateSize; } typedef struct ALCopenslBackendFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } ALCopenslBackendFactory; #define ALCOPENSLBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCopenslBackendFactory, ALCbackendFactory) } } static ALCboolean ALCopenslBackendFactory_init(ALCopenslBackendFactory* UNUSED(self)) { return ALC_TRUE; } static void ALCopenslBackendFactory_deinit(ALCopenslBackendFactory* UNUSED(self)) { } static ALCboolean ALCopenslBackendFactory_querySupport(ALCopenslBackendFactory* UNUSED(self), ALCbackend_Type type) { if(type == ALCbackend_Playback || type == ALCbackend_Capture) return ALC_TRUE; return ALC_FALSE; } static void ALCopenslBackendFactory_probe(ALCopenslBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames) { switch(type) { case ALL_DEVICE_PROBE: case CAPTURE_DEVICE_PROBE: alstr_append_range(outnames, opensl_device, opensl_device+sizeof(opensl_device)); break; } } static ALCbackend* ALCopenslBackendFactory_createBackend(ALCopenslBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) { if(type == ALCbackend_Playback) { ALCopenslPlayback *backend; NEW_OBJ(backend, ALCopenslPlayback)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } if(type == ALCbackend_Capture) { ALCopenslCapture *backend; NEW_OBJ(backend, ALCopenslCapture)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } return NULL; } DEFINE_ALCBACKENDFACTORY_VTABLE(ALCopenslBackendFactory); ALCbackendFactory *ALCopenslBackendFactory_getFactory(void) { static ALCopenslBackendFactory factory = ALCOPENSLBACKENDFACTORY_INITIALIZER; return STATIC_CAST(ALCbackendFactory, &factory); } openal-soft-openal-soft-1.19.1/Alc/backends/oss.c000066400000000000000000000602461335774445300215270ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include #include #include #include #include #include #include #include #include #include "alMain.h" #include "alu.h" #include "alconfig.h" #include "ringbuffer.h" #include "threads.h" #include "compat.h" #include "backends/base.h" #include /* * The OSS documentation talks about SOUND_MIXER_READ, but the header * only contains MIXER_READ. Play safe. Same for WRITE. */ #ifndef SOUND_MIXER_READ #define SOUND_MIXER_READ MIXER_READ #endif #ifndef SOUND_MIXER_WRITE #define SOUND_MIXER_WRITE MIXER_WRITE #endif #if defined(SOUND_VERSION) && (SOUND_VERSION < 0x040000) #define ALC_OSS_COMPAT #endif #ifndef SNDCTL_AUDIOINFO #define ALC_OSS_COMPAT #endif /* * FreeBSD strongly discourages the use of specific devices, * such as those returned in oss_audioinfo.devnode */ #ifdef __FreeBSD__ #define ALC_OSS_DEVNODE_TRUC #endif struct oss_device { const ALCchar *handle; const char *path; struct oss_device *next; }; static struct oss_device oss_playback = { "OSS Default", "/dev/dsp", NULL }; static struct oss_device oss_capture = { "OSS Default", "/dev/dsp", NULL }; #ifdef ALC_OSS_COMPAT #define DSP_CAP_OUTPUT 0x00020000 #define DSP_CAP_INPUT 0x00010000 static void ALCossListPopulate(struct oss_device *UNUSED(devlist), int UNUSED(type_flag)) { } #else #ifndef HAVE_STRNLEN static size_t strnlen(const char *str, size_t maxlen) { const char *end = memchr(str, 0, maxlen); if(!end) return maxlen; return end - str; } #endif static void ALCossListAppend(struct oss_device *list, const char *handle, size_t hlen, const char *path, size_t plen) { struct oss_device *next; struct oss_device *last; size_t i; /* skip the first item "OSS Default" */ last = list; next = list->next; #ifdef ALC_OSS_DEVNODE_TRUC for(i = 0;i < plen;i++) { if(path[i] == '.') { if(strncmp(path + i, handle + hlen + i - plen, plen - i) == 0) hlen = hlen + i - plen; plen = i; } } #else (void)i; #endif if(handle[0] == '\0') { handle = path; hlen = plen; } while(next != NULL) { if(strncmp(next->path, path, plen) == 0) return; last = next; next = next->next; } next = (struct oss_device*)malloc(sizeof(struct oss_device) + hlen + plen + 2); next->handle = (char*)(next + 1); next->path = next->handle + hlen + 1; next->next = NULL; last->next = next; strncpy((char*)next->handle, handle, hlen); ((char*)next->handle)[hlen] = '\0'; strncpy((char*)next->path, path, plen); ((char*)next->path)[plen] = '\0'; TRACE("Got device \"%s\", \"%s\"\n", next->handle, next->path); } static void ALCossListPopulate(struct oss_device *devlist, int type_flag) { struct oss_sysinfo si; struct oss_audioinfo ai; int fd, i; if((fd=open("/dev/mixer", O_RDONLY)) < 0) { TRACE("Could not open /dev/mixer: %s\n", strerror(errno)); return; } if(ioctl(fd, SNDCTL_SYSINFO, &si) == -1) { TRACE("SNDCTL_SYSINFO failed: %s\n", strerror(errno)); goto done; } for(i = 0;i < si.numaudios;i++) { const char *handle; size_t len; ai.dev = i; if(ioctl(fd, SNDCTL_AUDIOINFO, &ai) == -1) { ERR("SNDCTL_AUDIOINFO (%d) failed: %s\n", i, strerror(errno)); continue; } if(ai.devnode[0] == '\0') continue; if(ai.handle[0] != '\0') { len = strnlen(ai.handle, sizeof(ai.handle)); handle = ai.handle; } else { len = strnlen(ai.name, sizeof(ai.name)); handle = ai.name; } if((ai.caps&type_flag)) ALCossListAppend(devlist, handle, len, ai.devnode, strnlen(ai.devnode, sizeof(ai.devnode))); } done: close(fd); } #endif static void ALCossListFree(struct oss_device *list) { struct oss_device *cur; if(list == NULL) return; /* skip the first item "OSS Default" */ cur = list->next; list->next = NULL; while(cur != NULL) { struct oss_device *next = cur->next; free(cur); cur = next; } } static int log2i(ALCuint x) { int y = 0; while (x > 1) { x >>= 1; y++; } return y; } typedef struct ALCplaybackOSS { DERIVE_FROM_TYPE(ALCbackend); int fd; ALubyte *mix_data; int data_size; ATOMIC(ALenum) killNow; althrd_t thread; } ALCplaybackOSS; static int ALCplaybackOSS_mixerProc(void *ptr); static void ALCplaybackOSS_Construct(ALCplaybackOSS *self, ALCdevice *device); static void ALCplaybackOSS_Destruct(ALCplaybackOSS *self); static ALCenum ALCplaybackOSS_open(ALCplaybackOSS *self, const ALCchar *name); static ALCboolean ALCplaybackOSS_reset(ALCplaybackOSS *self); static ALCboolean ALCplaybackOSS_start(ALCplaybackOSS *self); static void ALCplaybackOSS_stop(ALCplaybackOSS *self); static DECLARE_FORWARD2(ALCplaybackOSS, ALCbackend, ALCenum, captureSamples, ALCvoid*, ALCuint) static DECLARE_FORWARD(ALCplaybackOSS, ALCbackend, ALCuint, availableSamples) static DECLARE_FORWARD(ALCplaybackOSS, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(ALCplaybackOSS, ALCbackend, void, lock) static DECLARE_FORWARD(ALCplaybackOSS, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCplaybackOSS) DEFINE_ALCBACKEND_VTABLE(ALCplaybackOSS); static int ALCplaybackOSS_mixerProc(void *ptr) { ALCplaybackOSS *self = (ALCplaybackOSS*)ptr; ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; struct timeval timeout; ALubyte *write_ptr; ALint frame_size; ALint to_write; ssize_t wrote; fd_set wfds; int sret; SetRTPriority(); althrd_setname(althrd_current(), MIXER_THREAD_NAME); frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); ALCplaybackOSS_lock(self); while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) && ATOMIC_LOAD(&device->Connected, almemory_order_acquire)) { FD_ZERO(&wfds); FD_SET(self->fd, &wfds); timeout.tv_sec = 1; timeout.tv_usec = 0; ALCplaybackOSS_unlock(self); sret = select(self->fd+1, NULL, &wfds, NULL, &timeout); ALCplaybackOSS_lock(self); if(sret < 0) { if(errno == EINTR) continue; ERR("select failed: %s\n", strerror(errno)); aluHandleDisconnect(device, "Failed waiting for playback buffer: %s", strerror(errno)); break; } else if(sret == 0) { WARN("select timeout\n"); continue; } write_ptr = self->mix_data; to_write = self->data_size; aluMixData(device, write_ptr, to_write/frame_size); while(to_write > 0 && !ATOMIC_LOAD_SEQ(&self->killNow)) { wrote = write(self->fd, write_ptr, to_write); if(wrote < 0) { if(errno == EAGAIN || errno == EWOULDBLOCK || errno == EINTR) continue; ERR("write failed: %s\n", strerror(errno)); aluHandleDisconnect(device, "Failed writing playback samples: %s", strerror(errno)); break; } to_write -= wrote; write_ptr += wrote; } } ALCplaybackOSS_unlock(self); return 0; } static void ALCplaybackOSS_Construct(ALCplaybackOSS *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCplaybackOSS, ALCbackend, self); self->fd = -1; ATOMIC_INIT(&self->killNow, AL_FALSE); } static void ALCplaybackOSS_Destruct(ALCplaybackOSS *self) { if(self->fd != -1) close(self->fd); self->fd = -1; ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static ALCenum ALCplaybackOSS_open(ALCplaybackOSS *self, const ALCchar *name) { struct oss_device *dev = &oss_playback; ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; if(!name || strcmp(name, dev->handle) == 0) name = dev->handle; else { if(!dev->next) { ALCossListPopulate(&oss_playback, DSP_CAP_OUTPUT); dev = &oss_playback; } while(dev != NULL) { if (strcmp(dev->handle, name) == 0) break; dev = dev->next; } if(dev == NULL) { WARN("Could not find \"%s\" in device list\n", name); return ALC_INVALID_VALUE; } } self->fd = open(dev->path, O_WRONLY); if(self->fd == -1) { ERR("Could not open %s: %s\n", dev->path, strerror(errno)); return ALC_INVALID_VALUE; } alstr_copy_cstr(&device->DeviceName, name); return ALC_NO_ERROR; } static ALCboolean ALCplaybackOSS_reset(ALCplaybackOSS *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; int numFragmentsLogSize; int log2FragmentSize; unsigned int periods; audio_buf_info info; ALuint frameSize; int numChannels; int ossFormat; int ossSpeed; char *err; switch(device->FmtType) { case DevFmtByte: ossFormat = AFMT_S8; break; case DevFmtUByte: ossFormat = AFMT_U8; break; case DevFmtUShort: case DevFmtInt: case DevFmtUInt: case DevFmtFloat: device->FmtType = DevFmtShort; /* fall-through */ case DevFmtShort: ossFormat = AFMT_S16_NE; break; } periods = device->NumUpdates; numChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); ossSpeed = device->Frequency; frameSize = numChannels * BytesFromDevFmt(device->FmtType); /* According to the OSS spec, 16 bytes (log2(16)) is the minimum. */ log2FragmentSize = maxi(log2i(device->UpdateSize*frameSize), 4); numFragmentsLogSize = (periods << 16) | log2FragmentSize; #define CHECKERR(func) if((func) < 0) { \ err = #func; \ goto err; \ } /* Don't fail if SETFRAGMENT fails. We can handle just about anything * that's reported back via GETOSPACE */ ioctl(self->fd, SNDCTL_DSP_SETFRAGMENT, &numFragmentsLogSize); CHECKERR(ioctl(self->fd, SNDCTL_DSP_SETFMT, &ossFormat)); CHECKERR(ioctl(self->fd, SNDCTL_DSP_CHANNELS, &numChannels)); CHECKERR(ioctl(self->fd, SNDCTL_DSP_SPEED, &ossSpeed)); CHECKERR(ioctl(self->fd, SNDCTL_DSP_GETOSPACE, &info)); if(0) { err: ERR("%s failed: %s\n", err, strerror(errno)); return ALC_FALSE; } #undef CHECKERR if((int)ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder) != numChannels) { ERR("Failed to set %s, got %d channels instead\n", DevFmtChannelsString(device->FmtChans), numChannels); return ALC_FALSE; } if(!((ossFormat == AFMT_S8 && device->FmtType == DevFmtByte) || (ossFormat == AFMT_U8 && device->FmtType == DevFmtUByte) || (ossFormat == AFMT_S16_NE && device->FmtType == DevFmtShort))) { ERR("Failed to set %s samples, got OSS format %#x\n", DevFmtTypeString(device->FmtType), ossFormat); return ALC_FALSE; } device->Frequency = ossSpeed; device->UpdateSize = info.fragsize / frameSize; device->NumUpdates = info.fragments; SetDefaultChannelOrder(device); return ALC_TRUE; } static ALCboolean ALCplaybackOSS_start(ALCplaybackOSS *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; self->data_size = device->UpdateSize * FrameSizeFromDevFmt( device->FmtChans, device->FmtType, device->AmbiOrder ); self->mix_data = calloc(1, self->data_size); ATOMIC_STORE_SEQ(&self->killNow, AL_FALSE); if(althrd_create(&self->thread, ALCplaybackOSS_mixerProc, self) != althrd_success) { free(self->mix_data); self->mix_data = NULL; return ALC_FALSE; } return ALC_TRUE; } static void ALCplaybackOSS_stop(ALCplaybackOSS *self) { int res; if(ATOMIC_EXCHANGE_SEQ(&self->killNow, AL_TRUE)) return; althrd_join(self->thread, &res); if(ioctl(self->fd, SNDCTL_DSP_RESET) != 0) ERR("Error resetting device: %s\n", strerror(errno)); free(self->mix_data); self->mix_data = NULL; } typedef struct ALCcaptureOSS { DERIVE_FROM_TYPE(ALCbackend); int fd; ll_ringbuffer_t *ring; ATOMIC(ALenum) killNow; althrd_t thread; } ALCcaptureOSS; static int ALCcaptureOSS_recordProc(void *ptr); static void ALCcaptureOSS_Construct(ALCcaptureOSS *self, ALCdevice *device); static void ALCcaptureOSS_Destruct(ALCcaptureOSS *self); static ALCenum ALCcaptureOSS_open(ALCcaptureOSS *self, const ALCchar *name); static DECLARE_FORWARD(ALCcaptureOSS, ALCbackend, ALCboolean, reset) static ALCboolean ALCcaptureOSS_start(ALCcaptureOSS *self); static void ALCcaptureOSS_stop(ALCcaptureOSS *self); static ALCenum ALCcaptureOSS_captureSamples(ALCcaptureOSS *self, ALCvoid *buffer, ALCuint samples); static ALCuint ALCcaptureOSS_availableSamples(ALCcaptureOSS *self); static DECLARE_FORWARD(ALCcaptureOSS, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(ALCcaptureOSS, ALCbackend, void, lock) static DECLARE_FORWARD(ALCcaptureOSS, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCcaptureOSS) DEFINE_ALCBACKEND_VTABLE(ALCcaptureOSS); static int ALCcaptureOSS_recordProc(void *ptr) { ALCcaptureOSS *self = (ALCcaptureOSS*)ptr; ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; struct timeval timeout; int frame_size; fd_set rfds; ssize_t amt; int sret; SetRTPriority(); althrd_setname(althrd_current(), RECORD_THREAD_NAME); frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); while(!ATOMIC_LOAD_SEQ(&self->killNow)) { ll_ringbuffer_data_t vec[2]; FD_ZERO(&rfds); FD_SET(self->fd, &rfds); timeout.tv_sec = 1; timeout.tv_usec = 0; sret = select(self->fd+1, &rfds, NULL, NULL, &timeout); if(sret < 0) { if(errno == EINTR) continue; ERR("select failed: %s\n", strerror(errno)); aluHandleDisconnect(device, "Failed to check capture samples: %s", strerror(errno)); break; } else if(sret == 0) { WARN("select timeout\n"); continue; } ll_ringbuffer_get_write_vector(self->ring, vec); if(vec[0].len > 0) { amt = read(self->fd, vec[0].buf, vec[0].len*frame_size); if(amt < 0) { ERR("read failed: %s\n", strerror(errno)); ALCcaptureOSS_lock(self); aluHandleDisconnect(device, "Failed reading capture samples: %s", strerror(errno)); ALCcaptureOSS_unlock(self); break; } ll_ringbuffer_write_advance(self->ring, amt/frame_size); } } return 0; } static void ALCcaptureOSS_Construct(ALCcaptureOSS *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCcaptureOSS, ALCbackend, self); self->fd = -1; self->ring = NULL; ATOMIC_INIT(&self->killNow, AL_FALSE); } static void ALCcaptureOSS_Destruct(ALCcaptureOSS *self) { if(self->fd != -1) close(self->fd); self->fd = -1; ll_ringbuffer_free(self->ring); self->ring = NULL; ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static ALCenum ALCcaptureOSS_open(ALCcaptureOSS *self, const ALCchar *name) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; struct oss_device *dev = &oss_capture; int numFragmentsLogSize; int log2FragmentSize; unsigned int periods; audio_buf_info info; ALuint frameSize; int numChannels; int ossFormat; int ossSpeed; char *err; if(!name || strcmp(name, dev->handle) == 0) name = dev->handle; else { if(!dev->next) { ALCossListPopulate(&oss_capture, DSP_CAP_INPUT); dev = &oss_capture; } while(dev != NULL) { if (strcmp(dev->handle, name) == 0) break; dev = dev->next; } if(dev == NULL) { WARN("Could not find \"%s\" in device list\n", name); return ALC_INVALID_VALUE; } } self->fd = open(dev->path, O_RDONLY); if(self->fd == -1) { ERR("Could not open %s: %s\n", dev->path, strerror(errno)); return ALC_INVALID_VALUE; } switch(device->FmtType) { case DevFmtByte: ossFormat = AFMT_S8; break; case DevFmtUByte: ossFormat = AFMT_U8; break; case DevFmtShort: ossFormat = AFMT_S16_NE; break; case DevFmtUShort: case DevFmtInt: case DevFmtUInt: case DevFmtFloat: ERR("%s capture samples not supported\n", DevFmtTypeString(device->FmtType)); return ALC_INVALID_VALUE; } periods = 4; numChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); frameSize = numChannels * BytesFromDevFmt(device->FmtType); ossSpeed = device->Frequency; log2FragmentSize = log2i(device->UpdateSize * device->NumUpdates * frameSize / periods); /* according to the OSS spec, 16 bytes are the minimum */ if (log2FragmentSize < 4) log2FragmentSize = 4; numFragmentsLogSize = (periods << 16) | log2FragmentSize; #define CHECKERR(func) if((func) < 0) { \ err = #func; \ goto err; \ } CHECKERR(ioctl(self->fd, SNDCTL_DSP_SETFRAGMENT, &numFragmentsLogSize)); CHECKERR(ioctl(self->fd, SNDCTL_DSP_SETFMT, &ossFormat)); CHECKERR(ioctl(self->fd, SNDCTL_DSP_CHANNELS, &numChannels)); CHECKERR(ioctl(self->fd, SNDCTL_DSP_SPEED, &ossSpeed)); CHECKERR(ioctl(self->fd, SNDCTL_DSP_GETISPACE, &info)); if(0) { err: ERR("%s failed: %s\n", err, strerror(errno)); close(self->fd); self->fd = -1; return ALC_INVALID_VALUE; } #undef CHECKERR if((int)ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder) != numChannels) { ERR("Failed to set %s, got %d channels instead\n", DevFmtChannelsString(device->FmtChans), numChannels); close(self->fd); self->fd = -1; return ALC_INVALID_VALUE; } if(!((ossFormat == AFMT_S8 && device->FmtType == DevFmtByte) || (ossFormat == AFMT_U8 && device->FmtType == DevFmtUByte) || (ossFormat == AFMT_S16_NE && device->FmtType == DevFmtShort))) { ERR("Failed to set %s samples, got OSS format %#x\n", DevFmtTypeString(device->FmtType), ossFormat); close(self->fd); self->fd = -1; return ALC_INVALID_VALUE; } self->ring = ll_ringbuffer_create(device->UpdateSize*device->NumUpdates, frameSize, false); if(!self->ring) { ERR("Ring buffer create failed\n"); close(self->fd); self->fd = -1; return ALC_OUT_OF_MEMORY; } alstr_copy_cstr(&device->DeviceName, name); return ALC_NO_ERROR; } static ALCboolean ALCcaptureOSS_start(ALCcaptureOSS *self) { ATOMIC_STORE_SEQ(&self->killNow, AL_FALSE); if(althrd_create(&self->thread, ALCcaptureOSS_recordProc, self) != althrd_success) return ALC_FALSE; return ALC_TRUE; } static void ALCcaptureOSS_stop(ALCcaptureOSS *self) { int res; if(ATOMIC_EXCHANGE_SEQ(&self->killNow, AL_TRUE)) return; althrd_join(self->thread, &res); if(ioctl(self->fd, SNDCTL_DSP_RESET) != 0) ERR("Error resetting device: %s\n", strerror(errno)); } static ALCenum ALCcaptureOSS_captureSamples(ALCcaptureOSS *self, ALCvoid *buffer, ALCuint samples) { ll_ringbuffer_read(self->ring, buffer, samples); return ALC_NO_ERROR; } static ALCuint ALCcaptureOSS_availableSamples(ALCcaptureOSS *self) { return ll_ringbuffer_read_space(self->ring); } typedef struct ALCossBackendFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } ALCossBackendFactory; #define ALCOSSBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCossBackendFactory, ALCbackendFactory) } } ALCbackendFactory *ALCossBackendFactory_getFactory(void); static ALCboolean ALCossBackendFactory_init(ALCossBackendFactory *self); static void ALCossBackendFactory_deinit(ALCossBackendFactory *self); static ALCboolean ALCossBackendFactory_querySupport(ALCossBackendFactory *self, ALCbackend_Type type); static void ALCossBackendFactory_probe(ALCossBackendFactory *self, enum DevProbe type, al_string *outnames); static ALCbackend* ALCossBackendFactory_createBackend(ALCossBackendFactory *self, ALCdevice *device, ALCbackend_Type type); DEFINE_ALCBACKENDFACTORY_VTABLE(ALCossBackendFactory); ALCbackendFactory *ALCossBackendFactory_getFactory(void) { static ALCossBackendFactory factory = ALCOSSBACKENDFACTORY_INITIALIZER; return STATIC_CAST(ALCbackendFactory, &factory); } ALCboolean ALCossBackendFactory_init(ALCossBackendFactory* UNUSED(self)) { ConfigValueStr(NULL, "oss", "device", &oss_playback.path); ConfigValueStr(NULL, "oss", "capture", &oss_capture.path); return ALC_TRUE; } void ALCossBackendFactory_deinit(ALCossBackendFactory* UNUSED(self)) { ALCossListFree(&oss_playback); ALCossListFree(&oss_capture); } ALCboolean ALCossBackendFactory_querySupport(ALCossBackendFactory* UNUSED(self), ALCbackend_Type type) { if(type == ALCbackend_Playback || type == ALCbackend_Capture) return ALC_TRUE; return ALC_FALSE; } void ALCossBackendFactory_probe(ALCossBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames) { struct oss_device *cur = NULL; switch(type) { case ALL_DEVICE_PROBE: ALCossListFree(&oss_playback); ALCossListPopulate(&oss_playback, DSP_CAP_OUTPUT); cur = &oss_playback; break; case CAPTURE_DEVICE_PROBE: ALCossListFree(&oss_capture); ALCossListPopulate(&oss_capture, DSP_CAP_INPUT); cur = &oss_capture; break; } while(cur != NULL) { #ifdef HAVE_STAT struct stat buf; if(stat(cur->path, &buf) == 0) #endif alstr_append_range(outnames, cur->handle, cur->handle+strlen(cur->handle)+1); cur = cur->next; } } ALCbackend* ALCossBackendFactory_createBackend(ALCossBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) { if(type == ALCbackend_Playback) { ALCplaybackOSS *backend; NEW_OBJ(backend, ALCplaybackOSS)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } if(type == ALCbackend_Capture) { ALCcaptureOSS *backend; NEW_OBJ(backend, ALCcaptureOSS)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } return NULL; } openal-soft-openal-soft-1.19.1/Alc/backends/portaudio.c000066400000000000000000000425421335774445300227300ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include "alMain.h" #include "alu.h" #include "alconfig.h" #include "ringbuffer.h" #include "compat.h" #include "backends/base.h" #include static const ALCchar pa_device[] = "PortAudio Default"; #ifdef HAVE_DYNLOAD static void *pa_handle; #define MAKE_FUNC(x) static __typeof(x) * p##x MAKE_FUNC(Pa_Initialize); MAKE_FUNC(Pa_Terminate); MAKE_FUNC(Pa_GetErrorText); MAKE_FUNC(Pa_StartStream); MAKE_FUNC(Pa_StopStream); MAKE_FUNC(Pa_OpenStream); MAKE_FUNC(Pa_CloseStream); MAKE_FUNC(Pa_GetDefaultOutputDevice); MAKE_FUNC(Pa_GetDefaultInputDevice); MAKE_FUNC(Pa_GetStreamInfo); #undef MAKE_FUNC #define Pa_Initialize pPa_Initialize #define Pa_Terminate pPa_Terminate #define Pa_GetErrorText pPa_GetErrorText #define Pa_StartStream pPa_StartStream #define Pa_StopStream pPa_StopStream #define Pa_OpenStream pPa_OpenStream #define Pa_CloseStream pPa_CloseStream #define Pa_GetDefaultOutputDevice pPa_GetDefaultOutputDevice #define Pa_GetDefaultInputDevice pPa_GetDefaultInputDevice #define Pa_GetStreamInfo pPa_GetStreamInfo #endif static ALCboolean pa_load(void) { PaError err; #ifdef HAVE_DYNLOAD if(!pa_handle) { #ifdef _WIN32 # define PALIB "portaudio.dll" #elif defined(__APPLE__) && defined(__MACH__) # define PALIB "libportaudio.2.dylib" #elif defined(__OpenBSD__) # define PALIB "libportaudio.so" #else # define PALIB "libportaudio.so.2" #endif pa_handle = LoadLib(PALIB); if(!pa_handle) return ALC_FALSE; #define LOAD_FUNC(f) do { \ p##f = GetSymbol(pa_handle, #f); \ if(p##f == NULL) \ { \ CloseLib(pa_handle); \ pa_handle = NULL; \ return ALC_FALSE; \ } \ } while(0) LOAD_FUNC(Pa_Initialize); LOAD_FUNC(Pa_Terminate); LOAD_FUNC(Pa_GetErrorText); LOAD_FUNC(Pa_StartStream); LOAD_FUNC(Pa_StopStream); LOAD_FUNC(Pa_OpenStream); LOAD_FUNC(Pa_CloseStream); LOAD_FUNC(Pa_GetDefaultOutputDevice); LOAD_FUNC(Pa_GetDefaultInputDevice); LOAD_FUNC(Pa_GetStreamInfo); #undef LOAD_FUNC if((err=Pa_Initialize()) != paNoError) { ERR("Pa_Initialize() returned an error: %s\n", Pa_GetErrorText(err)); CloseLib(pa_handle); pa_handle = NULL; return ALC_FALSE; } } #else if((err=Pa_Initialize()) != paNoError) { ERR("Pa_Initialize() returned an error: %s\n", Pa_GetErrorText(err)); return ALC_FALSE; } #endif return ALC_TRUE; } typedef struct ALCportPlayback { DERIVE_FROM_TYPE(ALCbackend); PaStream *stream; PaStreamParameters params; ALuint update_size; } ALCportPlayback; static int ALCportPlayback_WriteCallback(const void *inputBuffer, void *outputBuffer, unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *timeInfo, const PaStreamCallbackFlags statusFlags, void *userData); static void ALCportPlayback_Construct(ALCportPlayback *self, ALCdevice *device); static void ALCportPlayback_Destruct(ALCportPlayback *self); static ALCenum ALCportPlayback_open(ALCportPlayback *self, const ALCchar *name); static ALCboolean ALCportPlayback_reset(ALCportPlayback *self); static ALCboolean ALCportPlayback_start(ALCportPlayback *self); static void ALCportPlayback_stop(ALCportPlayback *self); static DECLARE_FORWARD2(ALCportPlayback, ALCbackend, ALCenum, captureSamples, ALCvoid*, ALCuint) static DECLARE_FORWARD(ALCportPlayback, ALCbackend, ALCuint, availableSamples) static DECLARE_FORWARD(ALCportPlayback, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(ALCportPlayback, ALCbackend, void, lock) static DECLARE_FORWARD(ALCportPlayback, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCportPlayback) DEFINE_ALCBACKEND_VTABLE(ALCportPlayback); static void ALCportPlayback_Construct(ALCportPlayback *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCportPlayback, ALCbackend, self); self->stream = NULL; } static void ALCportPlayback_Destruct(ALCportPlayback *self) { PaError err = self->stream ? Pa_CloseStream(self->stream) : paNoError; if(err != paNoError) ERR("Error closing stream: %s\n", Pa_GetErrorText(err)); self->stream = NULL; ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static int ALCportPlayback_WriteCallback(const void *UNUSED(inputBuffer), void *outputBuffer, unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *UNUSED(timeInfo), const PaStreamCallbackFlags UNUSED(statusFlags), void *userData) { ALCportPlayback *self = userData; ALCportPlayback_lock(self); aluMixData(STATIC_CAST(ALCbackend, self)->mDevice, outputBuffer, framesPerBuffer); ALCportPlayback_unlock(self); return 0; } static ALCenum ALCportPlayback_open(ALCportPlayback *self, const ALCchar *name) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; PaError err; if(!name) name = pa_device; else if(strcmp(name, pa_device) != 0) return ALC_INVALID_VALUE; self->update_size = device->UpdateSize; self->params.device = -1; if(!ConfigValueInt(NULL, "port", "device", &self->params.device) || self->params.device < 0) self->params.device = Pa_GetDefaultOutputDevice(); self->params.suggestedLatency = (device->UpdateSize*device->NumUpdates) / (float)device->Frequency; self->params.hostApiSpecificStreamInfo = NULL; self->params.channelCount = ((device->FmtChans == DevFmtMono) ? 1 : 2); switch(device->FmtType) { case DevFmtByte: self->params.sampleFormat = paInt8; break; case DevFmtUByte: self->params.sampleFormat = paUInt8; break; case DevFmtUShort: /* fall-through */ case DevFmtShort: self->params.sampleFormat = paInt16; break; case DevFmtUInt: /* fall-through */ case DevFmtInt: self->params.sampleFormat = paInt32; break; case DevFmtFloat: self->params.sampleFormat = paFloat32; break; } retry_open: err = Pa_OpenStream(&self->stream, NULL, &self->params, device->Frequency, device->UpdateSize, paNoFlag, ALCportPlayback_WriteCallback, self ); if(err != paNoError) { if(self->params.sampleFormat == paFloat32) { self->params.sampleFormat = paInt16; goto retry_open; } ERR("Pa_OpenStream() returned an error: %s\n", Pa_GetErrorText(err)); return ALC_INVALID_VALUE; } alstr_copy_cstr(&device->DeviceName, name); return ALC_NO_ERROR; } static ALCboolean ALCportPlayback_reset(ALCportPlayback *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; const PaStreamInfo *streamInfo; streamInfo = Pa_GetStreamInfo(self->stream); device->Frequency = streamInfo->sampleRate; device->UpdateSize = self->update_size; if(self->params.sampleFormat == paInt8) device->FmtType = DevFmtByte; else if(self->params.sampleFormat == paUInt8) device->FmtType = DevFmtUByte; else if(self->params.sampleFormat == paInt16) device->FmtType = DevFmtShort; else if(self->params.sampleFormat == paInt32) device->FmtType = DevFmtInt; else if(self->params.sampleFormat == paFloat32) device->FmtType = DevFmtFloat; else { ERR("Unexpected sample format: 0x%lx\n", self->params.sampleFormat); return ALC_FALSE; } if(self->params.channelCount == 2) device->FmtChans = DevFmtStereo; else if(self->params.channelCount == 1) device->FmtChans = DevFmtMono; else { ERR("Unexpected channel count: %u\n", self->params.channelCount); return ALC_FALSE; } SetDefaultChannelOrder(device); return ALC_TRUE; } static ALCboolean ALCportPlayback_start(ALCportPlayback *self) { PaError err; err = Pa_StartStream(self->stream); if(err != paNoError) { ERR("Pa_StartStream() returned an error: %s\n", Pa_GetErrorText(err)); return ALC_FALSE; } return ALC_TRUE; } static void ALCportPlayback_stop(ALCportPlayback *self) { PaError err = Pa_StopStream(self->stream); if(err != paNoError) ERR("Error stopping stream: %s\n", Pa_GetErrorText(err)); } typedef struct ALCportCapture { DERIVE_FROM_TYPE(ALCbackend); PaStream *stream; PaStreamParameters params; ll_ringbuffer_t *ring; } ALCportCapture; static int ALCportCapture_ReadCallback(const void *inputBuffer, void *outputBuffer, unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *timeInfo, const PaStreamCallbackFlags statusFlags, void *userData); static void ALCportCapture_Construct(ALCportCapture *self, ALCdevice *device); static void ALCportCapture_Destruct(ALCportCapture *self); static ALCenum ALCportCapture_open(ALCportCapture *self, const ALCchar *name); static DECLARE_FORWARD(ALCportCapture, ALCbackend, ALCboolean, reset) static ALCboolean ALCportCapture_start(ALCportCapture *self); static void ALCportCapture_stop(ALCportCapture *self); static ALCenum ALCportCapture_captureSamples(ALCportCapture *self, ALCvoid *buffer, ALCuint samples); static ALCuint ALCportCapture_availableSamples(ALCportCapture *self); static DECLARE_FORWARD(ALCportCapture, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(ALCportCapture, ALCbackend, void, lock) static DECLARE_FORWARD(ALCportCapture, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCportCapture) DEFINE_ALCBACKEND_VTABLE(ALCportCapture); static void ALCportCapture_Construct(ALCportCapture *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCportCapture, ALCbackend, self); self->stream = NULL; self->ring = NULL; } static void ALCportCapture_Destruct(ALCportCapture *self) { PaError err = self->stream ? Pa_CloseStream(self->stream) : paNoError; if(err != paNoError) ERR("Error closing stream: %s\n", Pa_GetErrorText(err)); self->stream = NULL; ll_ringbuffer_free(self->ring); self->ring = NULL; ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static int ALCportCapture_ReadCallback(const void *inputBuffer, void *UNUSED(outputBuffer), unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *UNUSED(timeInfo), const PaStreamCallbackFlags UNUSED(statusFlags), void *userData) { ALCportCapture *self = userData; size_t writable = ll_ringbuffer_write_space(self->ring); if(framesPerBuffer > writable) framesPerBuffer = writable; ll_ringbuffer_write(self->ring, inputBuffer, framesPerBuffer); return 0; } static ALCenum ALCportCapture_open(ALCportCapture *self, const ALCchar *name) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; ALuint samples, frame_size; PaError err; if(!name) name = pa_device; else if(strcmp(name, pa_device) != 0) return ALC_INVALID_VALUE; samples = device->UpdateSize * device->NumUpdates; samples = maxu(samples, 100 * device->Frequency / 1000); frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); self->ring = ll_ringbuffer_create(samples, frame_size, false); if(self->ring == NULL) return ALC_INVALID_VALUE; self->params.device = -1; if(!ConfigValueInt(NULL, "port", "capture", &self->params.device) || self->params.device < 0) self->params.device = Pa_GetDefaultInputDevice(); self->params.suggestedLatency = 0.0f; self->params.hostApiSpecificStreamInfo = NULL; switch(device->FmtType) { case DevFmtByte: self->params.sampleFormat = paInt8; break; case DevFmtUByte: self->params.sampleFormat = paUInt8; break; case DevFmtShort: self->params.sampleFormat = paInt16; break; case DevFmtInt: self->params.sampleFormat = paInt32; break; case DevFmtFloat: self->params.sampleFormat = paFloat32; break; case DevFmtUInt: case DevFmtUShort: ERR("%s samples not supported\n", DevFmtTypeString(device->FmtType)); return ALC_INVALID_VALUE; } self->params.channelCount = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); err = Pa_OpenStream(&self->stream, &self->params, NULL, device->Frequency, paFramesPerBufferUnspecified, paNoFlag, ALCportCapture_ReadCallback, self ); if(err != paNoError) { ERR("Pa_OpenStream() returned an error: %s\n", Pa_GetErrorText(err)); return ALC_INVALID_VALUE; } alstr_copy_cstr(&device->DeviceName, name); return ALC_NO_ERROR; } static ALCboolean ALCportCapture_start(ALCportCapture *self) { PaError err = Pa_StartStream(self->stream); if(err != paNoError) { ERR("Error starting stream: %s\n", Pa_GetErrorText(err)); return ALC_FALSE; } return ALC_TRUE; } static void ALCportCapture_stop(ALCportCapture *self) { PaError err = Pa_StopStream(self->stream); if(err != paNoError) ERR("Error stopping stream: %s\n", Pa_GetErrorText(err)); } static ALCuint ALCportCapture_availableSamples(ALCportCapture *self) { return ll_ringbuffer_read_space(self->ring); } static ALCenum ALCportCapture_captureSamples(ALCportCapture *self, ALCvoid *buffer, ALCuint samples) { ll_ringbuffer_read(self->ring, buffer, samples); return ALC_NO_ERROR; } typedef struct ALCportBackendFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } ALCportBackendFactory; #define ALCPORTBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCportBackendFactory, ALCbackendFactory) } } static ALCboolean ALCportBackendFactory_init(ALCportBackendFactory *self); static void ALCportBackendFactory_deinit(ALCportBackendFactory *self); static ALCboolean ALCportBackendFactory_querySupport(ALCportBackendFactory *self, ALCbackend_Type type); static void ALCportBackendFactory_probe(ALCportBackendFactory *self, enum DevProbe type, al_string *outnames); static ALCbackend* ALCportBackendFactory_createBackend(ALCportBackendFactory *self, ALCdevice *device, ALCbackend_Type type); DEFINE_ALCBACKENDFACTORY_VTABLE(ALCportBackendFactory); static ALCboolean ALCportBackendFactory_init(ALCportBackendFactory* UNUSED(self)) { if(!pa_load()) return ALC_FALSE; return ALC_TRUE; } static void ALCportBackendFactory_deinit(ALCportBackendFactory* UNUSED(self)) { #ifdef HAVE_DYNLOAD if(pa_handle) { Pa_Terminate(); CloseLib(pa_handle); pa_handle = NULL; } #else Pa_Terminate(); #endif } static ALCboolean ALCportBackendFactory_querySupport(ALCportBackendFactory* UNUSED(self), ALCbackend_Type type) { if(type == ALCbackend_Playback || type == ALCbackend_Capture) return ALC_TRUE; return ALC_FALSE; } static void ALCportBackendFactory_probe(ALCportBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames) { switch(type) { case ALL_DEVICE_PROBE: case CAPTURE_DEVICE_PROBE: alstr_append_range(outnames, pa_device, pa_device+sizeof(pa_device)); break; } } static ALCbackend* ALCportBackendFactory_createBackend(ALCportBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) { if(type == ALCbackend_Playback) { ALCportPlayback *backend; NEW_OBJ(backend, ALCportPlayback)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } if(type == ALCbackend_Capture) { ALCportCapture *backend; NEW_OBJ(backend, ALCportCapture)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } return NULL; } ALCbackendFactory *ALCportBackendFactory_getFactory(void) { static ALCportBackendFactory factory = ALCPORTBACKENDFACTORY_INITIALIZER; return STATIC_CAST(ALCbackendFactory, &factory); } openal-soft-openal-soft-1.19.1/Alc/backends/pulseaudio.c000066400000000000000000001772461335774445300231060ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2009 by Konstantinos Natsakis * Copyright (C) 2010 by Chris Robinson * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include "alMain.h" #include "alu.h" #include "alconfig.h" #include "threads.h" #include "compat.h" #include "backends/base.h" #include #if PA_API_VERSION == 12 #ifdef HAVE_DYNLOAD static void *pa_handle; #define MAKE_FUNC(x) static __typeof(x) * p##x MAKE_FUNC(pa_context_unref); MAKE_FUNC(pa_sample_spec_valid); MAKE_FUNC(pa_frame_size); MAKE_FUNC(pa_stream_drop); MAKE_FUNC(pa_strerror); MAKE_FUNC(pa_context_get_state); MAKE_FUNC(pa_stream_get_state); MAKE_FUNC(pa_threaded_mainloop_signal); MAKE_FUNC(pa_stream_peek); MAKE_FUNC(pa_threaded_mainloop_wait); MAKE_FUNC(pa_threaded_mainloop_unlock); MAKE_FUNC(pa_threaded_mainloop_in_thread); MAKE_FUNC(pa_context_new); MAKE_FUNC(pa_threaded_mainloop_stop); MAKE_FUNC(pa_context_disconnect); MAKE_FUNC(pa_threaded_mainloop_start); MAKE_FUNC(pa_threaded_mainloop_get_api); MAKE_FUNC(pa_context_set_state_callback); MAKE_FUNC(pa_stream_write); MAKE_FUNC(pa_xfree); MAKE_FUNC(pa_stream_connect_record); MAKE_FUNC(pa_stream_connect_playback); MAKE_FUNC(pa_stream_readable_size); MAKE_FUNC(pa_stream_writable_size); MAKE_FUNC(pa_stream_is_corked); MAKE_FUNC(pa_stream_cork); MAKE_FUNC(pa_stream_is_suspended); MAKE_FUNC(pa_stream_get_device_name); MAKE_FUNC(pa_stream_get_latency); MAKE_FUNC(pa_path_get_filename); MAKE_FUNC(pa_get_binary_name); MAKE_FUNC(pa_threaded_mainloop_free); MAKE_FUNC(pa_context_errno); MAKE_FUNC(pa_xmalloc); MAKE_FUNC(pa_stream_unref); MAKE_FUNC(pa_threaded_mainloop_accept); MAKE_FUNC(pa_stream_set_write_callback); MAKE_FUNC(pa_threaded_mainloop_new); MAKE_FUNC(pa_context_connect); MAKE_FUNC(pa_stream_set_buffer_attr); MAKE_FUNC(pa_stream_get_buffer_attr); MAKE_FUNC(pa_stream_get_sample_spec); MAKE_FUNC(pa_stream_get_time); MAKE_FUNC(pa_stream_set_read_callback); MAKE_FUNC(pa_stream_set_state_callback); MAKE_FUNC(pa_stream_set_moved_callback); MAKE_FUNC(pa_stream_set_underflow_callback); MAKE_FUNC(pa_stream_new_with_proplist); MAKE_FUNC(pa_stream_disconnect); MAKE_FUNC(pa_threaded_mainloop_lock); MAKE_FUNC(pa_channel_map_init_auto); MAKE_FUNC(pa_channel_map_parse); MAKE_FUNC(pa_channel_map_snprint); MAKE_FUNC(pa_channel_map_equal); MAKE_FUNC(pa_context_get_server_info); MAKE_FUNC(pa_context_get_sink_info_by_name); MAKE_FUNC(pa_context_get_sink_info_list); MAKE_FUNC(pa_context_get_source_info_by_name); MAKE_FUNC(pa_context_get_source_info_list); MAKE_FUNC(pa_operation_get_state); MAKE_FUNC(pa_operation_unref); MAKE_FUNC(pa_proplist_new); MAKE_FUNC(pa_proplist_free); MAKE_FUNC(pa_proplist_set); MAKE_FUNC(pa_channel_map_superset); MAKE_FUNC(pa_stream_set_buffer_attr_callback); MAKE_FUNC(pa_stream_begin_write); #undef MAKE_FUNC #define pa_context_unref ppa_context_unref #define pa_sample_spec_valid ppa_sample_spec_valid #define pa_frame_size ppa_frame_size #define pa_stream_drop ppa_stream_drop #define pa_strerror ppa_strerror #define pa_context_get_state ppa_context_get_state #define pa_stream_get_state ppa_stream_get_state #define pa_threaded_mainloop_signal ppa_threaded_mainloop_signal #define pa_stream_peek ppa_stream_peek #define pa_threaded_mainloop_wait ppa_threaded_mainloop_wait #define pa_threaded_mainloop_unlock ppa_threaded_mainloop_unlock #define pa_threaded_mainloop_in_thread ppa_threaded_mainloop_in_thread #define pa_context_new ppa_context_new #define pa_threaded_mainloop_stop ppa_threaded_mainloop_stop #define pa_context_disconnect ppa_context_disconnect #define pa_threaded_mainloop_start ppa_threaded_mainloop_start #define pa_threaded_mainloop_get_api ppa_threaded_mainloop_get_api #define pa_context_set_state_callback ppa_context_set_state_callback #define pa_stream_write ppa_stream_write #define pa_xfree ppa_xfree #define pa_stream_connect_record ppa_stream_connect_record #define pa_stream_connect_playback ppa_stream_connect_playback #define pa_stream_readable_size ppa_stream_readable_size #define pa_stream_writable_size ppa_stream_writable_size #define pa_stream_is_corked ppa_stream_is_corked #define pa_stream_cork ppa_stream_cork #define pa_stream_is_suspended ppa_stream_is_suspended #define pa_stream_get_device_name ppa_stream_get_device_name #define pa_stream_get_latency ppa_stream_get_latency #define pa_path_get_filename ppa_path_get_filename #define pa_get_binary_name ppa_get_binary_name #define pa_threaded_mainloop_free ppa_threaded_mainloop_free #define pa_context_errno ppa_context_errno #define pa_xmalloc ppa_xmalloc #define pa_stream_unref ppa_stream_unref #define pa_threaded_mainloop_accept ppa_threaded_mainloop_accept #define pa_stream_set_write_callback ppa_stream_set_write_callback #define pa_threaded_mainloop_new ppa_threaded_mainloop_new #define pa_context_connect ppa_context_connect #define pa_stream_set_buffer_attr ppa_stream_set_buffer_attr #define pa_stream_get_buffer_attr ppa_stream_get_buffer_attr #define pa_stream_get_sample_spec ppa_stream_get_sample_spec #define pa_stream_get_time ppa_stream_get_time #define pa_stream_set_read_callback ppa_stream_set_read_callback #define pa_stream_set_state_callback ppa_stream_set_state_callback #define pa_stream_set_moved_callback ppa_stream_set_moved_callback #define pa_stream_set_underflow_callback ppa_stream_set_underflow_callback #define pa_stream_new_with_proplist ppa_stream_new_with_proplist #define pa_stream_disconnect ppa_stream_disconnect #define pa_threaded_mainloop_lock ppa_threaded_mainloop_lock #define pa_channel_map_init_auto ppa_channel_map_init_auto #define pa_channel_map_parse ppa_channel_map_parse #define pa_channel_map_snprint ppa_channel_map_snprint #define pa_channel_map_equal ppa_channel_map_equal #define pa_context_get_server_info ppa_context_get_server_info #define pa_context_get_sink_info_by_name ppa_context_get_sink_info_by_name #define pa_context_get_sink_info_list ppa_context_get_sink_info_list #define pa_context_get_source_info_by_name ppa_context_get_source_info_by_name #define pa_context_get_source_info_list ppa_context_get_source_info_list #define pa_operation_get_state ppa_operation_get_state #define pa_operation_unref ppa_operation_unref #define pa_proplist_new ppa_proplist_new #define pa_proplist_free ppa_proplist_free #define pa_proplist_set ppa_proplist_set #define pa_channel_map_superset ppa_channel_map_superset #define pa_stream_set_buffer_attr_callback ppa_stream_set_buffer_attr_callback #define pa_stream_begin_write ppa_stream_begin_write #endif static ALCboolean pulse_load(void) { ALCboolean ret = ALC_TRUE; #ifdef HAVE_DYNLOAD if(!pa_handle) { al_string missing_funcs = AL_STRING_INIT_STATIC(); #ifdef _WIN32 #define PALIB "libpulse-0.dll" #elif defined(__APPLE__) && defined(__MACH__) #define PALIB "libpulse.0.dylib" #else #define PALIB "libpulse.so.0" #endif pa_handle = LoadLib(PALIB); if(!pa_handle) { WARN("Failed to load %s\n", PALIB); return ALC_FALSE; } #define LOAD_FUNC(x) do { \ p##x = GetSymbol(pa_handle, #x); \ if(!(p##x)) { \ ret = ALC_FALSE; \ alstr_append_cstr(&missing_funcs, "\n" #x); \ } \ } while(0) LOAD_FUNC(pa_context_unref); LOAD_FUNC(pa_sample_spec_valid); LOAD_FUNC(pa_stream_drop); LOAD_FUNC(pa_frame_size); LOAD_FUNC(pa_strerror); LOAD_FUNC(pa_context_get_state); LOAD_FUNC(pa_stream_get_state); LOAD_FUNC(pa_threaded_mainloop_signal); LOAD_FUNC(pa_stream_peek); LOAD_FUNC(pa_threaded_mainloop_wait); LOAD_FUNC(pa_threaded_mainloop_unlock); LOAD_FUNC(pa_threaded_mainloop_in_thread); LOAD_FUNC(pa_context_new); LOAD_FUNC(pa_threaded_mainloop_stop); LOAD_FUNC(pa_context_disconnect); LOAD_FUNC(pa_threaded_mainloop_start); LOAD_FUNC(pa_threaded_mainloop_get_api); LOAD_FUNC(pa_context_set_state_callback); LOAD_FUNC(pa_stream_write); LOAD_FUNC(pa_xfree); LOAD_FUNC(pa_stream_connect_record); LOAD_FUNC(pa_stream_connect_playback); LOAD_FUNC(pa_stream_readable_size); LOAD_FUNC(pa_stream_writable_size); LOAD_FUNC(pa_stream_is_corked); LOAD_FUNC(pa_stream_cork); LOAD_FUNC(pa_stream_is_suspended); LOAD_FUNC(pa_stream_get_device_name); LOAD_FUNC(pa_stream_get_latency); LOAD_FUNC(pa_path_get_filename); LOAD_FUNC(pa_get_binary_name); LOAD_FUNC(pa_threaded_mainloop_free); LOAD_FUNC(pa_context_errno); LOAD_FUNC(pa_xmalloc); LOAD_FUNC(pa_stream_unref); LOAD_FUNC(pa_threaded_mainloop_accept); LOAD_FUNC(pa_stream_set_write_callback); LOAD_FUNC(pa_threaded_mainloop_new); LOAD_FUNC(pa_context_connect); LOAD_FUNC(pa_stream_set_buffer_attr); LOAD_FUNC(pa_stream_get_buffer_attr); LOAD_FUNC(pa_stream_get_sample_spec); LOAD_FUNC(pa_stream_get_time); LOAD_FUNC(pa_stream_set_read_callback); LOAD_FUNC(pa_stream_set_state_callback); LOAD_FUNC(pa_stream_set_moved_callback); LOAD_FUNC(pa_stream_set_underflow_callback); LOAD_FUNC(pa_stream_new_with_proplist); LOAD_FUNC(pa_stream_disconnect); LOAD_FUNC(pa_threaded_mainloop_lock); LOAD_FUNC(pa_channel_map_init_auto); LOAD_FUNC(pa_channel_map_parse); LOAD_FUNC(pa_channel_map_snprint); LOAD_FUNC(pa_channel_map_equal); LOAD_FUNC(pa_context_get_server_info); LOAD_FUNC(pa_context_get_sink_info_by_name); LOAD_FUNC(pa_context_get_sink_info_list); LOAD_FUNC(pa_context_get_source_info_by_name); LOAD_FUNC(pa_context_get_source_info_list); LOAD_FUNC(pa_operation_get_state); LOAD_FUNC(pa_operation_unref); LOAD_FUNC(pa_proplist_new); LOAD_FUNC(pa_proplist_free); LOAD_FUNC(pa_proplist_set); LOAD_FUNC(pa_channel_map_superset); LOAD_FUNC(pa_stream_set_buffer_attr_callback); LOAD_FUNC(pa_stream_begin_write); #undef LOAD_FUNC if(ret == ALC_FALSE) { WARN("Missing expected functions:%s\n", alstr_get_cstr(missing_funcs)); CloseLib(pa_handle); pa_handle = NULL; } alstr_reset(&missing_funcs); } #endif /* HAVE_DYNLOAD */ return ret; } /* Global flags and properties */ static pa_context_flags_t pulse_ctx_flags; static pa_proplist *prop_filter; /* PulseAudio Event Callbacks */ static void context_state_callback(pa_context *context, void *pdata) { pa_threaded_mainloop *loop = pdata; pa_context_state_t state; state = pa_context_get_state(context); if(state == PA_CONTEXT_READY || !PA_CONTEXT_IS_GOOD(state)) pa_threaded_mainloop_signal(loop, 0); } static void stream_state_callback(pa_stream *stream, void *pdata) { pa_threaded_mainloop *loop = pdata; pa_stream_state_t state; state = pa_stream_get_state(stream); if(state == PA_STREAM_READY || !PA_STREAM_IS_GOOD(state)) pa_threaded_mainloop_signal(loop, 0); } static void stream_success_callback(pa_stream *UNUSED(stream), int UNUSED(success), void *pdata) { pa_threaded_mainloop *loop = pdata; pa_threaded_mainloop_signal(loop, 0); } static void wait_for_operation(pa_operation *op, pa_threaded_mainloop *loop) { if(op) { while(pa_operation_get_state(op) == PA_OPERATION_RUNNING) pa_threaded_mainloop_wait(loop); pa_operation_unref(op); } } static pa_context *connect_context(pa_threaded_mainloop *loop, ALboolean silent) { const char *name = "OpenAL Soft"; al_string binname = AL_STRING_INIT_STATIC(); pa_context_state_t state; pa_context *context; int err; GetProcBinary(NULL, &binname); if(!alstr_empty(binname)) name = alstr_get_cstr(binname); context = pa_context_new(pa_threaded_mainloop_get_api(loop), name); if(!context) { ERR("pa_context_new() failed\n"); alstr_reset(&binname); return NULL; } pa_context_set_state_callback(context, context_state_callback, loop); if((err=pa_context_connect(context, NULL, pulse_ctx_flags, NULL)) >= 0) { while((state=pa_context_get_state(context)) != PA_CONTEXT_READY) { if(!PA_CONTEXT_IS_GOOD(state)) { err = pa_context_errno(context); if(err > 0) err = -err; break; } pa_threaded_mainloop_wait(loop); } } pa_context_set_state_callback(context, NULL, NULL); if(err < 0) { if(!silent) ERR("Context did not connect: %s\n", pa_strerror(err)); pa_context_unref(context); context = NULL; } alstr_reset(&binname); return context; } static ALCboolean pulse_open(pa_threaded_mainloop **loop, pa_context **context, void(*state_cb)(pa_context*,void*), void *ptr) { if(!(*loop = pa_threaded_mainloop_new())) { ERR("pa_threaded_mainloop_new() failed!\n"); return ALC_FALSE; } if(pa_threaded_mainloop_start(*loop) < 0) { ERR("pa_threaded_mainloop_start() failed\n"); goto error; } pa_threaded_mainloop_lock(*loop); *context = connect_context(*loop, AL_FALSE); if(!*context) { pa_threaded_mainloop_unlock(*loop); pa_threaded_mainloop_stop(*loop); goto error; } pa_context_set_state_callback(*context, state_cb, ptr); pa_threaded_mainloop_unlock(*loop); return ALC_TRUE; error: pa_threaded_mainloop_free(*loop); *loop = NULL; return ALC_FALSE; } static void pulse_close(pa_threaded_mainloop *loop, pa_context *context, pa_stream *stream) { pa_threaded_mainloop_lock(loop); if(stream) { pa_stream_set_state_callback(stream, NULL, NULL); pa_stream_set_moved_callback(stream, NULL, NULL); pa_stream_set_write_callback(stream, NULL, NULL); pa_stream_set_buffer_attr_callback(stream, NULL, NULL); pa_stream_disconnect(stream); pa_stream_unref(stream); } pa_context_disconnect(context); pa_context_unref(context); pa_threaded_mainloop_unlock(loop); pa_threaded_mainloop_stop(loop); pa_threaded_mainloop_free(loop); } typedef struct { al_string name; al_string device_name; } DevMap; TYPEDEF_VECTOR(DevMap, vector_DevMap) static vector_DevMap PlaybackDevices; static vector_DevMap CaptureDevices; static void clear_devlist(vector_DevMap *list) { #define DEINIT_STRS(i) (AL_STRING_DEINIT((i)->name),AL_STRING_DEINIT((i)->device_name)) VECTOR_FOR_EACH(DevMap, *list, DEINIT_STRS); #undef DEINIT_STRS VECTOR_RESIZE(*list, 0, 0); } typedef struct ALCpulsePlayback { DERIVE_FROM_TYPE(ALCbackend); al_string device_name; pa_buffer_attr attr; pa_sample_spec spec; pa_threaded_mainloop *loop; pa_stream *stream; pa_context *context; ATOMIC(ALenum) killNow; althrd_t thread; } ALCpulsePlayback; static void ALCpulsePlayback_deviceCallback(pa_context *context, const pa_sink_info *info, int eol, void *pdata); static void ALCpulsePlayback_probeDevices(void); static void ALCpulsePlayback_bufferAttrCallback(pa_stream *stream, void *pdata); static void ALCpulsePlayback_contextStateCallback(pa_context *context, void *pdata); static void ALCpulsePlayback_streamStateCallback(pa_stream *stream, void *pdata); static void ALCpulsePlayback_streamWriteCallback(pa_stream *p, size_t nbytes, void *userdata); static void ALCpulsePlayback_sinkInfoCallback(pa_context *context, const pa_sink_info *info, int eol, void *pdata); static void ALCpulsePlayback_sinkNameCallback(pa_context *context, const pa_sink_info *info, int eol, void *pdata); static void ALCpulsePlayback_streamMovedCallback(pa_stream *stream, void *pdata); static pa_stream *ALCpulsePlayback_connectStream(const char *device_name, pa_threaded_mainloop *loop, pa_context *context, pa_stream_flags_t flags, pa_buffer_attr *attr, pa_sample_spec *spec, pa_channel_map *chanmap); static int ALCpulsePlayback_mixerProc(void *ptr); static void ALCpulsePlayback_Construct(ALCpulsePlayback *self, ALCdevice *device); static void ALCpulsePlayback_Destruct(ALCpulsePlayback *self); static ALCenum ALCpulsePlayback_open(ALCpulsePlayback *self, const ALCchar *name); static ALCboolean ALCpulsePlayback_reset(ALCpulsePlayback *self); static ALCboolean ALCpulsePlayback_start(ALCpulsePlayback *self); static void ALCpulsePlayback_stop(ALCpulsePlayback *self); static DECLARE_FORWARD2(ALCpulsePlayback, ALCbackend, ALCenum, captureSamples, ALCvoid*, ALCuint) static DECLARE_FORWARD(ALCpulsePlayback, ALCbackend, ALCuint, availableSamples) static ClockLatency ALCpulsePlayback_getClockLatency(ALCpulsePlayback *self); static void ALCpulsePlayback_lock(ALCpulsePlayback *self); static void ALCpulsePlayback_unlock(ALCpulsePlayback *self); DECLARE_DEFAULT_ALLOCATORS(ALCpulsePlayback) DEFINE_ALCBACKEND_VTABLE(ALCpulsePlayback); static void ALCpulsePlayback_Construct(ALCpulsePlayback *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCpulsePlayback, ALCbackend, self); self->loop = NULL; AL_STRING_INIT(self->device_name); ATOMIC_INIT(&self->killNow, AL_TRUE); } static void ALCpulsePlayback_Destruct(ALCpulsePlayback *self) { if(self->loop) { pulse_close(self->loop, self->context, self->stream); self->loop = NULL; self->context = NULL; self->stream = NULL; } AL_STRING_DEINIT(self->device_name); ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static void ALCpulsePlayback_deviceCallback(pa_context *UNUSED(context), const pa_sink_info *info, int eol, void *pdata) { pa_threaded_mainloop *loop = pdata; const DevMap *iter; DevMap entry; int count; if(eol) { pa_threaded_mainloop_signal(loop, 0); return; } #define MATCH_INFO_NAME(iter) (alstr_cmp_cstr((iter)->device_name, info->name) == 0) VECTOR_FIND_IF(iter, const DevMap, PlaybackDevices, MATCH_INFO_NAME); if(iter != VECTOR_END(PlaybackDevices)) return; #undef MATCH_INFO_NAME AL_STRING_INIT(entry.name); AL_STRING_INIT(entry.device_name); alstr_copy_cstr(&entry.device_name, info->name); count = 0; while(1) { alstr_copy_cstr(&entry.name, info->description); if(count != 0) { char str[64]; snprintf(str, sizeof(str), " #%d", count+1); alstr_append_cstr(&entry.name, str); } #define MATCH_ENTRY(i) (alstr_cmp(entry.name, (i)->name) == 0) VECTOR_FIND_IF(iter, const DevMap, PlaybackDevices, MATCH_ENTRY); if(iter == VECTOR_END(PlaybackDevices)) break; #undef MATCH_ENTRY count++; } TRACE("Got device \"%s\", \"%s\"\n", alstr_get_cstr(entry.name), alstr_get_cstr(entry.device_name)); VECTOR_PUSH_BACK(PlaybackDevices, entry); } static void ALCpulsePlayback_probeDevices(void) { pa_threaded_mainloop *loop; clear_devlist(&PlaybackDevices); if((loop=pa_threaded_mainloop_new()) && pa_threaded_mainloop_start(loop) >= 0) { pa_context *context; pa_threaded_mainloop_lock(loop); context = connect_context(loop, AL_FALSE); if(context) { pa_operation *o; pa_stream_flags_t flags; pa_sample_spec spec; pa_stream *stream; flags = PA_STREAM_FIX_FORMAT | PA_STREAM_FIX_RATE | PA_STREAM_FIX_CHANNELS | PA_STREAM_DONT_MOVE; spec.format = PA_SAMPLE_S16NE; spec.rate = 44100; spec.channels = 2; stream = ALCpulsePlayback_connectStream(NULL, loop, context, flags, NULL, &spec, NULL); if(stream) { o = pa_context_get_sink_info_by_name(context, pa_stream_get_device_name(stream), ALCpulsePlayback_deviceCallback, loop); wait_for_operation(o, loop); pa_stream_disconnect(stream); pa_stream_unref(stream); stream = NULL; } o = pa_context_get_sink_info_list(context, ALCpulsePlayback_deviceCallback, loop); wait_for_operation(o, loop); pa_context_disconnect(context); pa_context_unref(context); } pa_threaded_mainloop_unlock(loop); pa_threaded_mainloop_stop(loop); } if(loop) pa_threaded_mainloop_free(loop); } static void ALCpulsePlayback_bufferAttrCallback(pa_stream *stream, void *pdata) { ALCpulsePlayback *self = pdata; self->attr = *pa_stream_get_buffer_attr(stream); TRACE("minreq=%d, tlength=%d, prebuf=%d\n", self->attr.minreq, self->attr.tlength, self->attr.prebuf); /* FIXME: Update the device's UpdateSize (and/or NumUpdates) using the new * buffer attributes? Changing UpdateSize will change the ALC_REFRESH * property, which probably shouldn't change between device resets. But * leaving it alone means ALC_REFRESH will be off. */ } static void ALCpulsePlayback_contextStateCallback(pa_context *context, void *pdata) { ALCpulsePlayback *self = pdata; if(pa_context_get_state(context) == PA_CONTEXT_FAILED) { ERR("Received context failure!\n"); aluHandleDisconnect(STATIC_CAST(ALCbackend,self)->mDevice, "Playback state failure"); } pa_threaded_mainloop_signal(self->loop, 0); } static void ALCpulsePlayback_streamStateCallback(pa_stream *stream, void *pdata) { ALCpulsePlayback *self = pdata; if(pa_stream_get_state(stream) == PA_STREAM_FAILED) { ERR("Received stream failure!\n"); aluHandleDisconnect(STATIC_CAST(ALCbackend,self)->mDevice, "Playback stream failure"); } pa_threaded_mainloop_signal(self->loop, 0); } static void ALCpulsePlayback_streamWriteCallback(pa_stream* UNUSED(p), size_t UNUSED(nbytes), void *pdata) { ALCpulsePlayback *self = pdata; pa_threaded_mainloop_signal(self->loop, 0); } static void ALCpulsePlayback_sinkInfoCallback(pa_context *UNUSED(context), const pa_sink_info *info, int eol, void *pdata) { static const struct { enum DevFmtChannels chans; pa_channel_map map; } chanmaps[] = { { DevFmtX71, { 8, { PA_CHANNEL_POSITION_FRONT_LEFT, PA_CHANNEL_POSITION_FRONT_RIGHT, PA_CHANNEL_POSITION_FRONT_CENTER, PA_CHANNEL_POSITION_LFE, PA_CHANNEL_POSITION_REAR_LEFT, PA_CHANNEL_POSITION_REAR_RIGHT, PA_CHANNEL_POSITION_SIDE_LEFT, PA_CHANNEL_POSITION_SIDE_RIGHT } } }, { DevFmtX61, { 7, { PA_CHANNEL_POSITION_FRONT_LEFT, PA_CHANNEL_POSITION_FRONT_RIGHT, PA_CHANNEL_POSITION_FRONT_CENTER, PA_CHANNEL_POSITION_LFE, PA_CHANNEL_POSITION_REAR_CENTER, PA_CHANNEL_POSITION_SIDE_LEFT, PA_CHANNEL_POSITION_SIDE_RIGHT } } }, { DevFmtX51, { 6, { PA_CHANNEL_POSITION_FRONT_LEFT, PA_CHANNEL_POSITION_FRONT_RIGHT, PA_CHANNEL_POSITION_FRONT_CENTER, PA_CHANNEL_POSITION_LFE, PA_CHANNEL_POSITION_SIDE_LEFT, PA_CHANNEL_POSITION_SIDE_RIGHT } } }, { DevFmtX51Rear, { 6, { PA_CHANNEL_POSITION_FRONT_LEFT, PA_CHANNEL_POSITION_FRONT_RIGHT, PA_CHANNEL_POSITION_FRONT_CENTER, PA_CHANNEL_POSITION_LFE, PA_CHANNEL_POSITION_REAR_LEFT, PA_CHANNEL_POSITION_REAR_RIGHT } } }, { DevFmtQuad, { 4, { PA_CHANNEL_POSITION_FRONT_LEFT, PA_CHANNEL_POSITION_FRONT_RIGHT, PA_CHANNEL_POSITION_REAR_LEFT, PA_CHANNEL_POSITION_REAR_RIGHT } } }, { DevFmtStereo, { 2, { PA_CHANNEL_POSITION_FRONT_LEFT, PA_CHANNEL_POSITION_FRONT_RIGHT } } }, { DevFmtMono, { 1, {PA_CHANNEL_POSITION_MONO} } } }; ALCpulsePlayback *self = pdata; ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; size_t i; if(eol) { pa_threaded_mainloop_signal(self->loop, 0); return; } for(i = 0;i < COUNTOF(chanmaps);i++) { if(pa_channel_map_superset(&info->channel_map, &chanmaps[i].map)) { if(!(device->Flags&DEVICE_CHANNELS_REQUEST)) device->FmtChans = chanmaps[i].chans; break; } } if(i == COUNTOF(chanmaps)) { char chanmap_str[PA_CHANNEL_MAP_SNPRINT_MAX] = ""; pa_channel_map_snprint(chanmap_str, sizeof(chanmap_str), &info->channel_map); WARN("Failed to find format for channel map:\n %s\n", chanmap_str); } if(info->active_port) TRACE("Active port: %s (%s)\n", info->active_port->name, info->active_port->description); device->IsHeadphones = (info->active_port && strcmp(info->active_port->name, "analog-output-headphones") == 0 && device->FmtChans == DevFmtStereo); } static void ALCpulsePlayback_sinkNameCallback(pa_context *UNUSED(context), const pa_sink_info *info, int eol, void *pdata) { ALCpulsePlayback *self = pdata; ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; if(eol) { pa_threaded_mainloop_signal(self->loop, 0); return; } alstr_copy_cstr(&device->DeviceName, info->description); } static void ALCpulsePlayback_streamMovedCallback(pa_stream *stream, void *pdata) { ALCpulsePlayback *self = pdata; alstr_copy_cstr(&self->device_name, pa_stream_get_device_name(stream)); TRACE("Stream moved to %s\n", alstr_get_cstr(self->device_name)); } static pa_stream *ALCpulsePlayback_connectStream(const char *device_name, pa_threaded_mainloop *loop, pa_context *context, pa_stream_flags_t flags, pa_buffer_attr *attr, pa_sample_spec *spec, pa_channel_map *chanmap) { pa_stream_state_t state; pa_stream *stream; if(!device_name) { device_name = getenv("ALSOFT_PULSE_DEFAULT"); if(device_name && !device_name[0]) device_name = NULL; } stream = pa_stream_new_with_proplist(context, "Playback Stream", spec, chanmap, prop_filter); if(!stream) { ERR("pa_stream_new_with_proplist() failed: %s\n", pa_strerror(pa_context_errno(context))); return NULL; } pa_stream_set_state_callback(stream, stream_state_callback, loop); if(pa_stream_connect_playback(stream, device_name, attr, flags, NULL, NULL) < 0) { ERR("Stream did not connect: %s\n", pa_strerror(pa_context_errno(context))); pa_stream_unref(stream); return NULL; } while((state=pa_stream_get_state(stream)) != PA_STREAM_READY) { if(!PA_STREAM_IS_GOOD(state)) { ERR("Stream did not get ready: %s\n", pa_strerror(pa_context_errno(context))); pa_stream_unref(stream); return NULL; } pa_threaded_mainloop_wait(loop); } pa_stream_set_state_callback(stream, NULL, NULL); return stream; } static int ALCpulsePlayback_mixerProc(void *ptr) { ALCpulsePlayback *self = ptr; ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; ALuint buffer_size; size_t frame_size; ssize_t len; SetRTPriority(); althrd_setname(althrd_current(), MIXER_THREAD_NAME); pa_threaded_mainloop_lock(self->loop); frame_size = pa_frame_size(&self->spec); while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) && ATOMIC_LOAD(&device->Connected, almemory_order_acquire)) { void *buf; int ret; len = pa_stream_writable_size(self->stream); if(len < 0) { ERR("Failed to get writable size: %ld", (long)len); aluHandleDisconnect(device, "Failed to get writable size: %ld", (long)len); break; } /* Make sure we're going to write at least 2 'periods' (minreqs), in * case the server increased it since starting playback. Also round up * the number of writable periods if it's not an integer count. */ buffer_size = maxu((self->attr.tlength + self->attr.minreq/2) / self->attr.minreq, 2) * self->attr.minreq; /* NOTE: This assumes pa_stream_writable_size returns between 0 and * tlength, else there will be more latency than intended. */ len = mini(len - (ssize_t)self->attr.tlength, 0) + buffer_size; if(len < (int32_t)self->attr.minreq) { if(pa_stream_is_corked(self->stream)) { pa_operation *o; o = pa_stream_cork(self->stream, 0, NULL, NULL); if(o) pa_operation_unref(o); } pa_threaded_mainloop_wait(self->loop); continue; } len -= len%self->attr.minreq; len -= len%frame_size; buf = pa_xmalloc(len); aluMixData(device, buf, len/frame_size); ret = pa_stream_write(self->stream, buf, len, pa_xfree, 0, PA_SEEK_RELATIVE); if(ret != PA_OK) ERR("Failed to write to stream: %d, %s\n", ret, pa_strerror(ret)); } pa_threaded_mainloop_unlock(self->loop); return 0; } static ALCenum ALCpulsePlayback_open(ALCpulsePlayback *self, const ALCchar *name) { const_al_string dev_name = AL_STRING_INIT_STATIC(); const char *pulse_name = NULL; pa_stream_flags_t flags; pa_sample_spec spec; if(name) { const DevMap *iter; if(VECTOR_SIZE(PlaybackDevices) == 0) ALCpulsePlayback_probeDevices(); #define MATCH_NAME(iter) (alstr_cmp_cstr((iter)->name, name) == 0) VECTOR_FIND_IF(iter, const DevMap, PlaybackDevices, MATCH_NAME); #undef MATCH_NAME if(iter == VECTOR_END(PlaybackDevices)) return ALC_INVALID_VALUE; pulse_name = alstr_get_cstr(iter->device_name); dev_name = iter->name; } if(!pulse_open(&self->loop, &self->context, ALCpulsePlayback_contextStateCallback, self)) return ALC_INVALID_VALUE; pa_threaded_mainloop_lock(self->loop); flags = PA_STREAM_FIX_FORMAT | PA_STREAM_FIX_RATE | PA_STREAM_FIX_CHANNELS; if(!GetConfigValueBool(NULL, "pulse", "allow-moves", 0)) flags |= PA_STREAM_DONT_MOVE; spec.format = PA_SAMPLE_S16NE; spec.rate = 44100; spec.channels = 2; TRACE("Connecting to \"%s\"\n", pulse_name ? pulse_name : "(default)"); self->stream = ALCpulsePlayback_connectStream(pulse_name, self->loop, self->context, flags, NULL, &spec, NULL); if(!self->stream) { pa_threaded_mainloop_unlock(self->loop); pulse_close(self->loop, self->context, self->stream); self->loop = NULL; self->context = NULL; return ALC_INVALID_VALUE; } pa_stream_set_moved_callback(self->stream, ALCpulsePlayback_streamMovedCallback, self); alstr_copy_cstr(&self->device_name, pa_stream_get_device_name(self->stream)); if(alstr_empty(dev_name)) { pa_operation *o = pa_context_get_sink_info_by_name( self->context, alstr_get_cstr(self->device_name), ALCpulsePlayback_sinkNameCallback, self ); wait_for_operation(o, self->loop); } else { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; alstr_copy(&device->DeviceName, dev_name); } pa_threaded_mainloop_unlock(self->loop); return ALC_NO_ERROR; } static ALCboolean ALCpulsePlayback_reset(ALCpulsePlayback *self) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; pa_stream_flags_t flags = 0; const char *mapname = NULL; pa_channel_map chanmap; pa_operation *o; pa_threaded_mainloop_lock(self->loop); if(self->stream) { pa_stream_set_state_callback(self->stream, NULL, NULL); pa_stream_set_moved_callback(self->stream, NULL, NULL); pa_stream_set_write_callback(self->stream, NULL, NULL); pa_stream_set_buffer_attr_callback(self->stream, NULL, NULL); pa_stream_disconnect(self->stream); pa_stream_unref(self->stream); self->stream = NULL; } o = pa_context_get_sink_info_by_name(self->context, alstr_get_cstr(self->device_name), ALCpulsePlayback_sinkInfoCallback, self); wait_for_operation(o, self->loop); if(GetConfigValueBool(alstr_get_cstr(device->DeviceName), "pulse", "fix-rate", 0) || !(device->Flags&DEVICE_FREQUENCY_REQUEST)) flags |= PA_STREAM_FIX_RATE; flags |= PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE; flags |= PA_STREAM_ADJUST_LATENCY; flags |= PA_STREAM_START_CORKED; if(!GetConfigValueBool(NULL, "pulse", "allow-moves", 0)) flags |= PA_STREAM_DONT_MOVE; switch(device->FmtType) { case DevFmtByte: device->FmtType = DevFmtUByte; /* fall-through */ case DevFmtUByte: self->spec.format = PA_SAMPLE_U8; break; case DevFmtUShort: device->FmtType = DevFmtShort; /* fall-through */ case DevFmtShort: self->spec.format = PA_SAMPLE_S16NE; break; case DevFmtUInt: device->FmtType = DevFmtInt; /* fall-through */ case DevFmtInt: self->spec.format = PA_SAMPLE_S32NE; break; case DevFmtFloat: self->spec.format = PA_SAMPLE_FLOAT32NE; break; } self->spec.rate = device->Frequency; self->spec.channels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); if(pa_sample_spec_valid(&self->spec) == 0) { ERR("Invalid sample format\n"); pa_threaded_mainloop_unlock(self->loop); return ALC_FALSE; } switch(device->FmtChans) { case DevFmtMono: mapname = "mono"; break; case DevFmtAmbi3D: device->FmtChans = DevFmtStereo; /*fall-through*/ case DevFmtStereo: mapname = "front-left,front-right"; break; case DevFmtQuad: mapname = "front-left,front-right,rear-left,rear-right"; break; case DevFmtX51: mapname = "front-left,front-right,front-center,lfe,side-left,side-right"; break; case DevFmtX51Rear: mapname = "front-left,front-right,front-center,lfe,rear-left,rear-right"; break; case DevFmtX61: mapname = "front-left,front-right,front-center,lfe,rear-center,side-left,side-right"; break; case DevFmtX71: mapname = "front-left,front-right,front-center,lfe,rear-left,rear-right,side-left,side-right"; break; } if(!pa_channel_map_parse(&chanmap, mapname)) { ERR("Failed to build channel map for %s\n", DevFmtChannelsString(device->FmtChans)); pa_threaded_mainloop_unlock(self->loop); return ALC_FALSE; } SetDefaultWFXChannelOrder(device); self->attr.fragsize = -1; self->attr.prebuf = 0; self->attr.minreq = device->UpdateSize * pa_frame_size(&self->spec); self->attr.tlength = self->attr.minreq * maxu(device->NumUpdates, 2); self->attr.maxlength = -1; self->stream = ALCpulsePlayback_connectStream(alstr_get_cstr(self->device_name), self->loop, self->context, flags, &self->attr, &self->spec, &chanmap ); if(!self->stream) { pa_threaded_mainloop_unlock(self->loop); return ALC_FALSE; } pa_stream_set_state_callback(self->stream, ALCpulsePlayback_streamStateCallback, self); pa_stream_set_moved_callback(self->stream, ALCpulsePlayback_streamMovedCallback, self); pa_stream_set_write_callback(self->stream, ALCpulsePlayback_streamWriteCallback, self); self->spec = *(pa_stream_get_sample_spec(self->stream)); if(device->Frequency != self->spec.rate) { /* Server updated our playback rate, so modify the buffer attribs * accordingly. */ device->NumUpdates = (ALuint)clampd( (ALdouble)device->NumUpdates/device->Frequency*self->spec.rate + 0.5, 2.0, 16.0 ); self->attr.minreq = device->UpdateSize * pa_frame_size(&self->spec); self->attr.tlength = self->attr.minreq * device->NumUpdates; self->attr.maxlength = -1; self->attr.prebuf = 0; o = pa_stream_set_buffer_attr(self->stream, &self->attr, stream_success_callback, self->loop); wait_for_operation(o, self->loop); device->Frequency = self->spec.rate; } pa_stream_set_buffer_attr_callback(self->stream, ALCpulsePlayback_bufferAttrCallback, self); ALCpulsePlayback_bufferAttrCallback(self->stream, self); device->NumUpdates = (ALuint)clampu64( (self->attr.tlength + self->attr.minreq/2) / self->attr.minreq, 2, 16 ); device->UpdateSize = self->attr.minreq / pa_frame_size(&self->spec); /* HACK: prebuf should be 0 as that's what we set it to. However on some * systems it comes back as non-0, so we have to make sure the device will * write enough audio to start playback. The lack of manual start control * may have unintended consequences, but it's better than not starting at * all. */ if(self->attr.prebuf != 0) { ALuint len = self->attr.prebuf / pa_frame_size(&self->spec); if(len <= device->UpdateSize*device->NumUpdates) ERR("Non-0 prebuf, %u samples (%u bytes), device has %u samples\n", len, self->attr.prebuf, device->UpdateSize*device->NumUpdates); else { ERR("Large prebuf, %u samples (%u bytes), increasing device from %u samples", len, self->attr.prebuf, device->UpdateSize*device->NumUpdates); device->NumUpdates = (len+device->UpdateSize-1) / device->UpdateSize; } } pa_threaded_mainloop_unlock(self->loop); return ALC_TRUE; } static ALCboolean ALCpulsePlayback_start(ALCpulsePlayback *self) { ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release); if(althrd_create(&self->thread, ALCpulsePlayback_mixerProc, self) != althrd_success) return ALC_FALSE; return ALC_TRUE; } static void ALCpulsePlayback_stop(ALCpulsePlayback *self) { pa_operation *o; int res; if(!self->stream || ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel)) return; /* Signal the main loop in case PulseAudio isn't sending us audio requests * (e.g. if the device is suspended). We need to lock the mainloop in case * the mixer is between checking the killNow flag but before waiting for * the signal. */ pa_threaded_mainloop_lock(self->loop); pa_threaded_mainloop_unlock(self->loop); pa_threaded_mainloop_signal(self->loop, 0); althrd_join(self->thread, &res); pa_threaded_mainloop_lock(self->loop); o = pa_stream_cork(self->stream, 1, stream_success_callback, self->loop); wait_for_operation(o, self->loop); pa_threaded_mainloop_unlock(self->loop); } static ClockLatency ALCpulsePlayback_getClockLatency(ALCpulsePlayback *self) { ClockLatency ret; pa_usec_t latency; int neg, err; pa_threaded_mainloop_lock(self->loop); ret.ClockTime = GetDeviceClockTime(STATIC_CAST(ALCbackend,self)->mDevice); err = pa_stream_get_latency(self->stream, &latency, &neg); pa_threaded_mainloop_unlock(self->loop); if(UNLIKELY(err != 0)) { /* FIXME: if err = -PA_ERR_NODATA, it means we were called too soon * after starting the stream and no timing info has been received from * the server yet. Should we wait, possibly stalling the app, or give a * dummy value? Either way, it shouldn't be 0. */ if(err != -PA_ERR_NODATA) ERR("Failed to get stream latency: 0x%x\n", err); latency = 0; neg = 0; } else if(UNLIKELY(neg)) latency = 0; ret.Latency = (ALint64)minu64(latency, U64(0x7fffffffffffffff)/1000) * 1000; return ret; } static void ALCpulsePlayback_lock(ALCpulsePlayback *self) { pa_threaded_mainloop_lock(self->loop); } static void ALCpulsePlayback_unlock(ALCpulsePlayback *self) { pa_threaded_mainloop_unlock(self->loop); } typedef struct ALCpulseCapture { DERIVE_FROM_TYPE(ALCbackend); al_string device_name; const void *cap_store; size_t cap_len; size_t cap_remain; ALCuint last_readable; pa_buffer_attr attr; pa_sample_spec spec; pa_threaded_mainloop *loop; pa_stream *stream; pa_context *context; } ALCpulseCapture; static void ALCpulseCapture_deviceCallback(pa_context *context, const pa_source_info *info, int eol, void *pdata); static void ALCpulseCapture_probeDevices(void); static void ALCpulseCapture_contextStateCallback(pa_context *context, void *pdata); static void ALCpulseCapture_streamStateCallback(pa_stream *stream, void *pdata); static void ALCpulseCapture_sourceNameCallback(pa_context *context, const pa_source_info *info, int eol, void *pdata); static void ALCpulseCapture_streamMovedCallback(pa_stream *stream, void *pdata); static pa_stream *ALCpulseCapture_connectStream(const char *device_name, pa_threaded_mainloop *loop, pa_context *context, pa_stream_flags_t flags, pa_buffer_attr *attr, pa_sample_spec *spec, pa_channel_map *chanmap); static void ALCpulseCapture_Construct(ALCpulseCapture *self, ALCdevice *device); static void ALCpulseCapture_Destruct(ALCpulseCapture *self); static ALCenum ALCpulseCapture_open(ALCpulseCapture *self, const ALCchar *name); static DECLARE_FORWARD(ALCpulseCapture, ALCbackend, ALCboolean, reset) static ALCboolean ALCpulseCapture_start(ALCpulseCapture *self); static void ALCpulseCapture_stop(ALCpulseCapture *self); static ALCenum ALCpulseCapture_captureSamples(ALCpulseCapture *self, ALCvoid *buffer, ALCuint samples); static ALCuint ALCpulseCapture_availableSamples(ALCpulseCapture *self); static ClockLatency ALCpulseCapture_getClockLatency(ALCpulseCapture *self); static void ALCpulseCapture_lock(ALCpulseCapture *self); static void ALCpulseCapture_unlock(ALCpulseCapture *self); DECLARE_DEFAULT_ALLOCATORS(ALCpulseCapture) DEFINE_ALCBACKEND_VTABLE(ALCpulseCapture); static void ALCpulseCapture_Construct(ALCpulseCapture *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCpulseCapture, ALCbackend, self); self->loop = NULL; AL_STRING_INIT(self->device_name); } static void ALCpulseCapture_Destruct(ALCpulseCapture *self) { if(self->loop) { pulse_close(self->loop, self->context, self->stream); self->loop = NULL; self->context = NULL; self->stream = NULL; } AL_STRING_DEINIT(self->device_name); ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static void ALCpulseCapture_deviceCallback(pa_context *UNUSED(context), const pa_source_info *info, int eol, void *pdata) { pa_threaded_mainloop *loop = pdata; const DevMap *iter; DevMap entry; int count; if(eol) { pa_threaded_mainloop_signal(loop, 0); return; } #define MATCH_INFO_NAME(iter) (alstr_cmp_cstr((iter)->device_name, info->name) == 0) VECTOR_FIND_IF(iter, const DevMap, CaptureDevices, MATCH_INFO_NAME); if(iter != VECTOR_END(CaptureDevices)) return; #undef MATCH_INFO_NAME AL_STRING_INIT(entry.name); AL_STRING_INIT(entry.device_name); alstr_copy_cstr(&entry.device_name, info->name); count = 0; while(1) { alstr_copy_cstr(&entry.name, info->description); if(count != 0) { char str[64]; snprintf(str, sizeof(str), " #%d", count+1); alstr_append_cstr(&entry.name, str); } #define MATCH_ENTRY(i) (alstr_cmp(entry.name, (i)->name) == 0) VECTOR_FIND_IF(iter, const DevMap, CaptureDevices, MATCH_ENTRY); if(iter == VECTOR_END(CaptureDevices)) break; #undef MATCH_ENTRY count++; } TRACE("Got device \"%s\", \"%s\"\n", alstr_get_cstr(entry.name), alstr_get_cstr(entry.device_name)); VECTOR_PUSH_BACK(CaptureDevices, entry); } static void ALCpulseCapture_probeDevices(void) { pa_threaded_mainloop *loop; clear_devlist(&CaptureDevices); if((loop=pa_threaded_mainloop_new()) && pa_threaded_mainloop_start(loop) >= 0) { pa_context *context; pa_threaded_mainloop_lock(loop); context = connect_context(loop, AL_FALSE); if(context) { pa_operation *o; pa_stream_flags_t flags; pa_sample_spec spec; pa_stream *stream; flags = PA_STREAM_FIX_FORMAT | PA_STREAM_FIX_RATE | PA_STREAM_FIX_CHANNELS | PA_STREAM_DONT_MOVE; spec.format = PA_SAMPLE_S16NE; spec.rate = 44100; spec.channels = 1; stream = ALCpulseCapture_connectStream(NULL, loop, context, flags, NULL, &spec, NULL); if(stream) { o = pa_context_get_source_info_by_name(context, pa_stream_get_device_name(stream), ALCpulseCapture_deviceCallback, loop); wait_for_operation(o, loop); pa_stream_disconnect(stream); pa_stream_unref(stream); stream = NULL; } o = pa_context_get_source_info_list(context, ALCpulseCapture_deviceCallback, loop); wait_for_operation(o, loop); pa_context_disconnect(context); pa_context_unref(context); } pa_threaded_mainloop_unlock(loop); pa_threaded_mainloop_stop(loop); } if(loop) pa_threaded_mainloop_free(loop); } static void ALCpulseCapture_contextStateCallback(pa_context *context, void *pdata) { ALCpulseCapture *self = pdata; if(pa_context_get_state(context) == PA_CONTEXT_FAILED) { ERR("Received context failure!\n"); aluHandleDisconnect(STATIC_CAST(ALCbackend,self)->mDevice, "Capture state failure"); } pa_threaded_mainloop_signal(self->loop, 0); } static void ALCpulseCapture_streamStateCallback(pa_stream *stream, void *pdata) { ALCpulseCapture *self = pdata; if(pa_stream_get_state(stream) == PA_STREAM_FAILED) { ERR("Received stream failure!\n"); aluHandleDisconnect(STATIC_CAST(ALCbackend,self)->mDevice, "Capture stream failure"); } pa_threaded_mainloop_signal(self->loop, 0); } static void ALCpulseCapture_sourceNameCallback(pa_context *UNUSED(context), const pa_source_info *info, int eol, void *pdata) { ALCpulseCapture *self = pdata; ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; if(eol) { pa_threaded_mainloop_signal(self->loop, 0); return; } alstr_copy_cstr(&device->DeviceName, info->description); } static void ALCpulseCapture_streamMovedCallback(pa_stream *stream, void *pdata) { ALCpulseCapture *self = pdata; alstr_copy_cstr(&self->device_name, pa_stream_get_device_name(stream)); TRACE("Stream moved to %s\n", alstr_get_cstr(self->device_name)); } static pa_stream *ALCpulseCapture_connectStream(const char *device_name, pa_threaded_mainloop *loop, pa_context *context, pa_stream_flags_t flags, pa_buffer_attr *attr, pa_sample_spec *spec, pa_channel_map *chanmap) { pa_stream_state_t state; pa_stream *stream; stream = pa_stream_new_with_proplist(context, "Capture Stream", spec, chanmap, prop_filter); if(!stream) { ERR("pa_stream_new_with_proplist() failed: %s\n", pa_strerror(pa_context_errno(context))); return NULL; } pa_stream_set_state_callback(stream, stream_state_callback, loop); if(pa_stream_connect_record(stream, device_name, attr, flags) < 0) { ERR("Stream did not connect: %s\n", pa_strerror(pa_context_errno(context))); pa_stream_unref(stream); return NULL; } while((state=pa_stream_get_state(stream)) != PA_STREAM_READY) { if(!PA_STREAM_IS_GOOD(state)) { ERR("Stream did not get ready: %s\n", pa_strerror(pa_context_errno(context))); pa_stream_unref(stream); return NULL; } pa_threaded_mainloop_wait(loop); } pa_stream_set_state_callback(stream, NULL, NULL); return stream; } static ALCenum ALCpulseCapture_open(ALCpulseCapture *self, const ALCchar *name) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; const char *pulse_name = NULL; pa_stream_flags_t flags = 0; const char *mapname = NULL; pa_channel_map chanmap; ALuint samples; if(name) { const DevMap *iter; if(VECTOR_SIZE(CaptureDevices) == 0) ALCpulseCapture_probeDevices(); #define MATCH_NAME(iter) (alstr_cmp_cstr((iter)->name, name) == 0) VECTOR_FIND_IF(iter, const DevMap, CaptureDevices, MATCH_NAME); #undef MATCH_NAME if(iter == VECTOR_END(CaptureDevices)) return ALC_INVALID_VALUE; pulse_name = alstr_get_cstr(iter->device_name); alstr_copy(&device->DeviceName, iter->name); } if(!pulse_open(&self->loop, &self->context, ALCpulseCapture_contextStateCallback, self)) return ALC_INVALID_VALUE; pa_threaded_mainloop_lock(self->loop); switch(device->FmtType) { case DevFmtUByte: self->spec.format = PA_SAMPLE_U8; break; case DevFmtShort: self->spec.format = PA_SAMPLE_S16NE; break; case DevFmtInt: self->spec.format = PA_SAMPLE_S32NE; break; case DevFmtFloat: self->spec.format = PA_SAMPLE_FLOAT32NE; break; case DevFmtByte: case DevFmtUShort: case DevFmtUInt: ERR("%s capture samples not supported\n", DevFmtTypeString(device->FmtType)); pa_threaded_mainloop_unlock(self->loop); goto fail; } switch(device->FmtChans) { case DevFmtMono: mapname = "mono"; break; case DevFmtStereo: mapname = "front-left,front-right"; break; case DevFmtQuad: mapname = "front-left,front-right,rear-left,rear-right"; break; case DevFmtX51: mapname = "front-left,front-right,front-center,lfe,side-left,side-right"; break; case DevFmtX51Rear: mapname = "front-left,front-right,front-center,lfe,rear-left,rear-right"; break; case DevFmtX61: mapname = "front-left,front-right,front-center,lfe,rear-center,side-left,side-right"; break; case DevFmtX71: mapname = "front-left,front-right,front-center,lfe,rear-left,rear-right,side-left,side-right"; break; case DevFmtAmbi3D: ERR("%s capture samples not supported\n", DevFmtChannelsString(device->FmtChans)); pa_threaded_mainloop_unlock(self->loop); goto fail; } if(!pa_channel_map_parse(&chanmap, mapname)) { ERR("Failed to build channel map for %s\n", DevFmtChannelsString(device->FmtChans)); pa_threaded_mainloop_unlock(self->loop); return ALC_FALSE; } self->spec.rate = device->Frequency; self->spec.channels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); if(pa_sample_spec_valid(&self->spec) == 0) { ERR("Invalid sample format\n"); pa_threaded_mainloop_unlock(self->loop); goto fail; } if(!pa_channel_map_init_auto(&chanmap, self->spec.channels, PA_CHANNEL_MAP_WAVEEX)) { ERR("Couldn't build map for channel count (%d)!\n", self->spec.channels); pa_threaded_mainloop_unlock(self->loop); goto fail; } samples = device->UpdateSize * device->NumUpdates; samples = maxu(samples, 100 * device->Frequency / 1000); self->attr.minreq = -1; self->attr.prebuf = -1; self->attr.maxlength = samples * pa_frame_size(&self->spec); self->attr.tlength = -1; self->attr.fragsize = minu(samples, 50*device->Frequency/1000) * pa_frame_size(&self->spec); flags |= PA_STREAM_START_CORKED|PA_STREAM_ADJUST_LATENCY; if(!GetConfigValueBool(NULL, "pulse", "allow-moves", 0)) flags |= PA_STREAM_DONT_MOVE; TRACE("Connecting to \"%s\"\n", pulse_name ? pulse_name : "(default)"); self->stream = ALCpulseCapture_connectStream(pulse_name, self->loop, self->context, flags, &self->attr, &self->spec, &chanmap ); if(!self->stream) { pa_threaded_mainloop_unlock(self->loop); goto fail; } pa_stream_set_moved_callback(self->stream, ALCpulseCapture_streamMovedCallback, self); pa_stream_set_state_callback(self->stream, ALCpulseCapture_streamStateCallback, self); alstr_copy_cstr(&self->device_name, pa_stream_get_device_name(self->stream)); if(alstr_empty(device->DeviceName)) { pa_operation *o = pa_context_get_source_info_by_name( self->context, alstr_get_cstr(self->device_name), ALCpulseCapture_sourceNameCallback, self ); wait_for_operation(o, self->loop); } pa_threaded_mainloop_unlock(self->loop); return ALC_NO_ERROR; fail: pulse_close(self->loop, self->context, self->stream); self->loop = NULL; self->context = NULL; self->stream = NULL; return ALC_INVALID_VALUE; } static ALCboolean ALCpulseCapture_start(ALCpulseCapture *self) { pa_operation *o; pa_threaded_mainloop_lock(self->loop); o = pa_stream_cork(self->stream, 0, stream_success_callback, self->loop); wait_for_operation(o, self->loop); pa_threaded_mainloop_unlock(self->loop); return ALC_TRUE; } static void ALCpulseCapture_stop(ALCpulseCapture *self) { pa_operation *o; pa_threaded_mainloop_lock(self->loop); o = pa_stream_cork(self->stream, 1, stream_success_callback, self->loop); wait_for_operation(o, self->loop); pa_threaded_mainloop_unlock(self->loop); } static ALCenum ALCpulseCapture_captureSamples(ALCpulseCapture *self, ALCvoid *buffer, ALCuint samples) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; ALCuint todo = samples * pa_frame_size(&self->spec); /* Capture is done in fragment-sized chunks, so we loop until we get all * that's available */ self->last_readable -= todo; pa_threaded_mainloop_lock(self->loop); while(todo > 0) { size_t rem = todo; if(self->cap_len == 0) { pa_stream_state_t state; state = pa_stream_get_state(self->stream); if(!PA_STREAM_IS_GOOD(state)) { aluHandleDisconnect(device, "Bad capture state: %u", state); break; } if(pa_stream_peek(self->stream, &self->cap_store, &self->cap_len) < 0) { ERR("pa_stream_peek() failed: %s\n", pa_strerror(pa_context_errno(self->context))); aluHandleDisconnect(device, "Failed retrieving capture samples: %s", pa_strerror(pa_context_errno(self->context))); break; } self->cap_remain = self->cap_len; } if(rem > self->cap_remain) rem = self->cap_remain; memcpy(buffer, self->cap_store, rem); buffer = (ALbyte*)buffer + rem; todo -= rem; self->cap_store = (ALbyte*)self->cap_store + rem; self->cap_remain -= rem; if(self->cap_remain == 0) { pa_stream_drop(self->stream); self->cap_len = 0; } } pa_threaded_mainloop_unlock(self->loop); if(todo > 0) memset(buffer, ((device->FmtType==DevFmtUByte) ? 0x80 : 0), todo); return ALC_NO_ERROR; } static ALCuint ALCpulseCapture_availableSamples(ALCpulseCapture *self) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; size_t readable = self->cap_remain; if(ATOMIC_LOAD(&device->Connected, almemory_order_acquire)) { ssize_t got; pa_threaded_mainloop_lock(self->loop); got = pa_stream_readable_size(self->stream); if(got < 0) { ERR("pa_stream_readable_size() failed: %s\n", pa_strerror(got)); aluHandleDisconnect(device, "Failed getting readable size: %s", pa_strerror(got)); } else if((size_t)got > self->cap_len) readable += got - self->cap_len; pa_threaded_mainloop_unlock(self->loop); } if(self->last_readable < readable) self->last_readable = readable; return self->last_readable / pa_frame_size(&self->spec); } static ClockLatency ALCpulseCapture_getClockLatency(ALCpulseCapture *self) { ClockLatency ret; pa_usec_t latency; int neg, err; pa_threaded_mainloop_lock(self->loop); ret.ClockTime = GetDeviceClockTime(STATIC_CAST(ALCbackend,self)->mDevice); err = pa_stream_get_latency(self->stream, &latency, &neg); pa_threaded_mainloop_unlock(self->loop); if(UNLIKELY(err != 0)) { ERR("Failed to get stream latency: 0x%x\n", err); latency = 0; neg = 0; } else if(UNLIKELY(neg)) latency = 0; ret.Latency = (ALint64)minu64(latency, U64(0x7fffffffffffffff)/1000) * 1000; return ret; } static void ALCpulseCapture_lock(ALCpulseCapture *self) { pa_threaded_mainloop_lock(self->loop); } static void ALCpulseCapture_unlock(ALCpulseCapture *self) { pa_threaded_mainloop_unlock(self->loop); } typedef struct ALCpulseBackendFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } ALCpulseBackendFactory; #define ALCPULSEBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCpulseBackendFactory, ALCbackendFactory) } } static ALCboolean ALCpulseBackendFactory_init(ALCpulseBackendFactory *self); static void ALCpulseBackendFactory_deinit(ALCpulseBackendFactory *self); static ALCboolean ALCpulseBackendFactory_querySupport(ALCpulseBackendFactory *self, ALCbackend_Type type); static void ALCpulseBackendFactory_probe(ALCpulseBackendFactory *self, enum DevProbe type, al_string *outnames); static ALCbackend* ALCpulseBackendFactory_createBackend(ALCpulseBackendFactory *self, ALCdevice *device, ALCbackend_Type type); DEFINE_ALCBACKENDFACTORY_VTABLE(ALCpulseBackendFactory); static ALCboolean ALCpulseBackendFactory_init(ALCpulseBackendFactory* UNUSED(self)) { ALCboolean ret = ALC_FALSE; VECTOR_INIT(PlaybackDevices); VECTOR_INIT(CaptureDevices); if(pulse_load()) { pa_threaded_mainloop *loop; pulse_ctx_flags = 0; if(!GetConfigValueBool(NULL, "pulse", "spawn-server", 1)) pulse_ctx_flags |= PA_CONTEXT_NOAUTOSPAWN; if((loop=pa_threaded_mainloop_new()) && pa_threaded_mainloop_start(loop) >= 0) { pa_context *context; pa_threaded_mainloop_lock(loop); context = connect_context(loop, AL_TRUE); if(context) { ret = ALC_TRUE; /* Some libraries (Phonon, Qt) set some pulseaudio properties * through environment variables, which causes all streams in * the process to inherit them. This attempts to filter those * properties out by setting them to 0-length data. */ prop_filter = pa_proplist_new(); pa_proplist_set(prop_filter, PA_PROP_MEDIA_ROLE, NULL, 0); pa_proplist_set(prop_filter, "phonon.streamid", NULL, 0); pa_context_disconnect(context); pa_context_unref(context); } pa_threaded_mainloop_unlock(loop); pa_threaded_mainloop_stop(loop); } if(loop) pa_threaded_mainloop_free(loop); } return ret; } static void ALCpulseBackendFactory_deinit(ALCpulseBackendFactory* UNUSED(self)) { clear_devlist(&PlaybackDevices); VECTOR_DEINIT(PlaybackDevices); clear_devlist(&CaptureDevices); VECTOR_DEINIT(CaptureDevices); if(prop_filter) pa_proplist_free(prop_filter); prop_filter = NULL; /* PulseAudio doesn't like being CloseLib'd sometimes */ } static ALCboolean ALCpulseBackendFactory_querySupport(ALCpulseBackendFactory* UNUSED(self), ALCbackend_Type type) { if(type == ALCbackend_Playback || type == ALCbackend_Capture) return ALC_TRUE; return ALC_FALSE; } static void ALCpulseBackendFactory_probe(ALCpulseBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames) { switch(type) { #define APPEND_OUTNAME(e) do { \ if(!alstr_empty((e)->name)) \ alstr_append_range(outnames, VECTOR_BEGIN((e)->name), \ VECTOR_END((e)->name)+1); \ } while(0) case ALL_DEVICE_PROBE: ALCpulsePlayback_probeDevices(); VECTOR_FOR_EACH(const DevMap, PlaybackDevices, APPEND_OUTNAME); break; case CAPTURE_DEVICE_PROBE: ALCpulseCapture_probeDevices(); VECTOR_FOR_EACH(const DevMap, CaptureDevices, APPEND_OUTNAME); break; #undef APPEND_OUTNAME } } static ALCbackend* ALCpulseBackendFactory_createBackend(ALCpulseBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) { if(type == ALCbackend_Playback) { ALCpulsePlayback *backend; NEW_OBJ(backend, ALCpulsePlayback)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } if(type == ALCbackend_Capture) { ALCpulseCapture *backend; NEW_OBJ(backend, ALCpulseCapture)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } return NULL; } #else /* PA_API_VERSION == 12 */ #warning "Unsupported API version, backend will be unavailable!" typedef struct ALCpulseBackendFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } ALCpulseBackendFactory; #define ALCPULSEBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCpulseBackendFactory, ALCbackendFactory) } } static ALCboolean ALCpulseBackendFactory_init(ALCpulseBackendFactory* UNUSED(self)) { return ALC_FALSE; } static void ALCpulseBackendFactory_deinit(ALCpulseBackendFactory* UNUSED(self)) { } static ALCboolean ALCpulseBackendFactory_querySupport(ALCpulseBackendFactory* UNUSED(self), ALCbackend_Type UNUSED(type)) { return ALC_FALSE; } static void ALCpulseBackendFactory_probe(ALCpulseBackendFactory* UNUSED(self), enum DevProbe UNUSED(type), al_string* UNUSED(outnames)) { } static ALCbackend* ALCpulseBackendFactory_createBackend(ALCpulseBackendFactory* UNUSED(self), ALCdevice* UNUSED(device), ALCbackend_Type UNUSED(type)) { return NULL; } DEFINE_ALCBACKENDFACTORY_VTABLE(ALCpulseBackendFactory); #endif /* PA_API_VERSION == 12 */ ALCbackendFactory *ALCpulseBackendFactory_getFactory(void) { static ALCpulseBackendFactory factory = ALCPULSEBACKENDFACTORY_INITIALIZER; return STATIC_CAST(ALCbackendFactory, &factory); } openal-soft-openal-soft-1.19.1/Alc/backends/qsa.c000066400000000000000000000757121335774445300215130ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2011-2013 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include #include #include #include #include #include "alMain.h" #include "alu.h" #include "threads.h" #include "backends/base.h" typedef struct { snd_pcm_t* pcmHandle; int audio_fd; snd_pcm_channel_setup_t csetup; snd_pcm_channel_params_t cparams; ALvoid* buffer; ALsizei size; ATOMIC(ALenum) killNow; althrd_t thread; } qsa_data; typedef struct { ALCchar* name; int card; int dev; } DevMap; TYPEDEF_VECTOR(DevMap, vector_DevMap) static vector_DevMap DeviceNameMap; static vector_DevMap CaptureNameMap; static const ALCchar qsaDevice[] = "QSA Default"; static const struct { int32_t format; } formatlist[] = { {SND_PCM_SFMT_FLOAT_LE}, {SND_PCM_SFMT_S32_LE}, {SND_PCM_SFMT_U32_LE}, {SND_PCM_SFMT_S16_LE}, {SND_PCM_SFMT_U16_LE}, {SND_PCM_SFMT_S8}, {SND_PCM_SFMT_U8}, {0}, }; static const struct { int32_t rate; } ratelist[] = { {192000}, {176400}, {96000}, {88200}, {48000}, {44100}, {32000}, {24000}, {22050}, {16000}, {12000}, {11025}, {8000}, {0}, }; static const struct { int32_t channels; } channellist[] = { {8}, {7}, {6}, {4}, {2}, {1}, {0}, }; static void deviceList(int type, vector_DevMap *devmap) { snd_ctl_t* handle; snd_pcm_info_t pcminfo; int max_cards, card, err, dev; DevMap entry; char name[1024]; struct snd_ctl_hw_info info; max_cards = snd_cards(); if(max_cards < 0) return; #define FREE_NAME(iter) free((iter)->name) VECTOR_FOR_EACH(DevMap, *devmap, FREE_NAME); #undef FREE_NAME VECTOR_RESIZE(*devmap, 0, max_cards+1); entry.name = strdup(qsaDevice); entry.card = 0; entry.dev = 0; VECTOR_PUSH_BACK(*devmap, entry); for(card = 0;card < max_cards;card++) { if((err=snd_ctl_open(&handle, card)) < 0) continue; if((err=snd_ctl_hw_info(handle, &info)) < 0) { snd_ctl_close(handle); continue; } for(dev = 0;dev < (int)info.pcmdevs;dev++) { if((err=snd_ctl_pcm_info(handle, dev, &pcminfo)) < 0) continue; if((type==SND_PCM_CHANNEL_PLAYBACK && (pcminfo.flags&SND_PCM_INFO_PLAYBACK)) || (type==SND_PCM_CHANNEL_CAPTURE && (pcminfo.flags&SND_PCM_INFO_CAPTURE))) { snprintf(name, sizeof(name), "%s [%s] (hw:%d,%d)", info.name, pcminfo.name, card, dev); entry.name = strdup(name); entry.card = card; entry.dev = dev; VECTOR_PUSH_BACK(*devmap, entry); TRACE("Got device \"%s\", card %d, dev %d\n", name, card, dev); } } snd_ctl_close(handle); } } /* Wrappers to use an old-style backend with the new interface. */ typedef struct PlaybackWrapper { DERIVE_FROM_TYPE(ALCbackend); qsa_data *ExtraData; } PlaybackWrapper; static void PlaybackWrapper_Construct(PlaybackWrapper *self, ALCdevice *device); static void PlaybackWrapper_Destruct(PlaybackWrapper *self); static ALCenum PlaybackWrapper_open(PlaybackWrapper *self, const ALCchar *name); static ALCboolean PlaybackWrapper_reset(PlaybackWrapper *self); static ALCboolean PlaybackWrapper_start(PlaybackWrapper *self); static void PlaybackWrapper_stop(PlaybackWrapper *self); static DECLARE_FORWARD2(PlaybackWrapper, ALCbackend, ALCenum, captureSamples, void*, ALCuint) static DECLARE_FORWARD(PlaybackWrapper, ALCbackend, ALCuint, availableSamples) static DECLARE_FORWARD(PlaybackWrapper, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(PlaybackWrapper, ALCbackend, void, lock) static DECLARE_FORWARD(PlaybackWrapper, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(PlaybackWrapper) DEFINE_ALCBACKEND_VTABLE(PlaybackWrapper); FORCE_ALIGN static int qsa_proc_playback(void *ptr) { PlaybackWrapper *self = ptr; ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; qsa_data *data = self->ExtraData; snd_pcm_channel_status_t status; struct sched_param param; struct timeval timeout; char* write_ptr; fd_set wfds; ALint len; int sret; SetRTPriority(); althrd_setname(althrd_current(), MIXER_THREAD_NAME); /* Increase default 10 priority to 11 to avoid jerky sound */ SchedGet(0, 0, ¶m); param.sched_priority=param.sched_curpriority+1; SchedSet(0, 0, SCHED_NOCHANGE, ¶m); const ALint frame_size = FrameSizeFromDevFmt( device->FmtChans, device->FmtType, device->AmbiOrder ); V0(device->Backend,lock)(); while(!ATOMIC_LOAD(&data->killNow, almemory_order_acquire)) { FD_ZERO(&wfds); FD_SET(data->audio_fd, &wfds); timeout.tv_sec=2; timeout.tv_usec=0; /* Select also works like time slice to OS */ V0(device->Backend,unlock)(); sret = select(data->audio_fd+1, NULL, &wfds, NULL, &timeout); V0(device->Backend,lock)(); if(sret == -1) { ERR("select error: %s\n", strerror(errno)); aluHandleDisconnect(device, "Failed waiting for playback buffer: %s", strerror(errno)); break; } if(sret == 0) { ERR("select timeout\n"); continue; } len = data->size; write_ptr = data->buffer; aluMixData(device, write_ptr, len/frame_size); while(len>0 && !ATOMIC_LOAD(&data->killNow, almemory_order_acquire)) { int wrote = snd_pcm_plugin_write(data->pcmHandle, write_ptr, len); if(wrote <= 0) { if(errno==EAGAIN || errno==EWOULDBLOCK) continue; memset(&status, 0, sizeof(status)); status.channel = SND_PCM_CHANNEL_PLAYBACK; snd_pcm_plugin_status(data->pcmHandle, &status); /* we need to reinitialize the sound channel if we've underrun the buffer */ if(status.status == SND_PCM_STATUS_UNDERRUN || status.status == SND_PCM_STATUS_READY) { if(snd_pcm_plugin_prepare(data->pcmHandle, SND_PCM_CHANNEL_PLAYBACK) < 0) { aluHandleDisconnect(device, "Playback recovery failed"); break; } } } else { write_ptr += wrote; len -= wrote; } } } V0(device->Backend,unlock)(); return 0; } /************/ /* Playback */ /************/ static ALCenum qsa_open_playback(PlaybackWrapper *self, const ALCchar* deviceName) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; qsa_data *data; int card, dev; int status; data = (qsa_data*)calloc(1, sizeof(qsa_data)); if(data == NULL) return ALC_OUT_OF_MEMORY; ATOMIC_INIT(&data->killNow, AL_TRUE); if(!deviceName) deviceName = qsaDevice; if(strcmp(deviceName, qsaDevice) == 0) status = snd_pcm_open_preferred(&data->pcmHandle, &card, &dev, SND_PCM_OPEN_PLAYBACK); else { const DevMap *iter; if(VECTOR_SIZE(DeviceNameMap) == 0) deviceList(SND_PCM_CHANNEL_PLAYBACK, &DeviceNameMap); #define MATCH_DEVNAME(iter) ((iter)->name && strcmp(deviceName, (iter)->name)==0) VECTOR_FIND_IF(iter, const DevMap, DeviceNameMap, MATCH_DEVNAME); #undef MATCH_DEVNAME if(iter == VECTOR_END(DeviceNameMap)) { free(data); return ALC_INVALID_DEVICE; } status = snd_pcm_open(&data->pcmHandle, iter->card, iter->dev, SND_PCM_OPEN_PLAYBACK); } if(status < 0) { free(data); return ALC_INVALID_DEVICE; } data->audio_fd = snd_pcm_file_descriptor(data->pcmHandle, SND_PCM_CHANNEL_PLAYBACK); if(data->audio_fd < 0) { snd_pcm_close(data->pcmHandle); free(data); return ALC_INVALID_DEVICE; } alstr_copy_cstr(&device->DeviceName, deviceName); self->ExtraData = data; return ALC_NO_ERROR; } static void qsa_close_playback(PlaybackWrapper *self) { qsa_data *data = self->ExtraData; if (data->buffer!=NULL) { free(data->buffer); data->buffer=NULL; } snd_pcm_close(data->pcmHandle); free(data); self->ExtraData = NULL; } static ALCboolean qsa_reset_playback(PlaybackWrapper *self) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; qsa_data *data = self->ExtraData; int32_t format=-1; switch(device->FmtType) { case DevFmtByte: format=SND_PCM_SFMT_S8; break; case DevFmtUByte: format=SND_PCM_SFMT_U8; break; case DevFmtShort: format=SND_PCM_SFMT_S16_LE; break; case DevFmtUShort: format=SND_PCM_SFMT_U16_LE; break; case DevFmtInt: format=SND_PCM_SFMT_S32_LE; break; case DevFmtUInt: format=SND_PCM_SFMT_U32_LE; break; case DevFmtFloat: format=SND_PCM_SFMT_FLOAT_LE; break; } /* we actually don't want to block on writes */ snd_pcm_nonblock_mode(data->pcmHandle, 1); /* Disable mmap to control data transfer to the audio device */ snd_pcm_plugin_set_disable(data->pcmHandle, PLUGIN_DISABLE_MMAP); snd_pcm_plugin_set_disable(data->pcmHandle, PLUGIN_DISABLE_BUFFER_PARTIAL_BLOCKS); // configure a sound channel memset(&data->cparams, 0, sizeof(data->cparams)); data->cparams.channel=SND_PCM_CHANNEL_PLAYBACK; data->cparams.mode=SND_PCM_MODE_BLOCK; data->cparams.start_mode=SND_PCM_START_FULL; data->cparams.stop_mode=SND_PCM_STOP_STOP; data->cparams.buf.block.frag_size=device->UpdateSize * FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); data->cparams.buf.block.frags_max=device->NumUpdates; data->cparams.buf.block.frags_min=device->NumUpdates; data->cparams.format.interleave=1; data->cparams.format.rate=device->Frequency; data->cparams.format.voices=ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); data->cparams.format.format=format; if ((snd_pcm_plugin_params(data->pcmHandle, &data->cparams))<0) { int original_rate=data->cparams.format.rate; int original_voices=data->cparams.format.voices; int original_format=data->cparams.format.format; int it; int jt; for (it=0; it<1; it++) { /* Check for second pass */ if (it==1) { original_rate=ratelist[0].rate; original_voices=channellist[0].channels; original_format=formatlist[0].format; } do { /* At first downgrade sample format */ jt=0; do { if (formatlist[jt].format==data->cparams.format.format) { data->cparams.format.format=formatlist[jt+1].format; break; } if (formatlist[jt].format==0) { data->cparams.format.format=0; break; } jt++; } while(1); if (data->cparams.format.format==0) { data->cparams.format.format=original_format; /* At secod downgrade sample rate */ jt=0; do { if (ratelist[jt].rate==data->cparams.format.rate) { data->cparams.format.rate=ratelist[jt+1].rate; break; } if (ratelist[jt].rate==0) { data->cparams.format.rate=0; break; } jt++; } while(1); if (data->cparams.format.rate==0) { data->cparams.format.rate=original_rate; data->cparams.format.format=original_format; /* At third downgrade channels number */ jt=0; do { if(channellist[jt].channels==data->cparams.format.voices) { data->cparams.format.voices=channellist[jt+1].channels; break; } if (channellist[jt].channels==0) { data->cparams.format.voices=0; break; } jt++; } while(1); } if (data->cparams.format.voices==0) { break; } } data->cparams.buf.block.frag_size=device->UpdateSize* data->cparams.format.voices* snd_pcm_format_width(data->cparams.format.format)/8; data->cparams.buf.block.frags_max=device->NumUpdates; data->cparams.buf.block.frags_min=device->NumUpdates; if ((snd_pcm_plugin_params(data->pcmHandle, &data->cparams))<0) { continue; } else { break; } } while(1); if (data->cparams.format.voices!=0) { break; } } if (data->cparams.format.voices==0) { return ALC_FALSE; } } if ((snd_pcm_plugin_prepare(data->pcmHandle, SND_PCM_CHANNEL_PLAYBACK))<0) { return ALC_FALSE; } memset(&data->csetup, 0, sizeof(data->csetup)); data->csetup.channel=SND_PCM_CHANNEL_PLAYBACK; if (snd_pcm_plugin_setup(data->pcmHandle, &data->csetup)<0) { return ALC_FALSE; } /* now fill back to the our AL device */ device->Frequency=data->cparams.format.rate; switch (data->cparams.format.voices) { case 1: device->FmtChans=DevFmtMono; break; case 2: device->FmtChans=DevFmtStereo; break; case 4: device->FmtChans=DevFmtQuad; break; case 6: device->FmtChans=DevFmtX51; break; case 7: device->FmtChans=DevFmtX61; break; case 8: device->FmtChans=DevFmtX71; break; default: device->FmtChans=DevFmtMono; break; } switch (data->cparams.format.format) { case SND_PCM_SFMT_S8: device->FmtType=DevFmtByte; break; case SND_PCM_SFMT_U8: device->FmtType=DevFmtUByte; break; case SND_PCM_SFMT_S16_LE: device->FmtType=DevFmtShort; break; case SND_PCM_SFMT_U16_LE: device->FmtType=DevFmtUShort; break; case SND_PCM_SFMT_S32_LE: device->FmtType=DevFmtInt; break; case SND_PCM_SFMT_U32_LE: device->FmtType=DevFmtUInt; break; case SND_PCM_SFMT_FLOAT_LE: device->FmtType=DevFmtFloat; break; default: device->FmtType=DevFmtShort; break; } SetDefaultChannelOrder(device); device->UpdateSize=data->csetup.buf.block.frag_size/ FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); device->NumUpdates=data->csetup.buf.block.frags; data->size=data->csetup.buf.block.frag_size; data->buffer=malloc(data->size); if (!data->buffer) { return ALC_FALSE; } return ALC_TRUE; } static ALCboolean qsa_start_playback(PlaybackWrapper *self) { qsa_data *data = self->ExtraData; ATOMIC_STORE(&data->killNow, AL_FALSE, almemory_order_release); if(althrd_create(&data->thread, qsa_proc_playback, self) != althrd_success) return ALC_FALSE; return ALC_TRUE; } static void qsa_stop_playback(PlaybackWrapper *self) { qsa_data *data = self->ExtraData; int res; if(ATOMIC_EXCHANGE(&data->killNow, AL_TRUE, almemory_order_acq_rel)) return; althrd_join(data->thread, &res); } static void PlaybackWrapper_Construct(PlaybackWrapper *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(PlaybackWrapper, ALCbackend, self); self->ExtraData = NULL; } static void PlaybackWrapper_Destruct(PlaybackWrapper *self) { if(self->ExtraData) qsa_close_playback(self); ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static ALCenum PlaybackWrapper_open(PlaybackWrapper *self, const ALCchar *name) { return qsa_open_playback(self, name); } static ALCboolean PlaybackWrapper_reset(PlaybackWrapper *self) { return qsa_reset_playback(self); } static ALCboolean PlaybackWrapper_start(PlaybackWrapper *self) { return qsa_start_playback(self); } static void PlaybackWrapper_stop(PlaybackWrapper *self) { qsa_stop_playback(self); } /***********/ /* Capture */ /***********/ typedef struct CaptureWrapper { DERIVE_FROM_TYPE(ALCbackend); qsa_data *ExtraData; } CaptureWrapper; static void CaptureWrapper_Construct(CaptureWrapper *self, ALCdevice *device); static void CaptureWrapper_Destruct(CaptureWrapper *self); static ALCenum CaptureWrapper_open(CaptureWrapper *self, const ALCchar *name); static DECLARE_FORWARD(CaptureWrapper, ALCbackend, ALCboolean, reset) static ALCboolean CaptureWrapper_start(CaptureWrapper *self); static void CaptureWrapper_stop(CaptureWrapper *self); static ALCenum CaptureWrapper_captureSamples(CaptureWrapper *self, void *buffer, ALCuint samples); static ALCuint CaptureWrapper_availableSamples(CaptureWrapper *self); static DECLARE_FORWARD(CaptureWrapper, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(CaptureWrapper, ALCbackend, void, lock) static DECLARE_FORWARD(CaptureWrapper, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(CaptureWrapper) DEFINE_ALCBACKEND_VTABLE(CaptureWrapper); static ALCenum qsa_open_capture(CaptureWrapper *self, const ALCchar *deviceName) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; qsa_data *data; int card, dev; int format=-1; int status; data=(qsa_data*)calloc(1, sizeof(qsa_data)); if (data==NULL) { return ALC_OUT_OF_MEMORY; } if(!deviceName) deviceName = qsaDevice; if(strcmp(deviceName, qsaDevice) == 0) status = snd_pcm_open_preferred(&data->pcmHandle, &card, &dev, SND_PCM_OPEN_CAPTURE); else { const DevMap *iter; if(VECTOR_SIZE(CaptureNameMap) == 0) deviceList(SND_PCM_CHANNEL_CAPTURE, &CaptureNameMap); #define MATCH_DEVNAME(iter) ((iter)->name && strcmp(deviceName, (iter)->name)==0) VECTOR_FIND_IF(iter, const DevMap, CaptureNameMap, MATCH_DEVNAME); #undef MATCH_DEVNAME if(iter == VECTOR_END(CaptureNameMap)) { free(data); return ALC_INVALID_DEVICE; } status = snd_pcm_open(&data->pcmHandle, iter->card, iter->dev, SND_PCM_OPEN_CAPTURE); } if(status < 0) { free(data); return ALC_INVALID_DEVICE; } data->audio_fd = snd_pcm_file_descriptor(data->pcmHandle, SND_PCM_CHANNEL_CAPTURE); if(data->audio_fd < 0) { snd_pcm_close(data->pcmHandle); free(data); return ALC_INVALID_DEVICE; } alstr_copy_cstr(&device->DeviceName, deviceName); self->ExtraData = data; switch (device->FmtType) { case DevFmtByte: format=SND_PCM_SFMT_S8; break; case DevFmtUByte: format=SND_PCM_SFMT_U8; break; case DevFmtShort: format=SND_PCM_SFMT_S16_LE; break; case DevFmtUShort: format=SND_PCM_SFMT_U16_LE; break; case DevFmtInt: format=SND_PCM_SFMT_S32_LE; break; case DevFmtUInt: format=SND_PCM_SFMT_U32_LE; break; case DevFmtFloat: format=SND_PCM_SFMT_FLOAT_LE; break; } /* we actually don't want to block on reads */ snd_pcm_nonblock_mode(data->pcmHandle, 1); /* Disable mmap to control data transfer to the audio device */ snd_pcm_plugin_set_disable(data->pcmHandle, PLUGIN_DISABLE_MMAP); /* configure a sound channel */ memset(&data->cparams, 0, sizeof(data->cparams)); data->cparams.mode=SND_PCM_MODE_BLOCK; data->cparams.channel=SND_PCM_CHANNEL_CAPTURE; data->cparams.start_mode=SND_PCM_START_GO; data->cparams.stop_mode=SND_PCM_STOP_STOP; data->cparams.buf.block.frag_size=device->UpdateSize* FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); data->cparams.buf.block.frags_max=device->NumUpdates; data->cparams.buf.block.frags_min=device->NumUpdates; data->cparams.format.interleave=1; data->cparams.format.rate=device->Frequency; data->cparams.format.voices=ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); data->cparams.format.format=format; if(snd_pcm_plugin_params(data->pcmHandle, &data->cparams) < 0) { snd_pcm_close(data->pcmHandle); free(data); return ALC_INVALID_VALUE; } return ALC_NO_ERROR; } static void qsa_close_capture(CaptureWrapper *self) { qsa_data *data = self->ExtraData; if (data->pcmHandle!=NULL) snd_pcm_close(data->pcmHandle); free(data); self->ExtraData = NULL; } static void qsa_start_capture(CaptureWrapper *self) { qsa_data *data = self->ExtraData; int rstatus; if ((rstatus=snd_pcm_plugin_prepare(data->pcmHandle, SND_PCM_CHANNEL_CAPTURE))<0) { ERR("capture prepare failed: %s\n", snd_strerror(rstatus)); return; } memset(&data->csetup, 0, sizeof(data->csetup)); data->csetup.channel=SND_PCM_CHANNEL_CAPTURE; if ((rstatus=snd_pcm_plugin_setup(data->pcmHandle, &data->csetup))<0) { ERR("capture setup failed: %s\n", snd_strerror(rstatus)); return; } snd_pcm_capture_go(data->pcmHandle); } static void qsa_stop_capture(CaptureWrapper *self) { qsa_data *data = self->ExtraData; snd_pcm_capture_flush(data->pcmHandle); } static ALCuint qsa_available_samples(CaptureWrapper *self) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; qsa_data *data = self->ExtraData; snd_pcm_channel_status_t status; ALint frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); ALint free_size; int rstatus; memset(&status, 0, sizeof (status)); status.channel=SND_PCM_CHANNEL_CAPTURE; snd_pcm_plugin_status(data->pcmHandle, &status); if ((status.status==SND_PCM_STATUS_OVERRUN) || (status.status==SND_PCM_STATUS_READY)) { if ((rstatus=snd_pcm_plugin_prepare(data->pcmHandle, SND_PCM_CHANNEL_CAPTURE))<0) { ERR("capture prepare failed: %s\n", snd_strerror(rstatus)); aluHandleDisconnect(device, "Failed capture recovery: %s", snd_strerror(rstatus)); return 0; } snd_pcm_capture_go(data->pcmHandle); return 0; } free_size=data->csetup.buf.block.frag_size*data->csetup.buf.block.frags; free_size-=status.free; return free_size/frame_size; } static ALCenum qsa_capture_samples(CaptureWrapper *self, ALCvoid *buffer, ALCuint samples) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; qsa_data *data = self->ExtraData; char* read_ptr; snd_pcm_channel_status_t status; fd_set rfds; int selectret; struct timeval timeout; int bytes_read; ALint frame_size=FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); ALint len=samples*frame_size; int rstatus; read_ptr=buffer; while (len>0) { FD_ZERO(&rfds); FD_SET(data->audio_fd, &rfds); timeout.tv_sec=2; timeout.tv_usec=0; /* Select also works like time slice to OS */ bytes_read=0; selectret=select(data->audio_fd+1, &rfds, NULL, NULL, &timeout); switch (selectret) { case -1: aluHandleDisconnect(device, "Failed to check capture samples"); return ALC_INVALID_DEVICE; case 0: break; default: if (FD_ISSET(data->audio_fd, &rfds)) { bytes_read=snd_pcm_plugin_read(data->pcmHandle, read_ptr, len); break; } break; } if (bytes_read<=0) { if ((errno==EAGAIN) || (errno==EWOULDBLOCK)) { continue; } memset(&status, 0, sizeof (status)); status.channel=SND_PCM_CHANNEL_CAPTURE; snd_pcm_plugin_status(data->pcmHandle, &status); /* we need to reinitialize the sound channel if we've overrun the buffer */ if ((status.status==SND_PCM_STATUS_OVERRUN) || (status.status==SND_PCM_STATUS_READY)) { if ((rstatus=snd_pcm_plugin_prepare(data->pcmHandle, SND_PCM_CHANNEL_CAPTURE))<0) { ERR("capture prepare failed: %s\n", snd_strerror(rstatus)); aluHandleDisconnect(device, "Failed capture recovery: %s", snd_strerror(rstatus)); return ALC_INVALID_DEVICE; } snd_pcm_capture_go(data->pcmHandle); } } else { read_ptr+=bytes_read; len-=bytes_read; } } return ALC_NO_ERROR; } static void CaptureWrapper_Construct(CaptureWrapper *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(CaptureWrapper, ALCbackend, self); self->ExtraData = NULL; } static void CaptureWrapper_Destruct(CaptureWrapper *self) { if(self->ExtraData) qsa_close_capture(self); ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static ALCenum CaptureWrapper_open(CaptureWrapper *self, const ALCchar *name) { return qsa_open_capture(self, name); } static ALCboolean CaptureWrapper_start(CaptureWrapper *self) { qsa_start_capture(self); return ALC_TRUE; } static void CaptureWrapper_stop(CaptureWrapper *self) { qsa_stop_capture(self); } static ALCenum CaptureWrapper_captureSamples(CaptureWrapper *self, void *buffer, ALCuint samples) { return qsa_capture_samples(self, buffer, samples); } static ALCuint CaptureWrapper_availableSamples(CaptureWrapper *self) { return qsa_available_samples(self); } typedef struct ALCqsaBackendFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } ALCqsaBackendFactory; #define ALCQSABACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCqsaBackendFactory, ALCbackendFactory) } } static ALCboolean ALCqsaBackendFactory_init(ALCqsaBackendFactory* UNUSED(self)); static void ALCqsaBackendFactory_deinit(ALCqsaBackendFactory* UNUSED(self)); static ALCboolean ALCqsaBackendFactory_querySupport(ALCqsaBackendFactory* UNUSED(self), ALCbackend_Type type); static void ALCqsaBackendFactory_probe(ALCqsaBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames); static ALCbackend* ALCqsaBackendFactory_createBackend(ALCqsaBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type); DEFINE_ALCBACKENDFACTORY_VTABLE(ALCqsaBackendFactory); static ALCboolean ALCqsaBackendFactory_init(ALCqsaBackendFactory* UNUSED(self)) { return ALC_TRUE; } static void ALCqsaBackendFactory_deinit(ALCqsaBackendFactory* UNUSED(self)) { #define FREE_NAME(iter) free((iter)->name) VECTOR_FOR_EACH(DevMap, DeviceNameMap, FREE_NAME); VECTOR_DEINIT(DeviceNameMap); VECTOR_FOR_EACH(DevMap, CaptureNameMap, FREE_NAME); VECTOR_DEINIT(CaptureNameMap); #undef FREE_NAME } static ALCboolean ALCqsaBackendFactory_querySupport(ALCqsaBackendFactory* UNUSED(self), ALCbackend_Type type) { if(type == ALCbackend_Playback || type == ALCbackend_Capture) return ALC_TRUE; return ALC_FALSE; } static void ALCqsaBackendFactory_probe(ALCqsaBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames) { switch (type) { #define APPEND_OUTNAME(e) do { \ const char *n_ = (e)->name; \ if(n_ && n_[0]) \ alstr_append_range(outnames, n_, n_+strlen(n_)+1); \ } while(0) case ALL_DEVICE_PROBE: deviceList(SND_PCM_CHANNEL_PLAYBACK, &DeviceNameMap); VECTOR_FOR_EACH(const DevMap, DeviceNameMap, APPEND_OUTNAME); break; case CAPTURE_DEVICE_PROBE: deviceList(SND_PCM_CHANNEL_CAPTURE, &CaptureNameMap); VECTOR_FOR_EACH(const DevMap, CaptureNameMap, APPEND_OUTNAME); break; #undef APPEND_OUTNAME } } static ALCbackend* ALCqsaBackendFactory_createBackend(ALCqsaBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) { if(type == ALCbackend_Playback) { PlaybackWrapper *backend; NEW_OBJ(backend, PlaybackWrapper)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } if(type == ALCbackend_Capture) { CaptureWrapper *backend; NEW_OBJ(backend, CaptureWrapper)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } return NULL; } ALCbackendFactory *ALCqsaBackendFactory_getFactory(void) { static ALCqsaBackendFactory factory = ALCQSABACKENDFACTORY_INITIALIZER; return STATIC_CAST(ALCbackendFactory, &factory); } openal-soft-openal-soft-1.19.1/Alc/backends/sdl2.c000066400000000000000000000230731335774445300215640ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2018 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include "alMain.h" #include "alu.h" #include "threads.h" #include "compat.h" #include "backends/base.h" #ifdef _WIN32 #define DEVNAME_PREFIX "OpenAL Soft on " #else #define DEVNAME_PREFIX "" #endif typedef struct ALCsdl2Backend { DERIVE_FROM_TYPE(ALCbackend); SDL_AudioDeviceID deviceID; ALsizei frameSize; ALuint Frequency; enum DevFmtChannels FmtChans; enum DevFmtType FmtType; ALuint UpdateSize; } ALCsdl2Backend; static void ALCsdl2Backend_Construct(ALCsdl2Backend *self, ALCdevice *device); static void ALCsdl2Backend_Destruct(ALCsdl2Backend *self); static ALCenum ALCsdl2Backend_open(ALCsdl2Backend *self, const ALCchar *name); static ALCboolean ALCsdl2Backend_reset(ALCsdl2Backend *self); static ALCboolean ALCsdl2Backend_start(ALCsdl2Backend *self); static void ALCsdl2Backend_stop(ALCsdl2Backend *self); static DECLARE_FORWARD2(ALCsdl2Backend, ALCbackend, ALCenum, captureSamples, void*, ALCuint) static DECLARE_FORWARD(ALCsdl2Backend, ALCbackend, ALCuint, availableSamples) static DECLARE_FORWARD(ALCsdl2Backend, ALCbackend, ClockLatency, getClockLatency) static void ALCsdl2Backend_lock(ALCsdl2Backend *self); static void ALCsdl2Backend_unlock(ALCsdl2Backend *self); DECLARE_DEFAULT_ALLOCATORS(ALCsdl2Backend) DEFINE_ALCBACKEND_VTABLE(ALCsdl2Backend); static const ALCchar defaultDeviceName[] = DEVNAME_PREFIX "Default Device"; static void ALCsdl2Backend_Construct(ALCsdl2Backend *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCsdl2Backend, ALCbackend, self); self->deviceID = 0; self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); self->Frequency = device->Frequency; self->FmtChans = device->FmtChans; self->FmtType = device->FmtType; self->UpdateSize = device->UpdateSize; } static void ALCsdl2Backend_Destruct(ALCsdl2Backend *self) { if(self->deviceID) SDL_CloseAudioDevice(self->deviceID); self->deviceID = 0; ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static void ALCsdl2Backend_audioCallback(void *ptr, Uint8 *stream, int len) { ALCsdl2Backend *self = (ALCsdl2Backend*)ptr; ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; assert((len % self->frameSize) == 0); aluMixData(device, stream, len / self->frameSize); } static ALCenum ALCsdl2Backend_open(ALCsdl2Backend *self, const ALCchar *name) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; SDL_AudioSpec want, have; SDL_zero(want); SDL_zero(have); want.freq = device->Frequency; switch(device->FmtType) { case DevFmtUByte: want.format = AUDIO_U8; break; case DevFmtByte: want.format = AUDIO_S8; break; case DevFmtUShort: want.format = AUDIO_U16SYS; break; case DevFmtShort: want.format = AUDIO_S16SYS; break; case DevFmtUInt: /* fall-through */ case DevFmtInt: want.format = AUDIO_S32SYS; break; case DevFmtFloat: want.format = AUDIO_F32; break; } want.channels = (device->FmtChans == DevFmtMono) ? 1 : 2; want.samples = device->UpdateSize; want.callback = ALCsdl2Backend_audioCallback; want.userdata = self; /* Passing NULL to SDL_OpenAudioDevice opens a default, which isn't * necessarily the first in the list. */ if(!name || strcmp(name, defaultDeviceName) == 0) self->deviceID = SDL_OpenAudioDevice(NULL, SDL_FALSE, &want, &have, SDL_AUDIO_ALLOW_ANY_CHANGE); else { const size_t prefix_len = strlen(DEVNAME_PREFIX); if(strncmp(name, DEVNAME_PREFIX, prefix_len) == 0) self->deviceID = SDL_OpenAudioDevice(name+prefix_len, SDL_FALSE, &want, &have, SDL_AUDIO_ALLOW_ANY_CHANGE); else self->deviceID = SDL_OpenAudioDevice(name, SDL_FALSE, &want, &have, SDL_AUDIO_ALLOW_ANY_CHANGE); } if(self->deviceID == 0) return ALC_INVALID_VALUE; device->Frequency = have.freq; if(have.channels == 1) device->FmtChans = DevFmtMono; else if(have.channels == 2) device->FmtChans = DevFmtStereo; else { ERR("Got unhandled SDL channel count: %d\n", (int)have.channels); return ALC_INVALID_VALUE; } switch(have.format) { case AUDIO_U8: device->FmtType = DevFmtUByte; break; case AUDIO_S8: device->FmtType = DevFmtByte; break; case AUDIO_U16SYS: device->FmtType = DevFmtUShort; break; case AUDIO_S16SYS: device->FmtType = DevFmtShort; break; case AUDIO_S32SYS: device->FmtType = DevFmtInt; break; case AUDIO_F32SYS: device->FmtType = DevFmtFloat; break; default: ERR("Got unsupported SDL format: 0x%04x\n", have.format); return ALC_INVALID_VALUE; } device->UpdateSize = have.samples; device->NumUpdates = 2; /* SDL always (tries to) use two periods. */ self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); self->Frequency = device->Frequency; self->FmtChans = device->FmtChans; self->FmtType = device->FmtType; self->UpdateSize = device->UpdateSize; alstr_copy_cstr(&device->DeviceName, name ? name : defaultDeviceName); return ALC_NO_ERROR; } static ALCboolean ALCsdl2Backend_reset(ALCsdl2Backend *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; device->Frequency = self->Frequency; device->FmtChans = self->FmtChans; device->FmtType = self->FmtType; device->UpdateSize = self->UpdateSize; device->NumUpdates = 2; SetDefaultWFXChannelOrder(device); return ALC_TRUE; } static ALCboolean ALCsdl2Backend_start(ALCsdl2Backend *self) { SDL_PauseAudioDevice(self->deviceID, 0); return ALC_TRUE; } static void ALCsdl2Backend_stop(ALCsdl2Backend *self) { SDL_PauseAudioDevice(self->deviceID, 1); } static void ALCsdl2Backend_lock(ALCsdl2Backend *self) { SDL_LockAudioDevice(self->deviceID); } static void ALCsdl2Backend_unlock(ALCsdl2Backend *self) { SDL_UnlockAudioDevice(self->deviceID); } typedef struct ALCsdl2BackendFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } ALCsdl2BackendFactory; #define ALCsdl2BACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCsdl2BackendFactory, ALCbackendFactory) } } ALCbackendFactory *ALCsdl2BackendFactory_getFactory(void); static ALCboolean ALCsdl2BackendFactory_init(ALCsdl2BackendFactory *self); static void ALCsdl2BackendFactory_deinit(ALCsdl2BackendFactory *self); static ALCboolean ALCsdl2BackendFactory_querySupport(ALCsdl2BackendFactory *self, ALCbackend_Type type); static void ALCsdl2BackendFactory_probe(ALCsdl2BackendFactory *self, enum DevProbe type, al_string *outnames); static ALCbackend* ALCsdl2BackendFactory_createBackend(ALCsdl2BackendFactory *self, ALCdevice *device, ALCbackend_Type type); DEFINE_ALCBACKENDFACTORY_VTABLE(ALCsdl2BackendFactory); ALCbackendFactory *ALCsdl2BackendFactory_getFactory(void) { static ALCsdl2BackendFactory factory = ALCsdl2BACKENDFACTORY_INITIALIZER; return STATIC_CAST(ALCbackendFactory, &factory); } static ALCboolean ALCsdl2BackendFactory_init(ALCsdl2BackendFactory* UNUSED(self)) { if(SDL_InitSubSystem(SDL_INIT_AUDIO) == 0) return AL_TRUE; return ALC_FALSE; } static void ALCsdl2BackendFactory_deinit(ALCsdl2BackendFactory* UNUSED(self)) { SDL_QuitSubSystem(SDL_INIT_AUDIO); } static ALCboolean ALCsdl2BackendFactory_querySupport(ALCsdl2BackendFactory* UNUSED(self), ALCbackend_Type type) { if(type == ALCbackend_Playback) return ALC_TRUE; return ALC_FALSE; } static void ALCsdl2BackendFactory_probe(ALCsdl2BackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames) { int num_devices, i; al_string name; if(type != ALL_DEVICE_PROBE) return; AL_STRING_INIT(name); num_devices = SDL_GetNumAudioDevices(SDL_FALSE); alstr_append_range(outnames, defaultDeviceName, defaultDeviceName+sizeof(defaultDeviceName)); for(i = 0;i < num_devices;++i) { alstr_copy_cstr(&name, DEVNAME_PREFIX); alstr_append_cstr(&name, SDL_GetAudioDeviceName(i, SDL_FALSE)); if(!alstr_empty(name)) alstr_append_range(outnames, VECTOR_BEGIN(name), VECTOR_END(name)+1); } alstr_reset(&name); } static ALCbackend* ALCsdl2BackendFactory_createBackend(ALCsdl2BackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) { if(type == ALCbackend_Playback) { ALCsdl2Backend *backend; NEW_OBJ(backend, ALCsdl2Backend)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } return NULL; } openal-soft-openal-soft-1.19.1/Alc/backends/sndio.c000066400000000000000000000426511335774445300220370ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include "alMain.h" #include "alu.h" #include "threads.h" #include "ringbuffer.h" #include "backends/base.h" #include static const ALCchar sndio_device[] = "SndIO Default"; typedef struct SndioPlayback { DERIVE_FROM_TYPE(ALCbackend); struct sio_hdl *sndHandle; ALvoid *mix_data; ALsizei data_size; ATOMIC(int) killNow; althrd_t thread; } SndioPlayback; static int SndioPlayback_mixerProc(void *ptr); static void SndioPlayback_Construct(SndioPlayback *self, ALCdevice *device); static void SndioPlayback_Destruct(SndioPlayback *self); static ALCenum SndioPlayback_open(SndioPlayback *self, const ALCchar *name); static ALCboolean SndioPlayback_reset(SndioPlayback *self); static ALCboolean SndioPlayback_start(SndioPlayback *self); static void SndioPlayback_stop(SndioPlayback *self); static DECLARE_FORWARD2(SndioPlayback, ALCbackend, ALCenum, captureSamples, void*, ALCuint) static DECLARE_FORWARD(SndioPlayback, ALCbackend, ALCuint, availableSamples) static DECLARE_FORWARD(SndioPlayback, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(SndioPlayback, ALCbackend, void, lock) static DECLARE_FORWARD(SndioPlayback, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(SndioPlayback) DEFINE_ALCBACKEND_VTABLE(SndioPlayback); static void SndioPlayback_Construct(SndioPlayback *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(SndioPlayback, ALCbackend, self); self->sndHandle = NULL; self->mix_data = NULL; ATOMIC_INIT(&self->killNow, AL_TRUE); } static void SndioPlayback_Destruct(SndioPlayback *self) { if(self->sndHandle) sio_close(self->sndHandle); self->sndHandle = NULL; al_free(self->mix_data); self->mix_data = NULL; ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static int SndioPlayback_mixerProc(void *ptr) { SndioPlayback *self = (SndioPlayback*)ptr; ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; ALsizei frameSize; size_t wrote; SetRTPriority(); althrd_setname(althrd_current(), MIXER_THREAD_NAME); frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) && ATOMIC_LOAD(&device->Connected, almemory_order_acquire)) { ALsizei len = self->data_size; ALubyte *WritePtr = self->mix_data; SndioPlayback_lock(self); aluMixData(device, WritePtr, len/frameSize); SndioPlayback_unlock(self); while(len > 0 && !ATOMIC_LOAD(&self->killNow, almemory_order_acquire)) { wrote = sio_write(self->sndHandle, WritePtr, len); if(wrote == 0) { ERR("sio_write failed\n"); ALCdevice_Lock(device); aluHandleDisconnect(device, "Failed to write playback samples"); ALCdevice_Unlock(device); break; } len -= wrote; WritePtr += wrote; } } return 0; } static ALCenum SndioPlayback_open(SndioPlayback *self, const ALCchar *name) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; if(!name) name = sndio_device; else if(strcmp(name, sndio_device) != 0) return ALC_INVALID_VALUE; self->sndHandle = sio_open(NULL, SIO_PLAY, 0); if(self->sndHandle == NULL) { ERR("Could not open device\n"); return ALC_INVALID_VALUE; } alstr_copy_cstr(&device->DeviceName, name); return ALC_NO_ERROR; } static ALCboolean SndioPlayback_reset(SndioPlayback *self) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; struct sio_par par; sio_initpar(&par); par.rate = device->Frequency; par.pchan = ((device->FmtChans != DevFmtMono) ? 2 : 1); switch(device->FmtType) { case DevFmtByte: par.bits = 8; par.sig = 1; break; case DevFmtUByte: par.bits = 8; par.sig = 0; break; case DevFmtFloat: case DevFmtShort: par.bits = 16; par.sig = 1; break; case DevFmtUShort: par.bits = 16; par.sig = 0; break; case DevFmtInt: par.bits = 32; par.sig = 1; break; case DevFmtUInt: par.bits = 32; par.sig = 0; break; } par.le = SIO_LE_NATIVE; par.round = device->UpdateSize; par.appbufsz = device->UpdateSize * (device->NumUpdates-1); if(!par.appbufsz) par.appbufsz = device->UpdateSize; if(!sio_setpar(self->sndHandle, &par) || !sio_getpar(self->sndHandle, &par)) { ERR("Failed to set device parameters\n"); return ALC_FALSE; } if(par.bits != par.bps*8) { ERR("Padded samples not supported (%u of %u bits)\n", par.bits, par.bps*8); return ALC_FALSE; } device->Frequency = par.rate; device->FmtChans = ((par.pchan==1) ? DevFmtMono : DevFmtStereo); if(par.bits == 8 && par.sig == 1) device->FmtType = DevFmtByte; else if(par.bits == 8 && par.sig == 0) device->FmtType = DevFmtUByte; else if(par.bits == 16 && par.sig == 1) device->FmtType = DevFmtShort; else if(par.bits == 16 && par.sig == 0) device->FmtType = DevFmtUShort; else if(par.bits == 32 && par.sig == 1) device->FmtType = DevFmtInt; else if(par.bits == 32 && par.sig == 0) device->FmtType = DevFmtUInt; else { ERR("Unhandled sample format: %s %u-bit\n", (par.sig?"signed":"unsigned"), par.bits); return ALC_FALSE; } device->UpdateSize = par.round; device->NumUpdates = (par.bufsz/par.round) + 1; SetDefaultChannelOrder(device); return ALC_TRUE; } static ALCboolean SndioPlayback_start(SndioPlayback *self) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; self->data_size = device->UpdateSize * FrameSizeFromDevFmt( device->FmtChans, device->FmtType, device->AmbiOrder ); al_free(self->mix_data); self->mix_data = al_calloc(16, self->data_size); if(!sio_start(self->sndHandle)) { ERR("Error starting playback\n"); return ALC_FALSE; } ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release); if(althrd_create(&self->thread, SndioPlayback_mixerProc, self) != althrd_success) { sio_stop(self->sndHandle); return ALC_FALSE; } return ALC_TRUE; } static void SndioPlayback_stop(SndioPlayback *self) { int res; if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel)) return; althrd_join(self->thread, &res); if(!sio_stop(self->sndHandle)) ERR("Error stopping device\n"); al_free(self->mix_data); self->mix_data = NULL; } typedef struct SndioCapture { DERIVE_FROM_TYPE(ALCbackend); struct sio_hdl *sndHandle; ll_ringbuffer_t *ring; ATOMIC(int) killNow; althrd_t thread; } SndioCapture; static int SndioCapture_recordProc(void *ptr); static void SndioCapture_Construct(SndioCapture *self, ALCdevice *device); static void SndioCapture_Destruct(SndioCapture *self); static ALCenum SndioCapture_open(SndioCapture *self, const ALCchar *name); static DECLARE_FORWARD(SndioCapture, ALCbackend, ALCboolean, reset) static ALCboolean SndioCapture_start(SndioCapture *self); static void SndioCapture_stop(SndioCapture *self); static ALCenum SndioCapture_captureSamples(SndioCapture *self, void *buffer, ALCuint samples); static ALCuint SndioCapture_availableSamples(SndioCapture *self); static DECLARE_FORWARD(SndioCapture, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(SndioCapture, ALCbackend, void, lock) static DECLARE_FORWARD(SndioCapture, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(SndioCapture) DEFINE_ALCBACKEND_VTABLE(SndioCapture); static void SndioCapture_Construct(SndioCapture *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(SndioCapture, ALCbackend, self); self->sndHandle = NULL; self->ring = NULL; ATOMIC_INIT(&self->killNow, AL_TRUE); } static void SndioCapture_Destruct(SndioCapture *self) { if(self->sndHandle) sio_close(self->sndHandle); self->sndHandle = NULL; ll_ringbuffer_free(self->ring); self->ring = NULL; ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static int SndioCapture_recordProc(void* ptr) { SndioCapture *self = (SndioCapture*)ptr; ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; ALsizei frameSize; SetRTPriority(); althrd_setname(althrd_current(), RECORD_THREAD_NAME); frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) && ATOMIC_LOAD(&device->Connected, almemory_order_acquire)) { ll_ringbuffer_data_t data[2]; size_t total, todo; ll_ringbuffer_get_write_vector(self->ring, data); todo = data[0].len + data[1].len; if(todo == 0) { static char junk[4096]; sio_read(self->sndHandle, junk, minz(sizeof(junk)/frameSize, device->UpdateSize)*frameSize); continue; } total = 0; data[0].len *= frameSize; data[1].len *= frameSize; todo = minz(todo, device->UpdateSize) * frameSize; while(total < todo) { size_t got; if(!data[0].len) data[0] = data[1]; got = sio_read(self->sndHandle, data[0].buf, minz(todo-total, data[0].len)); if(!got) { SndioCapture_lock(self); aluHandleDisconnect(device, "Failed to read capture samples"); SndioCapture_unlock(self); break; } data[0].buf += got; data[0].len -= got; total += got; } ll_ringbuffer_write_advance(self->ring, total / frameSize); } return 0; } static ALCenum SndioCapture_open(SndioCapture *self, const ALCchar *name) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; struct sio_par par; if(!name) name = sndio_device; else if(strcmp(name, sndio_device) != 0) return ALC_INVALID_VALUE; self->sndHandle = sio_open(NULL, SIO_REC, 0); if(self->sndHandle == NULL) { ERR("Could not open device\n"); return ALC_INVALID_VALUE; } sio_initpar(&par); switch(device->FmtType) { case DevFmtByte: par.bps = 1; par.sig = 1; break; case DevFmtUByte: par.bps = 1; par.sig = 0; break; case DevFmtShort: par.bps = 2; par.sig = 1; break; case DevFmtUShort: par.bps = 2; par.sig = 0; break; case DevFmtInt: par.bps = 4; par.sig = 1; break; case DevFmtUInt: par.bps = 4; par.sig = 0; break; case DevFmtFloat: ERR("%s capture samples not supported\n", DevFmtTypeString(device->FmtType)); return ALC_INVALID_VALUE; } par.bits = par.bps * 8; par.le = SIO_LE_NATIVE; par.msb = SIO_LE_NATIVE ? 0 : 1; par.rchan = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); par.rate = device->Frequency; par.appbufsz = maxu(device->UpdateSize*device->NumUpdates, (device->Frequency+9)/10); par.round = clampu(par.appbufsz/device->NumUpdates, (device->Frequency+99)/100, (device->Frequency+19)/20); device->UpdateSize = par.round; device->NumUpdates = maxu(par.appbufsz/par.round, 1); if(!sio_setpar(self->sndHandle, &par) || !sio_getpar(self->sndHandle, &par)) { ERR("Failed to set device parameters\n"); return ALC_INVALID_VALUE; } if(par.bits != par.bps*8) { ERR("Padded samples not supported (%u of %u bits)\n", par.bits, par.bps*8); return ALC_INVALID_VALUE; } if(!((device->FmtType == DevFmtByte && par.bits == 8 && par.sig != 0) || (device->FmtType == DevFmtUByte && par.bits == 8 && par.sig == 0) || (device->FmtType == DevFmtShort && par.bits == 16 && par.sig != 0) || (device->FmtType == DevFmtUShort && par.bits == 16 && par.sig == 0) || (device->FmtType == DevFmtInt && par.bits == 32 && par.sig != 0) || (device->FmtType == DevFmtUInt && par.bits == 32 && par.sig == 0)) || ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder) != (ALsizei)par.rchan || device->Frequency != par.rate) { ERR("Failed to set format %s %s %uhz, got %c%u %u-channel %uhz instead\n", DevFmtTypeString(device->FmtType), DevFmtChannelsString(device->FmtChans), device->Frequency, par.sig?'s':'u', par.bits, par.rchan, par.rate); return ALC_INVALID_VALUE; } self->ring = ll_ringbuffer_create(device->UpdateSize*device->NumUpdates, par.bps*par.rchan, 0); if(!self->ring) { ERR("Failed to allocate %u-byte ringbuffer\n", device->UpdateSize*device->NumUpdates*par.bps*par.rchan); return ALC_OUT_OF_MEMORY; } SetDefaultChannelOrder(device); alstr_copy_cstr(&device->DeviceName, name); return ALC_NO_ERROR; } static ALCboolean SndioCapture_start(SndioCapture *self) { if(!sio_start(self->sndHandle)) { ERR("Error starting playback\n"); return ALC_FALSE; } ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release); if(althrd_create(&self->thread, SndioCapture_recordProc, self) != althrd_success) { sio_stop(self->sndHandle); return ALC_FALSE; } return ALC_TRUE; } static void SndioCapture_stop(SndioCapture *self) { int res; if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel)) return; althrd_join(self->thread, &res); if(!sio_stop(self->sndHandle)) ERR("Error stopping device\n"); } static ALCenum SndioCapture_captureSamples(SndioCapture *self, void *buffer, ALCuint samples) { ll_ringbuffer_read(self->ring, buffer, samples); return ALC_NO_ERROR; } static ALCuint SndioCapture_availableSamples(SndioCapture *self) { return ll_ringbuffer_read_space(self->ring); } typedef struct SndioBackendFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } SndioBackendFactory; #define SNDIOBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(SndioBackendFactory, ALCbackendFactory) } } ALCbackendFactory *SndioBackendFactory_getFactory(void); static ALCboolean SndioBackendFactory_init(SndioBackendFactory *self); static DECLARE_FORWARD(SndioBackendFactory, ALCbackendFactory, void, deinit) static ALCboolean SndioBackendFactory_querySupport(SndioBackendFactory *self, ALCbackend_Type type); static void SndioBackendFactory_probe(SndioBackendFactory *self, enum DevProbe type, al_string *outnames); static ALCbackend* SndioBackendFactory_createBackend(SndioBackendFactory *self, ALCdevice *device, ALCbackend_Type type); DEFINE_ALCBACKENDFACTORY_VTABLE(SndioBackendFactory); ALCbackendFactory *SndioBackendFactory_getFactory(void) { static SndioBackendFactory factory = SNDIOBACKENDFACTORY_INITIALIZER; return STATIC_CAST(ALCbackendFactory, &factory); } static ALCboolean SndioBackendFactory_init(SndioBackendFactory* UNUSED(self)) { /* No dynamic loading */ return ALC_TRUE; } static ALCboolean SndioBackendFactory_querySupport(SndioBackendFactory* UNUSED(self), ALCbackend_Type type) { if(type == ALCbackend_Playback || type == ALCbackend_Capture) return ALC_TRUE; return ALC_FALSE; } static void SndioBackendFactory_probe(SndioBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames) { switch(type) { case ALL_DEVICE_PROBE: case CAPTURE_DEVICE_PROBE: alstr_append_range(outnames, sndio_device, sndio_device+sizeof(sndio_device)); break; } } static ALCbackend* SndioBackendFactory_createBackend(SndioBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) { if(type == ALCbackend_Playback) { SndioPlayback *backend; NEW_OBJ(backend, SndioPlayback)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } if(type == ALCbackend_Capture) { SndioCapture *backend; NEW_OBJ(backend, SndioCapture)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } return NULL; } openal-soft-openal-soft-1.19.1/Alc/backends/solaris.c000066400000000000000000000260731335774445300223770ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include #include #include #include #include #include #include #include #include "alMain.h" #include "alu.h" #include "alconfig.h" #include "threads.h" #include "compat.h" #include "backends/base.h" #include typedef struct ALCsolarisBackend { DERIVE_FROM_TYPE(ALCbackend); int fd; ALubyte *mix_data; int data_size; ATOMIC(ALenum) killNow; althrd_t thread; } ALCsolarisBackend; static int ALCsolarisBackend_mixerProc(void *ptr); static void ALCsolarisBackend_Construct(ALCsolarisBackend *self, ALCdevice *device); static void ALCsolarisBackend_Destruct(ALCsolarisBackend *self); static ALCenum ALCsolarisBackend_open(ALCsolarisBackend *self, const ALCchar *name); static ALCboolean ALCsolarisBackend_reset(ALCsolarisBackend *self); static ALCboolean ALCsolarisBackend_start(ALCsolarisBackend *self); static void ALCsolarisBackend_stop(ALCsolarisBackend *self); static DECLARE_FORWARD2(ALCsolarisBackend, ALCbackend, ALCenum, captureSamples, void*, ALCuint) static DECLARE_FORWARD(ALCsolarisBackend, ALCbackend, ALCuint, availableSamples) static DECLARE_FORWARD(ALCsolarisBackend, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(ALCsolarisBackend, ALCbackend, void, lock) static DECLARE_FORWARD(ALCsolarisBackend, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCsolarisBackend) DEFINE_ALCBACKEND_VTABLE(ALCsolarisBackend); static const ALCchar solaris_device[] = "Solaris Default"; static const char *solaris_driver = "/dev/audio"; static void ALCsolarisBackend_Construct(ALCsolarisBackend *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCsolarisBackend, ALCbackend, self); self->fd = -1; self->mix_data = NULL; ATOMIC_INIT(&self->killNow, AL_FALSE); } static void ALCsolarisBackend_Destruct(ALCsolarisBackend *self) { if(self->fd != -1) close(self->fd); self->fd = -1; free(self->mix_data); self->mix_data = NULL; self->data_size = 0; ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static int ALCsolarisBackend_mixerProc(void *ptr) { ALCsolarisBackend *self = ptr; ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; struct timeval timeout; ALubyte *write_ptr; ALint frame_size; ALint to_write; ssize_t wrote; fd_set wfds; int sret; SetRTPriority(); althrd_setname(althrd_current(), MIXER_THREAD_NAME); frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); ALCsolarisBackend_lock(self); while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) && ATOMIC_LOAD(&device->Connected, almemory_order_acquire)) { FD_ZERO(&wfds); FD_SET(self->fd, &wfds); timeout.tv_sec = 1; timeout.tv_usec = 0; ALCsolarisBackend_unlock(self); sret = select(self->fd+1, NULL, &wfds, NULL, &timeout); ALCsolarisBackend_lock(self); if(sret < 0) { if(errno == EINTR) continue; ERR("select failed: %s\n", strerror(errno)); aluHandleDisconnect(device, "Failed to wait for playback buffer: %s", strerror(errno)); break; } else if(sret == 0) { WARN("select timeout\n"); continue; } write_ptr = self->mix_data; to_write = self->data_size; aluMixData(device, write_ptr, to_write/frame_size); while(to_write > 0 && !ATOMIC_LOAD_SEQ(&self->killNow)) { wrote = write(self->fd, write_ptr, to_write); if(wrote < 0) { if(errno == EAGAIN || errno == EWOULDBLOCK || errno == EINTR) continue; ERR("write failed: %s\n", strerror(errno)); aluHandleDisconnect(device, "Failed to write playback samples: %s", strerror(errno)); break; } to_write -= wrote; write_ptr += wrote; } } ALCsolarisBackend_unlock(self); return 0; } static ALCenum ALCsolarisBackend_open(ALCsolarisBackend *self, const ALCchar *name) { ALCdevice *device; if(!name) name = solaris_device; else if(strcmp(name, solaris_device) != 0) return ALC_INVALID_VALUE; self->fd = open(solaris_driver, O_WRONLY); if(self->fd == -1) { ERR("Could not open %s: %s\n", solaris_driver, strerror(errno)); return ALC_INVALID_VALUE; } device = STATIC_CAST(ALCbackend,self)->mDevice; alstr_copy_cstr(&device->DeviceName, name); return ALC_NO_ERROR; } static ALCboolean ALCsolarisBackend_reset(ALCsolarisBackend *self) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; audio_info_t info; ALsizei frameSize; ALsizei numChannels; AUDIO_INITINFO(&info); info.play.sample_rate = device->Frequency; if(device->FmtChans != DevFmtMono) device->FmtChans = DevFmtStereo; numChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); info.play.channels = numChannels; switch(device->FmtType) { case DevFmtByte: info.play.precision = 8; info.play.encoding = AUDIO_ENCODING_LINEAR; break; case DevFmtUByte: info.play.precision = 8; info.play.encoding = AUDIO_ENCODING_LINEAR8; break; case DevFmtUShort: case DevFmtInt: case DevFmtUInt: case DevFmtFloat: device->FmtType = DevFmtShort; /* fall-through */ case DevFmtShort: info.play.precision = 16; info.play.encoding = AUDIO_ENCODING_LINEAR; break; } frameSize = numChannels * BytesFromDevFmt(device->FmtType); info.play.buffer_size = device->UpdateSize*device->NumUpdates * frameSize; if(ioctl(self->fd, AUDIO_SETINFO, &info) < 0) { ERR("ioctl failed: %s\n", strerror(errno)); return ALC_FALSE; } if(ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder) != (ALsizei)info.play.channels) { ERR("Failed to set %s, got %u channels instead\n", DevFmtChannelsString(device->FmtChans), info.play.channels); return ALC_FALSE; } if(!((info.play.precision == 8 && info.play.encoding == AUDIO_ENCODING_LINEAR8 && device->FmtType == DevFmtUByte) || (info.play.precision == 8 && info.play.encoding == AUDIO_ENCODING_LINEAR && device->FmtType == DevFmtByte) || (info.play.precision == 16 && info.play.encoding == AUDIO_ENCODING_LINEAR && device->FmtType == DevFmtShort) || (info.play.precision == 32 && info.play.encoding == AUDIO_ENCODING_LINEAR && device->FmtType == DevFmtInt))) { ERR("Could not set %s samples, got %d (0x%x)\n", DevFmtTypeString(device->FmtType), info.play.precision, info.play.encoding); return ALC_FALSE; } device->Frequency = info.play.sample_rate; device->UpdateSize = (info.play.buffer_size/device->NumUpdates) + 1; SetDefaultChannelOrder(device); free(self->mix_data); self->data_size = device->UpdateSize * FrameSizeFromDevFmt( device->FmtChans, device->FmtType, device->AmbiOrder ); self->mix_data = calloc(1, self->data_size); return ALC_TRUE; } static ALCboolean ALCsolarisBackend_start(ALCsolarisBackend *self) { ATOMIC_STORE_SEQ(&self->killNow, AL_FALSE); if(althrd_create(&self->thread, ALCsolarisBackend_mixerProc, self) != althrd_success) return ALC_FALSE; return ALC_TRUE; } static void ALCsolarisBackend_stop(ALCsolarisBackend *self) { int res; if(ATOMIC_EXCHANGE_SEQ(&self->killNow, AL_TRUE)) return; althrd_join(self->thread, &res); if(ioctl(self->fd, AUDIO_DRAIN) < 0) ERR("Error draining device: %s\n", strerror(errno)); } typedef struct ALCsolarisBackendFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } ALCsolarisBackendFactory; #define ALCSOLARISBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCsolarisBackendFactory, ALCbackendFactory) } } ALCbackendFactory *ALCsolarisBackendFactory_getFactory(void); static ALCboolean ALCsolarisBackendFactory_init(ALCsolarisBackendFactory *self); static DECLARE_FORWARD(ALCsolarisBackendFactory, ALCbackendFactory, void, deinit) static ALCboolean ALCsolarisBackendFactory_querySupport(ALCsolarisBackendFactory *self, ALCbackend_Type type); static void ALCsolarisBackendFactory_probe(ALCsolarisBackendFactory *self, enum DevProbe type, al_string *outnames); static ALCbackend* ALCsolarisBackendFactory_createBackend(ALCsolarisBackendFactory *self, ALCdevice *device, ALCbackend_Type type); DEFINE_ALCBACKENDFACTORY_VTABLE(ALCsolarisBackendFactory); ALCbackendFactory *ALCsolarisBackendFactory_getFactory(void) { static ALCsolarisBackendFactory factory = ALCSOLARISBACKENDFACTORY_INITIALIZER; return STATIC_CAST(ALCbackendFactory, &factory); } static ALCboolean ALCsolarisBackendFactory_init(ALCsolarisBackendFactory* UNUSED(self)) { ConfigValueStr(NULL, "solaris", "device", &solaris_driver); return ALC_TRUE; } static ALCboolean ALCsolarisBackendFactory_querySupport(ALCsolarisBackendFactory* UNUSED(self), ALCbackend_Type type) { if(type == ALCbackend_Playback) return ALC_TRUE; return ALC_FALSE; } static void ALCsolarisBackendFactory_probe(ALCsolarisBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames) { switch(type) { case ALL_DEVICE_PROBE: { #ifdef HAVE_STAT struct stat buf; if(stat(solaris_driver, &buf) == 0) #endif alstr_append_range(outnames, solaris_device, solaris_device+sizeof(solaris_device)); } break; case CAPTURE_DEVICE_PROBE: break; } } ALCbackend* ALCsolarisBackendFactory_createBackend(ALCsolarisBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) { if(type == ALCbackend_Playback) { ALCsolarisBackend *backend; NEW_OBJ(backend, ALCsolarisBackend)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } return NULL; } openal-soft-openal-soft-1.19.1/Alc/backends/wasapi.c000066400000000000000000002026751335774445300222130ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2011 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #define COBJMACROS #include #include #include #include #include #include #include #include #include #include #include #include #ifndef _WAVEFORMATEXTENSIBLE_ #include #include #endif #include "alMain.h" #include "alu.h" #include "ringbuffer.h" #include "threads.h" #include "compat.h" #include "alstring.h" #include "converter.h" #include "backends/base.h" DEFINE_GUID(KSDATAFORMAT_SUBTYPE_PCM, 0x00000001, 0x0000, 0x0010, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71); DEFINE_GUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, 0x00000003, 0x0000, 0x0010, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71); DEFINE_DEVPROPKEY(DEVPKEY_Device_FriendlyName, 0xa45c254e, 0xdf1c, 0x4efd, 0x80,0x20, 0x67,0xd1,0x46,0xa8,0x50,0xe0, 14); DEFINE_PROPERTYKEY(PKEY_AudioEndpoint_FormFactor, 0x1da5d803, 0xd492, 0x4edd, 0x8c,0x23, 0xe0,0xc0,0xff,0xee,0x7f,0x0e, 0); DEFINE_PROPERTYKEY(PKEY_AudioEndpoint_GUID, 0x1da5d803, 0xd492, 0x4edd, 0x8c, 0x23,0xe0, 0xc0,0xff,0xee,0x7f,0x0e, 4 ); #define MONO SPEAKER_FRONT_CENTER #define STEREO (SPEAKER_FRONT_LEFT|SPEAKER_FRONT_RIGHT) #define QUAD (SPEAKER_FRONT_LEFT|SPEAKER_FRONT_RIGHT|SPEAKER_BACK_LEFT|SPEAKER_BACK_RIGHT) #define X5DOT1 (SPEAKER_FRONT_LEFT|SPEAKER_FRONT_RIGHT|SPEAKER_FRONT_CENTER|SPEAKER_LOW_FREQUENCY|SPEAKER_SIDE_LEFT|SPEAKER_SIDE_RIGHT) #define X5DOT1REAR (SPEAKER_FRONT_LEFT|SPEAKER_FRONT_RIGHT|SPEAKER_FRONT_CENTER|SPEAKER_LOW_FREQUENCY|SPEAKER_BACK_LEFT|SPEAKER_BACK_RIGHT) #define X6DOT1 (SPEAKER_FRONT_LEFT|SPEAKER_FRONT_RIGHT|SPEAKER_FRONT_CENTER|SPEAKER_LOW_FREQUENCY|SPEAKER_BACK_CENTER|SPEAKER_SIDE_LEFT|SPEAKER_SIDE_RIGHT) #define X7DOT1 (SPEAKER_FRONT_LEFT|SPEAKER_FRONT_RIGHT|SPEAKER_FRONT_CENTER|SPEAKER_LOW_FREQUENCY|SPEAKER_BACK_LEFT|SPEAKER_BACK_RIGHT|SPEAKER_SIDE_LEFT|SPEAKER_SIDE_RIGHT) #define X7DOT1_WIDE (SPEAKER_FRONT_LEFT|SPEAKER_FRONT_RIGHT|SPEAKER_FRONT_CENTER|SPEAKER_LOW_FREQUENCY|SPEAKER_BACK_LEFT|SPEAKER_BACK_RIGHT|SPEAKER_FRONT_LEFT_OF_CENTER|SPEAKER_FRONT_RIGHT_OF_CENTER) #define REFTIME_PER_SEC ((REFERENCE_TIME)10000000) #define DEVNAME_HEAD "OpenAL Soft on " /* Scales the given value using 64-bit integer math, ceiling the result. */ static inline ALuint64 ScaleCeil(ALuint64 val, ALuint64 new_scale, ALuint64 old_scale) { return (val*new_scale + old_scale-1) / old_scale; } typedef struct { al_string name; al_string endpoint_guid; // obtained from PKEY_AudioEndpoint_GUID , set to "Unknown device GUID" if absent. WCHAR *devid; } DevMap; TYPEDEF_VECTOR(DevMap, vector_DevMap) static void clear_devlist(vector_DevMap *list) { #define CLEAR_DEVMAP(i) do { \ AL_STRING_DEINIT((i)->name); \ AL_STRING_DEINIT((i)->endpoint_guid); \ free((i)->devid); \ (i)->devid = NULL; \ } while(0) VECTOR_FOR_EACH(DevMap, *list, CLEAR_DEVMAP); VECTOR_RESIZE(*list, 0, 0); #undef CLEAR_DEVMAP } static vector_DevMap PlaybackDevices; static vector_DevMap CaptureDevices; static HANDLE ThreadHdl; static DWORD ThreadID; typedef struct { HANDLE FinishedEvt; HRESULT result; } ThreadRequest; #define WM_USER_First (WM_USER+0) #define WM_USER_OpenDevice (WM_USER+0) #define WM_USER_ResetDevice (WM_USER+1) #define WM_USER_StartDevice (WM_USER+2) #define WM_USER_StopDevice (WM_USER+3) #define WM_USER_CloseDevice (WM_USER+4) #define WM_USER_Enumerate (WM_USER+5) #define WM_USER_Last (WM_USER+5) static const char MessageStr[WM_USER_Last+1-WM_USER][20] = { "Open Device", "Reset Device", "Start Device", "Stop Device", "Close Device", "Enumerate Devices", }; static inline void ReturnMsgResponse(ThreadRequest *req, HRESULT res) { req->result = res; SetEvent(req->FinishedEvt); } static HRESULT WaitForResponse(ThreadRequest *req) { if(WaitForSingleObject(req->FinishedEvt, INFINITE) == WAIT_OBJECT_0) return req->result; ERR("Message response error: %lu\n", GetLastError()); return E_FAIL; } static void get_device_name_and_guid(IMMDevice *device, al_string *name, al_string *guid) { IPropertyStore *ps; PROPVARIANT pvname; PROPVARIANT pvguid; HRESULT hr; alstr_copy_cstr(name, DEVNAME_HEAD); hr = IMMDevice_OpenPropertyStore(device, STGM_READ, &ps); if(FAILED(hr)) { WARN("OpenPropertyStore failed: 0x%08lx\n", hr); alstr_append_cstr(name, "Unknown Device Name"); if(guid!=NULL)alstr_copy_cstr(guid, "Unknown Device GUID"); return; } PropVariantInit(&pvname); hr = IPropertyStore_GetValue(ps, (const PROPERTYKEY*)&DEVPKEY_Device_FriendlyName, &pvname); if(FAILED(hr)) { WARN("GetValue Device_FriendlyName failed: 0x%08lx\n", hr); alstr_append_cstr(name, "Unknown Device Name"); } else if(pvname.vt == VT_LPWSTR) alstr_append_wcstr(name, pvname.pwszVal); else { WARN("Unexpected PROPVARIANT type: 0x%04x\n", pvname.vt); alstr_append_cstr(name, "Unknown Device Name"); } PropVariantClear(&pvname); if(guid!=NULL){ PropVariantInit(&pvguid); hr = IPropertyStore_GetValue(ps, (const PROPERTYKEY*)&PKEY_AudioEndpoint_GUID, &pvguid); if(FAILED(hr)) { WARN("GetValue AudioEndpoint_GUID failed: 0x%08lx\n", hr); alstr_copy_cstr(guid, "Unknown Device GUID"); } else if(pvguid.vt == VT_LPWSTR) alstr_copy_wcstr(guid, pvguid.pwszVal); else { WARN("Unexpected PROPVARIANT type: 0x%04x\n", pvguid.vt); alstr_copy_cstr(guid, "Unknown Device GUID"); } PropVariantClear(&pvguid); } IPropertyStore_Release(ps); } static void get_device_formfactor(IMMDevice *device, EndpointFormFactor *formfactor) { IPropertyStore *ps; PROPVARIANT pvform; HRESULT hr; hr = IMMDevice_OpenPropertyStore(device, STGM_READ, &ps); if(FAILED(hr)) { WARN("OpenPropertyStore failed: 0x%08lx\n", hr); return; } PropVariantInit(&pvform); hr = IPropertyStore_GetValue(ps, &PKEY_AudioEndpoint_FormFactor, &pvform); if(FAILED(hr)) WARN("GetValue AudioEndpoint_FormFactor failed: 0x%08lx\n", hr); else if(pvform.vt == VT_UI4) *formfactor = pvform.ulVal; else if(pvform.vt == VT_EMPTY) *formfactor = UnknownFormFactor; else WARN("Unexpected PROPVARIANT type: 0x%04x\n", pvform.vt); PropVariantClear(&pvform); IPropertyStore_Release(ps); } static void add_device(IMMDevice *device, const WCHAR *devid, vector_DevMap *list) { int count = 0; al_string tmpname; DevMap entry; AL_STRING_INIT(tmpname); AL_STRING_INIT(entry.name); AL_STRING_INIT(entry.endpoint_guid); entry.devid = strdupW(devid); get_device_name_and_guid(device, &tmpname, &entry.endpoint_guid); while(1) { const DevMap *iter; alstr_copy(&entry.name, tmpname); if(count != 0) { char str[64]; snprintf(str, sizeof(str), " #%d", count+1); alstr_append_cstr(&entry.name, str); } #define MATCH_ENTRY(i) (alstr_cmp(entry.name, (i)->name) == 0) VECTOR_FIND_IF(iter, const DevMap, *list, MATCH_ENTRY); if(iter == VECTOR_END(*list)) break; #undef MATCH_ENTRY count++; } TRACE("Got device \"%s\", \"%s\", \"%ls\"\n", alstr_get_cstr(entry.name), alstr_get_cstr(entry.endpoint_guid), entry.devid); VECTOR_PUSH_BACK(*list, entry); AL_STRING_DEINIT(tmpname); } static WCHAR *get_device_id(IMMDevice *device) { WCHAR *devid; HRESULT hr; hr = IMMDevice_GetId(device, &devid); if(FAILED(hr)) { ERR("Failed to get device id: %lx\n", hr); return NULL; } return devid; } static HRESULT probe_devices(IMMDeviceEnumerator *devenum, EDataFlow flowdir, vector_DevMap *list) { IMMDeviceCollection *coll; IMMDevice *defdev = NULL; WCHAR *defdevid = NULL; HRESULT hr; UINT count; UINT i; hr = IMMDeviceEnumerator_EnumAudioEndpoints(devenum, flowdir, DEVICE_STATE_ACTIVE, &coll); if(FAILED(hr)) { ERR("Failed to enumerate audio endpoints: 0x%08lx\n", hr); return hr; } count = 0; hr = IMMDeviceCollection_GetCount(coll, &count); if(SUCCEEDED(hr) && count > 0) { clear_devlist(list); VECTOR_RESIZE(*list, 0, count); hr = IMMDeviceEnumerator_GetDefaultAudioEndpoint(devenum, flowdir, eMultimedia, &defdev); } if(SUCCEEDED(hr) && defdev != NULL) { defdevid = get_device_id(defdev); if(defdevid) add_device(defdev, defdevid, list); } for(i = 0;i < count;++i) { IMMDevice *device; WCHAR *devid; hr = IMMDeviceCollection_Item(coll, i, &device); if(FAILED(hr)) continue; devid = get_device_id(device); if(devid) { if(wcscmp(devid, defdevid) != 0) add_device(device, devid, list); CoTaskMemFree(devid); } IMMDevice_Release(device); } if(defdev) IMMDevice_Release(defdev); if(defdevid) CoTaskMemFree(defdevid); IMMDeviceCollection_Release(coll); return S_OK; } /* Proxy interface used by the message handler. */ struct ALCwasapiProxyVtable; typedef struct ALCwasapiProxy { const struct ALCwasapiProxyVtable *vtbl; } ALCwasapiProxy; struct ALCwasapiProxyVtable { HRESULT (*const openProxy)(ALCwasapiProxy*); void (*const closeProxy)(ALCwasapiProxy*); HRESULT (*const resetProxy)(ALCwasapiProxy*); HRESULT (*const startProxy)(ALCwasapiProxy*); void (*const stopProxy)(ALCwasapiProxy*); }; #define DEFINE_ALCWASAPIPROXY_VTABLE(T) \ DECLARE_THUNK(T, ALCwasapiProxy, HRESULT, openProxy) \ DECLARE_THUNK(T, ALCwasapiProxy, void, closeProxy) \ DECLARE_THUNK(T, ALCwasapiProxy, HRESULT, resetProxy) \ DECLARE_THUNK(T, ALCwasapiProxy, HRESULT, startProxy) \ DECLARE_THUNK(T, ALCwasapiProxy, void, stopProxy) \ \ static const struct ALCwasapiProxyVtable T##_ALCwasapiProxy_vtable = { \ T##_ALCwasapiProxy_openProxy, \ T##_ALCwasapiProxy_closeProxy, \ T##_ALCwasapiProxy_resetProxy, \ T##_ALCwasapiProxy_startProxy, \ T##_ALCwasapiProxy_stopProxy, \ } static void ALCwasapiProxy_Construct(ALCwasapiProxy* UNUSED(self)) { } static void ALCwasapiProxy_Destruct(ALCwasapiProxy* UNUSED(self)) { } static DWORD CALLBACK ALCwasapiProxy_messageHandler(void *ptr) { ThreadRequest *req = ptr; IMMDeviceEnumerator *Enumerator; ALuint deviceCount = 0; ALCwasapiProxy *proxy; HRESULT hr, cohr; MSG msg; TRACE("Starting message thread\n"); cohr = CoInitializeEx(NULL, COINIT_MULTITHREADED); if(FAILED(cohr)) { WARN("Failed to initialize COM: 0x%08lx\n", cohr); ReturnMsgResponse(req, cohr); return 0; } hr = CoCreateInstance(&CLSID_MMDeviceEnumerator, NULL, CLSCTX_INPROC_SERVER, &IID_IMMDeviceEnumerator, &ptr); if(FAILED(hr)) { WARN("Failed to create IMMDeviceEnumerator instance: 0x%08lx\n", hr); CoUninitialize(); ReturnMsgResponse(req, hr); return 0; } Enumerator = ptr; IMMDeviceEnumerator_Release(Enumerator); Enumerator = NULL; CoUninitialize(); /* HACK: Force Windows to create a message queue for this thread before * returning success, otherwise PostThreadMessage may fail if it gets * called before GetMessage. */ PeekMessage(&msg, NULL, WM_USER, WM_USER, PM_NOREMOVE); TRACE("Message thread initialization complete\n"); ReturnMsgResponse(req, S_OK); TRACE("Starting message loop\n"); while(GetMessage(&msg, NULL, WM_USER_First, WM_USER_Last)) { TRACE("Got message \"%s\" (0x%04x, lparam=%p, wparam=%p)\n", (msg.message >= WM_USER && msg.message <= WM_USER_Last) ? MessageStr[msg.message-WM_USER] : "Unknown", msg.message, (void*)msg.lParam, (void*)msg.wParam ); switch(msg.message) { case WM_USER_OpenDevice: req = (ThreadRequest*)msg.wParam; proxy = (ALCwasapiProxy*)msg.lParam; hr = cohr = S_OK; if(++deviceCount == 1) hr = cohr = CoInitializeEx(NULL, COINIT_MULTITHREADED); if(SUCCEEDED(hr)) hr = V0(proxy,openProxy)(); if(FAILED(hr)) { if(--deviceCount == 0 && SUCCEEDED(cohr)) CoUninitialize(); } ReturnMsgResponse(req, hr); continue; case WM_USER_ResetDevice: req = (ThreadRequest*)msg.wParam; proxy = (ALCwasapiProxy*)msg.lParam; hr = V0(proxy,resetProxy)(); ReturnMsgResponse(req, hr); continue; case WM_USER_StartDevice: req = (ThreadRequest*)msg.wParam; proxy = (ALCwasapiProxy*)msg.lParam; hr = V0(proxy,startProxy)(); ReturnMsgResponse(req, hr); continue; case WM_USER_StopDevice: req = (ThreadRequest*)msg.wParam; proxy = (ALCwasapiProxy*)msg.lParam; V0(proxy,stopProxy)(); ReturnMsgResponse(req, S_OK); continue; case WM_USER_CloseDevice: req = (ThreadRequest*)msg.wParam; proxy = (ALCwasapiProxy*)msg.lParam; V0(proxy,closeProxy)(); if(--deviceCount == 0) CoUninitialize(); ReturnMsgResponse(req, S_OK); continue; case WM_USER_Enumerate: req = (ThreadRequest*)msg.wParam; hr = cohr = S_OK; if(++deviceCount == 1) hr = cohr = CoInitializeEx(NULL, COINIT_MULTITHREADED); if(SUCCEEDED(hr)) hr = CoCreateInstance(&CLSID_MMDeviceEnumerator, NULL, CLSCTX_INPROC_SERVER, &IID_IMMDeviceEnumerator, &ptr); if(SUCCEEDED(hr)) { Enumerator = ptr; if(msg.lParam == ALL_DEVICE_PROBE) hr = probe_devices(Enumerator, eRender, &PlaybackDevices); else if(msg.lParam == CAPTURE_DEVICE_PROBE) hr = probe_devices(Enumerator, eCapture, &CaptureDevices); IMMDeviceEnumerator_Release(Enumerator); Enumerator = NULL; } if(--deviceCount == 0 && SUCCEEDED(cohr)) CoUninitialize(); ReturnMsgResponse(req, hr); continue; default: ERR("Unexpected message: %u\n", msg.message); continue; } } TRACE("Message loop finished\n"); return 0; } typedef struct ALCwasapiPlayback { DERIVE_FROM_TYPE(ALCbackend); DERIVE_FROM_TYPE(ALCwasapiProxy); WCHAR *devid; IMMDevice *mmdev; IAudioClient *client; IAudioRenderClient *render; HANDLE NotifyEvent; HANDLE MsgEvent; ATOMIC(UINT32) Padding; ATOMIC(int) killNow; althrd_t thread; } ALCwasapiPlayback; static int ALCwasapiPlayback_mixerProc(void *arg); static void ALCwasapiPlayback_Construct(ALCwasapiPlayback *self, ALCdevice *device); static void ALCwasapiPlayback_Destruct(ALCwasapiPlayback *self); static ALCenum ALCwasapiPlayback_open(ALCwasapiPlayback *self, const ALCchar *name); static HRESULT ALCwasapiPlayback_openProxy(ALCwasapiPlayback *self); static void ALCwasapiPlayback_closeProxy(ALCwasapiPlayback *self); static ALCboolean ALCwasapiPlayback_reset(ALCwasapiPlayback *self); static HRESULT ALCwasapiPlayback_resetProxy(ALCwasapiPlayback *self); static ALCboolean ALCwasapiPlayback_start(ALCwasapiPlayback *self); static HRESULT ALCwasapiPlayback_startProxy(ALCwasapiPlayback *self); static void ALCwasapiPlayback_stop(ALCwasapiPlayback *self); static void ALCwasapiPlayback_stopProxy(ALCwasapiPlayback *self); static DECLARE_FORWARD2(ALCwasapiPlayback, ALCbackend, ALCenum, captureSamples, ALCvoid*, ALCuint) static DECLARE_FORWARD(ALCwasapiPlayback, ALCbackend, ALCuint, availableSamples) static ClockLatency ALCwasapiPlayback_getClockLatency(ALCwasapiPlayback *self); static DECLARE_FORWARD(ALCwasapiPlayback, ALCbackend, void, lock) static DECLARE_FORWARD(ALCwasapiPlayback, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCwasapiPlayback) DEFINE_ALCWASAPIPROXY_VTABLE(ALCwasapiPlayback); DEFINE_ALCBACKEND_VTABLE(ALCwasapiPlayback); static void ALCwasapiPlayback_Construct(ALCwasapiPlayback *self, ALCdevice *device) { SET_VTABLE2(ALCwasapiPlayback, ALCbackend, self); SET_VTABLE2(ALCwasapiPlayback, ALCwasapiProxy, self); ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); ALCwasapiProxy_Construct(STATIC_CAST(ALCwasapiProxy, self)); self->devid = NULL; self->mmdev = NULL; self->client = NULL; self->render = NULL; self->NotifyEvent = NULL; self->MsgEvent = NULL; ATOMIC_INIT(&self->Padding, 0); ATOMIC_INIT(&self->killNow, 0); } static void ALCwasapiPlayback_Destruct(ALCwasapiPlayback *self) { if(self->MsgEvent) { ThreadRequest req = { self->MsgEvent, 0 }; if(PostThreadMessage(ThreadID, WM_USER_CloseDevice, (WPARAM)&req, (LPARAM)STATIC_CAST(ALCwasapiProxy, self))) (void)WaitForResponse(&req); CloseHandle(self->MsgEvent); self->MsgEvent = NULL; } if(self->NotifyEvent) CloseHandle(self->NotifyEvent); self->NotifyEvent = NULL; free(self->devid); self->devid = NULL; if(self->NotifyEvent != NULL) CloseHandle(self->NotifyEvent); self->NotifyEvent = NULL; if(self->MsgEvent != NULL) CloseHandle(self->MsgEvent); self->MsgEvent = NULL; free(self->devid); self->devid = NULL; ALCwasapiProxy_Destruct(STATIC_CAST(ALCwasapiProxy, self)); ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } FORCE_ALIGN static int ALCwasapiPlayback_mixerProc(void *arg) { ALCwasapiPlayback *self = arg; ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; UINT32 buffer_len, written; ALuint update_size, len; BYTE *buffer; HRESULT hr; hr = CoInitializeEx(NULL, COINIT_MULTITHREADED); if(FAILED(hr)) { ERR("CoInitializeEx(NULL, COINIT_MULTITHREADED) failed: 0x%08lx\n", hr); V0(device->Backend,lock)(); aluHandleDisconnect(device, "COM init failed: 0x%08lx", hr); V0(device->Backend,unlock)(); return 1; } SetRTPriority(); althrd_setname(althrd_current(), MIXER_THREAD_NAME); update_size = device->UpdateSize; buffer_len = update_size * device->NumUpdates; while(!ATOMIC_LOAD(&self->killNow, almemory_order_relaxed)) { hr = IAudioClient_GetCurrentPadding(self->client, &written); if(FAILED(hr)) { ERR("Failed to get padding: 0x%08lx\n", hr); V0(device->Backend,lock)(); aluHandleDisconnect(device, "Failed to retrieve buffer padding: 0x%08lx", hr); V0(device->Backend,unlock)(); break; } ATOMIC_STORE(&self->Padding, written, almemory_order_relaxed); len = buffer_len - written; if(len < update_size) { DWORD res; res = WaitForSingleObjectEx(self->NotifyEvent, 2000, FALSE); if(res != WAIT_OBJECT_0) ERR("WaitForSingleObjectEx error: 0x%lx\n", res); continue; } len -= len%update_size; hr = IAudioRenderClient_GetBuffer(self->render, len, &buffer); if(SUCCEEDED(hr)) { ALCwasapiPlayback_lock(self); aluMixData(device, buffer, len); ATOMIC_STORE(&self->Padding, written + len, almemory_order_relaxed); ALCwasapiPlayback_unlock(self); hr = IAudioRenderClient_ReleaseBuffer(self->render, len, 0); } if(FAILED(hr)) { ERR("Failed to buffer data: 0x%08lx\n", hr); V0(device->Backend,lock)(); aluHandleDisconnect(device, "Failed to send playback samples: 0x%08lx", hr); V0(device->Backend,unlock)(); break; } } ATOMIC_STORE(&self->Padding, 0, almemory_order_release); CoUninitialize(); return 0; } static ALCboolean MakeExtensible(WAVEFORMATEXTENSIBLE *out, const WAVEFORMATEX *in) { memset(out, 0, sizeof(*out)); if(in->wFormatTag == WAVE_FORMAT_EXTENSIBLE) *out = *(const WAVEFORMATEXTENSIBLE*)in; else if(in->wFormatTag == WAVE_FORMAT_PCM) { out->Format = *in; out->Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE; out->Format.cbSize = sizeof(*out) - sizeof(*in); if(out->Format.nChannels == 1) out->dwChannelMask = MONO; else if(out->Format.nChannels == 2) out->dwChannelMask = STEREO; else ERR("Unhandled PCM channel count: %d\n", out->Format.nChannels); out->SubFormat = KSDATAFORMAT_SUBTYPE_PCM; } else if(in->wFormatTag == WAVE_FORMAT_IEEE_FLOAT) { out->Format = *in; out->Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE; out->Format.cbSize = sizeof(*out) - sizeof(*in); if(out->Format.nChannels == 1) out->dwChannelMask = MONO; else if(out->Format.nChannels == 2) out->dwChannelMask = STEREO; else ERR("Unhandled IEEE float channel count: %d\n", out->Format.nChannels); out->SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT; } else { ERR("Unhandled format tag: 0x%04x\n", in->wFormatTag); return ALC_FALSE; } return ALC_TRUE; } static ALCenum ALCwasapiPlayback_open(ALCwasapiPlayback *self, const ALCchar *deviceName) { HRESULT hr = S_OK; self->NotifyEvent = CreateEventW(NULL, FALSE, FALSE, NULL); self->MsgEvent = CreateEventW(NULL, FALSE, FALSE, NULL); if(self->NotifyEvent == NULL || self->MsgEvent == NULL) { ERR("Failed to create message events: %lu\n", GetLastError()); hr = E_FAIL; } if(SUCCEEDED(hr)) { if(deviceName) { const DevMap *iter; if(VECTOR_SIZE(PlaybackDevices) == 0) { ThreadRequest req = { self->MsgEvent, 0 }; if(PostThreadMessage(ThreadID, WM_USER_Enumerate, (WPARAM)&req, ALL_DEVICE_PROBE)) (void)WaitForResponse(&req); } hr = E_FAIL; #define MATCH_NAME(i) (alstr_cmp_cstr((i)->name, deviceName) == 0 || \ alstr_cmp_cstr((i)->endpoint_guid, deviceName) == 0) VECTOR_FIND_IF(iter, const DevMap, PlaybackDevices, MATCH_NAME); #undef MATCH_NAME if(iter == VECTOR_END(PlaybackDevices)) { int len; if((len=MultiByteToWideChar(CP_UTF8, 0, deviceName, -1, NULL, 0)) > 0) { WCHAR *wname = calloc(sizeof(WCHAR), len); MultiByteToWideChar(CP_UTF8, 0, deviceName, -1, wname, len); #define MATCH_NAME(i) (wcscmp((i)->devid, wname) == 0) VECTOR_FIND_IF(iter, const DevMap, PlaybackDevices, MATCH_NAME); #undef MATCH_NAME free(wname); } } if(iter == VECTOR_END(PlaybackDevices)) WARN("Failed to find device name matching \"%s\"\n", deviceName); else { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; self->devid = strdupW(iter->devid); alstr_copy(&device->DeviceName, iter->name); hr = S_OK; } } } if(SUCCEEDED(hr)) { ThreadRequest req = { self->MsgEvent, 0 }; hr = E_FAIL; if(PostThreadMessage(ThreadID, WM_USER_OpenDevice, (WPARAM)&req, (LPARAM)STATIC_CAST(ALCwasapiProxy, self))) hr = WaitForResponse(&req); else ERR("Failed to post thread message: %lu\n", GetLastError()); } if(FAILED(hr)) { if(self->NotifyEvent != NULL) CloseHandle(self->NotifyEvent); self->NotifyEvent = NULL; if(self->MsgEvent != NULL) CloseHandle(self->MsgEvent); self->MsgEvent = NULL; free(self->devid); self->devid = NULL; ERR("Device init failed: 0x%08lx\n", hr); return ALC_INVALID_VALUE; } return ALC_NO_ERROR; } static HRESULT ALCwasapiPlayback_openProxy(ALCwasapiPlayback *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; void *ptr; HRESULT hr; hr = CoCreateInstance(&CLSID_MMDeviceEnumerator, NULL, CLSCTX_INPROC_SERVER, &IID_IMMDeviceEnumerator, &ptr); if(SUCCEEDED(hr)) { IMMDeviceEnumerator *Enumerator = ptr; if(!self->devid) hr = IMMDeviceEnumerator_GetDefaultAudioEndpoint(Enumerator, eRender, eMultimedia, &self->mmdev); else hr = IMMDeviceEnumerator_GetDevice(Enumerator, self->devid, &self->mmdev); IMMDeviceEnumerator_Release(Enumerator); Enumerator = NULL; } if(SUCCEEDED(hr)) hr = IMMDevice_Activate(self->mmdev, &IID_IAudioClient, CLSCTX_INPROC_SERVER, NULL, &ptr); if(SUCCEEDED(hr)) { self->client = ptr; if(alstr_empty(device->DeviceName)) get_device_name_and_guid(self->mmdev, &device->DeviceName, NULL); } if(FAILED(hr)) { if(self->mmdev) IMMDevice_Release(self->mmdev); self->mmdev = NULL; } return hr; } static void ALCwasapiPlayback_closeProxy(ALCwasapiPlayback *self) { if(self->client) IAudioClient_Release(self->client); self->client = NULL; if(self->mmdev) IMMDevice_Release(self->mmdev); self->mmdev = NULL; } static ALCboolean ALCwasapiPlayback_reset(ALCwasapiPlayback *self) { ThreadRequest req = { self->MsgEvent, 0 }; HRESULT hr = E_FAIL; if(PostThreadMessage(ThreadID, WM_USER_ResetDevice, (WPARAM)&req, (LPARAM)STATIC_CAST(ALCwasapiProxy, self))) hr = WaitForResponse(&req); return SUCCEEDED(hr) ? ALC_TRUE : ALC_FALSE; } static HRESULT ALCwasapiPlayback_resetProxy(ALCwasapiPlayback *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; EndpointFormFactor formfactor = UnknownFormFactor; WAVEFORMATEXTENSIBLE OutputType; WAVEFORMATEX *wfx = NULL; REFERENCE_TIME min_per, buf_time; UINT32 buffer_len, min_len; void *ptr = NULL; HRESULT hr; if(self->client) IAudioClient_Release(self->client); self->client = NULL; hr = IMMDevice_Activate(self->mmdev, &IID_IAudioClient, CLSCTX_INPROC_SERVER, NULL, &ptr); if(FAILED(hr)) { ERR("Failed to reactivate audio client: 0x%08lx\n", hr); return hr; } self->client = ptr; hr = IAudioClient_GetMixFormat(self->client, &wfx); if(FAILED(hr)) { ERR("Failed to get mix format: 0x%08lx\n", hr); return hr; } if(!MakeExtensible(&OutputType, wfx)) { CoTaskMemFree(wfx); return E_FAIL; } CoTaskMemFree(wfx); wfx = NULL; buf_time = ScaleCeil(device->UpdateSize*device->NumUpdates, REFTIME_PER_SEC, device->Frequency); if(!(device->Flags&DEVICE_FREQUENCY_REQUEST)) device->Frequency = OutputType.Format.nSamplesPerSec; if(!(device->Flags&DEVICE_CHANNELS_REQUEST)) { if(OutputType.Format.nChannels == 1 && OutputType.dwChannelMask == MONO) device->FmtChans = DevFmtMono; else if(OutputType.Format.nChannels == 2 && OutputType.dwChannelMask == STEREO) device->FmtChans = DevFmtStereo; else if(OutputType.Format.nChannels == 4 && OutputType.dwChannelMask == QUAD) device->FmtChans = DevFmtQuad; else if(OutputType.Format.nChannels == 6 && OutputType.dwChannelMask == X5DOT1) device->FmtChans = DevFmtX51; else if(OutputType.Format.nChannels == 6 && OutputType.dwChannelMask == X5DOT1REAR) device->FmtChans = DevFmtX51Rear; else if(OutputType.Format.nChannels == 7 && OutputType.dwChannelMask == X6DOT1) device->FmtChans = DevFmtX61; else if(OutputType.Format.nChannels == 8 && (OutputType.dwChannelMask == X7DOT1 || OutputType.dwChannelMask == X7DOT1_WIDE)) device->FmtChans = DevFmtX71; else ERR("Unhandled channel config: %d -- 0x%08lx\n", OutputType.Format.nChannels, OutputType.dwChannelMask); } switch(device->FmtChans) { case DevFmtMono: OutputType.Format.nChannels = 1; OutputType.dwChannelMask = MONO; break; case DevFmtAmbi3D: device->FmtChans = DevFmtStereo; /*fall-through*/ case DevFmtStereo: OutputType.Format.nChannels = 2; OutputType.dwChannelMask = STEREO; break; case DevFmtQuad: OutputType.Format.nChannels = 4; OutputType.dwChannelMask = QUAD; break; case DevFmtX51: OutputType.Format.nChannels = 6; OutputType.dwChannelMask = X5DOT1; break; case DevFmtX51Rear: OutputType.Format.nChannels = 6; OutputType.dwChannelMask = X5DOT1REAR; break; case DevFmtX61: OutputType.Format.nChannels = 7; OutputType.dwChannelMask = X6DOT1; break; case DevFmtX71: OutputType.Format.nChannels = 8; OutputType.dwChannelMask = X7DOT1; break; } switch(device->FmtType) { case DevFmtByte: device->FmtType = DevFmtUByte; /* fall-through */ case DevFmtUByte: OutputType.Format.wBitsPerSample = 8; OutputType.Samples.wValidBitsPerSample = 8; OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; break; case DevFmtUShort: device->FmtType = DevFmtShort; /* fall-through */ case DevFmtShort: OutputType.Format.wBitsPerSample = 16; OutputType.Samples.wValidBitsPerSample = 16; OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; break; case DevFmtUInt: device->FmtType = DevFmtInt; /* fall-through */ case DevFmtInt: OutputType.Format.wBitsPerSample = 32; OutputType.Samples.wValidBitsPerSample = 32; OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; break; case DevFmtFloat: OutputType.Format.wBitsPerSample = 32; OutputType.Samples.wValidBitsPerSample = 32; OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT; break; } OutputType.Format.nSamplesPerSec = device->Frequency; OutputType.Format.nBlockAlign = OutputType.Format.nChannels * OutputType.Format.wBitsPerSample / 8; OutputType.Format.nAvgBytesPerSec = OutputType.Format.nSamplesPerSec * OutputType.Format.nBlockAlign; hr = IAudioClient_IsFormatSupported(self->client, AUDCLNT_SHAREMODE_SHARED, &OutputType.Format, &wfx); if(FAILED(hr)) { ERR("Failed to check format support: 0x%08lx\n", hr); hr = IAudioClient_GetMixFormat(self->client, &wfx); } if(FAILED(hr)) { ERR("Failed to find a supported format: 0x%08lx\n", hr); return hr; } if(wfx != NULL) { if(!MakeExtensible(&OutputType, wfx)) { CoTaskMemFree(wfx); return E_FAIL; } CoTaskMemFree(wfx); wfx = NULL; device->Frequency = OutputType.Format.nSamplesPerSec; if(OutputType.Format.nChannels == 1 && OutputType.dwChannelMask == MONO) device->FmtChans = DevFmtMono; else if(OutputType.Format.nChannels == 2 && OutputType.dwChannelMask == STEREO) device->FmtChans = DevFmtStereo; else if(OutputType.Format.nChannels == 4 && OutputType.dwChannelMask == QUAD) device->FmtChans = DevFmtQuad; else if(OutputType.Format.nChannels == 6 && OutputType.dwChannelMask == X5DOT1) device->FmtChans = DevFmtX51; else if(OutputType.Format.nChannels == 6 && OutputType.dwChannelMask == X5DOT1REAR) device->FmtChans = DevFmtX51Rear; else if(OutputType.Format.nChannels == 7 && OutputType.dwChannelMask == X6DOT1) device->FmtChans = DevFmtX61; else if(OutputType.Format.nChannels == 8 && (OutputType.dwChannelMask == X7DOT1 || OutputType.dwChannelMask == X7DOT1_WIDE)) device->FmtChans = DevFmtX71; else { ERR("Unhandled extensible channels: %d -- 0x%08lx\n", OutputType.Format.nChannels, OutputType.dwChannelMask); device->FmtChans = DevFmtStereo; OutputType.Format.nChannels = 2; OutputType.dwChannelMask = STEREO; } if(IsEqualGUID(&OutputType.SubFormat, &KSDATAFORMAT_SUBTYPE_PCM)) { if(OutputType.Format.wBitsPerSample == 8) device->FmtType = DevFmtUByte; else if(OutputType.Format.wBitsPerSample == 16) device->FmtType = DevFmtShort; else if(OutputType.Format.wBitsPerSample == 32) device->FmtType = DevFmtInt; else { device->FmtType = DevFmtShort; OutputType.Format.wBitsPerSample = 16; } } else if(IsEqualGUID(&OutputType.SubFormat, &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)) { device->FmtType = DevFmtFloat; OutputType.Format.wBitsPerSample = 32; } else { ERR("Unhandled format sub-type\n"); device->FmtType = DevFmtShort; OutputType.Format.wBitsPerSample = 16; OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; } OutputType.Samples.wValidBitsPerSample = OutputType.Format.wBitsPerSample; } get_device_formfactor(self->mmdev, &formfactor); device->IsHeadphones = (device->FmtChans == DevFmtStereo && (formfactor == Headphones || formfactor == Headset) ); SetDefaultWFXChannelOrder(device); hr = IAudioClient_Initialize(self->client, AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_EVENTCALLBACK, buf_time, 0, &OutputType.Format, NULL); if(FAILED(hr)) { ERR("Failed to initialize audio client: 0x%08lx\n", hr); return hr; } hr = IAudioClient_GetDevicePeriod(self->client, &min_per, NULL); if(SUCCEEDED(hr)) { min_len = (UINT32)ScaleCeil(min_per, device->Frequency, REFTIME_PER_SEC); /* Find the nearest multiple of the period size to the update size */ if(min_len < device->UpdateSize) min_len *= (device->UpdateSize + min_len/2)/min_len; hr = IAudioClient_GetBufferSize(self->client, &buffer_len); } if(FAILED(hr)) { ERR("Failed to get audio buffer info: 0x%08lx\n", hr); return hr; } device->UpdateSize = min_len; device->NumUpdates = buffer_len / device->UpdateSize; if(device->NumUpdates <= 1) { ERR("Audio client returned buffer_len < period*2; expect break up\n"); device->NumUpdates = 2; device->UpdateSize = buffer_len / device->NumUpdates; } hr = IAudioClient_SetEventHandle(self->client, self->NotifyEvent); if(FAILED(hr)) { ERR("Failed to set event handle: 0x%08lx\n", hr); return hr; } return hr; } static ALCboolean ALCwasapiPlayback_start(ALCwasapiPlayback *self) { ThreadRequest req = { self->MsgEvent, 0 }; HRESULT hr = E_FAIL; if(PostThreadMessage(ThreadID, WM_USER_StartDevice, (WPARAM)&req, (LPARAM)STATIC_CAST(ALCwasapiProxy, self))) hr = WaitForResponse(&req); return SUCCEEDED(hr) ? ALC_TRUE : ALC_FALSE; } static HRESULT ALCwasapiPlayback_startProxy(ALCwasapiPlayback *self) { HRESULT hr; void *ptr; ResetEvent(self->NotifyEvent); hr = IAudioClient_Start(self->client); if(FAILED(hr)) ERR("Failed to start audio client: 0x%08lx\n", hr); if(SUCCEEDED(hr)) hr = IAudioClient_GetService(self->client, &IID_IAudioRenderClient, &ptr); if(SUCCEEDED(hr)) { self->render = ptr; ATOMIC_STORE(&self->killNow, 0, almemory_order_release); if(althrd_create(&self->thread, ALCwasapiPlayback_mixerProc, self) != althrd_success) { if(self->render) IAudioRenderClient_Release(self->render); self->render = NULL; IAudioClient_Stop(self->client); ERR("Failed to start thread\n"); hr = E_FAIL; } } return hr; } static void ALCwasapiPlayback_stop(ALCwasapiPlayback *self) { ThreadRequest req = { self->MsgEvent, 0 }; if(PostThreadMessage(ThreadID, WM_USER_StopDevice, (WPARAM)&req, (LPARAM)STATIC_CAST(ALCwasapiProxy, self))) (void)WaitForResponse(&req); } static void ALCwasapiPlayback_stopProxy(ALCwasapiPlayback *self) { int res; if(!self->render) return; ATOMIC_STORE_SEQ(&self->killNow, 1); althrd_join(self->thread, &res); IAudioRenderClient_Release(self->render); self->render = NULL; IAudioClient_Stop(self->client); } static ClockLatency ALCwasapiPlayback_getClockLatency(ALCwasapiPlayback *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; ClockLatency ret; ALCwasapiPlayback_lock(self); ret.ClockTime = GetDeviceClockTime(device); ret.Latency = ATOMIC_LOAD(&self->Padding, almemory_order_relaxed) * DEVICE_CLOCK_RES / device->Frequency; ALCwasapiPlayback_unlock(self); return ret; } typedef struct ALCwasapiCapture { DERIVE_FROM_TYPE(ALCbackend); DERIVE_FROM_TYPE(ALCwasapiProxy); WCHAR *devid; IMMDevice *mmdev; IAudioClient *client; IAudioCaptureClient *capture; HANDLE NotifyEvent; HANDLE MsgEvent; ChannelConverter *ChannelConv; SampleConverter *SampleConv; ll_ringbuffer_t *Ring; ATOMIC(int) killNow; althrd_t thread; } ALCwasapiCapture; static int ALCwasapiCapture_recordProc(void *arg); static void ALCwasapiCapture_Construct(ALCwasapiCapture *self, ALCdevice *device); static void ALCwasapiCapture_Destruct(ALCwasapiCapture *self); static ALCenum ALCwasapiCapture_open(ALCwasapiCapture *self, const ALCchar *name); static HRESULT ALCwasapiCapture_openProxy(ALCwasapiCapture *self); static void ALCwasapiCapture_closeProxy(ALCwasapiCapture *self); static DECLARE_FORWARD(ALCwasapiCapture, ALCbackend, ALCboolean, reset) static HRESULT ALCwasapiCapture_resetProxy(ALCwasapiCapture *self); static ALCboolean ALCwasapiCapture_start(ALCwasapiCapture *self); static HRESULT ALCwasapiCapture_startProxy(ALCwasapiCapture *self); static void ALCwasapiCapture_stop(ALCwasapiCapture *self); static void ALCwasapiCapture_stopProxy(ALCwasapiCapture *self); static ALCenum ALCwasapiCapture_captureSamples(ALCwasapiCapture *self, ALCvoid *buffer, ALCuint samples); static ALuint ALCwasapiCapture_availableSamples(ALCwasapiCapture *self); static DECLARE_FORWARD(ALCwasapiCapture, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(ALCwasapiCapture, ALCbackend, void, lock) static DECLARE_FORWARD(ALCwasapiCapture, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCwasapiCapture) DEFINE_ALCWASAPIPROXY_VTABLE(ALCwasapiCapture); DEFINE_ALCBACKEND_VTABLE(ALCwasapiCapture); static void ALCwasapiCapture_Construct(ALCwasapiCapture *self, ALCdevice *device) { SET_VTABLE2(ALCwasapiCapture, ALCbackend, self); SET_VTABLE2(ALCwasapiCapture, ALCwasapiProxy, self); ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); ALCwasapiProxy_Construct(STATIC_CAST(ALCwasapiProxy, self)); self->devid = NULL; self->mmdev = NULL; self->client = NULL; self->capture = NULL; self->NotifyEvent = NULL; self->MsgEvent = NULL; self->ChannelConv = NULL; self->SampleConv = NULL; self->Ring = NULL; ATOMIC_INIT(&self->killNow, 0); } static void ALCwasapiCapture_Destruct(ALCwasapiCapture *self) { if(self->MsgEvent) { ThreadRequest req = { self->MsgEvent, 0 }; if(PostThreadMessage(ThreadID, WM_USER_CloseDevice, (WPARAM)&req, (LPARAM)STATIC_CAST(ALCwasapiProxy, self))) (void)WaitForResponse(&req); CloseHandle(self->MsgEvent); self->MsgEvent = NULL; } if(self->NotifyEvent != NULL) CloseHandle(self->NotifyEvent); self->NotifyEvent = NULL; ll_ringbuffer_free(self->Ring); self->Ring = NULL; DestroySampleConverter(&self->SampleConv); DestroyChannelConverter(&self->ChannelConv); free(self->devid); self->devid = NULL; ALCwasapiProxy_Destruct(STATIC_CAST(ALCwasapiProxy, self)); ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } FORCE_ALIGN int ALCwasapiCapture_recordProc(void *arg) { ALCwasapiCapture *self = arg; ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; ALfloat *samples = NULL; size_t samplesmax = 0; HRESULT hr; hr = CoInitializeEx(NULL, COINIT_MULTITHREADED); if(FAILED(hr)) { ERR("CoInitializeEx(NULL, COINIT_MULTITHREADED) failed: 0x%08lx\n", hr); V0(device->Backend,lock)(); aluHandleDisconnect(device, "COM init failed: 0x%08lx", hr); V0(device->Backend,unlock)(); return 1; } althrd_setname(althrd_current(), RECORD_THREAD_NAME); while(!ATOMIC_LOAD(&self->killNow, almemory_order_relaxed)) { UINT32 avail; DWORD res; hr = IAudioCaptureClient_GetNextPacketSize(self->capture, &avail); if(FAILED(hr)) ERR("Failed to get next packet size: 0x%08lx\n", hr); else if(avail > 0) { UINT32 numsamples; DWORD flags; BYTE *rdata; hr = IAudioCaptureClient_GetBuffer(self->capture, &rdata, &numsamples, &flags, NULL, NULL ); if(FAILED(hr)) ERR("Failed to get capture buffer: 0x%08lx\n", hr); else { ll_ringbuffer_data_t data[2]; size_t dstframes = 0; if(self->ChannelConv) { if(samplesmax < numsamples) { size_t newmax = RoundUp(numsamples, 4096); ALfloat *tmp = al_calloc(DEF_ALIGN, newmax*2*sizeof(ALfloat)); al_free(samples); samples = tmp; samplesmax = newmax; } ChannelConverterInput(self->ChannelConv, rdata, samples, numsamples); rdata = (BYTE*)samples; } ll_ringbuffer_get_write_vector(self->Ring, data); if(self->SampleConv) { const ALvoid *srcdata = rdata; ALsizei srcframes = numsamples; dstframes = SampleConverterInput(self->SampleConv, &srcdata, &srcframes, data[0].buf, (ALsizei)minz(data[0].len, INT_MAX) ); if(srcframes > 0 && dstframes == data[0].len && data[1].len > 0) { /* If some source samples remain, all of the first dest * block was filled, and there's space in the second * dest block, do another run for the second block. */ dstframes += SampleConverterInput(self->SampleConv, &srcdata, &srcframes, data[1].buf, (ALsizei)minz(data[1].len, INT_MAX) ); } } else { ALuint framesize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); size_t len1 = minz(data[0].len, numsamples); size_t len2 = minz(data[1].len, numsamples-len1); memcpy(data[0].buf, rdata, len1*framesize); if(len2 > 0) memcpy(data[1].buf, rdata+len1*framesize, len2*framesize); dstframes = len1 + len2; } ll_ringbuffer_write_advance(self->Ring, dstframes); hr = IAudioCaptureClient_ReleaseBuffer(self->capture, numsamples); if(FAILED(hr)) ERR("Failed to release capture buffer: 0x%08lx\n", hr); } } if(FAILED(hr)) { V0(device->Backend,lock)(); aluHandleDisconnect(device, "Failed to capture samples: 0x%08lx", hr); V0(device->Backend,unlock)(); break; } res = WaitForSingleObjectEx(self->NotifyEvent, 2000, FALSE); if(res != WAIT_OBJECT_0) ERR("WaitForSingleObjectEx error: 0x%lx\n", res); } al_free(samples); samples = NULL; samplesmax = 0; CoUninitialize(); return 0; } static ALCenum ALCwasapiCapture_open(ALCwasapiCapture *self, const ALCchar *deviceName) { HRESULT hr = S_OK; self->NotifyEvent = CreateEventW(NULL, FALSE, FALSE, NULL); self->MsgEvent = CreateEventW(NULL, FALSE, FALSE, NULL); if(self->NotifyEvent == NULL || self->MsgEvent == NULL) { ERR("Failed to create message events: %lu\n", GetLastError()); hr = E_FAIL; } if(SUCCEEDED(hr)) { if(deviceName) { const DevMap *iter; if(VECTOR_SIZE(CaptureDevices) == 0) { ThreadRequest req = { self->MsgEvent, 0 }; if(PostThreadMessage(ThreadID, WM_USER_Enumerate, (WPARAM)&req, CAPTURE_DEVICE_PROBE)) (void)WaitForResponse(&req); } hr = E_FAIL; #define MATCH_NAME(i) (alstr_cmp_cstr((i)->name, deviceName) == 0 || \ alstr_cmp_cstr((i)->endpoint_guid, deviceName) == 0) VECTOR_FIND_IF(iter, const DevMap, CaptureDevices, MATCH_NAME); #undef MATCH_NAME if(iter == VECTOR_END(CaptureDevices)) { int len; if((len=MultiByteToWideChar(CP_UTF8, 0, deviceName, -1, NULL, 0)) > 0) { WCHAR *wname = calloc(sizeof(WCHAR), len); MultiByteToWideChar(CP_UTF8, 0, deviceName, -1, wname, len); #define MATCH_NAME(i) (wcscmp((i)->devid, wname) == 0) VECTOR_FIND_IF(iter, const DevMap, CaptureDevices, MATCH_NAME); #undef MATCH_NAME free(wname); } } if(iter == VECTOR_END(CaptureDevices)) WARN("Failed to find device name matching \"%s\"\n", deviceName); else { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; self->devid = strdupW(iter->devid); alstr_copy(&device->DeviceName, iter->name); hr = S_OK; } } } if(SUCCEEDED(hr)) { ThreadRequest req = { self->MsgEvent, 0 }; hr = E_FAIL; if(PostThreadMessage(ThreadID, WM_USER_OpenDevice, (WPARAM)&req, (LPARAM)STATIC_CAST(ALCwasapiProxy, self))) hr = WaitForResponse(&req); else ERR("Failed to post thread message: %lu\n", GetLastError()); } if(FAILED(hr)) { if(self->NotifyEvent != NULL) CloseHandle(self->NotifyEvent); self->NotifyEvent = NULL; if(self->MsgEvent != NULL) CloseHandle(self->MsgEvent); self->MsgEvent = NULL; free(self->devid); self->devid = NULL; ERR("Device init failed: 0x%08lx\n", hr); return ALC_INVALID_VALUE; } else { ThreadRequest req = { self->MsgEvent, 0 }; hr = E_FAIL; if(PostThreadMessage(ThreadID, WM_USER_ResetDevice, (WPARAM)&req, (LPARAM)STATIC_CAST(ALCwasapiProxy, self))) hr = WaitForResponse(&req); else ERR("Failed to post thread message: %lu\n", GetLastError()); if(FAILED(hr)) { if(hr == E_OUTOFMEMORY) return ALC_OUT_OF_MEMORY; return ALC_INVALID_VALUE; } } return ALC_NO_ERROR; } static HRESULT ALCwasapiCapture_openProxy(ALCwasapiCapture *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; void *ptr; HRESULT hr; hr = CoCreateInstance(&CLSID_MMDeviceEnumerator, NULL, CLSCTX_INPROC_SERVER, &IID_IMMDeviceEnumerator, &ptr); if(SUCCEEDED(hr)) { IMMDeviceEnumerator *Enumerator = ptr; if(!self->devid) hr = IMMDeviceEnumerator_GetDefaultAudioEndpoint(Enumerator, eCapture, eMultimedia, &self->mmdev); else hr = IMMDeviceEnumerator_GetDevice(Enumerator, self->devid, &self->mmdev); IMMDeviceEnumerator_Release(Enumerator); Enumerator = NULL; } if(SUCCEEDED(hr)) hr = IMMDevice_Activate(self->mmdev, &IID_IAudioClient, CLSCTX_INPROC_SERVER, NULL, &ptr); if(SUCCEEDED(hr)) { self->client = ptr; if(alstr_empty(device->DeviceName)) get_device_name_and_guid(self->mmdev, &device->DeviceName, NULL); } if(FAILED(hr)) { if(self->mmdev) IMMDevice_Release(self->mmdev); self->mmdev = NULL; } return hr; } static void ALCwasapiCapture_closeProxy(ALCwasapiCapture *self) { if(self->client) IAudioClient_Release(self->client); self->client = NULL; if(self->mmdev) IMMDevice_Release(self->mmdev); self->mmdev = NULL; } static HRESULT ALCwasapiCapture_resetProxy(ALCwasapiCapture *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; WAVEFORMATEXTENSIBLE OutputType; WAVEFORMATEX *wfx = NULL; enum DevFmtType srcType; REFERENCE_TIME buf_time; UINT32 buffer_len; void *ptr = NULL; HRESULT hr; if(self->client) IAudioClient_Release(self->client); self->client = NULL; hr = IMMDevice_Activate(self->mmdev, &IID_IAudioClient, CLSCTX_INPROC_SERVER, NULL, &ptr); if(FAILED(hr)) { ERR("Failed to reactivate audio client: 0x%08lx\n", hr); return hr; } self->client = ptr; buf_time = ScaleCeil(device->UpdateSize*device->NumUpdates, REFTIME_PER_SEC, device->Frequency); // Make sure buffer is at least 100ms in size buf_time = maxu64(buf_time, REFTIME_PER_SEC/10); device->UpdateSize = (ALuint)ScaleCeil(buf_time, device->Frequency, REFTIME_PER_SEC) / device->NumUpdates; OutputType.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE; switch(device->FmtChans) { case DevFmtMono: OutputType.Format.nChannels = 1; OutputType.dwChannelMask = MONO; break; case DevFmtStereo: OutputType.Format.nChannels = 2; OutputType.dwChannelMask = STEREO; break; case DevFmtQuad: OutputType.Format.nChannels = 4; OutputType.dwChannelMask = QUAD; break; case DevFmtX51: OutputType.Format.nChannels = 6; OutputType.dwChannelMask = X5DOT1; break; case DevFmtX51Rear: OutputType.Format.nChannels = 6; OutputType.dwChannelMask = X5DOT1REAR; break; case DevFmtX61: OutputType.Format.nChannels = 7; OutputType.dwChannelMask = X6DOT1; break; case DevFmtX71: OutputType.Format.nChannels = 8; OutputType.dwChannelMask = X7DOT1; break; case DevFmtAmbi3D: return E_FAIL; } switch(device->FmtType) { /* NOTE: Signedness doesn't matter, the converter will handle it. */ case DevFmtByte: case DevFmtUByte: OutputType.Format.wBitsPerSample = 8; OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; break; case DevFmtShort: case DevFmtUShort: OutputType.Format.wBitsPerSample = 16; OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; break; case DevFmtInt: case DevFmtUInt: OutputType.Format.wBitsPerSample = 32; OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; break; case DevFmtFloat: OutputType.Format.wBitsPerSample = 32; OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT; break; } OutputType.Samples.wValidBitsPerSample = OutputType.Format.wBitsPerSample; OutputType.Format.nSamplesPerSec = device->Frequency; OutputType.Format.nBlockAlign = OutputType.Format.nChannels * OutputType.Format.wBitsPerSample / 8; OutputType.Format.nAvgBytesPerSec = OutputType.Format.nSamplesPerSec * OutputType.Format.nBlockAlign; OutputType.Format.cbSize = sizeof(OutputType) - sizeof(OutputType.Format); hr = IAudioClient_IsFormatSupported(self->client, AUDCLNT_SHAREMODE_SHARED, &OutputType.Format, &wfx ); if(FAILED(hr)) { ERR("Failed to check format support: 0x%08lx\n", hr); return hr; } DestroySampleConverter(&self->SampleConv); DestroyChannelConverter(&self->ChannelConv); if(wfx != NULL) { if(!(wfx->nChannels == OutputType.Format.nChannels || (wfx->nChannels == 1 && OutputType.Format.nChannels == 2) || (wfx->nChannels == 2 && OutputType.Format.nChannels == 1))) { ERR("Failed to get matching format, wanted: %s %s %uhz, got: %d channel%s %d-bit %luhz\n", DevFmtChannelsString(device->FmtChans), DevFmtTypeString(device->FmtType), device->Frequency, wfx->nChannels, (wfx->nChannels==1)?"":"s", wfx->wBitsPerSample, wfx->nSamplesPerSec); CoTaskMemFree(wfx); return E_FAIL; } if(!MakeExtensible(&OutputType, wfx)) { CoTaskMemFree(wfx); return E_FAIL; } CoTaskMemFree(wfx); wfx = NULL; } if(IsEqualGUID(&OutputType.SubFormat, &KSDATAFORMAT_SUBTYPE_PCM)) { if(OutputType.Format.wBitsPerSample == 8) srcType = DevFmtUByte; else if(OutputType.Format.wBitsPerSample == 16) srcType = DevFmtShort; else if(OutputType.Format.wBitsPerSample == 32) srcType = DevFmtInt; else { ERR("Unhandled integer bit depth: %d\n", OutputType.Format.wBitsPerSample); return E_FAIL; } } else if(IsEqualGUID(&OutputType.SubFormat, &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)) { if(OutputType.Format.wBitsPerSample == 32) srcType = DevFmtFloat; else { ERR("Unhandled float bit depth: %d\n", OutputType.Format.wBitsPerSample); return E_FAIL; } } else { ERR("Unhandled format sub-type\n"); return E_FAIL; } if(device->FmtChans == DevFmtMono && OutputType.Format.nChannels == 2) { self->ChannelConv = CreateChannelConverter(srcType, DevFmtStereo, device->FmtChans); if(!self->ChannelConv) { ERR("Failed to create %s stereo-to-mono converter\n", DevFmtTypeString(srcType)); return E_FAIL; } TRACE("Created %s stereo-to-mono converter\n", DevFmtTypeString(srcType)); /* The channel converter always outputs float, so change the input type * for the resampler/type-converter. */ srcType = DevFmtFloat; } else if(device->FmtChans == DevFmtStereo && OutputType.Format.nChannels == 1) { self->ChannelConv = CreateChannelConverter(srcType, DevFmtMono, device->FmtChans); if(!self->ChannelConv) { ERR("Failed to create %s mono-to-stereo converter\n", DevFmtTypeString(srcType)); return E_FAIL; } TRACE("Created %s mono-to-stereo converter\n", DevFmtTypeString(srcType)); srcType = DevFmtFloat; } if(device->Frequency != OutputType.Format.nSamplesPerSec || device->FmtType != srcType) { self->SampleConv = CreateSampleConverter( srcType, device->FmtType, ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder), OutputType.Format.nSamplesPerSec, device->Frequency ); if(!self->SampleConv) { ERR("Failed to create converter for %s format, dst: %s %uhz, src: %s %luhz\n", DevFmtChannelsString(device->FmtChans), DevFmtTypeString(device->FmtType), device->Frequency, DevFmtTypeString(srcType), OutputType.Format.nSamplesPerSec); return E_FAIL; } TRACE("Created converter for %s format, dst: %s %uhz, src: %s %luhz\n", DevFmtChannelsString(device->FmtChans), DevFmtTypeString(device->FmtType), device->Frequency, DevFmtTypeString(srcType), OutputType.Format.nSamplesPerSec); } hr = IAudioClient_Initialize(self->client, AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_EVENTCALLBACK, buf_time, 0, &OutputType.Format, NULL ); if(FAILED(hr)) { ERR("Failed to initialize audio client: 0x%08lx\n", hr); return hr; } hr = IAudioClient_GetBufferSize(self->client, &buffer_len); if(FAILED(hr)) { ERR("Failed to get buffer size: 0x%08lx\n", hr); return hr; } buffer_len = maxu(device->UpdateSize*device->NumUpdates, buffer_len); ll_ringbuffer_free(self->Ring); self->Ring = ll_ringbuffer_create(buffer_len, FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder), false ); if(!self->Ring) { ERR("Failed to allocate capture ring buffer\n"); return E_OUTOFMEMORY; } hr = IAudioClient_SetEventHandle(self->client, self->NotifyEvent); if(FAILED(hr)) { ERR("Failed to set event handle: 0x%08lx\n", hr); return hr; } return hr; } static ALCboolean ALCwasapiCapture_start(ALCwasapiCapture *self) { ThreadRequest req = { self->MsgEvent, 0 }; HRESULT hr = E_FAIL; if(PostThreadMessage(ThreadID, WM_USER_StartDevice, (WPARAM)&req, (LPARAM)STATIC_CAST(ALCwasapiProxy, self))) hr = WaitForResponse(&req); return SUCCEEDED(hr) ? ALC_TRUE : ALC_FALSE; } static HRESULT ALCwasapiCapture_startProxy(ALCwasapiCapture *self) { HRESULT hr; void *ptr; ResetEvent(self->NotifyEvent); hr = IAudioClient_Start(self->client); if(FAILED(hr)) { ERR("Failed to start audio client: 0x%08lx\n", hr); return hr; } hr = IAudioClient_GetService(self->client, &IID_IAudioCaptureClient, &ptr); if(SUCCEEDED(hr)) { self->capture = ptr; ATOMIC_STORE(&self->killNow, 0, almemory_order_release); if(althrd_create(&self->thread, ALCwasapiCapture_recordProc, self) != althrd_success) { ERR("Failed to start thread\n"); IAudioCaptureClient_Release(self->capture); self->capture = NULL; hr = E_FAIL; } } if(FAILED(hr)) { IAudioClient_Stop(self->client); IAudioClient_Reset(self->client); } return hr; } static void ALCwasapiCapture_stop(ALCwasapiCapture *self) { ThreadRequest req = { self->MsgEvent, 0 }; if(PostThreadMessage(ThreadID, WM_USER_StopDevice, (WPARAM)&req, (LPARAM)STATIC_CAST(ALCwasapiProxy, self))) (void)WaitForResponse(&req); } static void ALCwasapiCapture_stopProxy(ALCwasapiCapture *self) { int res; if(!self->capture) return; ATOMIC_STORE_SEQ(&self->killNow, 1); althrd_join(self->thread, &res); IAudioCaptureClient_Release(self->capture); self->capture = NULL; IAudioClient_Stop(self->client); IAudioClient_Reset(self->client); } ALuint ALCwasapiCapture_availableSamples(ALCwasapiCapture *self) { return (ALuint)ll_ringbuffer_read_space(self->Ring); } ALCenum ALCwasapiCapture_captureSamples(ALCwasapiCapture *self, ALCvoid *buffer, ALCuint samples) { if(ALCwasapiCapture_availableSamples(self) < samples) return ALC_INVALID_VALUE; ll_ringbuffer_read(self->Ring, buffer, samples); return ALC_NO_ERROR; } typedef struct ALCwasapiBackendFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } ALCwasapiBackendFactory; #define ALCWASAPIBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCwasapiBackendFactory, ALCbackendFactory) } } static ALCboolean ALCwasapiBackendFactory_init(ALCwasapiBackendFactory *self); static void ALCwasapiBackendFactory_deinit(ALCwasapiBackendFactory *self); static ALCboolean ALCwasapiBackendFactory_querySupport(ALCwasapiBackendFactory *self, ALCbackend_Type type); static void ALCwasapiBackendFactory_probe(ALCwasapiBackendFactory *self, enum DevProbe type, al_string *outnames); static ALCbackend* ALCwasapiBackendFactory_createBackend(ALCwasapiBackendFactory *self, ALCdevice *device, ALCbackend_Type type); DEFINE_ALCBACKENDFACTORY_VTABLE(ALCwasapiBackendFactory); static ALCboolean ALCwasapiBackendFactory_init(ALCwasapiBackendFactory* UNUSED(self)) { static HRESULT InitResult; VECTOR_INIT(PlaybackDevices); VECTOR_INIT(CaptureDevices); if(!ThreadHdl) { ThreadRequest req; InitResult = E_FAIL; req.FinishedEvt = CreateEventW(NULL, FALSE, FALSE, NULL); if(req.FinishedEvt == NULL) ERR("Failed to create event: %lu\n", GetLastError()); else { ThreadHdl = CreateThread(NULL, 0, ALCwasapiProxy_messageHandler, &req, 0, &ThreadID); if(ThreadHdl != NULL) InitResult = WaitForResponse(&req); CloseHandle(req.FinishedEvt); } } return SUCCEEDED(InitResult) ? ALC_TRUE : ALC_FALSE; } static void ALCwasapiBackendFactory_deinit(ALCwasapiBackendFactory* UNUSED(self)) { clear_devlist(&PlaybackDevices); VECTOR_DEINIT(PlaybackDevices); clear_devlist(&CaptureDevices); VECTOR_DEINIT(CaptureDevices); if(ThreadHdl) { TRACE("Sending WM_QUIT to Thread %04lx\n", ThreadID); PostThreadMessage(ThreadID, WM_QUIT, 0, 0); CloseHandle(ThreadHdl); ThreadHdl = NULL; } } static ALCboolean ALCwasapiBackendFactory_querySupport(ALCwasapiBackendFactory* UNUSED(self), ALCbackend_Type type) { if(type == ALCbackend_Playback || type == ALCbackend_Capture) return ALC_TRUE; return ALC_FALSE; } static void ALCwasapiBackendFactory_probe(ALCwasapiBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames) { ThreadRequest req = { NULL, 0 }; req.FinishedEvt = CreateEventW(NULL, FALSE, FALSE, NULL); if(req.FinishedEvt == NULL) ERR("Failed to create event: %lu\n", GetLastError()); else { HRESULT hr = E_FAIL; if(PostThreadMessage(ThreadID, WM_USER_Enumerate, (WPARAM)&req, type)) hr = WaitForResponse(&req); if(SUCCEEDED(hr)) switch(type) { #define APPEND_OUTNAME(e) do { \ if(!alstr_empty((e)->name)) \ alstr_append_range(outnames, VECTOR_BEGIN((e)->name), \ VECTOR_END((e)->name)+1); \ } while(0) case ALL_DEVICE_PROBE: VECTOR_FOR_EACH(const DevMap, PlaybackDevices, APPEND_OUTNAME); break; case CAPTURE_DEVICE_PROBE: VECTOR_FOR_EACH(const DevMap, CaptureDevices, APPEND_OUTNAME); break; #undef APPEND_OUTNAME } CloseHandle(req.FinishedEvt); req.FinishedEvt = NULL; } } static ALCbackend* ALCwasapiBackendFactory_createBackend(ALCwasapiBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) { if(type == ALCbackend_Playback) { ALCwasapiPlayback *backend; NEW_OBJ(backend, ALCwasapiPlayback)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } if(type == ALCbackend_Capture) { ALCwasapiCapture *backend; NEW_OBJ(backend, ALCwasapiCapture)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } return NULL; } ALCbackendFactory *ALCwasapiBackendFactory_getFactory(void) { static ALCwasapiBackendFactory factory = ALCWASAPIBACKENDFACTORY_INITIALIZER; return STATIC_CAST(ALCbackendFactory, &factory); } openal-soft-openal-soft-1.19.1/Alc/backends/wave.c000066400000000000000000000340161335774445300216610ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include #include "alMain.h" #include "alu.h" #include "alconfig.h" #include "threads.h" #include "compat.h" #include "backends/base.h" static const ALCchar waveDevice[] = "Wave File Writer"; static const ALubyte SUBTYPE_PCM[] = { 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x10, 0x00, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }; static const ALubyte SUBTYPE_FLOAT[] = { 0x03, 0x00, 0x00, 0x00, 0x00, 0x00, 0x10, 0x00, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }; static const ALubyte SUBTYPE_BFORMAT_PCM[] = { 0x01, 0x00, 0x00, 0x00, 0x21, 0x07, 0xd3, 0x11, 0x86, 0x44, 0xc8, 0xc1, 0xca, 0x00, 0x00, 0x00 }; static const ALubyte SUBTYPE_BFORMAT_FLOAT[] = { 0x03, 0x00, 0x00, 0x00, 0x21, 0x07, 0xd3, 0x11, 0x86, 0x44, 0xc8, 0xc1, 0xca, 0x00, 0x00, 0x00 }; static void fwrite16le(ALushort val, FILE *f) { ALubyte data[2] = { val&0xff, (val>>8)&0xff }; fwrite(data, 1, 2, f); } static void fwrite32le(ALuint val, FILE *f) { ALubyte data[4] = { val&0xff, (val>>8)&0xff, (val>>16)&0xff, (val>>24)&0xff }; fwrite(data, 1, 4, f); } typedef struct ALCwaveBackend { DERIVE_FROM_TYPE(ALCbackend); FILE *mFile; long mDataStart; ALvoid *mBuffer; ALuint mSize; ATOMIC(ALenum) killNow; althrd_t thread; } ALCwaveBackend; static int ALCwaveBackend_mixerProc(void *ptr); static void ALCwaveBackend_Construct(ALCwaveBackend *self, ALCdevice *device); static void ALCwaveBackend_Destruct(ALCwaveBackend *self); static ALCenum ALCwaveBackend_open(ALCwaveBackend *self, const ALCchar *name); static ALCboolean ALCwaveBackend_reset(ALCwaveBackend *self); static ALCboolean ALCwaveBackend_start(ALCwaveBackend *self); static void ALCwaveBackend_stop(ALCwaveBackend *self); static DECLARE_FORWARD2(ALCwaveBackend, ALCbackend, ALCenum, captureSamples, void*, ALCuint) static DECLARE_FORWARD(ALCwaveBackend, ALCbackend, ALCuint, availableSamples) static DECLARE_FORWARD(ALCwaveBackend, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(ALCwaveBackend, ALCbackend, void, lock) static DECLARE_FORWARD(ALCwaveBackend, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCwaveBackend) DEFINE_ALCBACKEND_VTABLE(ALCwaveBackend); static void ALCwaveBackend_Construct(ALCwaveBackend *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCwaveBackend, ALCbackend, self); self->mFile = NULL; self->mDataStart = -1; self->mBuffer = NULL; self->mSize = 0; ATOMIC_INIT(&self->killNow, AL_TRUE); } static void ALCwaveBackend_Destruct(ALCwaveBackend *self) { if(self->mFile) fclose(self->mFile); self->mFile = NULL; ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static int ALCwaveBackend_mixerProc(void *ptr) { ALCwaveBackend *self = (ALCwaveBackend*)ptr; ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; struct timespec now, start; ALint64 avail, done; ALuint frameSize; size_t fs; const long restTime = (long)((ALuint64)device->UpdateSize * 1000000000 / device->Frequency / 2); althrd_setname(althrd_current(), MIXER_THREAD_NAME); frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); done = 0; if(altimespec_get(&start, AL_TIME_UTC) != AL_TIME_UTC) { ERR("Failed to get starting time\n"); return 1; } while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) && ATOMIC_LOAD(&device->Connected, almemory_order_acquire)) { if(altimespec_get(&now, AL_TIME_UTC) != AL_TIME_UTC) { ERR("Failed to get current time\n"); return 1; } avail = (now.tv_sec - start.tv_sec) * device->Frequency; avail += (ALint64)(now.tv_nsec - start.tv_nsec) * device->Frequency / 1000000000; if(avail < done) { /* Oops, time skipped backwards. Reset the number of samples done * with one update available since we (likely) just came back from * sleeping. */ done = avail - device->UpdateSize; } if(avail-done < device->UpdateSize) al_nssleep(restTime); else while(avail-done >= device->UpdateSize) { ALCwaveBackend_lock(self); aluMixData(device, self->mBuffer, device->UpdateSize); ALCwaveBackend_unlock(self); done += device->UpdateSize; if(!IS_LITTLE_ENDIAN) { ALuint bytesize = BytesFromDevFmt(device->FmtType); ALuint i; if(bytesize == 2) { ALushort *samples = self->mBuffer; ALuint len = self->mSize / 2; for(i = 0;i < len;i++) { ALushort samp = samples[i]; samples[i] = (samp>>8) | (samp<<8); } } else if(bytesize == 4) { ALuint *samples = self->mBuffer; ALuint len = self->mSize / 4; for(i = 0;i < len;i++) { ALuint samp = samples[i]; samples[i] = (samp>>24) | ((samp>>8)&0x0000ff00) | ((samp<<8)&0x00ff0000) | (samp<<24); } } } fs = fwrite(self->mBuffer, frameSize, device->UpdateSize, self->mFile); (void)fs; if(ferror(self->mFile)) { ERR("Error writing to file\n"); ALCdevice_Lock(device); aluHandleDisconnect(device, "Failed to write playback samples"); ALCdevice_Unlock(device); break; } } } return 0; } static ALCenum ALCwaveBackend_open(ALCwaveBackend *self, const ALCchar *name) { ALCdevice *device; const char *fname; fname = GetConfigValue(NULL, "wave", "file", ""); if(!fname[0]) return ALC_INVALID_VALUE; if(!name) name = waveDevice; else if(strcmp(name, waveDevice) != 0) return ALC_INVALID_VALUE; self->mFile = al_fopen(fname, "wb"); if(!self->mFile) { ERR("Could not open file '%s': %s\n", fname, strerror(errno)); return ALC_INVALID_VALUE; } device = STATIC_CAST(ALCbackend, self)->mDevice; alstr_copy_cstr(&device->DeviceName, name); return ALC_NO_ERROR; } static ALCboolean ALCwaveBackend_reset(ALCwaveBackend *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; ALuint channels=0, bits=0, chanmask=0; int isbformat = 0; size_t val; fseek(self->mFile, 0, SEEK_SET); clearerr(self->mFile); if(GetConfigValueBool(NULL, "wave", "bformat", 0)) { device->FmtChans = DevFmtAmbi3D; device->AmbiOrder = 1; } switch(device->FmtType) { case DevFmtByte: device->FmtType = DevFmtUByte; break; case DevFmtUShort: device->FmtType = DevFmtShort; break; case DevFmtUInt: device->FmtType = DevFmtInt; break; case DevFmtUByte: case DevFmtShort: case DevFmtInt: case DevFmtFloat: break; } switch(device->FmtChans) { case DevFmtMono: chanmask = 0x04; break; case DevFmtStereo: chanmask = 0x01 | 0x02; break; case DevFmtQuad: chanmask = 0x01 | 0x02 | 0x10 | 0x20; break; case DevFmtX51: chanmask = 0x01 | 0x02 | 0x04 | 0x08 | 0x200 | 0x400; break; case DevFmtX51Rear: chanmask = 0x01 | 0x02 | 0x04 | 0x08 | 0x010 | 0x020; break; case DevFmtX61: chanmask = 0x01 | 0x02 | 0x04 | 0x08 | 0x100 | 0x200 | 0x400; break; case DevFmtX71: chanmask = 0x01 | 0x02 | 0x04 | 0x08 | 0x010 | 0x020 | 0x200 | 0x400; break; case DevFmtAmbi3D: /* .amb output requires FuMa */ device->AmbiLayout = AmbiLayout_FuMa; device->AmbiScale = AmbiNorm_FuMa; isbformat = 1; chanmask = 0; break; } bits = BytesFromDevFmt(device->FmtType) * 8; channels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); fputs("RIFF", self->mFile); fwrite32le(0xFFFFFFFF, self->mFile); // 'RIFF' header len; filled in at close fputs("WAVE", self->mFile); fputs("fmt ", self->mFile); fwrite32le(40, self->mFile); // 'fmt ' header len; 40 bytes for EXTENSIBLE // 16-bit val, format type id (extensible: 0xFFFE) fwrite16le(0xFFFE, self->mFile); // 16-bit val, channel count fwrite16le(channels, self->mFile); // 32-bit val, frequency fwrite32le(device->Frequency, self->mFile); // 32-bit val, bytes per second fwrite32le(device->Frequency * channels * bits / 8, self->mFile); // 16-bit val, frame size fwrite16le(channels * bits / 8, self->mFile); // 16-bit val, bits per sample fwrite16le(bits, self->mFile); // 16-bit val, extra byte count fwrite16le(22, self->mFile); // 16-bit val, valid bits per sample fwrite16le(bits, self->mFile); // 32-bit val, channel mask fwrite32le(chanmask, self->mFile); // 16 byte GUID, sub-type format val = fwrite((device->FmtType == DevFmtFloat) ? (isbformat ? SUBTYPE_BFORMAT_FLOAT : SUBTYPE_FLOAT) : (isbformat ? SUBTYPE_BFORMAT_PCM : SUBTYPE_PCM), 1, 16, self->mFile); (void)val; fputs("data", self->mFile); fwrite32le(0xFFFFFFFF, self->mFile); // 'data' header len; filled in at close if(ferror(self->mFile)) { ERR("Error writing header: %s\n", strerror(errno)); return ALC_FALSE; } self->mDataStart = ftell(self->mFile); SetDefaultWFXChannelOrder(device); return ALC_TRUE; } static ALCboolean ALCwaveBackend_start(ALCwaveBackend *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; self->mSize = device->UpdateSize * FrameSizeFromDevFmt( device->FmtChans, device->FmtType, device->AmbiOrder ); self->mBuffer = malloc(self->mSize); if(!self->mBuffer) { ERR("Buffer malloc failed\n"); return ALC_FALSE; } ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release); if(althrd_create(&self->thread, ALCwaveBackend_mixerProc, self) != althrd_success) { free(self->mBuffer); self->mBuffer = NULL; self->mSize = 0; return ALC_FALSE; } return ALC_TRUE; } static void ALCwaveBackend_stop(ALCwaveBackend *self) { ALuint dataLen; long size; int res; if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel)) return; althrd_join(self->thread, &res); free(self->mBuffer); self->mBuffer = NULL; size = ftell(self->mFile); if(size > 0) { dataLen = size - self->mDataStart; if(fseek(self->mFile, self->mDataStart-4, SEEK_SET) == 0) fwrite32le(dataLen, self->mFile); // 'data' header len if(fseek(self->mFile, 4, SEEK_SET) == 0) fwrite32le(size-8, self->mFile); // 'WAVE' header len } } typedef struct ALCwaveBackendFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } ALCwaveBackendFactory; #define ALCWAVEBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCwaveBackendFactory, ALCbackendFactory) } } ALCbackendFactory *ALCwaveBackendFactory_getFactory(void); static ALCboolean ALCwaveBackendFactory_init(ALCwaveBackendFactory *self); static DECLARE_FORWARD(ALCwaveBackendFactory, ALCbackendFactory, void, deinit) static ALCboolean ALCwaveBackendFactory_querySupport(ALCwaveBackendFactory *self, ALCbackend_Type type); static void ALCwaveBackendFactory_probe(ALCwaveBackendFactory *self, enum DevProbe type, al_string *outnames); static ALCbackend* ALCwaveBackendFactory_createBackend(ALCwaveBackendFactory *self, ALCdevice *device, ALCbackend_Type type); DEFINE_ALCBACKENDFACTORY_VTABLE(ALCwaveBackendFactory); ALCbackendFactory *ALCwaveBackendFactory_getFactory(void) { static ALCwaveBackendFactory factory = ALCWAVEBACKENDFACTORY_INITIALIZER; return STATIC_CAST(ALCbackendFactory, &factory); } static ALCboolean ALCwaveBackendFactory_init(ALCwaveBackendFactory* UNUSED(self)) { return ALC_TRUE; } static ALCboolean ALCwaveBackendFactory_querySupport(ALCwaveBackendFactory* UNUSED(self), ALCbackend_Type type) { if(type == ALCbackend_Playback) return !!ConfigValueExists(NULL, "wave", "file"); return ALC_FALSE; } static void ALCwaveBackendFactory_probe(ALCwaveBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames) { switch(type) { case ALL_DEVICE_PROBE: alstr_append_range(outnames, waveDevice, waveDevice+sizeof(waveDevice)); break; case CAPTURE_DEVICE_PROBE: break; } } static ALCbackend* ALCwaveBackendFactory_createBackend(ALCwaveBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) { if(type == ALCbackend_Playback) { ALCwaveBackend *backend; NEW_OBJ(backend, ALCwaveBackend)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } return NULL; } openal-soft-openal-soft-1.19.1/Alc/backends/winmm.c000066400000000000000000000602711335774445300220500ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include #include #include "alMain.h" #include "alu.h" #include "ringbuffer.h" #include "threads.h" #include "backends/base.h" #ifndef WAVE_FORMAT_IEEE_FLOAT #define WAVE_FORMAT_IEEE_FLOAT 0x0003 #endif #define DEVNAME_HEAD "OpenAL Soft on " static vector_al_string PlaybackDevices; static vector_al_string CaptureDevices; static void clear_devlist(vector_al_string *list) { VECTOR_FOR_EACH(al_string, *list, alstr_reset); VECTOR_RESIZE(*list, 0, 0); } static void ProbePlaybackDevices(void) { ALuint numdevs; ALuint i; clear_devlist(&PlaybackDevices); numdevs = waveOutGetNumDevs(); VECTOR_RESIZE(PlaybackDevices, 0, numdevs); for(i = 0;i < numdevs;i++) { WAVEOUTCAPSW WaveCaps; const al_string *iter; al_string dname; AL_STRING_INIT(dname); if(waveOutGetDevCapsW(i, &WaveCaps, sizeof(WaveCaps)) == MMSYSERR_NOERROR) { ALuint count = 0; while(1) { alstr_copy_cstr(&dname, DEVNAME_HEAD); alstr_append_wcstr(&dname, WaveCaps.szPname); if(count != 0) { char str[64]; snprintf(str, sizeof(str), " #%d", count+1); alstr_append_cstr(&dname, str); } count++; #define MATCH_ENTRY(i) (alstr_cmp(dname, *(i)) == 0) VECTOR_FIND_IF(iter, const al_string, PlaybackDevices, MATCH_ENTRY); if(iter == VECTOR_END(PlaybackDevices)) break; #undef MATCH_ENTRY } TRACE("Got device \"%s\", ID %u\n", alstr_get_cstr(dname), i); } VECTOR_PUSH_BACK(PlaybackDevices, dname); } } static void ProbeCaptureDevices(void) { ALuint numdevs; ALuint i; clear_devlist(&CaptureDevices); numdevs = waveInGetNumDevs(); VECTOR_RESIZE(CaptureDevices, 0, numdevs); for(i = 0;i < numdevs;i++) { WAVEINCAPSW WaveCaps; const al_string *iter; al_string dname; AL_STRING_INIT(dname); if(waveInGetDevCapsW(i, &WaveCaps, sizeof(WaveCaps)) == MMSYSERR_NOERROR) { ALuint count = 0; while(1) { alstr_copy_cstr(&dname, DEVNAME_HEAD); alstr_append_wcstr(&dname, WaveCaps.szPname); if(count != 0) { char str[64]; snprintf(str, sizeof(str), " #%d", count+1); alstr_append_cstr(&dname, str); } count++; #define MATCH_ENTRY(i) (alstr_cmp(dname, *(i)) == 0) VECTOR_FIND_IF(iter, const al_string, CaptureDevices, MATCH_ENTRY); if(iter == VECTOR_END(CaptureDevices)) break; #undef MATCH_ENTRY } TRACE("Got device \"%s\", ID %u\n", alstr_get_cstr(dname), i); } VECTOR_PUSH_BACK(CaptureDevices, dname); } } typedef struct ALCwinmmPlayback { DERIVE_FROM_TYPE(ALCbackend); RefCount WaveBuffersCommitted; WAVEHDR WaveBuffer[4]; HWAVEOUT OutHdl; WAVEFORMATEX Format; ATOMIC(ALenum) killNow; althrd_t thread; } ALCwinmmPlayback; static void ALCwinmmPlayback_Construct(ALCwinmmPlayback *self, ALCdevice *device); static void ALCwinmmPlayback_Destruct(ALCwinmmPlayback *self); static void CALLBACK ALCwinmmPlayback_waveOutProc(HWAVEOUT device, UINT msg, DWORD_PTR instance, DWORD_PTR param1, DWORD_PTR param2); static int ALCwinmmPlayback_mixerProc(void *arg); static ALCenum ALCwinmmPlayback_open(ALCwinmmPlayback *self, const ALCchar *name); static ALCboolean ALCwinmmPlayback_reset(ALCwinmmPlayback *self); static ALCboolean ALCwinmmPlayback_start(ALCwinmmPlayback *self); static void ALCwinmmPlayback_stop(ALCwinmmPlayback *self); static DECLARE_FORWARD2(ALCwinmmPlayback, ALCbackend, ALCenum, captureSamples, ALCvoid*, ALCuint) static DECLARE_FORWARD(ALCwinmmPlayback, ALCbackend, ALCuint, availableSamples) static DECLARE_FORWARD(ALCwinmmPlayback, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(ALCwinmmPlayback, ALCbackend, void, lock) static DECLARE_FORWARD(ALCwinmmPlayback, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCwinmmPlayback) DEFINE_ALCBACKEND_VTABLE(ALCwinmmPlayback); static void ALCwinmmPlayback_Construct(ALCwinmmPlayback *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCwinmmPlayback, ALCbackend, self); InitRef(&self->WaveBuffersCommitted, 0); self->OutHdl = NULL; ATOMIC_INIT(&self->killNow, AL_TRUE); } static void ALCwinmmPlayback_Destruct(ALCwinmmPlayback *self) { if(self->OutHdl) waveOutClose(self->OutHdl); self->OutHdl = 0; ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } /* ALCwinmmPlayback_waveOutProc * * Posts a message to 'ALCwinmmPlayback_mixerProc' everytime a WaveOut Buffer * is completed and returns to the application (for more data) */ static void CALLBACK ALCwinmmPlayback_waveOutProc(HWAVEOUT UNUSED(device), UINT msg, DWORD_PTR instance, DWORD_PTR param1, DWORD_PTR UNUSED(param2)) { ALCwinmmPlayback *self = (ALCwinmmPlayback*)instance; if(msg != WOM_DONE) return; DecrementRef(&self->WaveBuffersCommitted); PostThreadMessage(self->thread, msg, 0, param1); } FORCE_ALIGN static int ALCwinmmPlayback_mixerProc(void *arg) { ALCwinmmPlayback *self = arg; ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; WAVEHDR *WaveHdr; MSG msg; SetRTPriority(); althrd_setname(althrd_current(), MIXER_THREAD_NAME); while(GetMessage(&msg, NULL, 0, 0)) { if(msg.message != WOM_DONE) continue; if(ATOMIC_LOAD(&self->killNow, almemory_order_acquire)) { if(ReadRef(&self->WaveBuffersCommitted) == 0) break; continue; } WaveHdr = ((WAVEHDR*)msg.lParam); ALCwinmmPlayback_lock(self); aluMixData(device, WaveHdr->lpData, WaveHdr->dwBufferLength / self->Format.nBlockAlign); ALCwinmmPlayback_unlock(self); // Send buffer back to play more data waveOutWrite(self->OutHdl, WaveHdr, sizeof(WAVEHDR)); IncrementRef(&self->WaveBuffersCommitted); } return 0; } static ALCenum ALCwinmmPlayback_open(ALCwinmmPlayback *self, const ALCchar *deviceName) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; const al_string *iter; UINT DeviceID; MMRESULT res; if(VECTOR_SIZE(PlaybackDevices) == 0) ProbePlaybackDevices(); // Find the Device ID matching the deviceName if valid #define MATCH_DEVNAME(iter) (!alstr_empty(*(iter)) && \ (!deviceName || alstr_cmp_cstr(*(iter), deviceName) == 0)) VECTOR_FIND_IF(iter, const al_string, PlaybackDevices, MATCH_DEVNAME); if(iter == VECTOR_END(PlaybackDevices)) return ALC_INVALID_VALUE; #undef MATCH_DEVNAME DeviceID = (UINT)(iter - VECTOR_BEGIN(PlaybackDevices)); retry_open: memset(&self->Format, 0, sizeof(WAVEFORMATEX)); if(device->FmtType == DevFmtFloat) { self->Format.wFormatTag = WAVE_FORMAT_IEEE_FLOAT; self->Format.wBitsPerSample = 32; } else { self->Format.wFormatTag = WAVE_FORMAT_PCM; if(device->FmtType == DevFmtUByte || device->FmtType == DevFmtByte) self->Format.wBitsPerSample = 8; else self->Format.wBitsPerSample = 16; } self->Format.nChannels = ((device->FmtChans == DevFmtMono) ? 1 : 2); self->Format.nBlockAlign = self->Format.wBitsPerSample * self->Format.nChannels / 8; self->Format.nSamplesPerSec = device->Frequency; self->Format.nAvgBytesPerSec = self->Format.nSamplesPerSec * self->Format.nBlockAlign; self->Format.cbSize = 0; if((res=waveOutOpen(&self->OutHdl, DeviceID, &self->Format, (DWORD_PTR)&ALCwinmmPlayback_waveOutProc, (DWORD_PTR)self, CALLBACK_FUNCTION)) != MMSYSERR_NOERROR) { if(device->FmtType == DevFmtFloat) { device->FmtType = DevFmtShort; goto retry_open; } ERR("waveOutOpen failed: %u\n", res); goto failure; } alstr_copy(&device->DeviceName, VECTOR_ELEM(PlaybackDevices, DeviceID)); return ALC_NO_ERROR; failure: if(self->OutHdl) waveOutClose(self->OutHdl); self->OutHdl = NULL; return ALC_INVALID_VALUE; } static ALCboolean ALCwinmmPlayback_reset(ALCwinmmPlayback *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; device->UpdateSize = (ALuint)((ALuint64)device->UpdateSize * self->Format.nSamplesPerSec / device->Frequency); device->UpdateSize = (device->UpdateSize*device->NumUpdates + 3) / 4; device->NumUpdates = 4; device->Frequency = self->Format.nSamplesPerSec; if(self->Format.wFormatTag == WAVE_FORMAT_IEEE_FLOAT) { if(self->Format.wBitsPerSample == 32) device->FmtType = DevFmtFloat; else { ERR("Unhandled IEEE float sample depth: %d\n", self->Format.wBitsPerSample); return ALC_FALSE; } } else if(self->Format.wFormatTag == WAVE_FORMAT_PCM) { if(self->Format.wBitsPerSample == 16) device->FmtType = DevFmtShort; else if(self->Format.wBitsPerSample == 8) device->FmtType = DevFmtUByte; else { ERR("Unhandled PCM sample depth: %d\n", self->Format.wBitsPerSample); return ALC_FALSE; } } else { ERR("Unhandled format tag: 0x%04x\n", self->Format.wFormatTag); return ALC_FALSE; } if(self->Format.nChannels == 2) device->FmtChans = DevFmtStereo; else if(self->Format.nChannels == 1) device->FmtChans = DevFmtMono; else { ERR("Unhandled channel count: %d\n", self->Format.nChannels); return ALC_FALSE; } SetDefaultWFXChannelOrder(device); return ALC_TRUE; } static ALCboolean ALCwinmmPlayback_start(ALCwinmmPlayback *self) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; ALbyte *BufferData; ALint BufferSize; ALuint i; ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release); if(althrd_create(&self->thread, ALCwinmmPlayback_mixerProc, self) != althrd_success) return ALC_FALSE; InitRef(&self->WaveBuffersCommitted, 0); // Create 4 Buffers BufferSize = device->UpdateSize*device->NumUpdates / 4; BufferSize *= FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); BufferData = calloc(4, BufferSize); for(i = 0;i < 4;i++) { memset(&self->WaveBuffer[i], 0, sizeof(WAVEHDR)); self->WaveBuffer[i].dwBufferLength = BufferSize; self->WaveBuffer[i].lpData = ((i==0) ? (CHAR*)BufferData : (self->WaveBuffer[i-1].lpData + self->WaveBuffer[i-1].dwBufferLength)); waveOutPrepareHeader(self->OutHdl, &self->WaveBuffer[i], sizeof(WAVEHDR)); waveOutWrite(self->OutHdl, &self->WaveBuffer[i], sizeof(WAVEHDR)); IncrementRef(&self->WaveBuffersCommitted); } return ALC_TRUE; } static void ALCwinmmPlayback_stop(ALCwinmmPlayback *self) { void *buffer = NULL; int i; if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel)) return; althrd_join(self->thread, &i); // Release the wave buffers for(i = 0;i < 4;i++) { waveOutUnprepareHeader(self->OutHdl, &self->WaveBuffer[i], sizeof(WAVEHDR)); if(i == 0) buffer = self->WaveBuffer[i].lpData; self->WaveBuffer[i].lpData = NULL; } free(buffer); } typedef struct ALCwinmmCapture { DERIVE_FROM_TYPE(ALCbackend); RefCount WaveBuffersCommitted; WAVEHDR WaveBuffer[4]; HWAVEIN InHdl; ll_ringbuffer_t *Ring; WAVEFORMATEX Format; ATOMIC(ALenum) killNow; althrd_t thread; } ALCwinmmCapture; static void ALCwinmmCapture_Construct(ALCwinmmCapture *self, ALCdevice *device); static void ALCwinmmCapture_Destruct(ALCwinmmCapture *self); static void CALLBACK ALCwinmmCapture_waveInProc(HWAVEIN device, UINT msg, DWORD_PTR instance, DWORD_PTR param1, DWORD_PTR param2); static int ALCwinmmCapture_captureProc(void *arg); static ALCenum ALCwinmmCapture_open(ALCwinmmCapture *self, const ALCchar *name); static DECLARE_FORWARD(ALCwinmmCapture, ALCbackend, ALCboolean, reset) static ALCboolean ALCwinmmCapture_start(ALCwinmmCapture *self); static void ALCwinmmCapture_stop(ALCwinmmCapture *self); static ALCenum ALCwinmmCapture_captureSamples(ALCwinmmCapture *self, ALCvoid *buffer, ALCuint samples); static ALCuint ALCwinmmCapture_availableSamples(ALCwinmmCapture *self); static DECLARE_FORWARD(ALCwinmmCapture, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(ALCwinmmCapture, ALCbackend, void, lock) static DECLARE_FORWARD(ALCwinmmCapture, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(ALCwinmmCapture) DEFINE_ALCBACKEND_VTABLE(ALCwinmmCapture); static void ALCwinmmCapture_Construct(ALCwinmmCapture *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCwinmmCapture, ALCbackend, self); InitRef(&self->WaveBuffersCommitted, 0); self->InHdl = NULL; ATOMIC_INIT(&self->killNow, AL_TRUE); } static void ALCwinmmCapture_Destruct(ALCwinmmCapture *self) { void *buffer = NULL; int i; /* Tell the processing thread to quit and wait for it to do so. */ if(!ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel)) { PostThreadMessage(self->thread, WM_QUIT, 0, 0); althrd_join(self->thread, &i); /* Make sure capture is stopped and all pending buffers are flushed. */ waveInReset(self->InHdl); // Release the wave buffers for(i = 0;i < 4;i++) { waveInUnprepareHeader(self->InHdl, &self->WaveBuffer[i], sizeof(WAVEHDR)); if(i == 0) buffer = self->WaveBuffer[i].lpData; self->WaveBuffer[i].lpData = NULL; } free(buffer); } ll_ringbuffer_free(self->Ring); self->Ring = NULL; // Close the Wave device if(self->InHdl) waveInClose(self->InHdl); self->InHdl = 0; ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } /* ALCwinmmCapture_waveInProc * * Posts a message to 'ALCwinmmCapture_captureProc' everytime a WaveIn Buffer * is completed and returns to the application (with more data). */ static void CALLBACK ALCwinmmCapture_waveInProc(HWAVEIN UNUSED(device), UINT msg, DWORD_PTR instance, DWORD_PTR param1, DWORD_PTR UNUSED(param2)) { ALCwinmmCapture *self = (ALCwinmmCapture*)instance; if(msg != WIM_DATA) return; DecrementRef(&self->WaveBuffersCommitted); PostThreadMessage(self->thread, msg, 0, param1); } static int ALCwinmmCapture_captureProc(void *arg) { ALCwinmmCapture *self = arg; WAVEHDR *WaveHdr; MSG msg; althrd_setname(althrd_current(), RECORD_THREAD_NAME); while(GetMessage(&msg, NULL, 0, 0)) { if(msg.message != WIM_DATA) continue; /* Don't wait for other buffers to finish before quitting. We're * closing so we don't need them. */ if(ATOMIC_LOAD(&self->killNow, almemory_order_acquire)) break; WaveHdr = ((WAVEHDR*)msg.lParam); ll_ringbuffer_write(self->Ring, WaveHdr->lpData, WaveHdr->dwBytesRecorded / self->Format.nBlockAlign ); // Send buffer back to capture more data waveInAddBuffer(self->InHdl, WaveHdr, sizeof(WAVEHDR)); IncrementRef(&self->WaveBuffersCommitted); } return 0; } static ALCenum ALCwinmmCapture_open(ALCwinmmCapture *self, const ALCchar *name) { ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice; const al_string *iter; ALbyte *BufferData = NULL; DWORD CapturedDataSize; ALint BufferSize; UINT DeviceID; MMRESULT res; ALuint i; if(VECTOR_SIZE(CaptureDevices) == 0) ProbeCaptureDevices(); // Find the Device ID matching the deviceName if valid #define MATCH_DEVNAME(iter) (!alstr_empty(*(iter)) && (!name || alstr_cmp_cstr(*iter, name) == 0)) VECTOR_FIND_IF(iter, const al_string, CaptureDevices, MATCH_DEVNAME); if(iter == VECTOR_END(CaptureDevices)) return ALC_INVALID_VALUE; #undef MATCH_DEVNAME DeviceID = (UINT)(iter - VECTOR_BEGIN(CaptureDevices)); switch(device->FmtChans) { case DevFmtMono: case DevFmtStereo: break; case DevFmtQuad: case DevFmtX51: case DevFmtX51Rear: case DevFmtX61: case DevFmtX71: case DevFmtAmbi3D: return ALC_INVALID_ENUM; } switch(device->FmtType) { case DevFmtUByte: case DevFmtShort: case DevFmtInt: case DevFmtFloat: break; case DevFmtByte: case DevFmtUShort: case DevFmtUInt: return ALC_INVALID_ENUM; } memset(&self->Format, 0, sizeof(WAVEFORMATEX)); self->Format.wFormatTag = ((device->FmtType == DevFmtFloat) ? WAVE_FORMAT_IEEE_FLOAT : WAVE_FORMAT_PCM); self->Format.nChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); self->Format.wBitsPerSample = BytesFromDevFmt(device->FmtType) * 8; self->Format.nBlockAlign = self->Format.wBitsPerSample * self->Format.nChannels / 8; self->Format.nSamplesPerSec = device->Frequency; self->Format.nAvgBytesPerSec = self->Format.nSamplesPerSec * self->Format.nBlockAlign; self->Format.cbSize = 0; if((res=waveInOpen(&self->InHdl, DeviceID, &self->Format, (DWORD_PTR)&ALCwinmmCapture_waveInProc, (DWORD_PTR)self, CALLBACK_FUNCTION)) != MMSYSERR_NOERROR) { ERR("waveInOpen failed: %u\n", res); goto failure; } // Allocate circular memory buffer for the captured audio CapturedDataSize = device->UpdateSize*device->NumUpdates; // Make sure circular buffer is at least 100ms in size if(CapturedDataSize < (self->Format.nSamplesPerSec / 10)) CapturedDataSize = self->Format.nSamplesPerSec / 10; self->Ring = ll_ringbuffer_create(CapturedDataSize, self->Format.nBlockAlign, false); if(!self->Ring) goto failure; InitRef(&self->WaveBuffersCommitted, 0); // Create 4 Buffers of 50ms each BufferSize = self->Format.nAvgBytesPerSec / 20; BufferSize -= (BufferSize % self->Format.nBlockAlign); BufferData = calloc(4, BufferSize); if(!BufferData) goto failure; for(i = 0;i < 4;i++) { memset(&self->WaveBuffer[i], 0, sizeof(WAVEHDR)); self->WaveBuffer[i].dwBufferLength = BufferSize; self->WaveBuffer[i].lpData = ((i==0) ? (CHAR*)BufferData : (self->WaveBuffer[i-1].lpData + self->WaveBuffer[i-1].dwBufferLength)); self->WaveBuffer[i].dwFlags = 0; self->WaveBuffer[i].dwLoops = 0; waveInPrepareHeader(self->InHdl, &self->WaveBuffer[i], sizeof(WAVEHDR)); waveInAddBuffer(self->InHdl, &self->WaveBuffer[i], sizeof(WAVEHDR)); IncrementRef(&self->WaveBuffersCommitted); } ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release); if(althrd_create(&self->thread, ALCwinmmCapture_captureProc, self) != althrd_success) goto failure; alstr_copy(&device->DeviceName, VECTOR_ELEM(CaptureDevices, DeviceID)); return ALC_NO_ERROR; failure: if(BufferData) { for(i = 0;i < 4;i++) waveInUnprepareHeader(self->InHdl, &self->WaveBuffer[i], sizeof(WAVEHDR)); free(BufferData); } ll_ringbuffer_free(self->Ring); self->Ring = NULL; if(self->InHdl) waveInClose(self->InHdl); self->InHdl = NULL; return ALC_INVALID_VALUE; } static ALCboolean ALCwinmmCapture_start(ALCwinmmCapture *self) { waveInStart(self->InHdl); return ALC_TRUE; } static void ALCwinmmCapture_stop(ALCwinmmCapture *self) { waveInStop(self->InHdl); } static ALCenum ALCwinmmCapture_captureSamples(ALCwinmmCapture *self, ALCvoid *buffer, ALCuint samples) { ll_ringbuffer_read(self->Ring, buffer, samples); return ALC_NO_ERROR; } static ALCuint ALCwinmmCapture_availableSamples(ALCwinmmCapture *self) { return (ALCuint)ll_ringbuffer_read_space(self->Ring); } typedef struct ALCwinmmBackendFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } ALCwinmmBackendFactory; #define ALCWINMMBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCwinmmBackendFactory, ALCbackendFactory) } } static ALCboolean ALCwinmmBackendFactory_init(ALCwinmmBackendFactory *self); static void ALCwinmmBackendFactory_deinit(ALCwinmmBackendFactory *self); static ALCboolean ALCwinmmBackendFactory_querySupport(ALCwinmmBackendFactory *self, ALCbackend_Type type); static void ALCwinmmBackendFactory_probe(ALCwinmmBackendFactory *self, enum DevProbe type, al_string *outnames); static ALCbackend* ALCwinmmBackendFactory_createBackend(ALCwinmmBackendFactory *self, ALCdevice *device, ALCbackend_Type type); DEFINE_ALCBACKENDFACTORY_VTABLE(ALCwinmmBackendFactory); static ALCboolean ALCwinmmBackendFactory_init(ALCwinmmBackendFactory* UNUSED(self)) { VECTOR_INIT(PlaybackDevices); VECTOR_INIT(CaptureDevices); return ALC_TRUE; } static void ALCwinmmBackendFactory_deinit(ALCwinmmBackendFactory* UNUSED(self)) { clear_devlist(&PlaybackDevices); VECTOR_DEINIT(PlaybackDevices); clear_devlist(&CaptureDevices); VECTOR_DEINIT(CaptureDevices); } static ALCboolean ALCwinmmBackendFactory_querySupport(ALCwinmmBackendFactory* UNUSED(self), ALCbackend_Type type) { if(type == ALCbackend_Playback || type == ALCbackend_Capture) return ALC_TRUE; return ALC_FALSE; } static void ALCwinmmBackendFactory_probe(ALCwinmmBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames) { switch(type) { #define APPEND_OUTNAME(n) do { \ if(!alstr_empty(*(n))) \ alstr_append_range(outnames, VECTOR_BEGIN(*(n)), VECTOR_END(*(n))+1); \ } while(0) case ALL_DEVICE_PROBE: ProbePlaybackDevices(); VECTOR_FOR_EACH(const al_string, PlaybackDevices, APPEND_OUTNAME); break; case CAPTURE_DEVICE_PROBE: ProbeCaptureDevices(); VECTOR_FOR_EACH(const al_string, CaptureDevices, APPEND_OUTNAME); break; #undef APPEND_OUTNAME } } static ALCbackend* ALCwinmmBackendFactory_createBackend(ALCwinmmBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) { if(type == ALCbackend_Playback) { ALCwinmmPlayback *backend; NEW_OBJ(backend, ALCwinmmPlayback)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } if(type == ALCbackend_Capture) { ALCwinmmCapture *backend; NEW_OBJ(backend, ALCwinmmCapture)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } return NULL; } ALCbackendFactory *ALCwinmmBackendFactory_getFactory(void) { static ALCwinmmBackendFactory factory = ALCWINMMBACKENDFACTORY_INITIALIZER; return STATIC_CAST(ALCbackendFactory, &factory); } openal-soft-openal-soft-1.19.1/Alc/bformatdec.c000066400000000000000000000415571335774445300212630ustar00rootroot00000000000000 #include "config.h" #include "bformatdec.h" #include "ambdec.h" #include "filters/splitter.h" #include "alu.h" #include "bool.h" #include "threads.h" #include "almalloc.h" /* NOTE: These are scale factors as applied to Ambisonics content. Decoder * coefficients should be divided by these values to get proper N3D scalings. */ const ALfloat N3D2N3DScale[MAX_AMBI_COEFFS] = { 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f }; const ALfloat SN3D2N3DScale[MAX_AMBI_COEFFS] = { 1.000000000f, /* ACN 0 (W), sqrt(1) */ 1.732050808f, /* ACN 1 (Y), sqrt(3) */ 1.732050808f, /* ACN 2 (Z), sqrt(3) */ 1.732050808f, /* ACN 3 (X), sqrt(3) */ 2.236067978f, /* ACN 4 (V), sqrt(5) */ 2.236067978f, /* ACN 5 (T), sqrt(5) */ 2.236067978f, /* ACN 6 (R), sqrt(5) */ 2.236067978f, /* ACN 7 (S), sqrt(5) */ 2.236067978f, /* ACN 8 (U), sqrt(5) */ 2.645751311f, /* ACN 9 (Q), sqrt(7) */ 2.645751311f, /* ACN 10 (O), sqrt(7) */ 2.645751311f, /* ACN 11 (M), sqrt(7) */ 2.645751311f, /* ACN 12 (K), sqrt(7) */ 2.645751311f, /* ACN 13 (L), sqrt(7) */ 2.645751311f, /* ACN 14 (N), sqrt(7) */ 2.645751311f, /* ACN 15 (P), sqrt(7) */ }; const ALfloat FuMa2N3DScale[MAX_AMBI_COEFFS] = { 1.414213562f, /* ACN 0 (W), sqrt(2) */ 1.732050808f, /* ACN 1 (Y), sqrt(3) */ 1.732050808f, /* ACN 2 (Z), sqrt(3) */ 1.732050808f, /* ACN 3 (X), sqrt(3) */ 1.936491673f, /* ACN 4 (V), sqrt(15)/2 */ 1.936491673f, /* ACN 5 (T), sqrt(15)/2 */ 2.236067978f, /* ACN 6 (R), sqrt(5) */ 1.936491673f, /* ACN 7 (S), sqrt(15)/2 */ 1.936491673f, /* ACN 8 (U), sqrt(15)/2 */ 2.091650066f, /* ACN 9 (Q), sqrt(35/8) */ 1.972026594f, /* ACN 10 (O), sqrt(35)/3 */ 2.231093404f, /* ACN 11 (M), sqrt(224/45) */ 2.645751311f, /* ACN 12 (K), sqrt(7) */ 2.231093404f, /* ACN 13 (L), sqrt(224/45) */ 1.972026594f, /* ACN 14 (N), sqrt(35)/3 */ 2.091650066f, /* ACN 15 (P), sqrt(35/8) */ }; #define HF_BAND 0 #define LF_BAND 1 #define NUM_BANDS 2 /* These points are in AL coordinates! */ static const ALfloat Ambi3DPoints[8][3] = { { -0.577350269f, 0.577350269f, -0.577350269f }, { 0.577350269f, 0.577350269f, -0.577350269f }, { -0.577350269f, 0.577350269f, 0.577350269f }, { 0.577350269f, 0.577350269f, 0.577350269f }, { -0.577350269f, -0.577350269f, -0.577350269f }, { 0.577350269f, -0.577350269f, -0.577350269f }, { -0.577350269f, -0.577350269f, 0.577350269f }, { 0.577350269f, -0.577350269f, 0.577350269f }, }; static const ALfloat Ambi3DDecoder[8][MAX_AMBI_COEFFS] = { { 0.125f, 0.125f, 0.125f, 0.125f }, { 0.125f, -0.125f, 0.125f, 0.125f }, { 0.125f, 0.125f, 0.125f, -0.125f }, { 0.125f, -0.125f, 0.125f, -0.125f }, { 0.125f, 0.125f, -0.125f, 0.125f }, { 0.125f, -0.125f, -0.125f, 0.125f }, { 0.125f, 0.125f, -0.125f, -0.125f }, { 0.125f, -0.125f, -0.125f, -0.125f }, }; static const ALfloat Ambi3DDecoderHFScale[MAX_AMBI_COEFFS] = { 2.0f, 1.15470054f, 1.15470054f, 1.15470054f }; /* NOTE: BandSplitter filters are unused with single-band decoding */ typedef struct BFormatDec { ALuint Enabled; /* Bitfield of enabled channels. */ union { alignas(16) ALfloat Dual[MAX_OUTPUT_CHANNELS][NUM_BANDS][MAX_AMBI_COEFFS]; alignas(16) ALfloat Single[MAX_OUTPUT_CHANNELS][MAX_AMBI_COEFFS]; } Matrix; BandSplitter XOver[MAX_AMBI_COEFFS]; ALfloat (*Samples)[BUFFERSIZE]; /* These two alias into Samples */ ALfloat (*SamplesHF)[BUFFERSIZE]; ALfloat (*SamplesLF)[BUFFERSIZE]; alignas(16) ALfloat ChannelMix[BUFFERSIZE]; struct { BandSplitter XOver; ALfloat Gains[NUM_BANDS]; } UpSampler[4]; ALsizei NumChannels; ALboolean DualBand; } BFormatDec; BFormatDec *bformatdec_alloc() { return al_calloc(16, sizeof(BFormatDec)); } void bformatdec_free(BFormatDec **dec) { if(dec && *dec) { al_free((*dec)->Samples); (*dec)->Samples = NULL; (*dec)->SamplesHF = NULL; (*dec)->SamplesLF = NULL; al_free(*dec); *dec = NULL; } } void bformatdec_reset(BFormatDec *dec, const AmbDecConf *conf, ALsizei chancount, ALuint srate, const ALsizei chanmap[MAX_OUTPUT_CHANNELS]) { static const ALsizei map2DTo3D[MAX_AMBI2D_COEFFS] = { 0, 1, 3, 4, 8, 9, 15 }; const ALfloat *coeff_scale = N3D2N3DScale; bool periphonic; ALfloat ratio; ALsizei i; al_free(dec->Samples); dec->Samples = NULL; dec->SamplesHF = NULL; dec->SamplesLF = NULL; dec->NumChannels = chancount; dec->Samples = al_calloc(16, dec->NumChannels*2 * sizeof(dec->Samples[0])); dec->SamplesHF = dec->Samples; dec->SamplesLF = dec->SamplesHF + dec->NumChannels; dec->Enabled = 0; for(i = 0;i < conf->NumSpeakers;i++) dec->Enabled |= 1 << chanmap[i]; if(conf->CoeffScale == ADS_SN3D) coeff_scale = SN3D2N3DScale; else if(conf->CoeffScale == ADS_FuMa) coeff_scale = FuMa2N3DScale; memset(dec->UpSampler, 0, sizeof(dec->UpSampler)); ratio = 400.0f / (ALfloat)srate; for(i = 0;i < 4;i++) bandsplit_init(&dec->UpSampler[i].XOver, ratio); if((conf->ChanMask&AMBI_PERIPHONIC_MASK)) { periphonic = true; dec->UpSampler[0].Gains[HF_BAND] = (conf->ChanMask > 0x1ff) ? W_SCALE_3H3P : (conf->ChanMask > 0xf) ? W_SCALE_2H2P : 1.0f; dec->UpSampler[0].Gains[LF_BAND] = 1.0f; for(i = 1;i < 4;i++) { dec->UpSampler[i].Gains[HF_BAND] = (conf->ChanMask > 0x1ff) ? XYZ_SCALE_3H3P : (conf->ChanMask > 0xf) ? XYZ_SCALE_2H2P : 1.0f; dec->UpSampler[i].Gains[LF_BAND] = 1.0f; } } else { periphonic = false; dec->UpSampler[0].Gains[HF_BAND] = (conf->ChanMask > 0x1ff) ? W_SCALE_3H0P : (conf->ChanMask > 0xf) ? W_SCALE_2H0P : 1.0f; dec->UpSampler[0].Gains[LF_BAND] = 1.0f; for(i = 1;i < 3;i++) { dec->UpSampler[i].Gains[HF_BAND] = (conf->ChanMask > 0x1ff) ? XYZ_SCALE_3H0P : (conf->ChanMask > 0xf) ? XYZ_SCALE_2H0P : 1.0f; dec->UpSampler[i].Gains[LF_BAND] = 1.0f; } dec->UpSampler[3].Gains[HF_BAND] = 0.0f; dec->UpSampler[3].Gains[LF_BAND] = 0.0f; } memset(&dec->Matrix, 0, sizeof(dec->Matrix)); if(conf->FreqBands == 1) { dec->DualBand = AL_FALSE; for(i = 0;i < conf->NumSpeakers;i++) { ALsizei chan = chanmap[i]; ALfloat gain; ALsizei j, k; if(!periphonic) { for(j = 0,k = 0;j < MAX_AMBI2D_COEFFS;j++) { ALsizei l = map2DTo3D[j]; if(j == 0) gain = conf->HFOrderGain[0]; else if(j == 1) gain = conf->HFOrderGain[1]; else if(j == 3) gain = conf->HFOrderGain[2]; else if(j == 5) gain = conf->HFOrderGain[3]; if((conf->ChanMask&(1<Matrix.Single[chan][j] = conf->HFMatrix[i][k++] / coeff_scale[l] * gain; } } else { for(j = 0,k = 0;j < MAX_AMBI_COEFFS;j++) { if(j == 0) gain = conf->HFOrderGain[0]; else if(j == 1) gain = conf->HFOrderGain[1]; else if(j == 4) gain = conf->HFOrderGain[2]; else if(j == 9) gain = conf->HFOrderGain[3]; if((conf->ChanMask&(1<Matrix.Single[chan][j] = conf->HFMatrix[i][k++] / coeff_scale[j] * gain; } } } } else { dec->DualBand = AL_TRUE; ratio = conf->XOverFreq / (ALfloat)srate; for(i = 0;i < MAX_AMBI_COEFFS;i++) bandsplit_init(&dec->XOver[i], ratio); ratio = powf(10.0f, conf->XOverRatio / 40.0f); for(i = 0;i < conf->NumSpeakers;i++) { ALsizei chan = chanmap[i]; ALfloat gain; ALsizei j, k; if(!periphonic) { for(j = 0,k = 0;j < MAX_AMBI2D_COEFFS;j++) { ALsizei l = map2DTo3D[j]; if(j == 0) gain = conf->HFOrderGain[0] * ratio; else if(j == 1) gain = conf->HFOrderGain[1] * ratio; else if(j == 3) gain = conf->HFOrderGain[2] * ratio; else if(j == 5) gain = conf->HFOrderGain[3] * ratio; if((conf->ChanMask&(1<Matrix.Dual[chan][HF_BAND][j] = conf->HFMatrix[i][k++] / coeff_scale[l] * gain; } for(j = 0,k = 0;j < MAX_AMBI2D_COEFFS;j++) { ALsizei l = map2DTo3D[j]; if(j == 0) gain = conf->LFOrderGain[0] / ratio; else if(j == 1) gain = conf->LFOrderGain[1] / ratio; else if(j == 3) gain = conf->LFOrderGain[2] / ratio; else if(j == 5) gain = conf->LFOrderGain[3] / ratio; if((conf->ChanMask&(1<Matrix.Dual[chan][LF_BAND][j] = conf->LFMatrix[i][k++] / coeff_scale[l] * gain; } } else { for(j = 0,k = 0;j < MAX_AMBI_COEFFS;j++) { if(j == 0) gain = conf->HFOrderGain[0] * ratio; else if(j == 1) gain = conf->HFOrderGain[1] * ratio; else if(j == 4) gain = conf->HFOrderGain[2] * ratio; else if(j == 9) gain = conf->HFOrderGain[3] * ratio; if((conf->ChanMask&(1<Matrix.Dual[chan][HF_BAND][j] = conf->HFMatrix[i][k++] / coeff_scale[j] * gain; } for(j = 0,k = 0;j < MAX_AMBI_COEFFS;j++) { if(j == 0) gain = conf->LFOrderGain[0] / ratio; else if(j == 1) gain = conf->LFOrderGain[1] / ratio; else if(j == 4) gain = conf->LFOrderGain[2] / ratio; else if(j == 9) gain = conf->LFOrderGain[3] / ratio; if((conf->ChanMask&(1<Matrix.Dual[chan][LF_BAND][j] = conf->LFMatrix[i][k++] / coeff_scale[j] * gain; } } } } } void bformatdec_process(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALsizei OutChannels, const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo) { ALsizei chan, i; OutBuffer = ASSUME_ALIGNED(OutBuffer, 16); if(dec->DualBand) { for(i = 0;i < dec->NumChannels;i++) bandsplit_process(&dec->XOver[i], dec->SamplesHF[i], dec->SamplesLF[i], InSamples[i], SamplesToDo); for(chan = 0;chan < OutChannels;chan++) { if(!(dec->Enabled&(1<ChannelMix, 0, SamplesToDo*sizeof(ALfloat)); MixRowSamples(dec->ChannelMix, dec->Matrix.Dual[chan][HF_BAND], dec->SamplesHF, dec->NumChannels, 0, SamplesToDo ); MixRowSamples(dec->ChannelMix, dec->Matrix.Dual[chan][LF_BAND], dec->SamplesLF, dec->NumChannels, 0, SamplesToDo ); for(i = 0;i < SamplesToDo;i++) OutBuffer[chan][i] += dec->ChannelMix[i]; } } else { for(chan = 0;chan < OutChannels;chan++) { if(!(dec->Enabled&(1<ChannelMix, 0, SamplesToDo*sizeof(ALfloat)); MixRowSamples(dec->ChannelMix, dec->Matrix.Single[chan], InSamples, dec->NumChannels, 0, SamplesToDo); for(i = 0;i < SamplesToDo;i++) OutBuffer[chan][i] += dec->ChannelMix[i]; } } } void bformatdec_upSample(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BUFFERSIZE], const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei InChannels, ALsizei SamplesToDo) { ALsizei i; /* This up-sampler leverages the differences observed in dual-band second- * and third-order decoder matrices compared to first-order. For the same * output channel configuration, the low-frequency matrix has identical * coefficients in the shared input channels, while the high-frequency * matrix has extra scalars applied to the W channel and X/Y/Z channels. * Mixing the first-order content into the higher-order stream with the * appropriate counter-scales applied to the HF response results in the * subsequent higher-order decode generating the same response as a first- * order decode. */ for(i = 0;i < InChannels;i++) { /* First, split the first-order components into low and high frequency * bands. */ bandsplit_process(&dec->UpSampler[i].XOver, dec->Samples[HF_BAND], dec->Samples[LF_BAND], InSamples[i], SamplesToDo ); /* Now write each band to the output. */ MixRowSamples(OutBuffer[i], dec->UpSampler[i].Gains, dec->Samples, NUM_BANDS, 0, SamplesToDo ); } } #define INVALID_UPSAMPLE_INDEX INT_MAX static ALsizei GetACNIndex(const BFChannelConfig *chans, ALsizei numchans, ALsizei acn) { ALsizei i; for(i = 0;i < numchans;i++) { if(chans[i].Index == acn) return i; } return INVALID_UPSAMPLE_INDEX; } #define GetChannelForACN(b, a) GetACNIndex((b).Ambi.Map, (b).NumChannels, (a)) typedef struct AmbiUpsampler { alignas(16) ALfloat Samples[NUM_BANDS][BUFFERSIZE]; BandSplitter XOver[4]; ALfloat Gains[4][MAX_OUTPUT_CHANNELS][NUM_BANDS]; } AmbiUpsampler; AmbiUpsampler *ambiup_alloc() { return al_calloc(16, sizeof(AmbiUpsampler)); } void ambiup_free(struct AmbiUpsampler **ambiup) { if(ambiup) { al_free(*ambiup); *ambiup = NULL; } } void ambiup_reset(struct AmbiUpsampler *ambiup, const ALCdevice *device, ALfloat w_scale, ALfloat xyz_scale) { ALfloat ratio; ALsizei i; ratio = 400.0f / (ALfloat)device->Frequency; for(i = 0;i < 4;i++) bandsplit_init(&ambiup->XOver[i], ratio); memset(ambiup->Gains, 0, sizeof(ambiup->Gains)); if(device->Dry.CoeffCount > 0) { ALfloat encgains[8][MAX_OUTPUT_CHANNELS]; ALsizei j; size_t k; for(k = 0;k < COUNTOF(Ambi3DPoints);k++) { ALfloat coeffs[MAX_AMBI_COEFFS] = { 0.0f }; CalcDirectionCoeffs(Ambi3DPoints[k], 0.0f, coeffs); ComputePanGains(&device->Dry, coeffs, 1.0f, encgains[k]); } /* Combine the matrices that do the in->virt and virt->out conversions * so we get a single in->out conversion. NOTE: the Encoder matrix * (encgains) and output are transposed, so the input channels line up * with the rows and the output channels line up with the columns. */ for(i = 0;i < 4;i++) { for(j = 0;j < device->Dry.NumChannels;j++) { ALdouble gain = 0.0; for(k = 0;k < COUNTOF(Ambi3DDecoder);k++) gain += (ALdouble)Ambi3DDecoder[k][i] * encgains[k][j]; ambiup->Gains[i][j][HF_BAND] = (ALfloat)(gain * Ambi3DDecoderHFScale[i]); ambiup->Gains[i][j][LF_BAND] = (ALfloat)gain; } } } else { for(i = 0;i < 4;i++) { ALsizei index = GetChannelForACN(device->Dry, i); if(index != INVALID_UPSAMPLE_INDEX) { ALfloat scale = device->Dry.Ambi.Map[index].Scale; ambiup->Gains[i][index][HF_BAND] = scale * ((i==0) ? w_scale : xyz_scale); ambiup->Gains[i][index][LF_BAND] = scale; } } } } void ambiup_process(struct AmbiUpsampler *ambiup, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALsizei OutChannels, const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo) { ALsizei i, j; for(i = 0;i < 4;i++) { bandsplit_process(&ambiup->XOver[i], ambiup->Samples[HF_BAND], ambiup->Samples[LF_BAND], InSamples[i], SamplesToDo ); for(j = 0;j < OutChannels;j++) MixRowSamples(OutBuffer[j], ambiup->Gains[i][j], ambiup->Samples, NUM_BANDS, 0, SamplesToDo ); } } openal-soft-openal-soft-1.19.1/Alc/bformatdec.h000066400000000000000000000043631335774445300212620ustar00rootroot00000000000000#ifndef BFORMATDEC_H #define BFORMATDEC_H #include "alMain.h" /* These are the necessary scales for first-order HF responses to play over * higher-order 2D (non-periphonic) decoders. */ #define W_SCALE_2H0P 1.224744871f /* sqrt(1.5) */ #define XYZ_SCALE_2H0P 1.0f #define W_SCALE_3H0P 1.414213562f /* sqrt(2) */ #define XYZ_SCALE_3H0P 1.082392196f /* These are the necessary scales for first-order HF responses to play over * higher-order 3D (periphonic) decoders. */ #define W_SCALE_2H2P 1.341640787f /* sqrt(1.8) */ #define XYZ_SCALE_2H2P 1.0f #define W_SCALE_3H3P 1.695486018f #define XYZ_SCALE_3H3P 1.136697713f /* NOTE: These are scale factors as applied to Ambisonics content. Decoder * coefficients should be divided by these values to get proper N3D scalings. */ const ALfloat N3D2N3DScale[MAX_AMBI_COEFFS]; const ALfloat SN3D2N3DScale[MAX_AMBI_COEFFS]; const ALfloat FuMa2N3DScale[MAX_AMBI_COEFFS]; struct AmbDecConf; struct BFormatDec; struct AmbiUpsampler; struct BFormatDec *bformatdec_alloc(); void bformatdec_free(struct BFormatDec **dec); void bformatdec_reset(struct BFormatDec *dec, const struct AmbDecConf *conf, ALsizei chancount, ALuint srate, const ALsizei chanmap[MAX_OUTPUT_CHANNELS]); /* Decodes the ambisonic input to the given output channels. */ void bformatdec_process(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALsizei OutChannels, const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo); /* Up-samples a first-order input to the decoder's configuration. */ void bformatdec_upSample(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BUFFERSIZE], const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei InChannels, ALsizei SamplesToDo); /* Stand-alone first-order upsampler. Kept here because it shares some stuff * with bformatdec. Assumes a periphonic (4-channel) input mix! */ struct AmbiUpsampler *ambiup_alloc(); void ambiup_free(struct AmbiUpsampler **ambiup); void ambiup_reset(struct AmbiUpsampler *ambiup, const ALCdevice *device, ALfloat w_scale, ALfloat xyz_scale); void ambiup_process(struct AmbiUpsampler *ambiup, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALsizei OutChannels, const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo); #endif /* BFORMATDEC_H */ openal-soft-openal-soft-1.19.1/Alc/bs2b.c000066400000000000000000000131311335774445300177700ustar00rootroot00000000000000/*- * Copyright (c) 2005 Boris Mikhaylov * * Permission is hereby granted, free of charge, to any person obtaining * a copy of this software and associated documentation files (the * "Software"), to deal in the Software without restriction, including * without limitation the rights to use, copy, modify, merge, publish, * distribute, sublicense, and/or sell copies of the Software, and to * permit persons to whom the Software is furnished to do so, subject to * the following conditions: * * The above copyright notice and this permission notice shall be * included in all copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */ #include "config.h" #include #include #include "bs2b.h" #include "alu.h" /* Set up all data. */ static void init(struct bs2b *bs2b) { float Fc_lo, Fc_hi; float G_lo, G_hi; float x, g; switch(bs2b->level) { case BS2B_LOW_CLEVEL: /* Low crossfeed level */ Fc_lo = 360.0f; Fc_hi = 501.0f; G_lo = 0.398107170553497f; G_hi = 0.205671765275719f; break; case BS2B_MIDDLE_CLEVEL: /* Middle crossfeed level */ Fc_lo = 500.0f; Fc_hi = 711.0f; G_lo = 0.459726988530872f; G_hi = 0.228208484414988f; break; case BS2B_HIGH_CLEVEL: /* High crossfeed level (virtual speakers are closer to itself) */ Fc_lo = 700.0f; Fc_hi = 1021.0f; G_lo = 0.530884444230988f; G_hi = 0.250105790667544f; break; case BS2B_LOW_ECLEVEL: /* Low easy crossfeed level */ Fc_lo = 360.0f; Fc_hi = 494.0f; G_lo = 0.316227766016838f; G_hi = 0.168236228897329f; break; case BS2B_MIDDLE_ECLEVEL: /* Middle easy crossfeed level */ Fc_lo = 500.0f; Fc_hi = 689.0f; G_lo = 0.354813389233575f; G_hi = 0.187169483835901f; break; default: /* High easy crossfeed level */ bs2b->level = BS2B_HIGH_ECLEVEL; Fc_lo = 700.0f; Fc_hi = 975.0f; G_lo = 0.398107170553497f; G_hi = 0.205671765275719f; break; } /* switch */ g = 1.0f / (1.0f - G_hi + G_lo); /* $fc = $Fc / $s; * $d = 1 / 2 / pi / $fc; * $x = exp(-1 / $d); */ x = expf(-2.0f * F_PI * Fc_lo / bs2b->srate); bs2b->b1_lo = x; bs2b->a0_lo = G_lo * (1.0f - x) * g; x = expf(-2.0f * F_PI * Fc_hi / bs2b->srate); bs2b->b1_hi = x; bs2b->a0_hi = (1.0f - G_hi * (1.0f - x)) * g; bs2b->a1_hi = -x * g; } /* init */ /* Exported functions. * See descriptions in "bs2b.h" */ void bs2b_set_params(struct bs2b *bs2b, int level, int srate) { if(srate <= 0) srate = 1; bs2b->level = level; bs2b->srate = srate; init(bs2b); } /* bs2b_set_params */ int bs2b_get_level(struct bs2b *bs2b) { return bs2b->level; } /* bs2b_get_level */ int bs2b_get_srate(struct bs2b *bs2b) { return bs2b->srate; } /* bs2b_get_srate */ void bs2b_clear(struct bs2b *bs2b) { memset(&bs2b->last_sample, 0, sizeof(bs2b->last_sample)); } /* bs2b_clear */ void bs2b_cross_feed(struct bs2b *bs2b, float *restrict Left, float *restrict Right, int SamplesToDo) { float lsamples[128][2]; float rsamples[128][2]; int base; for(base = 0;base < SamplesToDo;) { int todo = mini(128, SamplesToDo-base); int i; /* Process left input */ lsamples[0][0] = bs2b->a0_lo*Left[0] + bs2b->b1_lo*bs2b->last_sample[0].lo; lsamples[0][1] = bs2b->a0_hi*Left[0] + bs2b->a1_hi*bs2b->last_sample[0].asis + bs2b->b1_hi*bs2b->last_sample[0].hi; for(i = 1;i < todo;i++) { lsamples[i][0] = bs2b->a0_lo*Left[i] + bs2b->b1_lo*lsamples[i-1][0]; lsamples[i][1] = bs2b->a0_hi*Left[i] + bs2b->a1_hi*Left[i-1] + bs2b->b1_hi*lsamples[i-1][1]; } bs2b->last_sample[0].asis = Left[i-1]; bs2b->last_sample[0].lo = lsamples[i-1][0]; bs2b->last_sample[0].hi = lsamples[i-1][1]; /* Process right input */ rsamples[0][0] = bs2b->a0_lo*Right[0] + bs2b->b1_lo*bs2b->last_sample[1].lo; rsamples[0][1] = bs2b->a0_hi*Right[0] + bs2b->a1_hi*bs2b->last_sample[1].asis + bs2b->b1_hi*bs2b->last_sample[1].hi; for(i = 1;i < todo;i++) { rsamples[i][0] = bs2b->a0_lo*Right[i] + bs2b->b1_lo*rsamples[i-1][0]; rsamples[i][1] = bs2b->a0_hi*Right[i] + bs2b->a1_hi*Right[i-1] + bs2b->b1_hi*rsamples[i-1][1]; } bs2b->last_sample[1].asis = Right[i-1]; bs2b->last_sample[1].lo = rsamples[i-1][0]; bs2b->last_sample[1].hi = rsamples[i-1][1]; /* Crossfeed */ for(i = 0;i < todo;i++) *(Left++) = lsamples[i][1] + rsamples[i][0]; for(i = 0;i < todo;i++) *(Right++) = rsamples[i][1] + lsamples[i][0]; base += todo; } } /* bs2b_cross_feed */ openal-soft-openal-soft-1.19.1/Alc/compat.h000066400000000000000000000017261335774445300204370ustar00rootroot00000000000000#ifndef AL_COMPAT_H #define AL_COMPAT_H #include "alstring.h" #ifdef __cplusplus extern "C" { #endif #ifdef _WIN32 #define WIN32_LEAN_AND_MEAN #include WCHAR *strdupW(const WCHAR *str); /* Opens a file with standard I/O. The filename is expected to be UTF-8. */ FILE *al_fopen(const char *fname, const char *mode); #define HAVE_DYNLOAD 1 #else #define al_fopen fopen #if defined(HAVE_DLFCN_H) && !defined(IN_IDE_PARSER) #define HAVE_DYNLOAD 1 #endif #endif struct FileMapping { #ifdef _WIN32 HANDLE file; HANDLE fmap; #else int fd; #endif void *ptr; size_t len; }; struct FileMapping MapFileToMem(const char *fname); void UnmapFileMem(const struct FileMapping *mapping); void GetProcBinary(al_string *path, al_string *fname); #ifdef HAVE_DYNLOAD void *LoadLib(const char *name); void CloseLib(void *handle); void *GetSymbol(void *handle, const char *name); #endif #ifdef __cplusplus } /* extern "C" */ #endif #endif /* AL_COMPAT_H */ openal-soft-openal-soft-1.19.1/Alc/converter.c000066400000000000000000000377171335774445300211670ustar00rootroot00000000000000 #include "config.h" #include "converter.h" #include "fpu_modes.h" #include "mixer/defs.h" SampleConverter *CreateSampleConverter(enum DevFmtType srcType, enum DevFmtType dstType, ALsizei numchans, ALsizei srcRate, ALsizei dstRate) { SampleConverter *converter; ALsizei step; if(numchans <= 0 || srcRate <= 0 || dstRate <= 0) return NULL; converter = al_calloc(16, FAM_SIZE(SampleConverter, Chan, numchans)); converter->mSrcType = srcType; converter->mDstType = dstType; converter->mNumChannels = numchans; converter->mSrcTypeSize = BytesFromDevFmt(srcType); converter->mDstTypeSize = BytesFromDevFmt(dstType); converter->mSrcPrepCount = 0; converter->mFracOffset = 0; /* Have to set the mixer FPU mode since that's what the resampler code expects. */ START_MIXER_MODE(); step = (ALsizei)mind(((ALdouble)srcRate/dstRate*FRACTIONONE) + 0.5, MAX_PITCH * FRACTIONONE); converter->mIncrement = maxi(step, 1); if(converter->mIncrement == FRACTIONONE) converter->mResample = Resample_copy_C; else { /* TODO: Allow other resamplers. */ BsincPrepare(converter->mIncrement, &converter->mState.bsinc, &bsinc12); converter->mResample = SelectResampler(BSinc12Resampler); } END_MIXER_MODE(); return converter; } void DestroySampleConverter(SampleConverter **converter) { if(converter) { al_free(*converter); *converter = NULL; } } static inline ALfloat Sample_ALbyte(ALbyte val) { return val * (1.0f/128.0f); } static inline ALfloat Sample_ALubyte(ALubyte val) { return Sample_ALbyte((ALint)val - 128); } static inline ALfloat Sample_ALshort(ALshort val) { return val * (1.0f/32768.0f); } static inline ALfloat Sample_ALushort(ALushort val) { return Sample_ALshort((ALint)val - 32768); } static inline ALfloat Sample_ALint(ALint val) { return (val>>7) * (1.0f/16777216.0f); } static inline ALfloat Sample_ALuint(ALuint val) { return Sample_ALint(val - INT_MAX - 1); } static inline ALfloat Sample_ALfloat(ALfloat val) { return val; } #define DECL_TEMPLATE(T) \ static inline void Load_##T(ALfloat *restrict dst, const T *restrict src, \ ALint srcstep, ALsizei samples) \ { \ ALsizei i; \ for(i = 0;i < samples;i++) \ dst[i] = Sample_##T(src[i*srcstep]); \ } DECL_TEMPLATE(ALbyte) DECL_TEMPLATE(ALubyte) DECL_TEMPLATE(ALshort) DECL_TEMPLATE(ALushort) DECL_TEMPLATE(ALint) DECL_TEMPLATE(ALuint) DECL_TEMPLATE(ALfloat) #undef DECL_TEMPLATE static void LoadSamples(ALfloat *dst, const ALvoid *src, ALint srcstep, enum DevFmtType srctype, ALsizei samples) { switch(srctype) { case DevFmtByte: Load_ALbyte(dst, src, srcstep, samples); break; case DevFmtUByte: Load_ALubyte(dst, src, srcstep, samples); break; case DevFmtShort: Load_ALshort(dst, src, srcstep, samples); break; case DevFmtUShort: Load_ALushort(dst, src, srcstep, samples); break; case DevFmtInt: Load_ALint(dst, src, srcstep, samples); break; case DevFmtUInt: Load_ALuint(dst, src, srcstep, samples); break; case DevFmtFloat: Load_ALfloat(dst, src, srcstep, samples); break; } } static inline ALbyte ALbyte_Sample(ALfloat val) { return fastf2i(clampf(val*128.0f, -128.0f, 127.0f)); } static inline ALubyte ALubyte_Sample(ALfloat val) { return ALbyte_Sample(val)+128; } static inline ALshort ALshort_Sample(ALfloat val) { return fastf2i(clampf(val*32768.0f, -32768.0f, 32767.0f)); } static inline ALushort ALushort_Sample(ALfloat val) { return ALshort_Sample(val)+32768; } static inline ALint ALint_Sample(ALfloat val) { return fastf2i(clampf(val*16777216.0f, -16777216.0f, 16777215.0f)) << 7; } static inline ALuint ALuint_Sample(ALfloat val) { return ALint_Sample(val)+INT_MAX+1; } static inline ALfloat ALfloat_Sample(ALfloat val) { return val; } #define DECL_TEMPLATE(T) \ static inline void Store_##T(T *restrict dst, const ALfloat *restrict src, \ ALint dststep, ALsizei samples) \ { \ ALsizei i; \ for(i = 0;i < samples;i++) \ dst[i*dststep] = T##_Sample(src[i]); \ } DECL_TEMPLATE(ALbyte) DECL_TEMPLATE(ALubyte) DECL_TEMPLATE(ALshort) DECL_TEMPLATE(ALushort) DECL_TEMPLATE(ALint) DECL_TEMPLATE(ALuint) DECL_TEMPLATE(ALfloat) #undef DECL_TEMPLATE static void StoreSamples(ALvoid *dst, const ALfloat *src, ALint dststep, enum DevFmtType dsttype, ALsizei samples) { switch(dsttype) { case DevFmtByte: Store_ALbyte(dst, src, dststep, samples); break; case DevFmtUByte: Store_ALubyte(dst, src, dststep, samples); break; case DevFmtShort: Store_ALshort(dst, src, dststep, samples); break; case DevFmtUShort: Store_ALushort(dst, src, dststep, samples); break; case DevFmtInt: Store_ALint(dst, src, dststep, samples); break; case DevFmtUInt: Store_ALuint(dst, src, dststep, samples); break; case DevFmtFloat: Store_ALfloat(dst, src, dststep, samples); break; } } ALsizei SampleConverterAvailableOut(SampleConverter *converter, ALsizei srcframes) { ALint prepcount = converter->mSrcPrepCount; ALsizei increment = converter->mIncrement; ALsizei DataPosFrac = converter->mFracOffset; ALuint64 DataSize64; if(prepcount < 0) { /* Negative prepcount means we need to skip that many input samples. */ if(-prepcount >= srcframes) return 0; srcframes += prepcount; prepcount = 0; } if(srcframes < 1) { /* No output samples if there's no input samples. */ return 0; } if(prepcount < MAX_RESAMPLE_PADDING*2 && MAX_RESAMPLE_PADDING*2 - prepcount >= srcframes) { /* Not enough input samples to generate an output sample. */ return 0; } DataSize64 = prepcount; DataSize64 += srcframes; DataSize64 -= MAX_RESAMPLE_PADDING*2; DataSize64 <<= FRACTIONBITS; DataSize64 -= DataPosFrac; /* If we have a full prep, we can generate at least one sample. */ return (ALsizei)clampu64((DataSize64 + increment-1)/increment, 1, BUFFERSIZE); } ALsizei SampleConverterInput(SampleConverter *converter, const ALvoid **src, ALsizei *srcframes, ALvoid *dst, ALsizei dstframes) { const ALsizei SrcFrameSize = converter->mNumChannels * converter->mSrcTypeSize; const ALsizei DstFrameSize = converter->mNumChannels * converter->mDstTypeSize; const ALsizei increment = converter->mIncrement; ALsizei pos = 0; START_MIXER_MODE(); while(pos < dstframes && *srcframes > 0) { ALfloat *restrict SrcData = ASSUME_ALIGNED(converter->mSrcSamples, 16); ALfloat *restrict DstData = ASSUME_ALIGNED(converter->mDstSamples, 16); ALint prepcount = converter->mSrcPrepCount; ALsizei DataPosFrac = converter->mFracOffset; ALuint64 DataSize64; ALsizei DstSize; ALint toread; ALsizei chan; if(prepcount < 0) { /* Negative prepcount means we need to skip that many input samples. */ if(-prepcount >= *srcframes) { converter->mSrcPrepCount = prepcount + *srcframes; *srcframes = 0; break; } *src = (const ALbyte*)*src + SrcFrameSize*-prepcount; *srcframes += prepcount; converter->mSrcPrepCount = 0; continue; } toread = mini(*srcframes, BUFFERSIZE - MAX_RESAMPLE_PADDING*2); if(prepcount < MAX_RESAMPLE_PADDING*2 && MAX_RESAMPLE_PADDING*2 - prepcount >= toread) { /* Not enough input samples to generate an output sample. Store * what we're given for later. */ for(chan = 0;chan < converter->mNumChannels;chan++) LoadSamples(&converter->Chan[chan].mPrevSamples[prepcount], (const ALbyte*)*src + converter->mSrcTypeSize*chan, converter->mNumChannels, converter->mSrcType, toread ); converter->mSrcPrepCount = prepcount + toread; *srcframes = 0; break; } DataSize64 = prepcount; DataSize64 += toread; DataSize64 -= MAX_RESAMPLE_PADDING*2; DataSize64 <<= FRACTIONBITS; DataSize64 -= DataPosFrac; /* If we have a full prep, we can generate at least one sample. */ DstSize = (ALsizei)clampu64((DataSize64 + increment-1)/increment, 1, BUFFERSIZE); DstSize = mini(DstSize, dstframes-pos); for(chan = 0;chan < converter->mNumChannels;chan++) { const ALbyte *SrcSamples = (const ALbyte*)*src + converter->mSrcTypeSize*chan; ALbyte *DstSamples = (ALbyte*)dst + converter->mDstTypeSize*chan; const ALfloat *ResampledData; ALsizei SrcDataEnd; /* Load the previous samples into the source data first, then the * new samples from the input buffer. */ memcpy(SrcData, converter->Chan[chan].mPrevSamples, prepcount*sizeof(ALfloat)); LoadSamples(SrcData + prepcount, SrcSamples, converter->mNumChannels, converter->mSrcType, toread ); /* Store as many prep samples for next time as possible, given the * number of output samples being generated. */ SrcDataEnd = (DataPosFrac + increment*DstSize)>>FRACTIONBITS; if(SrcDataEnd >= prepcount+toread) memset(converter->Chan[chan].mPrevSamples, 0, sizeof(converter->Chan[chan].mPrevSamples)); else { size_t len = mini(MAX_RESAMPLE_PADDING*2, prepcount+toread-SrcDataEnd); memcpy(converter->Chan[chan].mPrevSamples, &SrcData[SrcDataEnd], len*sizeof(ALfloat)); memset(converter->Chan[chan].mPrevSamples+len, 0, sizeof(converter->Chan[chan].mPrevSamples) - len*sizeof(ALfloat)); } /* Now resample, and store the result in the output buffer. */ ResampledData = converter->mResample(&converter->mState, SrcData+MAX_RESAMPLE_PADDING, DataPosFrac, increment, DstData, DstSize ); StoreSamples(DstSamples, ResampledData, converter->mNumChannels, converter->mDstType, DstSize); } /* Update the number of prep samples still available, as well as the * fractional offset. */ DataPosFrac += increment*DstSize; converter->mSrcPrepCount = mini(prepcount + toread - (DataPosFrac>>FRACTIONBITS), MAX_RESAMPLE_PADDING*2); converter->mFracOffset = DataPosFrac & FRACTIONMASK; /* Update the src and dst pointers in case there's still more to do. */ *src = (const ALbyte*)*src + SrcFrameSize*(DataPosFrac>>FRACTIONBITS); *srcframes -= mini(*srcframes, (DataPosFrac>>FRACTIONBITS)); dst = (ALbyte*)dst + DstFrameSize*DstSize; pos += DstSize; } END_MIXER_MODE(); return pos; } ChannelConverter *CreateChannelConverter(enum DevFmtType srcType, enum DevFmtChannels srcChans, enum DevFmtChannels dstChans) { ChannelConverter *converter; if(srcChans != dstChans && !((srcChans == DevFmtMono && dstChans == DevFmtStereo) || (srcChans == DevFmtStereo && dstChans == DevFmtMono))) return NULL; converter = al_calloc(DEF_ALIGN, sizeof(*converter)); converter->mSrcType = srcType; converter->mSrcChans = srcChans; converter->mDstChans = dstChans; return converter; } void DestroyChannelConverter(ChannelConverter **converter) { if(converter) { al_free(*converter); *converter = NULL; } } #define DECL_TEMPLATE(T) \ static void Mono2Stereo##T(ALfloat *restrict dst, const T *src, ALsizei frames)\ { \ ALsizei i; \ for(i = 0;i < frames;i++) \ dst[i*2 + 1] = dst[i*2 + 0] = Sample_##T(src[i]) * 0.707106781187f; \ } \ \ static void Stereo2Mono##T(ALfloat *restrict dst, const T *src, ALsizei frames)\ { \ ALsizei i; \ for(i = 0;i < frames;i++) \ dst[i] = (Sample_##T(src[i*2 + 0])+Sample_##T(src[i*2 + 1])) * \ 0.707106781187f; \ } DECL_TEMPLATE(ALbyte) DECL_TEMPLATE(ALubyte) DECL_TEMPLATE(ALshort) DECL_TEMPLATE(ALushort) DECL_TEMPLATE(ALint) DECL_TEMPLATE(ALuint) DECL_TEMPLATE(ALfloat) #undef DECL_TEMPLATE void ChannelConverterInput(ChannelConverter *converter, const ALvoid *src, ALfloat *dst, ALsizei frames) { if(converter->mSrcChans == converter->mDstChans) { LoadSamples(dst, src, 1, converter->mSrcType, frames*ChannelsFromDevFmt(converter->mSrcChans, 0)); return; } if(converter->mSrcChans == DevFmtStereo && converter->mDstChans == DevFmtMono) { switch(converter->mSrcType) { case DevFmtByte: Stereo2MonoALbyte(dst, src, frames); break; case DevFmtUByte: Stereo2MonoALubyte(dst, src, frames); break; case DevFmtShort: Stereo2MonoALshort(dst, src, frames); break; case DevFmtUShort: Stereo2MonoALushort(dst, src, frames); break; case DevFmtInt: Stereo2MonoALint(dst, src, frames); break; case DevFmtUInt: Stereo2MonoALuint(dst, src, frames); break; case DevFmtFloat: Stereo2MonoALfloat(dst, src, frames); break; } } else /*if(converter->mSrcChans == DevFmtMono && converter->mDstChans == DevFmtStereo)*/ { switch(converter->mSrcType) { case DevFmtByte: Mono2StereoALbyte(dst, src, frames); break; case DevFmtUByte: Mono2StereoALubyte(dst, src, frames); break; case DevFmtShort: Mono2StereoALshort(dst, src, frames); break; case DevFmtUShort: Mono2StereoALushort(dst, src, frames); break; case DevFmtInt: Mono2StereoALint(dst, src, frames); break; case DevFmtUInt: Mono2StereoALuint(dst, src, frames); break; case DevFmtFloat: Mono2StereoALfloat(dst, src, frames); break; } } } openal-soft-openal-soft-1.19.1/Alc/converter.h000066400000000000000000000030211335774445300211510ustar00rootroot00000000000000#ifndef CONVERTER_H #define CONVERTER_H #include "alMain.h" #include "alu.h" #ifdef __cpluspluc extern "C" { #endif typedef struct SampleConverter { enum DevFmtType mSrcType; enum DevFmtType mDstType; ALsizei mNumChannels; ALsizei mSrcTypeSize; ALsizei mDstTypeSize; ALint mSrcPrepCount; ALsizei mFracOffset; ALsizei mIncrement; InterpState mState; ResamplerFunc mResample; alignas(16) ALfloat mSrcSamples[BUFFERSIZE]; alignas(16) ALfloat mDstSamples[BUFFERSIZE]; struct { alignas(16) ALfloat mPrevSamples[MAX_RESAMPLE_PADDING*2]; } Chan[]; } SampleConverter; SampleConverter *CreateSampleConverter(enum DevFmtType srcType, enum DevFmtType dstType, ALsizei numchans, ALsizei srcRate, ALsizei dstRate); void DestroySampleConverter(SampleConverter **converter); ALsizei SampleConverterInput(SampleConverter *converter, const ALvoid **src, ALsizei *srcframes, ALvoid *dst, ALsizei dstframes); ALsizei SampleConverterAvailableOut(SampleConverter *converter, ALsizei srcframes); typedef struct ChannelConverter { enum DevFmtType mSrcType; enum DevFmtChannels mSrcChans; enum DevFmtChannels mDstChans; } ChannelConverter; ChannelConverter *CreateChannelConverter(enum DevFmtType srcType, enum DevFmtChannels srcChans, enum DevFmtChannels dstChans); void DestroyChannelConverter(ChannelConverter **converter); void ChannelConverterInput(ChannelConverter *converter, const ALvoid *src, ALfloat *dst, ALsizei frames); #ifdef __cpluspluc } #endif #endif /* CONVERTER_H */ openal-soft-openal-soft-1.19.1/Alc/cpu_caps.h000066400000000000000000000004131335774445300207410ustar00rootroot00000000000000#ifndef CPU_CAPS_H #define CPU_CAPS_H extern int CPUCapFlags; enum { CPU_CAP_SSE = 1<<0, CPU_CAP_SSE2 = 1<<1, CPU_CAP_SSE3 = 1<<2, CPU_CAP_SSE4_1 = 1<<3, CPU_CAP_NEON = 1<<4, }; void FillCPUCaps(int capfilter); #endif /* CPU_CAPS_H */ openal-soft-openal-soft-1.19.1/Alc/effects/000077500000000000000000000000001335774445300204145ustar00rootroot00000000000000openal-soft-openal-soft-1.19.1/Alc/effects/autowah.c000066400000000000000000000262531335774445300222400ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2018 by Raul Herraiz. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include "alMain.h" #include "alAuxEffectSlot.h" #include "alError.h" #include "alu.h" #include "filters/defs.h" #define MIN_FREQ 20.0f #define MAX_FREQ 2500.0f #define Q_FACTOR 5.0f typedef struct ALautowahState { DERIVE_FROM_TYPE(ALeffectState); /* Effect parameters */ ALfloat AttackRate; ALfloat ReleaseRate; ALfloat ResonanceGain; ALfloat PeakGain; ALfloat FreqMinNorm; ALfloat BandwidthNorm; ALfloat env_delay; /* Filter components derived from the envelope. */ struct { ALfloat cos_w0; ALfloat alpha; } Env[BUFFERSIZE]; struct { /* Effect filters' history. */ struct { ALfloat z1, z2; } Filter; /* Effect gains for each output channel */ ALfloat CurrentGains[MAX_OUTPUT_CHANNELS]; ALfloat TargetGains[MAX_OUTPUT_CHANNELS]; } Chans[MAX_EFFECT_CHANNELS]; /* Effects buffers */ alignas(16) ALfloat BufferOut[BUFFERSIZE]; } ALautowahState; static ALvoid ALautowahState_Destruct(ALautowahState *state); static ALboolean ALautowahState_deviceUpdate(ALautowahState *state, ALCdevice *device); static ALvoid ALautowahState_update(ALautowahState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props); static ALvoid ALautowahState_process(ALautowahState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels); DECLARE_DEFAULT_ALLOCATORS(ALautowahState) DEFINE_ALEFFECTSTATE_VTABLE(ALautowahState); static void ALautowahState_Construct(ALautowahState *state) { ALeffectState_Construct(STATIC_CAST(ALeffectState, state)); SET_VTABLE2(ALautowahState, ALeffectState, state); } static ALvoid ALautowahState_Destruct(ALautowahState *state) { ALeffectState_Destruct(STATIC_CAST(ALeffectState,state)); } static ALboolean ALautowahState_deviceUpdate(ALautowahState *state, ALCdevice *UNUSED(device)) { /* (Re-)initializing parameters and clear the buffers. */ ALsizei i, j; state->AttackRate = 1.0f; state->ReleaseRate = 1.0f; state->ResonanceGain = 10.0f; state->PeakGain = 4.5f; state->FreqMinNorm = 4.5e-4f; state->BandwidthNorm = 0.05f; state->env_delay = 0.0f; memset(state->Env, 0, sizeof(state->Env)); for(i = 0;i < MAX_EFFECT_CHANNELS;i++) { for(j = 0;j < MAX_OUTPUT_CHANNELS;j++) state->Chans[i].CurrentGains[j] = 0.0f; state->Chans[i].Filter.z1 = 0.0f; state->Chans[i].Filter.z2 = 0.0f; } return AL_TRUE; } static ALvoid ALautowahState_update(ALautowahState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props) { const ALCdevice *device = context->Device; ALfloat ReleaseTime; ALsizei i; ReleaseTime = clampf(props->Autowah.ReleaseTime, 0.001f, 1.0f); state->AttackRate = expf(-1.0f / (props->Autowah.AttackTime*device->Frequency)); state->ReleaseRate = expf(-1.0f / (ReleaseTime*device->Frequency)); /* 0-20dB Resonance Peak gain */ state->ResonanceGain = sqrtf(log10f(props->Autowah.Resonance)*10.0f / 3.0f); state->PeakGain = 1.0f - log10f(props->Autowah.PeakGain/AL_AUTOWAH_MAX_PEAK_GAIN); state->FreqMinNorm = MIN_FREQ / device->Frequency; state->BandwidthNorm = (MAX_FREQ-MIN_FREQ) / device->Frequency; STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer; STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels; for(i = 0;i < MAX_EFFECT_CHANNELS;i++) ComputePanGains(&device->FOAOut, IdentityMatrixf.m[i], slot->Params.Gain, state->Chans[i].TargetGains); } static ALvoid ALautowahState_process(ALautowahState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels) { const ALfloat attack_rate = state->AttackRate; const ALfloat release_rate = state->ReleaseRate; const ALfloat res_gain = state->ResonanceGain; const ALfloat peak_gain = state->PeakGain; const ALfloat freq_min = state->FreqMinNorm; const ALfloat bandwidth = state->BandwidthNorm; ALfloat env_delay; ALsizei c, i; env_delay = state->env_delay; for(i = 0;i < SamplesToDo;i++) { ALfloat w0, sample, a; /* Envelope follower described on the book: Audio Effects, Theory, * Implementation and Application. */ sample = peak_gain * fabsf(SamplesIn[0][i]); a = (sample > env_delay) ? attack_rate : release_rate; env_delay = lerp(sample, env_delay, a); /* Calculate the cos and alpha components for this sample's filter. */ w0 = minf((bandwidth*env_delay + freq_min), 0.46f) * F_TAU; state->Env[i].cos_w0 = cosf(w0); state->Env[i].alpha = sinf(w0)/(2.0f * Q_FACTOR); } state->env_delay = env_delay; for(c = 0;c < MAX_EFFECT_CHANNELS; c++) { /* This effectively inlines BiquadFilter_setParams for a peaking * filter and BiquadFilter_processC. The alpha and cosine components * for the filter coefficients were previously calculated with the * envelope. Because the filter changes for each sample, the * coefficients are transient and don't need to be held. */ ALfloat z1 = state->Chans[c].Filter.z1; ALfloat z2 = state->Chans[c].Filter.z2; for(i = 0;i < SamplesToDo;i++) { const ALfloat alpha = state->Env[i].alpha; const ALfloat cos_w0 = state->Env[i].cos_w0; ALfloat input, output; ALfloat a[3], b[3]; b[0] = 1.0f + alpha*res_gain; b[1] = -2.0f * cos_w0; b[2] = 1.0f - alpha*res_gain; a[0] = 1.0f + alpha/res_gain; a[1] = -2.0f * cos_w0; a[2] = 1.0f - alpha/res_gain; input = SamplesIn[c][i]; output = input*(b[0]/a[0]) + z1; z1 = input*(b[1]/a[0]) - output*(a[1]/a[0]) + z2; z2 = input*(b[2]/a[0]) - output*(a[2]/a[0]); state->BufferOut[i] = output; } state->Chans[c].Filter.z1 = z1; state->Chans[c].Filter.z2 = z2; /* Now, mix the processed sound data to the output. */ MixSamples(state->BufferOut, NumChannels, SamplesOut, state->Chans[c].CurrentGains, state->Chans[c].TargetGains, SamplesToDo, 0, SamplesToDo); } } typedef struct AutowahStateFactory { DERIVE_FROM_TYPE(EffectStateFactory); } AutowahStateFactory; static ALeffectState *AutowahStateFactory_create(AutowahStateFactory *UNUSED(factory)) { ALautowahState *state; NEW_OBJ0(state, ALautowahState)(); if(!state) return NULL; return STATIC_CAST(ALeffectState, state); } DEFINE_EFFECTSTATEFACTORY_VTABLE(AutowahStateFactory); EffectStateFactory *AutowahStateFactory_getFactory(void) { static AutowahStateFactory AutowahFactory = { { GET_VTABLE2(AutowahStateFactory, EffectStateFactory) } }; return STATIC_CAST(EffectStateFactory, &AutowahFactory); } void ALautowah_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_AUTOWAH_ATTACK_TIME: if(!(val >= AL_AUTOWAH_MIN_ATTACK_TIME && val <= AL_AUTOWAH_MAX_ATTACK_TIME)) SETERR_RETURN(context, AL_INVALID_VALUE,,"Autowah attack time out of range"); props->Autowah.AttackTime = val; break; case AL_AUTOWAH_RELEASE_TIME: if(!(val >= AL_AUTOWAH_MIN_RELEASE_TIME && val <= AL_AUTOWAH_MAX_RELEASE_TIME)) SETERR_RETURN(context, AL_INVALID_VALUE,,"Autowah release time out of range"); props->Autowah.ReleaseTime = val; break; case AL_AUTOWAH_RESONANCE: if(!(val >= AL_AUTOWAH_MIN_RESONANCE && val <= AL_AUTOWAH_MAX_RESONANCE)) SETERR_RETURN(context, AL_INVALID_VALUE,,"Autowah resonance out of range"); props->Autowah.Resonance = val; break; case AL_AUTOWAH_PEAK_GAIN: if(!(val >= AL_AUTOWAH_MIN_PEAK_GAIN && val <= AL_AUTOWAH_MAX_PEAK_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,,"Autowah peak gain out of range"); props->Autowah.PeakGain = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid autowah float property 0x%04x", param); } } void ALautowah_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { ALautowah_setParamf(effect, context, param, vals[0]); } void ALautowah_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid autowah integer property 0x%04x", param); } void ALautowah_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid autowah integer vector property 0x%04x", param); } void ALautowah_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid autowah integer property 0x%04x", param); } void ALautowah_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid autowah integer vector property 0x%04x", param); } void ALautowah_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_AUTOWAH_ATTACK_TIME: *val = props->Autowah.AttackTime; break; case AL_AUTOWAH_RELEASE_TIME: *val = props->Autowah.ReleaseTime; break; case AL_AUTOWAH_RESONANCE: *val = props->Autowah.Resonance; break; case AL_AUTOWAH_PEAK_GAIN: *val = props->Autowah.PeakGain; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid autowah float property 0x%04x", param); } } void ALautowah_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { ALautowah_getParamf(effect, context, param, vals); } DEFINE_ALEFFECT_VTABLE(ALautowah); openal-soft-openal-soft-1.19.1/Alc/effects/chorus.c000066400000000000000000000457501335774445300220760ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2013 by Mike Gorchak * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include "alMain.h" #include "alAuxEffectSlot.h" #include "alError.h" #include "alu.h" #include "filters/defs.h" static_assert(AL_CHORUS_WAVEFORM_SINUSOID == AL_FLANGER_WAVEFORM_SINUSOID, "Chorus/Flanger waveform value mismatch"); static_assert(AL_CHORUS_WAVEFORM_TRIANGLE == AL_FLANGER_WAVEFORM_TRIANGLE, "Chorus/Flanger waveform value mismatch"); enum WaveForm { WF_Sinusoid, WF_Triangle }; typedef struct ALchorusState { DERIVE_FROM_TYPE(ALeffectState); ALfloat *SampleBuffer; ALsizei BufferLength; ALsizei offset; ALsizei lfo_offset; ALsizei lfo_range; ALfloat lfo_scale; ALint lfo_disp; /* Gains for left and right sides */ struct { ALfloat Current[MAX_OUTPUT_CHANNELS]; ALfloat Target[MAX_OUTPUT_CHANNELS]; } Gains[2]; /* effect parameters */ enum WaveForm waveform; ALint delay; ALfloat depth; ALfloat feedback; } ALchorusState; static ALvoid ALchorusState_Destruct(ALchorusState *state); static ALboolean ALchorusState_deviceUpdate(ALchorusState *state, ALCdevice *Device); static ALvoid ALchorusState_update(ALchorusState *state, const ALCcontext *Context, const ALeffectslot *Slot, const ALeffectProps *props); static ALvoid ALchorusState_process(ALchorusState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels); DECLARE_DEFAULT_ALLOCATORS(ALchorusState) DEFINE_ALEFFECTSTATE_VTABLE(ALchorusState); static void ALchorusState_Construct(ALchorusState *state) { ALeffectState_Construct(STATIC_CAST(ALeffectState, state)); SET_VTABLE2(ALchorusState, ALeffectState, state); state->BufferLength = 0; state->SampleBuffer = NULL; state->offset = 0; state->lfo_offset = 0; state->lfo_range = 1; state->waveform = WF_Triangle; } static ALvoid ALchorusState_Destruct(ALchorusState *state) { al_free(state->SampleBuffer); state->SampleBuffer = NULL; ALeffectState_Destruct(STATIC_CAST(ALeffectState,state)); } static ALboolean ALchorusState_deviceUpdate(ALchorusState *state, ALCdevice *Device) { const ALfloat max_delay = maxf(AL_CHORUS_MAX_DELAY, AL_FLANGER_MAX_DELAY); ALsizei maxlen; maxlen = NextPowerOf2(float2int(max_delay*2.0f*Device->Frequency) + 1u); if(maxlen <= 0) return AL_FALSE; if(maxlen != state->BufferLength) { void *temp = al_calloc(16, maxlen * sizeof(ALfloat)); if(!temp) return AL_FALSE; al_free(state->SampleBuffer); state->SampleBuffer = temp; state->BufferLength = maxlen; } memset(state->SampleBuffer, 0, state->BufferLength*sizeof(ALfloat)); memset(state->Gains, 0, sizeof(state->Gains)); return AL_TRUE; } static ALvoid ALchorusState_update(ALchorusState *state, const ALCcontext *Context, const ALeffectslot *Slot, const ALeffectProps *props) { const ALsizei mindelay = MAX_RESAMPLE_PADDING << FRACTIONBITS; const ALCdevice *device = Context->Device; ALfloat frequency = (ALfloat)device->Frequency; ALfloat coeffs[MAX_AMBI_COEFFS]; ALfloat rate; ALint phase; switch(props->Chorus.Waveform) { case AL_CHORUS_WAVEFORM_TRIANGLE: state->waveform = WF_Triangle; break; case AL_CHORUS_WAVEFORM_SINUSOID: state->waveform = WF_Sinusoid; break; } /* The LFO depth is scaled to be relative to the sample delay. Clamp the * delay and depth to allow enough padding for resampling. */ state->delay = maxi(float2int(props->Chorus.Delay*frequency*FRACTIONONE + 0.5f), mindelay); state->depth = minf(props->Chorus.Depth * state->delay, (ALfloat)(state->delay - mindelay)); state->feedback = props->Chorus.Feedback; /* Gains for left and right sides */ CalcAngleCoeffs(-F_PI_2, 0.0f, 0.0f, coeffs); ComputePanGains(&device->Dry, coeffs, Slot->Params.Gain, state->Gains[0].Target); CalcAngleCoeffs( F_PI_2, 0.0f, 0.0f, coeffs); ComputePanGains(&device->Dry, coeffs, Slot->Params.Gain, state->Gains[1].Target); phase = props->Chorus.Phase; rate = props->Chorus.Rate; if(!(rate > 0.0f)) { state->lfo_offset = 0; state->lfo_range = 1; state->lfo_scale = 0.0f; state->lfo_disp = 0; } else { /* Calculate LFO coefficient (number of samples per cycle). Limit the * max range to avoid overflow when calculating the displacement. */ ALsizei lfo_range = float2int(minf(frequency/rate + 0.5f, (ALfloat)(INT_MAX/360 - 180))); state->lfo_offset = float2int((ALfloat)state->lfo_offset/state->lfo_range* lfo_range + 0.5f) % lfo_range; state->lfo_range = lfo_range; switch(state->waveform) { case WF_Triangle: state->lfo_scale = 4.0f / state->lfo_range; break; case WF_Sinusoid: state->lfo_scale = F_TAU / state->lfo_range; break; } /* Calculate lfo phase displacement */ if(phase < 0) phase = 360 + phase; state->lfo_disp = (state->lfo_range*phase + 180) / 360; } } static void GetTriangleDelays(ALint *restrict delays, ALsizei offset, const ALsizei lfo_range, const ALfloat lfo_scale, const ALfloat depth, const ALsizei delay, const ALsizei todo) { ALsizei i; for(i = 0;i < todo;i++) { delays[i] = fastf2i((1.0f - fabsf(2.0f - lfo_scale*offset)) * depth) + delay; offset = (offset+1)%lfo_range; } } static void GetSinusoidDelays(ALint *restrict delays, ALsizei offset, const ALsizei lfo_range, const ALfloat lfo_scale, const ALfloat depth, const ALsizei delay, const ALsizei todo) { ALsizei i; for(i = 0;i < todo;i++) { delays[i] = fastf2i(sinf(lfo_scale*offset) * depth) + delay; offset = (offset+1)%lfo_range; } } static ALvoid ALchorusState_process(ALchorusState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels) { const ALsizei bufmask = state->BufferLength-1; const ALfloat feedback = state->feedback; const ALsizei avgdelay = (state->delay + (FRACTIONONE>>1)) >> FRACTIONBITS; ALfloat *restrict delaybuf = state->SampleBuffer; ALsizei offset = state->offset; ALsizei i, c; ALsizei base; for(base = 0;base < SamplesToDo;) { const ALsizei todo = mini(256, SamplesToDo-base); ALint moddelays[2][256]; alignas(16) ALfloat temps[2][256]; if(state->waveform == WF_Sinusoid) { GetSinusoidDelays(moddelays[0], state->lfo_offset, state->lfo_range, state->lfo_scale, state->depth, state->delay, todo); GetSinusoidDelays(moddelays[1], (state->lfo_offset+state->lfo_disp)%state->lfo_range, state->lfo_range, state->lfo_scale, state->depth, state->delay, todo); } else /*if(state->waveform == WF_Triangle)*/ { GetTriangleDelays(moddelays[0], state->lfo_offset, state->lfo_range, state->lfo_scale, state->depth, state->delay, todo); GetTriangleDelays(moddelays[1], (state->lfo_offset+state->lfo_disp)%state->lfo_range, state->lfo_range, state->lfo_scale, state->depth, state->delay, todo); } state->lfo_offset = (state->lfo_offset+todo) % state->lfo_range; for(i = 0;i < todo;i++) { ALint delay; ALfloat mu; // Feed the buffer's input first (necessary for delays < 1). delaybuf[offset&bufmask] = SamplesIn[0][base+i]; // Tap for the left output. delay = offset - (moddelays[0][i]>>FRACTIONBITS); mu = (moddelays[0][i]&FRACTIONMASK) * (1.0f/FRACTIONONE); temps[0][i] = cubic(delaybuf[(delay+1) & bufmask], delaybuf[(delay ) & bufmask], delaybuf[(delay-1) & bufmask], delaybuf[(delay-2) & bufmask], mu); // Tap for the right output. delay = offset - (moddelays[1][i]>>FRACTIONBITS); mu = (moddelays[1][i]&FRACTIONMASK) * (1.0f/FRACTIONONE); temps[1][i] = cubic(delaybuf[(delay+1) & bufmask], delaybuf[(delay ) & bufmask], delaybuf[(delay-1) & bufmask], delaybuf[(delay-2) & bufmask], mu); // Accumulate feedback from the average delay of the taps. delaybuf[offset&bufmask] += delaybuf[(offset-avgdelay) & bufmask] * feedback; offset++; } for(c = 0;c < 2;c++) MixSamples(temps[c], NumChannels, SamplesOut, state->Gains[c].Current, state->Gains[c].Target, SamplesToDo-base, base, todo); base += todo; } state->offset = offset; } typedef struct ChorusStateFactory { DERIVE_FROM_TYPE(EffectStateFactory); } ChorusStateFactory; static ALeffectState *ChorusStateFactory_create(ChorusStateFactory *UNUSED(factory)) { ALchorusState *state; NEW_OBJ0(state, ALchorusState)(); if(!state) return NULL; return STATIC_CAST(ALeffectState, state); } DEFINE_EFFECTSTATEFACTORY_VTABLE(ChorusStateFactory); EffectStateFactory *ChorusStateFactory_getFactory(void) { static ChorusStateFactory ChorusFactory = { { GET_VTABLE2(ChorusStateFactory, EffectStateFactory) } }; return STATIC_CAST(EffectStateFactory, &ChorusFactory); } void ALchorus_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_CHORUS_WAVEFORM: if(!(val >= AL_CHORUS_MIN_WAVEFORM && val <= AL_CHORUS_MAX_WAVEFORM)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Invalid chorus waveform"); props->Chorus.Waveform = val; break; case AL_CHORUS_PHASE: if(!(val >= AL_CHORUS_MIN_PHASE && val <= AL_CHORUS_MAX_PHASE)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Chorus phase out of range"); props->Chorus.Phase = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid chorus integer property 0x%04x", param); } } void ALchorus_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals) { ALchorus_setParami(effect, context, param, vals[0]); } void ALchorus_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_CHORUS_RATE: if(!(val >= AL_CHORUS_MIN_RATE && val <= AL_CHORUS_MAX_RATE)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Chorus rate out of range"); props->Chorus.Rate = val; break; case AL_CHORUS_DEPTH: if(!(val >= AL_CHORUS_MIN_DEPTH && val <= AL_CHORUS_MAX_DEPTH)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Chorus depth out of range"); props->Chorus.Depth = val; break; case AL_CHORUS_FEEDBACK: if(!(val >= AL_CHORUS_MIN_FEEDBACK && val <= AL_CHORUS_MAX_FEEDBACK)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Chorus feedback out of range"); props->Chorus.Feedback = val; break; case AL_CHORUS_DELAY: if(!(val >= AL_CHORUS_MIN_DELAY && val <= AL_CHORUS_MAX_DELAY)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Chorus delay out of range"); props->Chorus.Delay = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid chorus float property 0x%04x", param); } } void ALchorus_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { ALchorus_setParamf(effect, context, param, vals[0]); } void ALchorus_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_CHORUS_WAVEFORM: *val = props->Chorus.Waveform; break; case AL_CHORUS_PHASE: *val = props->Chorus.Phase; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid chorus integer property 0x%04x", param); } } void ALchorus_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals) { ALchorus_getParami(effect, context, param, vals); } void ALchorus_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_CHORUS_RATE: *val = props->Chorus.Rate; break; case AL_CHORUS_DEPTH: *val = props->Chorus.Depth; break; case AL_CHORUS_FEEDBACK: *val = props->Chorus.Feedback; break; case AL_CHORUS_DELAY: *val = props->Chorus.Delay; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid chorus float property 0x%04x", param); } } void ALchorus_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { ALchorus_getParamf(effect, context, param, vals); } DEFINE_ALEFFECT_VTABLE(ALchorus); /* Flanger is basically a chorus with a really short delay. They can both use * the same processing functions, so piggyback flanger on the chorus functions. */ typedef struct FlangerStateFactory { DERIVE_FROM_TYPE(EffectStateFactory); } FlangerStateFactory; ALeffectState *FlangerStateFactory_create(FlangerStateFactory *UNUSED(factory)) { ALchorusState *state; NEW_OBJ0(state, ALchorusState)(); if(!state) return NULL; return STATIC_CAST(ALeffectState, state); } DEFINE_EFFECTSTATEFACTORY_VTABLE(FlangerStateFactory); EffectStateFactory *FlangerStateFactory_getFactory(void) { static FlangerStateFactory FlangerFactory = { { GET_VTABLE2(FlangerStateFactory, EffectStateFactory) } }; return STATIC_CAST(EffectStateFactory, &FlangerFactory); } void ALflanger_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_FLANGER_WAVEFORM: if(!(val >= AL_FLANGER_MIN_WAVEFORM && val <= AL_FLANGER_MAX_WAVEFORM)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Invalid flanger waveform"); props->Chorus.Waveform = val; break; case AL_FLANGER_PHASE: if(!(val >= AL_FLANGER_MIN_PHASE && val <= AL_FLANGER_MAX_PHASE)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Flanger phase out of range"); props->Chorus.Phase = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid flanger integer property 0x%04x", param); } } void ALflanger_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals) { ALflanger_setParami(effect, context, param, vals[0]); } void ALflanger_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_FLANGER_RATE: if(!(val >= AL_FLANGER_MIN_RATE && val <= AL_FLANGER_MAX_RATE)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Flanger rate out of range"); props->Chorus.Rate = val; break; case AL_FLANGER_DEPTH: if(!(val >= AL_FLANGER_MIN_DEPTH && val <= AL_FLANGER_MAX_DEPTH)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Flanger depth out of range"); props->Chorus.Depth = val; break; case AL_FLANGER_FEEDBACK: if(!(val >= AL_FLANGER_MIN_FEEDBACK && val <= AL_FLANGER_MAX_FEEDBACK)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Flanger feedback out of range"); props->Chorus.Feedback = val; break; case AL_FLANGER_DELAY: if(!(val >= AL_FLANGER_MIN_DELAY && val <= AL_FLANGER_MAX_DELAY)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Flanger delay out of range"); props->Chorus.Delay = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid flanger float property 0x%04x", param); } } void ALflanger_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { ALflanger_setParamf(effect, context, param, vals[0]); } void ALflanger_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_FLANGER_WAVEFORM: *val = props->Chorus.Waveform; break; case AL_FLANGER_PHASE: *val = props->Chorus.Phase; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid flanger integer property 0x%04x", param); } } void ALflanger_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals) { ALflanger_getParami(effect, context, param, vals); } void ALflanger_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_FLANGER_RATE: *val = props->Chorus.Rate; break; case AL_FLANGER_DEPTH: *val = props->Chorus.Depth; break; case AL_FLANGER_FEEDBACK: *val = props->Chorus.Feedback; break; case AL_FLANGER_DELAY: *val = props->Chorus.Delay; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid flanger float property 0x%04x", param); } } void ALflanger_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { ALflanger_getParamf(effect, context, param, vals); } DEFINE_ALEFFECT_VTABLE(ALflanger); openal-soft-openal-soft-1.19.1/Alc/effects/compressor.c000066400000000000000000000215621335774445300227620ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2013 by Anis A. Hireche * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include #include "config.h" #include "alError.h" #include "alMain.h" #include "alAuxEffectSlot.h" #include "alu.h" #define AMP_ENVELOPE_MIN 0.5f #define AMP_ENVELOPE_MAX 2.0f #define ATTACK_TIME 0.1f /* 100ms to rise from min to max */ #define RELEASE_TIME 0.2f /* 200ms to drop from max to min */ typedef struct ALcompressorState { DERIVE_FROM_TYPE(ALeffectState); /* Effect gains for each channel */ ALfloat Gain[MAX_EFFECT_CHANNELS][MAX_OUTPUT_CHANNELS]; /* Effect parameters */ ALboolean Enabled; ALfloat AttackMult; ALfloat ReleaseMult; ALfloat EnvFollower; } ALcompressorState; static ALvoid ALcompressorState_Destruct(ALcompressorState *state); static ALboolean ALcompressorState_deviceUpdate(ALcompressorState *state, ALCdevice *device); static ALvoid ALcompressorState_update(ALcompressorState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props); static ALvoid ALcompressorState_process(ALcompressorState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels); DECLARE_DEFAULT_ALLOCATORS(ALcompressorState) DEFINE_ALEFFECTSTATE_VTABLE(ALcompressorState); static void ALcompressorState_Construct(ALcompressorState *state) { ALeffectState_Construct(STATIC_CAST(ALeffectState, state)); SET_VTABLE2(ALcompressorState, ALeffectState, state); state->Enabled = AL_TRUE; state->AttackMult = 1.0f; state->ReleaseMult = 1.0f; state->EnvFollower = 1.0f; } static ALvoid ALcompressorState_Destruct(ALcompressorState *state) { ALeffectState_Destruct(STATIC_CAST(ALeffectState,state)); } static ALboolean ALcompressorState_deviceUpdate(ALcompressorState *state, ALCdevice *device) { /* Number of samples to do a full attack and release (non-integer sample * counts are okay). */ const ALfloat attackCount = (ALfloat)device->Frequency * ATTACK_TIME; const ALfloat releaseCount = (ALfloat)device->Frequency * RELEASE_TIME; /* Calculate per-sample multipliers to attack and release at the desired * rates. */ state->AttackMult = powf(AMP_ENVELOPE_MAX/AMP_ENVELOPE_MIN, 1.0f/attackCount); state->ReleaseMult = powf(AMP_ENVELOPE_MIN/AMP_ENVELOPE_MAX, 1.0f/releaseCount); return AL_TRUE; } static ALvoid ALcompressorState_update(ALcompressorState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props) { const ALCdevice *device = context->Device; ALuint i; state->Enabled = props->Compressor.OnOff; STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer; STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels; for(i = 0;i < 4;i++) ComputePanGains(&device->FOAOut, IdentityMatrixf.m[i], slot->Params.Gain, state->Gain[i]); } static ALvoid ALcompressorState_process(ALcompressorState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels) { ALsizei i, j, k; ALsizei base; for(base = 0;base < SamplesToDo;) { ALfloat gains[256]; ALsizei td = mini(256, SamplesToDo-base); ALfloat env = state->EnvFollower; /* Generate the per-sample gains from the signal envelope. */ if(state->Enabled) { for(i = 0;i < td;++i) { /* Clamp the absolute amplitude to the defined envelope limits, * then attack or release the envelope to reach it. */ ALfloat amplitude = clampf(fabsf(SamplesIn[0][base+i]), AMP_ENVELOPE_MIN, AMP_ENVELOPE_MAX); if(amplitude > env) env = minf(env*state->AttackMult, amplitude); else if(amplitude < env) env = maxf(env*state->ReleaseMult, amplitude); /* Apply the reciprocal of the envelope to normalize the volume * (compress the dynamic range). */ gains[i] = 1.0f / env; } } else { /* Same as above, except the amplitude is forced to 1. This helps * ensure smooth gain changes when the compressor is turned on and * off. */ for(i = 0;i < td;++i) { ALfloat amplitude = 1.0f; if(amplitude > env) env = minf(env*state->AttackMult, amplitude); else if(amplitude < env) env = maxf(env*state->ReleaseMult, amplitude); gains[i] = 1.0f / env; } } state->EnvFollower = env; /* Now compress the signal amplitude to output. */ for(j = 0;j < MAX_EFFECT_CHANNELS;j++) { for(k = 0;k < NumChannels;k++) { ALfloat gain = state->Gain[j][k]; if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD)) continue; for(i = 0;i < td;i++) SamplesOut[k][base+i] += SamplesIn[j][base+i] * gains[i] * gain; } } base += td; } } typedef struct CompressorStateFactory { DERIVE_FROM_TYPE(EffectStateFactory); } CompressorStateFactory; static ALeffectState *CompressorStateFactory_create(CompressorStateFactory *UNUSED(factory)) { ALcompressorState *state; NEW_OBJ0(state, ALcompressorState)(); if(!state) return NULL; return STATIC_CAST(ALeffectState, state); } DEFINE_EFFECTSTATEFACTORY_VTABLE(CompressorStateFactory); EffectStateFactory *CompressorStateFactory_getFactory(void) { static CompressorStateFactory CompressorFactory = { { GET_VTABLE2(CompressorStateFactory, EffectStateFactory) } }; return STATIC_CAST(EffectStateFactory, &CompressorFactory); } void ALcompressor_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_COMPRESSOR_ONOFF: if(!(val >= AL_COMPRESSOR_MIN_ONOFF && val <= AL_COMPRESSOR_MAX_ONOFF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Compressor state out of range"); props->Compressor.OnOff = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid compressor integer property 0x%04x", param); } } void ALcompressor_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals) { ALcompressor_setParami(effect, context, param, vals[0]); } void ALcompressor_setParamf(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid compressor float property 0x%04x", param); } void ALcompressor_setParamfv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALfloat *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid compressor float-vector property 0x%04x", param); } void ALcompressor_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_COMPRESSOR_ONOFF: *val = props->Compressor.OnOff; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid compressor integer property 0x%04x", param); } } void ALcompressor_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals) { ALcompressor_getParami(effect, context, param, vals); } void ALcompressor_getParamf(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat *UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid compressor float property 0x%04x", param); } void ALcompressor_getParamfv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid compressor float-vector property 0x%04x", param); } DEFINE_ALEFFECT_VTABLE(ALcompressor); openal-soft-openal-soft-1.19.1/Alc/effects/dedicated.c000066400000000000000000000157771335774445300225070ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2011 by Chris Robinson. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include "alMain.h" #include "alAuxEffectSlot.h" #include "alError.h" #include "alu.h" #include "filters/defs.h" typedef struct ALdedicatedState { DERIVE_FROM_TYPE(ALeffectState); ALfloat CurrentGains[MAX_OUTPUT_CHANNELS]; ALfloat TargetGains[MAX_OUTPUT_CHANNELS]; } ALdedicatedState; static ALvoid ALdedicatedState_Destruct(ALdedicatedState *state); static ALboolean ALdedicatedState_deviceUpdate(ALdedicatedState *state, ALCdevice *device); static ALvoid ALdedicatedState_update(ALdedicatedState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props); static ALvoid ALdedicatedState_process(ALdedicatedState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels); DECLARE_DEFAULT_ALLOCATORS(ALdedicatedState) DEFINE_ALEFFECTSTATE_VTABLE(ALdedicatedState); static void ALdedicatedState_Construct(ALdedicatedState *state) { ALeffectState_Construct(STATIC_CAST(ALeffectState, state)); SET_VTABLE2(ALdedicatedState, ALeffectState, state); } static ALvoid ALdedicatedState_Destruct(ALdedicatedState *state) { ALeffectState_Destruct(STATIC_CAST(ALeffectState,state)); } static ALboolean ALdedicatedState_deviceUpdate(ALdedicatedState *state, ALCdevice *UNUSED(device)) { ALsizei i; for(i = 0;i < MAX_OUTPUT_CHANNELS;i++) state->CurrentGains[i] = 0.0f; return AL_TRUE; } static ALvoid ALdedicatedState_update(ALdedicatedState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props) { const ALCdevice *device = context->Device; ALfloat Gain; ALsizei i; for(i = 0;i < MAX_OUTPUT_CHANNELS;i++) state->TargetGains[i] = 0.0f; Gain = slot->Params.Gain * props->Dedicated.Gain; if(slot->Params.EffectType == AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT) { int idx; if((idx=GetChannelIdxByName(&device->RealOut, LFE)) != -1) { STATIC_CAST(ALeffectState,state)->OutBuffer = device->RealOut.Buffer; STATIC_CAST(ALeffectState,state)->OutChannels = device->RealOut.NumChannels; state->TargetGains[idx] = Gain; } } else if(slot->Params.EffectType == AL_EFFECT_DEDICATED_DIALOGUE) { int idx; /* Dialog goes to the front-center speaker if it exists, otherwise it * plays from the front-center location. */ if((idx=GetChannelIdxByName(&device->RealOut, FrontCenter)) != -1) { STATIC_CAST(ALeffectState,state)->OutBuffer = device->RealOut.Buffer; STATIC_CAST(ALeffectState,state)->OutChannels = device->RealOut.NumChannels; state->TargetGains[idx] = Gain; } else { ALfloat coeffs[MAX_AMBI_COEFFS]; CalcAngleCoeffs(0.0f, 0.0f, 0.0f, coeffs); STATIC_CAST(ALeffectState,state)->OutBuffer = device->Dry.Buffer; STATIC_CAST(ALeffectState,state)->OutChannels = device->Dry.NumChannels; ComputePanGains(&device->Dry, coeffs, Gain, state->TargetGains); } } } static ALvoid ALdedicatedState_process(ALdedicatedState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels) { MixSamples(SamplesIn[0], NumChannels, SamplesOut, state->CurrentGains, state->TargetGains, SamplesToDo, 0, SamplesToDo); } typedef struct DedicatedStateFactory { DERIVE_FROM_TYPE(EffectStateFactory); } DedicatedStateFactory; ALeffectState *DedicatedStateFactory_create(DedicatedStateFactory *UNUSED(factory)) { ALdedicatedState *state; NEW_OBJ0(state, ALdedicatedState)(); if(!state) return NULL; return STATIC_CAST(ALeffectState, state); } DEFINE_EFFECTSTATEFACTORY_VTABLE(DedicatedStateFactory); EffectStateFactory *DedicatedStateFactory_getFactory(void) { static DedicatedStateFactory DedicatedFactory = { { GET_VTABLE2(DedicatedStateFactory, EffectStateFactory) } }; return STATIC_CAST(EffectStateFactory, &DedicatedFactory); } void ALdedicated_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid dedicated integer property 0x%04x", param); } void ALdedicated_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid dedicated integer-vector property 0x%04x", param); } void ALdedicated_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_DEDICATED_GAIN: if(!(val >= 0.0f && isfinite(val))) SETERR_RETURN(context, AL_INVALID_VALUE,, "Dedicated gain out of range"); props->Dedicated.Gain = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid dedicated float property 0x%04x", param); } } void ALdedicated_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { ALdedicated_setParamf(effect, context, param, vals[0]); } void ALdedicated_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid dedicated integer property 0x%04x", param); } void ALdedicated_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid dedicated integer-vector property 0x%04x", param); } void ALdedicated_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_DEDICATED_GAIN: *val = props->Dedicated.Gain; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid dedicated float property 0x%04x", param); } } void ALdedicated_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { ALdedicated_getParamf(effect, context, param, vals); } DEFINE_ALEFFECT_VTABLE(ALdedicated); openal-soft-openal-soft-1.19.1/Alc/effects/distortion.c000066400000000000000000000254131335774445300227630ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2013 by Mike Gorchak * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include "alMain.h" #include "alAuxEffectSlot.h" #include "alError.h" #include "alu.h" #include "filters/defs.h" typedef struct ALdistortionState { DERIVE_FROM_TYPE(ALeffectState); /* Effect gains for each channel */ ALfloat Gain[MAX_OUTPUT_CHANNELS]; /* Effect parameters */ BiquadFilter lowpass; BiquadFilter bandpass; ALfloat attenuation; ALfloat edge_coeff; ALfloat Buffer[2][BUFFERSIZE]; } ALdistortionState; static ALvoid ALdistortionState_Destruct(ALdistortionState *state); static ALboolean ALdistortionState_deviceUpdate(ALdistortionState *state, ALCdevice *device); static ALvoid ALdistortionState_update(ALdistortionState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props); static ALvoid ALdistortionState_process(ALdistortionState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels); DECLARE_DEFAULT_ALLOCATORS(ALdistortionState) DEFINE_ALEFFECTSTATE_VTABLE(ALdistortionState); static void ALdistortionState_Construct(ALdistortionState *state) { ALeffectState_Construct(STATIC_CAST(ALeffectState, state)); SET_VTABLE2(ALdistortionState, ALeffectState, state); } static ALvoid ALdistortionState_Destruct(ALdistortionState *state) { ALeffectState_Destruct(STATIC_CAST(ALeffectState,state)); } static ALboolean ALdistortionState_deviceUpdate(ALdistortionState *state, ALCdevice *UNUSED(device)) { BiquadFilter_clear(&state->lowpass); BiquadFilter_clear(&state->bandpass); return AL_TRUE; } static ALvoid ALdistortionState_update(ALdistortionState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props) { const ALCdevice *device = context->Device; ALfloat frequency = (ALfloat)device->Frequency; ALfloat coeffs[MAX_AMBI_COEFFS]; ALfloat bandwidth; ALfloat cutoff; ALfloat edge; /* Store waveshaper edge settings. */ edge = sinf(props->Distortion.Edge * (F_PI_2)); edge = minf(edge, 0.99f); state->edge_coeff = 2.0f * edge / (1.0f-edge); cutoff = props->Distortion.LowpassCutoff; /* Bandwidth value is constant in octaves. */ bandwidth = (cutoff / 2.0f) / (cutoff * 0.67f); /* Multiply sampling frequency by the amount of oversampling done during * processing. */ BiquadFilter_setParams(&state->lowpass, BiquadType_LowPass, 1.0f, cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth) ); cutoff = props->Distortion.EQCenter; /* Convert bandwidth in Hz to octaves. */ bandwidth = props->Distortion.EQBandwidth / (cutoff * 0.67f); BiquadFilter_setParams(&state->bandpass, BiquadType_BandPass, 1.0f, cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth) ); CalcAngleCoeffs(0.0f, 0.0f, 0.0f, coeffs); ComputePanGains(&device->Dry, coeffs, slot->Params.Gain*props->Distortion.Gain, state->Gain); } static ALvoid ALdistortionState_process(ALdistortionState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels) { ALfloat (*restrict buffer)[BUFFERSIZE] = state->Buffer; const ALfloat fc = state->edge_coeff; ALsizei base; ALsizei i, k; for(base = 0;base < SamplesToDo;) { /* Perform 4x oversampling to avoid aliasing. Oversampling greatly * improves distortion quality and allows to implement lowpass and * bandpass filters using high frequencies, at which classic IIR * filters became unstable. */ ALsizei todo = mini(BUFFERSIZE, (SamplesToDo-base) * 4); /* Fill oversample buffer using zero stuffing. Multiply the sample by * the amount of oversampling to maintain the signal's power. */ for(i = 0;i < todo;i++) buffer[0][i] = !(i&3) ? SamplesIn[0][(i>>2)+base] * 4.0f : 0.0f; /* First step, do lowpass filtering of original signal. Additionally * perform buffer interpolation and lowpass cutoff for oversampling * (which is fortunately first step of distortion). So combine three * operations into the one. */ BiquadFilter_process(&state->lowpass, buffer[1], buffer[0], todo); /* Second step, do distortion using waveshaper function to emulate * signal processing during tube overdriving. Three steps of * waveshaping are intended to modify waveform without boost/clipping/ * attenuation process. */ for(i = 0;i < todo;i++) { ALfloat smp = buffer[1][i]; smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp)); smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp)) * -1.0f; smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp)); buffer[0][i] = smp; } /* Third step, do bandpass filtering of distorted signal. */ BiquadFilter_process(&state->bandpass, buffer[1], buffer[0], todo); todo >>= 2; for(k = 0;k < NumChannels;k++) { /* Fourth step, final, do attenuation and perform decimation, * storing only one sample out of four. */ ALfloat gain = state->Gain[k]; if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD)) continue; for(i = 0;i < todo;i++) SamplesOut[k][base+i] += gain * buffer[1][i*4]; } base += todo; } } typedef struct DistortionStateFactory { DERIVE_FROM_TYPE(EffectStateFactory); } DistortionStateFactory; static ALeffectState *DistortionStateFactory_create(DistortionStateFactory *UNUSED(factory)) { ALdistortionState *state; NEW_OBJ0(state, ALdistortionState)(); if(!state) return NULL; return STATIC_CAST(ALeffectState, state); } DEFINE_EFFECTSTATEFACTORY_VTABLE(DistortionStateFactory); EffectStateFactory *DistortionStateFactory_getFactory(void) { static DistortionStateFactory DistortionFactory = { { GET_VTABLE2(DistortionStateFactory, EffectStateFactory) } }; return STATIC_CAST(EffectStateFactory, &DistortionFactory); } void ALdistortion_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid distortion integer property 0x%04x", param); } void ALdistortion_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid distortion integer-vector property 0x%04x", param); } void ALdistortion_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_DISTORTION_EDGE: if(!(val >= AL_DISTORTION_MIN_EDGE && val <= AL_DISTORTION_MAX_EDGE)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion edge out of range"); props->Distortion.Edge = val; break; case AL_DISTORTION_GAIN: if(!(val >= AL_DISTORTION_MIN_GAIN && val <= AL_DISTORTION_MAX_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion gain out of range"); props->Distortion.Gain = val; break; case AL_DISTORTION_LOWPASS_CUTOFF: if(!(val >= AL_DISTORTION_MIN_LOWPASS_CUTOFF && val <= AL_DISTORTION_MAX_LOWPASS_CUTOFF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion low-pass cutoff out of range"); props->Distortion.LowpassCutoff = val; break; case AL_DISTORTION_EQCENTER: if(!(val >= AL_DISTORTION_MIN_EQCENTER && val <= AL_DISTORTION_MAX_EQCENTER)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion EQ center out of range"); props->Distortion.EQCenter = val; break; case AL_DISTORTION_EQBANDWIDTH: if(!(val >= AL_DISTORTION_MIN_EQBANDWIDTH && val <= AL_DISTORTION_MAX_EQBANDWIDTH)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion EQ bandwidth out of range"); props->Distortion.EQBandwidth = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid distortion float property 0x%04x", param); } } void ALdistortion_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { ALdistortion_setParamf(effect, context, param, vals[0]); } void ALdistortion_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid distortion integer property 0x%04x", param); } void ALdistortion_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid distortion integer-vector property 0x%04x", param); } void ALdistortion_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_DISTORTION_EDGE: *val = props->Distortion.Edge; break; case AL_DISTORTION_GAIN: *val = props->Distortion.Gain; break; case AL_DISTORTION_LOWPASS_CUTOFF: *val = props->Distortion.LowpassCutoff; break; case AL_DISTORTION_EQCENTER: *val = props->Distortion.EQCenter; break; case AL_DISTORTION_EQBANDWIDTH: *val = props->Distortion.EQBandwidth; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid distortion float property 0x%04x", param); } } void ALdistortion_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { ALdistortion_getParamf(effect, context, param, vals); } DEFINE_ALEFFECT_VTABLE(ALdistortion); openal-soft-openal-soft-1.19.1/Alc/effects/echo.c000066400000000000000000000246711335774445300215100ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2009 by Chris Robinson. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include "alMain.h" #include "alFilter.h" #include "alAuxEffectSlot.h" #include "alError.h" #include "alu.h" #include "filters/defs.h" typedef struct ALechoState { DERIVE_FROM_TYPE(ALeffectState); ALfloat *SampleBuffer; ALsizei BufferLength; // The echo is two tap. The delay is the number of samples from before the // current offset struct { ALsizei delay; } Tap[2]; ALsizei Offset; /* The panning gains for the two taps */ struct { ALfloat Current[MAX_OUTPUT_CHANNELS]; ALfloat Target[MAX_OUTPUT_CHANNELS]; } Gains[2]; ALfloat FeedGain; BiquadFilter Filter; } ALechoState; static ALvoid ALechoState_Destruct(ALechoState *state); static ALboolean ALechoState_deviceUpdate(ALechoState *state, ALCdevice *Device); static ALvoid ALechoState_update(ALechoState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props); static ALvoid ALechoState_process(ALechoState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels); DECLARE_DEFAULT_ALLOCATORS(ALechoState) DEFINE_ALEFFECTSTATE_VTABLE(ALechoState); static void ALechoState_Construct(ALechoState *state) { ALeffectState_Construct(STATIC_CAST(ALeffectState, state)); SET_VTABLE2(ALechoState, ALeffectState, state); state->BufferLength = 0; state->SampleBuffer = NULL; state->Tap[0].delay = 0; state->Tap[1].delay = 0; state->Offset = 0; BiquadFilter_clear(&state->Filter); } static ALvoid ALechoState_Destruct(ALechoState *state) { al_free(state->SampleBuffer); state->SampleBuffer = NULL; ALeffectState_Destruct(STATIC_CAST(ALeffectState,state)); } static ALboolean ALechoState_deviceUpdate(ALechoState *state, ALCdevice *Device) { ALsizei maxlen; // Use the next power of 2 for the buffer length, so the tap offsets can be // wrapped using a mask instead of a modulo maxlen = float2int(AL_ECHO_MAX_DELAY*Device->Frequency + 0.5f) + float2int(AL_ECHO_MAX_LRDELAY*Device->Frequency + 0.5f); maxlen = NextPowerOf2(maxlen); if(maxlen <= 0) return AL_FALSE; if(maxlen != state->BufferLength) { void *temp = al_calloc(16, maxlen * sizeof(ALfloat)); if(!temp) return AL_FALSE; al_free(state->SampleBuffer); state->SampleBuffer = temp; state->BufferLength = maxlen; } memset(state->SampleBuffer, 0, state->BufferLength*sizeof(ALfloat)); memset(state->Gains, 0, sizeof(state->Gains)); return AL_TRUE; } static ALvoid ALechoState_update(ALechoState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props) { const ALCdevice *device = context->Device; ALuint frequency = device->Frequency; ALfloat coeffs[MAX_AMBI_COEFFS]; ALfloat gainhf, lrpan, spread; state->Tap[0].delay = maxi(float2int(props->Echo.Delay*frequency + 0.5f), 1); state->Tap[1].delay = float2int(props->Echo.LRDelay*frequency + 0.5f); state->Tap[1].delay += state->Tap[0].delay; spread = props->Echo.Spread; if(spread < 0.0f) lrpan = -1.0f; else lrpan = 1.0f; /* Convert echo spread (where 0 = omni, +/-1 = directional) to coverage * spread (where 0 = point, tau = omni). */ spread = asinf(1.0f - fabsf(spread))*4.0f; state->FeedGain = props->Echo.Feedback; gainhf = maxf(1.0f - props->Echo.Damping, 0.0625f); /* Limit -24dB */ BiquadFilter_setParams(&state->Filter, BiquadType_HighShelf, gainhf, LOWPASSFREQREF/frequency, calc_rcpQ_from_slope(gainhf, 1.0f) ); /* First tap panning */ CalcAngleCoeffs(-F_PI_2*lrpan, 0.0f, spread, coeffs); ComputePanGains(&device->Dry, coeffs, slot->Params.Gain, state->Gains[0].Target); /* Second tap panning */ CalcAngleCoeffs( F_PI_2*lrpan, 0.0f, spread, coeffs); ComputePanGains(&device->Dry, coeffs, slot->Params.Gain, state->Gains[1].Target); } static ALvoid ALechoState_process(ALechoState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels) { const ALsizei mask = state->BufferLength-1; const ALsizei tap1 = state->Tap[0].delay; const ALsizei tap2 = state->Tap[1].delay; ALfloat *restrict delaybuf = state->SampleBuffer; ALsizei offset = state->Offset; ALfloat z1, z2, in, out; ALsizei base; ALsizei c, i; z1 = state->Filter.z1; z2 = state->Filter.z2; for(base = 0;base < SamplesToDo;) { alignas(16) ALfloat temps[2][128]; ALsizei td = mini(128, SamplesToDo-base); for(i = 0;i < td;i++) { /* Feed the delay buffer's input first. */ delaybuf[offset&mask] = SamplesIn[0][i+base]; /* First tap */ temps[0][i] = delaybuf[(offset-tap1) & mask]; /* Second tap */ temps[1][i] = delaybuf[(offset-tap2) & mask]; /* Apply damping to the second tap, then add it to the buffer with * feedback attenuation. */ in = temps[1][i]; out = in*state->Filter.b0 + z1; z1 = in*state->Filter.b1 - out*state->Filter.a1 + z2; z2 = in*state->Filter.b2 - out*state->Filter.a2; delaybuf[offset&mask] += out * state->FeedGain; offset++; } for(c = 0;c < 2;c++) MixSamples(temps[c], NumChannels, SamplesOut, state->Gains[c].Current, state->Gains[c].Target, SamplesToDo-base, base, td); base += td; } state->Filter.z1 = z1; state->Filter.z2 = z2; state->Offset = offset; } typedef struct EchoStateFactory { DERIVE_FROM_TYPE(EffectStateFactory); } EchoStateFactory; ALeffectState *EchoStateFactory_create(EchoStateFactory *UNUSED(factory)) { ALechoState *state; NEW_OBJ0(state, ALechoState)(); if(!state) return NULL; return STATIC_CAST(ALeffectState, state); } DEFINE_EFFECTSTATEFACTORY_VTABLE(EchoStateFactory); EffectStateFactory *EchoStateFactory_getFactory(void) { static EchoStateFactory EchoFactory = { { GET_VTABLE2(EchoStateFactory, EffectStateFactory) } }; return STATIC_CAST(EffectStateFactory, &EchoFactory); } void ALecho_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid echo integer property 0x%04x", param); } void ALecho_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid echo integer-vector property 0x%04x", param); } void ALecho_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_ECHO_DELAY: if(!(val >= AL_ECHO_MIN_DELAY && val <= AL_ECHO_MAX_DELAY)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Echo delay out of range"); props->Echo.Delay = val; break; case AL_ECHO_LRDELAY: if(!(val >= AL_ECHO_MIN_LRDELAY && val <= AL_ECHO_MAX_LRDELAY)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Echo LR delay out of range"); props->Echo.LRDelay = val; break; case AL_ECHO_DAMPING: if(!(val >= AL_ECHO_MIN_DAMPING && val <= AL_ECHO_MAX_DAMPING)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Echo damping out of range"); props->Echo.Damping = val; break; case AL_ECHO_FEEDBACK: if(!(val >= AL_ECHO_MIN_FEEDBACK && val <= AL_ECHO_MAX_FEEDBACK)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Echo feedback out of range"); props->Echo.Feedback = val; break; case AL_ECHO_SPREAD: if(!(val >= AL_ECHO_MIN_SPREAD && val <= AL_ECHO_MAX_SPREAD)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Echo spread out of range"); props->Echo.Spread = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid echo float property 0x%04x", param); } } void ALecho_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { ALecho_setParamf(effect, context, param, vals[0]); } void ALecho_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid echo integer property 0x%04x", param); } void ALecho_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid echo integer-vector property 0x%04x", param); } void ALecho_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_ECHO_DELAY: *val = props->Echo.Delay; break; case AL_ECHO_LRDELAY: *val = props->Echo.LRDelay; break; case AL_ECHO_DAMPING: *val = props->Echo.Damping; break; case AL_ECHO_FEEDBACK: *val = props->Echo.Feedback; break; case AL_ECHO_SPREAD: *val = props->Echo.Spread; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid echo float property 0x%04x", param); } } void ALecho_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { ALecho_getParamf(effect, context, param, vals); } DEFINE_ALEFFECT_VTABLE(ALecho); openal-soft-openal-soft-1.19.1/Alc/effects/equalizer.c000066400000000000000000000362341335774445300225710ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2013 by Mike Gorchak * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include "alMain.h" #include "alAuxEffectSlot.h" #include "alError.h" #include "alu.h" #include "filters/defs.h" /* The document "Effects Extension Guide.pdf" says that low and high * * frequencies are cutoff frequencies. This is not fully correct, they * * are corner frequencies for low and high shelf filters. If they were * * just cutoff frequencies, there would be no need in cutoff frequency * * gains, which are present. Documentation for "Creative Proteus X2" * * software describes 4-band equalizer functionality in a much better * * way. This equalizer seems to be a predecessor of OpenAL 4-band * * equalizer. With low and high shelf filters we are able to cutoff * * frequencies below and/or above corner frequencies using attenuation * * gains (below 1.0) and amplify all low and/or high frequencies using * * gains above 1.0. * * * * Low-shelf Low Mid Band High Mid Band High-shelf * * corner center center corner * * frequency frequency frequency frequency * * 50Hz..800Hz 200Hz..3000Hz 1000Hz..8000Hz 4000Hz..16000Hz * * * * | | | | * * | | | | * * B -----+ /--+--\ /--+--\ +----- * * O |\ | | | | | | /| * * O | \ - | - - | - / | * * S + | \ | | | | | | / | * * T | | | | | | | | | | * * ---------+---------------+------------------+---------------+-------- * * C | | | | | | | | | | * * U - | / | | | | | | \ | * * T | / - | - - | - \ | * * O |/ | | | | | | \| * * F -----+ \--+--/ \--+--/ +----- * * F | | | | * * | | | | * * * * Gains vary from 0.126 up to 7.943, which means from -18dB attenuation * * up to +18dB amplification. Band width varies from 0.01 up to 1.0 in * * octaves for two mid bands. * * * * Implementation is based on the "Cookbook formulae for audio EQ biquad * * filter coefficients" by Robert Bristow-Johnson * * http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt */ typedef struct ALequalizerState { DERIVE_FROM_TYPE(ALeffectState); struct { /* Effect parameters */ BiquadFilter filter[4]; /* Effect gains for each channel */ ALfloat CurrentGains[MAX_OUTPUT_CHANNELS]; ALfloat TargetGains[MAX_OUTPUT_CHANNELS]; } Chans[MAX_EFFECT_CHANNELS]; ALfloat SampleBuffer[MAX_EFFECT_CHANNELS][BUFFERSIZE]; } ALequalizerState; static ALvoid ALequalizerState_Destruct(ALequalizerState *state); static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *state, ALCdevice *device); static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props); static ALvoid ALequalizerState_process(ALequalizerState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels); DECLARE_DEFAULT_ALLOCATORS(ALequalizerState) DEFINE_ALEFFECTSTATE_VTABLE(ALequalizerState); static void ALequalizerState_Construct(ALequalizerState *state) { ALeffectState_Construct(STATIC_CAST(ALeffectState, state)); SET_VTABLE2(ALequalizerState, ALeffectState, state); } static ALvoid ALequalizerState_Destruct(ALequalizerState *state) { ALeffectState_Destruct(STATIC_CAST(ALeffectState,state)); } static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *state, ALCdevice *UNUSED(device)) { ALsizei i, j; for(i = 0; i < MAX_EFFECT_CHANNELS;i++) { for(j = 0;j < 4;j++) BiquadFilter_clear(&state->Chans[i].filter[j]); for(j = 0;j < MAX_OUTPUT_CHANNELS;j++) state->Chans[i].CurrentGains[j] = 0.0f; } return AL_TRUE; } static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props) { const ALCdevice *device = context->Device; ALfloat frequency = (ALfloat)device->Frequency; ALfloat gain, f0norm; ALuint i; /* Calculate coefficients for the each type of filter. Note that the shelf * filters' gain is for the reference frequency, which is the centerpoint * of the transition band. */ gain = maxf(sqrtf(props->Equalizer.LowGain), 0.0625f); /* Limit -24dB */ f0norm = props->Equalizer.LowCutoff/frequency; BiquadFilter_setParams(&state->Chans[0].filter[0], BiquadType_LowShelf, gain, f0norm, calc_rcpQ_from_slope(gain, 0.75f) ); gain = maxf(props->Equalizer.Mid1Gain, 0.0625f); f0norm = props->Equalizer.Mid1Center/frequency; BiquadFilter_setParams(&state->Chans[0].filter[1], BiquadType_Peaking, gain, f0norm, calc_rcpQ_from_bandwidth( f0norm, props->Equalizer.Mid1Width ) ); gain = maxf(props->Equalizer.Mid2Gain, 0.0625f); f0norm = props->Equalizer.Mid2Center/frequency; BiquadFilter_setParams(&state->Chans[0].filter[2], BiquadType_Peaking, gain, f0norm, calc_rcpQ_from_bandwidth( f0norm, props->Equalizer.Mid2Width ) ); gain = maxf(sqrtf(props->Equalizer.HighGain), 0.0625f); f0norm = props->Equalizer.HighCutoff/frequency; BiquadFilter_setParams(&state->Chans[0].filter[3], BiquadType_HighShelf, gain, f0norm, calc_rcpQ_from_slope(gain, 0.75f) ); /* Copy the filter coefficients for the other input channels. */ for(i = 1;i < MAX_EFFECT_CHANNELS;i++) { BiquadFilter_copyParams(&state->Chans[i].filter[0], &state->Chans[0].filter[0]); BiquadFilter_copyParams(&state->Chans[i].filter[1], &state->Chans[0].filter[1]); BiquadFilter_copyParams(&state->Chans[i].filter[2], &state->Chans[0].filter[2]); BiquadFilter_copyParams(&state->Chans[i].filter[3], &state->Chans[0].filter[3]); } STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer; STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels; for(i = 0;i < MAX_EFFECT_CHANNELS;i++) ComputePanGains(&device->FOAOut, IdentityMatrixf.m[i], slot->Params.Gain, state->Chans[i].TargetGains); } static ALvoid ALequalizerState_process(ALequalizerState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels) { ALfloat (*restrict temps)[BUFFERSIZE] = state->SampleBuffer; ALsizei c; for(c = 0;c < MAX_EFFECT_CHANNELS;c++) { BiquadFilter_process(&state->Chans[c].filter[0], temps[0], SamplesIn[c], SamplesToDo); BiquadFilter_process(&state->Chans[c].filter[1], temps[1], temps[0], SamplesToDo); BiquadFilter_process(&state->Chans[c].filter[2], temps[2], temps[1], SamplesToDo); BiquadFilter_process(&state->Chans[c].filter[3], temps[3], temps[2], SamplesToDo); MixSamples(temps[3], NumChannels, SamplesOut, state->Chans[c].CurrentGains, state->Chans[c].TargetGains, SamplesToDo, 0, SamplesToDo ); } } typedef struct EqualizerStateFactory { DERIVE_FROM_TYPE(EffectStateFactory); } EqualizerStateFactory; ALeffectState *EqualizerStateFactory_create(EqualizerStateFactory *UNUSED(factory)) { ALequalizerState *state; NEW_OBJ0(state, ALequalizerState)(); if(!state) return NULL; return STATIC_CAST(ALeffectState, state); } DEFINE_EFFECTSTATEFACTORY_VTABLE(EqualizerStateFactory); EffectStateFactory *EqualizerStateFactory_getFactory(void) { static EqualizerStateFactory EqualizerFactory = { { GET_VTABLE2(EqualizerStateFactory, EffectStateFactory) } }; return STATIC_CAST(EffectStateFactory, &EqualizerFactory); } void ALequalizer_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid equalizer integer property 0x%04x", param); } void ALequalizer_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid equalizer integer-vector property 0x%04x", param); } void ALequalizer_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_EQUALIZER_LOW_GAIN: if(!(val >= AL_EQUALIZER_MIN_LOW_GAIN && val <= AL_EQUALIZER_MAX_LOW_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer low-band gain out of range"); props->Equalizer.LowGain = val; break; case AL_EQUALIZER_LOW_CUTOFF: if(!(val >= AL_EQUALIZER_MIN_LOW_CUTOFF && val <= AL_EQUALIZER_MAX_LOW_CUTOFF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer low-band cutoff out of range"); props->Equalizer.LowCutoff = val; break; case AL_EQUALIZER_MID1_GAIN: if(!(val >= AL_EQUALIZER_MIN_MID1_GAIN && val <= AL_EQUALIZER_MAX_MID1_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer mid1-band gain out of range"); props->Equalizer.Mid1Gain = val; break; case AL_EQUALIZER_MID1_CENTER: if(!(val >= AL_EQUALIZER_MIN_MID1_CENTER && val <= AL_EQUALIZER_MAX_MID1_CENTER)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer mid1-band center out of range"); props->Equalizer.Mid1Center = val; break; case AL_EQUALIZER_MID1_WIDTH: if(!(val >= AL_EQUALIZER_MIN_MID1_WIDTH && val <= AL_EQUALIZER_MAX_MID1_WIDTH)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer mid1-band width out of range"); props->Equalizer.Mid1Width = val; break; case AL_EQUALIZER_MID2_GAIN: if(!(val >= AL_EQUALIZER_MIN_MID2_GAIN && val <= AL_EQUALIZER_MAX_MID2_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer mid2-band gain out of range"); props->Equalizer.Mid2Gain = val; break; case AL_EQUALIZER_MID2_CENTER: if(!(val >= AL_EQUALIZER_MIN_MID2_CENTER && val <= AL_EQUALIZER_MAX_MID2_CENTER)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer mid2-band center out of range"); props->Equalizer.Mid2Center = val; break; case AL_EQUALIZER_MID2_WIDTH: if(!(val >= AL_EQUALIZER_MIN_MID2_WIDTH && val <= AL_EQUALIZER_MAX_MID2_WIDTH)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer mid2-band width out of range"); props->Equalizer.Mid2Width = val; break; case AL_EQUALIZER_HIGH_GAIN: if(!(val >= AL_EQUALIZER_MIN_HIGH_GAIN && val <= AL_EQUALIZER_MAX_HIGH_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer high-band gain out of range"); props->Equalizer.HighGain = val; break; case AL_EQUALIZER_HIGH_CUTOFF: if(!(val >= AL_EQUALIZER_MIN_HIGH_CUTOFF && val <= AL_EQUALIZER_MAX_HIGH_CUTOFF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer high-band cutoff out of range"); props->Equalizer.HighCutoff = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid equalizer float property 0x%04x", param); } } void ALequalizer_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { ALequalizer_setParamf(effect, context, param, vals[0]); } void ALequalizer_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid equalizer integer property 0x%04x", param); } void ALequalizer_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid equalizer integer-vector property 0x%04x", param); } void ALequalizer_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_EQUALIZER_LOW_GAIN: *val = props->Equalizer.LowGain; break; case AL_EQUALIZER_LOW_CUTOFF: *val = props->Equalizer.LowCutoff; break; case AL_EQUALIZER_MID1_GAIN: *val = props->Equalizer.Mid1Gain; break; case AL_EQUALIZER_MID1_CENTER: *val = props->Equalizer.Mid1Center; break; case AL_EQUALIZER_MID1_WIDTH: *val = props->Equalizer.Mid1Width; break; case AL_EQUALIZER_MID2_GAIN: *val = props->Equalizer.Mid2Gain; break; case AL_EQUALIZER_MID2_CENTER: *val = props->Equalizer.Mid2Center; break; case AL_EQUALIZER_MID2_WIDTH: *val = props->Equalizer.Mid2Width; break; case AL_EQUALIZER_HIGH_GAIN: *val = props->Equalizer.HighGain; break; case AL_EQUALIZER_HIGH_CUTOFF: *val = props->Equalizer.HighCutoff; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid equalizer float property 0x%04x", param); } } void ALequalizer_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { ALequalizer_getParamf(effect, context, param, vals); } DEFINE_ALEFFECT_VTABLE(ALequalizer); openal-soft-openal-soft-1.19.1/Alc/effects/fshifter.c000066400000000000000000000261261335774445300224010ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2018 by Raul Herraiz. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include "alMain.h" #include "alAuxEffectSlot.h" #include "alError.h" #include "alu.h" #include "filters/defs.h" #include "alcomplex.h" #define HIL_SIZE 1024 #define OVERSAMP (1<<2) #define HIL_STEP (HIL_SIZE / OVERSAMP) #define FIFO_LATENCY (HIL_STEP * (OVERSAMP-1)) typedef struct ALfshifterState { DERIVE_FROM_TYPE(ALeffectState); /* Effect parameters */ ALsizei count; ALsizei PhaseStep; ALsizei Phase; ALdouble ld_sign; /*Effects buffers*/ ALfloat InFIFO[HIL_SIZE]; ALcomplex OutFIFO[HIL_SIZE]; ALcomplex OutputAccum[HIL_SIZE]; ALcomplex Analytic[HIL_SIZE]; ALcomplex Outdata[BUFFERSIZE]; alignas(16) ALfloat BufferOut[BUFFERSIZE]; /* Effect gains for each output channel */ ALfloat CurrentGains[MAX_OUTPUT_CHANNELS]; ALfloat TargetGains[MAX_OUTPUT_CHANNELS]; } ALfshifterState; static ALvoid ALfshifterState_Destruct(ALfshifterState *state); static ALboolean ALfshifterState_deviceUpdate(ALfshifterState *state, ALCdevice *device); static ALvoid ALfshifterState_update(ALfshifterState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props); static ALvoid ALfshifterState_process(ALfshifterState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels); DECLARE_DEFAULT_ALLOCATORS(ALfshifterState) DEFINE_ALEFFECTSTATE_VTABLE(ALfshifterState); /* Define a Hann window, used to filter the HIL input and output. */ alignas(16) static ALdouble HannWindow[HIL_SIZE]; static void InitHannWindow(void) { ALsizei i; /* Create lookup table of the Hann window for the desired size, i.e. HIL_SIZE */ for(i = 0;i < HIL_SIZE>>1;i++) { ALdouble val = sin(M_PI * (ALdouble)i / (ALdouble)(HIL_SIZE-1)); HannWindow[i] = HannWindow[HIL_SIZE-1-i] = val * val; } } static alonce_flag HannInitOnce = AL_ONCE_FLAG_INIT; static void ALfshifterState_Construct(ALfshifterState *state) { ALeffectState_Construct(STATIC_CAST(ALeffectState, state)); SET_VTABLE2(ALfshifterState, ALeffectState, state); alcall_once(&HannInitOnce, InitHannWindow); } static ALvoid ALfshifterState_Destruct(ALfshifterState *state) { ALeffectState_Destruct(STATIC_CAST(ALeffectState,state)); } static ALboolean ALfshifterState_deviceUpdate(ALfshifterState *state, ALCdevice *UNUSED(device)) { /* (Re-)initializing parameters and clear the buffers. */ state->count = FIFO_LATENCY; state->PhaseStep = 0; state->Phase = 0; state->ld_sign = 1.0; memset(state->InFIFO, 0, sizeof(state->InFIFO)); memset(state->OutFIFO, 0, sizeof(state->OutFIFO)); memset(state->OutputAccum, 0, sizeof(state->OutputAccum)); memset(state->Analytic, 0, sizeof(state->Analytic)); memset(state->CurrentGains, 0, sizeof(state->CurrentGains)); memset(state->TargetGains, 0, sizeof(state->TargetGains)); return AL_TRUE; } static ALvoid ALfshifterState_update(ALfshifterState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props) { const ALCdevice *device = context->Device; ALfloat coeffs[MAX_AMBI_COEFFS]; ALfloat step; step = props->Fshifter.Frequency / (ALfloat)device->Frequency; state->PhaseStep = fastf2i(minf(step, 0.5f) * FRACTIONONE); switch(props->Fshifter.LeftDirection) { case AL_FREQUENCY_SHIFTER_DIRECTION_DOWN: state->ld_sign = -1.0; break; case AL_FREQUENCY_SHIFTER_DIRECTION_UP: state->ld_sign = 1.0; break; case AL_FREQUENCY_SHIFTER_DIRECTION_OFF: state->Phase = 0; state->PhaseStep = 0; break; } CalcAngleCoeffs(0.0f, 0.0f, 0.0f, coeffs); ComputePanGains(&device->Dry, coeffs, slot->Params.Gain, state->TargetGains); } static ALvoid ALfshifterState_process(ALfshifterState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels) { static const ALcomplex complex_zero = { 0.0, 0.0 }; ALfloat *restrict BufferOut = state->BufferOut; ALsizei j, k, base; for(base = 0;base < SamplesToDo;) { ALsizei todo = mini(HIL_SIZE-state->count, SamplesToDo-base); ASSUME(todo > 0); /* Fill FIFO buffer with samples data */ k = state->count; for(j = 0;j < todo;j++,k++) { state->InFIFO[k] = SamplesIn[0][base+j]; state->Outdata[base+j] = state->OutFIFO[k-FIFO_LATENCY]; } state->count += todo; base += todo; /* Check whether FIFO buffer is filled */ if(state->count < HIL_SIZE) continue; state->count = FIFO_LATENCY; /* Real signal windowing and store in Analytic buffer */ for(k = 0;k < HIL_SIZE;k++) { state->Analytic[k].Real = state->InFIFO[k] * HannWindow[k]; state->Analytic[k].Imag = 0.0; } /* Processing signal by Discrete Hilbert Transform (analytical signal). */ complex_hilbert(state->Analytic, HIL_SIZE); /* Windowing and add to output accumulator */ for(k = 0;k < HIL_SIZE;k++) { state->OutputAccum[k].Real += 2.0/OVERSAMP*HannWindow[k]*state->Analytic[k].Real; state->OutputAccum[k].Imag += 2.0/OVERSAMP*HannWindow[k]*state->Analytic[k].Imag; } /* Shift accumulator, input & output FIFO */ for(k = 0;k < HIL_STEP;k++) state->OutFIFO[k] = state->OutputAccum[k]; for(j = 0;k < HIL_SIZE;k++,j++) state->OutputAccum[j] = state->OutputAccum[k]; for(;j < HIL_SIZE;j++) state->OutputAccum[j] = complex_zero; for(k = 0;k < FIFO_LATENCY;k++) state->InFIFO[k] = state->InFIFO[k+HIL_STEP]; } /* Process frequency shifter using the analytic signal obtained. */ for(k = 0;k < SamplesToDo;k++) { ALdouble phase = state->Phase * ((1.0/FRACTIONONE) * 2.0*M_PI); BufferOut[k] = (ALfloat)(state->Outdata[k].Real*cos(phase) + state->Outdata[k].Imag*sin(phase)*state->ld_sign); state->Phase += state->PhaseStep; state->Phase &= FRACTIONMASK; } /* Now, mix the processed sound data to the output. */ MixSamples(BufferOut, NumChannels, SamplesOut, state->CurrentGains, state->TargetGains, maxi(SamplesToDo, 512), 0, SamplesToDo); } typedef struct FshifterStateFactory { DERIVE_FROM_TYPE(EffectStateFactory); } FshifterStateFactory; static ALeffectState *FshifterStateFactory_create(FshifterStateFactory *UNUSED(factory)) { ALfshifterState *state; NEW_OBJ0(state, ALfshifterState)(); if(!state) return NULL; return STATIC_CAST(ALeffectState, state); } DEFINE_EFFECTSTATEFACTORY_VTABLE(FshifterStateFactory); EffectStateFactory *FshifterStateFactory_getFactory(void) { static FshifterStateFactory FshifterFactory = { { GET_VTABLE2(FshifterStateFactory, EffectStateFactory) } }; return STATIC_CAST(EffectStateFactory, &FshifterFactory); } void ALfshifter_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_FREQUENCY_SHIFTER_FREQUENCY: if(!(val >= AL_FREQUENCY_SHIFTER_MIN_FREQUENCY && val <= AL_FREQUENCY_SHIFTER_MAX_FREQUENCY)) SETERR_RETURN(context, AL_INVALID_VALUE,,"Frequency shifter frequency out of range"); props->Fshifter.Frequency = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid frequency shifter float property 0x%04x", param); } } void ALfshifter_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { ALfshifter_setParamf(effect, context, param, vals[0]); } void ALfshifter_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_FREQUENCY_SHIFTER_LEFT_DIRECTION: if(!(val >= AL_FREQUENCY_SHIFTER_MIN_LEFT_DIRECTION && val <= AL_FREQUENCY_SHIFTER_MAX_LEFT_DIRECTION)) SETERR_RETURN(context, AL_INVALID_VALUE,,"Frequency shifter left direction out of range"); props->Fshifter.LeftDirection = val; break; case AL_FREQUENCY_SHIFTER_RIGHT_DIRECTION: if(!(val >= AL_FREQUENCY_SHIFTER_MIN_RIGHT_DIRECTION && val <= AL_FREQUENCY_SHIFTER_MAX_RIGHT_DIRECTION)) SETERR_RETURN(context, AL_INVALID_VALUE,,"Frequency shifter right direction out of range"); props->Fshifter.RightDirection = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid frequency shifter integer property 0x%04x", param); } } void ALfshifter_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals) { ALfshifter_setParami(effect, context, param, vals[0]); } void ALfshifter_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_FREQUENCY_SHIFTER_LEFT_DIRECTION: *val = props->Fshifter.LeftDirection; break; case AL_FREQUENCY_SHIFTER_RIGHT_DIRECTION: *val = props->Fshifter.RightDirection; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid frequency shifter integer property 0x%04x", param); } } void ALfshifter_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals) { ALfshifter_getParami(effect, context, param, vals); } void ALfshifter_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_FREQUENCY_SHIFTER_FREQUENCY: *val = props->Fshifter.Frequency; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid frequency shifter float property 0x%04x", param); } } void ALfshifter_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { ALfshifter_getParamf(effect, context, param, vals); } DEFINE_ALEFFECT_VTABLE(ALfshifter); openal-soft-openal-soft-1.19.1/Alc/effects/modulator.c000066400000000000000000000257331335774445300226000ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2009 by Chris Robinson. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include "alMain.h" #include "alAuxEffectSlot.h" #include "alError.h" #include "alu.h" #include "filters/defs.h" #define MAX_UPDATE_SAMPLES 128 typedef struct ALmodulatorState { DERIVE_FROM_TYPE(ALeffectState); void (*GetSamples)(ALfloat*, ALsizei, const ALsizei, ALsizei); ALsizei index; ALsizei step; struct { BiquadFilter Filter; ALfloat CurrentGains[MAX_OUTPUT_CHANNELS]; ALfloat TargetGains[MAX_OUTPUT_CHANNELS]; } Chans[MAX_EFFECT_CHANNELS]; } ALmodulatorState; static ALvoid ALmodulatorState_Destruct(ALmodulatorState *state); static ALboolean ALmodulatorState_deviceUpdate(ALmodulatorState *state, ALCdevice *device); static ALvoid ALmodulatorState_update(ALmodulatorState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props); static ALvoid ALmodulatorState_process(ALmodulatorState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels); DECLARE_DEFAULT_ALLOCATORS(ALmodulatorState) DEFINE_ALEFFECTSTATE_VTABLE(ALmodulatorState); #define WAVEFORM_FRACBITS 24 #define WAVEFORM_FRACONE (1<>(WAVEFORM_FRACBITS-2))&2) - 1); } static inline ALfloat One(ALsizei UNUSED(index)) { return 1.0f; } #define DECL_TEMPLATE(func) \ static void Modulate##func(ALfloat *restrict dst, ALsizei index, \ const ALsizei step, ALsizei todo) \ { \ ALsizei i; \ for(i = 0;i < todo;i++) \ { \ index += step; \ index &= WAVEFORM_FRACMASK; \ dst[i] = func(index); \ } \ } DECL_TEMPLATE(Sin) DECL_TEMPLATE(Saw) DECL_TEMPLATE(Square) DECL_TEMPLATE(One) #undef DECL_TEMPLATE static void ALmodulatorState_Construct(ALmodulatorState *state) { ALeffectState_Construct(STATIC_CAST(ALeffectState, state)); SET_VTABLE2(ALmodulatorState, ALeffectState, state); state->index = 0; state->step = 1; } static ALvoid ALmodulatorState_Destruct(ALmodulatorState *state) { ALeffectState_Destruct(STATIC_CAST(ALeffectState,state)); } static ALboolean ALmodulatorState_deviceUpdate(ALmodulatorState *state, ALCdevice *UNUSED(device)) { ALsizei i, j; for(i = 0;i < MAX_EFFECT_CHANNELS;i++) { BiquadFilter_clear(&state->Chans[i].Filter); for(j = 0;j < MAX_OUTPUT_CHANNELS;j++) state->Chans[i].CurrentGains[j] = 0.0f; } return AL_TRUE; } static ALvoid ALmodulatorState_update(ALmodulatorState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props) { const ALCdevice *device = context->Device; ALfloat f0norm; ALsizei i; state->step = fastf2i(props->Modulator.Frequency / (ALfloat)device->Frequency * WAVEFORM_FRACONE); state->step = clampi(state->step, 0, WAVEFORM_FRACONE-1); if(state->step == 0) state->GetSamples = ModulateOne; else if(props->Modulator.Waveform == AL_RING_MODULATOR_SINUSOID) state->GetSamples = ModulateSin; else if(props->Modulator.Waveform == AL_RING_MODULATOR_SAWTOOTH) state->GetSamples = ModulateSaw; else /*if(Slot->Params.EffectProps.Modulator.Waveform == AL_RING_MODULATOR_SQUARE)*/ state->GetSamples = ModulateSquare; f0norm = props->Modulator.HighPassCutoff / (ALfloat)device->Frequency; f0norm = clampf(f0norm, 1.0f/512.0f, 0.49f); /* Bandwidth value is constant in octaves. */ BiquadFilter_setParams(&state->Chans[0].Filter, BiquadType_HighPass, 1.0f, f0norm, calc_rcpQ_from_bandwidth(f0norm, 0.75f)); for(i = 1;i < MAX_EFFECT_CHANNELS;i++) BiquadFilter_copyParams(&state->Chans[i].Filter, &state->Chans[0].Filter); STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer; STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels; for(i = 0;i < MAX_EFFECT_CHANNELS;i++) ComputePanGains(&device->FOAOut, IdentityMatrixf.m[i], slot->Params.Gain, state->Chans[i].TargetGains); } static ALvoid ALmodulatorState_process(ALmodulatorState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels) { const ALsizei step = state->step; ALsizei base; for(base = 0;base < SamplesToDo;) { alignas(16) ALfloat modsamples[MAX_UPDATE_SAMPLES]; ALsizei td = mini(MAX_UPDATE_SAMPLES, SamplesToDo-base); ALsizei c, i; state->GetSamples(modsamples, state->index, step, td); state->index += (step*td) & WAVEFORM_FRACMASK; state->index &= WAVEFORM_FRACMASK; for(c = 0;c < MAX_EFFECT_CHANNELS;c++) { alignas(16) ALfloat temps[MAX_UPDATE_SAMPLES]; BiquadFilter_process(&state->Chans[c].Filter, temps, &SamplesIn[c][base], td); for(i = 0;i < td;i++) temps[i] *= modsamples[i]; MixSamples(temps, NumChannels, SamplesOut, state->Chans[c].CurrentGains, state->Chans[c].TargetGains, SamplesToDo-base, base, td); } base += td; } } typedef struct ModulatorStateFactory { DERIVE_FROM_TYPE(EffectStateFactory); } ModulatorStateFactory; static ALeffectState *ModulatorStateFactory_create(ModulatorStateFactory *UNUSED(factory)) { ALmodulatorState *state; NEW_OBJ0(state, ALmodulatorState)(); if(!state) return NULL; return STATIC_CAST(ALeffectState, state); } DEFINE_EFFECTSTATEFACTORY_VTABLE(ModulatorStateFactory); EffectStateFactory *ModulatorStateFactory_getFactory(void) { static ModulatorStateFactory ModulatorFactory = { { GET_VTABLE2(ModulatorStateFactory, EffectStateFactory) } }; return STATIC_CAST(EffectStateFactory, &ModulatorFactory); } void ALmodulator_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_RING_MODULATOR_FREQUENCY: if(!(val >= AL_RING_MODULATOR_MIN_FREQUENCY && val <= AL_RING_MODULATOR_MAX_FREQUENCY)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Modulator frequency out of range"); props->Modulator.Frequency = val; break; case AL_RING_MODULATOR_HIGHPASS_CUTOFF: if(!(val >= AL_RING_MODULATOR_MIN_HIGHPASS_CUTOFF && val <= AL_RING_MODULATOR_MAX_HIGHPASS_CUTOFF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Modulator high-pass cutoff out of range"); props->Modulator.HighPassCutoff = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid modulator float property 0x%04x", param); } } void ALmodulator_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { ALmodulator_setParamf(effect, context, param, vals[0]); } void ALmodulator_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_RING_MODULATOR_FREQUENCY: case AL_RING_MODULATOR_HIGHPASS_CUTOFF: ALmodulator_setParamf(effect, context, param, (ALfloat)val); break; case AL_RING_MODULATOR_WAVEFORM: if(!(val >= AL_RING_MODULATOR_MIN_WAVEFORM && val <= AL_RING_MODULATOR_MAX_WAVEFORM)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Invalid modulator waveform"); props->Modulator.Waveform = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid modulator integer property 0x%04x", param); } } void ALmodulator_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals) { ALmodulator_setParami(effect, context, param, vals[0]); } void ALmodulator_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_RING_MODULATOR_FREQUENCY: *val = (ALint)props->Modulator.Frequency; break; case AL_RING_MODULATOR_HIGHPASS_CUTOFF: *val = (ALint)props->Modulator.HighPassCutoff; break; case AL_RING_MODULATOR_WAVEFORM: *val = props->Modulator.Waveform; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid modulator integer property 0x%04x", param); } } void ALmodulator_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals) { ALmodulator_getParami(effect, context, param, vals); } void ALmodulator_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_RING_MODULATOR_FREQUENCY: *val = props->Modulator.Frequency; break; case AL_RING_MODULATOR_HIGHPASS_CUTOFF: *val = props->Modulator.HighPassCutoff; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid modulator float property 0x%04x", param); } } void ALmodulator_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { ALmodulator_getParamf(effect, context, param, vals); } DEFINE_ALEFFECT_VTABLE(ALmodulator); openal-soft-openal-soft-1.19.1/Alc/effects/null.c000066400000000000000000000133041335774445300215330ustar00rootroot00000000000000#include "config.h" #include #include "AL/al.h" #include "AL/alc.h" #include "alMain.h" #include "alAuxEffectSlot.h" #include "alError.h" typedef struct ALnullState { DERIVE_FROM_TYPE(ALeffectState); } ALnullState; /* Forward-declare "virtual" functions to define the vtable with. */ static ALvoid ALnullState_Destruct(ALnullState *state); static ALboolean ALnullState_deviceUpdate(ALnullState *state, ALCdevice *device); static ALvoid ALnullState_update(ALnullState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props); static ALvoid ALnullState_process(ALnullState *state, ALsizei samplesToDo, const ALfloat (*restrict samplesIn)[BUFFERSIZE], ALfloat (*restrict samplesOut)[BUFFERSIZE], ALsizei mumChannels); static void *ALnullState_New(size_t size); static void ALnullState_Delete(void *ptr); /* Define the ALeffectState vtable for this type. */ DEFINE_ALEFFECTSTATE_VTABLE(ALnullState); /* This constructs the effect state. It's called when the object is first * created. Make sure to call the parent Construct function first, and set the * vtable! */ static void ALnullState_Construct(ALnullState *state) { ALeffectState_Construct(STATIC_CAST(ALeffectState, state)); SET_VTABLE2(ALnullState, ALeffectState, state); } /* This destructs (not free!) the effect state. It's called only when the * effect slot is no longer used. Make sure to call the parent Destruct * function before returning! */ static ALvoid ALnullState_Destruct(ALnullState *state) { ALeffectState_Destruct(STATIC_CAST(ALeffectState,state)); } /* This updates the device-dependant effect state. This is called on * initialization and any time the device parameters (eg. playback frequency, * format) have been changed. */ static ALboolean ALnullState_deviceUpdate(ALnullState* UNUSED(state), ALCdevice* UNUSED(device)) { return AL_TRUE; } /* This updates the effect state. This is called any time the effect is * (re)loaded into a slot. */ static ALvoid ALnullState_update(ALnullState* UNUSED(state), const ALCcontext* UNUSED(context), const ALeffectslot* UNUSED(slot), const ALeffectProps* UNUSED(props)) { } /* This processes the effect state, for the given number of samples from the * input to the output buffer. The result should be added to the output buffer, * not replace it. */ static ALvoid ALnullState_process(ALnullState* UNUSED(state), ALsizei UNUSED(samplesToDo), const ALfloatBUFFERSIZE*restrict UNUSED(samplesIn), ALfloatBUFFERSIZE*restrict UNUSED(samplesOut), ALsizei UNUSED(numChannels)) { } /* This allocates memory to store the object, before it gets constructed. * DECLARE_DEFAULT_ALLOCATORS can be used to declare a default method. */ static void *ALnullState_New(size_t size) { return al_malloc(16, size); } /* This frees the memory used by the object, after it has been destructed. * DECLARE_DEFAULT_ALLOCATORS can be used to declare a default method. */ static void ALnullState_Delete(void *ptr) { al_free(ptr); } typedef struct NullStateFactory { DERIVE_FROM_TYPE(EffectStateFactory); } NullStateFactory; /* Creates ALeffectState objects of the appropriate type. */ ALeffectState *NullStateFactory_create(NullStateFactory *UNUSED(factory)) { ALnullState *state; NEW_OBJ0(state, ALnullState)(); if(!state) return NULL; return STATIC_CAST(ALeffectState, state); } /* Define the EffectStateFactory vtable for this type. */ DEFINE_EFFECTSTATEFACTORY_VTABLE(NullStateFactory); EffectStateFactory *NullStateFactory_getFactory(void) { static NullStateFactory NullFactory = { { GET_VTABLE2(NullStateFactory, EffectStateFactory) } }; return STATIC_CAST(EffectStateFactory, &NullFactory); } void ALnull_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val)) { switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid null effect integer property 0x%04x", param); } } void ALnull_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint* UNUSED(vals)) { switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid null effect integer-vector property 0x%04x", param); } } void ALnull_setParamf(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat UNUSED(val)) { switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid null effect float property 0x%04x", param); } } void ALnull_setParamfv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALfloat* UNUSED(vals)) { switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid null effect float-vector property 0x%04x", param); } } void ALnull_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint* UNUSED(val)) { switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid null effect integer property 0x%04x", param); } } void ALnull_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint* UNUSED(vals)) { switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid null effect integer-vector property 0x%04x", param); } } void ALnull_getParamf(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat* UNUSED(val)) { switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid null effect float property 0x%04x", param); } } void ALnull_getParamfv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat* UNUSED(vals)) { switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid null effect float-vector property 0x%04x", param); } } DEFINE_ALEFFECT_VTABLE(ALnull); openal-soft-openal-soft-1.19.1/Alc/effects/pshifter.c000066400000000000000000000354261335774445300224160ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2018 by Raul Herraiz. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include "alMain.h" #include "alAuxEffectSlot.h" #include "alError.h" #include "alu.h" #include "filters/defs.h" #include "alcomplex.h" #define STFT_SIZE 1024 #define STFT_HALF_SIZE (STFT_SIZE>>1) #define OVERSAMP (1<<2) #define STFT_STEP (STFT_SIZE / OVERSAMP) #define FIFO_LATENCY (STFT_STEP * (OVERSAMP-1)) typedef struct ALphasor { ALdouble Amplitude; ALdouble Phase; } ALphasor; typedef struct ALFrequencyDomain { ALdouble Amplitude; ALdouble Frequency; } ALfrequencyDomain; typedef struct ALpshifterState { DERIVE_FROM_TYPE(ALeffectState); /* Effect parameters */ ALsizei count; ALsizei PitchShiftI; ALfloat PitchShift; ALfloat FreqPerBin; /*Effects buffers*/ ALfloat InFIFO[STFT_SIZE]; ALfloat OutFIFO[STFT_STEP]; ALdouble LastPhase[STFT_HALF_SIZE+1]; ALdouble SumPhase[STFT_HALF_SIZE+1]; ALdouble OutputAccum[STFT_SIZE]; ALcomplex FFTbuffer[STFT_SIZE]; ALfrequencyDomain Analysis_buffer[STFT_HALF_SIZE+1]; ALfrequencyDomain Syntesis_buffer[STFT_HALF_SIZE+1]; alignas(16) ALfloat BufferOut[BUFFERSIZE]; /* Effect gains for each output channel */ ALfloat CurrentGains[MAX_OUTPUT_CHANNELS]; ALfloat TargetGains[MAX_OUTPUT_CHANNELS]; } ALpshifterState; static ALvoid ALpshifterState_Destruct(ALpshifterState *state); static ALboolean ALpshifterState_deviceUpdate(ALpshifterState *state, ALCdevice *device); static ALvoid ALpshifterState_update(ALpshifterState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props); static ALvoid ALpshifterState_process(ALpshifterState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels); DECLARE_DEFAULT_ALLOCATORS(ALpshifterState) DEFINE_ALEFFECTSTATE_VTABLE(ALpshifterState); /* Define a Hann window, used to filter the STFT input and output. */ alignas(16) static ALdouble HannWindow[STFT_SIZE]; static void InitHannWindow(void) { ALsizei i; /* Create lookup table of the Hann window for the desired size, i.e. STFT_SIZE */ for(i = 0;i < STFT_SIZE>>1;i++) { ALdouble val = sin(M_PI * (ALdouble)i / (ALdouble)(STFT_SIZE-1)); HannWindow[i] = HannWindow[STFT_SIZE-1-i] = val * val; } } static alonce_flag HannInitOnce = AL_ONCE_FLAG_INIT; static inline ALint double2int(ALdouble d) { #if ((defined(__GNUC__) || defined(__clang__)) && (defined(__i386__) || defined(__x86_64__)) && \ !defined(__SSE2_MATH__)) || (defined(_MSC_VER) && defined(_M_IX86_FP) && _M_IX86_FP < 2) ALint sign, shift; ALint64 mant; union { ALdouble d; ALint64 i64; } conv; conv.d = d; sign = (conv.i64>>63) | 1; shift = ((conv.i64>>52)&0x7ff) - (1023+52); /* Over/underflow */ if(UNLIKELY(shift >= 63 || shift < -52)) return 0; mant = (conv.i64&I64(0xfffffffffffff)) | I64(0x10000000000000); if(LIKELY(shift < 0)) return (ALint)(mant >> -shift) * sign; return (ALint)(mant << shift) * sign; #else return (ALint)d; #endif } /* Converts ALcomplex to ALphasor */ static inline ALphasor rect2polar(ALcomplex number) { ALphasor polar; polar.Amplitude = sqrt(number.Real*number.Real + number.Imag*number.Imag); polar.Phase = atan2(number.Imag, number.Real); return polar; } /* Converts ALphasor to ALcomplex */ static inline ALcomplex polar2rect(ALphasor number) { ALcomplex cartesian; cartesian.Real = number.Amplitude * cos(number.Phase); cartesian.Imag = number.Amplitude * sin(number.Phase); return cartesian; } static void ALpshifterState_Construct(ALpshifterState *state) { ALeffectState_Construct(STATIC_CAST(ALeffectState, state)); SET_VTABLE2(ALpshifterState, ALeffectState, state); alcall_once(&HannInitOnce, InitHannWindow); } static ALvoid ALpshifterState_Destruct(ALpshifterState *state) { ALeffectState_Destruct(STATIC_CAST(ALeffectState,state)); } static ALboolean ALpshifterState_deviceUpdate(ALpshifterState *state, ALCdevice *device) { /* (Re-)initializing parameters and clear the buffers. */ state->count = FIFO_LATENCY; state->PitchShiftI = FRACTIONONE; state->PitchShift = 1.0f; state->FreqPerBin = device->Frequency / (ALfloat)STFT_SIZE; memset(state->InFIFO, 0, sizeof(state->InFIFO)); memset(state->OutFIFO, 0, sizeof(state->OutFIFO)); memset(state->FFTbuffer, 0, sizeof(state->FFTbuffer)); memset(state->LastPhase, 0, sizeof(state->LastPhase)); memset(state->SumPhase, 0, sizeof(state->SumPhase)); memset(state->OutputAccum, 0, sizeof(state->OutputAccum)); memset(state->Analysis_buffer, 0, sizeof(state->Analysis_buffer)); memset(state->Syntesis_buffer, 0, sizeof(state->Syntesis_buffer)); memset(state->CurrentGains, 0, sizeof(state->CurrentGains)); memset(state->TargetGains, 0, sizeof(state->TargetGains)); return AL_TRUE; } static ALvoid ALpshifterState_update(ALpshifterState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props) { const ALCdevice *device = context->Device; ALfloat coeffs[MAX_AMBI_COEFFS]; float pitch; pitch = powf(2.0f, (ALfloat)(props->Pshifter.CoarseTune*100 + props->Pshifter.FineTune) / 1200.0f ); state->PitchShiftI = fastf2i(pitch*FRACTIONONE); state->PitchShift = state->PitchShiftI * (1.0f/FRACTIONONE); CalcAngleCoeffs(0.0f, 0.0f, 0.0f, coeffs); ComputePanGains(&device->Dry, coeffs, slot->Params.Gain, state->TargetGains); } static ALvoid ALpshifterState_process(ALpshifterState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels) { /* Pitch shifter engine based on the work of Stephan Bernsee. * http://blogs.zynaptiq.com/bernsee/pitch-shifting-using-the-ft/ */ static const ALdouble expected = M_PI*2.0 / OVERSAMP; const ALdouble freq_per_bin = state->FreqPerBin; ALfloat *restrict bufferOut = state->BufferOut; ALsizei count = state->count; ALsizei i, j, k; for(i = 0;i < SamplesToDo;) { do { /* Fill FIFO buffer with samples data */ state->InFIFO[count] = SamplesIn[0][i]; bufferOut[i] = state->OutFIFO[count - FIFO_LATENCY]; count++; } while(++i < SamplesToDo && count < STFT_SIZE); /* Check whether FIFO buffer is filled */ if(count < STFT_SIZE) break; count = FIFO_LATENCY; /* Real signal windowing and store in FFTbuffer */ for(k = 0;k < STFT_SIZE;k++) { state->FFTbuffer[k].Real = state->InFIFO[k] * HannWindow[k]; state->FFTbuffer[k].Imag = 0.0; } /* ANALYSIS */ /* Apply FFT to FFTbuffer data */ complex_fft(state->FFTbuffer, STFT_SIZE, -1.0); /* Analyze the obtained data. Since the real FFT is symmetric, only * STFT_HALF_SIZE+1 samples are needed. */ for(k = 0;k < STFT_HALF_SIZE+1;k++) { ALphasor component; ALdouble tmp; ALint qpd; /* Compute amplitude and phase */ component = rect2polar(state->FFTbuffer[k]); /* Compute phase difference and subtract expected phase difference */ tmp = (component.Phase - state->LastPhase[k]) - k*expected; /* Map delta phase into +/- Pi interval */ qpd = double2int(tmp / M_PI); tmp -= M_PI * (qpd + (qpd%2)); /* Get deviation from bin frequency from the +/- Pi interval */ tmp /= expected; /* Compute the k-th partials' true frequency, twice the amplitude * for maintain the gain (because half of bins are used) and store * amplitude and true frequency in analysis buffer. */ state->Analysis_buffer[k].Amplitude = 2.0 * component.Amplitude; state->Analysis_buffer[k].Frequency = (k + tmp) * freq_per_bin; /* Store actual phase[k] for the calculations in the next frame*/ state->LastPhase[k] = component.Phase; } /* PROCESSING */ /* pitch shifting */ for(k = 0;k < STFT_HALF_SIZE+1;k++) { state->Syntesis_buffer[k].Amplitude = 0.0; state->Syntesis_buffer[k].Frequency = 0.0; } for(k = 0;k < STFT_HALF_SIZE+1;k++) { j = (k*state->PitchShiftI) >> FRACTIONBITS; if(j >= STFT_HALF_SIZE+1) break; state->Syntesis_buffer[j].Amplitude += state->Analysis_buffer[k].Amplitude; state->Syntesis_buffer[j].Frequency = state->Analysis_buffer[k].Frequency * state->PitchShift; } /* SYNTHESIS */ /* Synthesis the processing data */ for(k = 0;k < STFT_HALF_SIZE+1;k++) { ALphasor component; ALdouble tmp; /* Compute bin deviation from scaled freq */ tmp = state->Syntesis_buffer[k].Frequency/freq_per_bin - k; /* Calculate actual delta phase and accumulate it to get bin phase */ state->SumPhase[k] += (k + tmp) * expected; component.Amplitude = state->Syntesis_buffer[k].Amplitude; component.Phase = state->SumPhase[k]; /* Compute phasor component to cartesian complex number and storage it into FFTbuffer*/ state->FFTbuffer[k] = polar2rect(component); } /* zero negative frequencies for recontruct a real signal */ for(k = STFT_HALF_SIZE+1;k < STFT_SIZE;k++) { state->FFTbuffer[k].Real = 0.0; state->FFTbuffer[k].Imag = 0.0; } /* Apply iFFT to buffer data */ complex_fft(state->FFTbuffer, STFT_SIZE, 1.0); /* Windowing and add to output */ for(k = 0;k < STFT_SIZE;k++) state->OutputAccum[k] += HannWindow[k] * state->FFTbuffer[k].Real / (0.5 * STFT_HALF_SIZE * OVERSAMP); /* Shift accumulator, input & output FIFO */ for(k = 0;k < STFT_STEP;k++) state->OutFIFO[k] = (ALfloat)state->OutputAccum[k]; for(j = 0;k < STFT_SIZE;k++,j++) state->OutputAccum[j] = state->OutputAccum[k]; for(;j < STFT_SIZE;j++) state->OutputAccum[j] = 0.0; for(k = 0;k < FIFO_LATENCY;k++) state->InFIFO[k] = state->InFIFO[k+STFT_STEP]; } state->count = count; /* Now, mix the processed sound data to the output. */ MixSamples(bufferOut, NumChannels, SamplesOut, state->CurrentGains, state->TargetGains, maxi(SamplesToDo, 512), 0, SamplesToDo); } typedef struct PshifterStateFactory { DERIVE_FROM_TYPE(EffectStateFactory); } PshifterStateFactory; static ALeffectState *PshifterStateFactory_create(PshifterStateFactory *UNUSED(factory)) { ALpshifterState *state; NEW_OBJ0(state, ALpshifterState)(); if(!state) return NULL; return STATIC_CAST(ALeffectState, state); } DEFINE_EFFECTSTATEFACTORY_VTABLE(PshifterStateFactory); EffectStateFactory *PshifterStateFactory_getFactory(void) { static PshifterStateFactory PshifterFactory = { { GET_VTABLE2(PshifterStateFactory, EffectStateFactory) } }; return STATIC_CAST(EffectStateFactory, &PshifterFactory); } void ALpshifter_setParamf(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat UNUSED(val)) { alSetError( context, AL_INVALID_ENUM, "Invalid pitch shifter float property 0x%04x", param ); } void ALpshifter_setParamfv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALfloat *UNUSED(vals)) { alSetError( context, AL_INVALID_ENUM, "Invalid pitch shifter float-vector property 0x%04x", param ); } void ALpshifter_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_PITCH_SHIFTER_COARSE_TUNE: if(!(val >= AL_PITCH_SHIFTER_MIN_COARSE_TUNE && val <= AL_PITCH_SHIFTER_MAX_COARSE_TUNE)) SETERR_RETURN(context, AL_INVALID_VALUE,,"Pitch shifter coarse tune out of range"); props->Pshifter.CoarseTune = val; break; case AL_PITCH_SHIFTER_FINE_TUNE: if(!(val >= AL_PITCH_SHIFTER_MIN_FINE_TUNE && val <= AL_PITCH_SHIFTER_MAX_FINE_TUNE)) SETERR_RETURN(context, AL_INVALID_VALUE,,"Pitch shifter fine tune out of range"); props->Pshifter.FineTune = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter integer property 0x%04x", param); } } void ALpshifter_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals) { ALpshifter_setParami(effect, context, param, vals[0]); } void ALpshifter_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_PITCH_SHIFTER_COARSE_TUNE: *val = (ALint)props->Pshifter.CoarseTune; break; case AL_PITCH_SHIFTER_FINE_TUNE: *val = (ALint)props->Pshifter.FineTune; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter integer property 0x%04x", param); } } void ALpshifter_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals) { ALpshifter_getParami(effect, context, param, vals); } void ALpshifter_getParamf(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat *UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter float property 0x%04x", param); } void ALpshifter_getParamfv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter float vector-property 0x%04x", param); } DEFINE_ALEFFECT_VTABLE(ALpshifter); openal-soft-openal-soft-1.19.1/Alc/effects/reverb.c000066400000000000000000002343121335774445300220520ustar00rootroot00000000000000/** * Ambisonic reverb engine for the OpenAL cross platform audio library * Copyright (C) 2008-2017 by Chris Robinson and Christopher Fitzgerald. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include "alMain.h" #include "alu.h" #include "alAuxEffectSlot.h" #include "alListener.h" #include "alError.h" #include "filters/defs.h" /* This is a user config option for modifying the overall output of the reverb * effect. */ ALfloat ReverbBoost = 1.0f; /* This is the maximum number of samples processed for each inner loop * iteration. */ #define MAX_UPDATE_SAMPLES 256 /* The number of samples used for cross-faded delay lines. This can be used * to balance the compensation for abrupt line changes and attenuation due to * minimally lengthed recursive lines. Try to keep this below the device * update size. */ #define FADE_SAMPLES 128 /* The number of spatialized lines or channels to process. Four channels allows * for a 3D A-Format response. NOTE: This can't be changed without taking care * of the conversion matrices, and a few places where the length arrays are * assumed to have 4 elements. */ #define NUM_LINES 4 /* The B-Format to A-Format conversion matrix. The arrangement of rows is * deliberately chosen to align the resulting lines to their spatial opposites * (0:above front left <-> 3:above back right, 1:below front right <-> 2:below * back left). It's not quite opposite, since the A-Format results in a * tetrahedron, but it's close enough. Should the model be extended to 8-lines * in the future, true opposites can be used. */ static const aluMatrixf B2A = {{ { 0.288675134595f, 0.288675134595f, 0.288675134595f, 0.288675134595f }, { 0.288675134595f, -0.288675134595f, -0.288675134595f, 0.288675134595f }, { 0.288675134595f, 0.288675134595f, -0.288675134595f, -0.288675134595f }, { 0.288675134595f, -0.288675134595f, 0.288675134595f, -0.288675134595f } }}; /* Converts A-Format to B-Format. */ static const aluMatrixf A2B = {{ { 0.866025403785f, 0.866025403785f, 0.866025403785f, 0.866025403785f }, { 0.866025403785f, -0.866025403785f, 0.866025403785f, -0.866025403785f }, { 0.866025403785f, -0.866025403785f, -0.866025403785f, 0.866025403785f }, { 0.866025403785f, 0.866025403785f, -0.866025403785f, -0.866025403785f } }}; static const ALfloat FadeStep = 1.0f / FADE_SAMPLES; /* The all-pass and delay lines have a variable length dependent on the * effect's density parameter, which helps alter the perceived environment * size. The size-to-density conversion is a cubed scale: * * density = min(1.0, pow(size, 3.0) / DENSITY_SCALE); * * The line lengths scale linearly with room size, so the inverse density * conversion is needed, taking the cube root of the re-scaled density to * calculate the line length multiplier: * * length_mult = max(5.0, cbrtf(density*DENSITY_SCALE)); * * The density scale below will result in a max line multiplier of 50, for an * effective size range of 5m to 50m. */ static const ALfloat DENSITY_SCALE = 125000.0f; /* All delay line lengths are specified in seconds. * * To approximate early reflections, we break them up into primary (those * arriving from the same direction as the source) and secondary (those * arriving from the opposite direction). * * The early taps decorrelate the 4-channel signal to approximate an average * room response for the primary reflections after the initial early delay. * * Given an average room dimension (d_a) and the speed of sound (c) we can * calculate the average reflection delay (r_a) regardless of listener and * source positions as: * * r_a = d_a / c * c = 343.3 * * This can extended to finding the average difference (r_d) between the * maximum (r_1) and minimum (r_0) reflection delays: * * r_0 = 2 / 3 r_a * = r_a - r_d / 2 * = r_d * r_1 = 4 / 3 r_a * = r_a + r_d / 2 * = 2 r_d * r_d = 2 / 3 r_a * = r_1 - r_0 * * As can be determined by integrating the 1D model with a source (s) and * listener (l) positioned across the dimension of length (d_a): * * r_d = int_(l=0)^d_a (int_(s=0)^d_a |2 d_a - 2 (l + s)| ds) dl / c * * The initial taps (T_(i=0)^N) are then specified by taking a power series * that ranges between r_0 and half of r_1 less r_0: * * R_i = 2^(i / (2 N - 1)) r_d * = r_0 + (2^(i / (2 N - 1)) - 1) r_d * = r_0 + T_i * T_i = R_i - r_0 * = (2^(i / (2 N - 1)) - 1) r_d * * Assuming an average of 1m, we get the following taps: */ static const ALfloat EARLY_TAP_LENGTHS[NUM_LINES] = { 0.0000000e+0f, 2.0213520e-4f, 4.2531060e-4f, 6.7171600e-4f }; /* The early all-pass filter lengths are based on the early tap lengths: * * A_i = R_i / a * * Where a is the approximate maximum all-pass cycle limit (20). */ static const ALfloat EARLY_ALLPASS_LENGTHS[NUM_LINES] = { 9.7096800e-5f, 1.0720356e-4f, 1.1836234e-4f, 1.3068260e-4f }; /* The early delay lines are used to transform the primary reflections into * the secondary reflections. The A-format is arranged in such a way that * the channels/lines are spatially opposite: * * C_i is opposite C_(N-i-1) * * The delays of the two opposing reflections (R_i and O_i) from a source * anywhere along a particular dimension always sum to twice its full delay: * * 2 r_a = R_i + O_i * * With that in mind we can determine the delay between the two reflections * and thus specify our early line lengths (L_(i=0)^N) using: * * O_i = 2 r_a - R_(N-i-1) * L_i = O_i - R_(N-i-1) * = 2 (r_a - R_(N-i-1)) * = 2 (r_a - T_(N-i-1) - r_0) * = 2 r_a (1 - (2 / 3) 2^((N - i - 1) / (2 N - 1))) * * Using an average dimension of 1m, we get: */ static const ALfloat EARLY_LINE_LENGTHS[NUM_LINES] = { 5.9850400e-4f, 1.0913150e-3f, 1.5376658e-3f, 1.9419362e-3f }; /* The late all-pass filter lengths are based on the late line lengths: * * A_i = (5 / 3) L_i / r_1 */ static const ALfloat LATE_ALLPASS_LENGTHS[NUM_LINES] = { 1.6182800e-4f, 2.0389060e-4f, 2.8159360e-4f, 3.2365600e-4f }; /* The late lines are used to approximate the decaying cycle of recursive * late reflections. * * Splitting the lines in half, we start with the shortest reflection paths * (L_(i=0)^(N/2)): * * L_i = 2^(i / (N - 1)) r_d * * Then for the opposite (longest) reflection paths (L_(i=N/2)^N): * * L_i = 2 r_a - L_(i-N/2) * = 2 r_a - 2^((i - N / 2) / (N - 1)) r_d * * For our 1m average room, we get: */ static const ALfloat LATE_LINE_LENGTHS[NUM_LINES] = { 1.9419362e-3f, 2.4466860e-3f, 3.3791220e-3f, 3.8838720e-3f }; typedef struct DelayLineI { /* The delay lines use interleaved samples, with the lengths being powers * of 2 to allow the use of bit-masking instead of a modulus for wrapping. */ ALsizei Mask; ALfloat (*Line)[NUM_LINES]; } DelayLineI; typedef struct VecAllpass { DelayLineI Delay; ALfloat Coeff; ALsizei Offset[NUM_LINES][2]; } VecAllpass; typedef struct T60Filter { /* Two filters are used to adjust the signal. One to control the low * frequencies, and one to control the high frequencies. */ ALfloat MidGain[2]; BiquadFilter HFFilter, LFFilter; } T60Filter; typedef struct EarlyReflections { /* A Gerzon vector all-pass filter is used to simulate initial diffusion. * The spread from this filter also helps smooth out the reverb tail. */ VecAllpass VecAp; /* An echo line is used to complete the second half of the early * reflections. */ DelayLineI Delay; ALsizei Offset[NUM_LINES][2]; ALfloat Coeff[NUM_LINES][2]; /* The gain for each output channel based on 3D panning. */ ALfloat CurrentGain[NUM_LINES][MAX_OUTPUT_CHANNELS]; ALfloat PanGain[NUM_LINES][MAX_OUTPUT_CHANNELS]; } EarlyReflections; typedef struct LateReverb { /* A recursive delay line is used fill in the reverb tail. */ DelayLineI Delay; ALsizei Offset[NUM_LINES][2]; /* Attenuation to compensate for the modal density and decay rate of the * late lines. */ ALfloat DensityGain[2]; /* T60 decay filters are used to simulate absorption. */ T60Filter T60[NUM_LINES]; /* A Gerzon vector all-pass filter is used to simulate diffusion. */ VecAllpass VecAp; /* The gain for each output channel based on 3D panning. */ ALfloat CurrentGain[NUM_LINES][MAX_OUTPUT_CHANNELS]; ALfloat PanGain[NUM_LINES][MAX_OUTPUT_CHANNELS]; } LateReverb; typedef struct ReverbState { DERIVE_FROM_TYPE(ALeffectState); /* All delay lines are allocated as a single buffer to reduce memory * fragmentation and management code. */ ALfloat *SampleBuffer; ALuint TotalSamples; struct { /* Calculated parameters which indicate if cross-fading is needed after * an update. */ ALfloat Density, Diffusion; ALfloat DecayTime, HFDecayTime, LFDecayTime; ALfloat HFReference, LFReference; } Params; /* Master effect filters */ struct { BiquadFilter Lp; BiquadFilter Hp; } Filter[NUM_LINES]; /* Core delay line (early reflections and late reverb tap from this). */ DelayLineI Delay; /* Tap points for early reflection delay. */ ALsizei EarlyDelayTap[NUM_LINES][2]; ALfloat EarlyDelayCoeff[NUM_LINES][2]; /* Tap points for late reverb feed and delay. */ ALsizei LateFeedTap; ALsizei LateDelayTap[NUM_LINES][2]; /* Coefficients for the all-pass and line scattering matrices. */ ALfloat MixX; ALfloat MixY; EarlyReflections Early; LateReverb Late; /* Indicates the cross-fade point for delay line reads [0,FADE_SAMPLES]. */ ALsizei FadeCount; /* Maximum number of samples to process at once. */ ALsizei MaxUpdate[2]; /* The current write offset for all delay lines. */ ALsizei Offset; /* Temporary storage used when processing. */ alignas(16) ALfloat TempSamples[NUM_LINES][MAX_UPDATE_SAMPLES]; alignas(16) ALfloat MixSamples[NUM_LINES][MAX_UPDATE_SAMPLES]; } ReverbState; static ALvoid ReverbState_Destruct(ReverbState *State); static ALboolean ReverbState_deviceUpdate(ReverbState *State, ALCdevice *Device); static ALvoid ReverbState_update(ReverbState *State, const ALCcontext *Context, const ALeffectslot *Slot, const ALeffectProps *props); static ALvoid ReverbState_process(ReverbState *State, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels); DECLARE_DEFAULT_ALLOCATORS(ReverbState) DEFINE_ALEFFECTSTATE_VTABLE(ReverbState); static void ReverbState_Construct(ReverbState *state) { ALsizei i, j; ALeffectState_Construct(STATIC_CAST(ALeffectState, state)); SET_VTABLE2(ReverbState, ALeffectState, state); state->TotalSamples = 0; state->SampleBuffer = NULL; state->Params.Density = AL_EAXREVERB_DEFAULT_DENSITY; state->Params.Diffusion = AL_EAXREVERB_DEFAULT_DIFFUSION; state->Params.DecayTime = AL_EAXREVERB_DEFAULT_DECAY_TIME; state->Params.HFDecayTime = AL_EAXREVERB_DEFAULT_DECAY_TIME*AL_EAXREVERB_DEFAULT_DECAY_HFRATIO; state->Params.LFDecayTime = AL_EAXREVERB_DEFAULT_DECAY_TIME*AL_EAXREVERB_DEFAULT_DECAY_LFRATIO; state->Params.HFReference = AL_EAXREVERB_DEFAULT_HFREFERENCE; state->Params.LFReference = AL_EAXREVERB_DEFAULT_LFREFERENCE; for(i = 0;i < NUM_LINES;i++) { BiquadFilter_clear(&state->Filter[i].Lp); BiquadFilter_clear(&state->Filter[i].Hp); } state->Delay.Mask = 0; state->Delay.Line = NULL; for(i = 0;i < NUM_LINES;i++) { state->EarlyDelayTap[i][0] = 0; state->EarlyDelayTap[i][1] = 0; state->EarlyDelayCoeff[i][0] = 0.0f; state->EarlyDelayCoeff[i][1] = 0.0f; } state->LateFeedTap = 0; for(i = 0;i < NUM_LINES;i++) { state->LateDelayTap[i][0] = 0; state->LateDelayTap[i][1] = 0; } state->MixX = 0.0f; state->MixY = 0.0f; state->Early.VecAp.Delay.Mask = 0; state->Early.VecAp.Delay.Line = NULL; state->Early.VecAp.Coeff = 0.0f; state->Early.Delay.Mask = 0; state->Early.Delay.Line = NULL; for(i = 0;i < NUM_LINES;i++) { state->Early.VecAp.Offset[i][0] = 0; state->Early.VecAp.Offset[i][1] = 0; state->Early.Offset[i][0] = 0; state->Early.Offset[i][1] = 0; state->Early.Coeff[i][0] = 0.0f; state->Early.Coeff[i][1] = 0.0f; } state->Late.DensityGain[0] = 0.0f; state->Late.DensityGain[1] = 0.0f; state->Late.Delay.Mask = 0; state->Late.Delay.Line = NULL; state->Late.VecAp.Delay.Mask = 0; state->Late.VecAp.Delay.Line = NULL; state->Late.VecAp.Coeff = 0.0f; for(i = 0;i < NUM_LINES;i++) { state->Late.Offset[i][0] = 0; state->Late.Offset[i][1] = 0; state->Late.VecAp.Offset[i][0] = 0; state->Late.VecAp.Offset[i][1] = 0; state->Late.T60[i].MidGain[0] = 0.0f; state->Late.T60[i].MidGain[1] = 0.0f; BiquadFilter_clear(&state->Late.T60[i].HFFilter); BiquadFilter_clear(&state->Late.T60[i].LFFilter); } for(i = 0;i < NUM_LINES;i++) { for(j = 0;j < MAX_OUTPUT_CHANNELS;j++) { state->Early.CurrentGain[i][j] = 0.0f; state->Early.PanGain[i][j] = 0.0f; state->Late.CurrentGain[i][j] = 0.0f; state->Late.PanGain[i][j] = 0.0f; } } state->FadeCount = 0; state->MaxUpdate[0] = MAX_UPDATE_SAMPLES; state->MaxUpdate[1] = MAX_UPDATE_SAMPLES; state->Offset = 0; } static ALvoid ReverbState_Destruct(ReverbState *State) { al_free(State->SampleBuffer); State->SampleBuffer = NULL; ALeffectState_Destruct(STATIC_CAST(ALeffectState,State)); } /************************************** * Device Update * **************************************/ static inline ALfloat CalcDelayLengthMult(ALfloat density) { return maxf(5.0f, cbrtf(density*DENSITY_SCALE)); } /* Given the allocated sample buffer, this function updates each delay line * offset. */ static inline ALvoid RealizeLineOffset(ALfloat *sampleBuffer, DelayLineI *Delay) { union { ALfloat *f; ALfloat (*f4)[NUM_LINES]; } u; u.f = &sampleBuffer[(ptrdiff_t)Delay->Line * NUM_LINES]; Delay->Line = u.f4; } /* Calculate the length of a delay line and store its mask and offset. */ static ALuint CalcLineLength(const ALfloat length, const ptrdiff_t offset, const ALuint frequency, const ALuint extra, DelayLineI *Delay) { ALuint samples; /* All line lengths are powers of 2, calculated from their lengths in * seconds, rounded up. */ samples = float2int(ceilf(length*frequency)); samples = NextPowerOf2(samples + extra); /* All lines share a single sample buffer. */ Delay->Mask = samples - 1; Delay->Line = (ALfloat(*)[NUM_LINES])offset; /* Return the sample count for accumulation. */ return samples; } /* Calculates the delay line metrics and allocates the shared sample buffer * for all lines given the sample rate (frequency). If an allocation failure * occurs, it returns AL_FALSE. */ static ALboolean AllocLines(const ALuint frequency, ReverbState *State) { ALuint totalSamples, i; ALfloat multiplier, length; /* All delay line lengths are calculated to accomodate the full range of * lengths given their respective paramters. */ totalSamples = 0; /* Multiplier for the maximum density value, i.e. density=1, which is * actually the least density... */ multiplier = CalcDelayLengthMult(AL_EAXREVERB_MAX_DENSITY); /* The main delay length includes the maximum early reflection delay, the * largest early tap width, the maximum late reverb delay, and the * largest late tap width. Finally, it must also be extended by the * update size (MAX_UPDATE_SAMPLES) for block processing. */ length = AL_EAXREVERB_MAX_REFLECTIONS_DELAY + EARLY_TAP_LENGTHS[NUM_LINES-1]*multiplier + AL_EAXREVERB_MAX_LATE_REVERB_DELAY + (LATE_LINE_LENGTHS[NUM_LINES-1] - LATE_LINE_LENGTHS[0])*0.25f*multiplier; totalSamples += CalcLineLength(length, totalSamples, frequency, MAX_UPDATE_SAMPLES, &State->Delay); /* The early vector all-pass line. */ length = EARLY_ALLPASS_LENGTHS[NUM_LINES-1] * multiplier; totalSamples += CalcLineLength(length, totalSamples, frequency, 0, &State->Early.VecAp.Delay); /* The early reflection line. */ length = EARLY_LINE_LENGTHS[NUM_LINES-1] * multiplier; totalSamples += CalcLineLength(length, totalSamples, frequency, 0, &State->Early.Delay); /* The late vector all-pass line. */ length = LATE_ALLPASS_LENGTHS[NUM_LINES-1] * multiplier; totalSamples += CalcLineLength(length, totalSamples, frequency, 0, &State->Late.VecAp.Delay); /* The late delay lines are calculated from the largest maximum density * line length. */ length = LATE_LINE_LENGTHS[NUM_LINES-1] * multiplier; totalSamples += CalcLineLength(length, totalSamples, frequency, 0, &State->Late.Delay); if(totalSamples != State->TotalSamples) { ALfloat *newBuffer; TRACE("New reverb buffer length: %ux4 samples\n", totalSamples); newBuffer = al_calloc(16, sizeof(ALfloat[NUM_LINES]) * totalSamples); if(!newBuffer) return AL_FALSE; al_free(State->SampleBuffer); State->SampleBuffer = newBuffer; State->TotalSamples = totalSamples; } /* Update all delays to reflect the new sample buffer. */ RealizeLineOffset(State->SampleBuffer, &State->Delay); RealizeLineOffset(State->SampleBuffer, &State->Early.VecAp.Delay); RealizeLineOffset(State->SampleBuffer, &State->Early.Delay); RealizeLineOffset(State->SampleBuffer, &State->Late.VecAp.Delay); RealizeLineOffset(State->SampleBuffer, &State->Late.Delay); /* Clear the sample buffer. */ for(i = 0;i < State->TotalSamples;i++) State->SampleBuffer[i] = 0.0f; return AL_TRUE; } static ALboolean ReverbState_deviceUpdate(ReverbState *State, ALCdevice *Device) { ALuint frequency = Device->Frequency; ALfloat multiplier; ALsizei i, j; /* Allocate the delay lines. */ if(!AllocLines(frequency, State)) return AL_FALSE; multiplier = CalcDelayLengthMult(AL_EAXREVERB_MAX_DENSITY); /* The late feed taps are set a fixed position past the latest delay tap. */ State->LateFeedTap = float2int((AL_EAXREVERB_MAX_REFLECTIONS_DELAY + EARLY_TAP_LENGTHS[NUM_LINES-1]*multiplier) * frequency); /* Clear filters and gain coefficients since the delay lines were all just * cleared (if not reallocated). */ for(i = 0;i < NUM_LINES;i++) { BiquadFilter_clear(&State->Filter[i].Lp); BiquadFilter_clear(&State->Filter[i].Hp); } for(i = 0;i < NUM_LINES;i++) { State->EarlyDelayCoeff[i][0] = 0.0f; State->EarlyDelayCoeff[i][1] = 0.0f; } for(i = 0;i < NUM_LINES;i++) { State->Early.Coeff[i][0] = 0.0f; State->Early.Coeff[i][1] = 0.0f; } State->Late.DensityGain[0] = 0.0f; State->Late.DensityGain[1] = 0.0f; for(i = 0;i < NUM_LINES;i++) { State->Late.T60[i].MidGain[0] = 0.0f; State->Late.T60[i].MidGain[1] = 0.0f; BiquadFilter_clear(&State->Late.T60[i].HFFilter); BiquadFilter_clear(&State->Late.T60[i].LFFilter); } for(i = 0;i < NUM_LINES;i++) { for(j = 0;j < MAX_OUTPUT_CHANNELS;j++) { State->Early.CurrentGain[i][j] = 0.0f; State->Early.PanGain[i][j] = 0.0f; State->Late.CurrentGain[i][j] = 0.0f; State->Late.PanGain[i][j] = 0.0f; } } /* Reset counters and offset base. */ State->FadeCount = 0; State->MaxUpdate[0] = MAX_UPDATE_SAMPLES; State->MaxUpdate[1] = MAX_UPDATE_SAMPLES; State->Offset = 0; return AL_TRUE; } /************************************** * Effect Update * **************************************/ /* Calculate a decay coefficient given the length of each cycle and the time * until the decay reaches -60 dB. */ static inline ALfloat CalcDecayCoeff(const ALfloat length, const ALfloat decayTime) { return powf(REVERB_DECAY_GAIN, length/decayTime); } /* Calculate a decay length from a coefficient and the time until the decay * reaches -60 dB. */ static inline ALfloat CalcDecayLength(const ALfloat coeff, const ALfloat decayTime) { return log10f(coeff) * decayTime / log10f(REVERB_DECAY_GAIN); } /* Calculate an attenuation to be applied to the input of any echo models to * compensate for modal density and decay time. */ static inline ALfloat CalcDensityGain(const ALfloat a) { /* The energy of a signal can be obtained by finding the area under the * squared signal. This takes the form of Sum(x_n^2), where x is the * amplitude for the sample n. * * Decaying feedback matches exponential decay of the form Sum(a^n), * where a is the attenuation coefficient, and n is the sample. The area * under this decay curve can be calculated as: 1 / (1 - a). * * Modifying the above equation to find the area under the squared curve * (for energy) yields: 1 / (1 - a^2). Input attenuation can then be * calculated by inverting the square root of this approximation, * yielding: 1 / sqrt(1 / (1 - a^2)), simplified to: sqrt(1 - a^2). */ return sqrtf(1.0f - a*a); } /* Calculate the scattering matrix coefficients given a diffusion factor. */ static inline ALvoid CalcMatrixCoeffs(const ALfloat diffusion, ALfloat *x, ALfloat *y) { ALfloat n, t; /* The matrix is of order 4, so n is sqrt(4 - 1). */ n = sqrtf(3.0f); t = diffusion * atanf(n); /* Calculate the first mixing matrix coefficient. */ *x = cosf(t); /* Calculate the second mixing matrix coefficient. */ *y = sinf(t) / n; } /* Calculate the limited HF ratio for use with the late reverb low-pass * filters. */ static ALfloat CalcLimitedHfRatio(const ALfloat hfRatio, const ALfloat airAbsorptionGainHF, const ALfloat decayTime, const ALfloat SpeedOfSound) { ALfloat limitRatio; /* Find the attenuation due to air absorption in dB (converting delay * time to meters using the speed of sound). Then reversing the decay * equation, solve for HF ratio. The delay length is cancelled out of * the equation, so it can be calculated once for all lines. */ limitRatio = 1.0f / (CalcDecayLength(airAbsorptionGainHF, decayTime) * SpeedOfSound); /* Using the limit calculated above, apply the upper bound to the HF ratio. */ return minf(limitRatio, hfRatio); } /* Calculates the 3-band T60 damping coefficients for a particular delay line * of specified length, using a combination of two shelf filter sections given * decay times for each band split at two reference frequencies. */ static void CalcT60DampingCoeffs(const ALfloat length, const ALfloat lfDecayTime, const ALfloat mfDecayTime, const ALfloat hfDecayTime, const ALfloat lf0norm, const ALfloat hf0norm, T60Filter *filter) { ALfloat lfGain = CalcDecayCoeff(length, lfDecayTime); ALfloat mfGain = CalcDecayCoeff(length, mfDecayTime); ALfloat hfGain = CalcDecayCoeff(length, hfDecayTime); filter->MidGain[1] = mfGain; BiquadFilter_setParams(&filter->LFFilter, BiquadType_LowShelf, lfGain/mfGain, lf0norm, calc_rcpQ_from_slope(lfGain/mfGain, 1.0f)); BiquadFilter_setParams(&filter->HFFilter, BiquadType_HighShelf, hfGain/mfGain, hf0norm, calc_rcpQ_from_slope(hfGain/mfGain, 1.0f)); } /* Update the offsets for the main effect delay line. */ static ALvoid UpdateDelayLine(const ALfloat earlyDelay, const ALfloat lateDelay, const ALfloat density, const ALfloat decayTime, const ALuint frequency, ReverbState *State) { ALfloat multiplier, length; ALuint i; multiplier = CalcDelayLengthMult(density); /* Early reflection taps are decorrelated by means of an average room * reflection approximation described above the definition of the taps. * This approximation is linear and so the above density multiplier can * be applied to adjust the width of the taps. A single-band decay * coefficient is applied to simulate initial attenuation and absorption. * * Late reverb taps are based on the late line lengths to allow a zero- * delay path and offsets that would continue the propagation naturally * into the late lines. */ for(i = 0;i < NUM_LINES;i++) { length = earlyDelay + EARLY_TAP_LENGTHS[i]*multiplier; State->EarlyDelayTap[i][1] = float2int(length * frequency); length = EARLY_TAP_LENGTHS[i]*multiplier; State->EarlyDelayCoeff[i][1] = CalcDecayCoeff(length, decayTime); length = lateDelay + (LATE_LINE_LENGTHS[i] - LATE_LINE_LENGTHS[0])*0.25f*multiplier; State->LateDelayTap[i][1] = State->LateFeedTap + float2int(length * frequency); } } /* Update the early reflection line lengths and gain coefficients. */ static ALvoid UpdateEarlyLines(const ALfloat density, const ALfloat diffusion, const ALfloat decayTime, const ALuint frequency, EarlyReflections *Early) { ALfloat multiplier, length; ALsizei i; multiplier = CalcDelayLengthMult(density); /* Calculate the all-pass feed-back/forward coefficient. */ Early->VecAp.Coeff = sqrtf(0.5f) * powf(diffusion, 2.0f); for(i = 0;i < NUM_LINES;i++) { /* Calculate the length (in seconds) of each all-pass line. */ length = EARLY_ALLPASS_LENGTHS[i] * multiplier; /* Calculate the delay offset for each all-pass line. */ Early->VecAp.Offset[i][1] = float2int(length * frequency); /* Calculate the length (in seconds) of each delay line. */ length = EARLY_LINE_LENGTHS[i] * multiplier; /* Calculate the delay offset for each delay line. */ Early->Offset[i][1] = float2int(length * frequency); /* Calculate the gain (coefficient) for each line. */ Early->Coeff[i][1] = CalcDecayCoeff(length, decayTime); } } /* Update the late reverb line lengths and T60 coefficients. */ static ALvoid UpdateLateLines(const ALfloat density, const ALfloat diffusion, const ALfloat lfDecayTime, const ALfloat mfDecayTime, const ALfloat hfDecayTime, const ALfloat lf0norm, const ALfloat hf0norm, const ALuint frequency, LateReverb *Late) { /* Scaling factor to convert the normalized reference frequencies from * representing 0...freq to 0...max_reference. */ const ALfloat norm_weight_factor = (ALfloat)frequency / AL_EAXREVERB_MAX_HFREFERENCE; ALfloat multiplier, length, bandWeights[3]; ALsizei i; /* To compensate for changes in modal density and decay time of the late * reverb signal, the input is attenuated based on the maximal energy of * the outgoing signal. This approximation is used to keep the apparent * energy of the signal equal for all ranges of density and decay time. * * The average length of the delay lines is used to calculate the * attenuation coefficient. */ multiplier = CalcDelayLengthMult(density); length = (LATE_LINE_LENGTHS[0] + LATE_LINE_LENGTHS[1] + LATE_LINE_LENGTHS[2] + LATE_LINE_LENGTHS[3]) / 4.0f * multiplier; length += (LATE_ALLPASS_LENGTHS[0] + LATE_ALLPASS_LENGTHS[1] + LATE_ALLPASS_LENGTHS[2] + LATE_ALLPASS_LENGTHS[3]) / 4.0f * multiplier; /* The density gain calculation uses an average decay time weighted by * approximate bandwidth. This attempts to compensate for losses of energy * that reduce decay time due to scattering into highly attenuated bands. */ bandWeights[0] = lf0norm*norm_weight_factor; bandWeights[1] = hf0norm*norm_weight_factor - lf0norm*norm_weight_factor; bandWeights[2] = 1.0f - hf0norm*norm_weight_factor; Late->DensityGain[1] = CalcDensityGain( CalcDecayCoeff(length, bandWeights[0]*lfDecayTime + bandWeights[1]*mfDecayTime + bandWeights[2]*hfDecayTime ) ); /* Calculate the all-pass feed-back/forward coefficient. */ Late->VecAp.Coeff = sqrtf(0.5f) * powf(diffusion, 2.0f); for(i = 0;i < NUM_LINES;i++) { /* Calculate the length (in seconds) of each all-pass line. */ length = LATE_ALLPASS_LENGTHS[i] * multiplier; /* Calculate the delay offset for each all-pass line. */ Late->VecAp.Offset[i][1] = float2int(length * frequency); /* Calculate the length (in seconds) of each delay line. */ length = LATE_LINE_LENGTHS[i] * multiplier; /* Calculate the delay offset for each delay line. */ Late->Offset[i][1] = float2int(length*frequency + 0.5f); /* Approximate the absorption that the vector all-pass would exhibit * given the current diffusion so we don't have to process a full T60 * filter for each of its four lines. */ length += lerp(LATE_ALLPASS_LENGTHS[i], (LATE_ALLPASS_LENGTHS[0] + LATE_ALLPASS_LENGTHS[1] + LATE_ALLPASS_LENGTHS[2] + LATE_ALLPASS_LENGTHS[3]) / 4.0f, diffusion) * multiplier; /* Calculate the T60 damping coefficients for each line. */ CalcT60DampingCoeffs(length, lfDecayTime, mfDecayTime, hfDecayTime, lf0norm, hf0norm, &Late->T60[i]); } } /* Creates a transform matrix given a reverb vector. The vector pans the reverb * reflections toward the given direction, using its magnitude (up to 1) as a * focal strength. This function results in a B-Format transformation matrix * that spatially focuses the signal in the desired direction. */ static aluMatrixf GetTransformFromVector(const ALfloat *vec) { aluMatrixf focus; ALfloat norm[3]; ALfloat mag; /* Normalize the panning vector according to the N3D scale, which has an * extra sqrt(3) term on the directional components. Converting from OpenAL * to B-Format also requires negating X (ACN 1) and Z (ACN 3). Note however * that the reverb panning vectors use left-handed coordinates, unlike the * rest of OpenAL which use right-handed. This is fixed by negating Z, * which cancels out with the B-Format Z negation. */ mag = sqrtf(vec[0]*vec[0] + vec[1]*vec[1] + vec[2]*vec[2]); if(mag > 1.0f) { norm[0] = vec[0] / mag * -SQRTF_3; norm[1] = vec[1] / mag * SQRTF_3; norm[2] = vec[2] / mag * SQRTF_3; mag = 1.0f; } else { /* If the magnitude is less than or equal to 1, just apply the sqrt(3) * term. There's no need to renormalize the magnitude since it would * just be reapplied in the matrix. */ norm[0] = vec[0] * -SQRTF_3; norm[1] = vec[1] * SQRTF_3; norm[2] = vec[2] * SQRTF_3; } aluMatrixfSet(&focus, 1.0f, 0.0f, 0.0f, 0.0f, norm[0], 1.0f-mag, 0.0f, 0.0f, norm[1], 0.0f, 1.0f-mag, 0.0f, norm[2], 0.0f, 0.0f, 1.0f-mag ); return focus; } /* Update the early and late 3D panning gains. */ static ALvoid Update3DPanning(const ALCdevice *Device, const ALfloat *ReflectionsPan, const ALfloat *LateReverbPan, const ALfloat earlyGain, const ALfloat lateGain, ReverbState *State) { aluMatrixf transform, rot; ALsizei i; STATIC_CAST(ALeffectState,State)->OutBuffer = Device->FOAOut.Buffer; STATIC_CAST(ALeffectState,State)->OutChannels = Device->FOAOut.NumChannels; /* Note: _res is transposed. */ #define MATRIX_MULT(_res, _m1, _m2) do { \ int row, col; \ for(col = 0;col < 4;col++) \ { \ for(row = 0;row < 4;row++) \ _res.m[col][row] = _m1.m[row][0]*_m2.m[0][col] + _m1.m[row][1]*_m2.m[1][col] + \ _m1.m[row][2]*_m2.m[2][col] + _m1.m[row][3]*_m2.m[3][col]; \ } \ } while(0) /* Create a matrix that first converts A-Format to B-Format, then * transforms the B-Format signal according to the panning vector. */ rot = GetTransformFromVector(ReflectionsPan); MATRIX_MULT(transform, rot, A2B); memset(&State->Early.PanGain, 0, sizeof(State->Early.PanGain)); for(i = 0;i < MAX_EFFECT_CHANNELS;i++) ComputePanGains(&Device->FOAOut, transform.m[i], earlyGain, State->Early.PanGain[i]); rot = GetTransformFromVector(LateReverbPan); MATRIX_MULT(transform, rot, A2B); memset(&State->Late.PanGain, 0, sizeof(State->Late.PanGain)); for(i = 0;i < MAX_EFFECT_CHANNELS;i++) ComputePanGains(&Device->FOAOut, transform.m[i], lateGain, State->Late.PanGain[i]); #undef MATRIX_MULT } static void ReverbState_update(ReverbState *State, const ALCcontext *Context, const ALeffectslot *Slot, const ALeffectProps *props) { const ALCdevice *Device = Context->Device; const ALlistener *Listener = Context->Listener; ALuint frequency = Device->Frequency; ALfloat lf0norm, hf0norm, hfRatio; ALfloat lfDecayTime, hfDecayTime; ALfloat gain, gainlf, gainhf; ALsizei i; /* Calculate the master filters */ hf0norm = minf(props->Reverb.HFReference / frequency, 0.49f); /* Restrict the filter gains from going below -60dB to keep the filter from * killing most of the signal. */ gainhf = maxf(props->Reverb.GainHF, 0.001f); BiquadFilter_setParams(&State->Filter[0].Lp, BiquadType_HighShelf, gainhf, hf0norm, calc_rcpQ_from_slope(gainhf, 1.0f)); lf0norm = minf(props->Reverb.LFReference / frequency, 0.49f); gainlf = maxf(props->Reverb.GainLF, 0.001f); BiquadFilter_setParams(&State->Filter[0].Hp, BiquadType_LowShelf, gainlf, lf0norm, calc_rcpQ_from_slope(gainlf, 1.0f)); for(i = 1;i < NUM_LINES;i++) { BiquadFilter_copyParams(&State->Filter[i].Lp, &State->Filter[0].Lp); BiquadFilter_copyParams(&State->Filter[i].Hp, &State->Filter[0].Hp); } /* Update the main effect delay and associated taps. */ UpdateDelayLine(props->Reverb.ReflectionsDelay, props->Reverb.LateReverbDelay, props->Reverb.Density, props->Reverb.DecayTime, frequency, State); /* Update the early lines. */ UpdateEarlyLines(props->Reverb.Density, props->Reverb.Diffusion, props->Reverb.DecayTime, frequency, &State->Early); /* Get the mixing matrix coefficients. */ CalcMatrixCoeffs(props->Reverb.Diffusion, &State->MixX, &State->MixY); /* If the HF limit parameter is flagged, calculate an appropriate limit * based on the air absorption parameter. */ hfRatio = props->Reverb.DecayHFRatio; if(props->Reverb.DecayHFLimit && props->Reverb.AirAbsorptionGainHF < 1.0f) hfRatio = CalcLimitedHfRatio(hfRatio, props->Reverb.AirAbsorptionGainHF, props->Reverb.DecayTime, Listener->Params.ReverbSpeedOfSound ); /* Calculate the LF/HF decay times. */ lfDecayTime = clampf(props->Reverb.DecayTime * props->Reverb.DecayLFRatio, AL_EAXREVERB_MIN_DECAY_TIME, AL_EAXREVERB_MAX_DECAY_TIME); hfDecayTime = clampf(props->Reverb.DecayTime * hfRatio, AL_EAXREVERB_MIN_DECAY_TIME, AL_EAXREVERB_MAX_DECAY_TIME); /* Update the late lines. */ UpdateLateLines(props->Reverb.Density, props->Reverb.Diffusion, lfDecayTime, props->Reverb.DecayTime, hfDecayTime, lf0norm, hf0norm, frequency, &State->Late ); /* Update early and late 3D panning. */ gain = props->Reverb.Gain * Slot->Params.Gain * ReverbBoost; Update3DPanning(Device, props->Reverb.ReflectionsPan, props->Reverb.LateReverbPan, props->Reverb.ReflectionsGain*gain, props->Reverb.LateReverbGain*gain, State); /* Calculate the max update size from the smallest relevant delay. */ State->MaxUpdate[1] = mini(MAX_UPDATE_SAMPLES, mini(State->Early.Offset[0][1], State->Late.Offset[0][1]) ); /* Determine if delay-line cross-fading is required. Density is essentially * a master control for the feedback delays, so changes the offsets of many * delay lines. */ if(State->Params.Density != props->Reverb.Density || /* Diffusion and decay times influences the decay rate (gain) of the * late reverb T60 filter. */ State->Params.Diffusion != props->Reverb.Diffusion || State->Params.DecayTime != props->Reverb.DecayTime || State->Params.HFDecayTime != hfDecayTime || State->Params.LFDecayTime != lfDecayTime || /* HF/LF References control the weighting used to calculate the density * gain. */ State->Params.HFReference != props->Reverb.HFReference || State->Params.LFReference != props->Reverb.LFReference) State->FadeCount = 0; State->Params.Density = props->Reverb.Density; State->Params.Diffusion = props->Reverb.Diffusion; State->Params.DecayTime = props->Reverb.DecayTime; State->Params.HFDecayTime = hfDecayTime; State->Params.LFDecayTime = lfDecayTime; State->Params.HFReference = props->Reverb.HFReference; State->Params.LFReference = props->Reverb.LFReference; } /************************************** * Effect Processing * **************************************/ /* Basic delay line input/output routines. */ static inline ALfloat DelayLineOut(const DelayLineI *Delay, const ALsizei offset, const ALsizei c) { return Delay->Line[offset&Delay->Mask][c]; } /* Cross-faded delay line output routine. Instead of interpolating the * offsets, this interpolates (cross-fades) the outputs at each offset. */ static inline ALfloat FadedDelayLineOut(const DelayLineI *Delay, const ALsizei off0, const ALsizei off1, const ALsizei c, const ALfloat sc0, const ALfloat sc1) { return Delay->Line[off0&Delay->Mask][c]*sc0 + Delay->Line[off1&Delay->Mask][c]*sc1; } static inline void DelayLineIn(const DelayLineI *Delay, ALsizei offset, const ALsizei c, const ALfloat *restrict in, ALsizei count) { ALsizei i; for(i = 0;i < count;i++) Delay->Line[(offset++)&Delay->Mask][c] = *(in++); } /* Applies a scattering matrix to the 4-line (vector) input. This is used * for both the below vector all-pass model and to perform modal feed-back * delay network (FDN) mixing. * * The matrix is derived from a skew-symmetric matrix to form a 4D rotation * matrix with a single unitary rotational parameter: * * [ d, a, b, c ] 1 = a^2 + b^2 + c^2 + d^2 * [ -a, d, c, -b ] * [ -b, -c, d, a ] * [ -c, b, -a, d ] * * The rotation is constructed from the effect's diffusion parameter, * yielding: * * 1 = x^2 + 3 y^2 * * Where a, b, and c are the coefficient y with differing signs, and d is the * coefficient x. The final matrix is thus: * * [ x, y, -y, y ] n = sqrt(matrix_order - 1) * [ -y, x, y, y ] t = diffusion_parameter * atan(n) * [ y, -y, x, y ] x = cos(t) * [ -y, -y, -y, x ] y = sin(t) / n * * Any square orthogonal matrix with an order that is a power of two will * work (where ^T is transpose, ^-1 is inverse): * * M^T = M^-1 * * Using that knowledge, finding an appropriate matrix can be accomplished * naively by searching all combinations of: * * M = D + S - S^T * * Where D is a diagonal matrix (of x), and S is a triangular matrix (of y) * whose combination of signs are being iterated. */ static inline void VectorPartialScatter(ALfloat *restrict out, const ALfloat *restrict in, const ALfloat xCoeff, const ALfloat yCoeff) { out[0] = xCoeff*in[0] + yCoeff*( in[1] + -in[2] + in[3]); out[1] = xCoeff*in[1] + yCoeff*(-in[0] + in[2] + in[3]); out[2] = xCoeff*in[2] + yCoeff*( in[0] + -in[1] + in[3]); out[3] = xCoeff*in[3] + yCoeff*(-in[0] + -in[1] + -in[2] ); } #define VectorScatterDelayIn(delay, o, in, xcoeff, ycoeff) \ VectorPartialScatter((delay)->Line[(o)&(delay)->Mask], in, xcoeff, ycoeff) /* Utilizes the above, but reverses the input channels. */ static inline void VectorScatterRevDelayIn(const DelayLineI *Delay, ALint offset, const ALfloat xCoeff, const ALfloat yCoeff, const ALfloat (*restrict in)[MAX_UPDATE_SAMPLES], const ALsizei count) { const DelayLineI delay = *Delay; ALsizei i, j; for(i = 0;i < count;++i) { ALfloat f[NUM_LINES]; for(j = 0;j < NUM_LINES;j++) f[NUM_LINES-1-j] = in[j][i]; VectorScatterDelayIn(&delay, offset++, f, xCoeff, yCoeff); } } /* This applies a Gerzon multiple-in/multiple-out (MIMO) vector all-pass * filter to the 4-line input. * * It works by vectorizing a regular all-pass filter and replacing the delay * element with a scattering matrix (like the one above) and a diagonal * matrix of delay elements. * * Two static specializations are used for transitional (cross-faded) delay * line processing and non-transitional processing. */ static void VectorAllpass_Unfaded(ALfloat (*restrict samples)[MAX_UPDATE_SAMPLES], ALsizei offset, const ALfloat xCoeff, const ALfloat yCoeff, ALsizei todo, VecAllpass *Vap) { const DelayLineI delay = Vap->Delay; const ALfloat feedCoeff = Vap->Coeff; ALsizei vap_offset[NUM_LINES]; ALsizei i, j; ASSUME(todo > 0); for(j = 0;j < NUM_LINES;j++) vap_offset[j] = offset-Vap->Offset[j][0]; for(i = 0;i < todo;i++) { ALfloat f[NUM_LINES]; for(j = 0;j < NUM_LINES;j++) { ALfloat input = samples[j][i]; ALfloat out = DelayLineOut(&delay, vap_offset[j]++, j) - feedCoeff*input; f[j] = input + feedCoeff*out; samples[j][i] = out; } VectorScatterDelayIn(&delay, offset, f, xCoeff, yCoeff); ++offset; } } static void VectorAllpass_Faded(ALfloat (*restrict samples)[MAX_UPDATE_SAMPLES], ALsizei offset, const ALfloat xCoeff, const ALfloat yCoeff, ALfloat fade, ALsizei todo, VecAllpass *Vap) { const DelayLineI delay = Vap->Delay; const ALfloat feedCoeff = Vap->Coeff; ALsizei vap_offset[NUM_LINES][2]; ALsizei i, j; ASSUME(todo > 0); fade *= 1.0f/FADE_SAMPLES; for(j = 0;j < NUM_LINES;j++) { vap_offset[j][0] = offset-Vap->Offset[j][0]; vap_offset[j][1] = offset-Vap->Offset[j][1]; } for(i = 0;i < todo;i++) { ALfloat f[NUM_LINES]; for(j = 0;j < NUM_LINES;j++) { ALfloat input = samples[j][i]; ALfloat out = FadedDelayLineOut(&delay, vap_offset[j][0]++, vap_offset[j][1]++, j, 1.0f-fade, fade ) - feedCoeff*input; f[j] = input + feedCoeff*out; samples[j][i] = out; } fade += FadeStep; VectorScatterDelayIn(&delay, offset, f, xCoeff, yCoeff); ++offset; } } /* This generates early reflections. * * This is done by obtaining the primary reflections (those arriving from the * same direction as the source) from the main delay line. These are * attenuated and all-pass filtered (based on the diffusion parameter). * * The early lines are then fed in reverse (according to the approximately * opposite spatial location of the A-Format lines) to create the secondary * reflections (those arriving from the opposite direction as the source). * * The early response is then completed by combining the primary reflections * with the delayed and attenuated output from the early lines. * * Finally, the early response is reversed, scattered (based on diffusion), * and fed into the late reverb section of the main delay line. * * Two static specializations are used for transitional (cross-faded) delay * line processing and non-transitional processing. */ static void EarlyReflection_Unfaded(ReverbState *State, ALsizei offset, const ALsizei todo, ALfloat (*restrict out)[MAX_UPDATE_SAMPLES]) { ALfloat (*restrict temps)[MAX_UPDATE_SAMPLES] = State->TempSamples; const DelayLineI early_delay = State->Early.Delay; const DelayLineI main_delay = State->Delay; const ALfloat mixX = State->MixX; const ALfloat mixY = State->MixY; ALsizei late_feed_tap; ALsizei i, j; ASSUME(todo > 0); /* First, load decorrelated samples from the main delay line as the primary * reflections. */ for(j = 0;j < NUM_LINES;j++) { ALsizei early_delay_tap = offset - State->EarlyDelayTap[j][0]; ALfloat coeff = State->EarlyDelayCoeff[j][0]; for(i = 0;i < todo;i++) temps[j][i] = DelayLineOut(&main_delay, early_delay_tap++, j) * coeff; } /* Apply a vector all-pass, to help color the initial reflections based on * the diffusion strength. */ VectorAllpass_Unfaded(temps, offset, mixX, mixY, todo, &State->Early.VecAp); /* Apply a delay and bounce to generate secondary reflections, combine with * the primary reflections and write out the result for mixing. */ for(j = 0;j < NUM_LINES;j++) { ALint early_feedb_tap = offset - State->Early.Offset[j][0]; ALfloat early_feedb_coeff = State->Early.Coeff[j][0]; for(i = 0;i < todo;i++) out[j][i] = DelayLineOut(&early_delay, early_feedb_tap++, j)*early_feedb_coeff + temps[j][i]; } for(j = 0;j < NUM_LINES;j++) DelayLineIn(&early_delay, offset, NUM_LINES-1-j, temps[j], todo); /* Also write the result back to the main delay line for the late reverb * stage to pick up at the appropriate time, appplying a scatter and * bounce to improve the initial diffusion in the late reverb. */ late_feed_tap = offset - State->LateFeedTap; VectorScatterRevDelayIn(&main_delay, late_feed_tap, mixX, mixY, out, todo); } static void EarlyReflection_Faded(ReverbState *State, ALsizei offset, const ALsizei todo, const ALfloat fade, ALfloat (*restrict out)[MAX_UPDATE_SAMPLES]) { ALfloat (*restrict temps)[MAX_UPDATE_SAMPLES] = State->TempSamples; const DelayLineI early_delay = State->Early.Delay; const DelayLineI main_delay = State->Delay; const ALfloat mixX = State->MixX; const ALfloat mixY = State->MixY; ALsizei late_feed_tap; ALsizei i, j; ASSUME(todo > 0); for(j = 0;j < NUM_LINES;j++) { ALsizei early_delay_tap0 = offset - State->EarlyDelayTap[j][0]; ALsizei early_delay_tap1 = offset - State->EarlyDelayTap[j][1]; ALfloat oldCoeff = State->EarlyDelayCoeff[j][0]; ALfloat oldCoeffStep = -oldCoeff / FADE_SAMPLES; ALfloat newCoeffStep = State->EarlyDelayCoeff[j][1] / FADE_SAMPLES; ALfloat fadeCount = fade; for(i = 0;i < todo;i++) { const ALfloat fade0 = oldCoeff + oldCoeffStep*fadeCount; const ALfloat fade1 = newCoeffStep*fadeCount; temps[j][i] = FadedDelayLineOut(&main_delay, early_delay_tap0++, early_delay_tap1++, j, fade0, fade1 ); fadeCount += 1.0f; } } VectorAllpass_Faded(temps, offset, mixX, mixY, fade, todo, &State->Early.VecAp); for(j = 0;j < NUM_LINES;j++) { ALint feedb_tap0 = offset - State->Early.Offset[j][0]; ALint feedb_tap1 = offset - State->Early.Offset[j][1]; ALfloat feedb_oldCoeff = State->Early.Coeff[j][0]; ALfloat feedb_oldCoeffStep = -feedb_oldCoeff / FADE_SAMPLES; ALfloat feedb_newCoeffStep = State->Early.Coeff[j][1] / FADE_SAMPLES; ALfloat fadeCount = fade; for(i = 0;i < todo;i++) { const ALfloat fade0 = feedb_oldCoeff + feedb_oldCoeffStep*fadeCount; const ALfloat fade1 = feedb_newCoeffStep*fadeCount; out[j][i] = FadedDelayLineOut(&early_delay, feedb_tap0++, feedb_tap1++, j, fade0, fade1 ) + temps[j][i]; fadeCount += 1.0f; } } for(j = 0;j < NUM_LINES;j++) DelayLineIn(&early_delay, offset, NUM_LINES-1-j, temps[j], todo); late_feed_tap = offset - State->LateFeedTap; VectorScatterRevDelayIn(&main_delay, late_feed_tap, mixX, mixY, out, todo); } /* Applies the two T60 damping filter sections. */ static inline void LateT60Filter(ALfloat *restrict samples, const ALsizei todo, T60Filter *filter) { ALfloat temp[MAX_UPDATE_SAMPLES]; BiquadFilter_process(&filter->HFFilter, temp, samples, todo); BiquadFilter_process(&filter->LFFilter, samples, temp, todo); } /* This generates the reverb tail using a modified feed-back delay network * (FDN). * * Results from the early reflections are mixed with the output from the late * delay lines. * * The late response is then completed by T60 and all-pass filtering the mix. * * Finally, the lines are reversed (so they feed their opposite directions) * and scattered with the FDN matrix before re-feeding the delay lines. * * Two variations are made, one for for transitional (cross-faded) delay line * processing and one for non-transitional processing. */ static void LateReverb_Unfaded(ReverbState *State, ALsizei offset, const ALsizei todo, ALfloat (*restrict out)[MAX_UPDATE_SAMPLES]) { ALfloat (*restrict temps)[MAX_UPDATE_SAMPLES] = State->TempSamples; const DelayLineI late_delay = State->Late.Delay; const DelayLineI main_delay = State->Delay; const ALfloat mixX = State->MixX; const ALfloat mixY = State->MixY; ALsizei i, j; ASSUME(todo > 0); /* First, load decorrelated samples from the main and feedback delay lines. * Filter the signal to apply its frequency-dependent decay. */ for(j = 0;j < NUM_LINES;j++) { ALsizei late_delay_tap = offset - State->LateDelayTap[j][0]; ALsizei late_feedb_tap = offset - State->Late.Offset[j][0]; ALfloat midGain = State->Late.T60[j].MidGain[0]; const ALfloat densityGain = State->Late.DensityGain[0] * midGain; for(i = 0;i < todo;i++) temps[j][i] = DelayLineOut(&main_delay, late_delay_tap++, j)*densityGain + DelayLineOut(&late_delay, late_feedb_tap++, j)*midGain; LateT60Filter(temps[j], todo, &State->Late.T60[j]); } /* Apply a vector all-pass to improve micro-surface diffusion, and write * out the results for mixing. */ VectorAllpass_Unfaded(temps, offset, mixX, mixY, todo, &State->Late.VecAp); for(j = 0;j < NUM_LINES;j++) memcpy(out[j], temps[j], todo*sizeof(ALfloat)); /* Finally, scatter and bounce the results to refeed the feedback buffer. */ VectorScatterRevDelayIn(&late_delay, offset, mixX, mixY, out, todo); } static void LateReverb_Faded(ReverbState *State, ALsizei offset, const ALsizei todo, const ALfloat fade, ALfloat (*restrict out)[MAX_UPDATE_SAMPLES]) { ALfloat (*restrict temps)[MAX_UPDATE_SAMPLES] = State->TempSamples; const DelayLineI late_delay = State->Late.Delay; const DelayLineI main_delay = State->Delay; const ALfloat mixX = State->MixX; const ALfloat mixY = State->MixY; ALsizei i, j; ASSUME(todo > 0); for(j = 0;j < NUM_LINES;j++) { const ALfloat oldMidGain = State->Late.T60[j].MidGain[0]; const ALfloat midGain = State->Late.T60[j].MidGain[1]; const ALfloat oldMidStep = -oldMidGain / FADE_SAMPLES; const ALfloat midStep = midGain / FADE_SAMPLES; const ALfloat oldDensityGain = State->Late.DensityGain[0] * oldMidGain; const ALfloat densityGain = State->Late.DensityGain[1] * midGain; const ALfloat oldDensityStep = -oldDensityGain / FADE_SAMPLES; const ALfloat densityStep = densityGain / FADE_SAMPLES; ALsizei late_delay_tap0 = offset - State->LateDelayTap[j][0]; ALsizei late_delay_tap1 = offset - State->LateDelayTap[j][1]; ALsizei late_feedb_tap0 = offset - State->Late.Offset[j][0]; ALsizei late_feedb_tap1 = offset - State->Late.Offset[j][1]; ALfloat fadeCount = fade; for(i = 0;i < todo;i++) { const ALfloat fade0 = oldDensityGain + oldDensityStep*fadeCount; const ALfloat fade1 = densityStep*fadeCount; const ALfloat gfade0 = oldMidGain + oldMidStep*fadeCount; const ALfloat gfade1 = midStep*fadeCount; temps[j][i] = FadedDelayLineOut(&main_delay, late_delay_tap0++, late_delay_tap1++, j, fade0, fade1) + FadedDelayLineOut(&late_delay, late_feedb_tap0++, late_feedb_tap1++, j, gfade0, gfade1); fadeCount += 1.0f; } LateT60Filter(temps[j], todo, &State->Late.T60[j]); } VectorAllpass_Faded(temps, offset, mixX, mixY, fade, todo, &State->Late.VecAp); for(j = 0;j < NUM_LINES;j++) memcpy(out[j], temps[j], todo*sizeof(ALfloat)); VectorScatterRevDelayIn(&late_delay, offset, mixX, mixY, temps, todo); } static ALvoid ReverbState_process(ReverbState *State, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels) { ALfloat (*restrict afmt)[MAX_UPDATE_SAMPLES] = State->TempSamples; ALfloat (*restrict samples)[MAX_UPDATE_SAMPLES] = State->MixSamples; ALsizei fadeCount = State->FadeCount; ALsizei offset = State->Offset; ALsizei base, c; /* Process reverb for these samples. */ for(base = 0;base < SamplesToDo;) { ALsizei todo = SamplesToDo - base; /* If cross-fading, don't do more samples than there are to fade. */ if(FADE_SAMPLES-fadeCount > 0) { todo = mini(todo, FADE_SAMPLES-fadeCount); todo = mini(todo, State->MaxUpdate[0]); } todo = mini(todo, State->MaxUpdate[1]); /* If this is not the final update, ensure the update size is a * multiple of 4 for the SIMD mixers. */ if(todo < SamplesToDo-base) todo &= ~3; /* Convert B-Format to A-Format for processing. */ memset(afmt, 0, sizeof(*afmt)*NUM_LINES); for(c = 0;c < NUM_LINES;c++) MixRowSamples(afmt[c], B2A.m[c], SamplesIn, MAX_EFFECT_CHANNELS, base, todo ); /* Process the samples for reverb. */ for(c = 0;c < NUM_LINES;c++) { /* Band-pass the incoming samples. */ BiquadFilter_process(&State->Filter[c].Lp, samples[0], afmt[c], todo); BiquadFilter_process(&State->Filter[c].Hp, samples[1], samples[0], todo); /* Feed the initial delay line. */ DelayLineIn(&State->Delay, offset, c, samples[1], todo); } if(UNLIKELY(fadeCount < FADE_SAMPLES)) { ALfloat fade = (ALfloat)fadeCount; /* Generate early reflections. */ EarlyReflection_Faded(State, offset, todo, fade, samples); /* Mix the A-Format results to output, implicitly converting back * to B-Format. */ for(c = 0;c < NUM_LINES;c++) MixSamples(samples[c], NumChannels, SamplesOut, State->Early.CurrentGain[c], State->Early.PanGain[c], SamplesToDo-base, base, todo ); /* Generate and mix late reverb. */ LateReverb_Faded(State, offset, todo, fade, samples); for(c = 0;c < NUM_LINES;c++) MixSamples(samples[c], NumChannels, SamplesOut, State->Late.CurrentGain[c], State->Late.PanGain[c], SamplesToDo-base, base, todo ); /* Step fading forward. */ fadeCount += todo; if(LIKELY(fadeCount >= FADE_SAMPLES)) { /* Update the cross-fading delay line taps. */ fadeCount = FADE_SAMPLES; for(c = 0;c < NUM_LINES;c++) { State->EarlyDelayTap[c][0] = State->EarlyDelayTap[c][1]; State->EarlyDelayCoeff[c][0] = State->EarlyDelayCoeff[c][1]; State->Early.VecAp.Offset[c][0] = State->Early.VecAp.Offset[c][1]; State->Early.Offset[c][0] = State->Early.Offset[c][1]; State->Early.Coeff[c][0] = State->Early.Coeff[c][1]; State->LateDelayTap[c][0] = State->LateDelayTap[c][1]; State->Late.VecAp.Offset[c][0] = State->Late.VecAp.Offset[c][1]; State->Late.Offset[c][0] = State->Late.Offset[c][1]; State->Late.T60[c].MidGain[0] = State->Late.T60[c].MidGain[1]; } State->Late.DensityGain[0] = State->Late.DensityGain[1]; State->MaxUpdate[0] = State->MaxUpdate[1]; } } else { /* Generate and mix early reflections. */ EarlyReflection_Unfaded(State, offset, todo, samples); for(c = 0;c < NUM_LINES;c++) MixSamples(samples[c], NumChannels, SamplesOut, State->Early.CurrentGain[c], State->Early.PanGain[c], SamplesToDo-base, base, todo ); /* Generate and mix late reverb. */ LateReverb_Unfaded(State, offset, todo, samples); for(c = 0;c < NUM_LINES;c++) MixSamples(samples[c], NumChannels, SamplesOut, State->Late.CurrentGain[c], State->Late.PanGain[c], SamplesToDo-base, base, todo ); } /* Step all delays forward. */ offset += todo; base += todo; } State->Offset = offset; State->FadeCount = fadeCount; } typedef struct ReverbStateFactory { DERIVE_FROM_TYPE(EffectStateFactory); } ReverbStateFactory; static ALeffectState *ReverbStateFactory_create(ReverbStateFactory* UNUSED(factory)) { ReverbState *state; NEW_OBJ0(state, ReverbState)(); if(!state) return NULL; return STATIC_CAST(ALeffectState, state); } DEFINE_EFFECTSTATEFACTORY_VTABLE(ReverbStateFactory); EffectStateFactory *ReverbStateFactory_getFactory(void) { static ReverbStateFactory ReverbFactory = { { GET_VTABLE2(ReverbStateFactory, EffectStateFactory) } }; return STATIC_CAST(EffectStateFactory, &ReverbFactory); } void ALeaxreverb_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_EAXREVERB_DECAY_HFLIMIT: if(!(val >= AL_EAXREVERB_MIN_DECAY_HFLIMIT && val <= AL_EAXREVERB_MAX_DECAY_HFLIMIT)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb decay hflimit out of range"); props->Reverb.DecayHFLimit = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid EAX reverb integer property 0x%04x", param); } } void ALeaxreverb_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals) { ALeaxreverb_setParami(effect, context, param, vals[0]); } void ALeaxreverb_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_EAXREVERB_DENSITY: if(!(val >= AL_EAXREVERB_MIN_DENSITY && val <= AL_EAXREVERB_MAX_DENSITY)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb density out of range"); props->Reverb.Density = val; break; case AL_EAXREVERB_DIFFUSION: if(!(val >= AL_EAXREVERB_MIN_DIFFUSION && val <= AL_EAXREVERB_MAX_DIFFUSION)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb diffusion out of range"); props->Reverb.Diffusion = val; break; case AL_EAXREVERB_GAIN: if(!(val >= AL_EAXREVERB_MIN_GAIN && val <= AL_EAXREVERB_MAX_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb gain out of range"); props->Reverb.Gain = val; break; case AL_EAXREVERB_GAINHF: if(!(val >= AL_EAXREVERB_MIN_GAINHF && val <= AL_EAXREVERB_MAX_GAINHF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb gainhf out of range"); props->Reverb.GainHF = val; break; case AL_EAXREVERB_GAINLF: if(!(val >= AL_EAXREVERB_MIN_GAINLF && val <= AL_EAXREVERB_MAX_GAINLF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb gainlf out of range"); props->Reverb.GainLF = val; break; case AL_EAXREVERB_DECAY_TIME: if(!(val >= AL_EAXREVERB_MIN_DECAY_TIME && val <= AL_EAXREVERB_MAX_DECAY_TIME)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb decay time out of range"); props->Reverb.DecayTime = val; break; case AL_EAXREVERB_DECAY_HFRATIO: if(!(val >= AL_EAXREVERB_MIN_DECAY_HFRATIO && val <= AL_EAXREVERB_MAX_DECAY_HFRATIO)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb decay hfratio out of range"); props->Reverb.DecayHFRatio = val; break; case AL_EAXREVERB_DECAY_LFRATIO: if(!(val >= AL_EAXREVERB_MIN_DECAY_LFRATIO && val <= AL_EAXREVERB_MAX_DECAY_LFRATIO)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb decay lfratio out of range"); props->Reverb.DecayLFRatio = val; break; case AL_EAXREVERB_REFLECTIONS_GAIN: if(!(val >= AL_EAXREVERB_MIN_REFLECTIONS_GAIN && val <= AL_EAXREVERB_MAX_REFLECTIONS_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb reflections gain out of range"); props->Reverb.ReflectionsGain = val; break; case AL_EAXREVERB_REFLECTIONS_DELAY: if(!(val >= AL_EAXREVERB_MIN_REFLECTIONS_DELAY && val <= AL_EAXREVERB_MAX_REFLECTIONS_DELAY)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb reflections delay out of range"); props->Reverb.ReflectionsDelay = val; break; case AL_EAXREVERB_LATE_REVERB_GAIN: if(!(val >= AL_EAXREVERB_MIN_LATE_REVERB_GAIN && val <= AL_EAXREVERB_MAX_LATE_REVERB_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb late reverb gain out of range"); props->Reverb.LateReverbGain = val; break; case AL_EAXREVERB_LATE_REVERB_DELAY: if(!(val >= AL_EAXREVERB_MIN_LATE_REVERB_DELAY && val <= AL_EAXREVERB_MAX_LATE_REVERB_DELAY)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb late reverb delay out of range"); props->Reverb.LateReverbDelay = val; break; case AL_EAXREVERB_AIR_ABSORPTION_GAINHF: if(!(val >= AL_EAXREVERB_MIN_AIR_ABSORPTION_GAINHF && val <= AL_EAXREVERB_MAX_AIR_ABSORPTION_GAINHF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb air absorption gainhf out of range"); props->Reverb.AirAbsorptionGainHF = val; break; case AL_EAXREVERB_ECHO_TIME: if(!(val >= AL_EAXREVERB_MIN_ECHO_TIME && val <= AL_EAXREVERB_MAX_ECHO_TIME)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb echo time out of range"); props->Reverb.EchoTime = val; break; case AL_EAXREVERB_ECHO_DEPTH: if(!(val >= AL_EAXREVERB_MIN_ECHO_DEPTH && val <= AL_EAXREVERB_MAX_ECHO_DEPTH)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb echo depth out of range"); props->Reverb.EchoDepth = val; break; case AL_EAXREVERB_MODULATION_TIME: if(!(val >= AL_EAXREVERB_MIN_MODULATION_TIME && val <= AL_EAXREVERB_MAX_MODULATION_TIME)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb modulation time out of range"); props->Reverb.ModulationTime = val; break; case AL_EAXREVERB_MODULATION_DEPTH: if(!(val >= AL_EAXREVERB_MIN_MODULATION_DEPTH && val <= AL_EAXREVERB_MAX_MODULATION_DEPTH)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb modulation depth out of range"); props->Reverb.ModulationDepth = val; break; case AL_EAXREVERB_HFREFERENCE: if(!(val >= AL_EAXREVERB_MIN_HFREFERENCE && val <= AL_EAXREVERB_MAX_HFREFERENCE)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb hfreference out of range"); props->Reverb.HFReference = val; break; case AL_EAXREVERB_LFREFERENCE: if(!(val >= AL_EAXREVERB_MIN_LFREFERENCE && val <= AL_EAXREVERB_MAX_LFREFERENCE)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb lfreference out of range"); props->Reverb.LFReference = val; break; case AL_EAXREVERB_ROOM_ROLLOFF_FACTOR: if(!(val >= AL_EAXREVERB_MIN_ROOM_ROLLOFF_FACTOR && val <= AL_EAXREVERB_MAX_ROOM_ROLLOFF_FACTOR)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb room rolloff factor out of range"); props->Reverb.RoomRolloffFactor = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid EAX reverb float property 0x%04x", param); } } void ALeaxreverb_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { ALeffectProps *props = &effect->Props; switch(param) { case AL_EAXREVERB_REFLECTIONS_PAN: if(!(isfinite(vals[0]) && isfinite(vals[1]) && isfinite(vals[2]))) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb reflections pan out of range"); props->Reverb.ReflectionsPan[0] = vals[0]; props->Reverb.ReflectionsPan[1] = vals[1]; props->Reverb.ReflectionsPan[2] = vals[2]; break; case AL_EAXREVERB_LATE_REVERB_PAN: if(!(isfinite(vals[0]) && isfinite(vals[1]) && isfinite(vals[2]))) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb late reverb pan out of range"); props->Reverb.LateReverbPan[0] = vals[0]; props->Reverb.LateReverbPan[1] = vals[1]; props->Reverb.LateReverbPan[2] = vals[2]; break; default: ALeaxreverb_setParamf(effect, context, param, vals[0]); break; } } void ALeaxreverb_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_EAXREVERB_DECAY_HFLIMIT: *val = props->Reverb.DecayHFLimit; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid EAX reverb integer property 0x%04x", param); } } void ALeaxreverb_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals) { ALeaxreverb_getParami(effect, context, param, vals); } void ALeaxreverb_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_EAXREVERB_DENSITY: *val = props->Reverb.Density; break; case AL_EAXREVERB_DIFFUSION: *val = props->Reverb.Diffusion; break; case AL_EAXREVERB_GAIN: *val = props->Reverb.Gain; break; case AL_EAXREVERB_GAINHF: *val = props->Reverb.GainHF; break; case AL_EAXREVERB_GAINLF: *val = props->Reverb.GainLF; break; case AL_EAXREVERB_DECAY_TIME: *val = props->Reverb.DecayTime; break; case AL_EAXREVERB_DECAY_HFRATIO: *val = props->Reverb.DecayHFRatio; break; case AL_EAXREVERB_DECAY_LFRATIO: *val = props->Reverb.DecayLFRatio; break; case AL_EAXREVERB_REFLECTIONS_GAIN: *val = props->Reverb.ReflectionsGain; break; case AL_EAXREVERB_REFLECTIONS_DELAY: *val = props->Reverb.ReflectionsDelay; break; case AL_EAXREVERB_LATE_REVERB_GAIN: *val = props->Reverb.LateReverbGain; break; case AL_EAXREVERB_LATE_REVERB_DELAY: *val = props->Reverb.LateReverbDelay; break; case AL_EAXREVERB_AIR_ABSORPTION_GAINHF: *val = props->Reverb.AirAbsorptionGainHF; break; case AL_EAXREVERB_ECHO_TIME: *val = props->Reverb.EchoTime; break; case AL_EAXREVERB_ECHO_DEPTH: *val = props->Reverb.EchoDepth; break; case AL_EAXREVERB_MODULATION_TIME: *val = props->Reverb.ModulationTime; break; case AL_EAXREVERB_MODULATION_DEPTH: *val = props->Reverb.ModulationDepth; break; case AL_EAXREVERB_HFREFERENCE: *val = props->Reverb.HFReference; break; case AL_EAXREVERB_LFREFERENCE: *val = props->Reverb.LFReference; break; case AL_EAXREVERB_ROOM_ROLLOFF_FACTOR: *val = props->Reverb.RoomRolloffFactor; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid EAX reverb float property 0x%04x", param); } } void ALeaxreverb_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_EAXREVERB_REFLECTIONS_PAN: vals[0] = props->Reverb.ReflectionsPan[0]; vals[1] = props->Reverb.ReflectionsPan[1]; vals[2] = props->Reverb.ReflectionsPan[2]; break; case AL_EAXREVERB_LATE_REVERB_PAN: vals[0] = props->Reverb.LateReverbPan[0]; vals[1] = props->Reverb.LateReverbPan[1]; vals[2] = props->Reverb.LateReverbPan[2]; break; default: ALeaxreverb_getParamf(effect, context, param, vals); break; } } DEFINE_ALEFFECT_VTABLE(ALeaxreverb); void ALreverb_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_REVERB_DECAY_HFLIMIT: if(!(val >= AL_REVERB_MIN_DECAY_HFLIMIT && val <= AL_REVERB_MAX_DECAY_HFLIMIT)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb decay hflimit out of range"); props->Reverb.DecayHFLimit = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid reverb integer property 0x%04x", param); } } void ALreverb_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals) { ALreverb_setParami(effect, context, param, vals[0]); } void ALreverb_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_REVERB_DENSITY: if(!(val >= AL_REVERB_MIN_DENSITY && val <= AL_REVERB_MAX_DENSITY)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb density out of range"); props->Reverb.Density = val; break; case AL_REVERB_DIFFUSION: if(!(val >= AL_REVERB_MIN_DIFFUSION && val <= AL_REVERB_MAX_DIFFUSION)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb diffusion out of range"); props->Reverb.Diffusion = val; break; case AL_REVERB_GAIN: if(!(val >= AL_REVERB_MIN_GAIN && val <= AL_REVERB_MAX_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb gain out of range"); props->Reverb.Gain = val; break; case AL_REVERB_GAINHF: if(!(val >= AL_REVERB_MIN_GAINHF && val <= AL_REVERB_MAX_GAINHF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb gainhf out of range"); props->Reverb.GainHF = val; break; case AL_REVERB_DECAY_TIME: if(!(val >= AL_REVERB_MIN_DECAY_TIME && val <= AL_REVERB_MAX_DECAY_TIME)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb decay time out of range"); props->Reverb.DecayTime = val; break; case AL_REVERB_DECAY_HFRATIO: if(!(val >= AL_REVERB_MIN_DECAY_HFRATIO && val <= AL_REVERB_MAX_DECAY_HFRATIO)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb decay hfratio out of range"); props->Reverb.DecayHFRatio = val; break; case AL_REVERB_REFLECTIONS_GAIN: if(!(val >= AL_REVERB_MIN_REFLECTIONS_GAIN && val <= AL_REVERB_MAX_REFLECTIONS_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb reflections gain out of range"); props->Reverb.ReflectionsGain = val; break; case AL_REVERB_REFLECTIONS_DELAY: if(!(val >= AL_REVERB_MIN_REFLECTIONS_DELAY && val <= AL_REVERB_MAX_REFLECTIONS_DELAY)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb reflections delay out of range"); props->Reverb.ReflectionsDelay = val; break; case AL_REVERB_LATE_REVERB_GAIN: if(!(val >= AL_REVERB_MIN_LATE_REVERB_GAIN && val <= AL_REVERB_MAX_LATE_REVERB_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb late reverb gain out of range"); props->Reverb.LateReverbGain = val; break; case AL_REVERB_LATE_REVERB_DELAY: if(!(val >= AL_REVERB_MIN_LATE_REVERB_DELAY && val <= AL_REVERB_MAX_LATE_REVERB_DELAY)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb late reverb delay out of range"); props->Reverb.LateReverbDelay = val; break; case AL_REVERB_AIR_ABSORPTION_GAINHF: if(!(val >= AL_REVERB_MIN_AIR_ABSORPTION_GAINHF && val <= AL_REVERB_MAX_AIR_ABSORPTION_GAINHF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb air absorption gainhf out of range"); props->Reverb.AirAbsorptionGainHF = val; break; case AL_REVERB_ROOM_ROLLOFF_FACTOR: if(!(val >= AL_REVERB_MIN_ROOM_ROLLOFF_FACTOR && val <= AL_REVERB_MAX_ROOM_ROLLOFF_FACTOR)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb room rolloff factor out of range"); props->Reverb.RoomRolloffFactor = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid reverb float property 0x%04x", param); } } void ALreverb_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { ALreverb_setParamf(effect, context, param, vals[0]); } void ALreverb_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_REVERB_DECAY_HFLIMIT: *val = props->Reverb.DecayHFLimit; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid reverb integer property 0x%04x", param); } } void ALreverb_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals) { ALreverb_getParami(effect, context, param, vals); } void ALreverb_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_REVERB_DENSITY: *val = props->Reverb.Density; break; case AL_REVERB_DIFFUSION: *val = props->Reverb.Diffusion; break; case AL_REVERB_GAIN: *val = props->Reverb.Gain; break; case AL_REVERB_GAINHF: *val = props->Reverb.GainHF; break; case AL_REVERB_DECAY_TIME: *val = props->Reverb.DecayTime; break; case AL_REVERB_DECAY_HFRATIO: *val = props->Reverb.DecayHFRatio; break; case AL_REVERB_REFLECTIONS_GAIN: *val = props->Reverb.ReflectionsGain; break; case AL_REVERB_REFLECTIONS_DELAY: *val = props->Reverb.ReflectionsDelay; break; case AL_REVERB_LATE_REVERB_GAIN: *val = props->Reverb.LateReverbGain; break; case AL_REVERB_LATE_REVERB_DELAY: *val = props->Reverb.LateReverbDelay; break; case AL_REVERB_AIR_ABSORPTION_GAINHF: *val = props->Reverb.AirAbsorptionGainHF; break; case AL_REVERB_ROOM_ROLLOFF_FACTOR: *val = props->Reverb.RoomRolloffFactor; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid reverb float property 0x%04x", param); } } void ALreverb_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { ALreverb_getParamf(effect, context, param, vals); } DEFINE_ALEFFECT_VTABLE(ALreverb); openal-soft-openal-soft-1.19.1/Alc/filters/000077500000000000000000000000001335774445300204455ustar00rootroot00000000000000openal-soft-openal-soft-1.19.1/Alc/filters/defs.h000066400000000000000000000076501335774445300215470ustar00rootroot00000000000000#ifndef ALC_FILTER_H #define ALC_FILTER_H #include "AL/al.h" #include "math_defs.h" /* Filters implementation is based on the "Cookbook formulae for audio * EQ biquad filter coefficients" by Robert Bristow-Johnson * http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt */ /* Implementation note: For the shelf filters, the specified gain is for the * reference frequency, which is the centerpoint of the transition band. This * better matches EFX filter design. To set the gain for the shelf itself, use * the square root of the desired linear gain (or halve the dB gain). */ typedef enum BiquadType { /** EFX-style low-pass filter, specifying a gain and reference frequency. */ BiquadType_HighShelf, /** EFX-style high-pass filter, specifying a gain and reference frequency. */ BiquadType_LowShelf, /** Peaking filter, specifying a gain and reference frequency. */ BiquadType_Peaking, /** Low-pass cut-off filter, specifying a cut-off frequency. */ BiquadType_LowPass, /** High-pass cut-off filter, specifying a cut-off frequency. */ BiquadType_HighPass, /** Band-pass filter, specifying a center frequency. */ BiquadType_BandPass, } BiquadType; typedef struct BiquadFilter { ALfloat z1, z2; /* Last two delayed components for direct form II. */ ALfloat b0, b1, b2; /* Transfer function coefficients "b" (numerator) */ ALfloat a1, a2; /* Transfer function coefficients "a" (denominator; a0 is * pre-applied). */ } BiquadFilter; /* Currently only a C-based filter process method is implemented. */ #define BiquadFilter_process BiquadFilter_processC /** * Calculates the rcpQ (i.e. 1/Q) coefficient for shelving filters, using the * reference gain and shelf slope parameter. * \param gain 0 < gain * \param slope 0 < slope <= 1 */ inline ALfloat calc_rcpQ_from_slope(ALfloat gain, ALfloat slope) { return sqrtf((gain + 1.0f/gain)*(1.0f/slope - 1.0f) + 2.0f); } /** * Calculates the rcpQ (i.e. 1/Q) coefficient for filters, using the normalized * reference frequency and bandwidth. * \param f0norm 0 < f0norm < 0.5. * \param bandwidth 0 < bandwidth */ inline ALfloat calc_rcpQ_from_bandwidth(ALfloat f0norm, ALfloat bandwidth) { ALfloat w0 = F_TAU * f0norm; return 2.0f*sinhf(logf(2.0f)/2.0f*bandwidth*w0/sinf(w0)); } inline void BiquadFilter_clear(BiquadFilter *filter) { filter->z1 = 0.0f; filter->z2 = 0.0f; } /** * Sets up the filter state for the specified filter type and its parameters. * * \param filter The filter object to prepare. * \param type The type of filter for the object to apply. * \param gain The gain for the reference frequency response. Only used by the * Shelf and Peaking filter types. * \param f0norm The normalized reference frequency (ref_freq / sample_rate). * This is the center point for the Shelf, Peaking, and BandPass * filter types, or the cutoff frequency for the LowPass and * HighPass filter types. * \param rcpQ The reciprocal of the Q coefficient for the filter's transition * band. Can be generated from calc_rcpQ_from_slope or * calc_rcpQ_from_bandwidth depending on the available data. */ void BiquadFilter_setParams(BiquadFilter *filter, BiquadType type, ALfloat gain, ALfloat f0norm, ALfloat rcpQ); inline void BiquadFilter_copyParams(BiquadFilter *restrict dst, const BiquadFilter *restrict src) { dst->b0 = src->b0; dst->b1 = src->b1; dst->b2 = src->b2; dst->a1 = src->a1; dst->a2 = src->a2; } void BiquadFilter_processC(BiquadFilter *filter, ALfloat *restrict dst, const ALfloat *restrict src, ALsizei numsamples); inline void BiquadFilter_passthru(BiquadFilter *filter, ALsizei numsamples) { if(LIKELY(numsamples >= 2)) { filter->z1 = 0.0f; filter->z2 = 0.0f; } else if(numsamples == 1) { filter->z1 = filter->z2; filter->z2 = 0.0f; } } #endif /* ALC_FILTER_H */ openal-soft-openal-soft-1.19.1/Alc/filters/filter.c000066400000000000000000000110251335774445300220750ustar00rootroot00000000000000 #include "config.h" #include "AL/alc.h" #include "AL/al.h" #include "alMain.h" #include "defs.h" extern inline void BiquadFilter_clear(BiquadFilter *filter); extern inline void BiquadFilter_copyParams(BiquadFilter *restrict dst, const BiquadFilter *restrict src); extern inline void BiquadFilter_passthru(BiquadFilter *filter, ALsizei numsamples); extern inline ALfloat calc_rcpQ_from_slope(ALfloat gain, ALfloat slope); extern inline ALfloat calc_rcpQ_from_bandwidth(ALfloat f0norm, ALfloat bandwidth); void BiquadFilter_setParams(BiquadFilter *filter, BiquadType type, ALfloat gain, ALfloat f0norm, ALfloat rcpQ) { ALfloat alpha, sqrtgain_alpha_2; ALfloat w0, sin_w0, cos_w0; ALfloat a[3] = { 1.0f, 0.0f, 0.0f }; ALfloat b[3] = { 1.0f, 0.0f, 0.0f }; // Limit gain to -100dB assert(gain > 0.00001f); w0 = F_TAU * f0norm; sin_w0 = sinf(w0); cos_w0 = cosf(w0); alpha = sin_w0/2.0f * rcpQ; /* Calculate filter coefficients depending on filter type */ switch(type) { case BiquadType_HighShelf: sqrtgain_alpha_2 = 2.0f * sqrtf(gain) * alpha; b[0] = gain*((gain+1.0f) + (gain-1.0f)*cos_w0 + sqrtgain_alpha_2); b[1] = -2.0f*gain*((gain-1.0f) + (gain+1.0f)*cos_w0 ); b[2] = gain*((gain+1.0f) + (gain-1.0f)*cos_w0 - sqrtgain_alpha_2); a[0] = (gain+1.0f) - (gain-1.0f)*cos_w0 + sqrtgain_alpha_2; a[1] = 2.0f* ((gain-1.0f) - (gain+1.0f)*cos_w0 ); a[2] = (gain+1.0f) - (gain-1.0f)*cos_w0 - sqrtgain_alpha_2; break; case BiquadType_LowShelf: sqrtgain_alpha_2 = 2.0f * sqrtf(gain) * alpha; b[0] = gain*((gain+1.0f) - (gain-1.0f)*cos_w0 + sqrtgain_alpha_2); b[1] = 2.0f*gain*((gain-1.0f) - (gain+1.0f)*cos_w0 ); b[2] = gain*((gain+1.0f) - (gain-1.0f)*cos_w0 - sqrtgain_alpha_2); a[0] = (gain+1.0f) + (gain-1.0f)*cos_w0 + sqrtgain_alpha_2; a[1] = -2.0f* ((gain-1.0f) + (gain+1.0f)*cos_w0 ); a[2] = (gain+1.0f) + (gain-1.0f)*cos_w0 - sqrtgain_alpha_2; break; case BiquadType_Peaking: gain = sqrtf(gain); b[0] = 1.0f + alpha * gain; b[1] = -2.0f * cos_w0; b[2] = 1.0f - alpha * gain; a[0] = 1.0f + alpha / gain; a[1] = -2.0f * cos_w0; a[2] = 1.0f - alpha / gain; break; case BiquadType_LowPass: b[0] = (1.0f - cos_w0) / 2.0f; b[1] = 1.0f - cos_w0; b[2] = (1.0f - cos_w0) / 2.0f; a[0] = 1.0f + alpha; a[1] = -2.0f * cos_w0; a[2] = 1.0f - alpha; break; case BiquadType_HighPass: b[0] = (1.0f + cos_w0) / 2.0f; b[1] = -(1.0f + cos_w0); b[2] = (1.0f + cos_w0) / 2.0f; a[0] = 1.0f + alpha; a[1] = -2.0f * cos_w0; a[2] = 1.0f - alpha; break; case BiquadType_BandPass: b[0] = alpha; b[1] = 0; b[2] = -alpha; a[0] = 1.0f + alpha; a[1] = -2.0f * cos_w0; a[2] = 1.0f - alpha; break; } filter->a1 = a[1] / a[0]; filter->a2 = a[2] / a[0]; filter->b0 = b[0] / a[0]; filter->b1 = b[1] / a[0]; filter->b2 = b[2] / a[0]; } void BiquadFilter_processC(BiquadFilter *filter, ALfloat *restrict dst, const ALfloat *restrict src, ALsizei numsamples) { const ALfloat a1 = filter->a1; const ALfloat a2 = filter->a2; const ALfloat b0 = filter->b0; const ALfloat b1 = filter->b1; const ALfloat b2 = filter->b2; ALfloat z1 = filter->z1; ALfloat z2 = filter->z2; ALsizei i; ASSUME(numsamples > 0); /* Processing loop is Transposed Direct Form II. This requires less storage * compared to Direct Form I (only two delay components, instead of a four- * sample history; the last two inputs and outputs), and works better for * floating-point which favors summing similarly-sized values while being * less bothered by overflow. * * See: http://www.earlevel.com/main/2003/02/28/biquads/ */ for(i = 0;i < numsamples;i++) { ALfloat input = src[i]; ALfloat output = input*b0 + z1; z1 = input*b1 - output*a1 + z2; z2 = input*b2 - output*a2; dst[i] = output; } filter->z1 = z1; filter->z2 = z2; } openal-soft-openal-soft-1.19.1/Alc/filters/nfc.c000066400000000000000000000261151335774445300213640ustar00rootroot00000000000000 #include "config.h" #include "nfc.h" #include "alMain.h" #include /* Near-field control filters are the basis for handling the near-field effect. * The near-field effect is a bass-boost present in the directional components * of a recorded signal, created as a result of the wavefront curvature (itself * a function of sound distance). Proper reproduction dictates this be * compensated for using a bass-cut given the playback speaker distance, to * avoid excessive bass in the playback. * * For real-time rendered audio, emulating the near-field effect based on the * sound source's distance, and subsequently compensating for it at output * based on the speaker distances, can create a more realistic perception of * sound distance beyond a simple 1/r attenuation. * * These filters do just that. Each one applies a low-shelf filter, created as * the combination of a bass-boost for a given sound source distance (near- * field emulation) along with a bass-cut for a given control/speaker distance * (near-field compensation). * * Note that it is necessary to apply a cut along with the boost, since the * boost alone is unstable in higher-order ambisonics as it causes an infinite * DC gain (even first-order ambisonics requires there to be no DC offset for * the boost to work). Consequently, ambisonics requires a control parameter to * be used to avoid an unstable boost-only filter. NFC-HOA defines this control * as a reference delay, calculated with: * * reference_delay = control_distance / speed_of_sound * * This means w0 (for input) or w1 (for output) should be set to: * * wN = 1 / (reference_delay * sample_rate) * * when dealing with NFC-HOA content. For FOA input content, which does not * specify a reference_delay variable, w0 should be set to 0 to apply only * near-field compensation for output. It's important that w1 be a finite, * positive, non-0 value or else the bass-boost will become unstable again. * Also, w0 should not be too large compared to w1, to avoid excessively loud * low frequencies. */ static const float B[4][3] = { { 0.0f }, { 1.0f }, { 3.0f, 3.0f }, { 3.6778f, 6.4595f, 2.3222f }, /*{ 4.2076f, 11.4877f, 5.7924f, 9.1401f }*/ }; static void NfcFilterCreate1(struct NfcFilter1 *nfc, const float w0, const float w1) { float b_00, g_0; float r; nfc->base_gain = 1.0f; nfc->gain = 1.0f; /* Calculate bass-boost coefficients. */ r = 0.5f * w0; b_00 = B[1][0] * r; g_0 = 1.0f + b_00; nfc->gain *= g_0; nfc->b1 = 2.0f * b_00 / g_0; /* Calculate bass-cut coefficients. */ r = 0.5f * w1; b_00 = B[1][0] * r; g_0 = 1.0f + b_00; nfc->base_gain /= g_0; nfc->gain /= g_0; nfc->a1 = 2.0f * b_00 / g_0; } static void NfcFilterAdjust1(struct NfcFilter1 *nfc, const float w0) { float b_00, g_0; float r; r = 0.5f * w0; b_00 = B[1][0] * r; g_0 = 1.0f + b_00; nfc->gain = nfc->base_gain * g_0; nfc->b1 = 2.0f * b_00 / g_0; } static void NfcFilterCreate2(struct NfcFilter2 *nfc, const float w0, const float w1) { float b_10, b_11, g_1; float r; nfc->base_gain = 1.0f; nfc->gain = 1.0f; /* Calculate bass-boost coefficients. */ r = 0.5f * w0; b_10 = B[2][0] * r; b_11 = B[2][1] * r * r; g_1 = 1.0f + b_10 + b_11; nfc->gain *= g_1; nfc->b1 = (2.0f*b_10 + 4.0f*b_11) / g_1; nfc->b2 = 4.0f * b_11 / g_1; /* Calculate bass-cut coefficients. */ r = 0.5f * w1; b_10 = B[2][0] * r; b_11 = B[2][1] * r * r; g_1 = 1.0f + b_10 + b_11; nfc->base_gain /= g_1; nfc->gain /= g_1; nfc->a1 = (2.0f*b_10 + 4.0f*b_11) / g_1; nfc->a2 = 4.0f * b_11 / g_1; } static void NfcFilterAdjust2(struct NfcFilter2 *nfc, const float w0) { float b_10, b_11, g_1; float r; r = 0.5f * w0; b_10 = B[2][0] * r; b_11 = B[2][1] * r * r; g_1 = 1.0f + b_10 + b_11; nfc->gain = nfc->base_gain * g_1; nfc->b1 = (2.0f*b_10 + 4.0f*b_11) / g_1; nfc->b2 = 4.0f * b_11 / g_1; } static void NfcFilterCreate3(struct NfcFilter3 *nfc, const float w0, const float w1) { float b_10, b_11, g_1; float b_00, g_0; float r; nfc->base_gain = 1.0f; nfc->gain = 1.0f; /* Calculate bass-boost coefficients. */ r = 0.5f * w0; b_10 = B[3][0] * r; b_11 = B[3][1] * r * r; g_1 = 1.0f + b_10 + b_11; nfc->gain *= g_1; nfc->b1 = (2.0f*b_10 + 4.0f*b_11) / g_1; nfc->b2 = 4.0f * b_11 / g_1; b_00 = B[3][2] * r; g_0 = 1.0f + b_00; nfc->gain *= g_0; nfc->b3 = 2.0f * b_00 / g_0; /* Calculate bass-cut coefficients. */ r = 0.5f * w1; b_10 = B[3][0] * r; b_11 = B[3][1] * r * r; g_1 = 1.0f + b_10 + b_11; nfc->base_gain /= g_1; nfc->gain /= g_1; nfc->a1 = (2.0f*b_10 + 4.0f*b_11) / g_1; nfc->a2 = 4.0f * b_11 / g_1; b_00 = B[3][2] * r; g_0 = 1.0f + b_00; nfc->base_gain /= g_0; nfc->gain /= g_0; nfc->a3 = 2.0f * b_00 / g_0; } static void NfcFilterAdjust3(struct NfcFilter3 *nfc, const float w0) { float b_10, b_11, g_1; float b_00, g_0; float r; r = 0.5f * w0; b_10 = B[3][0] * r; b_11 = B[3][1] * r * r; g_1 = 1.0f + b_10 + b_11; nfc->gain = nfc->base_gain * g_1; nfc->b1 = (2.0f*b_10 + 4.0f*b_11) / g_1; nfc->b2 = 4.0f * b_11 / g_1; b_00 = B[3][2] * r; g_0 = 1.0f + b_00; nfc->gain *= g_0; nfc->b3 = 2.0f * b_00 / g_0; } void NfcFilterCreate(NfcFilter *nfc, const float w0, const float w1) { memset(nfc, 0, sizeof(*nfc)); NfcFilterCreate1(&nfc->first, w0, w1); NfcFilterCreate2(&nfc->second, w0, w1); NfcFilterCreate3(&nfc->third, w0, w1); } void NfcFilterAdjust(NfcFilter *nfc, const float w0) { NfcFilterAdjust1(&nfc->first, w0); NfcFilterAdjust2(&nfc->second, w0); NfcFilterAdjust3(&nfc->third, w0); } void NfcFilterProcess1(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count) { const float gain = nfc->first.gain; const float b1 = nfc->first.b1; const float a1 = nfc->first.a1; float z1 = nfc->first.z[0]; int i; ASSUME(count > 0); for(i = 0;i < count;i++) { float y = src[i]*gain - a1*z1; float out = y + b1*z1; z1 += y; dst[i] = out; } nfc->first.z[0] = z1; } void NfcFilterProcess2(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count) { const float gain = nfc->second.gain; const float b1 = nfc->second.b1; const float b2 = nfc->second.b2; const float a1 = nfc->second.a1; const float a2 = nfc->second.a2; float z1 = nfc->second.z[0]; float z2 = nfc->second.z[1]; int i; ASSUME(count > 0); for(i = 0;i < count;i++) { float y = src[i]*gain - a1*z1 - a2*z2; float out = y + b1*z1 + b2*z2; z2 += z1; z1 += y; dst[i] = out; } nfc->second.z[0] = z1; nfc->second.z[1] = z2; } void NfcFilterProcess3(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count) { const float gain = nfc->third.gain; const float b1 = nfc->third.b1; const float b2 = nfc->third.b2; const float b3 = nfc->third.b3; const float a1 = nfc->third.a1; const float a2 = nfc->third.a2; const float a3 = nfc->third.a3; float z1 = nfc->third.z[0]; float z2 = nfc->third.z[1]; float z3 = nfc->third.z[2]; int i; ASSUME(count > 0); for(i = 0;i < count;i++) { float y = src[i]*gain - a1*z1 - a2*z2; float out = y + b1*z1 + b2*z2; z2 += z1; z1 += y; y = out - a3*z3; out = y + b3*z3; z3 += y; dst[i] = out; } nfc->third.z[0] = z1; nfc->third.z[1] = z2; nfc->third.z[2] = z3; } #if 0 /* Original methods the above are derived from. */ static void NfcFilterCreate(NfcFilter *nfc, const ALsizei order, const float src_dist, const float ctl_dist, const float rate) { static const float B[4][5] = { { }, { 1.0f }, { 3.0f, 3.0f }, { 3.6778f, 6.4595f, 2.3222f }, { 4.2076f, 11.4877f, 5.7924f, 9.1401f } }; float w0 = SPEEDOFSOUNDMETRESPERSEC / (src_dist * rate); float w1 = SPEEDOFSOUNDMETRESPERSEC / (ctl_dist * rate); ALsizei i; float r; nfc->g = 1.0f; nfc->coeffs[0] = 1.0f; /* NOTE: Slight adjustment from the literature to raise the center * frequency a bit (0.5 -> 1.0). */ r = 1.0f * w0; for(i = 0; i < (order-1);i += 2) { float b_10 = B[order][i ] * r; float b_11 = B[order][i+1] * r * r; float g_1 = 1.0f + b_10 + b_11; nfc->b[i] = b_10; nfc->b[i + 1] = b_11; nfc->coeffs[0] *= g_1; nfc->coeffs[i+1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1; nfc->coeffs[i+2] = (4.0f * b_11) / g_1; } if(i < order) { float b_00 = B[order][i] * r; float g_0 = 1.0f + b_00; nfc->b[i] = b_00; nfc->coeffs[0] *= g_0; nfc->coeffs[i+1] = (2.0f * b_00) / g_0; } r = 1.0f * w1; for(i = 0;i < (order-1);i += 2) { float b_10 = B[order][i ] * r; float b_11 = B[order][i+1] * r * r; float g_1 = 1.0f + b_10 + b_11; nfc->g /= g_1; nfc->coeffs[0] /= g_1; nfc->coeffs[order+i+1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1; nfc->coeffs[order+i+2] = (4.0f * b_11) / g_1; } if(i < order) { float b_00 = B[order][i] * r; float g_0 = 1.0f + b_00; nfc->g /= g_0; nfc->coeffs[0] /= g_0; nfc->coeffs[order+i+1] = (2.0f * b_00) / g_0; } for(i = 0; i < MAX_AMBI_ORDER; i++) nfc->history[i] = 0.0f; } static void NfcFilterAdjust(NfcFilter *nfc, const float distance) { int i; nfc->coeffs[0] = nfc->g; for(i = 0;i < (nfc->order-1);i += 2) { float b_10 = nfc->b[i] / distance; float b_11 = nfc->b[i+1] / (distance * distance); float g_1 = 1.0f + b_10 + b_11; nfc->coeffs[0] *= g_1; nfc->coeffs[i+1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1; nfc->coeffs[i+2] = (4.0f * b_11) / g_1; } if(i < nfc->order) { float b_00 = nfc->b[i] / distance; float g_0 = 1.0f + b_00; nfc->coeffs[0] *= g_0; nfc->coeffs[i+1] = (2.0f * b_00) / g_0; } } static float NfcFilterProcess(const float in, NfcFilter *nfc) { int i; float out = in * nfc->coeffs[0]; for(i = 0;i < (nfc->order-1);i += 2) { float y = out - (nfc->coeffs[nfc->order+i+1] * nfc->history[i]) - (nfc->coeffs[nfc->order+i+2] * nfc->history[i+1]) + 1.0e-30f; out = y + (nfc->coeffs[i+1]*nfc->history[i]) + (nfc->coeffs[i+2]*nfc->history[i+1]); nfc->history[i+1] += nfc->history[i]; nfc->history[i] += y; } if(i < nfc->order) { float y = out - (nfc->coeffs[nfc->order+i+1] * nfc->history[i]) + 1.0e-30f; out = y + (nfc->coeffs[i+1] * nfc->history[i]); nfc->history[i] += y; } return out; } #endif openal-soft-openal-soft-1.19.1/Alc/filters/nfc.h000066400000000000000000000030251335774445300213640ustar00rootroot00000000000000#ifndef FILTER_NFC_H #define FILTER_NFC_H struct NfcFilter1 { float base_gain, gain; float b1, a1; float z[1]; }; struct NfcFilter2 { float base_gain, gain; float b1, b2, a1, a2; float z[2]; }; struct NfcFilter3 { float base_gain, gain; float b1, b2, b3, a1, a2, a3; float z[3]; }; typedef struct NfcFilter { struct NfcFilter1 first; struct NfcFilter2 second; struct NfcFilter3 third; } NfcFilter; /* NOTE: * w0 = speed_of_sound / (source_distance * sample_rate); * w1 = speed_of_sound / (control_distance * sample_rate); * * Generally speaking, the control distance should be approximately the average * speaker distance, or based on the reference delay if outputing NFC-HOA. It * must not be negative, 0, or infinite. The source distance should not be too * small relative to the control distance. */ void NfcFilterCreate(NfcFilter *nfc, const float w0, const float w1); void NfcFilterAdjust(NfcFilter *nfc, const float w0); /* Near-field control filter for first-order ambisonic channels (1-3). */ void NfcFilterProcess1(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count); /* Near-field control filter for second-order ambisonic channels (4-8). */ void NfcFilterProcess2(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count); /* Near-field control filter for third-order ambisonic channels (9-15). */ void NfcFilterProcess3(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count); #endif /* FILTER_NFC_H */ openal-soft-openal-soft-1.19.1/Alc/filters/splitter.c000066400000000000000000000044741335774445300224700ustar00rootroot00000000000000 #include "config.h" #include "splitter.h" #include "math_defs.h" void bandsplit_init(BandSplitter *splitter, ALfloat f0norm) { ALfloat w = f0norm * F_TAU; ALfloat cw = cosf(w); if(cw > FLT_EPSILON) splitter->coeff = (sinf(w) - 1.0f) / cw; else splitter->coeff = cw * -0.5f; splitter->lp_z1 = 0.0f; splitter->lp_z2 = 0.0f; splitter->hp_z1 = 0.0f; } void bandsplit_clear(BandSplitter *splitter) { splitter->lp_z1 = 0.0f; splitter->lp_z2 = 0.0f; splitter->hp_z1 = 0.0f; } void bandsplit_process(BandSplitter *splitter, ALfloat *restrict hpout, ALfloat *restrict lpout, const ALfloat *input, ALsizei count) { ALfloat lp_coeff, hp_coeff, lp_y, hp_y, d; ALfloat lp_z1, lp_z2, hp_z1; ALsizei i; ASSUME(count > 0); hp_coeff = splitter->coeff; lp_coeff = splitter->coeff*0.5f + 0.5f; lp_z1 = splitter->lp_z1; lp_z2 = splitter->lp_z2; hp_z1 = splitter->hp_z1; for(i = 0;i < count;i++) { ALfloat in = input[i]; /* Low-pass sample processing. */ d = (in - lp_z1) * lp_coeff; lp_y = lp_z1 + d; lp_z1 = lp_y + d; d = (lp_y - lp_z2) * lp_coeff; lp_y = lp_z2 + d; lp_z2 = lp_y + d; lpout[i] = lp_y; /* All-pass sample processing. */ hp_y = in*hp_coeff + hp_z1; hp_z1 = in - hp_y*hp_coeff; /* High-pass generated from removing low-passed output. */ hpout[i] = hp_y - lp_y; } splitter->lp_z1 = lp_z1; splitter->lp_z2 = lp_z2; splitter->hp_z1 = hp_z1; } void splitterap_init(SplitterAllpass *splitter, ALfloat f0norm) { ALfloat w = f0norm * F_TAU; ALfloat cw = cosf(w); if(cw > FLT_EPSILON) splitter->coeff = (sinf(w) - 1.0f) / cw; else splitter->coeff = cw * -0.5f; splitter->z1 = 0.0f; } void splitterap_clear(SplitterAllpass *splitter) { splitter->z1 = 0.0f; } void splitterap_process(SplitterAllpass *splitter, ALfloat *restrict samples, ALsizei count) { ALfloat coeff, in, out; ALfloat z1; ALsizei i; ASSUME(count > 0); coeff = splitter->coeff; z1 = splitter->z1; for(i = 0;i < count;i++) { in = samples[i]; out = in*coeff + z1; z1 = in - out*coeff; samples[i] = out; } splitter->z1 = z1; } openal-soft-openal-soft-1.19.1/Alc/filters/splitter.h000066400000000000000000000022771335774445300224740ustar00rootroot00000000000000#ifndef FILTER_SPLITTER_H #define FILTER_SPLITTER_H #include "alMain.h" /* Band splitter. Splits a signal into two phase-matching frequency bands. */ typedef struct BandSplitter { ALfloat coeff; ALfloat lp_z1; ALfloat lp_z2; ALfloat hp_z1; } BandSplitter; void bandsplit_init(BandSplitter *splitter, ALfloat f0norm); void bandsplit_clear(BandSplitter *splitter); void bandsplit_process(BandSplitter *splitter, ALfloat *restrict hpout, ALfloat *restrict lpout, const ALfloat *input, ALsizei count); /* The all-pass portion of the band splitter. Applies the same phase shift * without splitting the signal. */ typedef struct SplitterAllpass { ALfloat coeff; ALfloat z1; } SplitterAllpass; void splitterap_init(SplitterAllpass *splitter, ALfloat f0norm); void splitterap_clear(SplitterAllpass *splitter); void splitterap_process(SplitterAllpass *splitter, ALfloat *restrict samples, ALsizei count); typedef struct FrontStablizer { SplitterAllpass APFilter[MAX_OUTPUT_CHANNELS]; BandSplitter LFilter, RFilter; alignas(16) ALfloat LSplit[2][BUFFERSIZE]; alignas(16) ALfloat RSplit[2][BUFFERSIZE]; } FrontStablizer; #endif /* FILTER_SPLITTER_H */ openal-soft-openal-soft-1.19.1/Alc/fpu_modes.h000066400000000000000000000016331335774445300211320ustar00rootroot00000000000000#ifndef FPU_MODES_H #define FPU_MODES_H #ifdef HAVE_FENV_H #include #endif typedef struct FPUCtl { #if defined(__GNUC__) && defined(HAVE_SSE) unsigned int sse_state; #elif defined(HAVE___CONTROL87_2) unsigned int state; unsigned int sse_state; #elif defined(HAVE__CONTROLFP) unsigned int state; #endif } FPUCtl; void SetMixerFPUMode(FPUCtl *ctl); void RestoreFPUMode(const FPUCtl *ctl); #ifdef __GNUC__ /* Use an alternate macro set with GCC to avoid accidental continue or break * statements within the mixer mode. */ #define START_MIXER_MODE() __extension__({ FPUCtl _oldMode; SetMixerFPUMode(&_oldMode) #define END_MIXER_MODE() RestoreFPUMode(&_oldMode); }) #else #define START_MIXER_MODE() do { FPUCtl _oldMode; SetMixerFPUMode(&_oldMode) #define END_MIXER_MODE() RestoreFPUMode(&_oldMode); } while(0) #endif #define LEAVE_MIXER_MODE() RestoreFPUMode(&_oldMode) #endif /* FPU_MODES_H */ openal-soft-openal-soft-1.19.1/Alc/helpers.c000066400000000000000000001002351335774445300206040ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2011 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #ifdef _WIN32 #ifdef __MINGW32__ #define _WIN32_IE 0x501 #else #define _WIN32_IE 0x400 #endif #endif #include "config.h" #include #include #include #include #include #ifdef HAVE_MALLOC_H #include #endif #ifdef HAVE_DIRENT_H #include #endif #ifdef HAVE_PROC_PIDPATH #include #endif #ifdef __FreeBSD__ #include #include #endif #ifndef AL_NO_UID_DEFS #if defined(HAVE_GUIDDEF_H) || defined(HAVE_INITGUID_H) #define INITGUID #include #ifdef HAVE_GUIDDEF_H #include #else #include #endif DEFINE_GUID(KSDATAFORMAT_SUBTYPE_PCM, 0x00000001, 0x0000, 0x0010, 0x80,0x00, 0x00,0xaa,0x00,0x38,0x9b,0x71); DEFINE_GUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, 0x00000003, 0x0000, 0x0010, 0x80,0x00, 0x00,0xaa,0x00,0x38,0x9b,0x71); DEFINE_GUID(IID_IDirectSoundNotify, 0xb0210783, 0x89cd, 0x11d0, 0xaf,0x08, 0x00,0xa0,0xc9,0x25,0xcd,0x16); DEFINE_GUID(CLSID_MMDeviceEnumerator, 0xbcde0395, 0xe52f, 0x467c, 0x8e,0x3d, 0xc4,0x57,0x92,0x91,0x69,0x2e); DEFINE_GUID(IID_IMMDeviceEnumerator, 0xa95664d2, 0x9614, 0x4f35, 0xa7,0x46, 0xde,0x8d,0xb6,0x36,0x17,0xe6); DEFINE_GUID(IID_IAudioClient, 0x1cb9ad4c, 0xdbfa, 0x4c32, 0xb1,0x78, 0xc2,0xf5,0x68,0xa7,0x03,0xb2); DEFINE_GUID(IID_IAudioRenderClient, 0xf294acfc, 0x3146, 0x4483, 0xa7,0xbf, 0xad,0xdc,0xa7,0xc2,0x60,0xe2); DEFINE_GUID(IID_IAudioCaptureClient, 0xc8adbd64, 0xe71e, 0x48a0, 0xa4,0xde, 0x18,0x5c,0x39,0x5c,0xd3,0x17); #ifdef HAVE_WASAPI #include #include #include DEFINE_DEVPROPKEY(DEVPKEY_Device_FriendlyName, 0xa45c254e, 0xdf1c, 0x4efd, 0x80,0x20, 0x67,0xd1,0x46,0xa8,0x50,0xe0, 14); DEFINE_PROPERTYKEY(PKEY_AudioEndpoint_FormFactor, 0x1da5d803, 0xd492, 0x4edd, 0x8c,0x23, 0xe0,0xc0,0xff,0xee,0x7f,0x0e, 0); DEFINE_PROPERTYKEY(PKEY_AudioEndpoint_GUID, 0x1da5d803, 0xd492, 0x4edd, 0x8c, 0x23,0xe0, 0xc0,0xff,0xee,0x7f,0x0e, 4 ); #endif #endif #endif /* AL_NO_UID_DEFS */ #ifdef HAVE_DLFCN_H #include #endif #ifdef HAVE_INTRIN_H #include #endif #ifdef HAVE_CPUID_H #include #endif #ifdef HAVE_SYS_SYSCONF_H #include #endif #ifdef HAVE_FLOAT_H #include #endif #ifdef HAVE_IEEEFP_H #include #endif #ifndef _WIN32 #include #include #include #include #include #elif defined(_WIN32_IE) #include #endif #include "alMain.h" #include "alu.h" #include "cpu_caps.h" #include "fpu_modes.h" #include "atomic.h" #include "uintmap.h" #include "vector.h" #include "alstring.h" #include "compat.h" #include "threads.h" extern inline ALuint NextPowerOf2(ALuint value); extern inline size_t RoundUp(size_t value, size_t r); extern inline ALint fastf2i(ALfloat f); extern inline int float2int(float f); extern inline float fast_roundf(float f); #ifndef __GNUC__ #if defined(HAVE_BITSCANFORWARD64_INTRINSIC) extern inline int msvc64_ctz64(ALuint64 v); #elif defined(HAVE_BITSCANFORWARD_INTRINSIC) extern inline int msvc_ctz64(ALuint64 v); #else extern inline int fallback_popcnt64(ALuint64 v); extern inline int fallback_ctz64(ALuint64 value); #endif #endif #if defined(HAVE_GCC_GET_CPUID) && (defined(__i386__) || defined(__x86_64__) || \ defined(_M_IX86) || defined(_M_X64)) typedef unsigned int reg_type; static inline void get_cpuid(int f, reg_type *regs) { __get_cpuid(f, ®s[0], ®s[1], ®s[2], ®s[3]); } #define CAN_GET_CPUID #elif defined(HAVE_CPUID_INTRINSIC) && (defined(__i386__) || defined(__x86_64__) || \ defined(_M_IX86) || defined(_M_X64)) typedef int reg_type; static inline void get_cpuid(int f, reg_type *regs) { (__cpuid)(regs, f); } #define CAN_GET_CPUID #endif int CPUCapFlags = 0; void FillCPUCaps(int capfilter) { int caps = 0; /* FIXME: We really should get this for all available CPUs in case different * CPUs have different caps (is that possible on one machine?). */ #ifdef CAN_GET_CPUID union { reg_type regs[4]; char str[sizeof(reg_type[4])]; } cpuinf[3] = {{ { 0, 0, 0, 0 } }}; get_cpuid(0, cpuinf[0].regs); if(cpuinf[0].regs[0] == 0) ERR("Failed to get CPUID\n"); else { unsigned int maxfunc = cpuinf[0].regs[0]; unsigned int maxextfunc; get_cpuid(0x80000000, cpuinf[0].regs); maxextfunc = cpuinf[0].regs[0]; TRACE("Detected max CPUID function: 0x%x (ext. 0x%x)\n", maxfunc, maxextfunc); TRACE("Vendor ID: \"%.4s%.4s%.4s\"\n", cpuinf[0].str+4, cpuinf[0].str+12, cpuinf[0].str+8); if(maxextfunc >= 0x80000004) { get_cpuid(0x80000002, cpuinf[0].regs); get_cpuid(0x80000003, cpuinf[1].regs); get_cpuid(0x80000004, cpuinf[2].regs); TRACE("Name: \"%.16s%.16s%.16s\"\n", cpuinf[0].str, cpuinf[1].str, cpuinf[2].str); } if(maxfunc >= 1) { get_cpuid(1, cpuinf[0].regs); if((cpuinf[0].regs[3]&(1<<25))) caps |= CPU_CAP_SSE; if((caps&CPU_CAP_SSE) && (cpuinf[0].regs[3]&(1<<26))) caps |= CPU_CAP_SSE2; if((caps&CPU_CAP_SSE2) && (cpuinf[0].regs[2]&(1<<0))) caps |= CPU_CAP_SSE3; if((caps&CPU_CAP_SSE3) && (cpuinf[0].regs[2]&(1<<19))) caps |= CPU_CAP_SSE4_1; } } #else /* Assume support for whatever's supported if we can't check for it */ #if defined(HAVE_SSE4_1) #warning "Assuming SSE 4.1 run-time support!" caps |= CPU_CAP_SSE | CPU_CAP_SSE2 | CPU_CAP_SSE3 | CPU_CAP_SSE4_1; #elif defined(HAVE_SSE3) #warning "Assuming SSE 3 run-time support!" caps |= CPU_CAP_SSE | CPU_CAP_SSE2 | CPU_CAP_SSE3; #elif defined(HAVE_SSE2) #warning "Assuming SSE 2 run-time support!" caps |= CPU_CAP_SSE | CPU_CAP_SSE2; #elif defined(HAVE_SSE) #warning "Assuming SSE run-time support!" caps |= CPU_CAP_SSE; #endif #endif #ifdef HAVE_NEON FILE *file = fopen("/proc/cpuinfo", "rt"); if(!file) ERR("Failed to open /proc/cpuinfo, cannot check for NEON support\n"); else { al_string features = AL_STRING_INIT_STATIC(); char buf[256]; while(fgets(buf, sizeof(buf), file) != NULL) { if(strncmp(buf, "Features\t:", 10) != 0) continue; alstr_copy_cstr(&features, buf+10); while(VECTOR_BACK(features) != '\n') { if(fgets(buf, sizeof(buf), file) == NULL) break; alstr_append_cstr(&features, buf); } break; } fclose(file); file = NULL; if(!alstr_empty(features)) { const char *str = alstr_get_cstr(features); while(isspace(str[0])) ++str; TRACE("Got features string:%s\n", str); while((str=strstr(str, "neon")) != NULL) { if(isspace(*(str-1)) && (str[4] == 0 || isspace(str[4]))) { caps |= CPU_CAP_NEON; break; } ++str; } } alstr_reset(&features); } #endif TRACE("Extensions:%s%s%s%s%s%s\n", ((capfilter&CPU_CAP_SSE) ? ((caps&CPU_CAP_SSE) ? " +SSE" : " -SSE") : ""), ((capfilter&CPU_CAP_SSE2) ? ((caps&CPU_CAP_SSE2) ? " +SSE2" : " -SSE2") : ""), ((capfilter&CPU_CAP_SSE3) ? ((caps&CPU_CAP_SSE3) ? " +SSE3" : " -SSE3") : ""), ((capfilter&CPU_CAP_SSE4_1) ? ((caps&CPU_CAP_SSE4_1) ? " +SSE4.1" : " -SSE4.1") : ""), ((capfilter&CPU_CAP_NEON) ? ((caps&CPU_CAP_NEON) ? " +NEON" : " -NEON") : ""), ((!capfilter) ? " -none-" : "") ); CPUCapFlags = caps & capfilter; } void SetMixerFPUMode(FPUCtl *ctl) { #if defined(__GNUC__) && defined(HAVE_SSE) if((CPUCapFlags&CPU_CAP_SSE)) { __asm__ __volatile__("stmxcsr %0" : "=m" (*&ctl->sse_state)); unsigned int sseState = ctl->sse_state; sseState |= 0x8000; /* set flush-to-zero */ if((CPUCapFlags&CPU_CAP_SSE2)) sseState |= 0x0040; /* set denormals-are-zero */ __asm__ __volatile__("ldmxcsr %0" : : "m" (*&sseState)); } #elif defined(HAVE___CONTROL87_2) __control87_2(0, 0, &ctl->state, &ctl->sse_state); _control87(_DN_FLUSH, _MCW_DN); #elif defined(HAVE__CONTROLFP) ctl->state = _controlfp(0, 0); _controlfp(_DN_FLUSH, _MCW_DN); #endif } void RestoreFPUMode(const FPUCtl *ctl) { #if defined(__GNUC__) && defined(HAVE_SSE) if((CPUCapFlags&CPU_CAP_SSE)) __asm__ __volatile__("ldmxcsr %0" : : "m" (*&ctl->sse_state)); #elif defined(HAVE___CONTROL87_2) int mode; __control87_2(ctl->state, _MCW_DN, &mode, NULL); __control87_2(ctl->sse_state, _MCW_DN, NULL, &mode); #elif defined(HAVE__CONTROLFP) _controlfp(ctl->state, _MCW_DN); #endif } static int StringSortCompare(const void *str1, const void *str2) { return alstr_cmp(*(const_al_string*)str1, *(const_al_string*)str2); } #ifdef _WIN32 static WCHAR *strrchrW(WCHAR *str, WCHAR ch) { WCHAR *ret = NULL; while(*str) { if(*str == ch) ret = str; ++str; } return ret; } void GetProcBinary(al_string *path, al_string *fname) { WCHAR *pathname, *sep; DWORD pathlen; DWORD len; pathlen = 256; pathname = malloc(pathlen * sizeof(pathname[0])); while(pathlen > 0 && (len=GetModuleFileNameW(NULL, pathname, pathlen)) == pathlen) { free(pathname); pathlen <<= 1; pathname = malloc(pathlen * sizeof(pathname[0])); } if(len == 0) { free(pathname); ERR("Failed to get process name: error %lu\n", GetLastError()); return; } pathname[len] = 0; if((sep=strrchrW(pathname, '\\')) != NULL) { WCHAR *sep2 = strrchrW(sep+1, '/'); if(sep2) sep = sep2; } else sep = strrchrW(pathname, '/'); if(sep) { if(path) alstr_copy_wrange(path, pathname, sep); if(fname) alstr_copy_wcstr(fname, sep+1); } else { if(path) alstr_clear(path); if(fname) alstr_copy_wcstr(fname, pathname); } free(pathname); if(path && fname) TRACE("Got: %s, %s\n", alstr_get_cstr(*path), alstr_get_cstr(*fname)); else if(path) TRACE("Got path: %s\n", alstr_get_cstr(*path)); else if(fname) TRACE("Got filename: %s\n", alstr_get_cstr(*fname)); } static WCHAR *FromUTF8(const char *str) { WCHAR *out = NULL; int len; if((len=MultiByteToWideChar(CP_UTF8, 0, str, -1, NULL, 0)) > 0) { out = calloc(sizeof(WCHAR), len); MultiByteToWideChar(CP_UTF8, 0, str, -1, out, len); } return out; } void *LoadLib(const char *name) { HANDLE hdl = NULL; WCHAR *wname; wname = FromUTF8(name); if(!wname) ERR("Failed to convert UTF-8 filename: \"%s\"\n", name); else { hdl = LoadLibraryW(wname); free(wname); } return hdl; } void CloseLib(void *handle) { FreeLibrary((HANDLE)handle); } void *GetSymbol(void *handle, const char *name) { void *ret; ret = (void*)GetProcAddress((HANDLE)handle, name); if(ret == NULL) ERR("Failed to load %s\n", name); return ret; } WCHAR *strdupW(const WCHAR *str) { const WCHAR *n; WCHAR *ret; size_t len; n = str; while(*n) n++; len = n - str; ret = calloc(sizeof(WCHAR), len+1); if(ret != NULL) memcpy(ret, str, sizeof(WCHAR)*len); return ret; } FILE *al_fopen(const char *fname, const char *mode) { WCHAR *wname=NULL, *wmode=NULL; FILE *file = NULL; wname = FromUTF8(fname); wmode = FromUTF8(mode); if(!wname) ERR("Failed to convert UTF-8 filename: \"%s\"\n", fname); else if(!wmode) ERR("Failed to convert UTF-8 mode: \"%s\"\n", mode); else file = _wfopen(wname, wmode); free(wname); free(wmode); return file; } void al_print(const char *type, const char *func, const char *fmt, ...) { char str[1024]; WCHAR *wstr; va_list ap; va_start(ap, fmt); vsnprintf(str, sizeof(str), fmt, ap); va_end(ap); str[sizeof(str)-1] = 0; wstr = FromUTF8(str); if(!wstr) fprintf(LogFile, "AL lib: %s %s: %s", type, func, str); else { fprintf(LogFile, "AL lib: %s %s: %ls", type, func, wstr); free(wstr); wstr = NULL; } fflush(LogFile); } static inline int is_slash(int c) { return (c == '\\' || c == '/'); } static void DirectorySearch(const char *path, const char *ext, vector_al_string *results) { al_string pathstr = AL_STRING_INIT_STATIC(); WIN32_FIND_DATAW fdata; WCHAR *wpath; HANDLE hdl; alstr_copy_cstr(&pathstr, path); alstr_append_cstr(&pathstr, "\\*"); alstr_append_cstr(&pathstr, ext); TRACE("Searching %s\n", alstr_get_cstr(pathstr)); wpath = FromUTF8(alstr_get_cstr(pathstr)); hdl = FindFirstFileW(wpath, &fdata); if(hdl != INVALID_HANDLE_VALUE) { size_t base = VECTOR_SIZE(*results); do { al_string str = AL_STRING_INIT_STATIC(); alstr_copy_cstr(&str, path); alstr_append_char(&str, '\\'); alstr_append_wcstr(&str, fdata.cFileName); TRACE("Got result %s\n", alstr_get_cstr(str)); VECTOR_PUSH_BACK(*results, str); } while(FindNextFileW(hdl, &fdata)); FindClose(hdl); if(VECTOR_SIZE(*results) > base) qsort(VECTOR_BEGIN(*results)+base, VECTOR_SIZE(*results)-base, sizeof(VECTOR_FRONT(*results)), StringSortCompare); } free(wpath); alstr_reset(&pathstr); } vector_al_string SearchDataFiles(const char *ext, const char *subdir) { static const int ids[2] = { CSIDL_APPDATA, CSIDL_COMMON_APPDATA }; static RefCount search_lock; vector_al_string results = VECTOR_INIT_STATIC(); size_t i; while(ATOMIC_EXCHANGE_SEQ(&search_lock, 1) == 1) althrd_yield(); /* If the path is absolute, use it directly. */ if(isalpha(subdir[0]) && subdir[1] == ':' && is_slash(subdir[2])) { al_string path = AL_STRING_INIT_STATIC(); alstr_copy_cstr(&path, subdir); #define FIX_SLASH(i) do { if(*(i) == '/') *(i) = '\\'; } while(0) VECTOR_FOR_EACH(char, path, FIX_SLASH); #undef FIX_SLASH DirectorySearch(alstr_get_cstr(path), ext, &results); alstr_reset(&path); } else if(subdir[0] == '\\' && subdir[1] == '\\' && subdir[2] == '?' && subdir[3] == '\\') DirectorySearch(subdir, ext, &results); else { al_string path = AL_STRING_INIT_STATIC(); WCHAR *cwdbuf; /* Search the app-local directory. */ if((cwdbuf=_wgetenv(L"ALSOFT_LOCAL_PATH")) && *cwdbuf != '\0') { alstr_copy_wcstr(&path, cwdbuf); if(is_slash(VECTOR_BACK(path))) { VECTOR_POP_BACK(path); *VECTOR_END(path) = 0; } } else if(!(cwdbuf=_wgetcwd(NULL, 0))) alstr_copy_cstr(&path, "."); else { alstr_copy_wcstr(&path, cwdbuf); if(is_slash(VECTOR_BACK(path))) { VECTOR_POP_BACK(path); *VECTOR_END(path) = 0; } free(cwdbuf); } #define FIX_SLASH(i) do { if(*(i) == '/') *(i) = '\\'; } while(0) VECTOR_FOR_EACH(char, path, FIX_SLASH); #undef FIX_SLASH DirectorySearch(alstr_get_cstr(path), ext, &results); /* Search the local and global data dirs. */ for(i = 0;i < COUNTOF(ids);i++) { WCHAR buffer[MAX_PATH]; if(SHGetSpecialFolderPathW(NULL, buffer, ids[i], FALSE) != FALSE) { alstr_copy_wcstr(&path, buffer); if(!is_slash(VECTOR_BACK(path))) alstr_append_char(&path, '\\'); alstr_append_cstr(&path, subdir); #define FIX_SLASH(i) do { if(*(i) == '/') *(i) = '\\'; } while(0) VECTOR_FOR_EACH(char, path, FIX_SLASH); #undef FIX_SLASH DirectorySearch(alstr_get_cstr(path), ext, &results); } } alstr_reset(&path); } ATOMIC_STORE_SEQ(&search_lock, 0); return results; } struct FileMapping MapFileToMem(const char *fname) { struct FileMapping ret = { NULL, NULL, NULL, 0 }; MEMORY_BASIC_INFORMATION meminfo; HANDLE file, fmap; WCHAR *wname; void *ptr; wname = FromUTF8(fname); file = CreateFileW(wname, GENERIC_READ, FILE_SHARE_READ, NULL, OPEN_EXISTING, FILE_ATTRIBUTE_NORMAL, NULL); if(file == INVALID_HANDLE_VALUE) { ERR("Failed to open %s: %lu\n", fname, GetLastError()); free(wname); return ret; } free(wname); wname = NULL; fmap = CreateFileMappingW(file, NULL, PAGE_READONLY, 0, 0, NULL); if(!fmap) { ERR("Failed to create map for %s: %lu\n", fname, GetLastError()); CloseHandle(file); return ret; } ptr = MapViewOfFile(fmap, FILE_MAP_READ, 0, 0, 0); if(!ptr) { ERR("Failed to map %s: %lu\n", fname, GetLastError()); CloseHandle(fmap); CloseHandle(file); return ret; } if(VirtualQuery(ptr, &meminfo, sizeof(meminfo)) != sizeof(meminfo)) { ERR("Failed to get map size for %s: %lu\n", fname, GetLastError()); UnmapViewOfFile(ptr); CloseHandle(fmap); CloseHandle(file); return ret; } ret.file = file; ret.fmap = fmap; ret.ptr = ptr; ret.len = meminfo.RegionSize; return ret; } void UnmapFileMem(const struct FileMapping *mapping) { UnmapViewOfFile(mapping->ptr); CloseHandle(mapping->fmap); CloseHandle(mapping->file); } #else void GetProcBinary(al_string *path, al_string *fname) { char *pathname = NULL; size_t pathlen; #ifdef __FreeBSD__ int mib[4] = { CTL_KERN, KERN_PROC, KERN_PROC_PATHNAME, -1 }; if(sysctl(mib, 4, NULL, &pathlen, NULL, 0) == -1) WARN("Failed to sysctl kern.proc.pathname: %s\n", strerror(errno)); else { pathname = malloc(pathlen + 1); sysctl(mib, 4, (void*)pathname, &pathlen, NULL, 0); pathname[pathlen] = 0; } #endif #ifdef HAVE_PROC_PIDPATH if(!pathname) { const pid_t pid = getpid(); char procpath[PROC_PIDPATHINFO_MAXSIZE]; int ret; ret = proc_pidpath(pid, procpath, sizeof(procpath)); if(ret < 1) { WARN("proc_pidpath(%d, ...) failed: %s\n", pid, strerror(errno)); free(pathname); pathname = NULL; } else { pathlen = strlen(procpath); pathname = strdup(procpath); } } #endif if(!pathname) { const char *selfname; ssize_t len; pathlen = 256; pathname = malloc(pathlen); selfname = "/proc/self/exe"; len = readlink(selfname, pathname, pathlen); if(len == -1 && errno == ENOENT) { selfname = "/proc/self/file"; len = readlink(selfname, pathname, pathlen); } if(len == -1 && errno == ENOENT) { selfname = "/proc/curproc/exe"; len = readlink(selfname, pathname, pathlen); } if(len == -1 && errno == ENOENT) { selfname = "/proc/curproc/file"; len = readlink(selfname, pathname, pathlen); } while(len > 0 && (size_t)len == pathlen) { free(pathname); pathlen <<= 1; pathname = malloc(pathlen); len = readlink(selfname, pathname, pathlen); } if(len <= 0) { free(pathname); WARN("Failed to readlink %s: %s\n", selfname, strerror(errno)); return; } pathname[len] = 0; } char *sep = strrchr(pathname, '/'); if(sep) { if(path) alstr_copy_range(path, pathname, sep); if(fname) alstr_copy_cstr(fname, sep+1); } else { if(path) alstr_clear(path); if(fname) alstr_copy_cstr(fname, pathname); } free(pathname); if(path && fname) TRACE("Got: %s, %s\n", alstr_get_cstr(*path), alstr_get_cstr(*fname)); else if(path) TRACE("Got path: %s\n", alstr_get_cstr(*path)); else if(fname) TRACE("Got filename: %s\n", alstr_get_cstr(*fname)); } #ifdef HAVE_DLFCN_H void *LoadLib(const char *name) { const char *err; void *handle; dlerror(); handle = dlopen(name, RTLD_NOW); if((err=dlerror()) != NULL) handle = NULL; return handle; } void CloseLib(void *handle) { dlclose(handle); } void *GetSymbol(void *handle, const char *name) { const char *err; void *sym; dlerror(); sym = dlsym(handle, name); if((err=dlerror()) != NULL) { WARN("Failed to load %s: %s\n", name, err); sym = NULL; } return sym; } #endif /* HAVE_DLFCN_H */ void al_print(const char *type, const char *func, const char *fmt, ...) { va_list ap; va_start(ap, fmt); fprintf(LogFile, "AL lib: %s %s: ", type, func); vfprintf(LogFile, fmt, ap); va_end(ap); fflush(LogFile); } static void DirectorySearch(const char *path, const char *ext, vector_al_string *results) { size_t extlen = strlen(ext); DIR *dir; TRACE("Searching %s for *%s\n", path, ext); dir = opendir(path); if(dir != NULL) { size_t base = VECTOR_SIZE(*results); struct dirent *dirent; while((dirent=readdir(dir)) != NULL) { al_string str; size_t len; if(strcmp(dirent->d_name, ".") == 0 || strcmp(dirent->d_name, "..") == 0) continue; len = strlen(dirent->d_name); if(!(len > extlen)) continue; if(strcasecmp(dirent->d_name+len-extlen, ext) != 0) continue; AL_STRING_INIT(str); alstr_copy_cstr(&str, path); if(VECTOR_BACK(str) != '/') alstr_append_char(&str, '/'); alstr_append_cstr(&str, dirent->d_name); TRACE("Got result %s\n", alstr_get_cstr(str)); VECTOR_PUSH_BACK(*results, str); } closedir(dir); if(VECTOR_SIZE(*results) > base) qsort(VECTOR_BEGIN(*results)+base, VECTOR_SIZE(*results)-base, sizeof(VECTOR_FRONT(*results)), StringSortCompare); } } vector_al_string SearchDataFiles(const char *ext, const char *subdir) { static RefCount search_lock; vector_al_string results = VECTOR_INIT_STATIC(); while(ATOMIC_EXCHANGE_SEQ(&search_lock, 1) == 1) althrd_yield(); if(subdir[0] == '/') DirectorySearch(subdir, ext, &results); else { al_string path = AL_STRING_INIT_STATIC(); const char *str, *next; /* Search the app-local directory. */ if((str=getenv("ALSOFT_LOCAL_PATH")) && *str != '\0') DirectorySearch(str, ext, &results); else { size_t cwdlen = 256; char *cwdbuf = malloc(cwdlen); while(!getcwd(cwdbuf, cwdlen)) { free(cwdbuf); cwdbuf = NULL; if(errno != ERANGE) break; cwdlen <<= 1; cwdbuf = malloc(cwdlen); } if(!cwdbuf) DirectorySearch(".", ext, &results); else { DirectorySearch(cwdbuf, ext, &results); free(cwdbuf); cwdbuf = NULL; } } // Search local data dir if((str=getenv("XDG_DATA_HOME")) != NULL && str[0] != '\0') { alstr_copy_cstr(&path, str); if(VECTOR_BACK(path) != '/') alstr_append_char(&path, '/'); alstr_append_cstr(&path, subdir); DirectorySearch(alstr_get_cstr(path), ext, &results); } else if((str=getenv("HOME")) != NULL && str[0] != '\0') { alstr_copy_cstr(&path, str); if(VECTOR_BACK(path) == '/') { VECTOR_POP_BACK(path); *VECTOR_END(path) = 0; } alstr_append_cstr(&path, "/.local/share/"); alstr_append_cstr(&path, subdir); DirectorySearch(alstr_get_cstr(path), ext, &results); } // Search global data dirs if((str=getenv("XDG_DATA_DIRS")) == NULL || str[0] == '\0') str = "/usr/local/share/:/usr/share/"; next = str; while((str=next) != NULL && str[0] != '\0') { next = strchr(str, ':'); if(!next) alstr_copy_cstr(&path, str); else { alstr_copy_range(&path, str, next); ++next; } if(!alstr_empty(path)) { if(VECTOR_BACK(path) != '/') alstr_append_char(&path, '/'); alstr_append_cstr(&path, subdir); DirectorySearch(alstr_get_cstr(path), ext, &results); } } alstr_reset(&path); } ATOMIC_STORE_SEQ(&search_lock, 0); return results; } struct FileMapping MapFileToMem(const char *fname) { struct FileMapping ret = { -1, NULL, 0 }; struct stat sbuf; void *ptr; int fd; fd = open(fname, O_RDONLY, 0); if(fd == -1) { ERR("Failed to open %s: (%d) %s\n", fname, errno, strerror(errno)); return ret; } if(fstat(fd, &sbuf) == -1) { ERR("Failed to stat %s: (%d) %s\n", fname, errno, strerror(errno)); close(fd); return ret; } ptr = mmap(NULL, sbuf.st_size, PROT_READ, MAP_PRIVATE, fd, 0); if(ptr == MAP_FAILED) { ERR("Failed to map %s: (%d) %s\n", fname, errno, strerror(errno)); close(fd); return ret; } ret.fd = fd; ret.ptr = ptr; ret.len = sbuf.st_size; return ret; } void UnmapFileMem(const struct FileMapping *mapping) { munmap(mapping->ptr, mapping->len); close(mapping->fd); } #endif void SetRTPriority(void) { ALboolean failed = AL_FALSE; #ifdef _WIN32 if(RTPrioLevel > 0) failed = !SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_TIME_CRITICAL); #elif defined(HAVE_PTHREAD_SETSCHEDPARAM) && !defined(__OpenBSD__) if(RTPrioLevel > 0) { struct sched_param param; /* Use the minimum real-time priority possible for now (on Linux this * should be 1 for SCHED_RR) */ param.sched_priority = sched_get_priority_min(SCHED_RR); failed = !!pthread_setschedparam(pthread_self(), SCHED_RR, ¶m); } #else /* Real-time priority not available */ failed = (RTPrioLevel>0); #endif if(failed) ERR("Failed to set priority level for thread\n"); } extern inline void alstr_reset(al_string *str); extern inline size_t alstr_length(const_al_string str); extern inline ALboolean alstr_empty(const_al_string str); extern inline const al_string_char_type *alstr_get_cstr(const_al_string str); void alstr_clear(al_string *str) { if(!alstr_empty(*str)) { /* Reserve one more character than the total size of the string. This * is to ensure we have space to add a null terminator in the string * data so it can be used as a C-style string. */ VECTOR_RESIZE(*str, 0, 1); VECTOR_ELEM(*str, 0) = 0; } } static inline int alstr_compare(const al_string_char_type *str1, size_t str1len, const al_string_char_type *str2, size_t str2len) { size_t complen = (str1len < str2len) ? str1len : str2len; int ret = memcmp(str1, str2, complen); if(ret == 0) { if(str1len > str2len) return 1; if(str1len < str2len) return -1; } return ret; } int alstr_cmp(const_al_string str1, const_al_string str2) { return alstr_compare(&VECTOR_FRONT(str1), alstr_length(str1), &VECTOR_FRONT(str2), alstr_length(str2)); } int alstr_cmp_cstr(const_al_string str1, const al_string_char_type *str2) { return alstr_compare(&VECTOR_FRONT(str1), alstr_length(str1), str2, strlen(str2)); } void alstr_copy(al_string *str, const_al_string from) { size_t len = alstr_length(from); size_t i; VECTOR_RESIZE(*str, len, len+1); for(i = 0;i < len;i++) VECTOR_ELEM(*str, i) = VECTOR_ELEM(from, i); VECTOR_ELEM(*str, i) = 0; } void alstr_copy_cstr(al_string *str, const al_string_char_type *from) { size_t len = strlen(from); size_t i; VECTOR_RESIZE(*str, len, len+1); for(i = 0;i < len;i++) VECTOR_ELEM(*str, i) = from[i]; VECTOR_ELEM(*str, i) = 0; } void alstr_copy_range(al_string *str, const al_string_char_type *from, const al_string_char_type *to) { size_t len = to - from; size_t i; VECTOR_RESIZE(*str, len, len+1); for(i = 0;i < len;i++) VECTOR_ELEM(*str, i) = from[i]; VECTOR_ELEM(*str, i) = 0; } void alstr_append_char(al_string *str, const al_string_char_type c) { size_t len = alstr_length(*str); VECTOR_RESIZE(*str, len+1, len+2); VECTOR_BACK(*str) = c; VECTOR_ELEM(*str, len+1) = 0; } void alstr_append_cstr(al_string *str, const al_string_char_type *from) { size_t len = strlen(from); if(len != 0) { size_t base = alstr_length(*str); size_t i; VECTOR_RESIZE(*str, base+len, base+len+1); for(i = 0;i < len;i++) VECTOR_ELEM(*str, base+i) = from[i]; VECTOR_ELEM(*str, base+i) = 0; } } void alstr_append_range(al_string *str, const al_string_char_type *from, const al_string_char_type *to) { size_t len = to - from; if(len != 0) { size_t base = alstr_length(*str); size_t i; VECTOR_RESIZE(*str, base+len, base+len+1); for(i = 0;i < len;i++) VECTOR_ELEM(*str, base+i) = from[i]; VECTOR_ELEM(*str, base+i) = 0; } } #ifdef _WIN32 void alstr_copy_wcstr(al_string *str, const wchar_t *from) { int len; if((len=WideCharToMultiByte(CP_UTF8, 0, from, -1, NULL, 0, NULL, NULL)) > 0) { VECTOR_RESIZE(*str, len-1, len); WideCharToMultiByte(CP_UTF8, 0, from, -1, &VECTOR_FRONT(*str), len, NULL, NULL); VECTOR_ELEM(*str, len-1) = 0; } } void alstr_append_wcstr(al_string *str, const wchar_t *from) { int len; if((len=WideCharToMultiByte(CP_UTF8, 0, from, -1, NULL, 0, NULL, NULL)) > 0) { size_t base = alstr_length(*str); VECTOR_RESIZE(*str, base+len-1, base+len); WideCharToMultiByte(CP_UTF8, 0, from, -1, &VECTOR_ELEM(*str, base), len, NULL, NULL); VECTOR_ELEM(*str, base+len-1) = 0; } } void alstr_copy_wrange(al_string *str, const wchar_t *from, const wchar_t *to) { int len; if((len=WideCharToMultiByte(CP_UTF8, 0, from, (int)(to-from), NULL, 0, NULL, NULL)) > 0) { VECTOR_RESIZE(*str, len, len+1); WideCharToMultiByte(CP_UTF8, 0, from, (int)(to-from), &VECTOR_FRONT(*str), len+1, NULL, NULL); VECTOR_ELEM(*str, len) = 0; } } void alstr_append_wrange(al_string *str, const wchar_t *from, const wchar_t *to) { int len; if((len=WideCharToMultiByte(CP_UTF8, 0, from, (int)(to-from), NULL, 0, NULL, NULL)) > 0) { size_t base = alstr_length(*str); VECTOR_RESIZE(*str, base+len, base+len+1); WideCharToMultiByte(CP_UTF8, 0, from, (int)(to-from), &VECTOR_ELEM(*str, base), len+1, NULL, NULL); VECTOR_ELEM(*str, base+len) = 0; } } #endif openal-soft-openal-soft-1.19.1/Alc/hrtf.c000066400000000000000000001261701335774445300201130ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2011 by Chris Robinson * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include "AL/al.h" #include "AL/alc.h" #include "alMain.h" #include "alSource.h" #include "alu.h" #include "hrtf.h" #include "alconfig.h" #include "filters/splitter.h" #include "compat.h" #include "almalloc.h" /* Current data set limits defined by the makehrtf utility. */ #define MIN_IR_SIZE (8) #define MAX_IR_SIZE (512) #define MOD_IR_SIZE (8) #define MIN_FD_COUNT (1) #define MAX_FD_COUNT (16) #define MIN_FD_DISTANCE (50) #define MAX_FD_DISTANCE (2500) #define MIN_EV_COUNT (5) #define MAX_EV_COUNT (128) #define MIN_AZ_COUNT (1) #define MAX_AZ_COUNT (128) #define MAX_HRIR_DELAY (HRTF_HISTORY_LENGTH-1) struct HrtfEntry { struct HrtfEntry *next; struct Hrtf *handle; char filename[]; }; static const ALchar magicMarker00[8] = "MinPHR00"; static const ALchar magicMarker01[8] = "MinPHR01"; static const ALchar magicMarker02[8] = "MinPHR02"; /* First value for pass-through coefficients (remaining are 0), used for omni- * directional sounds. */ static const ALfloat PassthruCoeff = 0.707106781187f/*sqrt(0.5)*/; static ATOMIC_FLAG LoadedHrtfLock = ATOMIC_FLAG_INIT; static struct HrtfEntry *LoadedHrtfs = NULL; /* Calculate the elevation index given the polar elevation in radians. This * will return an index between 0 and (evcount - 1). */ static ALsizei CalcEvIndex(ALsizei evcount, ALfloat ev, ALfloat *mu) { ALsizei idx; ev = (F_PI_2+ev) * (evcount-1) / F_PI; idx = float2int(ev); *mu = ev - idx; return mini(idx, evcount-1); } /* Calculate the azimuth index given the polar azimuth in radians. This will * return an index between 0 and (azcount - 1). */ static ALsizei CalcAzIndex(ALsizei azcount, ALfloat az, ALfloat *mu) { ALsizei idx; az = (F_TAU+az) * azcount / F_TAU; idx = float2int(az); *mu = az - idx; return idx % azcount; } /* Calculates static HRIR coefficients and delays for the given polar elevation * and azimuth in radians. The coefficients are normalized. */ void GetHrtfCoeffs(const struct Hrtf *Hrtf, ALfloat elevation, ALfloat azimuth, ALfloat spread, ALfloat (*restrict coeffs)[2], ALsizei *delays) { ALsizei evidx, azidx, idx[4]; ALsizei evoffset; ALfloat emu, amu[2]; ALfloat blend[4]; ALfloat dirfact; ALsizei i, c; dirfact = 1.0f - (spread / F_TAU); /* Claculate the lower elevation index. */ evidx = CalcEvIndex(Hrtf->evCount, elevation, &emu); evoffset = Hrtf->evOffset[evidx]; /* Calculate lower azimuth index. */ azidx= CalcAzIndex(Hrtf->azCount[evidx], azimuth, &amu[0]); /* Calculate the lower HRIR indices. */ idx[0] = evoffset + azidx; idx[1] = evoffset + ((azidx+1) % Hrtf->azCount[evidx]); if(evidx < Hrtf->evCount-1) { /* Increment elevation to the next (upper) index. */ evidx++; evoffset = Hrtf->evOffset[evidx]; /* Calculate upper azimuth index. */ azidx = CalcAzIndex(Hrtf->azCount[evidx], azimuth, &amu[1]); /* Calculate the upper HRIR indices. */ idx[2] = evoffset + azidx; idx[3] = evoffset + ((azidx+1) % Hrtf->azCount[evidx]); } else { /* If the lower elevation is the top index, the upper elevation is the * same as the lower. */ amu[1] = amu[0]; idx[2] = idx[0]; idx[3] = idx[1]; } /* Calculate bilinear blending weights, attenuated according to the * directional panning factor. */ blend[0] = (1.0f-emu) * (1.0f-amu[0]) * dirfact; blend[1] = (1.0f-emu) * ( amu[0]) * dirfact; blend[2] = ( emu) * (1.0f-amu[1]) * dirfact; blend[3] = ( emu) * ( amu[1]) * dirfact; /* Calculate the blended HRIR delays. */ delays[0] = fastf2i( Hrtf->delays[idx[0]][0]*blend[0] + Hrtf->delays[idx[1]][0]*blend[1] + Hrtf->delays[idx[2]][0]*blend[2] + Hrtf->delays[idx[3]][0]*blend[3] ); delays[1] = fastf2i( Hrtf->delays[idx[0]][1]*blend[0] + Hrtf->delays[idx[1]][1]*blend[1] + Hrtf->delays[idx[2]][1]*blend[2] + Hrtf->delays[idx[3]][1]*blend[3] ); /* Calculate the sample offsets for the HRIR indices. */ idx[0] *= Hrtf->irSize; idx[1] *= Hrtf->irSize; idx[2] *= Hrtf->irSize; idx[3] *= Hrtf->irSize; ASSUME(Hrtf->irSize >= MIN_IR_SIZE && (Hrtf->irSize%MOD_IR_SIZE) == 0); coeffs = ASSUME_ALIGNED(coeffs, 16); /* Calculate the blended HRIR coefficients. */ coeffs[0][0] = PassthruCoeff * (1.0f-dirfact); coeffs[0][1] = PassthruCoeff * (1.0f-dirfact); for(i = 1;i < Hrtf->irSize;i++) { coeffs[i][0] = 0.0f; coeffs[i][1] = 0.0f; } for(c = 0;c < 4;c++) { const ALfloat (*restrict srccoeffs)[2] = ASSUME_ALIGNED(Hrtf->coeffs+idx[c], 16); for(i = 0;i < Hrtf->irSize;i++) { coeffs[i][0] += srccoeffs[i][0] * blend[c]; coeffs[i][1] += srccoeffs[i][1] * blend[c]; } } } void BuildBFormatHrtf(const struct Hrtf *Hrtf, DirectHrtfState *state, ALsizei NumChannels, const struct AngularPoint *AmbiPoints, const ALfloat (*restrict AmbiMatrix)[MAX_AMBI_COEFFS], ALsizei AmbiCount, const ALfloat *restrict AmbiOrderHFGain) { /* Set this to 2 for dual-band HRTF processing. May require a higher quality * band-splitter, or better calculation of the new IR length to deal with the * tail generated by the filter. */ #define NUM_BANDS 2 BandSplitter splitter; ALdouble (*tmpres)[HRIR_LENGTH][2]; ALsizei *restrict idx; ALsizei min_delay = HRTF_HISTORY_LENGTH; ALsizei max_delay = 0; ALfloat temps[3][HRIR_LENGTH]; ALsizei max_length; ALsizei i, c, b; idx = al_calloc(DEF_ALIGN, AmbiCount*sizeof(*idx)); for(c = 0;c < AmbiCount;c++) { ALuint evidx, azidx; ALuint evoffset; ALuint azcount; /* Calculate elevation index. */ evidx = (ALsizei)((F_PI_2+AmbiPoints[c].Elev) * (Hrtf->evCount-1) / F_PI + 0.5f); evidx = clampi(evidx, 0, Hrtf->evCount-1); azcount = Hrtf->azCount[evidx]; evoffset = Hrtf->evOffset[evidx]; /* Calculate azimuth index for this elevation. */ azidx = (ALsizei)((F_TAU+AmbiPoints[c].Azim) * azcount / F_TAU + 0.5f) % azcount; /* Calculate indices for left and right channels. */ idx[c] = evoffset + azidx; min_delay = mini(min_delay, mini(Hrtf->delays[idx[c]][0], Hrtf->delays[idx[c]][1])); max_delay = maxi(max_delay, maxi(Hrtf->delays[idx[c]][0], Hrtf->delays[idx[c]][1])); } tmpres = al_calloc(16, NumChannels * sizeof(*tmpres)); memset(temps, 0, sizeof(temps)); bandsplit_init(&splitter, 400.0f / (ALfloat)Hrtf->sampleRate); for(c = 0;c < AmbiCount;c++) { const ALfloat (*fir)[2] = &Hrtf->coeffs[idx[c] * Hrtf->irSize]; ALsizei ldelay = Hrtf->delays[idx[c]][0] - min_delay; ALsizei rdelay = Hrtf->delays[idx[c]][1] - min_delay; if(NUM_BANDS == 1) { for(i = 0;i < NumChannels;++i) { ALdouble mult = (ALdouble)AmbiOrderHFGain[(ALsizei)sqrt(i)] * AmbiMatrix[c][i]; ALsizei lidx = ldelay, ridx = rdelay; ALsizei j = 0; while(lidx < HRIR_LENGTH && ridx < HRIR_LENGTH && j < Hrtf->irSize) { tmpres[i][lidx++][0] += fir[j][0] * mult; tmpres[i][ridx++][1] += fir[j][1] * mult; j++; } } } else { /* Band-split left HRIR into low and high frequency responses. */ bandsplit_clear(&splitter); for(i = 0;i < Hrtf->irSize;i++) temps[2][i] = fir[i][0]; bandsplit_process(&splitter, temps[0], temps[1], temps[2], HRIR_LENGTH); /* Apply left ear response with delay. */ for(i = 0;i < NumChannels;++i) { ALfloat hfgain = AmbiOrderHFGain[(ALsizei)sqrt(i)]; for(b = 0;b < NUM_BANDS;b++) { ALdouble mult = AmbiMatrix[c][i] * (ALdouble)((b==0) ? hfgain : 1.0); ALsizei lidx = ldelay; ALsizei j = 0; while(lidx < HRIR_LENGTH) tmpres[i][lidx++][0] += temps[b][j++] * mult; } } /* Band-split right HRIR into low and high frequency responses. */ bandsplit_clear(&splitter); for(i = 0;i < Hrtf->irSize;i++) temps[2][i] = fir[i][1]; bandsplit_process(&splitter, temps[0], temps[1], temps[2], HRIR_LENGTH); /* Apply right ear response with delay. */ for(i = 0;i < NumChannels;++i) { ALfloat hfgain = AmbiOrderHFGain[(ALsizei)sqrt(i)]; for(b = 0;b < NUM_BANDS;b++) { ALdouble mult = AmbiMatrix[c][i] * (ALdouble)((b==0) ? hfgain : 1.0); ALsizei ridx = rdelay; ALsizei j = 0; while(ridx < HRIR_LENGTH) tmpres[i][ridx++][1] += temps[b][j++] * mult; } } } } for(i = 0;i < NumChannels;++i) { int idx; for(idx = 0;idx < HRIR_LENGTH;idx++) { state->Chan[i].Coeffs[idx][0] = (ALfloat)tmpres[i][idx][0]; state->Chan[i].Coeffs[idx][1] = (ALfloat)tmpres[i][idx][1]; } } al_free(tmpres); tmpres = NULL; al_free(idx); idx = NULL; if(NUM_BANDS == 1) max_length = mini(max_delay-min_delay + Hrtf->irSize, HRIR_LENGTH); else { /* Increase the IR size by 2/3rds to account for the tail generated by * the band-split filter. */ const ALsizei irsize = mini(Hrtf->irSize*5/3, HRIR_LENGTH); max_length = mini(max_delay-min_delay + irsize, HRIR_LENGTH); } /* Round up to the next IR size multiple. */ max_length += MOD_IR_SIZE-1; max_length -= max_length%MOD_IR_SIZE; TRACE("Skipped delay: %d, max delay: %d, new FIR length: %d\n", min_delay, max_delay-min_delay, max_length); state->IrSize = max_length; #undef NUM_BANDS } static struct Hrtf *CreateHrtfStore(ALuint rate, ALsizei irSize, ALfloat distance, ALsizei evCount, ALsizei irCount, const ALubyte *azCount, const ALushort *evOffset, const ALfloat (*coeffs)[2], const ALubyte (*delays)[2], const char *filename) { struct Hrtf *Hrtf; size_t total; total = sizeof(struct Hrtf); total += sizeof(Hrtf->azCount[0])*evCount; total = RoundUp(total, sizeof(ALushort)); /* Align for ushort fields */ total += sizeof(Hrtf->evOffset[0])*evCount; total = RoundUp(total, 16); /* Align for coefficients using SIMD */ total += sizeof(Hrtf->coeffs[0])*irSize*irCount; total += sizeof(Hrtf->delays[0])*irCount; Hrtf = al_calloc(16, total); if(Hrtf == NULL) ERR("Out of memory allocating storage for %s.\n", filename); else { uintptr_t offset = sizeof(struct Hrtf); char *base = (char*)Hrtf; ALushort *_evOffset; ALubyte *_azCount; ALubyte (*_delays)[2]; ALfloat (*_coeffs)[2]; ALsizei i; InitRef(&Hrtf->ref, 0); Hrtf->sampleRate = rate; Hrtf->irSize = irSize; Hrtf->distance = distance; Hrtf->evCount = evCount; /* Set up pointers to storage following the main HRTF struct. */ _azCount = (ALubyte*)(base + offset); offset += sizeof(_azCount[0])*evCount; offset = RoundUp(offset, sizeof(ALushort)); /* Align for ushort fields */ _evOffset = (ALushort*)(base + offset); offset += sizeof(_evOffset[0])*evCount; offset = RoundUp(offset, 16); /* Align for coefficients using SIMD */ _coeffs = (ALfloat(*)[2])(base + offset); offset += sizeof(_coeffs[0])*irSize*irCount; _delays = (ALubyte(*)[2])(base + offset); offset += sizeof(_delays[0])*irCount; assert(offset == total); /* Copy input data to storage. */ for(i = 0;i < evCount;i++) _azCount[i] = azCount[i]; for(i = 0;i < evCount;i++) _evOffset[i] = evOffset[i]; for(i = 0;i < irSize*irCount;i++) { _coeffs[i][0] = coeffs[i][0]; _coeffs[i][1] = coeffs[i][1]; } for(i = 0;i < irCount;i++) { _delays[i][0] = delays[i][0]; _delays[i][1] = delays[i][1]; } /* Finally, assign the storage pointers. */ Hrtf->azCount = _azCount; Hrtf->evOffset = _evOffset; Hrtf->coeffs = _coeffs; Hrtf->delays = _delays; } return Hrtf; } static ALubyte GetLE_ALubyte(const ALubyte **data, size_t *len) { ALubyte ret = (*data)[0]; *data += 1; *len -= 1; return ret; } static ALshort GetLE_ALshort(const ALubyte **data, size_t *len) { ALshort ret = (*data)[0] | ((*data)[1]<<8); *data += 2; *len -= 2; return ret; } static ALushort GetLE_ALushort(const ALubyte **data, size_t *len) { ALushort ret = (*data)[0] | ((*data)[1]<<8); *data += 2; *len -= 2; return ret; } static ALint GetLE_ALint24(const ALubyte **data, size_t *len) { ALint ret = (*data)[0] | ((*data)[1]<<8) | ((*data)[2]<<16); *data += 3; *len -= 3; return (ret^0x800000) - 0x800000; } static ALuint GetLE_ALuint(const ALubyte **data, size_t *len) { ALuint ret = (*data)[0] | ((*data)[1]<<8) | ((*data)[2]<<16) | ((*data)[3]<<24); *data += 4; *len -= 4; return ret; } static const ALubyte *Get_ALubytePtr(const ALubyte **data, size_t *len, size_t size) { const ALubyte *ret = *data; *data += size; *len -= size; return ret; } static struct Hrtf *LoadHrtf00(const ALubyte *data, size_t datalen, const char *filename) { struct Hrtf *Hrtf = NULL; ALboolean failed = AL_FALSE; ALuint rate = 0; ALushort irCount = 0; ALushort irSize = 0; ALubyte evCount = 0; ALubyte *azCount = NULL; ALushort *evOffset = NULL; ALfloat (*coeffs)[2] = NULL; ALubyte (*delays)[2] = NULL; ALsizei i, j; if(datalen < 9) { ERR("Unexpected end of %s data (req %d, rem "SZFMT")\n", filename, 9, datalen); return NULL; } rate = GetLE_ALuint(&data, &datalen); irCount = GetLE_ALushort(&data, &datalen); irSize = GetLE_ALushort(&data, &datalen); evCount = GetLE_ALubyte(&data, &datalen); if(irSize < MIN_IR_SIZE || irSize > MAX_IR_SIZE || (irSize%MOD_IR_SIZE)) { ERR("Unsupported HRIR size: irSize=%d (%d to %d by %d)\n", irSize, MIN_IR_SIZE, MAX_IR_SIZE, MOD_IR_SIZE); failed = AL_TRUE; } if(evCount < MIN_EV_COUNT || evCount > MAX_EV_COUNT) { ERR("Unsupported elevation count: evCount=%d (%d to %d)\n", evCount, MIN_EV_COUNT, MAX_EV_COUNT); failed = AL_TRUE; } if(failed) return NULL; if(datalen < evCount*2u) { ERR("Unexpected end of %s data (req %d, rem "SZFMT")\n", filename, evCount*2, datalen); return NULL; } azCount = malloc(sizeof(azCount[0])*evCount); evOffset = malloc(sizeof(evOffset[0])*evCount); if(azCount == NULL || evOffset == NULL) { ERR("Out of memory.\n"); failed = AL_TRUE; } if(!failed) { evOffset[0] = GetLE_ALushort(&data, &datalen); for(i = 1;i < evCount;i++) { evOffset[i] = GetLE_ALushort(&data, &datalen); if(evOffset[i] <= evOffset[i-1]) { ERR("Invalid evOffset: evOffset[%d]=%d (last=%d)\n", i, evOffset[i], evOffset[i-1]); failed = AL_TRUE; } azCount[i-1] = evOffset[i] - evOffset[i-1]; if(azCount[i-1] < MIN_AZ_COUNT || azCount[i-1] > MAX_AZ_COUNT) { ERR("Unsupported azimuth count: azCount[%d]=%d (%d to %d)\n", i-1, azCount[i-1], MIN_AZ_COUNT, MAX_AZ_COUNT); failed = AL_TRUE; } } if(irCount <= evOffset[i-1]) { ERR("Invalid evOffset: evOffset[%d]=%d (irCount=%d)\n", i-1, evOffset[i-1], irCount); failed = AL_TRUE; } azCount[i-1] = irCount - evOffset[i-1]; if(azCount[i-1] < MIN_AZ_COUNT || azCount[i-1] > MAX_AZ_COUNT) { ERR("Unsupported azimuth count: azCount[%d]=%d (%d to %d)\n", i-1, azCount[i-1], MIN_AZ_COUNT, MAX_AZ_COUNT); failed = AL_TRUE; } } if(!failed) { coeffs = malloc(sizeof(coeffs[0])*irSize*irCount); delays = malloc(sizeof(delays[0])*irCount); if(coeffs == NULL || delays == NULL) { ERR("Out of memory.\n"); failed = AL_TRUE; } } if(!failed) { size_t reqsize = 2*irSize*irCount + irCount; if(datalen < reqsize) { ERR("Unexpected end of %s data (req "SZFMT", rem "SZFMT")\n", filename, reqsize, datalen); failed = AL_TRUE; } } if(!failed) { for(i = 0;i < irCount;i++) { for(j = 0;j < irSize;j++) coeffs[i*irSize + j][0] = GetLE_ALshort(&data, &datalen) / 32768.0f; } for(i = 0;i < irCount;i++) { delays[i][0] = GetLE_ALubyte(&data, &datalen); if(delays[i][0] > MAX_HRIR_DELAY) { ERR("Invalid delays[%d]: %d (%d)\n", i, delays[i][0], MAX_HRIR_DELAY); failed = AL_TRUE; } } } if(!failed) { /* Mirror the left ear responses to the right ear. */ for(i = 0;i < evCount;i++) { ALushort evoffset = evOffset[i]; ALubyte azcount = azCount[i]; for(j = 0;j < azcount;j++) { ALsizei lidx = evoffset + j; ALsizei ridx = evoffset + ((azcount-j) % azcount); ALsizei k; for(k = 0;k < irSize;k++) coeffs[ridx*irSize + k][1] = coeffs[lidx*irSize + k][0]; delays[ridx][1] = delays[lidx][0]; } } Hrtf = CreateHrtfStore(rate, irSize, 0.0f, evCount, irCount, azCount, evOffset, coeffs, delays, filename); } free(azCount); free(evOffset); free(coeffs); free(delays); return Hrtf; } static struct Hrtf *LoadHrtf01(const ALubyte *data, size_t datalen, const char *filename) { struct Hrtf *Hrtf = NULL; ALboolean failed = AL_FALSE; ALuint rate = 0; ALushort irCount = 0; ALushort irSize = 0; ALubyte evCount = 0; const ALubyte *azCount = NULL; ALushort *evOffset = NULL; ALfloat (*coeffs)[2] = NULL; ALubyte (*delays)[2] = NULL; ALsizei i, j; if(datalen < 6) { ERR("Unexpected end of %s data (req %d, rem "SZFMT"\n", filename, 6, datalen); return NULL; } rate = GetLE_ALuint(&data, &datalen); irSize = GetLE_ALubyte(&data, &datalen); evCount = GetLE_ALubyte(&data, &datalen); if(irSize < MIN_IR_SIZE || irSize > MAX_IR_SIZE || (irSize%MOD_IR_SIZE)) { ERR("Unsupported HRIR size: irSize=%d (%d to %d by %d)\n", irSize, MIN_IR_SIZE, MAX_IR_SIZE, MOD_IR_SIZE); failed = AL_TRUE; } if(evCount < MIN_EV_COUNT || evCount > MAX_EV_COUNT) { ERR("Unsupported elevation count: evCount=%d (%d to %d)\n", evCount, MIN_EV_COUNT, MAX_EV_COUNT); failed = AL_TRUE; } if(failed) return NULL; if(datalen < evCount) { ERR("Unexpected end of %s data (req %d, rem "SZFMT"\n", filename, evCount, datalen); return NULL; } azCount = Get_ALubytePtr(&data, &datalen, evCount); evOffset = malloc(sizeof(evOffset[0])*evCount); if(azCount == NULL || evOffset == NULL) { ERR("Out of memory.\n"); failed = AL_TRUE; } if(!failed) { for(i = 0;i < evCount;i++) { if(azCount[i] < MIN_AZ_COUNT || azCount[i] > MAX_AZ_COUNT) { ERR("Unsupported azimuth count: azCount[%d]=%d (%d to %d)\n", i, azCount[i], MIN_AZ_COUNT, MAX_AZ_COUNT); failed = AL_TRUE; } } } if(!failed) { evOffset[0] = 0; irCount = azCount[0]; for(i = 1;i < evCount;i++) { evOffset[i] = evOffset[i-1] + azCount[i-1]; irCount += azCount[i]; } coeffs = malloc(sizeof(coeffs[0])*irSize*irCount); delays = malloc(sizeof(delays[0])*irCount); if(coeffs == NULL || delays == NULL) { ERR("Out of memory.\n"); failed = AL_TRUE; } } if(!failed) { size_t reqsize = 2*irSize*irCount + irCount; if(datalen < reqsize) { ERR("Unexpected end of %s data (req "SZFMT", rem "SZFMT"\n", filename, reqsize, datalen); failed = AL_TRUE; } } if(!failed) { for(i = 0;i < irCount;i++) { for(j = 0;j < irSize;j++) coeffs[i*irSize + j][0] = GetLE_ALshort(&data, &datalen) / 32768.0f; } for(i = 0;i < irCount;i++) { delays[i][0] = GetLE_ALubyte(&data, &datalen); if(delays[i][0] > MAX_HRIR_DELAY) { ERR("Invalid delays[%d]: %d (%d)\n", i, delays[i][0], MAX_HRIR_DELAY); failed = AL_TRUE; } } } if(!failed) { /* Mirror the left ear responses to the right ear. */ for(i = 0;i < evCount;i++) { ALushort evoffset = evOffset[i]; ALubyte azcount = azCount[i]; for(j = 0;j < azcount;j++) { ALsizei lidx = evoffset + j; ALsizei ridx = evoffset + ((azcount-j) % azcount); ALsizei k; for(k = 0;k < irSize;k++) coeffs[ridx*irSize + k][1] = coeffs[lidx*irSize + k][0]; delays[ridx][1] = delays[lidx][0]; } } Hrtf = CreateHrtfStore(rate, irSize, 0.0f, evCount, irCount, azCount, evOffset, coeffs, delays, filename); } free(evOffset); free(coeffs); free(delays); return Hrtf; } #define SAMPLETYPE_S16 0 #define SAMPLETYPE_S24 1 #define CHANTYPE_LEFTONLY 0 #define CHANTYPE_LEFTRIGHT 1 static struct Hrtf *LoadHrtf02(const ALubyte *data, size_t datalen, const char *filename) { struct Hrtf *Hrtf = NULL; ALboolean failed = AL_FALSE; ALuint rate = 0; ALubyte sampleType; ALubyte channelType; ALushort irCount = 0; ALushort irSize = 0; ALubyte fdCount = 0; ALushort distance = 0; ALubyte evCount = 0; const ALubyte *azCount = NULL; ALushort *evOffset = NULL; ALfloat (*coeffs)[2] = NULL; ALubyte (*delays)[2] = NULL; ALsizei i, j; if(datalen < 8) { ERR("Unexpected end of %s data (req %d, rem "SZFMT"\n", filename, 8, datalen); return NULL; } rate = GetLE_ALuint(&data, &datalen); sampleType = GetLE_ALubyte(&data, &datalen); channelType = GetLE_ALubyte(&data, &datalen); irSize = GetLE_ALubyte(&data, &datalen); fdCount = GetLE_ALubyte(&data, &datalen); if(sampleType > SAMPLETYPE_S24) { ERR("Unsupported sample type: %d\n", sampleType); failed = AL_TRUE; } if(channelType > CHANTYPE_LEFTRIGHT) { ERR("Unsupported channel type: %d\n", channelType); failed = AL_TRUE; } if(irSize < MIN_IR_SIZE || irSize > MAX_IR_SIZE || (irSize%MOD_IR_SIZE)) { ERR("Unsupported HRIR size: irSize=%d (%d to %d by %d)\n", irSize, MIN_IR_SIZE, MAX_IR_SIZE, MOD_IR_SIZE); failed = AL_TRUE; } if(fdCount != 1) { ERR("Multiple field-depths not supported: fdCount=%d (%d to %d)\n", evCount, MIN_FD_COUNT, MAX_FD_COUNT); failed = AL_TRUE; } if(failed) return NULL; for(i = 0;i < fdCount;i++) { if(datalen < 3) { ERR("Unexpected end of %s data (req %d, rem "SZFMT"\n", filename, 3, datalen); return NULL; } distance = GetLE_ALushort(&data, &datalen); if(distance < MIN_FD_DISTANCE || distance > MAX_FD_DISTANCE) { ERR("Unsupported field distance: distance=%d (%dmm to %dmm)\n", distance, MIN_FD_DISTANCE, MAX_FD_DISTANCE); failed = AL_TRUE; } evCount = GetLE_ALubyte(&data, &datalen); if(evCount < MIN_EV_COUNT || evCount > MAX_EV_COUNT) { ERR("Unsupported elevation count: evCount=%d (%d to %d)\n", evCount, MIN_EV_COUNT, MAX_EV_COUNT); failed = AL_TRUE; } if(failed) return NULL; if(datalen < evCount) { ERR("Unexpected end of %s data (req %d, rem "SZFMT"\n", filename, evCount, datalen); return NULL; } azCount = Get_ALubytePtr(&data, &datalen, evCount); for(j = 0;j < evCount;j++) { if(azCount[j] < MIN_AZ_COUNT || azCount[j] > MAX_AZ_COUNT) { ERR("Unsupported azimuth count: azCount[%d]=%d (%d to %d)\n", j, azCount[j], MIN_AZ_COUNT, MAX_AZ_COUNT); failed = AL_TRUE; } } } if(failed) return NULL; evOffset = malloc(sizeof(evOffset[0])*evCount); if(azCount == NULL || evOffset == NULL) { ERR("Out of memory.\n"); failed = AL_TRUE; } if(!failed) { evOffset[0] = 0; irCount = azCount[0]; for(i = 1;i < evCount;i++) { evOffset[i] = evOffset[i-1] + azCount[i-1]; irCount += azCount[i]; } coeffs = malloc(sizeof(coeffs[0])*irSize*irCount); delays = malloc(sizeof(delays[0])*irCount); if(coeffs == NULL || delays == NULL) { ERR("Out of memory.\n"); failed = AL_TRUE; } } if(!failed) { size_t reqsize = 2*irSize*irCount + irCount; if(datalen < reqsize) { ERR("Unexpected end of %s data (req "SZFMT", rem "SZFMT"\n", filename, reqsize, datalen); failed = AL_TRUE; } } if(!failed) { if(channelType == CHANTYPE_LEFTONLY) { if(sampleType == SAMPLETYPE_S16) for(i = 0;i < irCount;i++) { for(j = 0;j < irSize;j++) coeffs[i*irSize + j][0] = GetLE_ALshort(&data, &datalen) / 32768.0f; } else if(sampleType == SAMPLETYPE_S24) for(i = 0;i < irCount;i++) { for(j = 0;j < irSize;j++) coeffs[i*irSize + j][0] = GetLE_ALint24(&data, &datalen) / 8388608.0f; } for(i = 0;i < irCount;i++) { delays[i][0] = GetLE_ALubyte(&data, &datalen); if(delays[i][0] > MAX_HRIR_DELAY) { ERR("Invalid delays[%d][0]: %d (%d)\n", i, delays[i][0], MAX_HRIR_DELAY); failed = AL_TRUE; } } } else if(channelType == CHANTYPE_LEFTRIGHT) { if(sampleType == SAMPLETYPE_S16) for(i = 0;i < irCount;i++) { for(j = 0;j < irSize;j++) { coeffs[i*irSize + j][0] = GetLE_ALshort(&data, &datalen) / 32768.0f; coeffs[i*irSize + j][1] = GetLE_ALshort(&data, &datalen) / 32768.0f; } } else if(sampleType == SAMPLETYPE_S24) for(i = 0;i < irCount;i++) { for(j = 0;j < irSize;j++) { coeffs[i*irSize + j][0] = GetLE_ALint24(&data, &datalen) / 8388608.0f; coeffs[i*irSize + j][1] = GetLE_ALint24(&data, &datalen) / 8388608.0f; } } for(i = 0;i < irCount;i++) { delays[i][0] = GetLE_ALubyte(&data, &datalen); if(delays[i][0] > MAX_HRIR_DELAY) { ERR("Invalid delays[%d][0]: %d (%d)\n", i, delays[i][0], MAX_HRIR_DELAY); failed = AL_TRUE; } delays[i][1] = GetLE_ALubyte(&data, &datalen); if(delays[i][1] > MAX_HRIR_DELAY) { ERR("Invalid delays[%d][1]: %d (%d)\n", i, delays[i][1], MAX_HRIR_DELAY); failed = AL_TRUE; } } } } if(!failed) { if(channelType == CHANTYPE_LEFTONLY) { /* Mirror the left ear responses to the right ear. */ for(i = 0;i < evCount;i++) { ALushort evoffset = evOffset[i]; ALubyte azcount = azCount[i]; for(j = 0;j < azcount;j++) { ALsizei lidx = evoffset + j; ALsizei ridx = evoffset + ((azcount-j) % azcount); ALsizei k; for(k = 0;k < irSize;k++) coeffs[ridx*irSize + k][1] = coeffs[lidx*irSize + k][0]; delays[ridx][1] = delays[lidx][0]; } } } Hrtf = CreateHrtfStore(rate, irSize, (ALfloat)distance / 1000.0f, evCount, irCount, azCount, evOffset, coeffs, delays, filename ); } free(evOffset); free(coeffs); free(delays); return Hrtf; } static void AddFileEntry(vector_EnumeratedHrtf *list, const_al_string filename) { EnumeratedHrtf entry = { AL_STRING_INIT_STATIC(), NULL }; struct HrtfEntry *loaded_entry; const EnumeratedHrtf *iter; const char *name; const char *ext; int i; /* Check if this file has already been loaded globally. */ loaded_entry = LoadedHrtfs; while(loaded_entry) { if(alstr_cmp_cstr(filename, loaded_entry->filename) == 0) { /* Check if this entry has already been added to the list. */ #define MATCH_ENTRY(i) (loaded_entry == (i)->hrtf) VECTOR_FIND_IF(iter, const EnumeratedHrtf, *list, MATCH_ENTRY); #undef MATCH_ENTRY if(iter != VECTOR_END(*list)) { TRACE("Skipping duplicate file entry %s\n", alstr_get_cstr(filename)); return; } break; } loaded_entry = loaded_entry->next; } if(!loaded_entry) { TRACE("Got new file \"%s\"\n", alstr_get_cstr(filename)); loaded_entry = al_calloc(DEF_ALIGN, FAM_SIZE(struct HrtfEntry, filename, alstr_length(filename)+1) ); loaded_entry->next = LoadedHrtfs; loaded_entry->handle = NULL; strcpy(loaded_entry->filename, alstr_get_cstr(filename)); LoadedHrtfs = loaded_entry; } /* TODO: Get a human-readable name from the HRTF data (possibly coming in a * format update). */ name = strrchr(alstr_get_cstr(filename), '/'); if(!name) name = strrchr(alstr_get_cstr(filename), '\\'); if(!name) name = alstr_get_cstr(filename); else ++name; ext = strrchr(name, '.'); i = 0; do { if(!ext) alstr_copy_cstr(&entry.name, name); else alstr_copy_range(&entry.name, name, ext); if(i != 0) { char str[64]; snprintf(str, sizeof(str), " #%d", i+1); alstr_append_cstr(&entry.name, str); } ++i; #define MATCH_NAME(i) (alstr_cmp(entry.name, (i)->name) == 0) VECTOR_FIND_IF(iter, const EnumeratedHrtf, *list, MATCH_NAME); #undef MATCH_NAME } while(iter != VECTOR_END(*list)); entry.hrtf = loaded_entry; TRACE("Adding entry \"%s\" from file \"%s\"\n", alstr_get_cstr(entry.name), alstr_get_cstr(filename)); VECTOR_PUSH_BACK(*list, entry); } /* Unfortunate that we have to duplicate AddFileEntry to take a memory buffer * for input instead of opening the given filename. */ static void AddBuiltInEntry(vector_EnumeratedHrtf *list, const_al_string filename, ALuint residx) { EnumeratedHrtf entry = { AL_STRING_INIT_STATIC(), NULL }; struct HrtfEntry *loaded_entry; struct Hrtf *hrtf = NULL; const EnumeratedHrtf *iter; const char *name; const char *ext; int i; loaded_entry = LoadedHrtfs; while(loaded_entry) { if(alstr_cmp_cstr(filename, loaded_entry->filename) == 0) { #define MATCH_ENTRY(i) (loaded_entry == (i)->hrtf) VECTOR_FIND_IF(iter, const EnumeratedHrtf, *list, MATCH_ENTRY); #undef MATCH_ENTRY if(iter != VECTOR_END(*list)) { TRACE("Skipping duplicate file entry %s\n", alstr_get_cstr(filename)); return; } break; } loaded_entry = loaded_entry->next; } if(!loaded_entry) { size_t namelen = alstr_length(filename)+32; TRACE("Got new file \"%s\"\n", alstr_get_cstr(filename)); loaded_entry = al_calloc(DEF_ALIGN, FAM_SIZE(struct HrtfEntry, filename, namelen) ); loaded_entry->next = LoadedHrtfs; loaded_entry->handle = hrtf; snprintf(loaded_entry->filename, namelen, "!%u_%s", residx, alstr_get_cstr(filename)); LoadedHrtfs = loaded_entry; } /* TODO: Get a human-readable name from the HRTF data (possibly coming in a * format update). */ name = strrchr(alstr_get_cstr(filename), '/'); if(!name) name = strrchr(alstr_get_cstr(filename), '\\'); if(!name) name = alstr_get_cstr(filename); else ++name; ext = strrchr(name, '.'); i = 0; do { if(!ext) alstr_copy_cstr(&entry.name, name); else alstr_copy_range(&entry.name, name, ext); if(i != 0) { char str[64]; snprintf(str, sizeof(str), " #%d", i+1); alstr_append_cstr(&entry.name, str); } ++i; #define MATCH_NAME(i) (alstr_cmp(entry.name, (i)->name) == 0) VECTOR_FIND_IF(iter, const EnumeratedHrtf, *list, MATCH_NAME); #undef MATCH_NAME } while(iter != VECTOR_END(*list)); entry.hrtf = loaded_entry; TRACE("Adding built-in entry \"%s\"\n", alstr_get_cstr(entry.name)); VECTOR_PUSH_BACK(*list, entry); } #define IDR_DEFAULT_44100_MHR 1 #define IDR_DEFAULT_48000_MHR 2 #ifndef ALSOFT_EMBED_HRTF_DATA static const ALubyte *GetResource(int UNUSED(name), size_t *size) { *size = 0; return NULL; } #else #include "default-44100.mhr.h" #include "default-48000.mhr.h" static const ALubyte *GetResource(int name, size_t *size) { if(name == IDR_DEFAULT_44100_MHR) { *size = sizeof(hrtf_default_44100); return hrtf_default_44100; } if(name == IDR_DEFAULT_48000_MHR) { *size = sizeof(hrtf_default_48000); return hrtf_default_48000; } *size = 0; return NULL; } #endif vector_EnumeratedHrtf EnumerateHrtf(const_al_string devname) { vector_EnumeratedHrtf list = VECTOR_INIT_STATIC(); const char *defaulthrtf = ""; const char *pathlist = ""; bool usedefaults = true; if(ConfigValueStr(alstr_get_cstr(devname), NULL, "hrtf-paths", &pathlist)) { al_string pname = AL_STRING_INIT_STATIC(); while(pathlist && *pathlist) { const char *next, *end; while(isspace(*pathlist) || *pathlist == ',') pathlist++; if(*pathlist == '\0') continue; next = strchr(pathlist, ','); if(next) end = next++; else { end = pathlist + strlen(pathlist); usedefaults = false; } while(end != pathlist && isspace(*(end-1))) --end; if(end != pathlist) { vector_al_string flist; size_t i; alstr_copy_range(&pname, pathlist, end); flist = SearchDataFiles(".mhr", alstr_get_cstr(pname)); for(i = 0;i < VECTOR_SIZE(flist);i++) AddFileEntry(&list, VECTOR_ELEM(flist, i)); VECTOR_FOR_EACH(al_string, flist, alstr_reset); VECTOR_DEINIT(flist); } pathlist = next; } alstr_reset(&pname); } else if(ConfigValueExists(alstr_get_cstr(devname), NULL, "hrtf_tables")) ERR("The hrtf_tables option is deprecated, please use hrtf-paths instead.\n"); if(usedefaults) { al_string ename = AL_STRING_INIT_STATIC(); vector_al_string flist; const ALubyte *rdata; size_t rsize, i; flist = SearchDataFiles(".mhr", "openal/hrtf"); for(i = 0;i < VECTOR_SIZE(flist);i++) AddFileEntry(&list, VECTOR_ELEM(flist, i)); VECTOR_FOR_EACH(al_string, flist, alstr_reset); VECTOR_DEINIT(flist); rdata = GetResource(IDR_DEFAULT_44100_MHR, &rsize); if(rdata != NULL && rsize > 0) { alstr_copy_cstr(&ename, "Built-In 44100hz"); AddBuiltInEntry(&list, ename, IDR_DEFAULT_44100_MHR); } rdata = GetResource(IDR_DEFAULT_48000_MHR, &rsize); if(rdata != NULL && rsize > 0) { alstr_copy_cstr(&ename, "Built-In 48000hz"); AddBuiltInEntry(&list, ename, IDR_DEFAULT_48000_MHR); } alstr_reset(&ename); } if(VECTOR_SIZE(list) > 1 && ConfigValueStr(alstr_get_cstr(devname), NULL, "default-hrtf", &defaulthrtf)) { const EnumeratedHrtf *iter; /* Find the preferred HRTF and move it to the front of the list. */ #define FIND_ENTRY(i) (alstr_cmp_cstr((i)->name, defaulthrtf) == 0) VECTOR_FIND_IF(iter, const EnumeratedHrtf, list, FIND_ENTRY); #undef FIND_ENTRY if(iter == VECTOR_END(list)) WARN("Failed to find default HRTF \"%s\"\n", defaulthrtf); else if(iter != VECTOR_BEGIN(list)) { EnumeratedHrtf entry = *iter; memmove(&VECTOR_ELEM(list,1), &VECTOR_ELEM(list,0), (iter-VECTOR_BEGIN(list))*sizeof(EnumeratedHrtf)); VECTOR_ELEM(list,0) = entry; } } return list; } void FreeHrtfList(vector_EnumeratedHrtf *list) { #define CLEAR_ENTRY(i) alstr_reset(&(i)->name) VECTOR_FOR_EACH(EnumeratedHrtf, *list, CLEAR_ENTRY); VECTOR_DEINIT(*list); #undef CLEAR_ENTRY } struct Hrtf *GetLoadedHrtf(struct HrtfEntry *entry) { struct Hrtf *hrtf = NULL; struct FileMapping fmap; const ALubyte *rdata; const char *name; ALuint residx; size_t rsize; char ch; while(ATOMIC_FLAG_TEST_AND_SET(&LoadedHrtfLock, almemory_order_seq_cst)) althrd_yield(); if(entry->handle) { hrtf = entry->handle; Hrtf_IncRef(hrtf); goto done; } fmap.ptr = NULL; fmap.len = 0; if(sscanf(entry->filename, "!%u%c", &residx, &ch) == 2 && ch == '_') { name = strchr(entry->filename, ch)+1; TRACE("Loading %s...\n", name); rdata = GetResource(residx, &rsize); if(rdata == NULL || rsize == 0) { ERR("Could not get resource %u, %s\n", residx, name); goto done; } } else { name = entry->filename; TRACE("Loading %s...\n", entry->filename); fmap = MapFileToMem(entry->filename); if(fmap.ptr == NULL) { ERR("Could not open %s\n", entry->filename); goto done; } rdata = fmap.ptr; rsize = fmap.len; } if(rsize < sizeof(magicMarker02)) ERR("%s data is too short ("SZFMT" bytes)\n", name, rsize); else if(memcmp(rdata, magicMarker02, sizeof(magicMarker02)) == 0) { TRACE("Detected data set format v2\n"); hrtf = LoadHrtf02(rdata+sizeof(magicMarker02), rsize-sizeof(magicMarker02), name ); } else if(memcmp(rdata, magicMarker01, sizeof(magicMarker01)) == 0) { TRACE("Detected data set format v1\n"); hrtf = LoadHrtf01(rdata+sizeof(magicMarker01), rsize-sizeof(magicMarker01), name ); } else if(memcmp(rdata, magicMarker00, sizeof(magicMarker00)) == 0) { TRACE("Detected data set format v0\n"); hrtf = LoadHrtf00(rdata+sizeof(magicMarker00), rsize-sizeof(magicMarker00), name ); } else ERR("Invalid header in %s: \"%.8s\"\n", name, (const char*)rdata); if(fmap.ptr) UnmapFileMem(&fmap); if(!hrtf) { ERR("Failed to load %s\n", name); goto done; } entry->handle = hrtf; Hrtf_IncRef(hrtf); TRACE("Loaded HRTF support for format: %s %uhz\n", DevFmtChannelsString(DevFmtStereo), hrtf->sampleRate); done: ATOMIC_FLAG_CLEAR(&LoadedHrtfLock, almemory_order_seq_cst); return hrtf; } void Hrtf_IncRef(struct Hrtf *hrtf) { uint ref = IncrementRef(&hrtf->ref); TRACEREF("%p increasing refcount to %u\n", hrtf, ref); } void Hrtf_DecRef(struct Hrtf *hrtf) { struct HrtfEntry *Hrtf; uint ref = DecrementRef(&hrtf->ref); TRACEREF("%p decreasing refcount to %u\n", hrtf, ref); if(ref == 0) { while(ATOMIC_FLAG_TEST_AND_SET(&LoadedHrtfLock, almemory_order_seq_cst)) althrd_yield(); Hrtf = LoadedHrtfs; while(Hrtf != NULL) { /* Need to double-check that it's still unused, as another device * could've reacquired this HRTF after its reference went to 0 and * before the lock was taken. */ if(hrtf == Hrtf->handle && ReadRef(&hrtf->ref) == 0) { al_free(Hrtf->handle); Hrtf->handle = NULL; TRACE("Unloaded unused HRTF %s\n", Hrtf->filename); } Hrtf = Hrtf->next; } ATOMIC_FLAG_CLEAR(&LoadedHrtfLock, almemory_order_seq_cst); } } void FreeHrtfs(void) { struct HrtfEntry *Hrtf = LoadedHrtfs; LoadedHrtfs = NULL; while(Hrtf != NULL) { struct HrtfEntry *next = Hrtf->next; al_free(Hrtf->handle); al_free(Hrtf); Hrtf = next; } } openal-soft-openal-soft-1.19.1/Alc/hrtf.h000066400000000000000000000042631335774445300201160ustar00rootroot00000000000000#ifndef ALC_HRTF_H #define ALC_HRTF_H #include "AL/al.h" #include "AL/alc.h" #include "alMain.h" #include "alstring.h" #include "atomic.h" #define HRTF_HISTORY_BITS (6) #define HRTF_HISTORY_LENGTH (1< #ifdef __GNUC__ #define DECL_FORMAT(x, y, z) __attribute__((format(x, (y), (z)))) #else #define DECL_FORMAT(x, y, z) #endif #ifdef __cplusplus extern "C" { #endif extern FILE *LogFile; #if defined(__GNUC__) && !defined(_WIN32) #define AL_PRINT(T, MSG, ...) fprintf(LogFile, "AL lib: %s %s: "MSG, T, __FUNCTION__ , ## __VA_ARGS__) #else void al_print(const char *type, const char *func, const char *fmt, ...) DECL_FORMAT(printf, 3,4); #define AL_PRINT(T, ...) al_print((T), __FUNCTION__, __VA_ARGS__) #endif #ifdef __ANDROID__ #include #define LOG_ANDROID(T, MSG, ...) __android_log_print(T, "openal", "AL lib: %s: "MSG, __FUNCTION__ , ## __VA_ARGS__) #else #define LOG_ANDROID(T, MSG, ...) ((void)0) #endif enum LogLevel { NoLog, LogError, LogWarning, LogTrace, LogRef }; extern enum LogLevel LogLevel; #define TRACEREF(...) do { \ if(LogLevel >= LogRef) \ AL_PRINT("(--)", __VA_ARGS__); \ } while(0) #define TRACE(...) do { \ if(LogLevel >= LogTrace) \ AL_PRINT("(II)", __VA_ARGS__); \ LOG_ANDROID(ANDROID_LOG_DEBUG, __VA_ARGS__); \ } while(0) #define WARN(...) do { \ if(LogLevel >= LogWarning) \ AL_PRINT("(WW)", __VA_ARGS__); \ LOG_ANDROID(ANDROID_LOG_WARN, __VA_ARGS__); \ } while(0) #define ERR(...) do { \ if(LogLevel >= LogError) \ AL_PRINT("(EE)", __VA_ARGS__); \ LOG_ANDROID(ANDROID_LOG_ERROR, __VA_ARGS__); \ } while(0) #ifdef __cplusplus } /* extern "C" */ #endif #endif /* LOGGING_H */ openal-soft-openal-soft-1.19.1/Alc/mastering.c000066400000000000000000000402401335774445300211320ustar00rootroot00000000000000#include "config.h" #include #include "mastering.h" #include "alu.h" #include "almalloc.h" #include "static_assert.h" /* These structures assume BUFFERSIZE is a power of 2. */ static_assert((BUFFERSIZE & (BUFFERSIZE-1)) == 0, "BUFFERSIZE is not a power of 2"); typedef struct SlidingHold { ALfloat Values[BUFFERSIZE]; ALsizei Expiries[BUFFERSIZE]; ALsizei LowerIndex; ALsizei UpperIndex; ALsizei Length; } SlidingHold; /* General topology and basic automation was based on the following paper: * * D. Giannoulis, M. Massberg and J. D. Reiss, * "Parameter Automation in a Dynamic Range Compressor," * Journal of the Audio Engineering Society, v61 (10), Oct. 2013 * * Available (along with supplemental reading) at: * * http://c4dm.eecs.qmul.ac.uk/audioengineering/compressors/ */ typedef struct Compressor { ALsizei NumChans; ALuint SampleRate; struct { ALuint Knee : 1; ALuint Attack : 1; ALuint Release : 1; ALuint PostGain : 1; ALuint Declip : 1; } Auto; ALsizei LookAhead; ALfloat PreGain; ALfloat PostGain; ALfloat Threshold; ALfloat Slope; ALfloat Knee; ALfloat Attack; ALfloat Release; alignas(16) ALfloat SideChain[2*BUFFERSIZE]; alignas(16) ALfloat CrestFactor[BUFFERSIZE]; SlidingHold *Hold; ALfloat (*Delay)[BUFFERSIZE]; ALsizei DelayIndex; ALfloat CrestCoeff; ALfloat GainEstimate; ALfloat AdaptCoeff; ALfloat LastPeakSq; ALfloat LastRmsSq; ALfloat LastRelease; ALfloat LastAttack; ALfloat LastGainDev; } Compressor; /* This sliding hold follows the input level with an instant attack and a * fixed duration hold before an instant release to the next highest level. * It is a sliding window maximum (descending maxima) implementation based on * Richard Harter's ascending minima algorithm available at: * * http://www.richardhartersworld.com/cri/2001/slidingmin.html */ static ALfloat UpdateSlidingHold(SlidingHold *Hold, const ALsizei i, const ALfloat in) { const ALsizei mask = BUFFERSIZE - 1; const ALsizei length = Hold->Length; ALfloat *restrict values = Hold->Values; ALsizei *restrict expiries = Hold->Expiries; ALsizei lowerIndex = Hold->LowerIndex; ALsizei upperIndex = Hold->UpperIndex; if(i >= expiries[upperIndex]) upperIndex = (upperIndex + 1) & mask; if(in >= values[upperIndex]) { values[upperIndex] = in; expiries[upperIndex] = i + length; lowerIndex = upperIndex; } else { do { do { if(!(in >= values[lowerIndex])) goto found_place; } while(lowerIndex--); lowerIndex = mask; } while(1); found_place: lowerIndex = (lowerIndex + 1) & mask; values[lowerIndex] = in; expiries[lowerIndex] = i + length; } Hold->LowerIndex = lowerIndex; Hold->UpperIndex = upperIndex; return values[upperIndex]; } static void ShiftSlidingHold(SlidingHold *Hold, const ALsizei n) { const ALsizei lowerIndex = Hold->LowerIndex; ALsizei *restrict expiries = Hold->Expiries; ALsizei i = Hold->UpperIndex; if(lowerIndex < i) { for(;i < BUFFERSIZE;i++) expiries[i] -= n; i = 0; } for(;i < lowerIndex;i++) expiries[i] -= n; expiries[i] -= n; } /* Multichannel compression is linked via the absolute maximum of all * channels. */ static void LinkChannels(Compressor *Comp, const ALsizei SamplesToDo, ALfloat (*restrict OutBuffer)[BUFFERSIZE]) { const ALsizei index = Comp->LookAhead; const ALsizei numChans = Comp->NumChans; ALfloat *restrict sideChain = Comp->SideChain; ALsizei c, i; ASSUME(SamplesToDo > 0); ASSUME(numChans > 0); for(i = 0;i < SamplesToDo;i++) sideChain[index + i] = 0.0f; for(c = 0;c < numChans;c++) { ALsizei offset = index; for(i = 0;i < SamplesToDo;i++) { sideChain[offset] = maxf(sideChain[offset], fabsf(OutBuffer[c][i])); ++offset; } } } /* This calculates the squared crest factor of the control signal for the * basic automation of the attack/release times. As suggested by the paper, * it uses an instantaneous squared peak detector and a squared RMS detector * both with 200ms release times. */ static void CrestDetector(Compressor *Comp, const ALsizei SamplesToDo) { const ALfloat a_crest = Comp->CrestCoeff; const ALsizei index = Comp->LookAhead; const ALfloat *restrict sideChain = Comp->SideChain; ALfloat *restrict crestFactor = Comp->CrestFactor; ALfloat y2_peak = Comp->LastPeakSq; ALfloat y2_rms = Comp->LastRmsSq; ALsizei i; ASSUME(SamplesToDo > 0); for(i = 0;i < SamplesToDo;i++) { ALfloat x_abs = sideChain[index + i]; ALfloat x2 = maxf(0.000001f, x_abs * x_abs); y2_peak = maxf(x2, lerp(x2, y2_peak, a_crest)); y2_rms = lerp(x2, y2_rms, a_crest); crestFactor[i] = y2_peak / y2_rms; } Comp->LastPeakSq = y2_peak; Comp->LastRmsSq = y2_rms; } /* The side-chain starts with a simple peak detector (based on the absolute * value of the incoming signal) and performs most of its operations in the * log domain. */ static void PeakDetector(Compressor *Comp, const ALsizei SamplesToDo) { const ALsizei index = Comp->LookAhead; ALfloat *restrict sideChain = Comp->SideChain; ALsizei i; ASSUME(SamplesToDo > 0); for(i = 0;i < SamplesToDo;i++) { const ALuint offset = index + i; const ALfloat x_abs = sideChain[offset]; sideChain[offset] = logf(maxf(0.000001f, x_abs)); } } /* An optional hold can be used to extend the peak detector so it can more * solidly detect fast transients. This is best used when operating as a * limiter. */ static void PeakHoldDetector(Compressor *Comp, const ALsizei SamplesToDo) { const ALsizei index = Comp->LookAhead; ALfloat *restrict sideChain = Comp->SideChain; SlidingHold *hold = Comp->Hold; ALsizei i; ASSUME(SamplesToDo > 0); for(i = 0;i < SamplesToDo;i++) { const ALsizei offset = index + i; const ALfloat x_abs = sideChain[offset]; const ALfloat x_G = logf(maxf(0.000001f, x_abs)); sideChain[offset] = UpdateSlidingHold(hold, i, x_G); } ShiftSlidingHold(hold, SamplesToDo); } /* This is the heart of the feed-forward compressor. It operates in the log * domain (to better match human hearing) and can apply some basic automation * to knee width, attack/release times, make-up/post gain, and clipping * reduction. */ static void GainCompressor(Compressor *Comp, const ALsizei SamplesToDo) { const bool autoKnee = Comp->Auto.Knee; const bool autoAttack = Comp->Auto.Attack; const bool autoRelease = Comp->Auto.Release; const bool autoPostGain = Comp->Auto.PostGain; const bool autoDeclip = Comp->Auto.Declip; const ALsizei lookAhead = Comp->LookAhead; const ALfloat threshold = Comp->Threshold; const ALfloat slope = Comp->Slope; const ALfloat attack = Comp->Attack; const ALfloat release = Comp->Release; const ALfloat c_est = Comp->GainEstimate; const ALfloat a_adp = Comp->AdaptCoeff; const ALfloat *restrict crestFactor = Comp->CrestFactor; ALfloat *restrict sideChain = Comp->SideChain; ALfloat postGain = Comp->PostGain; ALfloat knee = Comp->Knee; ALfloat t_att = attack; ALfloat t_rel = release - attack; ALfloat a_att = expf(-1.0f / t_att); ALfloat a_rel = expf(-1.0f / t_rel); ALfloat y_1 = Comp->LastRelease; ALfloat y_L = Comp->LastAttack; ALfloat c_dev = Comp->LastGainDev; ALsizei i; ASSUME(SamplesToDo > 0); for(i = 0;i < SamplesToDo;i++) { const ALfloat y2_crest = crestFactor[i]; const ALfloat x_G = sideChain[lookAhead + i]; const ALfloat x_over = x_G - threshold; ALfloat knee_h; ALfloat y_G; ALfloat x_L; if(autoKnee) knee = maxf(0.0f, 2.5f * (c_dev + c_est)); knee_h = 0.5f * knee; /* This is the gain computer. It applies a static compression curve * to the control signal. */ if(x_over <= -knee_h) y_G = 0.0f; else if(fabsf(x_over) < knee_h) y_G = (x_over + knee_h) * (x_over + knee_h) / (2.0f * knee); else y_G = x_over; x_L = -slope * y_G; if(autoAttack) { t_att = 2.0f * attack / y2_crest; a_att = expf(-1.0f / t_att); } if(autoRelease) { t_rel = 2.0f * release / y2_crest - t_att; a_rel = expf(-1.0f / t_rel); } /* Gain smoothing (ballistics) is done via a smooth decoupled peak * detector. The attack time is subtracted from the release time * above to compensate for the chained operating mode. */ y_1 = maxf(x_L, lerp(x_L, y_1, a_rel)); y_L = lerp(y_1, y_L, a_att); /* Knee width and make-up gain automation make use of a smoothed * measurement of deviation between the control signal and estimate. * The estimate is also used to bias the measurement to hot-start its * average. */ c_dev = lerp(-y_L - c_est, c_dev, a_adp); if(autoPostGain) { /* Clipping reduction is only viable when make-up gain is being * automated. It modifies the deviation to further attenuate the * control signal when clipping is detected. The adaptation * time is sufficiently long enough to suppress further clipping * at the same output level. */ if(autoDeclip) c_dev = maxf(c_dev, sideChain[i] - y_L - threshold - c_est); postGain = -(c_dev + c_est); } sideChain[i] = expf(postGain - y_L); } Comp->LastRelease = y_1; Comp->LastAttack = y_L; Comp->LastGainDev = c_dev; } /* Combined with the hold time, a look-ahead delay can improve handling of * fast transients by allowing the envelope time to converge prior to * reaching the offending impulse. This is best used when operating as a * limiter. */ static void SignalDelay(Compressor *Comp, const ALsizei SamplesToDo, ALfloat (*restrict OutBuffer)[BUFFERSIZE]) { const ALsizei mask = BUFFERSIZE - 1; const ALsizei numChans = Comp->NumChans; const ALsizei indexIn = Comp->DelayIndex; const ALsizei indexOut = Comp->DelayIndex - Comp->LookAhead; ALfloat (*restrict delay)[BUFFERSIZE] = Comp->Delay; ALsizei c, i; ASSUME(SamplesToDo > 0); ASSUME(numChans > 0); for(c = 0;c < numChans;c++) { for(i = 0;i < SamplesToDo;i++) { ALfloat sig = OutBuffer[c][i]; OutBuffer[c][i] = delay[c][(indexOut + i) & mask]; delay[c][(indexIn + i) & mask] = sig; } } Comp->DelayIndex = (indexIn + SamplesToDo) & mask; } /* The compressor is initialized with the following settings: * * NumChans - Number of channels to process. * SampleRate - Sample rate to process. * AutoKnee - Whether to automate the knee width parameter. * AutoAttack - Whether to automate the attack time parameter. * AutoRelease - Whether to automate the release time parameter. * AutoPostGain - Whether to automate the make-up (post) gain parameter. * AutoDeclip - Whether to automate clipping reduction. Ignored when * not automating make-up gain. * LookAheadTime - Look-ahead time (in seconds). * HoldTime - Peak hold-time (in seconds). * PreGainDb - Gain applied before detection (in dB). * PostGainDb - Make-up gain applied after compression (in dB). * ThresholdDb - Triggering threshold (in dB). * Ratio - Compression ratio (x:1). Set to INFINITY for true * limiting. Ignored when automating knee width. * KneeDb - Knee width (in dB). Ignored when automating knee * width. * AttackTimeMin - Attack time (in seconds). Acts as a maximum when * automating attack time. * ReleaseTimeMin - Release time (in seconds). Acts as a maximum when * automating release time. */ Compressor* CompressorInit(const ALsizei NumChans, const ALuint SampleRate, const ALboolean AutoKnee, const ALboolean AutoAttack, const ALboolean AutoRelease, const ALboolean AutoPostGain, const ALboolean AutoDeclip, const ALfloat LookAheadTime, const ALfloat HoldTime, const ALfloat PreGainDb, const ALfloat PostGainDb, const ALfloat ThresholdDb, const ALfloat Ratio, const ALfloat KneeDb, const ALfloat AttackTime, const ALfloat ReleaseTime) { Compressor *Comp; ALsizei lookAhead; ALsizei hold; size_t size; lookAhead = (ALsizei)clampf(roundf(LookAheadTime*SampleRate), 0.0f, BUFFERSIZE-1); hold = (ALsizei)clampf(roundf(HoldTime*SampleRate), 0.0f, BUFFERSIZE-1); /* The sliding hold implementation doesn't handle a length of 1. A 1-sample * hold is useless anyway, it would only ever give back what was just given * to it. */ if(hold == 1) hold = 0; size = sizeof(*Comp); if(lookAhead > 0) { size += sizeof(*Comp->Delay) * NumChans; if(hold > 0) size += sizeof(*Comp->Hold); } Comp = al_calloc(16, size); Comp->NumChans = NumChans; Comp->SampleRate = SampleRate; Comp->Auto.Knee = AutoKnee; Comp->Auto.Attack = AutoAttack; Comp->Auto.Release = AutoRelease; Comp->Auto.PostGain = AutoPostGain; Comp->Auto.Declip = AutoPostGain && AutoDeclip; Comp->LookAhead = lookAhead; Comp->PreGain = powf(10.0f, PreGainDb / 20.0f); Comp->PostGain = PostGainDb * logf(10.0f) / 20.0f; Comp->Threshold = ThresholdDb * logf(10.0f) / 20.0f; Comp->Slope = 1.0f / maxf(1.0f, Ratio) - 1.0f; Comp->Knee = maxf(0.0f, KneeDb * logf(10.0f) / 20.0f); Comp->Attack = maxf(1.0f, AttackTime * SampleRate); Comp->Release = maxf(1.0f, ReleaseTime * SampleRate); /* Knee width automation actually treats the compressor as a limiter. By * varying the knee width, it can effectively be seen as applying * compression over a wide range of ratios. */ if(AutoKnee) Comp->Slope = -1.0f; if(lookAhead > 0) { if(hold > 0) { Comp->Hold = (SlidingHold*)(Comp + 1); Comp->Hold->Values[0] = -INFINITY; Comp->Hold->Expiries[0] = hold; Comp->Hold->Length = hold; Comp->Delay = (ALfloat(*)[])(Comp->Hold + 1); } else { Comp->Delay = (ALfloat(*)[])(Comp + 1); } } Comp->CrestCoeff = expf(-1.0f / (0.200f * SampleRate)); // 200ms Comp->GainEstimate = Comp->Threshold * -0.5f * Comp->Slope; Comp->AdaptCoeff = expf(-1.0f / (2.0f * SampleRate)); // 2s return Comp; } void ApplyCompression(Compressor *Comp, const ALsizei SamplesToDo, ALfloat (*restrict OutBuffer)[BUFFERSIZE]) { const ALsizei numChans = Comp->NumChans; const ALfloat preGain = Comp->PreGain; ALfloat *restrict sideChain; ALsizei c, i; ASSUME(SamplesToDo > 0); ASSUME(numChans > 0); if(preGain != 1.0f) { for(c = 0;c < numChans;c++) { for(i = 0;i < SamplesToDo;i++) OutBuffer[c][i] *= preGain; } } LinkChannels(Comp, SamplesToDo, OutBuffer); if(Comp->Auto.Attack || Comp->Auto.Release) CrestDetector(Comp, SamplesToDo); if(Comp->Hold) PeakHoldDetector(Comp, SamplesToDo); else PeakDetector(Comp, SamplesToDo); GainCompressor(Comp, SamplesToDo); if(Comp->Delay) SignalDelay(Comp, SamplesToDo, OutBuffer); sideChain = Comp->SideChain; for(c = 0;c < numChans;c++) { for(i = 0;i < SamplesToDo;i++) OutBuffer[c][i] *= sideChain[i]; } memmove(sideChain, sideChain+SamplesToDo, Comp->LookAhead*sizeof(ALfloat)); } ALsizei GetCompressorLookAhead(const Compressor *Comp) { return Comp->LookAhead; } openal-soft-openal-soft-1.19.1/Alc/mastering.h000066400000000000000000000042061335774445300211410ustar00rootroot00000000000000#ifndef MASTERING_H #define MASTERING_H #include "AL/al.h" /* For BUFFERSIZE. */ #include "alMain.h" struct Compressor; /* The compressor is initialized with the following settings: * * NumChans - Number of channels to process. * SampleRate - Sample rate to process. * AutoKnee - Whether to automate the knee width parameter. * AutoAttack - Whether to automate the attack time parameter. * AutoRelease - Whether to automate the release time parameter. * AutoPostGain - Whether to automate the make-up (post) gain parameter. * AutoDeclip - Whether to automate clipping reduction. Ignored when * not automating make-up gain. * LookAheadTime - Look-ahead time (in seconds). * HoldTime - Peak hold-time (in seconds). * PreGainDb - Gain applied before detection (in dB). * PostGainDb - Make-up gain applied after compression (in dB). * ThresholdDb - Triggering threshold (in dB). * Ratio - Compression ratio (x:1). Set to INFINIFTY for true * limiting. Ignored when automating knee width. * KneeDb - Knee width (in dB). Ignored when automating knee * width. * AttackTimeMin - Attack time (in seconds). Acts as a maximum when * automating attack time. * ReleaseTimeMin - Release time (in seconds). Acts as a maximum when * automating release time. */ struct Compressor* CompressorInit(const ALsizei NumChans, const ALuint SampleRate, const ALboolean AutoKnee, const ALboolean AutoAttack, const ALboolean AutoRelease, const ALboolean AutoPostGain, const ALboolean AutoDeclip, const ALfloat LookAheadTime, const ALfloat HoldTime, const ALfloat PreGainDb, const ALfloat PostGainDb, const ALfloat ThresholdDb, const ALfloat Ratio, const ALfloat KneeDb, const ALfloat AttackTime, const ALfloat ReleaseTime); void ApplyCompression(struct Compressor *Comp, const ALsizei SamplesToDo, ALfloat (*restrict OutBuffer)[BUFFERSIZE]); ALsizei GetCompressorLookAhead(const struct Compressor *Comp); #endif /* MASTERING_H */ openal-soft-openal-soft-1.19.1/Alc/mixer/000077500000000000000000000000001335774445300201215ustar00rootroot00000000000000openal-soft-openal-soft-1.19.1/Alc/mixer/defs.h000066400000000000000000000147531335774445300212250ustar00rootroot00000000000000#ifndef MIXER_DEFS_H #define MIXER_DEFS_H #include "AL/alc.h" #include "AL/al.h" #include "alMain.h" #include "alu.h" struct MixGains; struct MixHrtfParams; struct HrtfState; /* C resamplers */ const ALfloat *Resample_copy_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen); const ALfloat *Resample_point_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen); const ALfloat *Resample_lerp_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen); const ALfloat *Resample_cubic_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen); const ALfloat *Resample_bsinc_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen); /* C mixers */ void MixHrtf_C(ALfloat *restrict LeftOut, ALfloat *restrict RightOut, const ALfloat *data, ALsizei Offset, ALsizei OutPos, const ALsizei IrSize, struct MixHrtfParams *hrtfparams, struct HrtfState *hrtfstate, ALsizei BufferSize); void MixHrtfBlend_C(ALfloat *restrict LeftOut, ALfloat *restrict RightOut, const ALfloat *data, ALsizei Offset, ALsizei OutPos, const ALsizei IrSize, const HrtfParams *oldparams, MixHrtfParams *newparams, HrtfState *hrtfstate, ALsizei BufferSize); void MixDirectHrtf_C(ALfloat *restrict LeftOut, ALfloat *restrict RightOut, const ALfloat *data, ALsizei Offset, const ALsizei IrSize, const ALfloat (*restrict Coeffs)[2], ALfloat (*restrict Values)[2], ALsizei BufferSize); void Mix_C(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos, ALsizei BufferSize); void MixRow_C(ALfloat *OutBuffer, const ALfloat *Gains, const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans, ALsizei InPos, ALsizei BufferSize); /* SSE mixers */ void MixHrtf_SSE(ALfloat *restrict LeftOut, ALfloat *restrict RightOut, const ALfloat *data, ALsizei Offset, ALsizei OutPos, const ALsizei IrSize, struct MixHrtfParams *hrtfparams, struct HrtfState *hrtfstate, ALsizei BufferSize); void MixHrtfBlend_SSE(ALfloat *restrict LeftOut, ALfloat *restrict RightOut, const ALfloat *data, ALsizei Offset, ALsizei OutPos, const ALsizei IrSize, const HrtfParams *oldparams, MixHrtfParams *newparams, HrtfState *hrtfstate, ALsizei BufferSize); void MixDirectHrtf_SSE(ALfloat *restrict LeftOut, ALfloat *restrict RightOut, const ALfloat *data, ALsizei Offset, const ALsizei IrSize, const ALfloat (*restrict Coeffs)[2], ALfloat (*restrict Values)[2], ALsizei BufferSize); void Mix_SSE(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos, ALsizei BufferSize); void MixRow_SSE(ALfloat *OutBuffer, const ALfloat *Gains, const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans, ALsizei InPos, ALsizei BufferSize); /* SSE resamplers */ inline void InitiatePositionArrays(ALsizei frac, ALint increment, ALsizei *restrict frac_arr, ALsizei *restrict pos_arr, ALsizei size) { ALsizei i; pos_arr[0] = 0; frac_arr[0] = frac; for(i = 1;i < size;i++) { ALint frac_tmp = frac_arr[i-1] + increment; pos_arr[i] = pos_arr[i-1] + (frac_tmp>>FRACTIONBITS); frac_arr[i] = frac_tmp&FRACTIONMASK; } } const ALfloat *Resample_lerp_SSE2(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei numsamples); const ALfloat *Resample_lerp_SSE41(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei numsamples); const ALfloat *Resample_bsinc_SSE(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen); /* Neon mixers */ void MixHrtf_Neon(ALfloat *restrict LeftOut, ALfloat *restrict RightOut, const ALfloat *data, ALsizei Offset, ALsizei OutPos, const ALsizei IrSize, struct MixHrtfParams *hrtfparams, struct HrtfState *hrtfstate, ALsizei BufferSize); void MixHrtfBlend_Neon(ALfloat *restrict LeftOut, ALfloat *restrict RightOut, const ALfloat *data, ALsizei Offset, ALsizei OutPos, const ALsizei IrSize, const HrtfParams *oldparams, MixHrtfParams *newparams, HrtfState *hrtfstate, ALsizei BufferSize); void MixDirectHrtf_Neon(ALfloat *restrict LeftOut, ALfloat *restrict RightOut, const ALfloat *data, ALsizei Offset, const ALsizei IrSize, const ALfloat (*restrict Coeffs)[2], ALfloat (*restrict Values)[2], ALsizei BufferSize); void Mix_Neon(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos, ALsizei BufferSize); void MixRow_Neon(ALfloat *OutBuffer, const ALfloat *Gains, const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans, ALsizei InPos, ALsizei BufferSize); /* Neon resamplers */ const ALfloat *Resample_lerp_Neon(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei numsamples); const ALfloat *Resample_bsinc_Neon(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen); #endif /* MIXER_DEFS_H */ openal-soft-openal-soft-1.19.1/Alc/mixer/hrtf_inc.c000066400000000000000000000110141335774445300220560ustar00rootroot00000000000000#include "config.h" #include "alMain.h" #include "alSource.h" #include "hrtf.h" #include "align.h" #include "alu.h" #include "defs.h" static inline void ApplyCoeffs(ALsizei Offset, ALfloat (*restrict Values)[2], const ALsizei irSize, const ALfloat (*restrict Coeffs)[2], ALfloat left, ALfloat right); void MixHrtf(ALfloat *restrict LeftOut, ALfloat *restrict RightOut, const ALfloat *data, ALsizei Offset, ALsizei OutPos, const ALsizei IrSize, MixHrtfParams *hrtfparams, HrtfState *hrtfstate, ALsizei BufferSize) { const ALfloat (*Coeffs)[2] = ASSUME_ALIGNED(hrtfparams->Coeffs, 16); const ALsizei Delay[2] = { hrtfparams->Delay[0], hrtfparams->Delay[1] }; const ALfloat gainstep = hrtfparams->GainStep; const ALfloat gain = hrtfparams->Gain; ALfloat g, stepcount = 0.0f; ALfloat left, right; ALsizei i; ASSUME(IrSize >= 4); ASSUME(BufferSize > 0); LeftOut += OutPos; RightOut += OutPos; for(i = 0;i < BufferSize;i++) { hrtfstate->History[Offset&HRTF_HISTORY_MASK] = *(data++); g = gain + gainstep*stepcount; left = hrtfstate->History[(Offset-Delay[0])&HRTF_HISTORY_MASK]*g; right = hrtfstate->History[(Offset-Delay[1])&HRTF_HISTORY_MASK]*g; hrtfstate->Values[(Offset+IrSize-1)&HRIR_MASK][0] = 0.0f; hrtfstate->Values[(Offset+IrSize-1)&HRIR_MASK][1] = 0.0f; ApplyCoeffs(Offset, hrtfstate->Values, IrSize, Coeffs, left, right); *(LeftOut++) += hrtfstate->Values[Offset&HRIR_MASK][0]; *(RightOut++) += hrtfstate->Values[Offset&HRIR_MASK][1]; stepcount += 1.0f; Offset++; } hrtfparams->Gain = gain + gainstep*stepcount; } void MixHrtfBlend(ALfloat *restrict LeftOut, ALfloat *restrict RightOut, const ALfloat *data, ALsizei Offset, ALsizei OutPos, const ALsizei IrSize, const HrtfParams *oldparams, MixHrtfParams *newparams, HrtfState *hrtfstate, ALsizei BufferSize) { const ALfloat (*OldCoeffs)[2] = ASSUME_ALIGNED(oldparams->Coeffs, 16); const ALsizei OldDelay[2] = { oldparams->Delay[0], oldparams->Delay[1] }; const ALfloat oldGain = oldparams->Gain; const ALfloat oldGainStep = -oldGain / (ALfloat)BufferSize; const ALfloat (*NewCoeffs)[2] = ASSUME_ALIGNED(newparams->Coeffs, 16); const ALsizei NewDelay[2] = { newparams->Delay[0], newparams->Delay[1] }; const ALfloat newGain = newparams->Gain; const ALfloat newGainStep = newparams->GainStep; ALfloat g, stepcount = 0.0f; ALfloat left, right; ALsizei i; ASSUME(IrSize >= 4); ASSUME(BufferSize > 0); LeftOut += OutPos; RightOut += OutPos; for(i = 0;i < BufferSize;i++) { hrtfstate->Values[(Offset+IrSize-1)&HRIR_MASK][0] = 0.0f; hrtfstate->Values[(Offset+IrSize-1)&HRIR_MASK][1] = 0.0f; hrtfstate->History[Offset&HRTF_HISTORY_MASK] = *(data++); g = oldGain + oldGainStep*stepcount; left = hrtfstate->History[(Offset-OldDelay[0])&HRTF_HISTORY_MASK]*g; right = hrtfstate->History[(Offset-OldDelay[1])&HRTF_HISTORY_MASK]*g; ApplyCoeffs(Offset, hrtfstate->Values, IrSize, OldCoeffs, left, right); g = newGain + newGainStep*stepcount; left = hrtfstate->History[(Offset-NewDelay[0])&HRTF_HISTORY_MASK]*g; right = hrtfstate->History[(Offset-NewDelay[1])&HRTF_HISTORY_MASK]*g; ApplyCoeffs(Offset, hrtfstate->Values, IrSize, NewCoeffs, left, right); *(LeftOut++) += hrtfstate->Values[Offset&HRIR_MASK][0]; *(RightOut++) += hrtfstate->Values[Offset&HRIR_MASK][1]; stepcount += 1.0f; Offset++; } newparams->Gain = newGain + newGainStep*stepcount; } void MixDirectHrtf(ALfloat *restrict LeftOut, ALfloat *restrict RightOut, const ALfloat *data, ALsizei Offset, const ALsizei IrSize, const ALfloat (*restrict Coeffs)[2], ALfloat (*restrict Values)[2], ALsizei BufferSize) { ALfloat insample; ALsizei i; ASSUME(IrSize >= 4); ASSUME(BufferSize > 0); for(i = 0;i < BufferSize;i++) { Values[(Offset+IrSize)&HRIR_MASK][0] = 0.0f; Values[(Offset+IrSize)&HRIR_MASK][1] = 0.0f; Offset++; insample = *(data++); ApplyCoeffs(Offset, Values, IrSize, Coeffs, insample, insample); *(LeftOut++) += Values[Offset&HRIR_MASK][0]; *(RightOut++) += Values[Offset&HRIR_MASK][1]; } } openal-soft-openal-soft-1.19.1/Alc/mixer/mixer_c.c000066400000000000000000000145031335774445300217160ustar00rootroot00000000000000#include "config.h" #include #include "alMain.h" #include "alu.h" #include "alSource.h" #include "alAuxEffectSlot.h" #include "defs.h" static inline ALfloat do_point(const InterpState* UNUSED(state), const ALfloat *restrict vals, ALsizei UNUSED(frac)) { return vals[0]; } static inline ALfloat do_lerp(const InterpState* UNUSED(state), const ALfloat *restrict vals, ALsizei frac) { return lerp(vals[0], vals[1], frac * (1.0f/FRACTIONONE)); } static inline ALfloat do_cubic(const InterpState* UNUSED(state), const ALfloat *restrict vals, ALsizei frac) { return cubic(vals[0], vals[1], vals[2], vals[3], frac * (1.0f/FRACTIONONE)); } static inline ALfloat do_bsinc(const InterpState *state, const ALfloat *restrict vals, ALsizei frac) { const ALfloat *fil, *scd, *phd, *spd; ALsizei j_f, pi; ALfloat pf, r; ASSUME(state->bsinc.m > 0); // Calculate the phase index and factor. #define FRAC_PHASE_BITDIFF (FRACTIONBITS-BSINC_PHASE_BITS) pi = frac >> FRAC_PHASE_BITDIFF; pf = (frac & ((1<bsinc.filter + state->bsinc.m*pi*4, 16); scd = ASSUME_ALIGNED(fil + state->bsinc.m, 16); phd = ASSUME_ALIGNED(scd + state->bsinc.m, 16); spd = ASSUME_ALIGNED(phd + state->bsinc.m, 16); // Apply the scale and phase interpolated filter. r = 0.0f; for(j_f = 0;j_f < state->bsinc.m;j_f++) r += (fil[j_f] + state->bsinc.sf*scd[j_f] + pf*(phd[j_f] + state->bsinc.sf*spd[j_f])) * vals[j_f]; return r; } const ALfloat *Resample_copy_C(const InterpState* UNUSED(state), const ALfloat *restrict src, ALsizei UNUSED(frac), ALint UNUSED(increment), ALfloat *restrict dst, ALsizei numsamples) { #if defined(HAVE_SSE) || defined(HAVE_NEON) /* Avoid copying the source data if it's aligned like the destination. */ if((((intptr_t)src)&15) == (((intptr_t)dst)&15)) return src; #endif memcpy(dst, src, numsamples*sizeof(ALfloat)); return dst; } #define DECL_TEMPLATE(Tag, Sampler, O) \ const ALfloat *Resample_##Tag##_C(const InterpState *state, \ const ALfloat *restrict src, ALsizei frac, ALint increment, \ ALfloat *restrict dst, ALsizei numsamples) \ { \ const InterpState istate = *state; \ ALsizei i; \ \ ASSUME(numsamples > 0); \ \ src -= O; \ for(i = 0;i < numsamples;i++) \ { \ dst[i] = Sampler(&istate, src, frac); \ \ frac += increment; \ src += frac>>FRACTIONBITS; \ frac &= FRACTIONMASK; \ } \ return dst; \ } DECL_TEMPLATE(point, do_point, 0) DECL_TEMPLATE(lerp, do_lerp, 0) DECL_TEMPLATE(cubic, do_cubic, 1) DECL_TEMPLATE(bsinc, do_bsinc, istate.bsinc.l) #undef DECL_TEMPLATE static inline void ApplyCoeffs(ALsizei Offset, ALfloat (*restrict Values)[2], const ALsizei IrSize, const ALfloat (*restrict Coeffs)[2], ALfloat left, ALfloat right) { ALsizei c; for(c = 0;c < IrSize;c++) { const ALsizei off = (Offset+c)&HRIR_MASK; Values[off][0] += Coeffs[c][0] * left; Values[off][1] += Coeffs[c][1] * right; } } #define MixHrtf MixHrtf_C #define MixHrtfBlend MixHrtfBlend_C #define MixDirectHrtf MixDirectHrtf_C #include "hrtf_inc.c" void Mix_C(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos, ALsizei BufferSize) { const ALfloat delta = (Counter > 0) ? 1.0f/(ALfloat)Counter : 0.0f; ALsizei c; ASSUME(OutChans > 0); ASSUME(BufferSize > 0); for(c = 0;c < OutChans;c++) { ALsizei pos = 0; ALfloat gain = CurrentGains[c]; const ALfloat diff = TargetGains[c] - gain; if(fabsf(diff) > FLT_EPSILON) { ALsizei minsize = mini(BufferSize, Counter); const ALfloat step = diff * delta; ALfloat step_count = 0.0f; for(;pos < minsize;pos++) { OutBuffer[c][OutPos+pos] += data[pos] * (gain + step*step_count); step_count += 1.0f; } if(pos == Counter) gain = TargetGains[c]; else gain += step*step_count; CurrentGains[c] = gain; } if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD)) continue; for(;pos < BufferSize;pos++) OutBuffer[c][OutPos+pos] += data[pos]*gain; } } /* Basically the inverse of the above. Rather than one input going to multiple * outputs (each with its own gain), it's multiple inputs (each with its own * gain) going to one output. This applies one row (vs one column) of a matrix * transform. And as the matrices are more or less static once set up, no * stepping is necessary. */ void MixRow_C(ALfloat *OutBuffer, const ALfloat *Gains, const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans, ALsizei InPos, ALsizei BufferSize) { ALsizei c, i; ASSUME(InChans > 0); ASSUME(BufferSize > 0); for(c = 0;c < InChans;c++) { const ALfloat gain = Gains[c]; if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD)) continue; for(i = 0;i < BufferSize;i++) OutBuffer[i] += data[c][InPos+i] * gain; } } openal-soft-openal-soft-1.19.1/Alc/mixer/mixer_neon.c000066400000000000000000000232751335774445300224410ustar00rootroot00000000000000#include "config.h" #include #include "AL/al.h" #include "AL/alc.h" #include "alMain.h" #include "alu.h" #include "hrtf.h" #include "defs.h" const ALfloat *Resample_lerp_Neon(const InterpState* UNUSED(state), const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei numsamples) { const int32x4_t increment4 = vdupq_n_s32(increment*4); const float32x4_t fracOne4 = vdupq_n_f32(1.0f/FRACTIONONE); const int32x4_t fracMask4 = vdupq_n_s32(FRACTIONMASK); alignas(16) ALsizei pos_[4], frac_[4]; int32x4_t pos4, frac4; ALsizei todo, pos, i; ASSUME(numsamples > 0); InitiatePositionArrays(frac, increment, frac_, pos_, 4); frac4 = vld1q_s32(frac_); pos4 = vld1q_s32(pos_); todo = numsamples & ~3; for(i = 0;i < todo;i += 4) { const int pos0 = vgetq_lane_s32(pos4, 0); const int pos1 = vgetq_lane_s32(pos4, 1); const int pos2 = vgetq_lane_s32(pos4, 2); const int pos3 = vgetq_lane_s32(pos4, 3); const float32x4_t val1 = (float32x4_t){src[pos0], src[pos1], src[pos2], src[pos3]}; const float32x4_t val2 = (float32x4_t){src[pos0+1], src[pos1+1], src[pos2+1], src[pos3+1]}; /* val1 + (val2-val1)*mu */ const float32x4_t r0 = vsubq_f32(val2, val1); const float32x4_t mu = vmulq_f32(vcvtq_f32_s32(frac4), fracOne4); const float32x4_t out = vmlaq_f32(val1, mu, r0); vst1q_f32(&dst[i], out); frac4 = vaddq_s32(frac4, increment4); pos4 = vaddq_s32(pos4, vshrq_n_s32(frac4, FRACTIONBITS)); frac4 = vandq_s32(frac4, fracMask4); } /* NOTE: These four elements represent the position *after* the last four * samples, so the lowest element is the next position to resample. */ pos = vgetq_lane_s32(pos4, 0); frac = vgetq_lane_s32(frac4, 0); for(;i < numsamples;++i) { dst[i] = lerp(src[pos], src[pos+1], frac * (1.0f/FRACTIONONE)); frac += increment; pos += frac>>FRACTIONBITS; frac &= FRACTIONMASK; } return dst; } const ALfloat *Resample_bsinc_Neon(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen) { const ALfloat *const filter = state->bsinc.filter; const float32x4_t sf4 = vdupq_n_f32(state->bsinc.sf); const ALsizei m = state->bsinc.m; const float32x4_t *fil, *scd, *phd, *spd; ALsizei pi, i, j, offset; float32x4_t r4; ALfloat pf; ASSUME(m > 0); ASSUME(dstlen > 0); src -= state->bsinc.l; for(i = 0;i < dstlen;i++) { // Calculate the phase index and factor. #define FRAC_PHASE_BITDIFF (FRACTIONBITS-BSINC_PHASE_BITS) pi = frac >> FRAC_PHASE_BITDIFF; pf = (frac & ((1<> 2; const float32x4_t pf4 = vdupq_n_f32(pf); ASSUME(count > 0); for(j = 0;j < count;j++) { /* f = ((fil + sf*scd) + pf*(phd + sf*spd)) */ const float32x4_t f4 = vmlaq_f32( vmlaq_f32(fil[j], sf4, scd[j]), pf4, vmlaq_f32(phd[j], sf4, spd[j]) ); /* r += f*src */ r4 = vmlaq_f32(r4, f4, vld1q_f32(&src[j*4])); } } r4 = vaddq_f32(r4, vcombine_f32(vrev64_f32(vget_high_f32(r4)), vrev64_f32(vget_low_f32(r4)))); dst[i] = vget_lane_f32(vadd_f32(vget_low_f32(r4), vget_high_f32(r4)), 0); frac += increment; src += frac>>FRACTIONBITS; frac &= FRACTIONMASK; } return dst; } static inline void ApplyCoeffs(ALsizei Offset, ALfloat (*restrict Values)[2], const ALsizei IrSize, const ALfloat (*restrict Coeffs)[2], ALfloat left, ALfloat right) { ALsizei c; float32x4_t leftright4; { float32x2_t leftright2 = vdup_n_f32(0.0); leftright2 = vset_lane_f32(left, leftright2, 0); leftright2 = vset_lane_f32(right, leftright2, 1); leftright4 = vcombine_f32(leftright2, leftright2); } Values = ASSUME_ALIGNED(Values, 16); Coeffs = ASSUME_ALIGNED(Coeffs, 16); for(c = 0;c < IrSize;c += 2) { const ALsizei o0 = (Offset+c)&HRIR_MASK; const ALsizei o1 = (o0+1)&HRIR_MASK; float32x4_t vals = vcombine_f32(vld1_f32((float32_t*)&Values[o0][0]), vld1_f32((float32_t*)&Values[o1][0])); float32x4_t coefs = vld1q_f32((float32_t*)&Coeffs[c][0]); vals = vmlaq_f32(vals, coefs, leftright4); vst1_f32((float32_t*)&Values[o0][0], vget_low_f32(vals)); vst1_f32((float32_t*)&Values[o1][0], vget_high_f32(vals)); } } #define MixHrtf MixHrtf_Neon #define MixHrtfBlend MixHrtfBlend_Neon #define MixDirectHrtf MixDirectHrtf_Neon #include "hrtf_inc.c" void Mix_Neon(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos, ALsizei BufferSize) { const ALfloat delta = (Counter > 0) ? 1.0f/(ALfloat)Counter : 0.0f; ALsizei c; ASSUME(OutChans > 0); ASSUME(BufferSize > 0); data = ASSUME_ALIGNED(data, 16); OutBuffer = ASSUME_ALIGNED(OutBuffer, 16); for(c = 0;c < OutChans;c++) { ALsizei pos = 0; ALfloat gain = CurrentGains[c]; const ALfloat diff = TargetGains[c] - gain; if(fabsf(diff) > FLT_EPSILON) { ALsizei minsize = mini(BufferSize, Counter); const ALfloat step = diff * delta; ALfloat step_count = 0.0f; /* Mix with applying gain steps in aligned multiples of 4. */ if(LIKELY(minsize > 3)) { const float32x4_t four4 = vdupq_n_f32(4.0f); const float32x4_t step4 = vdupq_n_f32(step); const float32x4_t gain4 = vdupq_n_f32(gain); float32x4_t step_count4 = vsetq_lane_f32(0.0f, vsetq_lane_f32(1.0f, vsetq_lane_f32(2.0f, vsetq_lane_f32(3.0f, vdupq_n_f32(0.0f), 3), 2), 1), 0 ); ALsizei todo = minsize >> 2; do { const float32x4_t val4 = vld1q_f32(&data[pos]); float32x4_t dry4 = vld1q_f32(&OutBuffer[c][OutPos+pos]); dry4 = vmlaq_f32(dry4, val4, vmlaq_f32(gain4, step4, step_count4)); step_count4 = vaddq_f32(step_count4, four4); vst1q_f32(&OutBuffer[c][OutPos+pos], dry4); pos += 4; } while(--todo); /* NOTE: step_count4 now represents the next four counts after * the last four mixed samples, so the lowest element * represents the next step count to apply. */ step_count = vgetq_lane_f32(step_count4, 0); } /* Mix with applying left over gain steps that aren't aligned multiples of 4. */ for(;pos < minsize;pos++) { OutBuffer[c][OutPos+pos] += data[pos]*(gain + step*step_count); step_count += 1.0f; } if(pos == Counter) gain = TargetGains[c]; else gain += step*step_count; CurrentGains[c] = gain; /* Mix until pos is aligned with 4 or the mix is done. */ minsize = mini(BufferSize, (pos+3)&~3); for(;pos < minsize;pos++) OutBuffer[c][OutPos+pos] += data[pos]*gain; } if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD)) continue; if(LIKELY(BufferSize-pos > 3)) { ALsizei todo = (BufferSize-pos) >> 2; const float32x4_t gain4 = vdupq_n_f32(gain); do { const float32x4_t val4 = vld1q_f32(&data[pos]); float32x4_t dry4 = vld1q_f32(&OutBuffer[c][OutPos+pos]); dry4 = vmlaq_f32(dry4, val4, gain4); vst1q_f32(&OutBuffer[c][OutPos+pos], dry4); pos += 4; } while(--todo); } for(;pos < BufferSize;pos++) OutBuffer[c][OutPos+pos] += data[pos]*gain; } } void MixRow_Neon(ALfloat *OutBuffer, const ALfloat *Gains, const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans, ALsizei InPos, ALsizei BufferSize) { ALsizei c; ASSUME(InChans > 0); ASSUME(BufferSize > 0); for(c = 0;c < InChans;c++) { ALsizei pos = 0; const ALfloat gain = Gains[c]; if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD)) continue; if(LIKELY(BufferSize > 3)) { ALsizei todo = BufferSize >> 2; float32x4_t gain4 = vdupq_n_f32(gain); do { const float32x4_t val4 = vld1q_f32(&data[c][InPos+pos]); float32x4_t dry4 = vld1q_f32(&OutBuffer[pos]); dry4 = vmlaq_f32(dry4, val4, gain4); vst1q_f32(&OutBuffer[pos], dry4); pos += 4; } while(--todo); } for(;pos < BufferSize;pos++) OutBuffer[pos] += data[c][InPos+pos]*gain; } } openal-soft-openal-soft-1.19.1/Alc/mixer/mixer_sse.c000066400000000000000000000206531335774445300222710ustar00rootroot00000000000000#include "config.h" #include #include "AL/al.h" #include "AL/alc.h" #include "alMain.h" #include "alu.h" #include "alSource.h" #include "alAuxEffectSlot.h" #include "defs.h" const ALfloat *Resample_bsinc_SSE(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen) { const ALfloat *const filter = state->bsinc.filter; const __m128 sf4 = _mm_set1_ps(state->bsinc.sf); const ALsizei m = state->bsinc.m; const __m128 *fil, *scd, *phd, *spd; ALsizei pi, i, j, offset; ALfloat pf; __m128 r4; ASSUME(m > 0); ASSUME(dstlen > 0); src -= state->bsinc.l; for(i = 0;i < dstlen;i++) { // Calculate the phase index and factor. #define FRAC_PHASE_BITDIFF (FRACTIONBITS-BSINC_PHASE_BITS) pi = frac >> FRAC_PHASE_BITDIFF; pf = (frac & ((1<> 2; const __m128 pf4 = _mm_set1_ps(pf); ASSUME(count > 0); #define MLA4(x, y, z) _mm_add_ps(x, _mm_mul_ps(y, z)) for(j = 0;j < count;j++) { /* f = ((fil + sf*scd) + pf*(phd + sf*spd)) */ const __m128 f4 = MLA4( MLA4(fil[j], sf4, scd[j]), pf4, MLA4(phd[j], sf4, spd[j]) ); /* r += f*src */ r4 = MLA4(r4, f4, _mm_loadu_ps(&src[j*4])); } #undef MLA4 } r4 = _mm_add_ps(r4, _mm_shuffle_ps(r4, r4, _MM_SHUFFLE(0, 1, 2, 3))); r4 = _mm_add_ps(r4, _mm_movehl_ps(r4, r4)); dst[i] = _mm_cvtss_f32(r4); frac += increment; src += frac>>FRACTIONBITS; frac &= FRACTIONMASK; } return dst; } static inline void ApplyCoeffs(ALsizei Offset, ALfloat (*restrict Values)[2], const ALsizei IrSize, const ALfloat (*restrict Coeffs)[2], ALfloat left, ALfloat right) { const __m128 lrlr = _mm_setr_ps(left, right, left, right); __m128 vals = _mm_setzero_ps(); __m128 coeffs; ALsizei i; Values = ASSUME_ALIGNED(Values, 16); Coeffs = ASSUME_ALIGNED(Coeffs, 16); if((Offset&1)) { const ALsizei o0 = Offset&HRIR_MASK; const ALsizei o1 = (Offset+IrSize-1)&HRIR_MASK; __m128 imp0, imp1; coeffs = _mm_load_ps(&Coeffs[0][0]); vals = _mm_loadl_pi(vals, (__m64*)&Values[o0][0]); imp0 = _mm_mul_ps(lrlr, coeffs); vals = _mm_add_ps(imp0, vals); _mm_storel_pi((__m64*)&Values[o0][0], vals); for(i = 1;i < IrSize-1;i += 2) { const ALsizei o2 = (Offset+i)&HRIR_MASK; coeffs = _mm_load_ps(&Coeffs[i+1][0]); vals = _mm_load_ps(&Values[o2][0]); imp1 = _mm_mul_ps(lrlr, coeffs); imp0 = _mm_shuffle_ps(imp0, imp1, _MM_SHUFFLE(1, 0, 3, 2)); vals = _mm_add_ps(imp0, vals); _mm_store_ps(&Values[o2][0], vals); imp0 = imp1; } vals = _mm_loadl_pi(vals, (__m64*)&Values[o1][0]); imp0 = _mm_movehl_ps(imp0, imp0); vals = _mm_add_ps(imp0, vals); _mm_storel_pi((__m64*)&Values[o1][0], vals); } else { for(i = 0;i < IrSize;i += 2) { const ALsizei o = (Offset + i)&HRIR_MASK; coeffs = _mm_load_ps(&Coeffs[i][0]); vals = _mm_load_ps(&Values[o][0]); vals = _mm_add_ps(vals, _mm_mul_ps(lrlr, coeffs)); _mm_store_ps(&Values[o][0], vals); } } } #define MixHrtf MixHrtf_SSE #define MixHrtfBlend MixHrtfBlend_SSE #define MixDirectHrtf MixDirectHrtf_SSE #include "hrtf_inc.c" void Mix_SSE(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos, ALsizei BufferSize) { const ALfloat delta = (Counter > 0) ? 1.0f/(ALfloat)Counter : 0.0f; ALsizei c; ASSUME(OutChans > 0); ASSUME(BufferSize > 0); for(c = 0;c < OutChans;c++) { ALsizei pos = 0; ALfloat gain = CurrentGains[c]; const ALfloat diff = TargetGains[c] - gain; if(fabsf(diff) > FLT_EPSILON) { ALsizei minsize = mini(BufferSize, Counter); const ALfloat step = diff * delta; ALfloat step_count = 0.0f; /* Mix with applying gain steps in aligned multiples of 4. */ if(LIKELY(minsize > 3)) { const __m128 four4 = _mm_set1_ps(4.0f); const __m128 step4 = _mm_set1_ps(step); const __m128 gain4 = _mm_set1_ps(gain); __m128 step_count4 = _mm_setr_ps(0.0f, 1.0f, 2.0f, 3.0f); ALsizei todo = minsize >> 2; do { const __m128 val4 = _mm_load_ps(&data[pos]); __m128 dry4 = _mm_load_ps(&OutBuffer[c][OutPos+pos]); #define MLA4(x, y, z) _mm_add_ps(x, _mm_mul_ps(y, z)) /* dry += val * (gain + step*step_count) */ dry4 = MLA4(dry4, val4, MLA4(gain4, step4, step_count4)); #undef MLA4 _mm_store_ps(&OutBuffer[c][OutPos+pos], dry4); step_count4 = _mm_add_ps(step_count4, four4); pos += 4; } while(--todo); /* NOTE: step_count4 now represents the next four counts after * the last four mixed samples, so the lowest element * represents the next step count to apply. */ step_count = _mm_cvtss_f32(step_count4); } /* Mix with applying left over gain steps that aren't aligned multiples of 4. */ for(;pos < minsize;pos++) { OutBuffer[c][OutPos+pos] += data[pos]*(gain + step*step_count); step_count += 1.0f; } if(pos == Counter) gain = TargetGains[c]; else gain += step*step_count; CurrentGains[c] = gain; /* Mix until pos is aligned with 4 or the mix is done. */ minsize = mini(BufferSize, (pos+3)&~3); for(;pos < minsize;pos++) OutBuffer[c][OutPos+pos] += data[pos]*gain; } if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD)) continue; if(LIKELY(BufferSize-pos > 3)) { ALsizei todo = (BufferSize-pos) >> 2; const __m128 gain4 = _mm_set1_ps(gain); do { const __m128 val4 = _mm_load_ps(&data[pos]); __m128 dry4 = _mm_load_ps(&OutBuffer[c][OutPos+pos]); dry4 = _mm_add_ps(dry4, _mm_mul_ps(val4, gain4)); _mm_store_ps(&OutBuffer[c][OutPos+pos], dry4); pos += 4; } while(--todo); } for(;pos < BufferSize;pos++) OutBuffer[c][OutPos+pos] += data[pos]*gain; } } void MixRow_SSE(ALfloat *OutBuffer, const ALfloat *Gains, const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans, ALsizei InPos, ALsizei BufferSize) { ALsizei c; ASSUME(InChans > 0); ASSUME(BufferSize > 0); for(c = 0;c < InChans;c++) { ALsizei pos = 0; const ALfloat gain = Gains[c]; if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD)) continue; if(LIKELY(BufferSize > 3)) { ALsizei todo = BufferSize >> 2; const __m128 gain4 = _mm_set1_ps(gain); do { const __m128 val4 = _mm_load_ps(&data[c][InPos+pos]); __m128 dry4 = _mm_load_ps(&OutBuffer[pos]); dry4 = _mm_add_ps(dry4, _mm_mul_ps(val4, gain4)); _mm_store_ps(&OutBuffer[pos], dry4); pos += 4; } while(--todo); } for(;pos < BufferSize;pos++) OutBuffer[pos] += data[c][InPos+pos]*gain; } } openal-soft-openal-soft-1.19.1/Alc/mixer/mixer_sse2.c000066400000000000000000000062221335774445300223470ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2014 by Timothy Arceri . * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include "alu.h" #include "defs.h" const ALfloat *Resample_lerp_SSE2(const InterpState* UNUSED(state), const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei numsamples) { const __m128i increment4 = _mm_set1_epi32(increment*4); const __m128 fracOne4 = _mm_set1_ps(1.0f/FRACTIONONE); const __m128i fracMask4 = _mm_set1_epi32(FRACTIONMASK); alignas(16) ALsizei pos_[4], frac_[4]; __m128i frac4, pos4; ALsizei todo, pos, i; ASSUME(numsamples > 0); InitiatePositionArrays(frac, increment, frac_, pos_, 4); frac4 = _mm_setr_epi32(frac_[0], frac_[1], frac_[2], frac_[3]); pos4 = _mm_setr_epi32(pos_[0], pos_[1], pos_[2], pos_[3]); todo = numsamples & ~3; for(i = 0;i < todo;i += 4) { const int pos0 = _mm_cvtsi128_si32(_mm_shuffle_epi32(pos4, _MM_SHUFFLE(0, 0, 0, 0))); const int pos1 = _mm_cvtsi128_si32(_mm_shuffle_epi32(pos4, _MM_SHUFFLE(1, 1, 1, 1))); const int pos2 = _mm_cvtsi128_si32(_mm_shuffle_epi32(pos4, _MM_SHUFFLE(2, 2, 2, 2))); const int pos3 = _mm_cvtsi128_si32(_mm_shuffle_epi32(pos4, _MM_SHUFFLE(3, 3, 3, 3))); const __m128 val1 = _mm_setr_ps(src[pos0 ], src[pos1 ], src[pos2 ], src[pos3 ]); const __m128 val2 = _mm_setr_ps(src[pos0+1], src[pos1+1], src[pos2+1], src[pos3+1]); /* val1 + (val2-val1)*mu */ const __m128 r0 = _mm_sub_ps(val2, val1); const __m128 mu = _mm_mul_ps(_mm_cvtepi32_ps(frac4), fracOne4); const __m128 out = _mm_add_ps(val1, _mm_mul_ps(mu, r0)); _mm_store_ps(&dst[i], out); frac4 = _mm_add_epi32(frac4, increment4); pos4 = _mm_add_epi32(pos4, _mm_srli_epi32(frac4, FRACTIONBITS)); frac4 = _mm_and_si128(frac4, fracMask4); } /* NOTE: These four elements represent the position *after* the last four * samples, so the lowest element is the next position to resample. */ pos = _mm_cvtsi128_si32(pos4); frac = _mm_cvtsi128_si32(frac4); for(;i < numsamples;++i) { dst[i] = lerp(src[pos], src[pos+1], frac * (1.0f/FRACTIONONE)); frac += increment; pos += frac>>FRACTIONBITS; frac &= FRACTIONMASK; } return dst; } openal-soft-openal-soft-1.19.1/Alc/mixer/mixer_sse3.c000066400000000000000000000000001335774445300223340ustar00rootroot00000000000000openal-soft-openal-soft-1.19.1/Alc/mixer/mixer_sse41.c000066400000000000000000000060061335774445300224320ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 2014 by Timothy Arceri . * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include "alu.h" #include "defs.h" const ALfloat *Resample_lerp_SSE41(const InterpState* UNUSED(state), const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei numsamples) { const __m128i increment4 = _mm_set1_epi32(increment*4); const __m128 fracOne4 = _mm_set1_ps(1.0f/FRACTIONONE); const __m128i fracMask4 = _mm_set1_epi32(FRACTIONMASK); alignas(16) ALsizei pos_[4], frac_[4]; __m128i frac4, pos4; ALsizei todo, pos, i; ASSUME(numsamples > 0); InitiatePositionArrays(frac, increment, frac_, pos_, 4); frac4 = _mm_setr_epi32(frac_[0], frac_[1], frac_[2], frac_[3]); pos4 = _mm_setr_epi32(pos_[0], pos_[1], pos_[2], pos_[3]); todo = numsamples & ~3; for(i = 0;i < todo;i += 4) { const int pos0 = _mm_extract_epi32(pos4, 0); const int pos1 = _mm_extract_epi32(pos4, 1); const int pos2 = _mm_extract_epi32(pos4, 2); const int pos3 = _mm_extract_epi32(pos4, 3); const __m128 val1 = _mm_setr_ps(src[pos0 ], src[pos1 ], src[pos2 ], src[pos3 ]); const __m128 val2 = _mm_setr_ps(src[pos0+1], src[pos1+1], src[pos2+1], src[pos3+1]); /* val1 + (val2-val1)*mu */ const __m128 r0 = _mm_sub_ps(val2, val1); const __m128 mu = _mm_mul_ps(_mm_cvtepi32_ps(frac4), fracOne4); const __m128 out = _mm_add_ps(val1, _mm_mul_ps(mu, r0)); _mm_store_ps(&dst[i], out); frac4 = _mm_add_epi32(frac4, increment4); pos4 = _mm_add_epi32(pos4, _mm_srli_epi32(frac4, FRACTIONBITS)); frac4 = _mm_and_si128(frac4, fracMask4); } /* NOTE: These four elements represent the position *after* the last four * samples, so the lowest element is the next position to resample. */ pos = _mm_cvtsi128_si32(pos4); frac = _mm_cvtsi128_si32(frac4); for(;i < numsamples;++i) { dst[i] = lerp(src[pos], src[pos+1], frac * (1.0f/FRACTIONONE)); frac += increment; pos += frac>>FRACTIONBITS; frac &= FRACTIONMASK; } return dst; } openal-soft-openal-soft-1.19.1/Alc/mixvoice.c000066400000000000000000000701571335774445300207760ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include #include #include "alMain.h" #include "AL/al.h" #include "AL/alc.h" #include "alSource.h" #include "alBuffer.h" #include "alListener.h" #include "alAuxEffectSlot.h" #include "sample_cvt.h" #include "alu.h" #include "alconfig.h" #include "ringbuffer.h" #include "cpu_caps.h" #include "mixer/defs.h" static_assert((INT_MAX>>FRACTIONBITS)/MAX_PITCH > BUFFERSIZE, "MAX_PITCH and/or BUFFERSIZE are too large for FRACTIONBITS!"); extern inline void InitiatePositionArrays(ALsizei frac, ALint increment, ALsizei *restrict frac_arr, ALsizei *restrict pos_arr, ALsizei size); /* BSinc24 requires up to 23 extra samples before the current position, and 24 after. */ static_assert(MAX_RESAMPLE_PADDING >= 24, "MAX_RESAMPLE_PADDING must be at least 24!"); enum Resampler ResamplerDefault = LinearResampler; MixerFunc MixSamples = Mix_C; RowMixerFunc MixRowSamples = MixRow_C; static HrtfMixerFunc MixHrtfSamples = MixHrtf_C; static HrtfMixerBlendFunc MixHrtfBlendSamples = MixHrtfBlend_C; static MixerFunc SelectMixer(void) { #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return Mix_Neon; #endif #ifdef HAVE_SSE if((CPUCapFlags&CPU_CAP_SSE)) return Mix_SSE; #endif return Mix_C; } static RowMixerFunc SelectRowMixer(void) { #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return MixRow_Neon; #endif #ifdef HAVE_SSE if((CPUCapFlags&CPU_CAP_SSE)) return MixRow_SSE; #endif return MixRow_C; } static inline HrtfMixerFunc SelectHrtfMixer(void) { #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return MixHrtf_Neon; #endif #ifdef HAVE_SSE if((CPUCapFlags&CPU_CAP_SSE)) return MixHrtf_SSE; #endif return MixHrtf_C; } static inline HrtfMixerBlendFunc SelectHrtfBlendMixer(void) { #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return MixHrtfBlend_Neon; #endif #ifdef HAVE_SSE if((CPUCapFlags&CPU_CAP_SSE)) return MixHrtfBlend_SSE; #endif return MixHrtfBlend_C; } ResamplerFunc SelectResampler(enum Resampler resampler) { switch(resampler) { case PointResampler: return Resample_point_C; case LinearResampler: #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return Resample_lerp_Neon; #endif #ifdef HAVE_SSE4_1 if((CPUCapFlags&CPU_CAP_SSE4_1)) return Resample_lerp_SSE41; #endif #ifdef HAVE_SSE2 if((CPUCapFlags&CPU_CAP_SSE2)) return Resample_lerp_SSE2; #endif return Resample_lerp_C; case FIR4Resampler: return Resample_cubic_C; case BSinc12Resampler: case BSinc24Resampler: #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return Resample_bsinc_Neon; #endif #ifdef HAVE_SSE if((CPUCapFlags&CPU_CAP_SSE)) return Resample_bsinc_SSE; #endif return Resample_bsinc_C; } return Resample_point_C; } void aluInitMixer(void) { const char *str; if(ConfigValueStr(NULL, NULL, "resampler", &str)) { if(strcasecmp(str, "point") == 0 || strcasecmp(str, "none") == 0) ResamplerDefault = PointResampler; else if(strcasecmp(str, "linear") == 0) ResamplerDefault = LinearResampler; else if(strcasecmp(str, "cubic") == 0) ResamplerDefault = FIR4Resampler; else if(strcasecmp(str, "bsinc12") == 0) ResamplerDefault = BSinc12Resampler; else if(strcasecmp(str, "bsinc24") == 0) ResamplerDefault = BSinc24Resampler; else if(strcasecmp(str, "bsinc") == 0) { WARN("Resampler option \"%s\" is deprecated, using bsinc12\n", str); ResamplerDefault = BSinc12Resampler; } else if(strcasecmp(str, "sinc4") == 0 || strcasecmp(str, "sinc8") == 0) { WARN("Resampler option \"%s\" is deprecated, using cubic\n", str); ResamplerDefault = FIR4Resampler; } else { char *end; long n = strtol(str, &end, 0); if(*end == '\0' && (n == PointResampler || n == LinearResampler || n == FIR4Resampler)) ResamplerDefault = n; else WARN("Invalid resampler: %s\n", str); } } MixHrtfBlendSamples = SelectHrtfBlendMixer(); MixHrtfSamples = SelectHrtfMixer(); MixSamples = SelectMixer(); MixRowSamples = SelectRowMixer(); } static void SendAsyncEvent(ALCcontext *context, ALuint enumtype, ALenum type, ALuint objid, ALuint param, const char *msg) { AsyncEvent evt = ASYNC_EVENT(enumtype); evt.u.user.type = type; evt.u.user.id = objid; evt.u.user.param = param; strcpy(evt.u.user.msg, msg); if(ll_ringbuffer_write(context->AsyncEvents, (const char*)&evt, 1) == 1) alsem_post(&context->EventSem); } static inline ALfloat Sample_ALubyte(ALubyte val) { return (val-128) * (1.0f/128.0f); } static inline ALfloat Sample_ALshort(ALshort val) { return val * (1.0f/32768.0f); } static inline ALfloat Sample_ALfloat(ALfloat val) { return val; } static inline ALfloat Sample_ALdouble(ALdouble val) { return (ALfloat)val; } typedef ALubyte ALmulaw; static inline ALfloat Sample_ALmulaw(ALmulaw val) { return muLawDecompressionTable[val] * (1.0f/32768.0f); } typedef ALubyte ALalaw; static inline ALfloat Sample_ALalaw(ALalaw val) { return aLawDecompressionTable[val] * (1.0f/32768.0f); } #define DECL_TEMPLATE(T) \ static inline void Load_##T(ALfloat *restrict dst, const T *restrict src, \ ALint srcstep, ALsizei samples) \ { \ ALsizei i; \ for(i = 0;i < samples;i++) \ dst[i] += Sample_##T(src[i*srcstep]); \ } DECL_TEMPLATE(ALubyte) DECL_TEMPLATE(ALshort) DECL_TEMPLATE(ALfloat) DECL_TEMPLATE(ALdouble) DECL_TEMPLATE(ALmulaw) DECL_TEMPLATE(ALalaw) #undef DECL_TEMPLATE static void LoadSamples(ALfloat *restrict dst, const ALvoid *restrict src, ALint srcstep, enum FmtType srctype, ALsizei samples) { #define HANDLE_FMT(ET, ST) case ET: Load_##ST(dst, src, srcstep, samples); break switch(srctype) { HANDLE_FMT(FmtUByte, ALubyte); HANDLE_FMT(FmtShort, ALshort); HANDLE_FMT(FmtFloat, ALfloat); HANDLE_FMT(FmtDouble, ALdouble); HANDLE_FMT(FmtMulaw, ALmulaw); HANDLE_FMT(FmtAlaw, ALalaw); } #undef HANDLE_FMT } static const ALfloat *DoFilters(BiquadFilter *lpfilter, BiquadFilter *hpfilter, ALfloat *restrict dst, const ALfloat *restrict src, ALsizei numsamples, enum ActiveFilters type) { ALsizei i; switch(type) { case AF_None: BiquadFilter_passthru(lpfilter, numsamples); BiquadFilter_passthru(hpfilter, numsamples); break; case AF_LowPass: BiquadFilter_process(lpfilter, dst, src, numsamples); BiquadFilter_passthru(hpfilter, numsamples); return dst; case AF_HighPass: BiquadFilter_passthru(lpfilter, numsamples); BiquadFilter_process(hpfilter, dst, src, numsamples); return dst; case AF_BandPass: for(i = 0;i < numsamples;) { ALfloat temp[256]; ALsizei todo = mini(256, numsamples-i); BiquadFilter_process(lpfilter, temp, src+i, todo); BiquadFilter_process(hpfilter, dst+i, temp, todo); i += todo; } return dst; } return src; } /* This function uses these device temp buffers. */ #define SOURCE_DATA_BUF 0 #define RESAMPLED_BUF 1 #define FILTERED_BUF 2 #define NFC_DATA_BUF 3 ALboolean MixSource(ALvoice *voice, ALuint SourceID, ALCcontext *Context, ALsizei SamplesToDo) { ALCdevice *Device = Context->Device; ALbufferlistitem *BufferListItem; ALbufferlistitem *BufferLoopItem; ALsizei NumChannels, SampleSize; ALbitfieldSOFT enabledevt; ALsizei buffers_done = 0; ResamplerFunc Resample; ALsizei DataPosInt; ALsizei DataPosFrac; ALint64 DataSize64; ALint increment; ALsizei Counter; ALsizei OutPos; ALsizei IrSize; bool isplaying; bool firstpass; bool isstatic; ALsizei chan; ALsizei send; /* Get source info */ isplaying = true; /* Will only be called while playing. */ isstatic = !!(voice->Flags&VOICE_IS_STATIC); DataPosInt = ATOMIC_LOAD(&voice->position, almemory_order_acquire); DataPosFrac = ATOMIC_LOAD(&voice->position_fraction, almemory_order_relaxed); BufferListItem = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed); BufferLoopItem = ATOMIC_LOAD(&voice->loop_buffer, almemory_order_relaxed); NumChannels = voice->NumChannels; SampleSize = voice->SampleSize; increment = voice->Step; IrSize = (Device->HrtfHandle ? Device->HrtfHandle->irSize : 0); Resample = ((increment == FRACTIONONE && DataPosFrac == 0) ? Resample_copy_C : voice->Resampler); Counter = (voice->Flags&VOICE_IS_FADING) ? SamplesToDo : 0; firstpass = true; OutPos = 0; do { ALsizei SrcBufferSize, DstBufferSize; /* Figure out how many buffer samples will be needed */ DataSize64 = SamplesToDo-OutPos; DataSize64 *= increment; DataSize64 += DataPosFrac+FRACTIONMASK; DataSize64 >>= FRACTIONBITS; DataSize64 += MAX_RESAMPLE_PADDING*2; SrcBufferSize = (ALsizei)mini64(DataSize64, BUFFERSIZE); /* Figure out how many samples we can actually mix from this. */ DataSize64 = SrcBufferSize; DataSize64 -= MAX_RESAMPLE_PADDING*2; DataSize64 <<= FRACTIONBITS; DataSize64 -= DataPosFrac; DstBufferSize = (ALsizei)mini64((DataSize64+(increment-1)) / increment, SamplesToDo - OutPos); /* Some mixers like having a multiple of 4, so try to give that unless * this is the last update. */ if(DstBufferSize < SamplesToDo-OutPos) DstBufferSize &= ~3; /* It's impossible to have a buffer list item with no entries. */ assert(BufferListItem->num_buffers > 0); for(chan = 0;chan < NumChannels;chan++) { const ALfloat *ResampledData; ALfloat *SrcData = Device->TempBuffer[SOURCE_DATA_BUF]; ALsizei FilledAmt; /* Load the previous samples into the source data first, and clear the rest. */ memcpy(SrcData, voice->PrevSamples[chan], MAX_RESAMPLE_PADDING*sizeof(ALfloat)); memset(SrcData+MAX_RESAMPLE_PADDING, 0, (BUFFERSIZE-MAX_RESAMPLE_PADDING)* sizeof(ALfloat)); FilledAmt = MAX_RESAMPLE_PADDING; if(isstatic) { /* TODO: For static sources, loop points are taken from the * first buffer (should be adjusted by any buffer offset, to * possibly be added later). */ const ALbuffer *Buffer0 = BufferListItem->buffers[0]; const ALsizei LoopStart = Buffer0->LoopStart; const ALsizei LoopEnd = Buffer0->LoopEnd; const ALsizei LoopSize = LoopEnd - LoopStart; /* If current pos is beyond the loop range, do not loop */ if(!BufferLoopItem || DataPosInt >= LoopEnd) { ALsizei SizeToDo = SrcBufferSize - FilledAmt; ALsizei CompLen = 0; ALsizei i; BufferLoopItem = NULL; for(i = 0;i < BufferListItem->num_buffers;i++) { const ALbuffer *buffer = BufferListItem->buffers[i]; const ALubyte *Data = buffer->data; ALsizei DataSize; if(DataPosInt >= buffer->SampleLen) continue; /* Load what's left to play from the buffer */ DataSize = mini(SizeToDo, buffer->SampleLen - DataPosInt); CompLen = maxi(CompLen, DataSize); LoadSamples(&SrcData[FilledAmt], &Data[(DataPosInt*NumChannels + chan)*SampleSize], NumChannels, buffer->FmtType, DataSize ); } FilledAmt += CompLen; } else { ALsizei SizeToDo = mini(SrcBufferSize - FilledAmt, LoopEnd - DataPosInt); ALsizei CompLen = 0; ALsizei i; for(i = 0;i < BufferListItem->num_buffers;i++) { const ALbuffer *buffer = BufferListItem->buffers[i]; const ALubyte *Data = buffer->data; ALsizei DataSize; if(DataPosInt >= buffer->SampleLen) continue; /* Load what's left of this loop iteration */ DataSize = mini(SizeToDo, buffer->SampleLen - DataPosInt); CompLen = maxi(CompLen, DataSize); LoadSamples(&SrcData[FilledAmt], &Data[(DataPosInt*NumChannels + chan)*SampleSize], NumChannels, buffer->FmtType, DataSize ); } FilledAmt += CompLen; while(SrcBufferSize > FilledAmt) { const ALsizei SizeToDo = mini(SrcBufferSize - FilledAmt, LoopSize); CompLen = 0; for(i = 0;i < BufferListItem->num_buffers;i++) { const ALbuffer *buffer = BufferListItem->buffers[i]; const ALubyte *Data = buffer->data; ALsizei DataSize; if(LoopStart >= buffer->SampleLen) continue; DataSize = mini(SizeToDo, buffer->SampleLen - LoopStart); CompLen = maxi(CompLen, DataSize); LoadSamples(&SrcData[FilledAmt], &Data[(LoopStart*NumChannels + chan)*SampleSize], NumChannels, buffer->FmtType, DataSize ); } FilledAmt += CompLen; } } } else { /* Crawl the buffer queue to fill in the temp buffer */ ALbufferlistitem *tmpiter = BufferListItem; ALsizei pos = DataPosInt; while(tmpiter && SrcBufferSize > FilledAmt) { ALsizei SizeToDo = SrcBufferSize - FilledAmt; ALsizei CompLen = 0; ALsizei i; for(i = 0;i < tmpiter->num_buffers;i++) { const ALbuffer *ALBuffer = tmpiter->buffers[i]; ALsizei DataSize = ALBuffer ? ALBuffer->SampleLen : 0; if(DataSize > pos) { const ALubyte *Data = ALBuffer->data; Data += (pos*NumChannels + chan)*SampleSize; DataSize = mini(SizeToDo, DataSize - pos); CompLen = maxi(CompLen, DataSize); LoadSamples(&SrcData[FilledAmt], Data, NumChannels, ALBuffer->FmtType, DataSize); } } if(UNLIKELY(!CompLen)) pos -= tmpiter->max_samples; else { FilledAmt += CompLen; if(SrcBufferSize <= FilledAmt) break; pos = 0; } tmpiter = ATOMIC_LOAD(&tmpiter->next, almemory_order_acquire); if(!tmpiter) tmpiter = BufferLoopItem; } } /* Store the last source samples used for next time. */ memcpy(voice->PrevSamples[chan], &SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS], MAX_RESAMPLE_PADDING*sizeof(ALfloat) ); /* Now resample, then filter and mix to the appropriate outputs. */ ResampledData = Resample(&voice->ResampleState, &SrcData[MAX_RESAMPLE_PADDING], DataPosFrac, increment, Device->TempBuffer[RESAMPLED_BUF], DstBufferSize ); { DirectParams *parms = &voice->Direct.Params[chan]; const ALfloat *samples; samples = DoFilters( &parms->LowPass, &parms->HighPass, Device->TempBuffer[FILTERED_BUF], ResampledData, DstBufferSize, voice->Direct.FilterType ); if(!(voice->Flags&VOICE_HAS_HRTF)) { if(!Counter) memcpy(parms->Gains.Current, parms->Gains.Target, sizeof(parms->Gains.Current)); if(!(voice->Flags&VOICE_HAS_NFC)) MixSamples(samples, voice->Direct.Channels, voice->Direct.Buffer, parms->Gains.Current, parms->Gains.Target, Counter, OutPos, DstBufferSize ); else { ALfloat *nfcsamples = Device->TempBuffer[NFC_DATA_BUF]; ALsizei chanoffset = 0; MixSamples(samples, voice->Direct.ChannelsPerOrder[0], voice->Direct.Buffer, parms->Gains.Current, parms->Gains.Target, Counter, OutPos, DstBufferSize ); chanoffset += voice->Direct.ChannelsPerOrder[0]; #define APPLY_NFC_MIX(order) \ if(voice->Direct.ChannelsPerOrder[order] > 0) \ { \ NfcFilterProcess##order(&parms->NFCtrlFilter, nfcsamples, samples, \ DstBufferSize); \ MixSamples(nfcsamples, voice->Direct.ChannelsPerOrder[order], \ voice->Direct.Buffer+chanoffset, parms->Gains.Current+chanoffset, \ parms->Gains.Target+chanoffset, Counter, OutPos, DstBufferSize \ ); \ chanoffset += voice->Direct.ChannelsPerOrder[order]; \ } APPLY_NFC_MIX(1) APPLY_NFC_MIX(2) APPLY_NFC_MIX(3) #undef APPLY_NFC_MIX } } else { MixHrtfParams hrtfparams; ALsizei fademix = 0; int lidx, ridx; lidx = GetChannelIdxByName(&Device->RealOut, FrontLeft); ridx = GetChannelIdxByName(&Device->RealOut, FrontRight); assert(lidx != -1 && ridx != -1); if(!Counter) { /* No fading, just overwrite the old HRTF params. */ parms->Hrtf.Old = parms->Hrtf.Target; } else if(!(parms->Hrtf.Old.Gain > GAIN_SILENCE_THRESHOLD)) { /* The old HRTF params are silent, so overwrite the old * coefficients with the new, and reset the old gain to * 0. The future mix will then fade from silence. */ parms->Hrtf.Old = parms->Hrtf.Target; parms->Hrtf.Old.Gain = 0.0f; } else if(firstpass) { ALfloat gain; /* Fade between the coefficients over 128 samples. */ fademix = mini(DstBufferSize, 128); /* The new coefficients need to fade in completely * since they're replacing the old ones. To keep the * gain fading consistent, interpolate between the old * and new target gains given how much of the fade time * this mix handles. */ gain = lerp(parms->Hrtf.Old.Gain, parms->Hrtf.Target.Gain, minf(1.0f, (ALfloat)fademix/Counter)); hrtfparams.Coeffs = parms->Hrtf.Target.Coeffs; hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0]; hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1]; hrtfparams.Gain = 0.0f; hrtfparams.GainStep = gain / (ALfloat)fademix; MixHrtfBlendSamples( voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx], samples, voice->Offset, OutPos, IrSize, &parms->Hrtf.Old, &hrtfparams, &parms->Hrtf.State, fademix ); /* Update the old parameters with the result. */ parms->Hrtf.Old = parms->Hrtf.Target; if(fademix < Counter) parms->Hrtf.Old.Gain = hrtfparams.Gain; } if(fademix < DstBufferSize) { ALsizei todo = DstBufferSize - fademix; ALfloat gain = parms->Hrtf.Target.Gain; /* Interpolate the target gain if the gain fading lasts * longer than this mix. */ if(Counter > DstBufferSize) gain = lerp(parms->Hrtf.Old.Gain, gain, (ALfloat)todo/(Counter-fademix)); hrtfparams.Coeffs = parms->Hrtf.Target.Coeffs; hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0]; hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1]; hrtfparams.Gain = parms->Hrtf.Old.Gain; hrtfparams.GainStep = (gain - parms->Hrtf.Old.Gain) / (ALfloat)todo; MixHrtfSamples( voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx], samples+fademix, voice->Offset+fademix, OutPos+fademix, IrSize, &hrtfparams, &parms->Hrtf.State, todo ); /* Store the interpolated gain or the final target gain * depending if the fade is done. */ if(DstBufferSize < Counter) parms->Hrtf.Old.Gain = gain; else parms->Hrtf.Old.Gain = parms->Hrtf.Target.Gain; } } } for(send = 0;send < Device->NumAuxSends;send++) { SendParams *parms = &voice->Send[send].Params[chan]; const ALfloat *samples; if(!voice->Send[send].Buffer) continue; samples = DoFilters( &parms->LowPass, &parms->HighPass, Device->TempBuffer[FILTERED_BUF], ResampledData, DstBufferSize, voice->Send[send].FilterType ); if(!Counter) memcpy(parms->Gains.Current, parms->Gains.Target, sizeof(parms->Gains.Current)); MixSamples(samples, voice->Send[send].Channels, voice->Send[send].Buffer, parms->Gains.Current, parms->Gains.Target, Counter, OutPos, DstBufferSize ); } } /* Update positions */ DataPosFrac += increment*DstBufferSize; DataPosInt += DataPosFrac>>FRACTIONBITS; DataPosFrac &= FRACTIONMASK; OutPos += DstBufferSize; voice->Offset += DstBufferSize; Counter = maxi(DstBufferSize, Counter) - DstBufferSize; firstpass = false; if(isstatic) { if(BufferLoopItem) { /* Handle looping static source */ const ALbuffer *Buffer = BufferListItem->buffers[0]; ALsizei LoopStart = Buffer->LoopStart; ALsizei LoopEnd = Buffer->LoopEnd; if(DataPosInt >= LoopEnd) { assert(LoopEnd > LoopStart); DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart; } } else { /* Handle non-looping static source */ if(DataPosInt >= BufferListItem->max_samples) { isplaying = false; BufferListItem = NULL; DataPosInt = 0; DataPosFrac = 0; break; } } } else while(1) { /* Handle streaming source */ if(BufferListItem->max_samples > DataPosInt) break; DataPosInt -= BufferListItem->max_samples; buffers_done += BufferListItem->num_buffers; BufferListItem = ATOMIC_LOAD(&BufferListItem->next, almemory_order_relaxed); if(!BufferListItem && !(BufferListItem=BufferLoopItem)) { isplaying = false; DataPosInt = 0; DataPosFrac = 0; break; } } } while(isplaying && OutPos < SamplesToDo); voice->Flags |= VOICE_IS_FADING; /* Update source info */ ATOMIC_STORE(&voice->position, DataPosInt, almemory_order_relaxed); ATOMIC_STORE(&voice->position_fraction, DataPosFrac, almemory_order_relaxed); ATOMIC_STORE(&voice->current_buffer, BufferListItem, almemory_order_release); /* Send any events now, after the position/buffer info was updated. */ enabledevt = ATOMIC_LOAD(&Context->EnabledEvts, almemory_order_acquire); if(buffers_done > 0 && (enabledevt&EventType_BufferCompleted)) SendAsyncEvent(Context, EventType_BufferCompleted, AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT, SourceID, buffers_done, "Buffer completed" ); return isplaying; } openal-soft-openal-soft-1.19.1/Alc/panning.c000066400000000000000000001331561335774445300206040ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2010 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include #include #include "alMain.h" #include "alAuxEffectSlot.h" #include "alu.h" #include "alconfig.h" #include "bool.h" #include "ambdec.h" #include "bformatdec.h" #include "filters/splitter.h" #include "uhjfilter.h" #include "bs2b.h" extern inline void CalcDirectionCoeffs(const ALfloat dir[3], ALfloat spread, ALfloat coeffs[MAX_AMBI_COEFFS]); extern inline void CalcAngleCoeffs(ALfloat azimuth, ALfloat elevation, ALfloat spread, ALfloat coeffs[MAX_AMBI_COEFFS]); extern inline float ScaleAzimuthFront(float azimuth, float scale); extern inline void ComputePanGains(const MixParams *dry, const ALfloat*restrict coeffs, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]); static const ALsizei FuMa2ACN[MAX_AMBI_COEFFS] = { 0, /* W */ 3, /* X */ 1, /* Y */ 2, /* Z */ 6, /* R */ 7, /* S */ 5, /* T */ 8, /* U */ 4, /* V */ 12, /* K */ 13, /* L */ 11, /* M */ 14, /* N */ 10, /* O */ 15, /* P */ 9, /* Q */ }; static const ALsizei ACN2ACN[MAX_AMBI_COEFFS] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15 }; void CalcAmbiCoeffs(const ALfloat y, const ALfloat z, const ALfloat x, const ALfloat spread, ALfloat coeffs[MAX_AMBI_COEFFS]) { /* Zeroth-order */ coeffs[0] = 1.0f; /* ACN 0 = 1 */ /* First-order */ coeffs[1] = SQRTF_3 * y; /* ACN 1 = sqrt(3) * Y */ coeffs[2] = SQRTF_3 * z; /* ACN 2 = sqrt(3) * Z */ coeffs[3] = SQRTF_3 * x; /* ACN 3 = sqrt(3) * X */ /* Second-order */ coeffs[4] = 3.872983346f * x * y; /* ACN 4 = sqrt(15) * X * Y */ coeffs[5] = 3.872983346f * y * z; /* ACN 5 = sqrt(15) * Y * Z */ coeffs[6] = 1.118033989f * (3.0f*z*z - 1.0f); /* ACN 6 = sqrt(5)/2 * (3*Z*Z - 1) */ coeffs[7] = 3.872983346f * x * z; /* ACN 7 = sqrt(15) * X * Z */ coeffs[8] = 1.936491673f * (x*x - y*y); /* ACN 8 = sqrt(15)/2 * (X*X - Y*Y) */ /* Third-order */ coeffs[9] = 2.091650066f * y * (3.0f*x*x - y*y); /* ACN 9 = sqrt(35/8) * Y * (3*X*X - Y*Y) */ coeffs[10] = 10.246950766f * z * x * y; /* ACN 10 = sqrt(105) * Z * X * Y */ coeffs[11] = 1.620185175f * y * (5.0f*z*z - 1.0f); /* ACN 11 = sqrt(21/8) * Y * (5*Z*Z - 1) */ coeffs[12] = 1.322875656f * z * (5.0f*z*z - 3.0f); /* ACN 12 = sqrt(7)/2 * Z * (5*Z*Z - 3) */ coeffs[13] = 1.620185175f * x * (5.0f*z*z - 1.0f); /* ACN 13 = sqrt(21/8) * X * (5*Z*Z - 1) */ coeffs[14] = 5.123475383f * z * (x*x - y*y); /* ACN 14 = sqrt(105)/2 * Z * (X*X - Y*Y) */ coeffs[15] = 2.091650066f * x * (x*x - 3.0f*y*y); /* ACN 15 = sqrt(35/8) * X * (X*X - 3*Y*Y) */ if(spread > 0.0f) { /* Implement the spread by using a spherical source that subtends the * angle spread. See: * http://www.ppsloan.org/publications/StupidSH36.pdf - Appendix A3 * * When adjusted for N3D normalization instead of SN3D, these * calculations are: * * ZH0 = -sqrt(pi) * (-1+ca); * ZH1 = 0.5*sqrt(pi) * sa*sa; * ZH2 = -0.5*sqrt(pi) * ca*(-1+ca)*(ca+1); * ZH3 = -0.125*sqrt(pi) * (-1+ca)*(ca+1)*(5*ca*ca - 1); * ZH4 = -0.125*sqrt(pi) * ca*(-1+ca)*(ca+1)*(7*ca*ca - 3); * ZH5 = -0.0625*sqrt(pi) * (-1+ca)*(ca+1)*(21*ca*ca*ca*ca - 14*ca*ca + 1); * * The gain of the source is compensated for size, so that the * loundness doesn't depend on the spread. Thus: * * ZH0 = 1.0f; * ZH1 = 0.5f * (ca+1.0f); * ZH2 = 0.5f * (ca+1.0f)*ca; * ZH3 = 0.125f * (ca+1.0f)*(5.0f*ca*ca - 1.0f); * ZH4 = 0.125f * (ca+1.0f)*(7.0f*ca*ca - 3.0f)*ca; * ZH5 = 0.0625f * (ca+1.0f)*(21.0f*ca*ca*ca*ca - 14.0f*ca*ca + 1.0f); */ ALfloat ca = cosf(spread * 0.5f); /* Increase the source volume by up to +3dB for a full spread. */ ALfloat scale = sqrtf(1.0f + spread/F_TAU); ALfloat ZH0_norm = scale; ALfloat ZH1_norm = 0.5f * (ca+1.f) * scale; ALfloat ZH2_norm = 0.5f * (ca+1.f)*ca * scale; ALfloat ZH3_norm = 0.125f * (ca+1.f)*(5.f*ca*ca-1.f) * scale; /* Zeroth-order */ coeffs[0] *= ZH0_norm; /* First-order */ coeffs[1] *= ZH1_norm; coeffs[2] *= ZH1_norm; coeffs[3] *= ZH1_norm; /* Second-order */ coeffs[4] *= ZH2_norm; coeffs[5] *= ZH2_norm; coeffs[6] *= ZH2_norm; coeffs[7] *= ZH2_norm; coeffs[8] *= ZH2_norm; /* Third-order */ coeffs[9] *= ZH3_norm; coeffs[10] *= ZH3_norm; coeffs[11] *= ZH3_norm; coeffs[12] *= ZH3_norm; coeffs[13] *= ZH3_norm; coeffs[14] *= ZH3_norm; coeffs[15] *= ZH3_norm; } } void ComputePanningGainsMC(const ChannelConfig *chancoeffs, ALsizei numchans, ALsizei numcoeffs, const ALfloat*restrict coeffs, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]) { ALsizei i, j; for(i = 0;i < numchans;i++) { float gain = 0.0f; for(j = 0;j < numcoeffs;j++) gain += chancoeffs[i][j]*coeffs[j]; gains[i] = clampf(gain, 0.0f, 1.0f) * ingain; } for(;i < MAX_OUTPUT_CHANNELS;i++) gains[i] = 0.0f; } void ComputePanningGainsBF(const BFChannelConfig *chanmap, ALsizei numchans, const ALfloat*restrict coeffs, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]) { ALsizei i; for(i = 0;i < numchans;i++) gains[i] = chanmap[i].Scale * coeffs[chanmap[i].Index] * ingain; for(;i < MAX_OUTPUT_CHANNELS;i++) gains[i] = 0.0f; } static inline const char *GetLabelFromChannel(enum Channel channel) { switch(channel) { case FrontLeft: return "front-left"; case FrontRight: return "front-right"; case FrontCenter: return "front-center"; case LFE: return "lfe"; case BackLeft: return "back-left"; case BackRight: return "back-right"; case BackCenter: return "back-center"; case SideLeft: return "side-left"; case SideRight: return "side-right"; case UpperFrontLeft: return "upper-front-left"; case UpperFrontRight: return "upper-front-right"; case UpperBackLeft: return "upper-back-left"; case UpperBackRight: return "upper-back-right"; case LowerFrontLeft: return "lower-front-left"; case LowerFrontRight: return "lower-front-right"; case LowerBackLeft: return "lower-back-left"; case LowerBackRight: return "lower-back-right"; case Aux0: return "aux-0"; case Aux1: return "aux-1"; case Aux2: return "aux-2"; case Aux3: return "aux-3"; case Aux4: return "aux-4"; case Aux5: return "aux-5"; case Aux6: return "aux-6"; case Aux7: return "aux-7"; case Aux8: return "aux-8"; case Aux9: return "aux-9"; case Aux10: return "aux-10"; case Aux11: return "aux-11"; case Aux12: return "aux-12"; case Aux13: return "aux-13"; case Aux14: return "aux-14"; case Aux15: return "aux-15"; case InvalidChannel: break; } return "(unknown)"; } typedef struct ChannelMap { enum Channel ChanName; ChannelConfig Config; } ChannelMap; static void SetChannelMap(const enum Channel devchans[MAX_OUTPUT_CHANNELS], ChannelConfig *ambicoeffs, const ChannelMap *chanmap, ALsizei count, ALsizei *outcount) { ALsizei maxchans = 0; ALsizei i, j; for(i = 0;i < count;i++) { ALint idx = GetChannelIndex(devchans, chanmap[i].ChanName); if(idx < 0) { ERR("Failed to find %s channel in device\n", GetLabelFromChannel(chanmap[i].ChanName)); continue; } maxchans = maxi(maxchans, idx+1); for(j = 0;j < MAX_AMBI_COEFFS;j++) ambicoeffs[idx][j] = chanmap[i].Config[j]; } *outcount = mini(maxchans, MAX_OUTPUT_CHANNELS); } static bool MakeSpeakerMap(ALCdevice *device, const AmbDecConf *conf, ALsizei speakermap[MAX_OUTPUT_CHANNELS]) { ALsizei i; for(i = 0;i < conf->NumSpeakers;i++) { enum Channel ch; int chidx = -1; /* NOTE: AmbDec does not define any standard speaker names, however * for this to work we have to by able to find the output channel * the speaker definition corresponds to. Therefore, OpenAL Soft * requires these channel labels to be recognized: * * LF = Front left * RF = Front right * LS = Side left * RS = Side right * LB = Back left * RB = Back right * CE = Front center * CB = Back center * * Additionally, surround51 will acknowledge back speakers for side * channels, and surround51rear will acknowledge side speakers for * back channels, to avoid issues with an ambdec expecting 5.1 to * use the side channels when the device is configured for back, * and vice-versa. */ if(alstr_cmp_cstr(conf->Speakers[i].Name, "LF") == 0) ch = FrontLeft; else if(alstr_cmp_cstr(conf->Speakers[i].Name, "RF") == 0) ch = FrontRight; else if(alstr_cmp_cstr(conf->Speakers[i].Name, "CE") == 0) ch = FrontCenter; else if(alstr_cmp_cstr(conf->Speakers[i].Name, "LS") == 0) { if(device->FmtChans == DevFmtX51Rear) ch = BackLeft; else ch = SideLeft; } else if(alstr_cmp_cstr(conf->Speakers[i].Name, "RS") == 0) { if(device->FmtChans == DevFmtX51Rear) ch = BackRight; else ch = SideRight; } else if(alstr_cmp_cstr(conf->Speakers[i].Name, "LB") == 0) { if(device->FmtChans == DevFmtX51) ch = SideLeft; else ch = BackLeft; } else if(alstr_cmp_cstr(conf->Speakers[i].Name, "RB") == 0) { if(device->FmtChans == DevFmtX51) ch = SideRight; else ch = BackRight; } else if(alstr_cmp_cstr(conf->Speakers[i].Name, "CB") == 0) ch = BackCenter; else { const char *name = alstr_get_cstr(conf->Speakers[i].Name); unsigned int n; char c; if(sscanf(name, "AUX%u%c", &n, &c) == 1 && n < 16) ch = Aux0+n; else { ERR("AmbDec speaker label \"%s\" not recognized\n", name); return false; } } chidx = GetChannelIdxByName(&device->RealOut, ch); if(chidx == -1) { ERR("Failed to lookup AmbDec speaker label %s\n", alstr_get_cstr(conf->Speakers[i].Name)); return false; } speakermap[i] = chidx; } return true; } static const ChannelMap MonoCfg[1] = { { FrontCenter, { 1.0f } }, }, StereoCfg[2] = { { FrontLeft, { 5.00000000e-1f, 2.88675135e-1f, 0.0f, 5.52305643e-2f } }, { FrontRight, { 5.00000000e-1f, -2.88675135e-1f, 0.0f, 5.52305643e-2f } }, }, QuadCfg[4] = { { BackLeft, { 3.53553391e-1f, 2.04124145e-1f, 0.0f, -2.04124145e-1f } }, { FrontLeft, { 3.53553391e-1f, 2.04124145e-1f, 0.0f, 2.04124145e-1f } }, { FrontRight, { 3.53553391e-1f, -2.04124145e-1f, 0.0f, 2.04124145e-1f } }, { BackRight, { 3.53553391e-1f, -2.04124145e-1f, 0.0f, -2.04124145e-1f } }, }, X51SideCfg[4] = { { SideLeft, { 3.33000782e-1f, 1.89084803e-1f, 0.0f, -2.00042375e-1f, -2.12307769e-2f, 0.0f, 0.0f, 0.0f, -1.14579885e-2f } }, { FrontLeft, { 1.88542860e-1f, 1.27709292e-1f, 0.0f, 1.66295695e-1f, 7.30571517e-2f, 0.0f, 0.0f, 0.0f, 2.10901184e-2f } }, { FrontRight, { 1.88542860e-1f, -1.27709292e-1f, 0.0f, 1.66295695e-1f, -7.30571517e-2f, 0.0f, 0.0f, 0.0f, 2.10901184e-2f } }, { SideRight, { 3.33000782e-1f, -1.89084803e-1f, 0.0f, -2.00042375e-1f, 2.12307769e-2f, 0.0f, 0.0f, 0.0f, -1.14579885e-2f } }, }, X51RearCfg[4] = { { BackLeft, { 3.33000782e-1f, 1.89084803e-1f, 0.0f, -2.00042375e-1f, -2.12307769e-2f, 0.0f, 0.0f, 0.0f, -1.14579885e-2f } }, { FrontLeft, { 1.88542860e-1f, 1.27709292e-1f, 0.0f, 1.66295695e-1f, 7.30571517e-2f, 0.0f, 0.0f, 0.0f, 2.10901184e-2f } }, { FrontRight, { 1.88542860e-1f, -1.27709292e-1f, 0.0f, 1.66295695e-1f, -7.30571517e-2f, 0.0f, 0.0f, 0.0f, 2.10901184e-2f } }, { BackRight, { 3.33000782e-1f, -1.89084803e-1f, 0.0f, -2.00042375e-1f, 2.12307769e-2f, 0.0f, 0.0f, 0.0f, -1.14579885e-2f } }, }, X61Cfg[6] = { { SideLeft, { 2.04460341e-1f, 2.17177926e-1f, 0.0f, -4.39996780e-2f, -2.60790269e-2f, 0.0f, 0.0f, 0.0f, -6.87239792e-2f } }, { FrontLeft, { 1.58923161e-1f, 9.21772680e-2f, 0.0f, 1.59658796e-1f, 6.66278083e-2f, 0.0f, 0.0f, 0.0f, 3.84686854e-2f } }, { FrontRight, { 1.58923161e-1f, -9.21772680e-2f, 0.0f, 1.59658796e-1f, -6.66278083e-2f, 0.0f, 0.0f, 0.0f, 3.84686854e-2f } }, { SideRight, { 2.04460341e-1f, -2.17177926e-1f, 0.0f, -4.39996780e-2f, 2.60790269e-2f, 0.0f, 0.0f, 0.0f, -6.87239792e-2f } }, { BackCenter, { 2.50001688e-1f, 0.00000000e+0f, 0.0f, -2.50000094e-1f, 0.00000000e+0f, 0.0f, 0.0f, 0.0f, 6.05133395e-2f } }, }, X71Cfg[6] = { { BackLeft, { 2.04124145e-1f, 1.08880247e-1f, 0.0f, -1.88586120e-1f, -1.29099444e-1f, 0.0f, 0.0f, 0.0f, 7.45355993e-2f, 3.73460789e-2f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.00000000e+0f } }, { SideLeft, { 2.04124145e-1f, 2.17760495e-1f, 0.0f, 0.00000000e+0f, 0.00000000e+0f, 0.0f, 0.0f, 0.0f, -1.49071198e-1f, -3.73460789e-2f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.00000000e+0f } }, { FrontLeft, { 2.04124145e-1f, 1.08880247e-1f, 0.0f, 1.88586120e-1f, 1.29099444e-1f, 0.0f, 0.0f, 0.0f, 7.45355993e-2f, 3.73460789e-2f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.00000000e+0f } }, { FrontRight, { 2.04124145e-1f, -1.08880247e-1f, 0.0f, 1.88586120e-1f, -1.29099444e-1f, 0.0f, 0.0f, 0.0f, 7.45355993e-2f, -3.73460789e-2f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.00000000e+0f } }, { SideRight, { 2.04124145e-1f, -2.17760495e-1f, 0.0f, 0.00000000e+0f, 0.00000000e+0f, 0.0f, 0.0f, 0.0f, -1.49071198e-1f, 3.73460789e-2f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.00000000e+0f } }, { BackRight, { 2.04124145e-1f, -1.08880247e-1f, 0.0f, -1.88586120e-1f, 1.29099444e-1f, 0.0f, 0.0f, 0.0f, 7.45355993e-2f, -3.73460789e-2f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.00000000e+0f } }, }; static void InitNearFieldCtrl(ALCdevice *device, ALfloat ctrl_dist, ALsizei order, const ALsizei *restrict chans_per_order) { const char *devname = alstr_get_cstr(device->DeviceName); ALsizei i; if(GetConfigValueBool(devname, "decoder", "nfc", 1) && ctrl_dist > 0.0f) { /* NFC is only used when AvgSpeakerDist is greater than 0, and can only * be used when rendering to an ambisonic buffer. */ device->AvgSpeakerDist = minf(ctrl_dist, 10.0f); TRACE("Using near-field reference distance: %.2f meters\n", device->AvgSpeakerDist); for(i = 0;i < order+1;i++) device->NumChannelsPerOrder[i] = chans_per_order[i]; for(;i < MAX_AMBI_ORDER+1;i++) device->NumChannelsPerOrder[i] = 0; } } static void InitDistanceComp(ALCdevice *device, const AmbDecConf *conf, const ALsizei speakermap[MAX_OUTPUT_CHANNELS]) { const char *devname = alstr_get_cstr(device->DeviceName); ALfloat maxdist = 0.0f; size_t total = 0; ALsizei i; for(i = 0;i < conf->NumSpeakers;i++) maxdist = maxf(maxdist, conf->Speakers[i].Distance); if(GetConfigValueBool(devname, "decoder", "distance-comp", 1) && maxdist > 0.0f) { ALfloat srate = (ALfloat)device->Frequency; for(i = 0;i < conf->NumSpeakers;i++) { ALsizei chan = speakermap[i]; ALfloat delay; /* Distance compensation only delays in steps of the sample rate. * This is a bit less accurate since the delay time falls to the * nearest sample time, but it's far simpler as it doesn't have to * deal with phase offsets. This means at 48khz, for instance, the * distance delay will be in steps of about 7 millimeters. */ delay = floorf((maxdist-conf->Speakers[i].Distance) / SPEEDOFSOUNDMETRESPERSEC * srate + 0.5f); if(delay >= (ALfloat)MAX_DELAY_LENGTH) ERR("Delay for speaker \"%s\" exceeds buffer length (%f >= %u)\n", alstr_get_cstr(conf->Speakers[i].Name), delay, MAX_DELAY_LENGTH); device->ChannelDelay[chan].Length = (ALsizei)clampf( delay, 0.0f, (ALfloat)(MAX_DELAY_LENGTH-1) ); device->ChannelDelay[chan].Gain = conf->Speakers[i].Distance / maxdist; TRACE("Channel %u \"%s\" distance compensation: %d samples, %f gain\n", chan, alstr_get_cstr(conf->Speakers[i].Name), device->ChannelDelay[chan].Length, device->ChannelDelay[chan].Gain ); /* Round up to the next 4th sample, so each channel buffer starts * 16-byte aligned. */ total += RoundUp(device->ChannelDelay[chan].Length, 4); } } if(total > 0) { device->ChannelDelay[0].Buffer = al_calloc(16, total * sizeof(ALfloat)); for(i = 1;i < MAX_OUTPUT_CHANNELS;i++) { size_t len = RoundUp(device->ChannelDelay[i-1].Length, 4); device->ChannelDelay[i].Buffer = device->ChannelDelay[i-1].Buffer + len; } } } static void InitPanning(ALCdevice *device) { const ChannelMap *chanmap = NULL; ALsizei coeffcount = 0; ALsizei count = 0; ALsizei i, j; switch(device->FmtChans) { case DevFmtMono: count = COUNTOF(MonoCfg); chanmap = MonoCfg; coeffcount = 1; break; case DevFmtStereo: count = COUNTOF(StereoCfg); chanmap = StereoCfg; coeffcount = 4; break; case DevFmtQuad: count = COUNTOF(QuadCfg); chanmap = QuadCfg; coeffcount = 4; break; case DevFmtX51: count = COUNTOF(X51SideCfg); chanmap = X51SideCfg; coeffcount = 9; break; case DevFmtX51Rear: count = COUNTOF(X51RearCfg); chanmap = X51RearCfg; coeffcount = 9; break; case DevFmtX61: count = COUNTOF(X61Cfg); chanmap = X61Cfg; coeffcount = 9; break; case DevFmtX71: count = COUNTOF(X71Cfg); chanmap = X71Cfg; coeffcount = 16; break; case DevFmtAmbi3D: break; } if(device->FmtChans == DevFmtAmbi3D) { const char *devname = alstr_get_cstr(device->DeviceName); const ALsizei *acnmap = (device->AmbiLayout == AmbiLayout_FuMa) ? FuMa2ACN : ACN2ACN; const ALfloat *n3dscale = (device->AmbiScale == AmbiNorm_FuMa) ? FuMa2N3DScale : (device->AmbiScale == AmbiNorm_SN3D) ? SN3D2N3DScale : /*(device->AmbiScale == AmbiNorm_N3D) ?*/ N3D2N3DScale; ALfloat nfc_delay = 0.0f; count = (device->AmbiOrder == 3) ? 16 : (device->AmbiOrder == 2) ? 9 : (device->AmbiOrder == 1) ? 4 : 1; for(i = 0;i < count;i++) { ALsizei acn = acnmap[i]; device->Dry.Ambi.Map[i].Scale = 1.0f/n3dscale[acn]; device->Dry.Ambi.Map[i].Index = acn; } device->Dry.CoeffCount = 0; device->Dry.NumChannels = count; if(device->AmbiOrder < 2) { device->FOAOut.Ambi = device->Dry.Ambi; device->FOAOut.CoeffCount = device->Dry.CoeffCount; device->FOAOut.NumChannels = 0; } else { ALfloat w_scale=1.0f, xyz_scale=1.0f; /* FOA output is always ACN+N3D for higher-order ambisonic output. * The upsampler expects this and will convert it for output. */ memset(&device->FOAOut.Ambi, 0, sizeof(device->FOAOut.Ambi)); for(i = 0;i < 4;i++) { device->FOAOut.Ambi.Map[i].Scale = 1.0f; device->FOAOut.Ambi.Map[i].Index = i; } device->FOAOut.CoeffCount = 0; device->FOAOut.NumChannels = 4; if(device->AmbiOrder >= 3) { w_scale = W_SCALE_3H3P; xyz_scale = XYZ_SCALE_3H3P; } else { w_scale = W_SCALE_2H2P; xyz_scale = XYZ_SCALE_2H2P; } ambiup_reset(device->AmbiUp, device, w_scale, xyz_scale); } if(ConfigValueFloat(devname, "decoder", "nfc-ref-delay", &nfc_delay) && nfc_delay > 0.0f) { static const ALsizei chans_per_order[MAX_AMBI_ORDER+1] = { 1, 3, 5, 7 }; nfc_delay = clampf(nfc_delay, 0.001f, 1000.0f); InitNearFieldCtrl(device, nfc_delay * SPEEDOFSOUNDMETRESPERSEC, device->AmbiOrder, chans_per_order); } } else { ALfloat w_scale, xyz_scale; SetChannelMap(device->RealOut.ChannelName, device->Dry.Ambi.Coeffs, chanmap, count, &device->Dry.NumChannels); device->Dry.CoeffCount = coeffcount; w_scale = (device->Dry.CoeffCount > 9) ? W_SCALE_3H0P : (device->Dry.CoeffCount > 4) ? W_SCALE_2H0P : 1.0f; xyz_scale = (device->Dry.CoeffCount > 9) ? XYZ_SCALE_3H0P : (device->Dry.CoeffCount > 4) ? XYZ_SCALE_2H0P : 1.0f; memset(&device->FOAOut.Ambi, 0, sizeof(device->FOAOut.Ambi)); for(i = 0;i < device->Dry.NumChannels;i++) { device->FOAOut.Ambi.Coeffs[i][0] = device->Dry.Ambi.Coeffs[i][0] * w_scale; for(j = 1;j < 4;j++) device->FOAOut.Ambi.Coeffs[i][j] = device->Dry.Ambi.Coeffs[i][j] * xyz_scale; } device->FOAOut.CoeffCount = 4; device->FOAOut.NumChannels = 0; } device->RealOut.NumChannels = 0; } static void InitCustomPanning(ALCdevice *device, const AmbDecConf *conf, const ALsizei speakermap[MAX_OUTPUT_CHANNELS]) { ChannelMap chanmap[MAX_OUTPUT_CHANNELS]; const ALfloat *coeff_scale = N3D2N3DScale; ALfloat w_scale = 1.0f; ALfloat xyz_scale = 1.0f; ALsizei i, j; if(conf->FreqBands != 1) ERR("Basic renderer uses the high-frequency matrix as single-band (xover_freq = %.0fhz)\n", conf->XOverFreq); if((conf->ChanMask&AMBI_PERIPHONIC_MASK)) { if(conf->ChanMask > 0x1ff) { w_scale = W_SCALE_3H3P; xyz_scale = XYZ_SCALE_3H3P; } else if(conf->ChanMask > 0xf) { w_scale = W_SCALE_2H2P; xyz_scale = XYZ_SCALE_2H2P; } } else { if(conf->ChanMask > 0x1ff) { w_scale = W_SCALE_3H0P; xyz_scale = XYZ_SCALE_3H0P; } else if(conf->ChanMask > 0xf) { w_scale = W_SCALE_2H0P; xyz_scale = XYZ_SCALE_2H0P; } } if(conf->CoeffScale == ADS_SN3D) coeff_scale = SN3D2N3DScale; else if(conf->CoeffScale == ADS_FuMa) coeff_scale = FuMa2N3DScale; for(i = 0;i < conf->NumSpeakers;i++) { ALsizei chan = speakermap[i]; ALfloat gain; ALsizei k = 0; for(j = 0;j < MAX_AMBI_COEFFS;j++) chanmap[i].Config[j] = 0.0f; chanmap[i].ChanName = device->RealOut.ChannelName[chan]; for(j = 0;j < MAX_AMBI_COEFFS;j++) { if(j == 0) gain = conf->HFOrderGain[0]; else if(j == 1) gain = conf->HFOrderGain[1]; else if(j == 4) gain = conf->HFOrderGain[2]; else if(j == 9) gain = conf->HFOrderGain[3]; if((conf->ChanMask&(1<HFMatrix[i][k++] / coeff_scale[j] * gain; } } SetChannelMap(device->RealOut.ChannelName, device->Dry.Ambi.Coeffs, chanmap, conf->NumSpeakers, &device->Dry.NumChannels); device->Dry.CoeffCount = (conf->ChanMask > 0x1ff) ? 16 : (conf->ChanMask > 0xf) ? 9 : 4; memset(&device->FOAOut.Ambi, 0, sizeof(device->FOAOut.Ambi)); for(i = 0;i < device->Dry.NumChannels;i++) { device->FOAOut.Ambi.Coeffs[i][0] = device->Dry.Ambi.Coeffs[i][0] * w_scale; for(j = 1;j < 4;j++) device->FOAOut.Ambi.Coeffs[i][j] = device->Dry.Ambi.Coeffs[i][j] * xyz_scale; } device->FOAOut.CoeffCount = 4; device->FOAOut.NumChannels = 0; device->RealOut.NumChannels = 0; InitDistanceComp(device, conf, speakermap); } static void InitHQPanning(ALCdevice *device, const AmbDecConf *conf, const ALsizei speakermap[MAX_OUTPUT_CHANNELS]) { static const ALsizei chans_per_order2d[MAX_AMBI_ORDER+1] = { 1, 2, 2, 2 }; static const ALsizei chans_per_order3d[MAX_AMBI_ORDER+1] = { 1, 3, 5, 7 }; ALfloat avg_dist; ALsizei count; ALsizei i; if((conf->ChanMask&AMBI_PERIPHONIC_MASK)) { count = (conf->ChanMask > 0x1ff) ? 16 : (conf->ChanMask > 0xf) ? 9 : 4; for(i = 0;i < count;i++) { device->Dry.Ambi.Map[i].Scale = 1.0f; device->Dry.Ambi.Map[i].Index = i; } } else { static const int map[MAX_AMBI2D_COEFFS] = { 0, 1, 3, 4, 8, 9, 15 }; count = (conf->ChanMask > 0x1ff) ? 7 : (conf->ChanMask > 0xf) ? 5 : 3; for(i = 0;i < count;i++) { device->Dry.Ambi.Map[i].Scale = 1.0f; device->Dry.Ambi.Map[i].Index = map[i]; } } device->Dry.CoeffCount = 0; device->Dry.NumChannels = count; TRACE("Enabling %s-band %s-order%s ambisonic decoder\n", (conf->FreqBands == 1) ? "single" : "dual", (conf->ChanMask > 0xf) ? (conf->ChanMask > 0x1ff) ? "third" : "second" : "first", (conf->ChanMask&AMBI_PERIPHONIC_MASK) ? " periphonic" : "" ); bformatdec_reset(device->AmbiDecoder, conf, count, device->Frequency, speakermap); if(!(conf->ChanMask > 0xf)) { device->FOAOut.Ambi = device->Dry.Ambi; device->FOAOut.CoeffCount = device->Dry.CoeffCount; device->FOAOut.NumChannels = 0; } else { memset(&device->FOAOut.Ambi, 0, sizeof(device->FOAOut.Ambi)); if((conf->ChanMask&AMBI_PERIPHONIC_MASK)) { count = 4; for(i = 0;i < count;i++) { device->FOAOut.Ambi.Map[i].Scale = 1.0f; device->FOAOut.Ambi.Map[i].Index = i; } } else { static const int map[3] = { 0, 1, 3 }; count = 3; for(i = 0;i < count;i++) { device->FOAOut.Ambi.Map[i].Scale = 1.0f; device->FOAOut.Ambi.Map[i].Index = map[i]; } } device->FOAOut.CoeffCount = 0; device->FOAOut.NumChannels = count; } device->RealOut.NumChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); avg_dist = 0.0f; for(i = 0;i < conf->NumSpeakers;i++) avg_dist += conf->Speakers[i].Distance; avg_dist /= (ALfloat)conf->NumSpeakers; InitNearFieldCtrl(device, avg_dist, (conf->ChanMask > 0x1ff) ? 3 : (conf->ChanMask > 0xf) ? 2 : 1, (conf->ChanMask&AMBI_PERIPHONIC_MASK) ? chans_per_order3d : chans_per_order2d ); InitDistanceComp(device, conf, speakermap); } static void InitHrtfPanning(ALCdevice *device) { /* NOTE: azimuth goes clockwise. */ static const struct AngularPoint AmbiPoints[] = { { DEG2RAD( 90.0f), DEG2RAD( 0.0f) }, { DEG2RAD( 35.2643897f), DEG2RAD( 45.0f) }, { DEG2RAD( 35.2643897f), DEG2RAD( 135.0f) }, { DEG2RAD( 35.2643897f), DEG2RAD(-135.0f) }, { DEG2RAD( 35.2643897f), DEG2RAD( -45.0f) }, { DEG2RAD( 0.0f), DEG2RAD( 0.0f) }, { DEG2RAD( 0.0f), DEG2RAD( 45.0f) }, { DEG2RAD( 0.0f), DEG2RAD( 90.0f) }, { DEG2RAD( 0.0f), DEG2RAD( 135.0f) }, { DEG2RAD( 0.0f), DEG2RAD( 180.0f) }, { DEG2RAD( 0.0f), DEG2RAD(-135.0f) }, { DEG2RAD( 0.0f), DEG2RAD( -90.0f) }, { DEG2RAD( 0.0f), DEG2RAD( -45.0f) }, { DEG2RAD(-35.2643897f), DEG2RAD( 45.0f) }, { DEG2RAD(-35.2643897f), DEG2RAD( 135.0f) }, { DEG2RAD(-35.2643897f), DEG2RAD(-135.0f) }, { DEG2RAD(-35.2643897f), DEG2RAD( -45.0f) }, { DEG2RAD(-90.0f), DEG2RAD( 0.0f) }, }; static const ALfloat AmbiMatrixFOA[][MAX_AMBI_COEFFS] = { { 5.55555556e-02f, 0.00000000e+00f, 1.23717915e-01f, 0.00000000e+00f }, { 5.55555556e-02f, -5.00000000e-02f, 7.14285715e-02f, 5.00000000e-02f }, { 5.55555556e-02f, -5.00000000e-02f, 7.14285715e-02f, -5.00000000e-02f }, { 5.55555556e-02f, 5.00000000e-02f, 7.14285715e-02f, -5.00000000e-02f }, { 5.55555556e-02f, 5.00000000e-02f, 7.14285715e-02f, 5.00000000e-02f }, { 5.55555556e-02f, 0.00000000e+00f, 0.00000000e+00f, 8.66025404e-02f }, { 5.55555556e-02f, -6.12372435e-02f, 0.00000000e+00f, 6.12372435e-02f }, { 5.55555556e-02f, -8.66025404e-02f, 0.00000000e+00f, 0.00000000e+00f }, { 5.55555556e-02f, -6.12372435e-02f, 0.00000000e+00f, -6.12372435e-02f }, { 5.55555556e-02f, 0.00000000e+00f, 0.00000000e+00f, -8.66025404e-02f }, { 5.55555556e-02f, 6.12372435e-02f, 0.00000000e+00f, -6.12372435e-02f }, { 5.55555556e-02f, 8.66025404e-02f, 0.00000000e+00f, 0.00000000e+00f }, { 5.55555556e-02f, 6.12372435e-02f, 0.00000000e+00f, 6.12372435e-02f }, { 5.55555556e-02f, -5.00000000e-02f, -7.14285715e-02f, 5.00000000e-02f }, { 5.55555556e-02f, -5.00000000e-02f, -7.14285715e-02f, -5.00000000e-02f }, { 5.55555556e-02f, 5.00000000e-02f, -7.14285715e-02f, -5.00000000e-02f }, { 5.55555556e-02f, 5.00000000e-02f, -7.14285715e-02f, 5.00000000e-02f }, { 5.55555556e-02f, 0.00000000e+00f, -1.23717915e-01f, 0.00000000e+00f }, }, AmbiMatrixHOA[][MAX_AMBI_COEFFS] = { { 5.55555556e-02f, 0.00000000e+00f, 1.23717915e-01f, 0.00000000e+00f, 0.00000000e+00f, 0.00000000e+00f }, { 5.55555556e-02f, -5.00000000e-02f, 7.14285715e-02f, 5.00000000e-02f, -4.55645099e-02f, 0.00000000e+00f }, { 5.55555556e-02f, -5.00000000e-02f, 7.14285715e-02f, -5.00000000e-02f, 4.55645099e-02f, 0.00000000e+00f }, { 5.55555556e-02f, 5.00000000e-02f, 7.14285715e-02f, -5.00000000e-02f, -4.55645099e-02f, 0.00000000e+00f }, { 5.55555556e-02f, 5.00000000e-02f, 7.14285715e-02f, 5.00000000e-02f, 4.55645099e-02f, 0.00000000e+00f }, { 5.55555556e-02f, 0.00000000e+00f, 0.00000000e+00f, 8.66025404e-02f, 0.00000000e+00f, 1.29099445e-01f }, { 5.55555556e-02f, -6.12372435e-02f, 0.00000000e+00f, 6.12372435e-02f, -6.83467648e-02f, 0.00000000e+00f }, { 5.55555556e-02f, -8.66025404e-02f, 0.00000000e+00f, 0.00000000e+00f, 0.00000000e+00f, -1.29099445e-01f }, { 5.55555556e-02f, -6.12372435e-02f, 0.00000000e+00f, -6.12372435e-02f, 6.83467648e-02f, 0.00000000e+00f }, { 5.55555556e-02f, 0.00000000e+00f, 0.00000000e+00f, -8.66025404e-02f, 0.00000000e+00f, 1.29099445e-01f }, { 5.55555556e-02f, 6.12372435e-02f, 0.00000000e+00f, -6.12372435e-02f, -6.83467648e-02f, 0.00000000e+00f }, { 5.55555556e-02f, 8.66025404e-02f, 0.00000000e+00f, 0.00000000e+00f, 0.00000000e+00f, -1.29099445e-01f }, { 5.55555556e-02f, 6.12372435e-02f, 0.00000000e+00f, 6.12372435e-02f, 6.83467648e-02f, 0.00000000e+00f }, { 5.55555556e-02f, -5.00000000e-02f, -7.14285715e-02f, 5.00000000e-02f, -4.55645099e-02f, 0.00000000e+00f }, { 5.55555556e-02f, -5.00000000e-02f, -7.14285715e-02f, -5.00000000e-02f, 4.55645099e-02f, 0.00000000e+00f }, { 5.55555556e-02f, 5.00000000e-02f, -7.14285715e-02f, -5.00000000e-02f, -4.55645099e-02f, 0.00000000e+00f }, { 5.55555556e-02f, 5.00000000e-02f, -7.14285715e-02f, 5.00000000e-02f, 4.55645099e-02f, 0.00000000e+00f }, { 5.55555556e-02f, 0.00000000e+00f, -1.23717915e-01f, 0.00000000e+00f, 0.00000000e+00f, 0.00000000e+00f }, }; static const ALfloat AmbiOrderHFGainFOA[MAX_AMBI_ORDER+1] = { 3.00000000e+00f, 1.73205081e+00f }, AmbiOrderHFGainHOA[MAX_AMBI_ORDER+1] = { 2.40192231e+00f, 1.86052102e+00f, 9.60768923e-01f }; static const ALsizei IndexMap[6] = { 0, 1, 2, 3, 4, 8 }; static const ALsizei ChansPerOrder[MAX_AMBI_ORDER+1] = { 1, 3, 2, 0 }; const ALfloat (*restrict AmbiMatrix)[MAX_AMBI_COEFFS] = AmbiMatrixFOA; const ALfloat *restrict AmbiOrderHFGain = AmbiOrderHFGainFOA; ALsizei count = 4; ALsizei i; static_assert(COUNTOF(AmbiPoints) == COUNTOF(AmbiMatrixFOA), "FOA Ambisonic HRTF mismatch"); static_assert(COUNTOF(AmbiPoints) == COUNTOF(AmbiMatrixHOA), "HOA Ambisonic HRTF mismatch"); if(device->AmbiUp) { AmbiMatrix = AmbiMatrixHOA; AmbiOrderHFGain = AmbiOrderHFGainHOA; count = COUNTOF(IndexMap); } device->Hrtf = al_calloc(16, FAM_SIZE(DirectHrtfState, Chan, count)); for(i = 0;i < count;i++) { device->Dry.Ambi.Map[i].Scale = 1.0f; device->Dry.Ambi.Map[i].Index = IndexMap[i]; } device->Dry.CoeffCount = 0; device->Dry.NumChannels = count; if(device->AmbiUp) { memset(&device->FOAOut.Ambi, 0, sizeof(device->FOAOut.Ambi)); for(i = 0;i < 4;i++) { device->FOAOut.Ambi.Map[i].Scale = 1.0f; device->FOAOut.Ambi.Map[i].Index = i; } device->FOAOut.CoeffCount = 0; device->FOAOut.NumChannels = 4; ambiup_reset(device->AmbiUp, device, AmbiOrderHFGainFOA[0] / AmbiOrderHFGain[0], AmbiOrderHFGainFOA[1] / AmbiOrderHFGain[1]); } else { device->FOAOut.Ambi = device->Dry.Ambi; device->FOAOut.CoeffCount = device->Dry.CoeffCount; device->FOAOut.NumChannels = 0; } device->RealOut.NumChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); BuildBFormatHrtf(device->HrtfHandle, device->Hrtf, device->Dry.NumChannels, AmbiPoints, AmbiMatrix, COUNTOF(AmbiPoints), AmbiOrderHFGain ); InitNearFieldCtrl(device, device->HrtfHandle->distance, device->AmbiUp ? 2 : 1, ChansPerOrder); } static void InitUhjPanning(ALCdevice *device) { ALsizei count = 3; ALsizei i; for(i = 0;i < count;i++) { ALsizei acn = FuMa2ACN[i]; device->Dry.Ambi.Map[i].Scale = 1.0f/FuMa2N3DScale[acn]; device->Dry.Ambi.Map[i].Index = acn; } device->Dry.CoeffCount = 0; device->Dry.NumChannels = count; device->FOAOut.Ambi = device->Dry.Ambi; device->FOAOut.CoeffCount = device->Dry.CoeffCount; device->FOAOut.NumChannels = 0; device->RealOut.NumChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder); } void aluInitRenderer(ALCdevice *device, ALint hrtf_id, enum HrtfRequestMode hrtf_appreq, enum HrtfRequestMode hrtf_userreq) { /* Hold the HRTF the device last used, in case it's used again. */ struct Hrtf *old_hrtf = device->HrtfHandle; const char *mode; bool headphones; int bs2blevel; size_t i; al_free(device->Hrtf); device->Hrtf = NULL; device->HrtfHandle = NULL; alstr_clear(&device->HrtfName); device->Render_Mode = NormalRender; memset(&device->Dry.Ambi, 0, sizeof(device->Dry.Ambi)); device->Dry.CoeffCount = 0; device->Dry.NumChannels = 0; for(i = 0;i < MAX_AMBI_ORDER+1;i++) device->NumChannelsPerOrder[i] = 0; device->AvgSpeakerDist = 0.0f; memset(device->ChannelDelay, 0, sizeof(device->ChannelDelay)); for(i = 0;i < MAX_OUTPUT_CHANNELS;i++) { device->ChannelDelay[i].Gain = 1.0f; device->ChannelDelay[i].Length = 0; } al_free(device->Stablizer); device->Stablizer = NULL; if(device->FmtChans != DevFmtStereo) { ALsizei speakermap[MAX_OUTPUT_CHANNELS]; const char *devname, *layout = NULL; AmbDecConf conf, *pconf = NULL; if(old_hrtf) Hrtf_DecRef(old_hrtf); old_hrtf = NULL; if(hrtf_appreq == Hrtf_Enable) device->HrtfStatus = ALC_HRTF_UNSUPPORTED_FORMAT_SOFT; ambdec_init(&conf); devname = alstr_get_cstr(device->DeviceName); switch(device->FmtChans) { case DevFmtQuad: layout = "quad"; break; case DevFmtX51: /* fall-through */ case DevFmtX51Rear: layout = "surround51"; break; case DevFmtX61: layout = "surround61"; break; case DevFmtX71: layout = "surround71"; break; /* Mono, Stereo, and Ambisonics output don't use custom decoders. */ case DevFmtMono: case DevFmtStereo: case DevFmtAmbi3D: break; } if(layout) { const char *fname; if(ConfigValueStr(devname, "decoder", layout, &fname)) { if(!ambdec_load(&conf, fname)) ERR("Failed to load layout file %s\n", fname); else { if(conf.ChanMask > 0xffff) ERR("Unsupported channel mask 0x%04x (max 0xffff)\n", conf.ChanMask); else { if(MakeSpeakerMap(device, &conf, speakermap)) pconf = &conf; } } } } if(pconf && GetConfigValueBool(devname, "decoder", "hq-mode", 0)) { ambiup_free(&device->AmbiUp); if(!device->AmbiDecoder) device->AmbiDecoder = bformatdec_alloc(); } else { bformatdec_free(&device->AmbiDecoder); if(device->FmtChans != DevFmtAmbi3D || device->AmbiOrder < 2) ambiup_free(&device->AmbiUp); else { if(!device->AmbiUp) device->AmbiUp = ambiup_alloc(); } } if(!pconf) InitPanning(device); else if(device->AmbiDecoder) InitHQPanning(device, pconf, speakermap); else InitCustomPanning(device, pconf, speakermap); /* Enable the stablizer only for formats that have front-left, front- * right, and front-center outputs. */ switch(device->FmtChans) { case DevFmtX51: case DevFmtX51Rear: case DevFmtX61: case DevFmtX71: if(GetConfigValueBool(devname, NULL, "front-stablizer", 0)) { /* Initialize band-splitting filters for the front-left and * front-right channels, with a crossover at 5khz (could be * higher). */ ALfloat scale = (ALfloat)(5000.0 / device->Frequency); FrontStablizer *stablizer = al_calloc(16, sizeof(*stablizer)); bandsplit_init(&stablizer->LFilter, scale); stablizer->RFilter = stablizer->LFilter; /* Initialize all-pass filters for all other channels. */ splitterap_init(&stablizer->APFilter[0], scale); for(i = 1;i < (size_t)device->RealOut.NumChannels;i++) stablizer->APFilter[i] = stablizer->APFilter[0]; device->Stablizer = stablizer; } break; case DevFmtMono: case DevFmtStereo: case DevFmtQuad: case DevFmtAmbi3D: break; } TRACE("Front stablizer %s\n", device->Stablizer ? "enabled" : "disabled"); ambdec_deinit(&conf); return; } bformatdec_free(&device->AmbiDecoder); headphones = device->IsHeadphones; if(device->Type != Loopback) { const char *mode; if(ConfigValueStr(alstr_get_cstr(device->DeviceName), NULL, "stereo-mode", &mode)) { if(strcasecmp(mode, "headphones") == 0) headphones = true; else if(strcasecmp(mode, "speakers") == 0) headphones = false; else if(strcasecmp(mode, "auto") != 0) ERR("Unexpected stereo-mode: %s\n", mode); } } if(hrtf_userreq == Hrtf_Default) { bool usehrtf = (headphones && hrtf_appreq != Hrtf_Disable) || (hrtf_appreq == Hrtf_Enable); if(!usehrtf) goto no_hrtf; device->HrtfStatus = ALC_HRTF_ENABLED_SOFT; if(headphones && hrtf_appreq != Hrtf_Disable) device->HrtfStatus = ALC_HRTF_HEADPHONES_DETECTED_SOFT; } else { if(hrtf_userreq != Hrtf_Enable) { if(hrtf_appreq == Hrtf_Enable) device->HrtfStatus = ALC_HRTF_DENIED_SOFT; goto no_hrtf; } device->HrtfStatus = ALC_HRTF_REQUIRED_SOFT; } if(VECTOR_SIZE(device->HrtfList) == 0) { VECTOR_DEINIT(device->HrtfList); device->HrtfList = EnumerateHrtf(device->DeviceName); } if(hrtf_id >= 0 && (size_t)hrtf_id < VECTOR_SIZE(device->HrtfList)) { const EnumeratedHrtf *entry = &VECTOR_ELEM(device->HrtfList, hrtf_id); struct Hrtf *hrtf = GetLoadedHrtf(entry->hrtf); if(hrtf && hrtf->sampleRate == device->Frequency) { device->HrtfHandle = hrtf; alstr_copy(&device->HrtfName, entry->name); } else if(hrtf) Hrtf_DecRef(hrtf); } for(i = 0;!device->HrtfHandle && i < VECTOR_SIZE(device->HrtfList);i++) { const EnumeratedHrtf *entry = &VECTOR_ELEM(device->HrtfList, i); struct Hrtf *hrtf = GetLoadedHrtf(entry->hrtf); if(hrtf && hrtf->sampleRate == device->Frequency) { device->HrtfHandle = hrtf; alstr_copy(&device->HrtfName, entry->name); } else if(hrtf) Hrtf_DecRef(hrtf); } if(device->HrtfHandle) { if(old_hrtf) Hrtf_DecRef(old_hrtf); old_hrtf = NULL; device->Render_Mode = HrtfRender; if(ConfigValueStr(alstr_get_cstr(device->DeviceName), NULL, "hrtf-mode", &mode)) { if(strcasecmp(mode, "full") == 0) device->Render_Mode = HrtfRender; else if(strcasecmp(mode, "basic") == 0) device->Render_Mode = NormalRender; else ERR("Unexpected hrtf-mode: %s\n", mode); } if(device->Render_Mode == HrtfRender) { /* Don't bother with HOA when using full HRTF rendering. Nothing * needs it, and it eases the CPU/memory load. */ ambiup_free(&device->AmbiUp); } else { if(!device->AmbiUp) device->AmbiUp = ambiup_alloc(); } TRACE("%s HRTF rendering enabled, using \"%s\"\n", ((device->Render_Mode == HrtfRender) ? "Full" : "Basic"), alstr_get_cstr(device->HrtfName) ); InitHrtfPanning(device); return; } device->HrtfStatus = ALC_HRTF_UNSUPPORTED_FORMAT_SOFT; no_hrtf: if(old_hrtf) Hrtf_DecRef(old_hrtf); old_hrtf = NULL; TRACE("HRTF disabled\n"); device->Render_Mode = StereoPair; ambiup_free(&device->AmbiUp); bs2blevel = ((headphones && hrtf_appreq != Hrtf_Disable) || (hrtf_appreq == Hrtf_Enable)) ? 5 : 0; if(device->Type != Loopback) ConfigValueInt(alstr_get_cstr(device->DeviceName), NULL, "cf_level", &bs2blevel); if(bs2blevel > 0 && bs2blevel <= 6) { device->Bs2b = al_calloc(16, sizeof(*device->Bs2b)); bs2b_set_params(device->Bs2b, bs2blevel, device->Frequency); TRACE("BS2B enabled\n"); InitPanning(device); return; } TRACE("BS2B disabled\n"); if(ConfigValueStr(alstr_get_cstr(device->DeviceName), NULL, "stereo-encoding", &mode)) { if(strcasecmp(mode, "uhj") == 0) device->Render_Mode = NormalRender; else if(strcasecmp(mode, "panpot") != 0) ERR("Unexpected stereo-encoding: %s\n", mode); } if(device->Render_Mode == NormalRender) { device->Uhj_Encoder = al_calloc(16, sizeof(Uhj2Encoder)); TRACE("UHJ enabled\n"); InitUhjPanning(device); return; } TRACE("UHJ disabled\n"); InitPanning(device); } void aluInitEffectPanning(ALeffectslot *slot) { ALsizei i; memset(slot->ChanMap, 0, sizeof(slot->ChanMap)); slot->NumChannels = 0; for(i = 0;i < MAX_EFFECT_CHANNELS;i++) { slot->ChanMap[i].Scale = 1.0f; slot->ChanMap[i].Index = i; } slot->NumChannels = i; } openal-soft-openal-soft-1.19.1/Alc/polymorphism.h000066400000000000000000000126571335774445300217230ustar00rootroot00000000000000#ifndef POLYMORPHISM_H #define POLYMORPHISM_H /* Macros to declare inheriting types, and to (down-)cast and up-cast. */ #define DERIVE_FROM_TYPE(t) t t##_parent #define STATIC_CAST(to, obj) (&(obj)->to##_parent) #ifdef __GNUC__ #define STATIC_UPCAST(to, from, obj) __extension__({ \ static_assert(__builtin_types_compatible_p(from, __typeof(*(obj))), \ "Invalid upcast object from type"); \ (to*)((char*)(obj) - offsetof(to, from##_parent)); \ }) #else #define STATIC_UPCAST(to, from, obj) ((to*)((char*)(obj) - offsetof(to, from##_parent))) #endif /* Defines method forwards, which call the given parent's (T2's) implementation. */ #define DECLARE_FORWARD(T1, T2, rettype, func) \ rettype T1##_##func(T1 *obj) \ { return T2##_##func(STATIC_CAST(T2, obj)); } #define DECLARE_FORWARD1(T1, T2, rettype, func, argtype1) \ rettype T1##_##func(T1 *obj, argtype1 a) \ { return T2##_##func(STATIC_CAST(T2, obj), a); } #define DECLARE_FORWARD2(T1, T2, rettype, func, argtype1, argtype2) \ rettype T1##_##func(T1 *obj, argtype1 a, argtype2 b) \ { return T2##_##func(STATIC_CAST(T2, obj), a, b); } #define DECLARE_FORWARD3(T1, T2, rettype, func, argtype1, argtype2, argtype3) \ rettype T1##_##func(T1 *obj, argtype1 a, argtype2 b, argtype3 c) \ { return T2##_##func(STATIC_CAST(T2, obj), a, b, c); } /* Defines method thunks, functions that call to the child's method. */ #define DECLARE_THUNK(T1, T2, rettype, func) \ static rettype T1##_##T2##_##func(T2 *obj) \ { return T1##_##func(STATIC_UPCAST(T1, T2, obj)); } #define DECLARE_THUNK1(T1, T2, rettype, func, argtype1) \ static rettype T1##_##T2##_##func(T2 *obj, argtype1 a) \ { return T1##_##func(STATIC_UPCAST(T1, T2, obj), a); } #define DECLARE_THUNK2(T1, T2, rettype, func, argtype1, argtype2) \ static rettype T1##_##T2##_##func(T2 *obj, argtype1 a, argtype2 b) \ { return T1##_##func(STATIC_UPCAST(T1, T2, obj), a, b); } #define DECLARE_THUNK3(T1, T2, rettype, func, argtype1, argtype2, argtype3) \ static rettype T1##_##T2##_##func(T2 *obj, argtype1 a, argtype2 b, argtype3 c) \ { return T1##_##func(STATIC_UPCAST(T1, T2, obj), a, b, c); } #define DECLARE_THUNK4(T1, T2, rettype, func, argtype1, argtype2, argtype3, argtype4) \ static rettype T1##_##T2##_##func(T2 *obj, argtype1 a, argtype2 b, argtype3 c, argtype4 d) \ { return T1##_##func(STATIC_UPCAST(T1, T2, obj), a, b, c, d); } /* Defines the default functions used to (de)allocate a polymorphic object. */ #define DECLARE_DEFAULT_ALLOCATORS(T) \ static void* T##_New(size_t size) { return al_malloc(16, size); } \ static void T##_Delete(void *ptr) { al_free(ptr); } /* Helper to extract an argument list for virtual method calls. */ #define EXTRACT_VCALL_ARGS(...) __VA_ARGS__)) /* Call a "virtual" method on an object, with arguments. */ #define V(obj, func) ((obj)->vtbl->func((obj), EXTRACT_VCALL_ARGS /* Call a "virtual" method on an object, with no arguments. */ #define V0(obj, func) ((obj)->vtbl->func((obj) EXTRACT_VCALL_ARGS /* Helper to extract an argument list for NEW_OBJ calls. */ #define EXTRACT_NEW_ARGS(...) __VA_ARGS__); \ } \ } while(0) /* Allocate and construct an object, with arguments. */ #define NEW_OBJ(_res, T) do { \ _res = T##_New(sizeof(T)); \ if(_res) \ { \ memset(_res, 0, sizeof(T)); \ T##_Construct(_res, EXTRACT_NEW_ARGS /* Allocate and construct an object, with no arguments. */ #define NEW_OBJ0(_res, T) do { \ _res = T##_New(sizeof(T)); \ if(_res) \ { \ memset(_res, 0, sizeof(T)); \ T##_Construct(_res EXTRACT_NEW_ARGS /* Destructs and deallocate an object. */ #define DELETE_OBJ(obj) do { \ if((obj) != NULL) \ { \ V0((obj),Destruct)(); \ V0((obj),Delete)(); \ } \ } while(0) /* Helper to get a type's vtable thunk for a child type. */ #define GET_VTABLE2(T1, T2) (&(T1##_##T2##_vtable)) /* Helper to set an object's vtable thunk for a child type. Used when constructing an object. */ #define SET_VTABLE2(T1, T2, obj) (STATIC_CAST(T2, obj)->vtbl = GET_VTABLE2(T1, T2)) #endif /* POLYMORPHISM_H */ openal-soft-openal-soft-1.19.1/Alc/ringbuffer.c000066400000000000000000000202501335774445300212710ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include "ringbuffer.h" #include "align.h" #include "atomic.h" #include "threads.h" #include "almalloc.h" #include "compat.h" /* NOTE: This lockless ringbuffer implementation is copied from JACK, extended * to include an element size. Consequently, parameters and return values for a * size or count is in 'elements', not bytes. Additionally, it only supports * single-consumer/single-provider operation. */ struct ll_ringbuffer { ATOMIC(size_t) write_ptr; ATOMIC(size_t) read_ptr; size_t size; size_t size_mask; size_t elem_size; alignas(16) char buf[]; }; ll_ringbuffer_t *ll_ringbuffer_create(size_t sz, size_t elem_sz, int limit_writes) { ll_ringbuffer_t *rb; size_t power_of_two = 0; if(sz > 0) { power_of_two = sz; power_of_two |= power_of_two>>1; power_of_two |= power_of_two>>2; power_of_two |= power_of_two>>4; power_of_two |= power_of_two>>8; power_of_two |= power_of_two>>16; #if SIZE_MAX > UINT_MAX power_of_two |= power_of_two>>32; #endif } power_of_two++; if(power_of_two < sz) return NULL; rb = al_malloc(16, sizeof(*rb) + power_of_two*elem_sz); if(!rb) return NULL; ATOMIC_INIT(&rb->write_ptr, 0); ATOMIC_INIT(&rb->read_ptr, 0); rb->size = limit_writes ? sz : power_of_two; rb->size_mask = power_of_two - 1; rb->elem_size = elem_sz; return rb; } void ll_ringbuffer_free(ll_ringbuffer_t *rb) { al_free(rb); } void ll_ringbuffer_reset(ll_ringbuffer_t *rb) { ATOMIC_STORE(&rb->write_ptr, 0, almemory_order_release); ATOMIC_STORE(&rb->read_ptr, 0, almemory_order_release); memset(rb->buf, 0, (rb->size_mask+1)*rb->elem_size); } size_t ll_ringbuffer_read_space(const ll_ringbuffer_t *rb) { size_t w = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->write_ptr, almemory_order_acquire); size_t r = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->read_ptr, almemory_order_acquire); return (w-r) & rb->size_mask; } size_t ll_ringbuffer_write_space(const ll_ringbuffer_t *rb) { size_t w = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->write_ptr, almemory_order_acquire); size_t r = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->read_ptr, almemory_order_acquire); w = (r-w-1) & rb->size_mask; return (w > rb->size) ? rb->size : w; } size_t ll_ringbuffer_read(ll_ringbuffer_t *rb, char *dest, size_t cnt) { size_t read_ptr; size_t free_cnt; size_t cnt2; size_t to_read; size_t n1, n2; free_cnt = ll_ringbuffer_read_space(rb); if(free_cnt == 0) return 0; to_read = (cnt > free_cnt) ? free_cnt : cnt; read_ptr = ATOMIC_LOAD(&rb->read_ptr, almemory_order_relaxed) & rb->size_mask; cnt2 = read_ptr + to_read; if(cnt2 > rb->size_mask+1) { n1 = rb->size_mask+1 - read_ptr; n2 = cnt2 & rb->size_mask; } else { n1 = to_read; n2 = 0; } memcpy(dest, &rb->buf[read_ptr*rb->elem_size], n1*rb->elem_size); read_ptr += n1; if(n2) { memcpy(dest + n1*rb->elem_size, &rb->buf[(read_ptr&rb->size_mask)*rb->elem_size], n2*rb->elem_size); read_ptr += n2; } ATOMIC_STORE(&rb->read_ptr, read_ptr, almemory_order_release); return to_read; } size_t ll_ringbuffer_peek(ll_ringbuffer_t *rb, char *dest, size_t cnt) { size_t free_cnt; size_t cnt2; size_t to_read; size_t n1, n2; size_t read_ptr; free_cnt = ll_ringbuffer_read_space(rb); if(free_cnt == 0) return 0; to_read = (cnt > free_cnt) ? free_cnt : cnt; read_ptr = ATOMIC_LOAD(&rb->read_ptr, almemory_order_relaxed) & rb->size_mask; cnt2 = read_ptr + to_read; if(cnt2 > rb->size_mask+1) { n1 = rb->size_mask+1 - read_ptr; n2 = cnt2 & rb->size_mask; } else { n1 = to_read; n2 = 0; } memcpy(dest, &rb->buf[read_ptr*rb->elem_size], n1*rb->elem_size); if(n2) { read_ptr += n1; memcpy(dest + n1*rb->elem_size, &rb->buf[(read_ptr&rb->size_mask)*rb->elem_size], n2*rb->elem_size); } return to_read; } size_t ll_ringbuffer_write(ll_ringbuffer_t *rb, const char *src, size_t cnt) { size_t write_ptr; size_t free_cnt; size_t cnt2; size_t to_write; size_t n1, n2; free_cnt = ll_ringbuffer_write_space(rb); if(free_cnt == 0) return 0; to_write = (cnt > free_cnt) ? free_cnt : cnt; write_ptr = ATOMIC_LOAD(&rb->write_ptr, almemory_order_relaxed) & rb->size_mask; cnt2 = write_ptr + to_write; if(cnt2 > rb->size_mask+1) { n1 = rb->size_mask+1 - write_ptr; n2 = cnt2 & rb->size_mask; } else { n1 = to_write; n2 = 0; } memcpy(&rb->buf[write_ptr*rb->elem_size], src, n1*rb->elem_size); write_ptr += n1; if(n2) { memcpy(&rb->buf[(write_ptr&rb->size_mask)*rb->elem_size], src + n1*rb->elem_size, n2*rb->elem_size); write_ptr += n2; } ATOMIC_STORE(&rb->write_ptr, write_ptr, almemory_order_release); return to_write; } void ll_ringbuffer_read_advance(ll_ringbuffer_t *rb, size_t cnt) { ATOMIC_ADD(&rb->read_ptr, cnt, almemory_order_acq_rel); } void ll_ringbuffer_write_advance(ll_ringbuffer_t *rb, size_t cnt) { ATOMIC_ADD(&rb->write_ptr, cnt, almemory_order_acq_rel); } void ll_ringbuffer_get_read_vector(const ll_ringbuffer_t *rb, ll_ringbuffer_data_t vec[2]) { size_t free_cnt; size_t cnt2; size_t w, r; w = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->write_ptr, almemory_order_acquire); r = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->read_ptr, almemory_order_acquire); w &= rb->size_mask; r &= rb->size_mask; free_cnt = (w-r) & rb->size_mask; cnt2 = r + free_cnt; if(cnt2 > rb->size_mask+1) { /* Two part vector: the rest of the buffer after the current write ptr, * plus some from the start of the buffer. */ vec[0].buf = (char*)&rb->buf[r*rb->elem_size]; vec[0].len = rb->size_mask+1 - r; vec[1].buf = (char*)rb->buf; vec[1].len = cnt2 & rb->size_mask; } else { /* Single part vector: just the rest of the buffer */ vec[0].buf = (char*)&rb->buf[r*rb->elem_size]; vec[0].len = free_cnt; vec[1].buf = NULL; vec[1].len = 0; } } void ll_ringbuffer_get_write_vector(const ll_ringbuffer_t *rb, ll_ringbuffer_data_t vec[2]) { size_t free_cnt; size_t cnt2; size_t w, r; w = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->write_ptr, almemory_order_acquire); r = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->read_ptr, almemory_order_acquire); w &= rb->size_mask; r &= rb->size_mask; free_cnt = (r-w-1) & rb->size_mask; if(free_cnt > rb->size) free_cnt = rb->size; cnt2 = w + free_cnt; if(cnt2 > rb->size_mask+1) { /* Two part vector: the rest of the buffer after the current write ptr, * plus some from the start of the buffer. */ vec[0].buf = (char*)&rb->buf[w*rb->elem_size]; vec[0].len = rb->size_mask+1 - w; vec[1].buf = (char*)rb->buf; vec[1].len = cnt2 & rb->size_mask; } else { vec[0].buf = (char*)&rb->buf[w*rb->elem_size]; vec[0].len = free_cnt; vec[1].buf = NULL; vec[1].len = 0; } } openal-soft-openal-soft-1.19.1/Alc/ringbuffer.h000066400000000000000000000053711335774445300213050ustar00rootroot00000000000000#ifndef RINGBUFFER_H #define RINGBUFFER_H #include #ifdef __cplusplus extern "C" { #endif typedef struct ll_ringbuffer ll_ringbuffer_t; typedef struct ll_ringbuffer_data { char *buf; size_t len; } ll_ringbuffer_data_t; /** * Create a new ringbuffer to hold at least `sz' elements of `elem_sz' bytes. * The number of elements is rounded up to the next power of two (even if it is * already a power of two, to ensure the requested amount can be written). */ ll_ringbuffer_t *ll_ringbuffer_create(size_t sz, size_t elem_sz, int limit_writes); /** Free all data associated with the ringbuffer `rb'. */ void ll_ringbuffer_free(ll_ringbuffer_t *rb); /** Reset the read and write pointers to zero. This is not thread safe. */ void ll_ringbuffer_reset(ll_ringbuffer_t *rb); /** * The non-copying data reader. `vec' is an array of two places. Set the values * at `vec' to hold the current readable data at `rb'. If the readable data is * in one segment the second segment has zero length. */ void ll_ringbuffer_get_read_vector(const ll_ringbuffer_t *rb, ll_ringbuffer_data_t vec[2]); /** * The non-copying data writer. `vec' is an array of two places. Set the values * at `vec' to hold the current writeable data at `rb'. If the writeable data * is in one segment the second segment has zero length. */ void ll_ringbuffer_get_write_vector(const ll_ringbuffer_t *rb, ll_ringbuffer_data_t vec[2]); /** * Return the number of elements available for reading. This is the number of * elements in front of the read pointer and behind the write pointer. */ size_t ll_ringbuffer_read_space(const ll_ringbuffer_t *rb); /** * The copying data reader. Copy at most `cnt' elements from `rb' to `dest'. * Returns the actual number of elements copied. */ size_t ll_ringbuffer_read(ll_ringbuffer_t *rb, char *dest, size_t cnt); /** * The copying data reader w/o read pointer advance. Copy at most `cnt' * elements from `rb' to `dest'. Returns the actual number of elements copied. */ size_t ll_ringbuffer_peek(ll_ringbuffer_t *rb, char *dest, size_t cnt); /** Advance the read pointer `cnt' places. */ void ll_ringbuffer_read_advance(ll_ringbuffer_t *rb, size_t cnt); /** * Return the number of elements available for writing. This is the number of * elements in front of the write pointer and behind the read pointer. */ size_t ll_ringbuffer_write_space(const ll_ringbuffer_t *rb); /** * The copying data writer. Copy at most `cnt' elements to `rb' from `src'. * Returns the actual number of elements copied. */ size_t ll_ringbuffer_write(ll_ringbuffer_t *rb, const char *src, size_t cnt); /** Advance the write pointer `cnt' places. */ void ll_ringbuffer_write_advance(ll_ringbuffer_t *rb, size_t cnt); #ifdef __cplusplus } /* extern "C" */ #endif #endif /* RINGBUFFER_H */ openal-soft-openal-soft-1.19.1/Alc/uhjfilter.c000066400000000000000000000106621335774445300211420ustar00rootroot00000000000000 #include "config.h" #include "alu.h" #include "uhjfilter.h" /* This is the maximum number of samples processed for each inner loop * iteration. */ #define MAX_UPDATE_SAMPLES 128 static const ALfloat Filter1CoeffSqr[4] = { 0.479400865589f, 0.876218493539f, 0.976597589508f, 0.997499255936f }; static const ALfloat Filter2CoeffSqr[4] = { 0.161758498368f, 0.733028932341f, 0.945349700329f, 0.990599156685f }; static void allpass_process(AllPassState *state, ALfloat *restrict dst, const ALfloat *restrict src, const ALfloat aa, ALsizei todo) { ALfloat z1 = state->z[0]; ALfloat z2 = state->z[1]; ALsizei i; for(i = 0;i < todo;i++) { ALfloat input = src[i]; ALfloat output = input*aa + z1; z1 = z2; z2 = output*aa - input; dst[i] = output; } state->z[0] = z1; state->z[1] = z2; } /* NOTE: There seems to be a bit of an inconsistency in how this encoding is * supposed to work. Some references, such as * * http://members.tripod.com/martin_leese/Ambisonic/UHJ_file_format.html * * specify a pre-scaling of sqrt(2) on the W channel input, while other * references, such as * * https://en.wikipedia.org/wiki/Ambisonic_UHJ_format#Encoding.5B1.5D * and * https://wiki.xiph.org/Ambisonics#UHJ_format * * do not. The sqrt(2) scaling is in line with B-Format decoder coefficients * which include such a scaling for the W channel input, however the original * source for this equation is a 1985 paper by Michael Gerzon, which does not * apparently include the scaling. Applying the extra scaling creates a louder * result with a narrower stereo image compared to not scaling, and I don't * know which is the intended result. */ void EncodeUhj2(Uhj2Encoder *enc, ALfloat *restrict LeftOut, ALfloat *restrict RightOut, ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo) { ALfloat D[MAX_UPDATE_SAMPLES], S[MAX_UPDATE_SAMPLES]; ALfloat temp[2][MAX_UPDATE_SAMPLES]; ALsizei base, i; ASSUME(SamplesToDo > 0); for(base = 0;base < SamplesToDo;) { ALsizei todo = mini(SamplesToDo - base, MAX_UPDATE_SAMPLES); ASSUME(todo > 0); /* D = 0.6554516*Y */ for(i = 0;i < todo;i++) temp[0][i] = 0.6554516f*InSamples[2][base+i]; allpass_process(&enc->Filter1_Y[0], temp[1], temp[0], Filter1CoeffSqr[0], todo); allpass_process(&enc->Filter1_Y[1], temp[0], temp[1], Filter1CoeffSqr[1], todo); allpass_process(&enc->Filter1_Y[2], temp[1], temp[0], Filter1CoeffSqr[2], todo); allpass_process(&enc->Filter1_Y[3], temp[0], temp[1], Filter1CoeffSqr[3], todo); /* NOTE: Filter1 requires a 1 sample delay for the final output, so * take the last processed sample from the previous run as the first * output sample. */ D[0] = enc->LastY; for(i = 1;i < todo;i++) D[i] = temp[0][i-1]; enc->LastY = temp[0][i-1]; /* D += j(-0.3420201*W + 0.5098604*X) */ for(i = 0;i < todo;i++) temp[0][i] = -0.3420201f*InSamples[0][base+i] + 0.5098604f*InSamples[1][base+i]; allpass_process(&enc->Filter2_WX[0], temp[1], temp[0], Filter2CoeffSqr[0], todo); allpass_process(&enc->Filter2_WX[1], temp[0], temp[1], Filter2CoeffSqr[1], todo); allpass_process(&enc->Filter2_WX[2], temp[1], temp[0], Filter2CoeffSqr[2], todo); allpass_process(&enc->Filter2_WX[3], temp[0], temp[1], Filter2CoeffSqr[3], todo); for(i = 0;i < todo;i++) D[i] += temp[0][i]; /* S = 0.9396926*W + 0.1855740*X */ for(i = 0;i < todo;i++) temp[0][i] = 0.9396926f*InSamples[0][base+i] + 0.1855740f*InSamples[1][base+i]; allpass_process(&enc->Filter1_WX[0], temp[1], temp[0], Filter1CoeffSqr[0], todo); allpass_process(&enc->Filter1_WX[1], temp[0], temp[1], Filter1CoeffSqr[1], todo); allpass_process(&enc->Filter1_WX[2], temp[1], temp[0], Filter1CoeffSqr[2], todo); allpass_process(&enc->Filter1_WX[3], temp[0], temp[1], Filter1CoeffSqr[3], todo); S[0] = enc->LastWX; for(i = 1;i < todo;i++) S[i] = temp[0][i-1]; enc->LastWX = temp[0][i-1]; /* Left = (S + D)/2.0 */ for(i = 0;i < todo;i++) *(LeftOut++) += (S[i] + D[i]) * 0.5f; /* Right = (S - D)/2.0 */ for(i = 0;i < todo;i++) *(RightOut++) += (S[i] - D[i]) * 0.5f; base += todo; } } openal-soft-openal-soft-1.19.1/Alc/uhjfilter.h000066400000000000000000000032711335774445300211450ustar00rootroot00000000000000#ifndef UHJFILTER_H #define UHJFILTER_H #include "AL/al.h" #include "alMain.h" typedef struct AllPassState { ALfloat z[2]; } AllPassState; /* Encoding 2-channel UHJ from B-Format is done as: * * S = 0.9396926*W + 0.1855740*X * D = j(-0.3420201*W + 0.5098604*X) + 0.6554516*Y * * Left = (S + D)/2.0 * Right = (S - D)/2.0 * * where j is a wide-band +90 degree phase shift. * * The phase shift is done using a Hilbert transform, described here: * https://web.archive.org/web/20060708031958/http://www.biochem.oulu.fi/~oniemita/dsp/hilbert/ * It works using 2 sets of 4 chained filters. The first filter chain produces * a phase shift of varying magnitude over a wide range of frequencies, while * the second filter chain produces a phase shift 90 degrees ahead of the * first over the same range. * * Combining these two stages requires the use of three filter chains. S- * channel output uses a Filter1 chain on the W and X channel mix, while the D- * channel output uses a Filter1 chain on the Y channel plus a Filter2 chain on * the W and X channel mix. This results in the W and X input mix on the D- * channel output having the required +90 degree phase shift relative to the * other inputs. */ typedef struct Uhj2Encoder { AllPassState Filter1_Y[4]; AllPassState Filter2_WX[4]; AllPassState Filter1_WX[4]; ALfloat LastY, LastWX; } Uhj2Encoder; /* Encodes a 2-channel UHJ (stereo-compatible) signal from a B-Format input * signal. The input must use FuMa channel ordering and scaling. */ void EncodeUhj2(Uhj2Encoder *enc, ALfloat *restrict LeftOut, ALfloat *restrict RightOut, ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo); #endif /* UHJFILTER_H */ openal-soft-openal-soft-1.19.1/Alc/vector.h000066400000000000000000000120311335774445300204450ustar00rootroot00000000000000#ifndef AL_VECTOR_H #define AL_VECTOR_H #include #include #include "almalloc.h" #define TYPEDEF_VECTOR(T, N) typedef struct { \ size_t Capacity; \ size_t Size; \ T Data[]; \ } _##N; \ typedef _##N* N; \ typedef const _##N* const_##N; #define VECTOR(T) struct { \ size_t Capacity; \ size_t Size; \ T Data[]; \ }* #define VECTOR_INIT(_x) do { (_x) = NULL; } while(0) #define VECTOR_INIT_STATIC() NULL #define VECTOR_DEINIT(_x) do { al_free((_x)); (_x) = NULL; } while(0) #define VECTOR_RESIZE(_x, _s, _c) do { \ size_t _size = (_s); \ size_t _cap = (_c); \ if(_size > _cap) \ _cap = _size; \ \ if(!(_x) && _cap == 0) \ break; \ \ if(((_x) ? (_x)->Capacity : 0) < _cap) \ { \ ptrdiff_t data_offset = (_x) ? (char*)((_x)->Data) - (char*)(_x) : \ sizeof(*(_x)); \ size_t old_size = ((_x) ? (_x)->Size : 0); \ void *temp; \ \ temp = al_calloc(16, data_offset + sizeof((_x)->Data[0])*_cap); \ assert(temp != NULL); \ if((_x)) \ memcpy(((char*)temp)+data_offset, (_x)->Data, \ sizeof((_x)->Data[0])*old_size); \ \ al_free((_x)); \ (_x) = temp; \ (_x)->Capacity = _cap; \ } \ (_x)->Size = _size; \ } while(0) \ #define VECTOR_CAPACITY(_x) ((_x) ? (_x)->Capacity : 0) #define VECTOR_SIZE(_x) ((_x) ? (_x)->Size : 0) #define VECTOR_BEGIN(_x) ((_x) ? (_x)->Data + 0 : NULL) #define VECTOR_END(_x) ((_x) ? (_x)->Data + (_x)->Size : NULL) #define VECTOR_PUSH_BACK(_x, _obj) do { \ size_t _pbsize = VECTOR_SIZE(_x)+1; \ VECTOR_RESIZE(_x, _pbsize, _pbsize); \ (_x)->Data[(_x)->Size-1] = (_obj); \ } while(0) #define VECTOR_POP_BACK(_x) ((void)((_x)->Size--)) #define VECTOR_BACK(_x) ((_x)->Data[(_x)->Size-1]) #define VECTOR_FRONT(_x) ((_x)->Data[0]) #define VECTOR_ELEM(_x, _o) ((_x)->Data[(_o)]) #define VECTOR_FOR_EACH(_t, _x, _f) do { \ _t *_iter = VECTOR_BEGIN((_x)); \ _t *_end = VECTOR_END((_x)); \ for(;_iter != _end;++_iter) \ _f(_iter); \ } while(0) #define VECTOR_FIND_IF(_i, _t, _x, _f) do { \ _t *_iter = VECTOR_BEGIN((_x)); \ _t *_end = VECTOR_END((_x)); \ for(;_iter != _end;++_iter) \ { \ if(_f(_iter)) \ break; \ } \ (_i) = _iter; \ } while(0) #endif /* AL_VECTOR_H */ openal-soft-openal-soft-1.19.1/CMakeLists.txt000066400000000000000000001656741335774445300210610ustar00rootroot00000000000000# CMake build file list for OpenAL CMAKE_MINIMUM_REQUIRED(VERSION 3.0.2) PROJECT(OpenAL) IF(COMMAND CMAKE_POLICY) CMAKE_POLICY(SET CMP0003 NEW) CMAKE_POLICY(SET CMP0005 NEW) IF(POLICY CMP0020) CMAKE_POLICY(SET CMP0020 NEW) ENDIF(POLICY CMP0020) IF(POLICY CMP0042) CMAKE_POLICY(SET CMP0042 NEW) ENDIF(POLICY CMP0042) IF(POLICY CMP0054) CMAKE_POLICY(SET CMP0054 NEW) ENDIF(POLICY CMP0054) ENDIF(COMMAND CMAKE_POLICY) SET(CMAKE_MODULE_PATH "${OpenAL_SOURCE_DIR}/cmake") INCLUDE(CheckFunctionExists) INCLUDE(CheckLibraryExists) INCLUDE(CheckSharedFunctionExists) INCLUDE(CheckIncludeFile) INCLUDE(CheckIncludeFiles) INCLUDE(CheckSymbolExists) INCLUDE(CheckCCompilerFlag) INCLUDE(CheckCXXCompilerFlag) INCLUDE(CheckCSourceCompiles) INCLUDE(CheckTypeSize) include(CheckStructHasMember) include(CheckFileOffsetBits) include(GNUInstallDirs) SET(CMAKE_ALLOW_LOOSE_LOOP_CONSTRUCTS TRUE) OPTION(ALSOFT_DLOPEN "Check for the dlopen API for loading optional libs" ON) OPTION(ALSOFT_WERROR "Treat compile warnings as errors" OFF) OPTION(ALSOFT_UTILS "Build and install utility programs" ON) OPTION(ALSOFT_NO_CONFIG_UTIL "Disable building the alsoft-config utility" OFF) OPTION(ALSOFT_EXAMPLES "Build and install example programs" ON) OPTION(ALSOFT_TESTS "Build and install test programs" ON) OPTION(ALSOFT_CONFIG "Install alsoft.conf sample configuration file" ON) OPTION(ALSOFT_HRTF_DEFS "Install HRTF definition files" ON) OPTION(ALSOFT_AMBDEC_PRESETS "Install AmbDec preset files" ON) OPTION(ALSOFT_INSTALL "Install headers and libraries" ON) if(DEFINED SHARE_INSTALL_DIR) message(WARNING "SHARE_INSTALL_DIR is deprecated. 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FORCE) ENDIF() IF(MSVC) SET(CPP_DEFS ${CPP_DEFS} _CRT_SECURE_NO_WARNINGS _CRT_NONSTDC_NO_DEPRECATE) SET(C_FLAGS ${C_FLAGS} /wd4098) IF(NOT DXSDK_DIR) STRING(REGEX REPLACE "\\\\" "/" DXSDK_DIR "$ENV{DXSDK_DIR}") ELSE() STRING(REGEX REPLACE "\\\\" "/" DXSDK_DIR "${DXSDK_DIR}") ENDIF() IF(DXSDK_DIR) MESSAGE(STATUS "Using DirectX SDK directory: ${DXSDK_DIR}") ENDIF() OPTION(FORCE_STATIC_VCRT "Force /MT for static VC runtimes" OFF) IF(FORCE_STATIC_VCRT) FOREACH(flag_var CMAKE_C_FLAGS CMAKE_C_FLAGS_DEBUG CMAKE_C_FLAGS_RELEASE CMAKE_C_FLAGS_MINSIZEREL CMAKE_C_FLAGS_RELWITHDEBINFO) IF(${flag_var} MATCHES "/MD") STRING(REGEX REPLACE "/MD" "/MT" ${flag_var} "${${flag_var}}") ENDIF() ENDFOREACH(flag_var) ENDIF() ELSE() SET(C_FLAGS ${C_FLAGS} -Winline -Wall) CHECK_C_COMPILER_FLAG(-Wextra HAVE_W_EXTRA) IF(HAVE_W_EXTRA) SET(C_FLAGS ${C_FLAGS} -Wextra) ENDIF() IF(ALSOFT_WERROR) SET(C_FLAGS ${C_FLAGS} -Werror) ENDIF() # We want RelWithDebInfo to actually include debug stuff (define _DEBUG # instead of NDEBUG) FOREACH(flag_var CMAKE_C_FLAGS_RELWITHDEBINFO CMAKE_CXX_FLAGS_RELWITHDEBINFO) IF(${flag_var} MATCHES "-DNDEBUG") STRING(REGEX REPLACE "-DNDEBUG" "-D_DEBUG" ${flag_var} "${${flag_var}}") ENDIF() ENDFOREACH() CHECK_C_COMPILER_FLAG(-fno-math-errno HAVE_FNO_MATH_ERRNO) IF(HAVE_FNO_MATH_ERRNO) SET(C_FLAGS ${C_FLAGS} -fno-math-errno) ENDIF() CHECK_C_SOURCE_COMPILES("int foo() __attribute__((destructor)); int main() {return 0;}" HAVE_GCC_DESTRUCTOR) option(ALSOFT_STATIC_LIBGCC "Force -static-libgcc for static GCC runtimes" OFF) if(ALSOFT_STATIC_LIBGCC) set(OLD_REQUIRED_LIBRARIES ${CMAKE_REQUIRED_LIBRARIES}) set(CMAKE_REQUIRED_LIBRARIES ${CMAKE_REQUIRED_LIBRARIES} -static-libgcc) check_c_source_compiles( "#include int main() { return 0; }" HAVE_STATIC_LIBGCC_SWITCH ) if(HAVE_STATIC_LIBGCC_SWITCH) SET(LINKER_FLAGS ${LINKER_FLAGS} -static-libgcc) endif() set(CMAKE_REQUIRED_LIBRARIES ${OLD_REQUIRED_LIBRARIES}) unset(OLD_REQUIRED_LIBRARIES) endif() ENDIF() # Set visibility/export options if available IF(WIN32) SET(EXPORT_DECL "__declspec(dllexport)") IF(NOT MINGW) SET(ALIGN_DECL "__declspec(align(x))") ELSE() SET(ALIGN_DECL "__declspec(aligned(x))") ENDIF() ELSE() SET(OLD_REQUIRED_FLAGS "${CMAKE_REQUIRED_FLAGS}") # Yes GCC, really don't accept visibility modes you don't support SET(CMAKE_REQUIRED_FLAGS "${OLD_REQUIRED_FLAGS} -Wattributes -Werror") CHECK_C_SOURCE_COMPILES("int foo() __attribute__((visibility(\"protected\"))); int main() {return 0;}" HAVE_GCC_PROTECTED_VISIBILITY) IF(HAVE_GCC_PROTECTED_VISIBILITY) SET(EXPORT_DECL "__attribute__((visibility(\"protected\")))") ELSE() CHECK_C_SOURCE_COMPILES("int foo() __attribute__((visibility(\"default\"))); int main() {return 0;}" HAVE_GCC_DEFAULT_VISIBILITY) IF(HAVE_GCC_DEFAULT_VISIBILITY) SET(EXPORT_DECL "__attribute__((visibility(\"default\")))") ENDIF() ENDIF() IF(HAVE_GCC_PROTECTED_VISIBILITY OR HAVE_GCC_DEFAULT_VISIBILITY) CHECK_C_COMPILER_FLAG(-fvisibility=hidden HAVE_VISIBILITY_HIDDEN_SWITCH) IF(HAVE_VISIBILITY_HIDDEN_SWITCH) SET(C_FLAGS ${C_FLAGS} -fvisibility=hidden) ENDIF() ENDIF() CHECK_C_SOURCE_COMPILES("int foo __attribute__((aligned(16))); int main() {return 0;}" HAVE_ATTRIBUTE_ALIGNED) IF(HAVE_ATTRIBUTE_ALIGNED) SET(ALIGN_DECL "__attribute__((aligned(x)))") ENDIF() SET(CMAKE_REQUIRED_FLAGS "${OLD_REQUIRED_FLAGS}") ENDIF() CHECK_C_SOURCE_COMPILES(" int main() { float *ptr; ptr = __builtin_assume_aligned(ptr, 16); return 0; }" HAVE___BUILTIN_ASSUME_ALIGNED) IF(HAVE___BUILTIN_ASSUME_ALIGNED) SET(ASSUME_ALIGNED_DECL "__builtin_assume_aligned(x, y)") ELSE() SET(ASSUME_ALIGNED_DECL "x") ENDIF() SET(SSE_SWITCH "") SET(SSE2_SWITCH "") SET(SSE3_SWITCH "") SET(SSE4_1_SWITCH "") SET(FPU_NEON_SWITCH "") CHECK_C_COMPILER_FLAG(-msse HAVE_MSSE_SWITCH) IF(HAVE_MSSE_SWITCH) SET(SSE_SWITCH "-msse") ENDIF() CHECK_C_COMPILER_FLAG(-msse2 HAVE_MSSE2_SWITCH) IF(HAVE_MSSE2_SWITCH) SET(SSE2_SWITCH "-msse2") ENDIF() CHECK_C_COMPILER_FLAG(-msse3 HAVE_MSSE3_SWITCH) IF(HAVE_MSSE3_SWITCH) SET(SSE3_SWITCH "-msse3") ENDIF() CHECK_C_COMPILER_FLAG(-msse4.1 HAVE_MSSE4_1_SWITCH) IF(HAVE_MSSE4_1_SWITCH) SET(SSE4_1_SWITCH "-msse4.1") ENDIF() CHECK_C_COMPILER_FLAG(-mfpu=neon HAVE_MFPU_NEON_SWITCH) IF(HAVE_MFPU_NEON_SWITCH) SET(FPU_NEON_SWITCH "-mfpu=neon") ENDIF() SET(FPMATH_SET "0") IF(CMAKE_SIZEOF_VOID_P MATCHES "4") IF(SSE_SWITCH OR MSVC) OPTION(ALSOFT_ENABLE_SSE_CODEGEN "Enable SSE code generation instead of x87 for 32-bit targets." TRUE) ENDIF() IF(SSE2_SWITCH OR MSVC) OPTION(ALSOFT_ENABLE_SSE2_CODEGEN "Enable SSE2 code generation instead of x87 for 32-bit targets." TRUE) ENDIF() IF(ALSOFT_ENABLE_SSE2_CODEGEN) IF(SSE2_SWITCH) CHECK_C_COMPILER_FLAG("${SSE2_SWITCH} -mfpmath=sse" HAVE_MFPMATH_SSE_2) IF(HAVE_MFPMATH_SSE_2) SET(C_FLAGS ${C_FLAGS} ${SSE2_SWITCH} -mfpmath=sse) SET(FPMATH_SET 2) ENDIF() ELSEIF(MSVC) CHECK_C_COMPILER_FLAG("/arch:SSE2" HAVE_ARCH_SSE2) IF(HAVE_ARCH_SSE2) SET(C_FLAGS ${C_FLAGS} "/arch:SSE2") SET(FPMATH_SET 2) ENDIF() ENDIF() ENDIF() IF(ALSOFT_ENABLE_SSE_CODEGEN AND NOT FPMATH_SET) IF(SSE_SWITCH) CHECK_C_COMPILER_FLAG("${SSE_SWITCH} -mfpmath=sse" HAVE_MFPMATH_SSE) IF(HAVE_MFPMATH_SSE) SET(C_FLAGS ${C_FLAGS} ${SSE_SWITCH} -mfpmath=sse) SET(FPMATH_SET 1) ENDIF() ELSEIF(MSVC) CHECK_C_COMPILER_FLAG("/arch:SSE" HAVE_ARCH_SSE) IF(HAVE_ARCH_SSE) SET(C_FLAGS ${C_FLAGS} "/arch:SSE") SET(FPMATH_SET 1) ENDIF() ENDIF() ENDIF() ENDIF() CHECK_C_SOURCE_COMPILES("int foo(const char *str, ...) __attribute__((format(printf, 1, 2))); int main() {return 0;}" HAVE_GCC_FORMAT) CHECK_INCLUDE_FILE(stdbool.h HAVE_STDBOOL_H) CHECK_INCLUDE_FILE(stdalign.h HAVE_STDALIGN_H) CHECK_INCLUDE_FILE(malloc.h HAVE_MALLOC_H) CHECK_INCLUDE_FILE(dirent.h HAVE_DIRENT_H) CHECK_INCLUDE_FILE(strings.h HAVE_STRINGS_H) CHECK_INCLUDE_FILE(cpuid.h HAVE_CPUID_H) CHECK_INCLUDE_FILE(intrin.h HAVE_INTRIN_H) CHECK_INCLUDE_FILE(sys/sysconf.h HAVE_SYS_SYSCONF_H) CHECK_INCLUDE_FILE(fenv.h HAVE_FENV_H) CHECK_INCLUDE_FILE(float.h HAVE_FLOAT_H) CHECK_INCLUDE_FILE(ieeefp.h HAVE_IEEEFP_H) CHECK_INCLUDE_FILE(guiddef.h HAVE_GUIDDEF_H) IF(NOT HAVE_GUIDDEF_H) CHECK_INCLUDE_FILE(initguid.h HAVE_INITGUID_H) ENDIF() # Some systems need libm for some of the following math functions to work SET(MATH_LIB ) CHECK_LIBRARY_EXISTS(m pow "" HAVE_LIBM) IF(HAVE_LIBM) SET(MATH_LIB ${MATH_LIB} m) SET(CMAKE_REQUIRED_LIBRARIES ${CMAKE_REQUIRED_LIBRARIES} m) ENDIF() # Check for the dlopen API (for dynamicly loading backend libs) IF(ALSOFT_DLOPEN) CHECK_LIBRARY_EXISTS(dl dlopen "" HAVE_LIBDL) IF(HAVE_LIBDL) SET(EXTRA_LIBS dl ${EXTRA_LIBS}) SET(CMAKE_REQUIRED_LIBRARIES ${CMAKE_REQUIRED_LIBRARIES} dl) ENDIF() CHECK_INCLUDE_FILE(dlfcn.h HAVE_DLFCN_H) ENDIF() # Check for a cpuid intrinsic IF(HAVE_CPUID_H) CHECK_C_SOURCE_COMPILES("#include int main() { unsigned int eax, ebx, ecx, edx; return __get_cpuid(0, &eax, &ebx, &ecx, &edx); }" HAVE_GCC_GET_CPUID) ENDIF() IF(HAVE_INTRIN_H) CHECK_C_SOURCE_COMPILES("#include int main() { int regs[4]; __cpuid(regs, 0); return regs[0]; }" HAVE_CPUID_INTRINSIC) CHECK_C_SOURCE_COMPILES("#include int main() { unsigned long idx = 0; _BitScanForward64(&idx, 1); return idx; }" HAVE_BITSCANFORWARD64_INTRINSIC) CHECK_C_SOURCE_COMPILES("#include int main() { unsigned long idx = 0; _BitScanForward(&idx, 1); return idx; }" HAVE_BITSCANFORWARD_INTRINSIC) ENDIF() CHECK_SYMBOL_EXISTS(sysconf unistd.h HAVE_SYSCONF) CHECK_SYMBOL_EXISTS(aligned_alloc stdlib.h HAVE_ALIGNED_ALLOC) CHECK_SYMBOL_EXISTS(posix_memalign stdlib.h HAVE_POSIX_MEMALIGN) CHECK_SYMBOL_EXISTS(_aligned_malloc malloc.h HAVE__ALIGNED_MALLOC) CHECK_SYMBOL_EXISTS(proc_pidpath libproc.h HAVE_PROC_PIDPATH) CHECK_SYMBOL_EXISTS(lrintf math.h HAVE_LRINTF) CHECK_SYMBOL_EXISTS(modff math.h HAVE_MODFF) CHECK_SYMBOL_EXISTS(log2f math.h HAVE_LOG2F) CHECK_SYMBOL_EXISTS(cbrtf math.h HAVE_CBRTF) CHECK_SYMBOL_EXISTS(copysignf math.h HAVE_COPYSIGNF) IF(HAVE_FLOAT_H) CHECK_SYMBOL_EXISTS(_controlfp float.h HAVE__CONTROLFP) CHECK_SYMBOL_EXISTS(__control87_2 float.h HAVE___CONTROL87_2) ENDIF() CHECK_FUNCTION_EXISTS(stat HAVE_STAT) CHECK_FUNCTION_EXISTS(strtof HAVE_STRTOF) CHECK_FUNCTION_EXISTS(strcasecmp HAVE_STRCASECMP) IF(NOT HAVE_STRCASECMP) CHECK_FUNCTION_EXISTS(_stricmp HAVE__STRICMP) IF(NOT HAVE__STRICMP) MESSAGE(FATAL_ERROR "No case-insensitive compare function found, please report!") ENDIF() SET(CPP_DEFS ${CPP_DEFS} strcasecmp=_stricmp) ENDIF() CHECK_FUNCTION_EXISTS(strncasecmp HAVE_STRNCASECMP) IF(NOT HAVE_STRNCASECMP) CHECK_FUNCTION_EXISTS(_strnicmp HAVE__STRNICMP) IF(NOT HAVE__STRNICMP) MESSAGE(FATAL_ERROR "No case-insensitive size-limitted compare function found, please report!") ENDIF() SET(CPP_DEFS ${CPP_DEFS} strncasecmp=_strnicmp) ENDIF() CHECK_SYMBOL_EXISTS(strnlen string.h HAVE_STRNLEN) CHECK_SYMBOL_EXISTS(snprintf stdio.h HAVE_SNPRINTF) IF(NOT HAVE_SNPRINTF) CHECK_FUNCTION_EXISTS(_snprintf HAVE__SNPRINTF) IF(NOT HAVE__SNPRINTF) MESSAGE(FATAL_ERROR "No snprintf function found, please report!") ENDIF() SET(CPP_DEFS ${CPP_DEFS} snprintf=_snprintf) ENDIF() CHECK_SYMBOL_EXISTS(isfinite math.h HAVE_ISFINITE) IF(NOT HAVE_ISFINITE) CHECK_FUNCTION_EXISTS(finite HAVE_FINITE) IF(NOT HAVE_FINITE) CHECK_FUNCTION_EXISTS(_finite HAVE__FINITE) IF(NOT HAVE__FINITE) MESSAGE(FATAL_ERROR "No isfinite function found, please report!") ENDIF() SET(CPP_DEFS ${CPP_DEFS} isfinite=_finite) ELSE() SET(CPP_DEFS ${CPP_DEFS} isfinite=finite) ENDIF() ENDIF() CHECK_SYMBOL_EXISTS(isnan math.h HAVE_ISNAN) IF(NOT HAVE_ISNAN) CHECK_FUNCTION_EXISTS(_isnan HAVE__ISNAN) IF(NOT HAVE__ISNAN) MESSAGE(FATAL_ERROR "No isnan function found, please report!") ENDIF() SET(CPP_DEFS ${CPP_DEFS} isnan=_isnan) ENDIF() # Check if we have Windows headers SET(OLD_REQUIRED_DEFINITIONS ${CMAKE_REQUIRED_DEFINITIONS}) SET(CMAKE_REQUIRED_DEFINITIONS ${CMAKE_REQUIRED_DEFINITIONS} -D_WIN32_WINNT=0x0502) CHECK_INCLUDE_FILE(windows.h HAVE_WINDOWS_H) SET(CMAKE_REQUIRED_DEFINITIONS ${OLD_REQUIRED_DEFINITIONS}) UNSET(OLD_REQUIRED_DEFINITIONS) IF(NOT HAVE_WINDOWS_H) CHECK_SYMBOL_EXISTS(gettimeofday sys/time.h HAVE_GETTIMEOFDAY) IF(NOT HAVE_GETTIMEOFDAY) MESSAGE(FATAL_ERROR "No timing function found!") ENDIF() CHECK_SYMBOL_EXISTS(nanosleep time.h HAVE_NANOSLEEP) IF(NOT HAVE_NANOSLEEP) MESSAGE(FATAL_ERROR "No sleep function found!") ENDIF() # We need pthreads outside of Windows CHECK_INCLUDE_FILE(pthread.h HAVE_PTHREAD_H) IF(NOT HAVE_PTHREAD_H) MESSAGE(FATAL_ERROR "PThreads is required for non-Windows builds!") ENDIF() # Some systems need pthread_np.h to get recursive mutexes CHECK_INCLUDE_FILES("pthread.h;pthread_np.h" HAVE_PTHREAD_NP_H) CHECK_C_COMPILER_FLAG(-pthread HAVE_PTHREAD) IF(HAVE_PTHREAD) SET(CMAKE_REQUIRED_FLAGS "${CMAKE_REQUIRED_FLAGS} -pthread") SET(C_FLAGS ${C_FLAGS} -pthread) SET(LINKER_FLAGS ${LINKER_FLAGS} -pthread) ENDIF() CHECK_LIBRARY_EXISTS(pthread pthread_create "" HAVE_LIBPTHREAD) IF(HAVE_LIBPTHREAD) SET(EXTRA_LIBS pthread ${EXTRA_LIBS}) ENDIF() CHECK_SYMBOL_EXISTS(pthread_setschedparam pthread.h HAVE_PTHREAD_SETSCHEDPARAM) IF(HAVE_PTHREAD_NP_H) CHECK_SYMBOL_EXISTS(pthread_setname_np "pthread.h;pthread_np.h" HAVE_PTHREAD_SETNAME_NP) IF(NOT HAVE_PTHREAD_SETNAME_NP) CHECK_SYMBOL_EXISTS(pthread_set_name_np "pthread.h;pthread_np.h" HAVE_PTHREAD_SET_NAME_NP) ELSE() CHECK_C_SOURCE_COMPILES(" #include #include int main() { pthread_setname_np(\"testname\"); return 0; }" PTHREAD_SETNAME_NP_ONE_PARAM ) CHECK_C_SOURCE_COMPILES(" #include #include int main() { pthread_setname_np(pthread_self(), \"%s\", \"testname\"); return 0; }" PTHREAD_SETNAME_NP_THREE_PARAMS ) ENDIF() CHECK_SYMBOL_EXISTS(pthread_mutexattr_setkind_np "pthread.h;pthread_np.h" HAVE_PTHREAD_MUTEXATTR_SETKIND_NP) ELSE() CHECK_SYMBOL_EXISTS(pthread_setname_np pthread.h HAVE_PTHREAD_SETNAME_NP) IF(NOT HAVE_PTHREAD_SETNAME_NP) CHECK_SYMBOL_EXISTS(pthread_set_name_np pthread.h HAVE_PTHREAD_SET_NAME_NP) ELSE() CHECK_C_SOURCE_COMPILES(" #include int main() { pthread_setname_np(\"testname\"); return 0; }" PTHREAD_SETNAME_NP_ONE_PARAM ) CHECK_C_SOURCE_COMPILES(" #include int main() { pthread_setname_np(pthread_self(), \"%s\", \"testname\"); return 0; }" PTHREAD_SETNAME_NP_THREE_PARAMS ) ENDIF() CHECK_SYMBOL_EXISTS(pthread_mutexattr_setkind_np pthread.h HAVE_PTHREAD_MUTEXATTR_SETKIND_NP) ENDIF() CHECK_SYMBOL_EXISTS(pthread_mutex_timedlock pthread.h HAVE_PTHREAD_MUTEX_TIMEDLOCK) CHECK_LIBRARY_EXISTS(rt clock_gettime "" HAVE_LIBRT) IF(HAVE_LIBRT) SET(EXTRA_LIBS rt ${EXTRA_LIBS}) ENDIF() ENDIF() CHECK_SYMBOL_EXISTS(getopt unistd.h HAVE_GETOPT) # Check for a 64-bit type CHECK_INCLUDE_FILE(stdint.h HAVE_STDINT_H) IF(NOT HAVE_STDINT_H) IF(HAVE_WINDOWS_H) CHECK_C_SOURCE_COMPILES("#define _WIN32_WINNT 0x0502 #include __int64 foo; int main() {return 0;}" HAVE___INT64) ENDIF() IF(NOT HAVE___INT64) IF(NOT SIZEOF_LONG MATCHES "8") IF(NOT SIZEOF_LONG_LONG MATCHES "8") MESSAGE(FATAL_ERROR "No 64-bit types found, please report!") ENDIF() ENDIF() ENDIF() ENDIF() SET(COMMON_OBJS common/alcomplex.c common/alcomplex.h common/align.h common/almalloc.c common/almalloc.h common/atomic.c common/atomic.h common/bool.h common/math_defs.h common/rwlock.c common/rwlock.h common/static_assert.h common/threads.c common/threads.h common/uintmap.c common/uintmap.h ) SET(OPENAL_OBJS OpenAL32/Include/bs2b.h OpenAL32/Include/alMain.h OpenAL32/Include/alu.h OpenAL32/Include/alAuxEffectSlot.h OpenAL32/alAuxEffectSlot.c OpenAL32/Include/alBuffer.h OpenAL32/alBuffer.c OpenAL32/Include/alEffect.h OpenAL32/alEffect.c OpenAL32/Include/alError.h OpenAL32/alError.c OpenAL32/alExtension.c OpenAL32/Include/alFilter.h OpenAL32/alFilter.c OpenAL32/Include/alListener.h OpenAL32/alListener.c OpenAL32/Include/alSource.h OpenAL32/alSource.c OpenAL32/alState.c OpenAL32/event.c OpenAL32/Include/sample_cvt.h OpenAL32/sample_cvt.c ) SET(ALC_OBJS Alc/ALc.c Alc/ALu.c Alc/alconfig.c Alc/alconfig.h Alc/bs2b.c Alc/converter.c Alc/converter.h Alc/inprogext.h Alc/mastering.c Alc/mastering.h Alc/ringbuffer.c Alc/ringbuffer.h Alc/effects/autowah.c Alc/effects/chorus.c Alc/effects/compressor.c Alc/effects/dedicated.c Alc/effects/distortion.c Alc/effects/echo.c Alc/effects/equalizer.c Alc/effects/fshifter.c Alc/effects/modulator.c Alc/effects/null.c Alc/effects/pshifter.c Alc/effects/reverb.c Alc/filters/defs.h Alc/filters/filter.c Alc/filters/nfc.c Alc/filters/nfc.h Alc/filters/splitter.c Alc/filters/splitter.h Alc/helpers.c Alc/alstring.h Alc/compat.h Alc/cpu_caps.h Alc/fpu_modes.h Alc/logging.h Alc/vector.h Alc/hrtf.c Alc/hrtf.h Alc/uhjfilter.c Alc/uhjfilter.h Alc/ambdec.c Alc/ambdec.h Alc/bformatdec.c Alc/bformatdec.h Alc/panning.c Alc/polymorphism.h Alc/mixvoice.c Alc/mixer/defs.h Alc/mixer/mixer_c.c ) SET(CPU_EXTS "Default") SET(HAVE_SSE 0) SET(HAVE_SSE2 0) SET(HAVE_SSE3 0) SET(HAVE_SSE4_1 0) SET(HAVE_NEON 0) SET(HAVE_ALSA 0) SET(HAVE_OSS 0) SET(HAVE_SOLARIS 0) SET(HAVE_SNDIO 0) SET(HAVE_QSA 0) SET(HAVE_DSOUND 0) SET(HAVE_WASAPI 0) SET(HAVE_WINMM 0) SET(HAVE_PORTAUDIO 0) SET(HAVE_PULSEAUDIO 0) SET(HAVE_COREAUDIO 0) SET(HAVE_OPENSL 0) SET(HAVE_WAVE 0) SET(HAVE_SDL2 0) # Check for SSE support OPTION(ALSOFT_REQUIRE_SSE "Require SSE support" OFF) CHECK_INCLUDE_FILE(xmmintrin.h HAVE_XMMINTRIN_H "${SSE_SWITCH}") IF(HAVE_XMMINTRIN_H) OPTION(ALSOFT_CPUEXT_SSE "Enable SSE support" ON) IF(ALSOFT_CPUEXT_SSE) IF(ALIGN_DECL OR HAVE_C11_ALIGNAS) SET(HAVE_SSE 1) SET(ALC_OBJS ${ALC_OBJS} Alc/mixer/mixer_sse.c) IF(SSE_SWITCH) SET_SOURCE_FILES_PROPERTIES(Alc/mixer/mixer_sse.c PROPERTIES COMPILE_FLAGS "${SSE_SWITCH}") ENDIF() SET(CPU_EXTS "${CPU_EXTS}, SSE") ENDIF() ENDIF() ENDIF() IF(ALSOFT_REQUIRE_SSE AND NOT HAVE_SSE) MESSAGE(FATAL_ERROR "Failed to enabled required SSE CPU extensions") ENDIF() OPTION(ALSOFT_REQUIRE_SSE2 "Require SSE2 support" OFF) CHECK_INCLUDE_FILE(emmintrin.h HAVE_EMMINTRIN_H "${SSE2_SWITCH}") IF(HAVE_EMMINTRIN_H) OPTION(ALSOFT_CPUEXT_SSE2 "Enable SSE2 support" ON) IF(HAVE_SSE AND ALSOFT_CPUEXT_SSE2) IF(ALIGN_DECL OR HAVE_C11_ALIGNAS) SET(HAVE_SSE2 1) SET(ALC_OBJS ${ALC_OBJS} Alc/mixer/mixer_sse2.c) IF(SSE2_SWITCH) SET_SOURCE_FILES_PROPERTIES(Alc/mixer/mixer_sse2.c PROPERTIES COMPILE_FLAGS "${SSE2_SWITCH}") ENDIF() SET(CPU_EXTS "${CPU_EXTS}, SSE2") ENDIF() ENDIF() ENDIF() IF(ALSOFT_REQUIRE_SSE2 AND NOT HAVE_SSE2) MESSAGE(FATAL_ERROR "Failed to enable required SSE2 CPU extensions") ENDIF() OPTION(ALSOFT_REQUIRE_SSE3 "Require SSE3 support" OFF) CHECK_INCLUDE_FILE(pmmintrin.h HAVE_PMMINTRIN_H "${SSE3_SWITCH}") IF(HAVE_EMMINTRIN_H) OPTION(ALSOFT_CPUEXT_SSE3 "Enable SSE3 support" ON) IF(HAVE_SSE2 AND ALSOFT_CPUEXT_SSE3) IF(ALIGN_DECL OR HAVE_C11_ALIGNAS) SET(HAVE_SSE3 1) SET(ALC_OBJS ${ALC_OBJS} Alc/mixer/mixer_sse3.c) IF(SSE2_SWITCH) SET_SOURCE_FILES_PROPERTIES(Alc/mixer/mixer_sse3.c PROPERTIES COMPILE_FLAGS "${SSE3_SWITCH}") ENDIF() SET(CPU_EXTS "${CPU_EXTS}, SSE3") ENDIF() ENDIF() ENDIF() IF(ALSOFT_REQUIRE_SSE3 AND NOT HAVE_SSE3) MESSAGE(FATAL_ERROR "Failed to enable required SSE3 CPU extensions") ENDIF() OPTION(ALSOFT_REQUIRE_SSE4_1 "Require SSE4.1 support" OFF) CHECK_INCLUDE_FILE(smmintrin.h HAVE_SMMINTRIN_H "${SSE4_1_SWITCH}") IF(HAVE_SMMINTRIN_H) OPTION(ALSOFT_CPUEXT_SSE4_1 "Enable SSE4.1 support" ON) IF(HAVE_SSE2 AND ALSOFT_CPUEXT_SSE4_1) IF(ALIGN_DECL OR HAVE_C11_ALIGNAS) SET(HAVE_SSE4_1 1) SET(ALC_OBJS ${ALC_OBJS} Alc/mixer/mixer_sse41.c) IF(SSE4_1_SWITCH) SET_SOURCE_FILES_PROPERTIES(Alc/mixer/mixer_sse41.c PROPERTIES COMPILE_FLAGS "${SSE4_1_SWITCH}") ENDIF() SET(CPU_EXTS "${CPU_EXTS}, SSE4.1") ENDIF() ENDIF() ENDIF() IF(ALSOFT_REQUIRE_SSE4_1 AND NOT HAVE_SSE4_1) MESSAGE(FATAL_ERROR "Failed to enable required SSE4.1 CPU extensions") ENDIF() # Check for ARM Neon support OPTION(ALSOFT_REQUIRE_NEON "Require ARM Neon support" OFF) CHECK_INCLUDE_FILE(arm_neon.h HAVE_ARM_NEON_H ${FPU_NEON_SWITCH}) IF(HAVE_ARM_NEON_H) OPTION(ALSOFT_CPUEXT_NEON "Enable ARM Neon support" ON) IF(ALSOFT_CPUEXT_NEON) SET(HAVE_NEON 1) SET(ALC_OBJS ${ALC_OBJS} Alc/mixer/mixer_neon.c) IF(FPU_NEON_SWITCH) SET_SOURCE_FILES_PROPERTIES(Alc/mixer/mixer_neon.c PROPERTIES COMPILE_FLAGS "${FPU_NEON_SWITCH}") ENDIF() SET(CPU_EXTS "${CPU_EXTS}, Neon") ENDIF() ENDIF() IF(ALSOFT_REQUIRE_NEON AND NOT HAVE_NEON) MESSAGE(FATAL_ERROR "Failed to enabled required ARM Neon CPU extensions") ENDIF() IF(WIN32 OR HAVE_DLFCN_H) SET(IS_LINKED "") MACRO(ADD_BACKEND_LIBS _LIBS) ENDMACRO() ELSE() SET(IS_LINKED " (linked)") MACRO(ADD_BACKEND_LIBS _LIBS) SET(EXTRA_LIBS ${_LIBS} ${EXTRA_LIBS}) ENDMACRO() ENDIF() SET(BACKENDS "") SET(ALC_OBJS ${ALC_OBJS} Alc/backends/base.c Alc/backends/base.h # Default backends, always available Alc/backends/loopback.c Alc/backends/null.c ) # Check ALSA backend OPTION(ALSOFT_REQUIRE_ALSA "Require ALSA backend" OFF) FIND_PACKAGE(ALSA) IF(ALSA_FOUND) OPTION(ALSOFT_BACKEND_ALSA "Enable ALSA backend" ON) IF(ALSOFT_BACKEND_ALSA) SET(HAVE_ALSA 1) SET(BACKENDS "${BACKENDS} ALSA${IS_LINKED},") SET(ALC_OBJS ${ALC_OBJS} Alc/backends/alsa.c) ADD_BACKEND_LIBS(${ALSA_LIBRARIES}) SET(INC_PATHS ${INC_PATHS} ${ALSA_INCLUDE_DIRS}) ENDIF() ENDIF() IF(ALSOFT_REQUIRE_ALSA AND NOT HAVE_ALSA) MESSAGE(FATAL_ERROR "Failed to enabled required ALSA backend") ENDIF() # Check OSS backend OPTION(ALSOFT_REQUIRE_OSS "Require OSS backend" OFF) FIND_PACKAGE(OSS) IF(OSS_FOUND) OPTION(ALSOFT_BACKEND_OSS "Enable OSS backend" ON) IF(ALSOFT_BACKEND_OSS) SET(HAVE_OSS 1) SET(BACKENDS "${BACKENDS} OSS,") SET(ALC_OBJS ${ALC_OBJS} Alc/backends/oss.c) IF(OSS_LIBRARIES) SET(EXTRA_LIBS ${OSS_LIBRARIES} ${EXTRA_LIBS}) ENDIF() SET(INC_PATHS ${INC_PATHS} ${OSS_INCLUDE_DIRS}) ENDIF() ENDIF() IF(ALSOFT_REQUIRE_OSS AND NOT HAVE_OSS) MESSAGE(FATAL_ERROR "Failed to enabled required OSS backend") ENDIF() # Check Solaris backend OPTION(ALSOFT_REQUIRE_SOLARIS "Require Solaris backend" OFF) FIND_PACKAGE(AudioIO) IF(AUDIOIO_FOUND) OPTION(ALSOFT_BACKEND_SOLARIS "Enable Solaris backend" ON) IF(ALSOFT_BACKEND_SOLARIS) SET(HAVE_SOLARIS 1) SET(BACKENDS "${BACKENDS} Solaris,") SET(ALC_OBJS ${ALC_OBJS} Alc/backends/solaris.c) SET(INC_PATHS ${INC_PATHS} ${AUDIOIO_INCLUDE_DIRS}) ENDIF() ENDIF() IF(ALSOFT_REQUIRE_SOLARIS AND NOT HAVE_SOLARIS) MESSAGE(FATAL_ERROR "Failed to enabled required Solaris backend") ENDIF() # Check SndIO backend OPTION(ALSOFT_REQUIRE_SNDIO "Require SndIO backend" OFF) FIND_PACKAGE(SoundIO) IF(SOUNDIO_FOUND) OPTION(ALSOFT_BACKEND_SNDIO "Enable SndIO backend" ON) IF(ALSOFT_BACKEND_SNDIO) SET(HAVE_SNDIO 1) SET(BACKENDS "${BACKENDS} SndIO (linked),") SET(ALC_OBJS ${ALC_OBJS} Alc/backends/sndio.c) SET(EXTRA_LIBS ${SOUNDIO_LIBRARIES} ${EXTRA_LIBS}) SET(INC_PATHS ${INC_PATHS} ${SOUNDIO_INCLUDE_DIRS}) ENDIF() ENDIF() IF(ALSOFT_REQUIRE_SNDIO AND NOT HAVE_SNDIO) MESSAGE(FATAL_ERROR "Failed to enabled required SndIO backend") ENDIF() # Check QSA backend OPTION(ALSOFT_REQUIRE_QSA "Require QSA backend" OFF) FIND_PACKAGE(QSA) IF(QSA_FOUND) OPTION(ALSOFT_BACKEND_QSA "Enable QSA backend" ON) IF(ALSOFT_BACKEND_QSA) SET(HAVE_QSA 1) SET(BACKENDS "${BACKENDS} QSA (linked),") SET(ALC_OBJS ${ALC_OBJS} Alc/backends/qsa.c) SET(EXTRA_LIBS ${QSA_LIBRARIES} ${EXTRA_LIBS}) SET(INC_PATHS ${INC_PATHS} ${QSA_INCLUDE_DIRS}) ENDIF() ENDIF() IF(ALSOFT_REQUIRE_QSA AND NOT HAVE_QSA) MESSAGE(FATAL_ERROR "Failed to enabled required QSA backend") ENDIF() # Check Windows-only backends OPTION(ALSOFT_REQUIRE_WINMM "Require Windows Multimedia backend" OFF) OPTION(ALSOFT_REQUIRE_DSOUND "Require DirectSound backend" OFF) OPTION(ALSOFT_REQUIRE_WASAPI "Require WASAPI backend" OFF) IF(HAVE_WINDOWS_H) SET(OLD_REQUIRED_DEFINITIONS ${CMAKE_REQUIRED_DEFINITIONS}) SET(CMAKE_REQUIRED_DEFINITIONS ${CMAKE_REQUIRED_DEFINITIONS} -D_WIN32_WINNT=0x0502) # Check MMSystem backend CHECK_INCLUDE_FILES("windows.h;mmsystem.h" HAVE_MMSYSTEM_H) IF(HAVE_MMSYSTEM_H) CHECK_SHARED_FUNCTION_EXISTS(waveOutOpen "windows.h;mmsystem.h" winmm "" HAVE_LIBWINMM) IF(HAVE_LIBWINMM) OPTION(ALSOFT_BACKEND_WINMM "Enable Windows Multimedia backend" ON) IF(ALSOFT_BACKEND_WINMM) SET(HAVE_WINMM 1) SET(BACKENDS "${BACKENDS} WinMM,") SET(ALC_OBJS ${ALC_OBJS} Alc/backends/winmm.c) SET(EXTRA_LIBS winmm ${EXTRA_LIBS}) ENDIF() ENDIF() ENDIF() # Check DSound backend FIND_PACKAGE(DSound) IF(DSOUND_FOUND) OPTION(ALSOFT_BACKEND_DSOUND "Enable DirectSound backend" ON) IF(ALSOFT_BACKEND_DSOUND) SET(HAVE_DSOUND 1) SET(BACKENDS "${BACKENDS} DirectSound${IS_LINKED},") SET(ALC_OBJS ${ALC_OBJS} Alc/backends/dsound.c) ADD_BACKEND_LIBS(${DSOUND_LIBRARIES}) SET(INC_PATHS ${INC_PATHS} ${DSOUND_INCLUDE_DIRS}) ENDIF() ENDIF() # Check for WASAPI backend CHECK_INCLUDE_FILE(mmdeviceapi.h HAVE_MMDEVICEAPI_H) IF(HAVE_MMDEVICEAPI_H) OPTION(ALSOFT_BACKEND_WASAPI "Enable WASAPI backend" ON) IF(ALSOFT_BACKEND_WASAPI) SET(HAVE_WASAPI 1) SET(BACKENDS "${BACKENDS} WASAPI,") SET(ALC_OBJS ${ALC_OBJS} Alc/backends/wasapi.c) ENDIF() ENDIF() SET(CMAKE_REQUIRED_DEFINITIONS ${OLD_REQUIRED_DEFINITIONS}) UNSET(OLD_REQUIRED_DEFINITIONS) ENDIF() IF(ALSOFT_REQUIRE_WINMM AND NOT HAVE_WINMM) MESSAGE(FATAL_ERROR "Failed to enabled required WinMM backend") ENDIF() IF(ALSOFT_REQUIRE_DSOUND AND NOT HAVE_DSOUND) MESSAGE(FATAL_ERROR "Failed to enabled required DSound backend") ENDIF() IF(ALSOFT_REQUIRE_WASAPI AND NOT HAVE_WASAPI) MESSAGE(FATAL_ERROR "Failed to enabled required WASAPI backend") ENDIF() # Check PortAudio backend OPTION(ALSOFT_REQUIRE_PORTAUDIO "Require PortAudio backend" OFF) FIND_PACKAGE(PortAudio) IF(PORTAUDIO_FOUND) OPTION(ALSOFT_BACKEND_PORTAUDIO "Enable PortAudio backend" ON) IF(ALSOFT_BACKEND_PORTAUDIO) SET(HAVE_PORTAUDIO 1) SET(BACKENDS "${BACKENDS} PortAudio${IS_LINKED},") SET(ALC_OBJS ${ALC_OBJS} Alc/backends/portaudio.c) ADD_BACKEND_LIBS(${PORTAUDIO_LIBRARIES}) SET(INC_PATHS ${INC_PATHS} ${PORTAUDIO_INCLUDE_DIRS}) ENDIF() ENDIF() IF(ALSOFT_REQUIRE_PORTAUDIO AND NOT HAVE_PORTAUDIO) MESSAGE(FATAL_ERROR "Failed to enabled required PortAudio backend") ENDIF() # Check PulseAudio backend OPTION(ALSOFT_REQUIRE_PULSEAUDIO "Require PulseAudio backend" OFF) FIND_PACKAGE(PulseAudio) IF(PULSEAUDIO_FOUND) OPTION(ALSOFT_BACKEND_PULSEAUDIO "Enable PulseAudio backend" ON) IF(ALSOFT_BACKEND_PULSEAUDIO) SET(HAVE_PULSEAUDIO 1) SET(BACKENDS "${BACKENDS} PulseAudio${IS_LINKED},") SET(ALC_OBJS ${ALC_OBJS} Alc/backends/pulseaudio.c) ADD_BACKEND_LIBS(${PULSEAUDIO_LIBRARIES}) SET(INC_PATHS ${INC_PATHS} ${PULSEAUDIO_INCLUDE_DIRS}) ENDIF() ENDIF() IF(ALSOFT_REQUIRE_PULSEAUDIO AND NOT HAVE_PULSEAUDIO) MESSAGE(FATAL_ERROR "Failed to enabled required PulseAudio backend") ENDIF() # Check JACK backend OPTION(ALSOFT_REQUIRE_JACK "Require JACK backend" OFF) FIND_PACKAGE(JACK) IF(JACK_FOUND) OPTION(ALSOFT_BACKEND_JACK "Enable JACK backend" ON) IF(ALSOFT_BACKEND_JACK) SET(HAVE_JACK 1) SET(BACKENDS "${BACKENDS} JACK${IS_LINKED},") SET(ALC_OBJS ${ALC_OBJS} Alc/backends/jack.c) ADD_BACKEND_LIBS(${JACK_LIBRARIES}) SET(INC_PATHS ${INC_PATHS} ${JACK_INCLUDE_DIRS}) ENDIF() ENDIF() IF(ALSOFT_REQUIRE_JACK AND NOT HAVE_JACK) MESSAGE(FATAL_ERROR "Failed to enabled required JACK backend") ENDIF() # Check CoreAudio backend OPTION(ALSOFT_REQUIRE_COREAUDIO "Require CoreAudio backend" OFF) FIND_LIBRARY(COREAUDIO_FRAMEWORK NAMES CoreAudio PATHS /System/Library/Frameworks ) IF(COREAUDIO_FRAMEWORK) OPTION(ALSOFT_BACKEND_COREAUDIO "Enable CoreAudio backend" ON) IF(ALSOFT_BACKEND_COREAUDIO) SET(HAVE_COREAUDIO 1) SET(ALC_OBJS ${ALC_OBJS} Alc/backends/coreaudio.c) SET(BACKENDS "${BACKENDS} CoreAudio,") SET(EXTRA_LIBS ${COREAUDIO_FRAMEWORK} ${EXTRA_LIBS}) SET(EXTRA_LIBS /System/Library/Frameworks/AudioUnit.framework ${EXTRA_LIBS}) SET(EXTRA_LIBS /System/Library/Frameworks/ApplicationServices.framework ${EXTRA_LIBS}) # Some versions of OSX may need the AudioToolbox framework. Add it if # it's found. FIND_LIBRARY(AUDIOTOOLBOX_LIBRARY NAMES AudioToolbox PATHS ~/Library/Frameworks /Library/Frameworks /System/Library/Frameworks ) IF(AUDIOTOOLBOX_LIBRARY) SET(EXTRA_LIBS ${AUDIOTOOLBOX_LIBRARY} ${EXTRA_LIBS}) ENDIF() ENDIF() ENDIF() IF(ALSOFT_REQUIRE_COREAUDIO AND NOT HAVE_COREAUDIO) MESSAGE(FATAL_ERROR "Failed to enabled required CoreAudio backend") ENDIF() # Check for OpenSL (Android) backend OPTION(ALSOFT_REQUIRE_OPENSL "Require OpenSL backend" OFF) CHECK_INCLUDE_FILES("SLES/OpenSLES.h;SLES/OpenSLES_Android.h" HAVE_SLES_OPENSLES_ANDROID_H) IF(HAVE_SLES_OPENSLES_ANDROID_H) CHECK_SHARED_FUNCTION_EXISTS(slCreateEngine "SLES/OpenSLES.h" OpenSLES "" HAVE_LIBOPENSLES) IF(HAVE_LIBOPENSLES) OPTION(ALSOFT_BACKEND_OPENSL "Enable OpenSL backend" ON) IF(ALSOFT_BACKEND_OPENSL) SET(HAVE_OPENSL 1) SET(ALC_OBJS ${ALC_OBJS} Alc/backends/opensl.c) SET(BACKENDS "${BACKENDS} OpenSL,") SET(EXTRA_LIBS OpenSLES ${EXTRA_LIBS}) ENDIF() ENDIF() ENDIF() IF(ALSOFT_REQUIRE_OPENSL AND NOT HAVE_OPENSL) MESSAGE(FATAL_ERROR "Failed to enabled required OpenSL backend") ENDIF() # Check for SDL2 backend OPTION(ALSOFT_REQUIRE_SDL2 "Require SDL2 backend" OFF) FIND_PACKAGE(SDL2) IF(SDL2_FOUND) # Off by default, since it adds a runtime dependency OPTION(ALSOFT_BACKEND_SDL2 "Enable SDL2 backend" OFF) IF(ALSOFT_BACKEND_SDL2) SET(HAVE_SDL2 1) SET(ALC_OBJS ${ALC_OBJS} Alc/backends/sdl2.c) SET(BACKENDS "${BACKENDS} SDL2,") SET(EXTRA_LIBS ${SDL2_LIBRARY} ${EXTRA_LIBS}) SET(INC_PATHS ${INC_PATHS} ${SDL2_INCLUDE_DIR}) ENDIF() ENDIF() IF(ALSOFT_REQUIRE_SDL2 AND NOT SDL2_FOUND) MESSAGE(FATAL_ERROR "Failed to enabled required SDL2 backend") ENDIF() # Optionally enable the Wave Writer backend OPTION(ALSOFT_BACKEND_WAVE "Enable Wave Writer backend" ON) IF(ALSOFT_BACKEND_WAVE) SET(HAVE_WAVE 1) SET(ALC_OBJS ${ALC_OBJS} Alc/backends/wave.c) SET(BACKENDS "${BACKENDS} WaveFile,") ENDIF() # This is always available SET(BACKENDS "${BACKENDS} Null") FIND_PACKAGE(Git) IF(GIT_FOUND AND EXISTS "${OpenAL_SOURCE_DIR}/.git") # Get the current working branch and its latest abbreviated commit hash ADD_CUSTOM_TARGET(build_version ${CMAKE_COMMAND} -D GIT_EXECUTABLE=${GIT_EXECUTABLE} -D LIB_VERSION=${LIB_VERSION} -D SRC=${OpenAL_SOURCE_DIR}/version.h.in -D DST=${OpenAL_BINARY_DIR}/version.h -P ${OpenAL_SOURCE_DIR}/version.cmake WORKING_DIRECTORY "${OpenAL_SOURCE_DIR}" VERBATIM ) ELSE() SET(GIT_BRANCH "UNKNOWN") SET(GIT_COMMIT_HASH "unknown") CONFIGURE_FILE( "${OpenAL_SOURCE_DIR}/version.h.in" "${OpenAL_BINARY_DIR}/version.h") ENDIF() SET(NATIVE_SRC_DIR "${OpenAL_SOURCE_DIR}/native-tools/") SET(NATIVE_BIN_DIR "${OpenAL_BINARY_DIR}/native-tools/") FILE(MAKE_DIRECTORY "${NATIVE_BIN_DIR}") SET(BIN2H_COMMAND "${NATIVE_BIN_DIR}bin2h") SET(BSINCGEN_COMMAND "${NATIVE_BIN_DIR}bsincgen") ADD_CUSTOM_COMMAND(OUTPUT "${BIN2H_COMMAND}" "${BSINCGEN_COMMAND}" COMMAND ${CMAKE_COMMAND} -G "${CMAKE_GENERATOR}" "${NATIVE_SRC_DIR}" COMMAND ${CMAKE_COMMAND} -E remove "${BIN2H_COMMAND}" "${BSINCGEN_COMMAND}" COMMAND ${CMAKE_COMMAND} --build . --config "Release" WORKING_DIRECTORY "${NATIVE_BIN_DIR}" DEPENDS "${NATIVE_SRC_DIR}CMakeLists.txt" IMPLICIT_DEPENDS C "${NATIVE_SRC_DIR}bin2h.c" C "${NATIVE_SRC_DIR}bsincgen.c" VERBATIM ) ADD_CUSTOM_TARGET(native-tools DEPENDS "${BIN2H_COMMAND}" "${BSINCGEN_COMMAND}" VERBATIM ) option(ALSOFT_EMBED_HRTF_DATA "Embed the HRTF data files (increases library footprint)" ON) if(ALSOFT_EMBED_HRTF_DATA) MACRO(make_hrtf_header FILENAME VARNAME) SET(infile "${OpenAL_SOURCE_DIR}/hrtf/${FILENAME}") SET(outfile "${OpenAL_BINARY_DIR}/${FILENAME}.h") ADD_CUSTOM_COMMAND(OUTPUT "${outfile}" COMMAND "${BIN2H_COMMAND}" "${infile}" "${outfile}" ${VARNAME} DEPENDS native-tools "${infile}" VERBATIM ) SET(ALC_OBJS ${ALC_OBJS} "${outfile}") ENDMACRO() make_hrtf_header(default-44100.mhr "hrtf_default_44100") make_hrtf_header(default-48000.mhr "hrtf_default_48000") endif() ADD_CUSTOM_COMMAND(OUTPUT "${OpenAL_BINARY_DIR}/bsinc_inc.h" COMMAND "${BSINCGEN_COMMAND}" "${OpenAL_BINARY_DIR}/bsinc_inc.h" DEPENDS native-tools "${NATIVE_SRC_DIR}bsincgen.c" VERBATIM ) SET(ALC_OBJS ${ALC_OBJS} "${OpenAL_BINARY_DIR}/bsinc_inc.h") IF(ALSOFT_UTILS AND NOT ALSOFT_NO_CONFIG_UTIL) add_subdirectory(utils/alsoft-config) ENDIF() IF(ALSOFT_EXAMPLES) IF(NOT SDL2_FOUND) FIND_PACKAGE(SDL2) ENDIF() IF(SDL2_FOUND) FIND_PACKAGE(SDL_sound) FIND_PACKAGE(FFmpeg COMPONENTS AVFORMAT AVCODEC AVUTIL SWSCALE SWRESAMPLE) ENDIF() ENDIF() IF(NOT WIN32) SET(LIBNAME "openal") ELSE() SET(LIBNAME "OpenAL32") ENDIF() # Needed for openal.pc.in SET(prefix ${CMAKE_INSTALL_PREFIX}) SET(exec_prefix "\${prefix}") SET(libdir "\${exec_prefix}/${CMAKE_INSTALL_LIBDIR}") SET(bindir "\${exec_prefix}/${CMAKE_INSTALL_BINDIR}") SET(includedir "\${prefix}/${CMAKE_INSTALL_INCLUDEDIR}") SET(PACKAGE_VERSION "${LIB_VERSION}") SET(PKG_CONFIG_CFLAGS ) SET(PKG_CONFIG_PRIVATE_LIBS ) IF(LIBTYPE STREQUAL "STATIC") SET(PKG_CONFIG_CFLAGS -DAL_LIBTYPE_STATIC) FOREACH(FLAG ${LINKER_FLAGS} ${EXTRA_LIBS} ${MATH_LIB}) # If this is already a linker flag, or is a full path+file, add it # as-is. Otherwise, it's a name intended to be dressed as -lname. IF(FLAG MATCHES "^-.*" OR EXISTS "${FLAG}") SET(PKG_CONFIG_PRIVATE_LIBS "${PKG_CONFIG_PRIVATE_LIBS} ${FLAG}") ELSE() SET(PKG_CONFIG_PRIVATE_LIBS "${PKG_CONFIG_PRIVATE_LIBS} -l${FLAG}") ENDIF() ENDFOREACH() ENDIF() # End configuration CONFIGURE_FILE( "${OpenAL_SOURCE_DIR}/config.h.in" "${OpenAL_BINARY_DIR}/config.h") CONFIGURE_FILE( "${OpenAL_SOURCE_DIR}/openal.pc.in" "${OpenAL_BINARY_DIR}/openal.pc" @ONLY) # Add a static library with common functions used by multiple targets ADD_LIBRARY(common STATIC ${COMMON_OBJS}) TARGET_COMPILE_DEFINITIONS(common PRIVATE ${CPP_DEFS}) TARGET_COMPILE_OPTIONS(common PRIVATE ${C_FLAGS}) UNSET(HAS_ROUTER) SET(IMPL_TARGET OpenAL) SET(COMMON_LIB ) SET(SUBSYS_FLAG ) # Build main library IF(LIBTYPE STREQUAL "STATIC") SET(CPP_DEFS ${CPP_DEFS} AL_LIBTYPE_STATIC) IF(WIN32 AND ALSOFT_NO_UID_DEFS) SET(CPP_DEFS ${CPP_DEFS} AL_NO_UID_DEFS) ENDIF() ADD_LIBRARY(OpenAL STATIC ${COMMON_OBJS} ${OPENAL_OBJS} ${ALC_OBJS}) ELSE() # Make sure to compile the common code with PIC, since it'll be linked into # shared libs that needs it. SET_PROPERTY(TARGET common PROPERTY POSITION_INDEPENDENT_CODE TRUE) SET(COMMON_LIB common) IF(WIN32) IF(MSVC) SET(SUBSYS_FLAG ${SUBSYS_FLAG} "/SUBSYSTEM:WINDOWS") ELSEIF(CMAKE_COMPILER_IS_GNUCC) SET(SUBSYS_FLAG ${SUBSYS_FLAG} "-mwindows") ENDIF() ENDIF() IF(WIN32 AND ALSOFT_BUILD_ROUTER) ADD_LIBRARY(OpenAL SHARED router/router.c router/router.h router/alc.c router/al.c) TARGET_COMPILE_DEFINITIONS(OpenAL PRIVATE AL_BUILD_LIBRARY AL_ALEXT_PROTOTYPES ${CPP_DEFS}) TARGET_COMPILE_OPTIONS(OpenAL PRIVATE ${C_FLAGS}) TARGET_LINK_LIBRARIES(OpenAL PRIVATE ${LINKER_FLAGS} ${COMMON_LIB}) SET_TARGET_PROPERTIES(OpenAL PROPERTIES PREFIX "") SET_TARGET_PROPERTIES(OpenAL PROPERTIES OUTPUT_NAME ${LIBNAME}) IF(TARGET build_version) ADD_DEPENDENCIES(OpenAL build_version) ENDIF() SET(HAS_ROUTER 1) SET(LIBNAME "soft_oal") SET(IMPL_TARGET soft_oal) ENDIF() ADD_LIBRARY(${IMPL_TARGET} SHARED ${OPENAL_OBJS} ${ALC_OBJS}) IF(WIN32) SET_TARGET_PROPERTIES(${IMPL_TARGET} PROPERTIES PREFIX "") ENDIF() ENDIF() SET_TARGET_PROPERTIES(${IMPL_TARGET} PROPERTIES OUTPUT_NAME ${LIBNAME} VERSION ${LIB_VERSION} SOVERSION ${LIB_MAJOR_VERSION} ) TARGET_COMPILE_DEFINITIONS(${IMPL_TARGET} PRIVATE AL_BUILD_LIBRARY AL_ALEXT_PROTOTYPES ${CPP_DEFS}) TARGET_INCLUDE_DIRECTORIES(${IMPL_TARGET} PRIVATE "${OpenAL_SOURCE_DIR}/OpenAL32/Include" "${OpenAL_SOURCE_DIR}/Alc" ${INC_PATHS}) TARGET_COMPILE_OPTIONS(${IMPL_TARGET} PRIVATE ${C_FLAGS}) TARGET_LINK_LIBRARIES(${IMPL_TARGET} PRIVATE ${LINKER_FLAGS} ${COMMON_LIB} ${EXTRA_LIBS} ${MATH_LIB}) IF(TARGET build_version) ADD_DEPENDENCIES(${IMPL_TARGET} build_version) ENDIF() IF(WIN32 AND MINGW AND ALSOFT_BUILD_IMPORT_LIB AND NOT LIBTYPE STREQUAL "STATIC") FIND_PROGRAM(SED_EXECUTABLE NAMES sed DOC "sed executable") FIND_PROGRAM(DLLTOOL_EXECUTABLE NAMES "${DLLTOOL}" DOC "dlltool executable") IF(NOT SED_EXECUTABLE OR NOT DLLTOOL_EXECUTABLE) MESSAGE(STATUS "") IF(NOT SED_EXECUTABLE) MESSAGE(STATUS "WARNING: Cannot find sed, disabling .def/.lib generation") ENDIF() IF(NOT DLLTOOL_EXECUTABLE) MESSAGE(STATUS "WARNING: Cannot find dlltool, disabling .def/.lib generation") ENDIF() ELSE() SET_PROPERTY(TARGET OpenAL APPEND_STRING PROPERTY LINK_FLAGS " -Wl,--output-def,OpenAL32.def") ADD_CUSTOM_COMMAND(TARGET OpenAL POST_BUILD COMMAND "${SED_EXECUTABLE}" -i -e "s/ @[^ ]*//" OpenAL32.def COMMAND "${DLLTOOL_EXECUTABLE}" -d OpenAL32.def -l OpenAL32.lib -D OpenAL32.dll COMMENT "Stripping ordinals from OpenAL32.def and generating OpenAL32.lib..." VERBATIM ) ENDIF() ENDIF() IF(ALSOFT_INSTALL) INSTALL(TARGETS OpenAL EXPORT OpenAL RUNTIME DESTINATION ${CMAKE_INSTALL_BINDIR} LIBRARY DESTINATION ${CMAKE_INSTALL_LIBDIR} ARCHIVE DESTINATION ${CMAKE_INSTALL_LIBDIR} INCLUDES DESTINATION ${CMAKE_INSTALL_INCLUDEDIR} ${CMAKE_INSTALL_INCLUDEDIR}/AL ) EXPORT(TARGETS OpenAL NAMESPACE OpenAL:: FILE OpenALConfig.cmake) INSTALL(EXPORT OpenAL DESTINATION ${CMAKE_INSTALL_LIBDIR}/cmake/OpenAL NAMESPACE OpenAL:: FILE OpenALConfig.cmake) INSTALL(FILES include/AL/al.h include/AL/alc.h include/AL/alext.h include/AL/efx.h include/AL/efx-creative.h include/AL/efx-presets.h DESTINATION ${CMAKE_INSTALL_INCLUDEDIR}/AL ) INSTALL(FILES "${OpenAL_BINARY_DIR}/openal.pc" DESTINATION "${CMAKE_INSTALL_LIBDIR}/pkgconfig") IF(TARGET soft_oal) INSTALL(TARGETS soft_oal RUNTIME DESTINATION ${CMAKE_INSTALL_BINDIR} ) ENDIF() ENDIF() if(HAS_ROUTER) message(STATUS "") message(STATUS "Building DLL router") endif() MESSAGE(STATUS "") MESSAGE(STATUS "Building OpenAL with support for the following backends:") MESSAGE(STATUS " ${BACKENDS}") MESSAGE(STATUS "") MESSAGE(STATUS "Building with support for CPU extensions:") MESSAGE(STATUS " ${CPU_EXTS}") MESSAGE(STATUS "") IF(FPMATH_SET) MESSAGE(STATUS "Building with SSE${FPMATH_SET} codegen") MESSAGE(STATUS "") ENDIF() IF(WIN32) IF(NOT HAVE_DSOUND) MESSAGE(STATUS "WARNING: Building the Windows version without DirectSound output") MESSAGE(STATUS " This is probably NOT what you want!") MESSAGE(STATUS "") ENDIF() ENDIF() if(ALSOFT_EMBED_HRTF_DATA) message(STATUS "Embedding HRTF datasets") message(STATUS "") endif() # Install alsoft.conf configuration file IF(ALSOFT_CONFIG) INSTALL(FILES alsoftrc.sample DESTINATION ${CMAKE_INSTALL_DATADIR}/openal ) MESSAGE(STATUS "Installing sample configuration") MESSAGE(STATUS "") ENDIF() # Install HRTF definitions IF(ALSOFT_HRTF_DEFS) INSTALL(FILES hrtf/default-44100.mhr hrtf/default-48000.mhr DESTINATION ${CMAKE_INSTALL_DATADIR}/openal/hrtf ) MESSAGE(STATUS "Installing HRTF definitions") MESSAGE(STATUS "") ENDIF() # Install AmbDec presets IF(ALSOFT_AMBDEC_PRESETS) INSTALL(FILES presets/3D7.1.ambdec presets/hexagon.ambdec presets/itu5.1.ambdec presets/itu5.1-nocenter.ambdec presets/rectangle.ambdec presets/square.ambdec presets/presets.txt DESTINATION ${CMAKE_INSTALL_DATADIR}/openal/presets ) MESSAGE(STATUS "Installing AmbDec presets") MESSAGE(STATUS "") ENDIF() IF(ALSOFT_UTILS) ADD_EXECUTABLE(openal-info utils/openal-info.c) TARGET_COMPILE_OPTIONS(openal-info PRIVATE ${C_FLAGS}) TARGET_LINK_LIBRARIES(openal-info PRIVATE ${LINKER_FLAGS} OpenAL) SET(MAKEHRTF_SRCS utils/makehrtf.c) IF(NOT HAVE_GETOPT) SET(MAKEHRTF_SRCS ${MAKEHRTF_SRCS} utils/getopt.c utils/getopt.h) ENDIF() ADD_EXECUTABLE(makehrtf ${MAKEHRTF_SRCS}) TARGET_COMPILE_DEFINITIONS(makehrtf PRIVATE ${CPP_DEFS}) TARGET_COMPILE_OPTIONS(makehrtf PRIVATE ${C_FLAGS}) IF(HAVE_LIBM) TARGET_LINK_LIBRARIES(makehrtf PRIVATE ${LINKER_FLAGS} m) ENDIF() IF(ALSOFT_INSTALL) INSTALL(TARGETS openal-info makehrtf RUNTIME DESTINATION ${CMAKE_INSTALL_BINDIR} LIBRARY DESTINATION ${CMAKE_INSTALL_LIBDIR} ARCHIVE DESTINATION ${CMAKE_INSTALL_LIBDIR} ) ENDIF() MESSAGE(STATUS "Building utility programs") IF(TARGET alsoft-config) MESSAGE(STATUS "Building configuration program") ENDIF() MESSAGE(STATUS "") ENDIF() IF(ALSOFT_TESTS) SET(TEST_COMMON_OBJS examples/common/alhelpers.c) ADD_EXECUTABLE(altonegen examples/altonegen.c ${TEST_COMMON_OBJS}) TARGET_COMPILE_DEFINITIONS(altonegen PRIVATE ${CPP_DEFS}) TARGET_COMPILE_OPTIONS(altonegen PRIVATE ${C_FLAGS}) TARGET_LINK_LIBRARIES(altonegen PRIVATE ${LINKER_FLAGS} common OpenAL ${MATH_LIB}) IF(ALSOFT_INSTALL) INSTALL(TARGETS altonegen RUNTIME DESTINATION ${CMAKE_INSTALL_BINDIR} LIBRARY DESTINATION ${CMAKE_INSTALL_LIBDIR} ARCHIVE DESTINATION ${CMAKE_INSTALL_LIBDIR} ) ENDIF() MESSAGE(STATUS "Building test programs") MESSAGE(STATUS "") ENDIF() IF(ALSOFT_EXAMPLES) ADD_EXECUTABLE(alrecord examples/alrecord.c) TARGET_COMPILE_DEFINITIONS(alrecord PRIVATE ${CPP_DEFS}) TARGET_COMPILE_OPTIONS(alrecord PRIVATE ${C_FLAGS}) TARGET_LINK_LIBRARIES(alrecord PRIVATE ${LINKER_FLAGS} common OpenAL) IF(ALSOFT_INSTALL) INSTALL(TARGETS alrecord RUNTIME DESTINATION ${CMAKE_INSTALL_BINDIR} LIBRARY DESTINATION ${CMAKE_INSTALL_LIBDIR} ARCHIVE DESTINATION ${CMAKE_INSTALL_LIBDIR} ) ENDIF() MESSAGE(STATUS "Building example programs") IF(SDL2_FOUND) IF(SDL_SOUND_FOUND) # Add a static library with common functions used by multiple targets ADD_LIBRARY(ex-common STATIC examples/common/alhelpers.c) TARGET_COMPILE_DEFINITIONS(ex-common PRIVATE ${CPP_DEFS}) TARGET_COMPILE_OPTIONS(ex-common PRIVATE ${C_FLAGS}) ADD_EXECUTABLE(alplay examples/alplay.c) TARGET_COMPILE_DEFINITIONS(alplay PRIVATE ${CPP_DEFS}) TARGET_INCLUDE_DIRECTORIES(alplay PRIVATE ${SDL2_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR}) TARGET_COMPILE_OPTIONS(alplay PRIVATE ${C_FLAGS}) TARGET_LINK_LIBRARIES(alplay PRIVATE ${LINKER_FLAGS} ${SDL_SOUND_LIBRARIES} ${SDL2_LIBRARY} ex-common common OpenAL) ADD_EXECUTABLE(alstream examples/alstream.c) TARGET_COMPILE_DEFINITIONS(alstream PRIVATE ${CPP_DEFS}) TARGET_INCLUDE_DIRECTORIES(alstream PRIVATE ${SDL2_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR}) TARGET_COMPILE_OPTIONS(alstream PRIVATE ${C_FLAGS}) TARGET_LINK_LIBRARIES(alstream PRIVATE ${LINKER_FLAGS} ${SDL_SOUND_LIBRARIES} ${SDL2_LIBRARY} ex-common common OpenAL) ADD_EXECUTABLE(alreverb examples/alreverb.c) TARGET_COMPILE_DEFINITIONS(alreverb PRIVATE ${CPP_DEFS}) TARGET_INCLUDE_DIRECTORIES(alreverb PRIVATE ${SDL2_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR}) TARGET_COMPILE_OPTIONS(alreverb PRIVATE ${C_FLAGS}) TARGET_LINK_LIBRARIES(alreverb PRIVATE ${LINKER_FLAGS} ${SDL_SOUND_LIBRARIES} ${SDL2_LIBRARY} ex-common common OpenAL) ADD_EXECUTABLE(almultireverb examples/almultireverb.c) TARGET_COMPILE_DEFINITIONS(almultireverb PRIVATE ${CPP_DEFS}) TARGET_INCLUDE_DIRECTORIES(almultireverb PRIVATE ${SDL2_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR}) TARGET_COMPILE_OPTIONS(almultireverb PRIVATE ${C_FLAGS}) TARGET_LINK_LIBRARIES(almultireverb PRIVATE ${LINKER_FLAGS} ${SDL_SOUND_LIBRARIES} ${SDL2_LIBRARY} ex-common common OpenAL ${MATH_LIB}) ADD_EXECUTABLE(allatency examples/allatency.c) TARGET_COMPILE_DEFINITIONS(allatency PRIVATE ${CPP_DEFS}) TARGET_INCLUDE_DIRECTORIES(allatency PRIVATE ${SDL2_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR}) TARGET_COMPILE_OPTIONS(allatency PRIVATE ${C_FLAGS}) TARGET_LINK_LIBRARIES(allatency PRIVATE ${LINKER_FLAGS} ${SDL_SOUND_LIBRARIES} ${SDL2_LIBRARY} ex-common common OpenAL) ADD_EXECUTABLE(alloopback examples/alloopback.c) TARGET_COMPILE_DEFINITIONS(alloopback PRIVATE ${CPP_DEFS}) TARGET_INCLUDE_DIRECTORIES(alloopback PRIVATE ${SDL2_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR}) TARGET_COMPILE_OPTIONS(alloopback PRIVATE ${C_FLAGS}) TARGET_LINK_LIBRARIES(alloopback PRIVATE ${LINKER_FLAGS} ${SDL_SOUND_LIBRARIES} ${SDL2_LIBRARY} ex-common common OpenAL ${MATH_LIB}) ADD_EXECUTABLE(alhrtf examples/alhrtf.c) TARGET_COMPILE_DEFINITIONS(alhrtf PRIVATE ${CPP_DEFS}) TARGET_INCLUDE_DIRECTORIES(alhrtf PRIVATE ${SDL2_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR}) TARGET_COMPILE_OPTIONS(alhrtf PRIVATE ${C_FLAGS}) TARGET_LINK_LIBRARIES(alhrtf PRIVATE ${LINKER_FLAGS} ${SDL_SOUND_LIBRARIES} ${SDL2_LIBRARY} ex-common common OpenAL ${MATH_LIB}) IF(ALSOFT_INSTALL) INSTALL(TARGETS alplay alstream alreverb almultireverb allatency alloopback alhrtf RUNTIME DESTINATION ${CMAKE_INSTALL_BINDIR} LIBRARY DESTINATION ${CMAKE_INSTALL_LIBDIR} ARCHIVE DESTINATION ${CMAKE_INSTALL_LIBDIR} ) ENDIF() MESSAGE(STATUS "Building SDL_sound example programs") ENDIF() SET(FFVER_OK FALSE) IF(FFMPEG_FOUND) SET(FFVER_OK TRUE) IF(AVFORMAT_VERSION VERSION_LESS "57.56.101") MESSAGE(STATUS "libavformat is too old! (${AVFORMAT_VERSION}, wanted 57.56.101)") SET(FFVER_OK FALSE) ENDIF() IF(AVCODEC_VERSION VERSION_LESS "57.64.101") MESSAGE(STATUS "libavcodec is too old! (${AVCODEC_VERSION}, wanted 57.64.101)") SET(FFVER_OK FALSE) ENDIF() IF(AVUTIL_VERSION VERSION_LESS "55.34.101") MESSAGE(STATUS "libavutil is too old! (${AVUTIL_VERSION}, wanted 55.34.101)") SET(FFVER_OK FALSE) ENDIF() IF(SWSCALE_VERSION VERSION_LESS "4.2.100") MESSAGE(STATUS "libswscale is too old! (${SWSCALE_VERSION}, wanted 4.2.100)") SET(FFVER_OK FALSE) ENDIF() IF(SWRESAMPLE_VERSION VERSION_LESS "2.3.100") MESSAGE(STATUS "libswresample is too old! (${SWRESAMPLE_VERSION}, wanted 2.3.100)") SET(FFVER_OK FALSE) ENDIF() ENDIF() IF(FFVER_OK) ADD_EXECUTABLE(alffplay examples/alffplay.cpp) TARGET_COMPILE_DEFINITIONS(alffplay PRIVATE ${CPP_DEFS}) TARGET_INCLUDE_DIRECTORIES(alffplay PRIVATE ${SDL2_INCLUDE_DIR} ${FFMPEG_INCLUDE_DIRS}) TARGET_COMPILE_OPTIONS(alffplay PRIVATE ${C_FLAGS}) TARGET_LINK_LIBRARIES(alffplay PRIVATE ${LINKER_FLAGS} ${SDL2_LIBRARY} ${FFMPEG_LIBRARIES} ex-common common OpenAL) IF(ALSOFT_INSTALL) INSTALL(TARGETS alffplay RUNTIME DESTINATION ${CMAKE_INSTALL_BINDIR} LIBRARY DESTINATION ${CMAKE_INSTALL_LIBDIR} ARCHIVE DESTINATION ${CMAKE_INSTALL_LIBDIR} ) ENDIF() MESSAGE(STATUS "Building SDL+FFmpeg example programs") ENDIF() MESSAGE(STATUS "") ENDIF() ENDIF() openal-soft-openal-soft-1.19.1/COPYING000066400000000000000000000554421335774445300173430ustar00rootroot00000000000000 GNU LIBRARY GENERAL PUBLIC LICENSE Version 2, June 1991 Copyright (C) 1991 Free Software Foundation, Inc. 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA Everyone is permitted to copy and distribute verbatim copies of this license document, but changing it is not allowed. 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IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY AND/OR REDISTRIBUTE THE LIBRARY AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES, INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING OUT OF THE USE OR INABILITY TO USE THE LIBRARY (INCLUDING BUT NOT LIMITED TO LOSS OF DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY YOU OR THIRD PARTIES OR A FAILURE OF THE LIBRARY TO OPERATE WITH ANY OTHER SOFTWARE), EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES. END OF TERMS AND CONDITIONS openal-soft-openal-soft-1.19.1/ChangeLog000066400000000000000000000272651335774445300200640ustar00rootroot00000000000000openal-soft-1.19.1: Implemented capture support for the SoundIO backend. Fixed source buffer queues potentially not playing properly when a queue entry completes. Fixed possible unexpected failures when generating auxiliary effect slots. Fixed a crash with certain reverb or device settings. Fixed OpenSL capture. Improved output limiter response, better ensuring the sample amplitude is clamped for output. openal-soft-1.19.0: Implemented the ALC_SOFT_device_clock extension. Implemented the Pitch Shifter, Frequency Shifter, and Autowah effects. Fixed compiling on FreeBSD systems that use freebsd-lib 9.1. Fixed compiling on NetBSD. Fixed the reverb effect's density scale and panning parameters. Fixed use of the WASAPI backend with certain games, which caused odd COM initialization errors. Increased the number of virtual channels for decoding Ambisonics to HRTF output. Changed 32-bit x86 builds to use SSE2 math by default for performance. Build-time options are available to use just SSE1 or x87 instead. Replaced the 4-point Sinc resampler with a more efficient cubic resampler. Renamed the MMDevAPI backend to WASAPI. Added support for 24-bit, dual-ear HRTF data sets. The built-in data set has been updated to 24-bit. Added a 24- to 48-point band-limited Sinc resampler. Added an SDL2 playback backend. Disabled by default to avoid a dependency on SDL2. Improved the performance and quality of the Chorus and Flanger effects. Improved the efficiency of the band-limited Sinc resampler. Improved the Sinc resampler's transition band to avoid over-attenuating higher frequencies. Improved the performance of some filter operations. Improved the efficiency of object ID lookups. Improved the efficienty of internal voice/source synchronization. Improved AL call error logging with contextualized messages. Removed the reverb effect's modulation stage. Due to the lack of reference for its intended behavior and strength. openal-soft-1.18.2: Fixed resetting the FPU rounding mode after certain function calls on Windows. Fixed use of SSE intrinsics when building with Clang on Windows. Fixed a crash with the JACK backend when using JACK1. Fixed use of pthread_setnane_np on NetBSD. Fixed building on FreeBSD with an older freebsd-lib. OSS now links with libossaudio if found at build time (for NetBSD). openal-soft-1.18.1: Fixed an issue where resuming a source might not restart playing it. Fixed PulseAudio playback when the configured stream length is much less than the requested length. Fixed MMDevAPI capture with sample rates not matching the backing device. Fixed int32 output for the Wave Writer. Fixed enumeration of OSS devices that are missing device files. Added correct retrieval of the executable's path on FreeBSD. Added a config option to specify the dithering depth. Added a 5.1 decoder preset that excludes front-center output. openal-soft-1.18.0: Implemented the AL_EXT_STEREO_ANGLES and AL_EXT_SOURCE_RADIUS extensions. Implemented the AL_SOFT_gain_clamp_ex, AL_SOFT_source_resampler, AL_SOFT_source_spatialize, and ALC_SOFT_output_limiter extensions. Implemented 3D processing for some effects. Currently implemented for Reverb, Compressor, Equalizer, and Ring Modulator. Implemented 2-channel UHJ output encoding. This needs to be enabled with a config option to be used. Implemented dual-band processing for high-quality ambisonic decoding. Implemented distance-compensation for surround sound output. Implemented near-field emulation and compensation with ambisonic rendering. Currently only applies when using the high-quality ambisonic decoder or ambisonic output, with appropriate config options. Implemented an output limiter to reduce the amount of distortion from clipping. Implemented dithering for 8-bit and 16-bit output. Implemented a config option to select a preferred HRTF. Implemented a run-time check for NEON extensions using /proc/cpuinfo. Implemented experimental capture support for the OpenSL backend. Fixed building on compilers with NEON support but don't default to having NEON enabled. Fixed support for JACK on Windows. Fixed starting a source while alcSuspendContext is in effect. Fixed detection of headsets as headphones, with MMDevAPI. Added support for AmbDec config files, for custom ambisonic decoder configurations. Version 3 files only. Added backend-specific options to alsoft-config. Added first-, second-, and third-order ambisonic output formats. Currently only works with backends that don't rely on channel labels, like JACK, ALSA, and OSS. Added a build option to embed the default HRTFs into the lib. Added AmbDec presets to enable high-quality ambisonic decoding. Added an AmbDec preset for 3D7.1 speaker setups. Added documentation regarding Ambisonics, 3D7.1, AmbDec config files, and the provided ambdec presets. Added the ability for MMDevAPI to open devices given a Device ID or GUID string. Added an option to the example apps to open a specific device. Increased the maximum auxiliary send limit to 16 (up from 4). Requires requesting them with the ALC_MAX_AUXILIARY_SENDS context creation attribute. Increased the default auxiliary effect slot count to 64 (up from 4). Reduced the default period count to 3 (down from 4). Slightly improved automatic naming for enumerated HRTFs. Improved B-Format decoding with HRTF output. Improved internal property handling for better batching behavior. Improved performance of certain filter uses. Removed support for the AL_SOFT_buffer_samples and AL_SOFT_buffer_sub_data extensions. Due to conflicts with AL_EXT_SOURCE_RADIUS. openal-soft-1.17.2: Implemented device enumeration for OSSv4. Fixed building on OSX. Fixed building on non-Windows systems without POSIX-2008. Fixed Dedicated Dialog and Dedicated LFE effect output. Added a build option to override the share install dir. Added a build option to static-link libgcc for MinGW. openal-soft-1.17.1: Fixed building with JACK and without PulseAudio. Fixed building on FreeBSD. Fixed the ALSA backend's allow-resampler option. Fixed handling of inexact ALSA period counts. Altered device naming scheme on Windows backends to better match other drivers. Updated the CoreAudio backend to use the AudioComponent API. This clears up deprecation warnings for OSX 10.11, although requires OSX 10.6 or newer. openal-soft-1.17.0: Implemented a JACK playback backend. Implemented the AL_EXT_BFORMAT and AL_EXT_MULAW_BFORMAT extensions. Implemented the ALC_SOFT_HRTF extension. Implemented C, SSE3, and SSE4.1 based 4- and 8-point Sinc resamplers. Implemented a C and SSE based band-limited Sinc resampler. This does 12- to 24-point Sinc resampling, and performs anti-aliasing. Implemented B-Format output support for the wave file writer. This creates FuMa-style first-order Ambisonics wave files (AMB format). Implemented a stereo-mode config option for treating stereo modes as either speakers or headphones. Implemented per-device configuration options. Fixed handling of PulseAudio and MMDevAPI devices that have identical descriptions. Fixed a potential lockup when stopping playback of suspended PulseAudio devices. Fixed logging of Unicode characters on Windows. Fixed 5.1 surround sound channels. By default it will now use the side channels for the surround output. A configuration using rear channels is still available. Fixed the QSA backend potentially altering the capture format. Fixed detecting MMDevAPI's default device. Fixed returning the default capture device name. Fixed mixing property calculations when deferring context updates. Altered the behavior of alcSuspendContext and alcProcessContext to better match certain Windows drivers. Altered the panning algorithm, utilizing Ambisonics for better side and back positioning cues with surround sound output. Improved support for certain older Windows apps. Improved the alffplay example to support surround sound streams. Improved support for building as a sub-project. Added an HRTF playback example. Added a tone generator output test. Added a toolchain to help with cross-compiling to Android. openal-soft-1.16.0: Implemented EFX Chorus, Flanger, Distortion, Equalizer, and Compressor effects. Implemented high-pass and band-pass EFX filters. Implemented the high-pass filter for the EAXReverb effect. Implemented SSE2 and SSE4.1 linear resamplers. Implemented Neon-enhanced non-HRTF mixers. Implemented a QSA backend, for QNX. Implemented the ALC_SOFT_pause_device, AL_SOFT_deferred_updates, AL_SOFT_block_alignment, AL_SOFT_MSADPCM, and AL_SOFT_source_length extensions. Fixed resetting mmdevapi backend devices. Fixed clamping when converting 32-bit float samples to integer. Fixed modulation range in the Modulator effect. Several fixes for the OpenSL playback backend. Fixed device specifier names that have Unicode characters on Windows. Added support for filenames and paths with Unicode (UTF-8) characters on Windows. Added support for alsoft.conf config files found in XDG Base Directory Specification locations (XDG_CONFIG_DIRS and XDG_CONFIG_HOME, or their defaults) on non-Windows systems. Added a GUI configuration utility (requires Qt 4.8). Added support for environment variable expansion in config options (not keys or section names). Added an example that uses SDL2 and ffmpeg. Modified examples to use SDL_sound. Modified CMake config option names for better sorting. HRTF data sets specified in the hrtf_tables config option may now be relative or absolute filenames. Made the default HRTF data set an external file, and added a data set for 48khz playback in addition to 44.1khz. Added support for C11 atomic methods. Improved support for some non-GNU build systems. openal-soft-1.15.1: Fixed a regression with retrieving the source's AL_GAIN property. openal-soft-1.15: Fixed device enumeration with the OSS backend. Reorganized internal mixing logic, so unneeded steps can potentially be skipped for better performance. Removed the lookup table for calculating the mixing pans. The panning is now calculated directly for better precision. Improved the panning of stereo source channels when using stereo output. Improved source filter quality on send paths. Added a config option to allow PulseAudio to move streams between devices. The PulseAudio backend will now attempt to spawn a server by default. Added a workaround for a DirectSound bug relating to float32 output. Added SSE-based mixers, for HRTF and non-HRTF mixing. Added support for the new AL_SOFT_source_latency extension. Improved ALSA capture by avoiding an extra buffer when using sizes supported by the underlying device. Improved the makehrtf utility to support new options and input formats. Modified the CFLAGS declared in the pkg-config file so the "AL/" portion of the header includes can optionally be omitted. Added a couple example code programs to show how to apply reverb, and retrieve latency. The configuration sample is now installed into the share/openal/ directory instead of /etc/openal. The configuration sample now gets installed by default. openal-soft-openal-soft-1.19.1/OpenAL32/000077500000000000000000000000001335774445300175615ustar00rootroot00000000000000openal-soft-openal-soft-1.19.1/OpenAL32/Include/000077500000000000000000000000001335774445300211445ustar00rootroot00000000000000openal-soft-openal-soft-1.19.1/OpenAL32/Include/alAuxEffectSlot.h000066400000000000000000000140721335774445300243520ustar00rootroot00000000000000#ifndef _AL_AUXEFFECTSLOT_H_ #define _AL_AUXEFFECTSLOT_H_ #include "alMain.h" #include "alEffect.h" #include "atomic.h" #include "align.h" #ifdef __cplusplus extern "C" { #endif struct ALeffectStateVtable; struct ALeffectslot; typedef struct ALeffectState { RefCount Ref; const struct ALeffectStateVtable *vtbl; ALfloat (*OutBuffer)[BUFFERSIZE]; ALsizei OutChannels; } ALeffectState; void ALeffectState_Construct(ALeffectState *state); void ALeffectState_Destruct(ALeffectState *state); struct ALeffectStateVtable { void (*const Destruct)(ALeffectState *state); ALboolean (*const deviceUpdate)(ALeffectState *state, ALCdevice *device); void (*const update)(ALeffectState *state, const ALCcontext *context, const struct ALeffectslot *slot, const union ALeffectProps *props); void (*const process)(ALeffectState *state, ALsizei samplesToDo, const ALfloat (*restrict samplesIn)[BUFFERSIZE], ALfloat (*restrict samplesOut)[BUFFERSIZE], ALsizei numChannels); void (*const Delete)(void *ptr); }; /* Small hack to use a pointer-to-array types as a normal argument type. * Shouldn't be used directly. */ typedef ALfloat ALfloatBUFFERSIZE[BUFFERSIZE]; #define DEFINE_ALEFFECTSTATE_VTABLE(T) \ DECLARE_THUNK(T, ALeffectState, void, Destruct) \ DECLARE_THUNK1(T, ALeffectState, ALboolean, deviceUpdate, ALCdevice*) \ DECLARE_THUNK3(T, ALeffectState, void, update, const ALCcontext*, const ALeffectslot*, const ALeffectProps*) \ DECLARE_THUNK4(T, ALeffectState, void, process, ALsizei, const ALfloatBUFFERSIZE*restrict, ALfloatBUFFERSIZE*restrict, ALsizei) \ static void T##_ALeffectState_Delete(void *ptr) \ { return T##_Delete(STATIC_UPCAST(T, ALeffectState, (ALeffectState*)ptr)); } \ \ static const struct ALeffectStateVtable T##_ALeffectState_vtable = { \ T##_ALeffectState_Destruct, \ \ T##_ALeffectState_deviceUpdate, \ T##_ALeffectState_update, \ T##_ALeffectState_process, \ \ T##_ALeffectState_Delete, \ } struct EffectStateFactoryVtable; typedef struct EffectStateFactory { const struct EffectStateFactoryVtable *vtab; } EffectStateFactory; struct EffectStateFactoryVtable { ALeffectState *(*const create)(EffectStateFactory *factory); }; #define EffectStateFactory_create(x) ((x)->vtab->create((x))) #define DEFINE_EFFECTSTATEFACTORY_VTABLE(T) \ DECLARE_THUNK(T, EffectStateFactory, ALeffectState*, create) \ \ static const struct EffectStateFactoryVtable T##_EffectStateFactory_vtable = { \ T##_EffectStateFactory_create, \ } #define MAX_EFFECT_CHANNELS (4) struct ALeffectslotArray { ALsizei count; struct ALeffectslot *slot[]; }; struct ALeffectslotProps { ALfloat Gain; ALboolean AuxSendAuto; ALenum Type; ALeffectProps Props; ALeffectState *State; ATOMIC(struct ALeffectslotProps*) next; }; typedef struct ALeffectslot { ALfloat Gain; ALboolean AuxSendAuto; struct { ALenum Type; ALeffectProps Props; ALeffectState *State; } Effect; ATOMIC_FLAG PropsClean; RefCount ref; ATOMIC(struct ALeffectslotProps*) Update; struct { ALfloat Gain; ALboolean AuxSendAuto; ALenum EffectType; ALeffectProps EffectProps; ALeffectState *EffectState; ALfloat RoomRolloff; /* Added to the source's room rolloff, not multiplied. */ ALfloat DecayTime; ALfloat DecayLFRatio; ALfloat DecayHFRatio; ALboolean DecayHFLimit; ALfloat AirAbsorptionGainHF; } Params; /* Self ID */ ALuint id; ALsizei NumChannels; BFChannelConfig ChanMap[MAX_EFFECT_CHANNELS]; /* Wet buffer configuration is ACN channel order with N3D scaling: * * Channel 0 is the unattenuated mono signal. * * Channel 1 is OpenAL -X * sqrt(3) * * Channel 2 is OpenAL Y * sqrt(3) * * Channel 3 is OpenAL -Z * sqrt(3) * Consequently, effects that only want to work with mono input can use * channel 0 by itself. Effects that want multichannel can process the * ambisonics signal and make a B-Format source pan for first-order device * output (FOAOut). */ alignas(16) ALfloat WetBuffer[MAX_EFFECT_CHANNELS][BUFFERSIZE]; } ALeffectslot; ALenum InitEffectSlot(ALeffectslot *slot); void DeinitEffectSlot(ALeffectslot *slot); void UpdateEffectSlotProps(ALeffectslot *slot, ALCcontext *context); void UpdateAllEffectSlotProps(ALCcontext *context); ALvoid ReleaseALAuxiliaryEffectSlots(ALCcontext *Context); EffectStateFactory *NullStateFactory_getFactory(void); EffectStateFactory *ReverbStateFactory_getFactory(void); EffectStateFactory *AutowahStateFactory_getFactory(void); EffectStateFactory *ChorusStateFactory_getFactory(void); EffectStateFactory *CompressorStateFactory_getFactory(void); EffectStateFactory *DistortionStateFactory_getFactory(void); EffectStateFactory *EchoStateFactory_getFactory(void); EffectStateFactory *EqualizerStateFactory_getFactory(void); EffectStateFactory *FlangerStateFactory_getFactory(void); EffectStateFactory *FshifterStateFactory_getFactory(void); EffectStateFactory *ModulatorStateFactory_getFactory(void); EffectStateFactory *PshifterStateFactory_getFactory(void); EffectStateFactory *DedicatedStateFactory_getFactory(void); ALenum InitializeEffect(ALCcontext *Context, ALeffectslot *EffectSlot, ALeffect *effect); void ALeffectState_DecRef(ALeffectState *state); #ifdef __cplusplus } #endif #endif openal-soft-openal-soft-1.19.1/OpenAL32/Include/alBuffer.h000066400000000000000000000046001335774445300230430ustar00rootroot00000000000000#ifndef _AL_BUFFER_H_ #define _AL_BUFFER_H_ #include "AL/alc.h" #include "AL/al.h" #include "AL/alext.h" #include "inprogext.h" #include "atomic.h" #include "rwlock.h" #ifdef __cplusplus extern "C" { #endif /* User formats */ enum UserFmtType { UserFmtUByte, UserFmtShort, UserFmtFloat, UserFmtDouble, UserFmtMulaw, UserFmtAlaw, UserFmtIMA4, UserFmtMSADPCM, }; enum UserFmtChannels { UserFmtMono, UserFmtStereo, UserFmtRear, UserFmtQuad, UserFmtX51, /* (WFX order) */ UserFmtX61, /* (WFX order) */ UserFmtX71, /* (WFX order) */ UserFmtBFormat2D, /* WXY */ UserFmtBFormat3D, /* WXYZ */ }; ALsizei BytesFromUserFmt(enum UserFmtType type); ALsizei ChannelsFromUserFmt(enum UserFmtChannels chans); inline ALsizei FrameSizeFromUserFmt(enum UserFmtChannels chans, enum UserFmtType type) { return ChannelsFromUserFmt(chans) * BytesFromUserFmt(type); } /* Storable formats */ enum FmtType { FmtUByte = UserFmtUByte, FmtShort = UserFmtShort, FmtFloat = UserFmtFloat, FmtDouble = UserFmtDouble, FmtMulaw = UserFmtMulaw, FmtAlaw = UserFmtAlaw, }; enum FmtChannels { FmtMono = UserFmtMono, FmtStereo = UserFmtStereo, FmtRear = UserFmtRear, FmtQuad = UserFmtQuad, FmtX51 = UserFmtX51, FmtX61 = UserFmtX61, FmtX71 = UserFmtX71, FmtBFormat2D = UserFmtBFormat2D, FmtBFormat3D = UserFmtBFormat3D, }; #define MAX_INPUT_CHANNELS (8) ALsizei BytesFromFmt(enum FmtType type); ALsizei ChannelsFromFmt(enum FmtChannels chans); inline ALsizei FrameSizeFromFmt(enum FmtChannels chans, enum FmtType type) { return ChannelsFromFmt(chans) * BytesFromFmt(type); } typedef struct ALbuffer { ALvoid *data; ALsizei Frequency; ALbitfieldSOFT Access; ALsizei SampleLen; enum FmtChannels FmtChannels; enum FmtType FmtType; ALsizei BytesAlloc; enum UserFmtType OriginalType; ALsizei OriginalSize; ALsizei OriginalAlign; ALsizei LoopStart; ALsizei LoopEnd; ATOMIC(ALsizei) UnpackAlign; ATOMIC(ALsizei) PackAlign; ALbitfieldSOFT MappedAccess; ALsizei MappedOffset; ALsizei MappedSize; /* Number of times buffer was attached to a source (deletion can only occur when 0) */ RefCount ref; /* Self ID */ ALuint id; } ALbuffer; ALvoid ReleaseALBuffers(ALCdevice *device); #ifdef __cplusplus } #endif #endif openal-soft-openal-soft-1.19.1/OpenAL32/Include/alEffect.h000066400000000000000000000134001335774445300230240ustar00rootroot00000000000000#ifndef _AL_EFFECT_H_ #define _AL_EFFECT_H_ #include "alMain.h" #ifdef __cplusplus extern "C" { #endif struct ALeffect; enum { EAXREVERB_EFFECT = 0, REVERB_EFFECT, AUTOWAH_EFFECT, CHORUS_EFFECT, COMPRESSOR_EFFECT, DISTORTION_EFFECT, ECHO_EFFECT, EQUALIZER_EFFECT, FLANGER_EFFECT, FSHIFTER_EFFECT, MODULATOR_EFFECT, PSHIFTER_EFFECT, DEDICATED_EFFECT, MAX_EFFECTS }; extern ALboolean DisabledEffects[MAX_EFFECTS]; extern ALfloat ReverbBoost; struct EffectList { const char name[16]; int type; ALenum val; }; #define EFFECTLIST_SIZE 14 extern const struct EffectList EffectList[EFFECTLIST_SIZE]; struct ALeffectVtable { void (*const setParami)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALint val); void (*const setParamiv)(struct ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals); void (*const setParamf)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val); void (*const setParamfv)(struct ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals); void (*const getParami)(const struct ALeffect *effect, ALCcontext *context, ALenum param, ALint *val); void (*const getParamiv)(const struct ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals); void (*const getParamf)(const struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val); void (*const getParamfv)(const struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals); }; #define DEFINE_ALEFFECT_VTABLE(T) \ const struct ALeffectVtable T##_vtable = { \ T##_setParami, T##_setParamiv, \ T##_setParamf, T##_setParamfv, \ T##_getParami, T##_getParamiv, \ T##_getParamf, T##_getParamfv, \ } extern const struct ALeffectVtable ALeaxreverb_vtable; extern const struct ALeffectVtable ALreverb_vtable; extern const struct ALeffectVtable ALautowah_vtable; extern const struct ALeffectVtable ALchorus_vtable; extern const struct ALeffectVtable ALcompressor_vtable; extern const struct ALeffectVtable ALdistortion_vtable; extern const struct ALeffectVtable ALecho_vtable; extern const struct ALeffectVtable ALequalizer_vtable; extern const struct ALeffectVtable ALflanger_vtable; extern const struct ALeffectVtable ALfshifter_vtable; extern const struct ALeffectVtable ALmodulator_vtable; extern const struct ALeffectVtable ALnull_vtable; extern const struct ALeffectVtable ALpshifter_vtable; extern const struct ALeffectVtable ALdedicated_vtable; typedef union ALeffectProps { struct { // Shared Reverb Properties ALfloat Density; ALfloat Diffusion; ALfloat Gain; ALfloat GainHF; ALfloat DecayTime; ALfloat DecayHFRatio; ALfloat ReflectionsGain; ALfloat ReflectionsDelay; ALfloat LateReverbGain; ALfloat LateReverbDelay; ALfloat AirAbsorptionGainHF; ALfloat RoomRolloffFactor; ALboolean DecayHFLimit; // Additional EAX Reverb Properties ALfloat GainLF; ALfloat DecayLFRatio; ALfloat ReflectionsPan[3]; ALfloat LateReverbPan[3]; ALfloat EchoTime; ALfloat EchoDepth; ALfloat ModulationTime; ALfloat ModulationDepth; ALfloat HFReference; ALfloat LFReference; } Reverb; struct { ALfloat AttackTime; ALfloat ReleaseTime; ALfloat Resonance; ALfloat PeakGain; } Autowah; struct { ALint Waveform; ALint Phase; ALfloat Rate; ALfloat Depth; ALfloat Feedback; ALfloat Delay; } Chorus; /* Also Flanger */ struct { ALboolean OnOff; } Compressor; struct { ALfloat Edge; ALfloat Gain; ALfloat LowpassCutoff; ALfloat EQCenter; ALfloat EQBandwidth; } Distortion; struct { ALfloat Delay; ALfloat LRDelay; ALfloat Damping; ALfloat Feedback; ALfloat Spread; } Echo; struct { ALfloat LowCutoff; ALfloat LowGain; ALfloat Mid1Center; ALfloat Mid1Gain; ALfloat Mid1Width; ALfloat Mid2Center; ALfloat Mid2Gain; ALfloat Mid2Width; ALfloat HighCutoff; ALfloat HighGain; } Equalizer; struct { ALfloat Frequency; ALint LeftDirection; ALint RightDirection; } Fshifter; struct { ALfloat Frequency; ALfloat HighPassCutoff; ALint Waveform; } Modulator; struct { ALint CoarseTune; ALint FineTune; } Pshifter; struct { ALfloat Gain; } Dedicated; } ALeffectProps; typedef struct ALeffect { // Effect type (AL_EFFECT_NULL, ...) ALenum type; ALeffectProps Props; const struct ALeffectVtable *vtab; /* Self ID */ ALuint id; } ALeffect; #define ALeffect_setParami(o, c, p, v) ((o)->vtab->setParami(o, c, p, v)) #define ALeffect_setParamf(o, c, p, v) ((o)->vtab->setParamf(o, c, p, v)) #define ALeffect_setParamiv(o, c, p, v) ((o)->vtab->setParamiv(o, c, p, v)) #define ALeffect_setParamfv(o, c, p, v) ((o)->vtab->setParamfv(o, c, p, v)) #define ALeffect_getParami(o, c, p, v) ((o)->vtab->getParami(o, c, p, v)) #define ALeffect_getParamf(o, c, p, v) ((o)->vtab->getParamf(o, c, p, v)) #define ALeffect_getParamiv(o, c, p, v) ((o)->vtab->getParamiv(o, c, p, v)) #define ALeffect_getParamfv(o, c, p, v) ((o)->vtab->getParamfv(o, c, p, v)) inline ALboolean IsReverbEffect(ALenum type) { return type == AL_EFFECT_REVERB || type == AL_EFFECT_EAXREVERB; } void InitEffect(ALeffect *effect); void ReleaseALEffects(ALCdevice *device); void LoadReverbPreset(const char *name, ALeffect *effect); #ifdef __cplusplus } #endif #endif openal-soft-openal-soft-1.19.1/OpenAL32/Include/alError.h000066400000000000000000000014471335774445300227310ustar00rootroot00000000000000#ifndef _AL_ERROR_H_ #define _AL_ERROR_H_ #include "alMain.h" #include "logging.h" #ifdef __cplusplus extern "C" { #endif extern ALboolean TrapALError; void alSetError(ALCcontext *context, ALenum errorCode, const char *msg, ...) DECL_FORMAT(printf, 3, 4); #define SETERR_GOTO(ctx, err, lbl, ...) do { \ alSetError((ctx), (err), __VA_ARGS__); \ goto lbl; \ } while(0) #define SETERR_RETURN(ctx, err, retval, ...) do { \ alSetError((ctx), (err), __VA_ARGS__); \ return retval; \ } while(0) #ifdef __cplusplus } #endif #endif openal-soft-openal-soft-1.19.1/OpenAL32/Include/alFilter.h000066400000000000000000000044551335774445300230670ustar00rootroot00000000000000#ifndef _AL_FILTER_H_ #define _AL_FILTER_H_ #include "AL/alc.h" #include "AL/al.h" #ifdef __cplusplus extern "C" { #endif #define LOWPASSFREQREF (5000.0f) #define HIGHPASSFREQREF (250.0f) struct ALfilter; typedef struct ALfilterVtable { void (*const setParami)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALint val); void (*const setParamiv)(struct ALfilter *filter, ALCcontext *context, ALenum param, const ALint *vals); void (*const setParamf)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val); void (*const setParamfv)(struct ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals); void (*const getParami)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALint *val); void (*const getParamiv)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALint *vals); void (*const getParamf)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val); void (*const getParamfv)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals); } ALfilterVtable; #define DEFINE_ALFILTER_VTABLE(T) \ const struct ALfilterVtable T##_vtable = { \ T##_setParami, T##_setParamiv, \ T##_setParamf, T##_setParamfv, \ T##_getParami, T##_getParamiv, \ T##_getParamf, T##_getParamfv, \ } typedef struct ALfilter { // Filter type (AL_FILTER_NULL, ...) ALenum type; ALfloat Gain; ALfloat GainHF; ALfloat HFReference; ALfloat GainLF; ALfloat LFReference; const struct ALfilterVtable *vtab; /* Self ID */ ALuint id; } ALfilter; #define ALfilter_setParami(o, c, p, v) ((o)->vtab->setParami(o, c, p, v)) #define ALfilter_setParamf(o, c, p, v) ((o)->vtab->setParamf(o, c, p, v)) #define ALfilter_setParamiv(o, c, p, v) ((o)->vtab->setParamiv(o, c, p, v)) #define ALfilter_setParamfv(o, c, p, v) ((o)->vtab->setParamfv(o, c, p, v)) #define ALfilter_getParami(o, c, p, v) ((o)->vtab->getParami(o, c, p, v)) #define ALfilter_getParamf(o, c, p, v) ((o)->vtab->getParamf(o, c, p, v)) #define ALfilter_getParamiv(o, c, p, v) ((o)->vtab->getParamiv(o, c, p, v)) #define ALfilter_getParamfv(o, c, p, v) ((o)->vtab->getParamfv(o, c, p, v)) void ReleaseALFilters(ALCdevice *device); #ifdef __cplusplus } #endif #endif openal-soft-openal-soft-1.19.1/OpenAL32/Include/alListener.h000066400000000000000000000024741335774445300234260ustar00rootroot00000000000000#ifndef _AL_LISTENER_H_ #define _AL_LISTENER_H_ #include "alMain.h" #include "alu.h" #ifdef __cplusplus extern "C" { #endif struct ALcontextProps { ALfloat DopplerFactor; ALfloat DopplerVelocity; ALfloat SpeedOfSound; ALboolean SourceDistanceModel; enum DistanceModel DistanceModel; ALfloat MetersPerUnit; ATOMIC(struct ALcontextProps*) next; }; struct ALlistenerProps { ALfloat Position[3]; ALfloat Velocity[3]; ALfloat Forward[3]; ALfloat Up[3]; ALfloat Gain; ATOMIC(struct ALlistenerProps*) next; }; typedef struct ALlistener { alignas(16) ALfloat Position[3]; ALfloat Velocity[3]; ALfloat Forward[3]; ALfloat Up[3]; ALfloat Gain; ATOMIC_FLAG PropsClean; /* Pointer to the most recent property values that are awaiting an update. */ ATOMIC(struct ALlistenerProps*) Update; struct { aluMatrixf Matrix; aluVector Velocity; ALfloat Gain; ALfloat MetersPerUnit; ALfloat DopplerFactor; ALfloat SpeedOfSound; /* in units per sec! */ ALfloat ReverbSpeedOfSound; /* in meters per sec! */ ALboolean SourceDistanceModel; enum DistanceModel DistanceModel; } Params; } ALlistener; void UpdateListenerProps(ALCcontext *context); #ifdef __cplusplus } #endif #endif openal-soft-openal-soft-1.19.1/OpenAL32/Include/alMain.h000066400000000000000000000566271335774445300225360ustar00rootroot00000000000000#ifndef AL_MAIN_H #define AL_MAIN_H #include #include #include #include #include #include #include #ifdef HAVE_STRINGS_H #include #endif #ifdef HAVE_INTRIN_H #include #endif #include "AL/al.h" #include "AL/alc.h" #include "AL/alext.h" #include "inprogext.h" #include "logging.h" #include "polymorphism.h" #include "static_assert.h" #include "align.h" #include "atomic.h" #include "vector.h" #include "alstring.h" #include "almalloc.h" #include "threads.h" #if defined(_WIN64) #define SZFMT "%I64u" #elif defined(_WIN32) #define SZFMT "%u" #else #define SZFMT "%zu" #endif #ifdef __has_builtin #define HAS_BUILTIN __has_builtin #else #define HAS_BUILTIN(x) (0) #endif #ifdef __GNUC__ /* LIKELY optimizes the case where the condition is true. The condition is not * required to be true, but it can result in more optimal code for the true * path at the expense of a less optimal false path. */ #define LIKELY(x) __builtin_expect(!!(x), !0) /* The opposite of LIKELY, optimizing the case where the condition is false. */ #define UNLIKELY(x) __builtin_expect(!!(x), 0) /* Unlike LIKELY, ASSUME requires the condition to be true or else it invokes * undefined behavior. It's essentially an assert without actually checking the * condition at run-time, allowing for stronger optimizations than LIKELY. */ #if HAS_BUILTIN(__builtin_assume) #define ASSUME __builtin_assume #else #define ASSUME(x) do { if(!(x)) __builtin_unreachable(); } while(0) #endif #else #define LIKELY(x) (!!(x)) #define UNLIKELY(x) (!!(x)) #ifdef _MSC_VER #define ASSUME __assume #else #define ASSUME(x) ((void)0) #endif #endif #ifndef UINT64_MAX #define UINT64_MAX U64(18446744073709551615) #endif #ifndef UNUSED #if defined(__cplusplus) #define UNUSED(x) #elif defined(__GNUC__) #define UNUSED(x) UNUSED_##x __attribute__((unused)) #elif defined(__LCLINT__) #define UNUSED(x) /*@unused@*/ x #else #define UNUSED(x) x #endif #endif /* Calculates the size of a struct with N elements of a flexible array member. * GCC and Clang allow offsetof(Type, fam[N]) for this, but MSVC seems to have * trouble, so a bit more verbose workaround is needed. */ #define FAM_SIZE(T, M, N) (offsetof(T, M) + sizeof(((T*)NULL)->M[0])*(N)) #ifdef __cplusplus extern "C" { #endif typedef ALint64SOFT ALint64; typedef ALuint64SOFT ALuint64; #ifndef U64 #if defined(_MSC_VER) #define U64(x) ((ALuint64)(x##ui64)) #elif SIZEOF_LONG == 8 #define U64(x) ((ALuint64)(x##ul)) #elif SIZEOF_LONG_LONG == 8 #define U64(x) ((ALuint64)(x##ull)) #endif #endif #ifndef I64 #if defined(_MSC_VER) #define I64(x) ((ALint64)(x##i64)) #elif SIZEOF_LONG == 8 #define I64(x) ((ALint64)(x##l)) #elif SIZEOF_LONG_LONG == 8 #define I64(x) ((ALint64)(x##ll)) #endif #endif /* Define a CTZ64 macro (count trailing zeros, for 64-bit integers). The result * is *UNDEFINED* if the value is 0. */ #ifdef __GNUC__ #if SIZEOF_LONG == 8 #define CTZ64 __builtin_ctzl #else #define CTZ64 __builtin_ctzll #endif #elif defined(HAVE_BITSCANFORWARD64_INTRINSIC) inline int msvc64_ctz64(ALuint64 v) { unsigned long idx = 64; _BitScanForward64(&idx, v); return (int)idx; } #define CTZ64 msvc64_ctz64 #elif defined(HAVE_BITSCANFORWARD_INTRINSIC) inline int msvc_ctz64(ALuint64 v) { unsigned long idx = 64; if(!_BitScanForward(&idx, v&0xffffffff)) { if(_BitScanForward(&idx, v>>32)) idx += 32; } return (int)idx; } #define CTZ64 msvc_ctz64 #else /* There be black magics here. The popcnt64 method is derived from * https://graphics.stanford.edu/~seander/bithacks.html#CountBitsSetParallel * while the ctz-utilizing-popcnt algorithm is shown here * http://www.hackersdelight.org/hdcodetxt/ntz.c.txt * as the ntz2 variant. These likely aren't the most efficient methods, but * they're good enough if the GCC or MSVC intrinsics aren't available. */ inline int fallback_popcnt64(ALuint64 v) { v = v - ((v >> 1) & U64(0x5555555555555555)); v = (v & U64(0x3333333333333333)) + ((v >> 2) & U64(0x3333333333333333)); v = (v + (v >> 4)) & U64(0x0f0f0f0f0f0f0f0f); return (int)((v * U64(0x0101010101010101)) >> 56); } inline int fallback_ctz64(ALuint64 value) { return fallback_popcnt64(~value & (value - 1)); } #define CTZ64 fallback_ctz64 #endif #if defined(__BYTE_ORDER__) && defined(__ORDER_LITTLE_ENDIAN__) #define IS_LITTLE_ENDIAN (__BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__) #else static const union { ALuint u; ALubyte b[sizeof(ALuint)]; } EndianTest = { 1 }; #define IS_LITTLE_ENDIAN (EndianTest.b[0] == 1) #endif #define COUNTOF(x) (sizeof(x) / sizeof(0[x])) struct ll_ringbuffer; struct Hrtf; struct HrtfEntry; struct DirectHrtfState; struct FrontStablizer; struct Compressor; struct ALCbackend; struct ALbuffer; struct ALeffect; struct ALfilter; struct ALsource; struct ALcontextProps; struct ALlistenerProps; struct ALvoiceProps; struct ALeffectslotProps; #define DEFAULT_OUTPUT_RATE (44100) #define MIN_OUTPUT_RATE (8000) /* Find the next power-of-2 for non-power-of-2 numbers. */ inline ALuint NextPowerOf2(ALuint value) { if(value > 0) { value--; value |= value>>1; value |= value>>2; value |= value>>4; value |= value>>8; value |= value>>16; } return value+1; } /** Round up a value to the next multiple. */ inline size_t RoundUp(size_t value, size_t r) { value += r-1; return value - (value%r); } /* Fast float-to-int conversion. No particular rounding mode is assumed; the * IEEE-754 default is round-to-nearest with ties-to-even, though an app could * change it on its own threads. On some systems, a truncating conversion may * always be the fastest method. */ inline ALint fastf2i(ALfloat f) { #if defined(HAVE_INTRIN_H) && ((defined(_M_IX86_FP) && (_M_IX86_FP > 0)) || defined(_M_X64)) return _mm_cvt_ss2si(_mm_set1_ps(f)); #elif defined(_MSC_VER) && defined(_M_IX86_FP) ALint i; __asm fld f __asm fistp i return i; #elif (defined(__GNUC__) || defined(__clang__)) && (defined(__i386__) || defined(__x86_64__)) ALint i; #ifdef __SSE_MATH__ __asm__("cvtss2si %1, %0" : "=r"(i) : "x"(f)); #else __asm__ __volatile__("fistpl %0" : "=m"(i) : "t"(f) : "st"); #endif return i; /* On GCC when compiling with -fno-math-errno, lrintf can be inlined to * some simple instructions. Clang does not inline it, always generating a * libc call, while MSVC's implementation is horribly slow, so always fall * back to a normal integer conversion for them. */ #elif defined(HAVE_LRINTF) && !defined(_MSC_VER) && !defined(__clang__) return lrintf(f); #else return (ALint)f; #endif } /* Converts float-to-int using standard behavior (truncation). */ inline int float2int(float f) { #if ((defined(__GNUC__) || defined(__clang__)) && (defined(__i386__) || defined(__x86_64__)) && \ !defined(__SSE_MATH__)) || (defined(_MSC_VER) && defined(_M_IX86_FP) && _M_IX86_FP == 0) ALint sign, shift, mant; union { ALfloat f; ALint i; } conv; conv.f = f; sign = (conv.i>>31) | 1; shift = ((conv.i>>23)&0xff) - (127+23); /* Over/underflow */ if(UNLIKELY(shift >= 31 || shift < -23)) return 0; mant = (conv.i&0x7fffff) | 0x800000; if(LIKELY(shift < 0)) return (mant >> -shift) * sign; return (mant << shift) * sign; #else return (ALint)f; #endif } /* Rounds a float to the nearest integral value, according to the current * rounding mode. This is essentially an inlined version of rintf, although * makes fewer promises (e.g. -0 or -0.25 rounded to 0 may result in +0). */ inline float fast_roundf(float f) { #if (defined(__GNUC__) || defined(__clang__)) && (defined(__i386__) || defined(__x86_64__)) && \ !defined(__SSE_MATH__) float out; __asm__ __volatile__("frndint" : "=t"(out) : "0"(f)); return out; #else /* Integral limit, where sub-integral precision is not available for * floats. */ static const float ilim[2] = { 8388608.0f /* 0x1.0p+23 */, -8388608.0f /* -0x1.0p+23 */ }; ALuint sign, expo; union { ALfloat f; ALuint i; } conv; conv.f = f; sign = (conv.i>>31)&0x01; expo = (conv.i>>23)&0xff; if(UNLIKELY(expo >= 150/*+23*/)) { /* An exponent (base-2) of 23 or higher is incapable of sub-integral * precision, so it's already an integral value. We don't need to worry * about infinity or NaN here. */ return f; } /* Adding the integral limit to the value (with a matching sign) forces a * result that has no sub-integral precision, and is consequently forced to * round to an integral value. Removing the integral limit then restores * the initial value rounded to the integral. The compiler should not * optimize this out because of non-associative rules on floating-point * math (as long as you don't use -fassociative-math, * -funsafe-math-optimizations, -ffast-math, or -Ofast, in which case this * may break). */ f += ilim[sign]; return f - ilim[sign]; #endif } enum DevProbe { ALL_DEVICE_PROBE, CAPTURE_DEVICE_PROBE }; enum DistanceModel { InverseDistanceClamped = AL_INVERSE_DISTANCE_CLAMPED, LinearDistanceClamped = AL_LINEAR_DISTANCE_CLAMPED, ExponentDistanceClamped = AL_EXPONENT_DISTANCE_CLAMPED, InverseDistance = AL_INVERSE_DISTANCE, LinearDistance = AL_LINEAR_DISTANCE, ExponentDistance = AL_EXPONENT_DISTANCE, DisableDistance = AL_NONE, DefaultDistanceModel = InverseDistanceClamped }; enum Channel { FrontLeft = 0, FrontRight, FrontCenter, LFE, BackLeft, BackRight, BackCenter, SideLeft, SideRight, UpperFrontLeft, UpperFrontRight, UpperBackLeft, UpperBackRight, LowerFrontLeft, LowerFrontRight, LowerBackLeft, LowerBackRight, Aux0, Aux1, Aux2, Aux3, Aux4, Aux5, Aux6, Aux7, Aux8, Aux9, Aux10, Aux11, Aux12, Aux13, Aux14, Aux15, InvalidChannel }; /* Device formats */ enum DevFmtType { DevFmtByte = ALC_BYTE_SOFT, DevFmtUByte = ALC_UNSIGNED_BYTE_SOFT, DevFmtShort = ALC_SHORT_SOFT, DevFmtUShort = ALC_UNSIGNED_SHORT_SOFT, DevFmtInt = ALC_INT_SOFT, DevFmtUInt = ALC_UNSIGNED_INT_SOFT, DevFmtFloat = ALC_FLOAT_SOFT, DevFmtTypeDefault = DevFmtFloat }; enum DevFmtChannels { DevFmtMono = ALC_MONO_SOFT, DevFmtStereo = ALC_STEREO_SOFT, DevFmtQuad = ALC_QUAD_SOFT, DevFmtX51 = ALC_5POINT1_SOFT, DevFmtX61 = ALC_6POINT1_SOFT, DevFmtX71 = ALC_7POINT1_SOFT, DevFmtAmbi3D = ALC_BFORMAT3D_SOFT, /* Similar to 5.1, except using rear channels instead of sides */ DevFmtX51Rear = 0x80000000, DevFmtChannelsDefault = DevFmtStereo }; #define MAX_OUTPUT_CHANNELS (16) ALsizei BytesFromDevFmt(enum DevFmtType type); ALsizei ChannelsFromDevFmt(enum DevFmtChannels chans, ALsizei ambiorder); inline ALsizei FrameSizeFromDevFmt(enum DevFmtChannels chans, enum DevFmtType type, ALsizei ambiorder) { return ChannelsFromDevFmt(chans, ambiorder) * BytesFromDevFmt(type); } enum AmbiLayout { AmbiLayout_FuMa = ALC_FUMA_SOFT, /* FuMa channel order */ AmbiLayout_ACN = ALC_ACN_SOFT, /* ACN channel order */ AmbiLayout_Default = AmbiLayout_ACN }; enum AmbiNorm { AmbiNorm_FuMa = ALC_FUMA_SOFT, /* FuMa normalization */ AmbiNorm_SN3D = ALC_SN3D_SOFT, /* SN3D normalization */ AmbiNorm_N3D = ALC_N3D_SOFT, /* N3D normalization */ AmbiNorm_Default = AmbiNorm_SN3D }; enum DeviceType { Playback, Capture, Loopback }; enum RenderMode { NormalRender, StereoPair, HrtfRender }; /* The maximum number of Ambisonics coefficients. For a given order (o), the * size needed will be (o+1)**2, thus zero-order has 1, first-order has 4, * second-order has 9, third-order has 16, and fourth-order has 25. */ #define MAX_AMBI_ORDER 3 #define MAX_AMBI_COEFFS ((MAX_AMBI_ORDER+1) * (MAX_AMBI_ORDER+1)) /* A bitmask of ambisonic channels with height information. If none of these * channels are used/needed, there's no height (e.g. with most surround sound * speaker setups). This only specifies up to 4th order, which is the highest * order a 32-bit mask value can specify (a 64-bit mask could handle up to 7th * order). This is ACN ordering, with bit 0 being ACN 0, etc. */ #define AMBI_PERIPHONIC_MASK (0xfe7ce4) /* The maximum number of Ambisonic coefficients for 2D (non-periphonic) * representation. This is 2 per each order above zero-order, plus 1 for zero- * order. Or simply, o*2 + 1. */ #define MAX_AMBI2D_COEFFS (MAX_AMBI_ORDER*2 + 1) typedef ALfloat ChannelConfig[MAX_AMBI_COEFFS]; typedef struct BFChannelConfig { ALfloat Scale; ALsizei Index; } BFChannelConfig; typedef union AmbiConfig { /* Ambisonic coefficients for mixing to the dry buffer. */ ChannelConfig Coeffs[MAX_OUTPUT_CHANNELS]; /* Coefficient channel mapping for mixing to the dry buffer. */ BFChannelConfig Map[MAX_OUTPUT_CHANNELS]; } AmbiConfig; typedef struct BufferSubList { ALuint64 FreeMask; struct ALbuffer *Buffers; /* 64 */ } BufferSubList; TYPEDEF_VECTOR(BufferSubList, vector_BufferSubList) typedef struct EffectSubList { ALuint64 FreeMask; struct ALeffect *Effects; /* 64 */ } EffectSubList; TYPEDEF_VECTOR(EffectSubList, vector_EffectSubList) typedef struct FilterSubList { ALuint64 FreeMask; struct ALfilter *Filters; /* 64 */ } FilterSubList; TYPEDEF_VECTOR(FilterSubList, vector_FilterSubList) typedef struct SourceSubList { ALuint64 FreeMask; struct ALsource *Sources; /* 64 */ } SourceSubList; TYPEDEF_VECTOR(SourceSubList, vector_SourceSubList) /* Effect slots are rather large, and apps aren't likely to have more than one * or two (let alone 64), so hold them individually. */ typedef struct ALeffectslot *ALeffectslotPtr; TYPEDEF_VECTOR(ALeffectslotPtr, vector_ALeffectslotPtr) typedef struct EnumeratedHrtf { al_string name; struct HrtfEntry *hrtf; } EnumeratedHrtf; TYPEDEF_VECTOR(EnumeratedHrtf, vector_EnumeratedHrtf) /* Maximum delay in samples for speaker distance compensation. */ #define MAX_DELAY_LENGTH 1024 typedef struct DistanceComp { ALfloat Gain; ALsizei Length; /* Valid range is [0...MAX_DELAY_LENGTH). */ ALfloat *Buffer; } DistanceComp; /* Size for temporary storage of buffer data, in ALfloats. Larger values need * more memory, while smaller values may need more iterations. The value needs * to be a sensible size, however, as it constrains the max stepping value used * for mixing, as well as the maximum number of samples per mixing iteration. */ #define BUFFERSIZE 2048 typedef struct MixParams { AmbiConfig Ambi; /* Number of coefficients in each Ambi.Coeffs to mix together (4 for first- * order, 9 for second-order, etc). If the count is 0, Ambi.Map is used * instead to map each output to a coefficient index. */ ALsizei CoeffCount; ALfloat (*Buffer)[BUFFERSIZE]; ALsizei NumChannels; } MixParams; typedef struct RealMixParams { enum Channel ChannelName[MAX_OUTPUT_CHANNELS]; ALfloat (*Buffer)[BUFFERSIZE]; ALsizei NumChannels; } RealMixParams; typedef void (*POSTPROCESS)(ALCdevice *device, ALsizei SamplesToDo); struct ALCdevice_struct { RefCount ref; ATOMIC(ALenum) Connected; enum DeviceType Type; ALuint Frequency; ALuint UpdateSize; ALuint NumUpdates; enum DevFmtChannels FmtChans; enum DevFmtType FmtType; ALboolean IsHeadphones; ALsizei AmbiOrder; /* For DevFmtAmbi* output only, specifies the channel order and * normalization. */ enum AmbiLayout AmbiLayout; enum AmbiNorm AmbiScale; ALCenum LimiterState; al_string DeviceName; ATOMIC(ALCenum) LastError; // Maximum number of sources that can be created ALuint SourcesMax; // Maximum number of slots that can be created ALuint AuxiliaryEffectSlotMax; ALCuint NumMonoSources; ALCuint NumStereoSources; ALsizei NumAuxSends; // Map of Buffers for this device vector_BufferSubList BufferList; almtx_t BufferLock; // Map of Effects for this device vector_EffectSubList EffectList; almtx_t EffectLock; // Map of Filters for this device vector_FilterSubList FilterList; almtx_t FilterLock; POSTPROCESS PostProcess; /* HRTF state and info */ struct DirectHrtfState *Hrtf; al_string HrtfName; struct Hrtf *HrtfHandle; vector_EnumeratedHrtf HrtfList; ALCenum HrtfStatus; /* UHJ encoder state */ struct Uhj2Encoder *Uhj_Encoder; /* High quality Ambisonic decoder */ struct BFormatDec *AmbiDecoder; /* Stereo-to-binaural filter */ struct bs2b *Bs2b; /* First-order ambisonic upsampler for higher-order output */ struct AmbiUpsampler *AmbiUp; /* Rendering mode. */ enum RenderMode Render_Mode; // Device flags ALuint Flags; ALuint64 ClockBase; ALuint SamplesDone; ALuint FixedLatency; /* Temp storage used for mixer processing. */ alignas(16) ALfloat TempBuffer[4][BUFFERSIZE]; /* The "dry" path corresponds to the main output. */ MixParams Dry; ALsizei NumChannelsPerOrder[MAX_AMBI_ORDER+1]; /* First-order ambisonics output, to be upsampled to the dry buffer if different. */ MixParams FOAOut; /* "Real" output, which will be written to the device buffer. May alias the * dry buffer. */ RealMixParams RealOut; struct FrontStablizer *Stablizer; struct Compressor *Limiter; /* The average speaker distance as determined by the ambdec configuration * (or alternatively, by the NFC-HOA reference delay). Only used for NFC. */ ALfloat AvgSpeakerDist; /* Delay buffers used to compensate for speaker distances. */ DistanceComp ChannelDelay[MAX_OUTPUT_CHANNELS]; /* Dithering control. */ ALfloat DitherDepth; ALuint DitherSeed; /* Running count of the mixer invocations, in 31.1 fixed point. This * actually increments *twice* when mixing, first at the start and then at * the end, so the bottom bit indicates if the device is currently mixing * and the upper bits indicates how many mixes have been done. */ RefCount MixCount; // Contexts created on this device ATOMIC(ALCcontext*) ContextList; almtx_t BackendLock; struct ALCbackend *Backend; ATOMIC(ALCdevice*) next; }; // Frequency was requested by the app or config file #define DEVICE_FREQUENCY_REQUEST (1u<<1) // Channel configuration was requested by the config file #define DEVICE_CHANNELS_REQUEST (1u<<2) // Sample type was requested by the config file #define DEVICE_SAMPLE_TYPE_REQUEST (1u<<3) // Specifies if the DSP is paused at user request #define DEVICE_PAUSED (1u<<30) // Specifies if the device is currently running #define DEVICE_RUNNING (1u<<31) /* Nanosecond resolution for the device clock time. */ #define DEVICE_CLOCK_RES U64(1000000000) /* Must be less than 15 characters (16 including terminating null) for * compatibility with pthread_setname_np limitations. */ #define MIXER_THREAD_NAME "alsoft-mixer" #define RECORD_THREAD_NAME "alsoft-record" enum { /* End event thread processing. */ EventType_KillThread = 0, /* User event types. */ EventType_SourceStateChange = 1<<0, EventType_BufferCompleted = 1<<1, EventType_Error = 1<<2, EventType_Performance = 1<<3, EventType_Deprecated = 1<<4, EventType_Disconnected = 1<<5, /* Internal events. */ EventType_ReleaseEffectState = 65536, }; typedef struct AsyncEvent { unsigned int EnumType; union { char dummy; struct { ALenum type; ALuint id; ALuint param; ALchar msg[1008]; } user; struct ALeffectState *EffectState; } u; } AsyncEvent; #define ASYNC_EVENT(t) { t, { 0 } } struct ALCcontext_struct { RefCount ref; struct ALlistener *Listener; vector_SourceSubList SourceList; ALuint NumSources; almtx_t SourceLock; vector_ALeffectslotPtr EffectSlotList; almtx_t EffectSlotLock; ATOMIC(ALenum) LastError; enum DistanceModel DistanceModel; ALboolean SourceDistanceModel; ALfloat DopplerFactor; ALfloat DopplerVelocity; ALfloat SpeedOfSound; ALfloat MetersPerUnit; ATOMIC_FLAG PropsClean; ATOMIC(ALenum) DeferUpdates; almtx_t PropLock; /* Counter for the pre-mixing updates, in 31.1 fixed point (lowest bit * indicates if updates are currently happening). */ RefCount UpdateCount; ATOMIC(ALenum) HoldUpdates; ALfloat GainBoost; ATOMIC(struct ALcontextProps*) Update; /* Linked lists of unused property containers, free to use for future * updates. */ ATOMIC(struct ALcontextProps*) FreeContextProps; ATOMIC(struct ALlistenerProps*) FreeListenerProps; ATOMIC(struct ALvoiceProps*) FreeVoiceProps; ATOMIC(struct ALeffectslotProps*) FreeEffectslotProps; struct ALvoice **Voices; ALsizei VoiceCount; ALsizei MaxVoices; ATOMIC(struct ALeffectslotArray*) ActiveAuxSlots; althrd_t EventThread; alsem_t EventSem; struct ll_ringbuffer *AsyncEvents; ATOMIC(ALbitfieldSOFT) EnabledEvts; almtx_t EventCbLock; ALEVENTPROCSOFT EventCb; void *EventParam; /* Default effect slot */ struct ALeffectslot *DefaultSlot; ALCdevice *Device; const ALCchar *ExtensionList; ATOMIC(ALCcontext*) next; /* Memory space used by the listener (and possibly default effect slot) */ alignas(16) ALCbyte _listener_mem[]; }; ALCcontext *GetContextRef(void); void ALCcontext_DecRef(ALCcontext *context); void ALCcontext_DeferUpdates(ALCcontext *context); void ALCcontext_ProcessUpdates(ALCcontext *context); void AllocateVoices(ALCcontext *context, ALsizei num_voices, ALsizei old_sends); extern ALint RTPrioLevel; void SetRTPriority(void); void SetDefaultChannelOrder(ALCdevice *device); void SetDefaultWFXChannelOrder(ALCdevice *device); const ALCchar *DevFmtTypeString(enum DevFmtType type); const ALCchar *DevFmtChannelsString(enum DevFmtChannels chans); inline ALint GetChannelIndex(const enum Channel names[MAX_OUTPUT_CHANNELS], enum Channel chan) { ALint i; for(i = 0;i < MAX_OUTPUT_CHANNELS;i++) { if(names[i] == chan) return i; } return -1; } /** * GetChannelIdxByName * * Returns the index for the given channel name (e.g. FrontCenter), or -1 if it * doesn't exist. */ inline ALint GetChannelIdxByName(const RealMixParams *real, enum Channel chan) { return GetChannelIndex(real->ChannelName, chan); } inline void LockBufferList(ALCdevice *device) { almtx_lock(&device->BufferLock); } inline void UnlockBufferList(ALCdevice *device) { almtx_unlock(&device->BufferLock); } inline void LockEffectList(ALCdevice *device) { almtx_lock(&device->EffectLock); } inline void UnlockEffectList(ALCdevice *device) { almtx_unlock(&device->EffectLock); } inline void LockFilterList(ALCdevice *device) { almtx_lock(&device->FilterLock); } inline void UnlockFilterList(ALCdevice *device) { almtx_unlock(&device->FilterLock); } inline void LockEffectSlotList(ALCcontext *context) { almtx_lock(&context->EffectSlotLock); } inline void UnlockEffectSlotList(ALCcontext *context) { almtx_unlock(&context->EffectSlotLock); } int EventThread(void *arg); vector_al_string SearchDataFiles(const char *match, const char *subdir); #ifdef __cplusplus } #endif #endif openal-soft-openal-soft-1.19.1/OpenAL32/Include/alSource.h000066400000000000000000000047741335774445300231060ustar00rootroot00000000000000#ifndef _AL_SOURCE_H_ #define _AL_SOURCE_H_ #include "bool.h" #include "alMain.h" #include "alu.h" #include "hrtf.h" #include "atomic.h" #define MAX_SENDS 16 #define DEFAULT_SENDS 2 #ifdef __cplusplus extern "C" { #endif struct ALbuffer; struct ALsource; typedef struct ALbufferlistitem { ATOMIC(struct ALbufferlistitem*) next; ALsizei max_samples; ALsizei num_buffers; struct ALbuffer *buffers[]; } ALbufferlistitem; typedef struct ALsource { /** Source properties. */ ALfloat Pitch; ALfloat Gain; ALfloat OuterGain; ALfloat MinGain; ALfloat MaxGain; ALfloat InnerAngle; ALfloat OuterAngle; ALfloat RefDistance; ALfloat MaxDistance; ALfloat RolloffFactor; ALfloat Position[3]; ALfloat Velocity[3]; ALfloat Direction[3]; ALfloat Orientation[2][3]; ALboolean HeadRelative; ALboolean Looping; enum DistanceModel DistanceModel; enum Resampler Resampler; ALboolean DirectChannels; enum SpatializeMode Spatialize; ALboolean DryGainHFAuto; ALboolean WetGainAuto; ALboolean WetGainHFAuto; ALfloat OuterGainHF; ALfloat AirAbsorptionFactor; ALfloat RoomRolloffFactor; ALfloat DopplerFactor; /* NOTE: Stereo pan angles are specified in radians, counter-clockwise * rather than clockwise. */ ALfloat StereoPan[2]; ALfloat Radius; /** Direct filter and auxiliary send info. */ struct { ALfloat Gain; ALfloat GainHF; ALfloat HFReference; ALfloat GainLF; ALfloat LFReference; } Direct; struct { struct ALeffectslot *Slot; ALfloat Gain; ALfloat GainHF; ALfloat HFReference; ALfloat GainLF; ALfloat LFReference; } *Send; /** * Last user-specified offset, and the offset type (bytes, samples, or * seconds). */ ALdouble Offset; ALenum OffsetType; /** Source type (static, streaming, or undetermined) */ ALint SourceType; /** Source state (initial, playing, paused, or stopped) */ ALenum state; /** Source Buffer Queue head. */ ALbufferlistitem *queue; ATOMIC_FLAG PropsClean; /* Index into the context's Voices array. Lazily updated, only checked and * reset when looking up the voice. */ ALint VoiceIdx; /** Self ID */ ALuint id; } ALsource; void UpdateAllSourceProps(ALCcontext *context); ALvoid ReleaseALSources(ALCcontext *Context); #ifdef __cplusplus } #endif #endif openal-soft-openal-soft-1.19.1/OpenAL32/Include/alu.h000066400000000000000000000364501335774445300221060ustar00rootroot00000000000000#ifndef _ALU_H_ #define _ALU_H_ #include #include #ifdef HAVE_FLOAT_H #include #endif #ifdef HAVE_IEEEFP_H #include #endif #include "alMain.h" #include "alBuffer.h" #include "hrtf.h" #include "align.h" #include "math_defs.h" #include "filters/defs.h" #include "filters/nfc.h" #define MAX_PITCH (255) /* Maximum number of samples to pad on either end of a buffer for resampling. * Note that both the beginning and end need padding! */ #define MAX_RESAMPLE_PADDING 24 #ifdef __cplusplus extern "C" { #endif struct BSincTable; struct ALsource; struct ALbufferlistitem; struct ALvoice; struct ALeffectslot; #define DITHER_RNG_SEED 22222 enum SpatializeMode { SpatializeOff = AL_FALSE, SpatializeOn = AL_TRUE, SpatializeAuto = AL_AUTO_SOFT }; enum Resampler { PointResampler, LinearResampler, FIR4Resampler, BSinc12Resampler, BSinc24Resampler, ResamplerMax = BSinc24Resampler }; extern enum Resampler ResamplerDefault; /* The number of distinct scale and phase intervals within the bsinc filter * table. */ #define BSINC_SCALE_BITS 4 #define BSINC_SCALE_COUNT (1<v[0] = x; vector->v[1] = y; vector->v[2] = z; vector->v[3] = w; } typedef union aluMatrixf { alignas(16) ALfloat m[4][4]; } aluMatrixf; extern const aluMatrixf IdentityMatrixf; inline void aluMatrixfSetRow(aluMatrixf *matrix, ALuint row, ALfloat m0, ALfloat m1, ALfloat m2, ALfloat m3) { matrix->m[row][0] = m0; matrix->m[row][1] = m1; matrix->m[row][2] = m2; matrix->m[row][3] = m3; } inline void aluMatrixfSet(aluMatrixf *matrix, ALfloat m00, ALfloat m01, ALfloat m02, ALfloat m03, ALfloat m10, ALfloat m11, ALfloat m12, ALfloat m13, ALfloat m20, ALfloat m21, ALfloat m22, ALfloat m23, ALfloat m30, ALfloat m31, ALfloat m32, ALfloat m33) { aluMatrixfSetRow(matrix, 0, m00, m01, m02, m03); aluMatrixfSetRow(matrix, 1, m10, m11, m12, m13); aluMatrixfSetRow(matrix, 2, m20, m21, m22, m23); aluMatrixfSetRow(matrix, 3, m30, m31, m32, m33); } enum ActiveFilters { AF_None = 0, AF_LowPass = 1, AF_HighPass = 2, AF_BandPass = AF_LowPass | AF_HighPass }; typedef struct MixHrtfParams { const ALfloat (*Coeffs)[2]; ALsizei Delay[2]; ALfloat Gain; ALfloat GainStep; } MixHrtfParams; typedef struct DirectParams { BiquadFilter LowPass; BiquadFilter HighPass; NfcFilter NFCtrlFilter; struct { HrtfParams Old; HrtfParams Target; HrtfState State; } Hrtf; struct { ALfloat Current[MAX_OUTPUT_CHANNELS]; ALfloat Target[MAX_OUTPUT_CHANNELS]; } Gains; } DirectParams; typedef struct SendParams { BiquadFilter LowPass; BiquadFilter HighPass; struct { ALfloat Current[MAX_OUTPUT_CHANNELS]; ALfloat Target[MAX_OUTPUT_CHANNELS]; } Gains; } SendParams; struct ALvoiceProps { ATOMIC(struct ALvoiceProps*) next; ALfloat Pitch; ALfloat Gain; ALfloat OuterGain; ALfloat MinGain; ALfloat MaxGain; ALfloat InnerAngle; ALfloat OuterAngle; ALfloat RefDistance; ALfloat MaxDistance; ALfloat RolloffFactor; ALfloat Position[3]; ALfloat Velocity[3]; ALfloat Direction[3]; ALfloat Orientation[2][3]; ALboolean HeadRelative; enum DistanceModel DistanceModel; enum Resampler Resampler; ALboolean DirectChannels; enum SpatializeMode SpatializeMode; ALboolean DryGainHFAuto; ALboolean WetGainAuto; ALboolean WetGainHFAuto; ALfloat OuterGainHF; ALfloat AirAbsorptionFactor; ALfloat RoomRolloffFactor; ALfloat DopplerFactor; ALfloat StereoPan[2]; ALfloat Radius; /** Direct filter and auxiliary send info. */ struct { ALfloat Gain; ALfloat GainHF; ALfloat HFReference; ALfloat GainLF; ALfloat LFReference; } Direct; struct { struct ALeffectslot *Slot; ALfloat Gain; ALfloat GainHF; ALfloat HFReference; ALfloat GainLF; ALfloat LFReference; } Send[]; }; #define VOICE_IS_STATIC (1<<0) #define VOICE_IS_FADING (1<<1) /* Fading sources use gain stepping for smooth transitions. */ #define VOICE_HAS_HRTF (1<<2) #define VOICE_HAS_NFC (1<<3) typedef struct ALvoice { struct ALvoiceProps *Props; ATOMIC(struct ALvoiceProps*) Update; ATOMIC(struct ALsource*) Source; ATOMIC(bool) Playing; /** * Source offset in samples, relative to the currently playing buffer, NOT * the whole queue, and the fractional (fixed-point) offset to the next * sample. */ ATOMIC(ALuint) position; ATOMIC(ALsizei) position_fraction; /* Current buffer queue item being played. */ ATOMIC(struct ALbufferlistitem*) current_buffer; /* Buffer queue item to loop to at end of queue (will be NULL for non- * looping voices). */ ATOMIC(struct ALbufferlistitem*) loop_buffer; /** * Number of channels and bytes-per-sample for the attached source's * buffer(s). */ ALsizei NumChannels; ALsizei SampleSize; /** Current target parameters used for mixing. */ ALint Step; ResamplerFunc Resampler; ALuint Flags; ALuint Offset; /* Number of output samples mixed since starting. */ alignas(16) ALfloat PrevSamples[MAX_INPUT_CHANNELS][MAX_RESAMPLE_PADDING]; InterpState ResampleState; struct { enum ActiveFilters FilterType; DirectParams Params[MAX_INPUT_CHANNELS]; ALfloat (*Buffer)[BUFFERSIZE]; ALsizei Channels; ALsizei ChannelsPerOrder[MAX_AMBI_ORDER+1]; } Direct; struct { enum ActiveFilters FilterType; SendParams Params[MAX_INPUT_CHANNELS]; ALfloat (*Buffer)[BUFFERSIZE]; ALsizei Channels; } Send[]; } ALvoice; void DeinitVoice(ALvoice *voice); typedef void (*MixerFunc)(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos, ALsizei BufferSize); typedef void (*RowMixerFunc)(ALfloat *OutBuffer, const ALfloat *gains, const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans, ALsizei InPos, ALsizei BufferSize); typedef void (*HrtfMixerFunc)(ALfloat *restrict LeftOut, ALfloat *restrict RightOut, const ALfloat *data, ALsizei Offset, ALsizei OutPos, const ALsizei IrSize, MixHrtfParams *hrtfparams, HrtfState *hrtfstate, ALsizei BufferSize); typedef void (*HrtfMixerBlendFunc)(ALfloat *restrict LeftOut, ALfloat *restrict RightOut, const ALfloat *data, ALsizei Offset, ALsizei OutPos, const ALsizei IrSize, const HrtfParams *oldparams, MixHrtfParams *newparams, HrtfState *hrtfstate, ALsizei BufferSize); typedef void (*HrtfDirectMixerFunc)(ALfloat *restrict LeftOut, ALfloat *restrict RightOut, const ALfloat *data, ALsizei Offset, const ALsizei IrSize, const ALfloat (*restrict Coeffs)[2], ALfloat (*restrict Values)[2], ALsizei BufferSize); #define GAIN_MIX_MAX (16.0f) /* +24dB */ #define GAIN_SILENCE_THRESHOLD (0.00001f) /* -100dB */ #define SPEEDOFSOUNDMETRESPERSEC (343.3f) #define AIRABSORBGAINHF (0.99426f) /* -0.05dB */ /* Target gain for the reverb decay feedback reaching the decay time. */ #define REVERB_DECAY_GAIN (0.001f) /* -60 dB */ #define FRACTIONBITS (12) #define FRACTIONONE (1< b) ? b : a); } inline ALfloat maxf(ALfloat a, ALfloat b) { return ((a > b) ? a : b); } inline ALfloat clampf(ALfloat val, ALfloat min, ALfloat max) { return minf(max, maxf(min, val)); } inline ALdouble mind(ALdouble a, ALdouble b) { return ((a > b) ? b : a); } inline ALdouble maxd(ALdouble a, ALdouble b) { return ((a > b) ? a : b); } inline ALdouble clampd(ALdouble val, ALdouble min, ALdouble max) { return mind(max, maxd(min, val)); } inline ALuint minu(ALuint a, ALuint b) { return ((a > b) ? b : a); } inline ALuint maxu(ALuint a, ALuint b) { return ((a > b) ? a : b); } inline ALuint clampu(ALuint val, ALuint min, ALuint max) { return minu(max, maxu(min, val)); } inline ALint mini(ALint a, ALint b) { return ((a > b) ? b : a); } inline ALint maxi(ALint a, ALint b) { return ((a > b) ? a : b); } inline ALint clampi(ALint val, ALint min, ALint max) { return mini(max, maxi(min, val)); } inline ALint64 mini64(ALint64 a, ALint64 b) { return ((a > b) ? b : a); } inline ALint64 maxi64(ALint64 a, ALint64 b) { return ((a > b) ? a : b); } inline ALint64 clampi64(ALint64 val, ALint64 min, ALint64 max) { return mini64(max, maxi64(min, val)); } inline ALuint64 minu64(ALuint64 a, ALuint64 b) { return ((a > b) ? b : a); } inline ALuint64 maxu64(ALuint64 a, ALuint64 b) { return ((a > b) ? a : b); } inline ALuint64 clampu64(ALuint64 val, ALuint64 min, ALuint64 max) { return minu64(max, maxu64(min, val)); } inline size_t minz(size_t a, size_t b) { return ((a > b) ? b : a); } inline size_t maxz(size_t a, size_t b) { return ((a > b) ? a : b); } inline size_t clampz(size_t val, size_t min, size_t max) { return minz(max, maxz(min, val)); } inline ALfloat lerp(ALfloat val1, ALfloat val2, ALfloat mu) { return val1 + (val2-val1)*mu; } inline ALfloat cubic(ALfloat val1, ALfloat val2, ALfloat val3, ALfloat val4, ALfloat mu) { ALfloat mu2 = mu*mu, mu3 = mu2*mu; ALfloat a0 = -0.5f*mu3 + mu2 + -0.5f*mu; ALfloat a1 = 1.5f*mu3 + -2.5f*mu2 + 1.0f; ALfloat a2 = -1.5f*mu3 + 2.0f*mu2 + 0.5f*mu; ALfloat a3 = 0.5f*mu3 + -0.5f*mu2; return val1*a0 + val2*a1 + val3*a2 + val4*a3; } enum HrtfRequestMode { Hrtf_Default = 0, Hrtf_Enable = 1, Hrtf_Disable = 2, }; void aluInit(void); void aluInitMixer(void); ResamplerFunc SelectResampler(enum Resampler resampler); /* aluInitRenderer * * Set up the appropriate panning method and mixing method given the device * properties. */ void aluInitRenderer(ALCdevice *device, ALint hrtf_id, enum HrtfRequestMode hrtf_appreq, enum HrtfRequestMode hrtf_userreq); void aluInitEffectPanning(struct ALeffectslot *slot); void aluSelectPostProcess(ALCdevice *device); /** * Calculates ambisonic encoder coefficients using the X, Y, and Z direction * components, which must represent a normalized (unit length) vector, and the * spread is the angular width of the sound (0...tau). * * NOTE: The components use ambisonic coordinates. As a result: * * Ambisonic Y = OpenAL -X * Ambisonic Z = OpenAL Y * Ambisonic X = OpenAL -Z * * The components are ordered such that OpenAL's X, Y, and Z are the first, * second, and third parameters respectively -- simply negate X and Z. */ void CalcAmbiCoeffs(const ALfloat y, const ALfloat z, const ALfloat x, const ALfloat spread, ALfloat coeffs[MAX_AMBI_COEFFS]); /** * CalcDirectionCoeffs * * Calculates ambisonic coefficients based on an OpenAL direction vector. The * vector must be normalized (unit length), and the spread is the angular width * of the sound (0...tau). */ inline void CalcDirectionCoeffs(const ALfloat dir[3], ALfloat spread, ALfloat coeffs[MAX_AMBI_COEFFS]) { /* Convert from OpenAL coords to Ambisonics. */ CalcAmbiCoeffs(-dir[0], dir[1], -dir[2], spread, coeffs); } /** * CalcAngleCoeffs * * Calculates ambisonic coefficients based on azimuth and elevation. The * azimuth and elevation parameters are in radians, going right and up * respectively. */ inline void CalcAngleCoeffs(ALfloat azimuth, ALfloat elevation, ALfloat spread, ALfloat coeffs[MAX_AMBI_COEFFS]) { ALfloat x = -sinf(azimuth) * cosf(elevation); ALfloat y = sinf(elevation); ALfloat z = cosf(azimuth) * cosf(elevation); CalcAmbiCoeffs(x, y, z, spread, coeffs); } /** * ScaleAzimuthFront * * Scales the given azimuth toward the side (+/- pi/2 radians) for positions in * front. */ inline float ScaleAzimuthFront(float azimuth, float scale) { ALfloat sign = copysignf(1.0f, azimuth); if(!(fabsf(azimuth) > F_PI_2)) return minf(fabsf(azimuth) * scale, F_PI_2) * sign; return azimuth; } void ComputePanningGainsMC(const ChannelConfig *chancoeffs, ALsizei numchans, ALsizei numcoeffs, const ALfloat*restrict coeffs, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]); void ComputePanningGainsBF(const BFChannelConfig *chanmap, ALsizei numchans, const ALfloat*restrict coeffs, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]); /** * ComputePanGains * * Computes panning gains using the given channel decoder coefficients and the * pre-calculated direction or angle coefficients. For B-Format sources, the * coeffs are a 'slice' of a transform matrix for the input channel, used to * scale and orient the sound samples. */ inline void ComputePanGains(const MixParams *dry, const ALfloat*restrict coeffs, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]) { if(dry->CoeffCount > 0) ComputePanningGainsMC(dry->Ambi.Coeffs, dry->NumChannels, dry->CoeffCount, coeffs, ingain, gains); else ComputePanningGainsBF(dry->Ambi.Map, dry->NumChannels, coeffs, ingain, gains); } ALboolean MixSource(struct ALvoice *voice, ALuint SourceID, ALCcontext *Context, ALsizei SamplesToDo); void aluMixData(ALCdevice *device, ALvoid *OutBuffer, ALsizei NumSamples); /* Caller must lock the device, and the mixer must not be running. */ void aluHandleDisconnect(ALCdevice *device, const char *msg, ...) DECL_FORMAT(printf, 2, 3); void UpdateContextProps(ALCcontext *context); extern MixerFunc MixSamples; extern RowMixerFunc MixRowSamples; extern ALfloat ConeScale; extern ALfloat ZScale; extern ALboolean OverrideReverbSpeedOfSound; #ifdef __cplusplus } #endif #endif openal-soft-openal-soft-1.19.1/OpenAL32/Include/bs2b.h000066400000000000000000000055301335774445300221500ustar00rootroot00000000000000/*- * Copyright (c) 2005 Boris Mikhaylov * * Permission is hereby granted, free of charge, to any person obtaining * a copy of this software and associated documentation files (the * "Software"), to deal in the Software without restriction, including * without limitation the rights to use, copy, modify, merge, publish, * distribute, sublicense, and/or sell copies of the Software, and to * permit persons to whom the Software is furnished to do so, subject to * the following conditions: * * The above copyright notice and this permission notice shall be * included in all copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */ #ifndef BS2B_H #define BS2B_H /* Number of crossfeed levels */ #define BS2B_CLEVELS 3 /* Normal crossfeed levels */ #define BS2B_HIGH_CLEVEL 3 #define BS2B_MIDDLE_CLEVEL 2 #define BS2B_LOW_CLEVEL 1 /* Easy crossfeed levels */ #define BS2B_HIGH_ECLEVEL BS2B_HIGH_CLEVEL + BS2B_CLEVELS #define BS2B_MIDDLE_ECLEVEL BS2B_MIDDLE_CLEVEL + BS2B_CLEVELS #define BS2B_LOW_ECLEVEL BS2B_LOW_CLEVEL + BS2B_CLEVELS /* Default crossfeed levels */ #define BS2B_DEFAULT_CLEVEL BS2B_HIGH_ECLEVEL /* Default sample rate (Hz) */ #define BS2B_DEFAULT_SRATE 44100 #ifdef __cplusplus extern "C" { #endif /* __cplusplus */ struct bs2b { int level; /* Crossfeed level */ int srate; /* Sample rate (Hz) */ /* Lowpass IIR filter coefficients */ float a0_lo; float b1_lo; /* Highboost IIR filter coefficients */ float a0_hi; float a1_hi; float b1_hi; /* Buffer of last filtered sample. * [0] - first channel, [1] - second channel */ struct t_last_sample { float asis; float lo; float hi; } last_sample[2]; }; /* Clear buffers and set new coefficients with new crossfeed level and sample * rate values. * level - crossfeed level of *LEVEL values. * srate - sample rate by Hz. */ void bs2b_set_params(struct bs2b *bs2b, int level, int srate); /* Return current crossfeed level value */ int bs2b_get_level(struct bs2b *bs2b); /* Return current sample rate value */ int bs2b_get_srate(struct bs2b *bs2b); /* Clear buffer */ void bs2b_clear(struct bs2b *bs2b); void bs2b_cross_feed(struct bs2b *bs2b, float *restrict Left, float *restrict Right, int SamplesToDo); #ifdef __cplusplus } /* extern "C" */ #endif /* __cplusplus */ #endif /* BS2B_H */ openal-soft-openal-soft-1.19.1/OpenAL32/Include/sample_cvt.h000066400000000000000000000007561335774445300234620ustar00rootroot00000000000000#ifndef SAMPLE_CVT_H #define SAMPLE_CVT_H #include "AL/al.h" #include "alBuffer.h" extern const ALshort muLawDecompressionTable[256]; extern const ALshort aLawDecompressionTable[256]; void Convert_ALshort_ALima4(ALshort *dst, const ALubyte *src, ALsizei numchans, ALsizei len, ALsizei align); void Convert_ALshort_ALmsadpcm(ALshort *dst, const ALubyte *src, ALsizei numchans, ALsizei len, ALsizei align); #endif /* SAMPLE_CVT_H */ openal-soft-openal-soft-1.19.1/OpenAL32/alAuxEffectSlot.c000066400000000000000000000574701335774445300227730ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include "AL/al.h" #include "AL/alc.h" #include "alMain.h" #include "alAuxEffectSlot.h" #include "alError.h" #include "alListener.h" #include "alSource.h" #include "fpu_modes.h" #include "almalloc.h" extern inline void LockEffectSlotList(ALCcontext *context); extern inline void UnlockEffectSlotList(ALCcontext *context); static void AddActiveEffectSlots(const ALuint *slotids, ALsizei count, ALCcontext *context); static void RemoveActiveEffectSlots(const ALuint *slotids, ALsizei count, ALCcontext *context); static const struct { ALenum Type; EffectStateFactory* (*GetFactory)(void); } FactoryList[] = { { AL_EFFECT_NULL, NullStateFactory_getFactory }, { AL_EFFECT_EAXREVERB, ReverbStateFactory_getFactory }, { AL_EFFECT_REVERB, ReverbStateFactory_getFactory }, { AL_EFFECT_AUTOWAH, AutowahStateFactory_getFactory }, { AL_EFFECT_CHORUS, ChorusStateFactory_getFactory }, { AL_EFFECT_COMPRESSOR, CompressorStateFactory_getFactory }, { AL_EFFECT_DISTORTION, DistortionStateFactory_getFactory }, { AL_EFFECT_ECHO, EchoStateFactory_getFactory }, { AL_EFFECT_EQUALIZER, EqualizerStateFactory_getFactory }, { AL_EFFECT_FLANGER, FlangerStateFactory_getFactory }, { AL_EFFECT_FREQUENCY_SHIFTER, FshifterStateFactory_getFactory }, { AL_EFFECT_RING_MODULATOR, ModulatorStateFactory_getFactory }, { AL_EFFECT_PITCH_SHIFTER, PshifterStateFactory_getFactory}, { AL_EFFECT_DEDICATED_DIALOGUE, DedicatedStateFactory_getFactory }, { AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT, DedicatedStateFactory_getFactory } }; static inline EffectStateFactory *getFactoryByType(ALenum type) { size_t i; for(i = 0;i < COUNTOF(FactoryList);i++) { if(FactoryList[i].Type == type) return FactoryList[i].GetFactory(); } return NULL; } static void ALeffectState_IncRef(ALeffectState *state); static inline ALeffectslot *LookupEffectSlot(ALCcontext *context, ALuint id) { id--; if(UNLIKELY(id >= VECTOR_SIZE(context->EffectSlotList))) return NULL; return VECTOR_ELEM(context->EffectSlotList, id); } static inline ALeffect *LookupEffect(ALCdevice *device, ALuint id) { EffectSubList *sublist; ALuint lidx = (id-1) >> 6; ALsizei slidx = (id-1) & 0x3f; if(UNLIKELY(lidx >= VECTOR_SIZE(device->EffectList))) return NULL; sublist = &VECTOR_ELEM(device->EffectList, lidx); if(UNLIKELY(sublist->FreeMask & (U64(1)<Effects + slidx; } #define DO_UPDATEPROPS() do { \ if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire)) \ UpdateEffectSlotProps(slot, context); \ else \ ATOMIC_FLAG_CLEAR(&slot->PropsClean, almemory_order_release); \ } while(0) AL_API ALvoid AL_APIENTRY alGenAuxiliaryEffectSlots(ALsizei n, ALuint *effectslots) { ALCdevice *device; ALCcontext *context; ALsizei cur; context = GetContextRef(); if(!context) return; if(!(n >= 0)) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Generating %d effect slots", n); if(n == 0) goto done; LockEffectSlotList(context); device = context->Device; for(cur = 0;cur < n;cur++) { ALeffectslotPtr *iter = VECTOR_BEGIN(context->EffectSlotList); ALeffectslotPtr *end = VECTOR_END(context->EffectSlotList); ALeffectslot *slot = NULL; ALenum err = AL_OUT_OF_MEMORY; for(;iter != end;iter++) { if(!*iter) break; } if(iter == end) { if(device->AuxiliaryEffectSlotMax == VECTOR_SIZE(context->EffectSlotList)) { UnlockEffectSlotList(context); alDeleteAuxiliaryEffectSlots(cur, effectslots); SETERR_GOTO(context, AL_OUT_OF_MEMORY, done, "Exceeding %u auxiliary effect slot limit", device->AuxiliaryEffectSlotMax); } VECTOR_PUSH_BACK(context->EffectSlotList, NULL); iter = &VECTOR_BACK(context->EffectSlotList); } slot = al_calloc(16, sizeof(ALeffectslot)); if(!slot || (err=InitEffectSlot(slot)) != AL_NO_ERROR) { al_free(slot); UnlockEffectSlotList(context); alDeleteAuxiliaryEffectSlots(cur, effectslots); SETERR_GOTO(context, err, done, "Effect slot object allocation failed"); } aluInitEffectPanning(slot); slot->id = (iter - VECTOR_BEGIN(context->EffectSlotList)) + 1; *iter = slot; effectslots[cur] = slot->id; } AddActiveEffectSlots(effectslots, n, context); UnlockEffectSlotList(context); done: ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alDeleteAuxiliaryEffectSlots(ALsizei n, const ALuint *effectslots) { ALCcontext *context; ALeffectslot *slot; ALsizei i; context = GetContextRef(); if(!context) return; LockEffectSlotList(context); if(!(n >= 0)) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Deleting %d effect slots", n); if(n == 0) goto done; for(i = 0;i < n;i++) { if((slot=LookupEffectSlot(context, effectslots[i])) == NULL) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslots[i]); if(ReadRef(&slot->ref) != 0) SETERR_GOTO(context, AL_INVALID_NAME, done, "Deleting in-use effect slot %u", effectslots[i]); } // All effectslots are valid RemoveActiveEffectSlots(effectslots, n, context); for(i = 0;i < n;i++) { if((slot=LookupEffectSlot(context, effectslots[i])) == NULL) continue; VECTOR_ELEM(context->EffectSlotList, effectslots[i]-1) = NULL; DeinitEffectSlot(slot); memset(slot, 0, sizeof(*slot)); al_free(slot); } done: UnlockEffectSlotList(context); ALCcontext_DecRef(context); } AL_API ALboolean AL_APIENTRY alIsAuxiliaryEffectSlot(ALuint effectslot) { ALCcontext *context; ALboolean ret; context = GetContextRef(); if(!context) return AL_FALSE; LockEffectSlotList(context); ret = (LookupEffectSlot(context, effectslot) ? AL_TRUE : AL_FALSE); UnlockEffectSlotList(context); ALCcontext_DecRef(context); return ret; } AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint value) { ALCdevice *device; ALCcontext *context; ALeffectslot *slot; ALeffect *effect = NULL; ALenum err; context = GetContextRef(); if(!context) return; almtx_lock(&context->PropLock); LockEffectSlotList(context); if((slot=LookupEffectSlot(context, effectslot)) == NULL) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot); switch(param) { case AL_EFFECTSLOT_EFFECT: device = context->Device; LockEffectList(device); effect = (value ? LookupEffect(device, value) : NULL); if(!(value == 0 || effect != NULL)) { UnlockEffectList(device); SETERR_GOTO(context, AL_INVALID_VALUE, done, "Invalid effect ID %u", value); } err = InitializeEffect(context, slot, effect); UnlockEffectList(device); if(err != AL_NO_ERROR) SETERR_GOTO(context, err, done, "Effect initialization failed"); break; case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO: if(!(value == AL_TRUE || value == AL_FALSE)) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Effect slot auxiliary send auto out of range"); slot->AuxSendAuto = value; break; default: SETERR_GOTO(context, AL_INVALID_ENUM, done, "Invalid effect slot integer property 0x%04x", param); } DO_UPDATEPROPS(); done: UnlockEffectSlotList(context); almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotiv(ALuint effectslot, ALenum param, const ALint *values) { ALCcontext *context; switch(param) { case AL_EFFECTSLOT_EFFECT: case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO: alAuxiliaryEffectSloti(effectslot, param, values[0]); return; } context = GetContextRef(); if(!context) return; LockEffectSlotList(context); if(LookupEffectSlot(context, effectslot) == NULL) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot); switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid effect slot integer-vector property 0x%04x", param); } done: UnlockEffectSlotList(context); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat value) { ALCcontext *context; ALeffectslot *slot; context = GetContextRef(); if(!context) return; almtx_lock(&context->PropLock); LockEffectSlotList(context); if((slot=LookupEffectSlot(context, effectslot)) == NULL) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot); switch(param) { case AL_EFFECTSLOT_GAIN: if(!(value >= 0.0f && value <= 1.0f)) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Effect slot gain out of range"); slot->Gain = value; break; default: SETERR_GOTO(context, AL_INVALID_ENUM, done, "Invalid effect slot float property 0x%04x", param); } DO_UPDATEPROPS(); done: UnlockEffectSlotList(context); almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotfv(ALuint effectslot, ALenum param, const ALfloat *values) { ALCcontext *context; switch(param) { case AL_EFFECTSLOT_GAIN: alAuxiliaryEffectSlotf(effectslot, param, values[0]); return; } context = GetContextRef(); if(!context) return; LockEffectSlotList(context); if(LookupEffectSlot(context, effectslot) == NULL) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot); switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid effect slot float-vector property 0x%04x", param); } done: UnlockEffectSlotList(context); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint *value) { ALCcontext *context; ALeffectslot *slot; context = GetContextRef(); if(!context) return; LockEffectSlotList(context); if((slot=LookupEffectSlot(context, effectslot)) == NULL) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot); switch(param) { case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO: *value = slot->AuxSendAuto; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid effect slot integer property 0x%04x", param); } done: UnlockEffectSlotList(context); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotiv(ALuint effectslot, ALenum param, ALint *values) { ALCcontext *context; switch(param) { case AL_EFFECTSLOT_EFFECT: case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO: alGetAuxiliaryEffectSloti(effectslot, param, values); return; } context = GetContextRef(); if(!context) return; LockEffectSlotList(context); if(LookupEffectSlot(context, effectslot) == NULL) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot); switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid effect slot integer-vector property 0x%04x", param); } done: UnlockEffectSlotList(context); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat *value) { ALCcontext *context; ALeffectslot *slot; context = GetContextRef(); if(!context) return; LockEffectSlotList(context); if((slot=LookupEffectSlot(context, effectslot)) == NULL) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot); switch(param) { case AL_EFFECTSLOT_GAIN: *value = slot->Gain; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid effect slot float property 0x%04x", param); } done: UnlockEffectSlotList(context); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotfv(ALuint effectslot, ALenum param, ALfloat *values) { ALCcontext *context; switch(param) { case AL_EFFECTSLOT_GAIN: alGetAuxiliaryEffectSlotf(effectslot, param, values); return; } context = GetContextRef(); if(!context) return; LockEffectSlotList(context); if(LookupEffectSlot(context, effectslot) == NULL) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot); switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid effect slot float-vector property 0x%04x", param); } done: UnlockEffectSlotList(context); ALCcontext_DecRef(context); } ALenum InitializeEffect(ALCcontext *Context, ALeffectslot *EffectSlot, ALeffect *effect) { ALCdevice *Device = Context->Device; ALenum newtype = (effect ? effect->type : AL_EFFECT_NULL); struct ALeffectslotProps *props; ALeffectState *State; if(newtype != EffectSlot->Effect.Type) { EffectStateFactory *factory; factory = getFactoryByType(newtype); if(!factory) { ERR("Failed to find factory for effect type 0x%04x\n", newtype); return AL_INVALID_ENUM; } State = EffectStateFactory_create(factory); if(!State) return AL_OUT_OF_MEMORY; START_MIXER_MODE(); almtx_lock(&Device->BackendLock); State->OutBuffer = Device->Dry.Buffer; State->OutChannels = Device->Dry.NumChannels; if(V(State,deviceUpdate)(Device) == AL_FALSE) { almtx_unlock(&Device->BackendLock); LEAVE_MIXER_MODE(); ALeffectState_DecRef(State); return AL_OUT_OF_MEMORY; } almtx_unlock(&Device->BackendLock); END_MIXER_MODE(); if(!effect) { EffectSlot->Effect.Type = AL_EFFECT_NULL; memset(&EffectSlot->Effect.Props, 0, sizeof(EffectSlot->Effect.Props)); } else { EffectSlot->Effect.Type = effect->type; EffectSlot->Effect.Props = effect->Props; } ALeffectState_DecRef(EffectSlot->Effect.State); EffectSlot->Effect.State = State; } else if(effect) EffectSlot->Effect.Props = effect->Props; /* Remove state references from old effect slot property updates. */ props = ATOMIC_LOAD_SEQ(&Context->FreeEffectslotProps); while(props) { if(props->State) ALeffectState_DecRef(props->State); props->State = NULL; props = ATOMIC_LOAD(&props->next, almemory_order_relaxed); } return AL_NO_ERROR; } static void ALeffectState_IncRef(ALeffectState *state) { uint ref; ref = IncrementRef(&state->Ref); TRACEREF("%p increasing refcount to %u\n", state, ref); } void ALeffectState_DecRef(ALeffectState *state) { uint ref; ref = DecrementRef(&state->Ref); TRACEREF("%p decreasing refcount to %u\n", state, ref); if(ref == 0) DELETE_OBJ(state); } void ALeffectState_Construct(ALeffectState *state) { InitRef(&state->Ref, 1); state->OutBuffer = NULL; state->OutChannels = 0; } void ALeffectState_Destruct(ALeffectState *UNUSED(state)) { } static void AddActiveEffectSlots(const ALuint *slotids, ALsizei count, ALCcontext *context) { struct ALeffectslotArray *curarray = ATOMIC_LOAD(&context->ActiveAuxSlots, almemory_order_acquire); struct ALeffectslotArray *newarray = NULL; ALsizei newcount = curarray->count + count; ALCdevice *device = context->Device; ALsizei i, j; /* Insert the new effect slots into the head of the array, followed by the * existing ones. */ newarray = al_calloc(DEF_ALIGN, FAM_SIZE(struct ALeffectslotArray, slot, newcount)); newarray->count = newcount; for(i = 0;i < count;i++) newarray->slot[i] = LookupEffectSlot(context, slotids[i]); for(j = 0;i < newcount;) newarray->slot[i++] = curarray->slot[j++]; /* Remove any duplicates (first instance of each will be kept). */ for(i = 1;i < newcount;i++) { for(j = i;j != 0;) { if(UNLIKELY(newarray->slot[i] == newarray->slot[--j])) { newcount--; for(j = i;j < newcount;j++) newarray->slot[j] = newarray->slot[j+1]; i--; break; } } } /* Reallocate newarray if the new size ended up smaller from duplicate * removal. */ if(UNLIKELY(newcount < newarray->count)) { struct ALeffectslotArray *tmpnewarray = al_calloc(DEF_ALIGN, FAM_SIZE(struct ALeffectslotArray, slot, newcount)); memcpy(tmpnewarray, newarray, FAM_SIZE(struct ALeffectslotArray, slot, newcount)); al_free(newarray); newarray = tmpnewarray; newarray->count = newcount; } curarray = ATOMIC_EXCHANGE_PTR(&context->ActiveAuxSlots, newarray, almemory_order_acq_rel); while((ATOMIC_LOAD(&device->MixCount, almemory_order_acquire)&1)) althrd_yield(); al_free(curarray); } static void RemoveActiveEffectSlots(const ALuint *slotids, ALsizei count, ALCcontext *context) { struct ALeffectslotArray *curarray = ATOMIC_LOAD(&context->ActiveAuxSlots, almemory_order_acquire); struct ALeffectslotArray *newarray = NULL; ALCdevice *device = context->Device; ALsizei i, j; /* Don't shrink the allocated array size since we don't know how many (if * any) of the effect slots to remove are in the array. */ newarray = al_calloc(DEF_ALIGN, FAM_SIZE(struct ALeffectslotArray, slot, curarray->count)); newarray->count = 0; for(i = 0;i < curarray->count;i++) { /* Insert this slot into the new array only if it's not one to remove. */ ALeffectslot *slot = curarray->slot[i]; for(j = count;j != 0;) { if(slot->id == slotids[--j]) goto skip_ins; } newarray->slot[newarray->count++] = slot; skip_ins: ; } /* TODO: Could reallocate newarray now that we know it's needed size. */ curarray = ATOMIC_EXCHANGE_PTR(&context->ActiveAuxSlots, newarray, almemory_order_acq_rel); while((ATOMIC_LOAD(&device->MixCount, almemory_order_acquire)&1)) althrd_yield(); al_free(curarray); } ALenum InitEffectSlot(ALeffectslot *slot) { EffectStateFactory *factory; slot->Effect.Type = AL_EFFECT_NULL; factory = getFactoryByType(AL_EFFECT_NULL); slot->Effect.State = EffectStateFactory_create(factory); if(!slot->Effect.State) return AL_OUT_OF_MEMORY; slot->Gain = 1.0; slot->AuxSendAuto = AL_TRUE; ATOMIC_FLAG_TEST_AND_SET(&slot->PropsClean, almemory_order_relaxed); InitRef(&slot->ref, 0); ATOMIC_INIT(&slot->Update, NULL); slot->Params.Gain = 1.0f; slot->Params.AuxSendAuto = AL_TRUE; ALeffectState_IncRef(slot->Effect.State); slot->Params.EffectState = slot->Effect.State; slot->Params.RoomRolloff = 0.0f; slot->Params.DecayTime = 0.0f; slot->Params.DecayLFRatio = 0.0f; slot->Params.DecayHFRatio = 0.0f; slot->Params.DecayHFLimit = AL_FALSE; slot->Params.AirAbsorptionGainHF = 1.0f; return AL_NO_ERROR; } void DeinitEffectSlot(ALeffectslot *slot) { struct ALeffectslotProps *props; props = ATOMIC_LOAD_SEQ(&slot->Update); if(props) { if(props->State) ALeffectState_DecRef(props->State); TRACE("Freed unapplied AuxiliaryEffectSlot update %p\n", props); al_free(props); } ALeffectState_DecRef(slot->Effect.State); if(slot->Params.EffectState) ALeffectState_DecRef(slot->Params.EffectState); } void UpdateEffectSlotProps(ALeffectslot *slot, ALCcontext *context) { struct ALeffectslotProps *props; ALeffectState *oldstate; /* Get an unused property container, or allocate a new one as needed. */ props = ATOMIC_LOAD(&context->FreeEffectslotProps, almemory_order_relaxed); if(!props) props = al_calloc(16, sizeof(*props)); else { struct ALeffectslotProps *next; do { next = ATOMIC_LOAD(&props->next, almemory_order_relaxed); } while(ATOMIC_COMPARE_EXCHANGE_PTR_WEAK(&context->FreeEffectslotProps, &props, next, almemory_order_seq_cst, almemory_order_acquire) == 0); } /* Copy in current property values. */ props->Gain = slot->Gain; props->AuxSendAuto = slot->AuxSendAuto; props->Type = slot->Effect.Type; props->Props = slot->Effect.Props; /* Swap out any stale effect state object there may be in the container, to * delete it. */ ALeffectState_IncRef(slot->Effect.State); oldstate = props->State; props->State = slot->Effect.State; /* Set the new container for updating internal parameters. */ props = ATOMIC_EXCHANGE_PTR(&slot->Update, props, almemory_order_acq_rel); if(props) { /* If there was an unused update container, put it back in the * freelist. */ if(props->State) ALeffectState_DecRef(props->State); props->State = NULL; ATOMIC_REPLACE_HEAD(struct ALeffectslotProps*, &context->FreeEffectslotProps, props); } if(oldstate) ALeffectState_DecRef(oldstate); } void UpdateAllEffectSlotProps(ALCcontext *context) { struct ALeffectslotArray *auxslots; ALsizei i; LockEffectSlotList(context); auxslots = ATOMIC_LOAD(&context->ActiveAuxSlots, almemory_order_acquire); for(i = 0;i < auxslots->count;i++) { ALeffectslot *slot = auxslots->slot[i]; if(!ATOMIC_FLAG_TEST_AND_SET(&slot->PropsClean, almemory_order_acq_rel)) UpdateEffectSlotProps(slot, context); } UnlockEffectSlotList(context); } ALvoid ReleaseALAuxiliaryEffectSlots(ALCcontext *context) { ALeffectslotPtr *iter = VECTOR_BEGIN(context->EffectSlotList); ALeffectslotPtr *end = VECTOR_END(context->EffectSlotList); size_t leftover = 0; for(;iter != end;iter++) { ALeffectslot *slot = *iter; if(!slot) continue; *iter = NULL; DeinitEffectSlot(slot); memset(slot, 0, sizeof(*slot)); al_free(slot); ++leftover; } if(leftover > 0) WARN("(%p) Deleted "SZFMT" AuxiliaryEffectSlot%s\n", context, leftover, (leftover==1)?"":"s"); } openal-soft-openal-soft-1.19.1/OpenAL32/alBuffer.c000066400000000000000000001250011335774445300214520ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include #ifdef HAVE_MALLOC_H #include #endif #include "alMain.h" #include "alu.h" #include "alError.h" #include "alBuffer.h" #include "sample_cvt.h" extern inline void LockBufferList(ALCdevice *device); extern inline void UnlockBufferList(ALCdevice *device); extern inline ALsizei FrameSizeFromUserFmt(enum UserFmtChannels chans, enum UserFmtType type); extern inline ALsizei FrameSizeFromFmt(enum FmtChannels chans, enum FmtType type); static ALbuffer *AllocBuffer(ALCcontext *context); static void FreeBuffer(ALCdevice *device, ALbuffer *buffer); static const ALchar *NameFromUserFmtType(enum UserFmtType type); static void LoadData(ALCcontext *context, ALbuffer *buffer, ALuint freq, ALsizei size, enum UserFmtChannels SrcChannels, enum UserFmtType SrcType, const ALvoid *data, ALbitfieldSOFT access); static ALboolean DecomposeUserFormat(ALenum format, enum UserFmtChannels *chans, enum UserFmtType *type); static ALsizei SanitizeAlignment(enum UserFmtType type, ALsizei align); static inline ALbuffer *LookupBuffer(ALCdevice *device, ALuint id) { BufferSubList *sublist; ALuint lidx = (id-1) >> 6; ALsizei slidx = (id-1) & 0x3f; if(UNLIKELY(lidx >= VECTOR_SIZE(device->BufferList))) return NULL; sublist = &VECTOR_ELEM(device->BufferList, lidx); if(UNLIKELY(sublist->FreeMask & (U64(1)<Buffers + slidx; } #define INVALID_STORAGE_MASK ~(AL_MAP_READ_BIT_SOFT | AL_MAP_WRITE_BIT_SOFT | AL_PRESERVE_DATA_BIT_SOFT | AL_MAP_PERSISTENT_BIT_SOFT) #define MAP_READ_WRITE_FLAGS (AL_MAP_READ_BIT_SOFT | AL_MAP_WRITE_BIT_SOFT) #define INVALID_MAP_FLAGS ~(AL_MAP_READ_BIT_SOFT | AL_MAP_WRITE_BIT_SOFT | AL_MAP_PERSISTENT_BIT_SOFT) AL_API ALvoid AL_APIENTRY alGenBuffers(ALsizei n, ALuint *buffers) { ALCcontext *context; ALsizei cur = 0; context = GetContextRef(); if(!context) return; if(!(n >= 0)) alSetError(context, AL_INVALID_VALUE, "Generating %d buffers", n); else for(cur = 0;cur < n;cur++) { ALbuffer *buffer = AllocBuffer(context); if(!buffer) { alDeleteBuffers(cur, buffers); break; } buffers[cur] = buffer->id; } ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alDeleteBuffers(ALsizei n, const ALuint *buffers) { ALCdevice *device; ALCcontext *context; ALbuffer *ALBuf; ALsizei i; context = GetContextRef(); if(!context) return; device = context->Device; LockBufferList(device); if(UNLIKELY(n < 0)) { alSetError(context, AL_INVALID_VALUE, "Deleting %d buffers", n); goto done; } for(i = 0;i < n;i++) { if(!buffers[i]) continue; /* Check for valid Buffer ID, and make sure it's not in use. */ if((ALBuf=LookupBuffer(device, buffers[i])) == NULL) { alSetError(context, AL_INVALID_NAME, "Invalid buffer ID %u", buffers[i]); goto done; } if(ReadRef(&ALBuf->ref) != 0) { alSetError(context, AL_INVALID_OPERATION, "Deleting in-use buffer %u", buffers[i]); goto done; } } for(i = 0;i < n;i++) { if((ALBuf=LookupBuffer(device, buffers[i])) != NULL) FreeBuffer(device, ALBuf); } done: UnlockBufferList(device); ALCcontext_DecRef(context); } AL_API ALboolean AL_APIENTRY alIsBuffer(ALuint buffer) { ALCcontext *context; ALboolean ret; context = GetContextRef(); if(!context) return AL_FALSE; LockBufferList(context->Device); ret = ((!buffer || LookupBuffer(context->Device, buffer)) ? AL_TRUE : AL_FALSE); UnlockBufferList(context->Device); ALCcontext_DecRef(context); return ret; } AL_API ALvoid AL_APIENTRY alBufferData(ALuint buffer, ALenum format, const ALvoid *data, ALsizei size, ALsizei freq) { alBufferStorageSOFT(buffer, format, data, size, freq, 0); } AL_API void AL_APIENTRY alBufferStorageSOFT(ALuint buffer, ALenum format, const ALvoid *data, ALsizei size, ALsizei freq, ALbitfieldSOFT flags) { enum UserFmtChannels srcchannels = UserFmtMono; enum UserFmtType srctype = UserFmtUByte; ALCdevice *device; ALCcontext *context; ALbuffer *albuf; context = GetContextRef(); if(!context) return; device = context->Device; LockBufferList(device); if(UNLIKELY((albuf=LookupBuffer(device, buffer)) == NULL)) alSetError(context, AL_INVALID_NAME, "Invalid buffer ID %u", buffer); else if(UNLIKELY(size < 0)) alSetError(context, AL_INVALID_VALUE, "Negative storage size %d", size); else if(UNLIKELY(freq < 1)) alSetError(context, AL_INVALID_VALUE, "Invalid sample rate %d", freq); else if(UNLIKELY((flags&INVALID_STORAGE_MASK) != 0)) alSetError(context, AL_INVALID_VALUE, "Invalid storage flags 0x%x", flags&INVALID_STORAGE_MASK); else if(UNLIKELY((flags&AL_MAP_PERSISTENT_BIT_SOFT) && !(flags&MAP_READ_WRITE_FLAGS))) alSetError(context, AL_INVALID_VALUE, "Declaring persistently mapped storage without read or write access"); else if(UNLIKELY(DecomposeUserFormat(format, &srcchannels, &srctype) == AL_FALSE)) alSetError(context, AL_INVALID_ENUM, "Invalid format 0x%04x", format); else LoadData(context, albuf, freq, size, srcchannels, srctype, data, flags); UnlockBufferList(device); ALCcontext_DecRef(context); } AL_API void* AL_APIENTRY alMapBufferSOFT(ALuint buffer, ALsizei offset, ALsizei length, ALbitfieldSOFT access) { void *retval = NULL; ALCdevice *device; ALCcontext *context; ALbuffer *albuf; context = GetContextRef(); if(!context) return retval; device = context->Device; LockBufferList(device); if(UNLIKELY((albuf=LookupBuffer(device, buffer)) == NULL)) alSetError(context, AL_INVALID_NAME, "Invalid buffer ID %u", buffer); else if(UNLIKELY((access&INVALID_MAP_FLAGS) != 0)) alSetError(context, AL_INVALID_VALUE, "Invalid map flags 0x%x", access&INVALID_MAP_FLAGS); else if(UNLIKELY(!(access&MAP_READ_WRITE_FLAGS))) alSetError(context, AL_INVALID_VALUE, "Mapping buffer %u without read or write access", buffer); else { ALbitfieldSOFT unavailable = (albuf->Access^access) & access; if(UNLIKELY(ReadRef(&albuf->ref) != 0 && !(access&AL_MAP_PERSISTENT_BIT_SOFT))) alSetError(context, AL_INVALID_OPERATION, "Mapping in-use buffer %u without persistent mapping", buffer); else if(UNLIKELY(albuf->MappedAccess != 0)) alSetError(context, AL_INVALID_OPERATION, "Mapping already-mapped buffer %u", buffer); else if(UNLIKELY((unavailable&AL_MAP_READ_BIT_SOFT))) alSetError(context, AL_INVALID_VALUE, "Mapping buffer %u for reading without read access", buffer); else if(UNLIKELY((unavailable&AL_MAP_WRITE_BIT_SOFT))) alSetError(context, AL_INVALID_VALUE, "Mapping buffer %u for writing without write access", buffer); else if(UNLIKELY((unavailable&AL_MAP_PERSISTENT_BIT_SOFT))) alSetError(context, AL_INVALID_VALUE, "Mapping buffer %u persistently without persistent access", buffer); else if(UNLIKELY(offset < 0 || offset >= albuf->OriginalSize || length <= 0 || length > albuf->OriginalSize - offset)) alSetError(context, AL_INVALID_VALUE, "Mapping invalid range %d+%d for buffer %u", offset, length, buffer); else { retval = (ALbyte*)albuf->data + offset; albuf->MappedAccess = access; albuf->MappedOffset = offset; albuf->MappedSize = length; } } UnlockBufferList(device); ALCcontext_DecRef(context); return retval; } AL_API void AL_APIENTRY alUnmapBufferSOFT(ALuint buffer) { ALCdevice *device; ALCcontext *context; ALbuffer *albuf; context = GetContextRef(); if(!context) return; device = context->Device; LockBufferList(device); if((albuf=LookupBuffer(device, buffer)) == NULL) alSetError(context, AL_INVALID_NAME, "Invalid buffer ID %u", buffer); else if(albuf->MappedAccess == 0) alSetError(context, AL_INVALID_OPERATION, "Unmapping unmapped buffer %u", buffer); else { albuf->MappedAccess = 0; albuf->MappedOffset = 0; albuf->MappedSize = 0; } UnlockBufferList(device); ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alFlushMappedBufferSOFT(ALuint buffer, ALsizei offset, ALsizei length) { ALCdevice *device; ALCcontext *context; ALbuffer *albuf; context = GetContextRef(); if(!context) return; device = context->Device; LockBufferList(device); if(UNLIKELY((albuf=LookupBuffer(device, buffer)) == NULL)) alSetError(context, AL_INVALID_NAME, "Invalid buffer ID %u", buffer); else if(UNLIKELY(!(albuf->MappedAccess&AL_MAP_WRITE_BIT_SOFT))) alSetError(context, AL_INVALID_OPERATION, "Flushing buffer %u while not mapped for writing", buffer); else if(UNLIKELY(offset < albuf->MappedOffset || offset >= albuf->MappedOffset+albuf->MappedSize || length <= 0 || length > albuf->MappedOffset+albuf->MappedSize-offset)) alSetError(context, AL_INVALID_VALUE, "Flushing invalid range %d+%d on buffer %u", offset, length, buffer); else { /* FIXME: Need to use some method of double-buffering for the mixer and * app to hold separate memory, which can be safely transfered * asynchronously. Currently we just say the app shouldn't write where * OpenAL's reading, and hope for the best... */ ATOMIC_THREAD_FENCE(almemory_order_seq_cst); } UnlockBufferList(device); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alBufferSubDataSOFT(ALuint buffer, ALenum format, const ALvoid *data, ALsizei offset, ALsizei length) { enum UserFmtChannels srcchannels = UserFmtMono; enum UserFmtType srctype = UserFmtUByte; ALCdevice *device; ALCcontext *context; ALbuffer *albuf; context = GetContextRef(); if(!context) return; device = context->Device; LockBufferList(device); if(UNLIKELY((albuf=LookupBuffer(device, buffer)) == NULL)) alSetError(context, AL_INVALID_NAME, "Invalid buffer ID %u", buffer); else if(UNLIKELY(DecomposeUserFormat(format, &srcchannels, &srctype) == AL_FALSE)) alSetError(context, AL_INVALID_ENUM, "Invalid format 0x%04x", format); else { ALsizei unpack_align, align; ALsizei byte_align; ALsizei frame_size; ALsizei num_chans; void *dst; unpack_align = ATOMIC_LOAD_SEQ(&albuf->UnpackAlign); align = SanitizeAlignment(srctype, unpack_align); if(UNLIKELY(align < 1)) alSetError(context, AL_INVALID_VALUE, "Invalid unpack alignment %d", unpack_align); else if(UNLIKELY((long)srcchannels != (long)albuf->FmtChannels || srctype != albuf->OriginalType)) alSetError(context, AL_INVALID_ENUM, "Unpacking data with mismatched format"); else if(UNLIKELY(align != albuf->OriginalAlign)) alSetError(context, AL_INVALID_VALUE, "Unpacking data with alignment %u does not match original alignment %u", align, albuf->OriginalAlign); else if(UNLIKELY(albuf->MappedAccess != 0)) alSetError(context, AL_INVALID_OPERATION, "Unpacking data into mapped buffer %u", buffer); else { num_chans = ChannelsFromFmt(albuf->FmtChannels); frame_size = num_chans * BytesFromFmt(albuf->FmtType); if(albuf->OriginalType == UserFmtIMA4) byte_align = ((align-1)/2 + 4) * num_chans; else if(albuf->OriginalType == UserFmtMSADPCM) byte_align = ((align-2)/2 + 7) * num_chans; else byte_align = align * frame_size; if(UNLIKELY(offset < 0 || length < 0 || offset > albuf->OriginalSize || length > albuf->OriginalSize-offset)) alSetError(context, AL_INVALID_VALUE, "Invalid data sub-range %d+%d on buffer %u", offset, length, buffer); else if(UNLIKELY((offset%byte_align) != 0)) alSetError(context, AL_INVALID_VALUE, "Sub-range offset %d is not a multiple of frame size %d (%d unpack alignment)", offset, byte_align, align); else if(UNLIKELY((length%byte_align) != 0)) alSetError(context, AL_INVALID_VALUE, "Sub-range length %d is not a multiple of frame size %d (%d unpack alignment)", length, byte_align, align); else { /* offset -> byte offset, length -> sample count */ offset = offset/byte_align * align * frame_size; length = length/byte_align * align; dst = (ALbyte*)albuf->data + offset; if(srctype == UserFmtIMA4 && albuf->FmtType == FmtShort) Convert_ALshort_ALima4(dst, data, num_chans, length, align); else if(srctype == UserFmtMSADPCM && albuf->FmtType == FmtShort) Convert_ALshort_ALmsadpcm(dst, data, num_chans, length, align); else { assert((long)srctype == (long)albuf->FmtType); memcpy(dst, data, length*frame_size); } } } } UnlockBufferList(device); ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alBufferSamplesSOFT(ALuint UNUSED(buffer), ALuint UNUSED(samplerate), ALenum UNUSED(internalformat), ALsizei UNUSED(samples), ALenum UNUSED(channels), ALenum UNUSED(type), const ALvoid *UNUSED(data)) { ALCcontext *context; context = GetContextRef(); if(!context) return; alSetError(context, AL_INVALID_OPERATION, "alBufferSamplesSOFT not supported"); ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alBufferSubSamplesSOFT(ALuint UNUSED(buffer), ALsizei UNUSED(offset), ALsizei UNUSED(samples), ALenum UNUSED(channels), ALenum UNUSED(type), const ALvoid *UNUSED(data)) { ALCcontext *context; context = GetContextRef(); if(!context) return; alSetError(context, AL_INVALID_OPERATION, "alBufferSubSamplesSOFT not supported"); ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alGetBufferSamplesSOFT(ALuint UNUSED(buffer), ALsizei UNUSED(offset), ALsizei UNUSED(samples), ALenum UNUSED(channels), ALenum UNUSED(type), ALvoid *UNUSED(data)) { ALCcontext *context; context = GetContextRef(); if(!context) return; alSetError(context, AL_INVALID_OPERATION, "alGetBufferSamplesSOFT not supported"); ALCcontext_DecRef(context); } AL_API ALboolean AL_APIENTRY alIsBufferFormatSupportedSOFT(ALenum UNUSED(format)) { ALCcontext *context; context = GetContextRef(); if(!context) return AL_FALSE; alSetError(context, AL_INVALID_OPERATION, "alIsBufferFormatSupportedSOFT not supported"); ALCcontext_DecRef(context); return AL_FALSE; } AL_API void AL_APIENTRY alBufferf(ALuint buffer, ALenum param, ALfloat UNUSED(value)) { ALCdevice *device; ALCcontext *context; context = GetContextRef(); if(!context) return; device = context->Device; LockBufferList(device); if(UNLIKELY(LookupBuffer(device, buffer) == NULL)) alSetError(context, AL_INVALID_NAME, "Invalid buffer ID %u", buffer); else switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid buffer float property 0x%04x", param); } UnlockBufferList(device); ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alBuffer3f(ALuint buffer, ALenum param, ALfloat UNUSED(value1), ALfloat UNUSED(value2), ALfloat UNUSED(value3)) { ALCdevice *device; ALCcontext *context; context = GetContextRef(); if(!context) return; device = context->Device; LockBufferList(device); if(UNLIKELY(LookupBuffer(device, buffer) == NULL)) alSetError(context, AL_INVALID_NAME, "Invalid buffer ID %u", buffer); else switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid buffer 3-float property 0x%04x", param); } UnlockBufferList(device); ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alBufferfv(ALuint buffer, ALenum param, const ALfloat *values) { ALCdevice *device; ALCcontext *context; context = GetContextRef(); if(!context) return; device = context->Device; LockBufferList(device); if(UNLIKELY(LookupBuffer(device, buffer) == NULL)) alSetError(context, AL_INVALID_NAME, "Invalid buffer ID %u", buffer); else if(UNLIKELY(!values)) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); else switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid buffer float-vector property 0x%04x", param); } UnlockBufferList(device); ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alBufferi(ALuint buffer, ALenum param, ALint value) { ALCdevice *device; ALCcontext *context; ALbuffer *albuf; context = GetContextRef(); if(!context) return; device = context->Device; LockBufferList(device); if(UNLIKELY((albuf=LookupBuffer(device, buffer)) == NULL)) alSetError(context, AL_INVALID_NAME, "Invalid buffer ID %u", buffer); else switch(param) { case AL_UNPACK_BLOCK_ALIGNMENT_SOFT: if(UNLIKELY(value < 0)) alSetError(context, AL_INVALID_VALUE, "Invalid unpack block alignment %d", value); else ATOMIC_STORE_SEQ(&albuf->UnpackAlign, value); break; case AL_PACK_BLOCK_ALIGNMENT_SOFT: if(UNLIKELY(value < 0)) alSetError(context, AL_INVALID_VALUE, "Invalid pack block alignment %d", value); else ATOMIC_STORE_SEQ(&albuf->PackAlign, value); break; default: alSetError(context, AL_INVALID_ENUM, "Invalid buffer integer property 0x%04x", param); } UnlockBufferList(device); ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alBuffer3i(ALuint buffer, ALenum param, ALint UNUSED(value1), ALint UNUSED(value2), ALint UNUSED(value3)) { ALCdevice *device; ALCcontext *context; context = GetContextRef(); if(!context) return; device = context->Device; LockBufferList(device); if(UNLIKELY(LookupBuffer(device, buffer) == NULL)) alSetError(context, AL_INVALID_NAME, "Invalid buffer ID %u", buffer); else switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid buffer 3-integer property 0x%04x", param); } UnlockBufferList(device); ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alBufferiv(ALuint buffer, ALenum param, const ALint *values) { ALCdevice *device; ALCcontext *context; ALbuffer *albuf; if(values) { switch(param) { case AL_UNPACK_BLOCK_ALIGNMENT_SOFT: case AL_PACK_BLOCK_ALIGNMENT_SOFT: alBufferi(buffer, param, values[0]); return; } } context = GetContextRef(); if(!context) return; device = context->Device; LockBufferList(device); if(UNLIKELY((albuf=LookupBuffer(device, buffer)) == NULL)) alSetError(context, AL_INVALID_NAME, "Invalid buffer ID %u", buffer); else if(UNLIKELY(!values)) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); else switch(param) { case AL_LOOP_POINTS_SOFT: if(UNLIKELY(ReadRef(&albuf->ref) != 0)) alSetError(context, AL_INVALID_OPERATION, "Modifying in-use buffer %u's loop points", buffer); else if(UNLIKELY(values[0] >= values[1] || values[0] < 0 || values[1] > albuf->SampleLen)) alSetError(context, AL_INVALID_VALUE, "Invalid loop point range %d -> %d o buffer %u", values[0], values[1], buffer); else { albuf->LoopStart = values[0]; albuf->LoopEnd = values[1]; } break; default: alSetError(context, AL_INVALID_ENUM, "Invalid buffer integer-vector property 0x%04x", param); } UnlockBufferList(device); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alGetBufferf(ALuint buffer, ALenum param, ALfloat *value) { ALCdevice *device; ALCcontext *context; ALbuffer *albuf; context = GetContextRef(); if(!context) return; device = context->Device; LockBufferList(device); if(UNLIKELY((albuf=LookupBuffer(device, buffer)) == NULL)) alSetError(context, AL_INVALID_NAME, "Invalid buffer ID %u", buffer); else if(UNLIKELY(!value)) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); else switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid buffer float property 0x%04x", param); } UnlockBufferList(device); ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alGetBuffer3f(ALuint buffer, ALenum param, ALfloat *value1, ALfloat *value2, ALfloat *value3) { ALCdevice *device; ALCcontext *context; context = GetContextRef(); if(!context) return; device = context->Device; LockBufferList(device); if(UNLIKELY(LookupBuffer(device, buffer) == NULL)) alSetError(context, AL_INVALID_NAME, "Invalid buffer ID %u", buffer); else if(UNLIKELY(!value1 || !value2 || !value3)) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); else switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid buffer 3-float property 0x%04x", param); } UnlockBufferList(device); ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alGetBufferfv(ALuint buffer, ALenum param, ALfloat *values) { ALCdevice *device; ALCcontext *context; switch(param) { case AL_SEC_LENGTH_SOFT: alGetBufferf(buffer, param, values); return; } context = GetContextRef(); if(!context) return; device = context->Device; LockBufferList(device); if(UNLIKELY(LookupBuffer(device, buffer) == NULL)) alSetError(context, AL_INVALID_NAME, "Invalid buffer ID %u", buffer); else if(UNLIKELY(!values)) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); else switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid buffer float-vector property 0x%04x", param); } UnlockBufferList(device); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alGetBufferi(ALuint buffer, ALenum param, ALint *value) { ALCdevice *device; ALCcontext *context; ALbuffer *albuf; context = GetContextRef(); if(!context) return; device = context->Device; LockBufferList(device); if(UNLIKELY((albuf=LookupBuffer(device, buffer)) == NULL)) alSetError(context, AL_INVALID_NAME, "Invalid buffer ID %u", buffer); else if(UNLIKELY(!value)) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); else switch(param) { case AL_FREQUENCY: *value = albuf->Frequency; break; case AL_BITS: *value = BytesFromFmt(albuf->FmtType) * 8; break; case AL_CHANNELS: *value = ChannelsFromFmt(albuf->FmtChannels); break; case AL_SIZE: *value = albuf->SampleLen * FrameSizeFromFmt(albuf->FmtChannels, albuf->FmtType); break; case AL_UNPACK_BLOCK_ALIGNMENT_SOFT: *value = ATOMIC_LOAD_SEQ(&albuf->UnpackAlign); break; case AL_PACK_BLOCK_ALIGNMENT_SOFT: *value = ATOMIC_LOAD_SEQ(&albuf->PackAlign); break; default: alSetError(context, AL_INVALID_ENUM, "Invalid buffer integer property 0x%04x", param); } UnlockBufferList(device); ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alGetBuffer3i(ALuint buffer, ALenum param, ALint *value1, ALint *value2, ALint *value3) { ALCdevice *device; ALCcontext *context; context = GetContextRef(); if(!context) return; device = context->Device; LockBufferList(device); if(UNLIKELY(LookupBuffer(device, buffer) == NULL)) alSetError(context, AL_INVALID_NAME, "Invalid buffer ID %u", buffer); else if(UNLIKELY(!value1 || !value2 || !value3)) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); else switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid buffer 3-integer property 0x%04x", param); } UnlockBufferList(device); ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alGetBufferiv(ALuint buffer, ALenum param, ALint *values) { ALCdevice *device; ALCcontext *context; ALbuffer *albuf; switch(param) { case AL_FREQUENCY: case AL_BITS: case AL_CHANNELS: case AL_SIZE: case AL_INTERNAL_FORMAT_SOFT: case AL_BYTE_LENGTH_SOFT: case AL_SAMPLE_LENGTH_SOFT: case AL_UNPACK_BLOCK_ALIGNMENT_SOFT: case AL_PACK_BLOCK_ALIGNMENT_SOFT: alGetBufferi(buffer, param, values); return; } context = GetContextRef(); if(!context) return; device = context->Device; LockBufferList(device); if(UNLIKELY((albuf=LookupBuffer(device, buffer)) == NULL)) alSetError(context, AL_INVALID_NAME, "Invalid buffer ID %u", buffer); else if(UNLIKELY(!values)) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); else switch(param) { case AL_LOOP_POINTS_SOFT: values[0] = albuf->LoopStart; values[1] = albuf->LoopEnd; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid buffer integer-vector property 0x%04x", param); } UnlockBufferList(device); ALCcontext_DecRef(context); } static const ALchar *NameFromUserFmtType(enum UserFmtType type) { switch(type) { case UserFmtUByte: return "Unsigned Byte"; case UserFmtShort: return "Signed Short"; case UserFmtFloat: return "Float32"; case UserFmtDouble: return "Float64"; case UserFmtMulaw: return "muLaw"; case UserFmtAlaw: return "aLaw"; case UserFmtIMA4: return "IMA4 ADPCM"; case UserFmtMSADPCM: return "MSADPCM"; } return ""; } /* * LoadData * * Loads the specified data into the buffer, using the specified format. */ static void LoadData(ALCcontext *context, ALbuffer *ALBuf, ALuint freq, ALsizei size, enum UserFmtChannels SrcChannels, enum UserFmtType SrcType, const ALvoid *data, ALbitfieldSOFT access) { enum FmtChannels DstChannels = FmtMono; enum FmtType DstType = FmtUByte; ALsizei NumChannels, FrameSize; ALsizei SrcByteAlign; ALsizei unpackalign; ALsizei newsize; ALsizei frames; ALsizei align; if(UNLIKELY(ReadRef(&ALBuf->ref) != 0 || ALBuf->MappedAccess != 0)) SETERR_RETURN(context, AL_INVALID_OPERATION,, "Modifying storage for in-use buffer %u", ALBuf->id); /* Currently no channel configurations need to be converted. */ switch(SrcChannels) { case UserFmtMono: DstChannels = FmtMono; break; case UserFmtStereo: DstChannels = FmtStereo; break; case UserFmtRear: DstChannels = FmtRear; break; case UserFmtQuad: DstChannels = FmtQuad; break; case UserFmtX51: DstChannels = FmtX51; break; case UserFmtX61: DstChannels = FmtX61; break; case UserFmtX71: DstChannels = FmtX71; break; case UserFmtBFormat2D: DstChannels = FmtBFormat2D; break; case UserFmtBFormat3D: DstChannels = FmtBFormat3D; break; } if(UNLIKELY((long)SrcChannels != (long)DstChannels)) SETERR_RETURN(context, AL_INVALID_ENUM,, "Invalid format"); /* IMA4 and MSADPCM convert to 16-bit short. */ switch(SrcType) { case UserFmtUByte: DstType = FmtUByte; break; case UserFmtShort: DstType = FmtShort; break; case UserFmtFloat: DstType = FmtFloat; break; case UserFmtDouble: DstType = FmtDouble; break; case UserFmtAlaw: DstType = FmtAlaw; break; case UserFmtMulaw: DstType = FmtMulaw; break; case UserFmtIMA4: DstType = FmtShort; break; case UserFmtMSADPCM: DstType = FmtShort; break; } /* TODO: Currently we can only map samples when they're not converted. To * allow it would need some kind of double-buffering to hold onto a copy of * the original data. */ if((access&MAP_READ_WRITE_FLAGS)) { if(UNLIKELY((long)SrcType != (long)DstType)) SETERR_RETURN(context, AL_INVALID_VALUE,, "%s samples cannot be mapped", NameFromUserFmtType(SrcType)); } unpackalign = ATOMIC_LOAD_SEQ(&ALBuf->UnpackAlign); if(UNLIKELY((align=SanitizeAlignment(SrcType, unpackalign)) < 1)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Invalid unpack alignment %d for %s samples", unpackalign, NameFromUserFmtType(SrcType)); if((access&AL_PRESERVE_DATA_BIT_SOFT)) { /* Can only preserve data with the same format and alignment. */ if(UNLIKELY(ALBuf->FmtChannels != DstChannels || ALBuf->OriginalType != SrcType)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Preserving data of mismatched format"); if(UNLIKELY(ALBuf->OriginalAlign != align)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Preserving data of mismatched alignment"); } /* Convert the input/source size in bytes to sample frames using the unpack * block alignment. */ if(SrcType == UserFmtIMA4) SrcByteAlign = ((align-1)/2 + 4) * ChannelsFromUserFmt(SrcChannels); else if(SrcType == UserFmtMSADPCM) SrcByteAlign = ((align-2)/2 + 7) * ChannelsFromUserFmt(SrcChannels); else SrcByteAlign = align * FrameSizeFromUserFmt(SrcChannels, SrcType); if(UNLIKELY((size%SrcByteAlign) != 0)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Data size %d is not a multiple of frame size %d (%d unpack alignment)", size, SrcByteAlign, align); if(UNLIKELY(size / SrcByteAlign > INT_MAX / align)) SETERR_RETURN(context, AL_OUT_OF_MEMORY,, "Buffer size overflow, %d blocks x %d samples per block", size/SrcByteAlign, align); frames = size / SrcByteAlign * align; /* Convert the sample frames to the number of bytes needed for internal * storage. */ NumChannels = ChannelsFromFmt(DstChannels); FrameSize = NumChannels * BytesFromFmt(DstType); if(UNLIKELY(frames > INT_MAX/FrameSize)) SETERR_RETURN(context, AL_OUT_OF_MEMORY,, "Buffer size overflow, %d frames x %d bytes per frame", frames, FrameSize); newsize = frames*FrameSize; /* Round up to the next 16-byte multiple. This could reallocate only when * increasing or the new size is less than half the current, but then the * buffer's AL_SIZE would not be very reliable for accounting buffer memory * usage, and reporting the real size could cause problems for apps that * use AL_SIZE to try to get the buffer's play length. */ if(LIKELY(newsize <= INT_MAX-15)) newsize = (newsize+15) & ~0xf; if(newsize != ALBuf->BytesAlloc) { void *temp = al_malloc(16, (size_t)newsize); if(UNLIKELY(!temp && newsize)) SETERR_RETURN(context, AL_OUT_OF_MEMORY,, "Failed to allocate %d bytes of storage", newsize); if((access&AL_PRESERVE_DATA_BIT_SOFT)) { ALsizei tocopy = mini(newsize, ALBuf->BytesAlloc); if(tocopy > 0) memcpy(temp, ALBuf->data, tocopy); } al_free(ALBuf->data); ALBuf->data = temp; ALBuf->BytesAlloc = newsize; } if(SrcType == UserFmtIMA4) { assert(DstType == FmtShort); if(data != NULL && ALBuf->data != NULL) Convert_ALshort_ALima4(ALBuf->data, data, NumChannels, frames, align); ALBuf->OriginalAlign = align; } else if(SrcType == UserFmtMSADPCM) { assert(DstType == FmtShort); if(data != NULL && ALBuf->data != NULL) Convert_ALshort_ALmsadpcm(ALBuf->data, data, NumChannels, frames, align); ALBuf->OriginalAlign = align; } else { assert((long)SrcType == (long)DstType); if(data != NULL && ALBuf->data != NULL) memcpy(ALBuf->data, data, frames*FrameSize); ALBuf->OriginalAlign = 1; } ALBuf->OriginalSize = size; ALBuf->OriginalType = SrcType; ALBuf->Frequency = freq; ALBuf->FmtChannels = DstChannels; ALBuf->FmtType = DstType; ALBuf->Access = access; ALBuf->SampleLen = frames; ALBuf->LoopStart = 0; ALBuf->LoopEnd = ALBuf->SampleLen; } ALsizei BytesFromUserFmt(enum UserFmtType type) { switch(type) { case UserFmtUByte: return sizeof(ALubyte); case UserFmtShort: return sizeof(ALshort); case UserFmtFloat: return sizeof(ALfloat); case UserFmtDouble: return sizeof(ALdouble); case UserFmtMulaw: return sizeof(ALubyte); case UserFmtAlaw: return sizeof(ALubyte); case UserFmtIMA4: break; /* not handled here */ case UserFmtMSADPCM: break; /* not handled here */ } return 0; } ALsizei ChannelsFromUserFmt(enum UserFmtChannels chans) { switch(chans) { case UserFmtMono: return 1; case UserFmtStereo: return 2; case UserFmtRear: return 2; case UserFmtQuad: return 4; case UserFmtX51: return 6; case UserFmtX61: return 7; case UserFmtX71: return 8; case UserFmtBFormat2D: return 3; case UserFmtBFormat3D: return 4; } return 0; } static ALboolean DecomposeUserFormat(ALenum format, enum UserFmtChannels *chans, enum UserFmtType *type) { static const struct { ALenum format; enum UserFmtChannels channels; enum UserFmtType type; } list[] = { { AL_FORMAT_MONO8, UserFmtMono, UserFmtUByte }, { AL_FORMAT_MONO16, UserFmtMono, UserFmtShort }, { AL_FORMAT_MONO_FLOAT32, UserFmtMono, UserFmtFloat }, { AL_FORMAT_MONO_DOUBLE_EXT, UserFmtMono, UserFmtDouble }, { AL_FORMAT_MONO_IMA4, UserFmtMono, UserFmtIMA4 }, { AL_FORMAT_MONO_MSADPCM_SOFT, UserFmtMono, UserFmtMSADPCM }, { AL_FORMAT_MONO_MULAW, UserFmtMono, UserFmtMulaw }, { AL_FORMAT_MONO_ALAW_EXT, UserFmtMono, UserFmtAlaw }, { AL_FORMAT_STEREO8, UserFmtStereo, UserFmtUByte }, { AL_FORMAT_STEREO16, UserFmtStereo, UserFmtShort }, { AL_FORMAT_STEREO_FLOAT32, UserFmtStereo, UserFmtFloat }, { AL_FORMAT_STEREO_DOUBLE_EXT, UserFmtStereo, UserFmtDouble }, { AL_FORMAT_STEREO_IMA4, UserFmtStereo, UserFmtIMA4 }, { AL_FORMAT_STEREO_MSADPCM_SOFT, UserFmtStereo, UserFmtMSADPCM }, { AL_FORMAT_STEREO_MULAW, UserFmtStereo, UserFmtMulaw }, { AL_FORMAT_STEREO_ALAW_EXT, UserFmtStereo, UserFmtAlaw }, { AL_FORMAT_REAR8, UserFmtRear, UserFmtUByte }, { AL_FORMAT_REAR16, UserFmtRear, UserFmtShort }, { AL_FORMAT_REAR32, UserFmtRear, UserFmtFloat }, { AL_FORMAT_REAR_MULAW, UserFmtRear, UserFmtMulaw }, { AL_FORMAT_QUAD8_LOKI, UserFmtQuad, UserFmtUByte }, { AL_FORMAT_QUAD16_LOKI, UserFmtQuad, UserFmtShort }, { AL_FORMAT_QUAD8, UserFmtQuad, UserFmtUByte }, { AL_FORMAT_QUAD16, UserFmtQuad, UserFmtShort }, { AL_FORMAT_QUAD32, UserFmtQuad, UserFmtFloat }, { AL_FORMAT_QUAD_MULAW, UserFmtQuad, UserFmtMulaw }, { AL_FORMAT_51CHN8, UserFmtX51, UserFmtUByte }, { AL_FORMAT_51CHN16, UserFmtX51, UserFmtShort }, { AL_FORMAT_51CHN32, UserFmtX51, UserFmtFloat }, { AL_FORMAT_51CHN_MULAW, UserFmtX51, UserFmtMulaw }, { AL_FORMAT_61CHN8, UserFmtX61, UserFmtUByte }, { AL_FORMAT_61CHN16, UserFmtX61, UserFmtShort }, { AL_FORMAT_61CHN32, UserFmtX61, UserFmtFloat }, { AL_FORMAT_61CHN_MULAW, UserFmtX61, UserFmtMulaw }, { AL_FORMAT_71CHN8, UserFmtX71, UserFmtUByte }, { AL_FORMAT_71CHN16, UserFmtX71, UserFmtShort }, { AL_FORMAT_71CHN32, UserFmtX71, UserFmtFloat }, { AL_FORMAT_71CHN_MULAW, UserFmtX71, UserFmtMulaw }, { AL_FORMAT_BFORMAT2D_8, UserFmtBFormat2D, UserFmtUByte }, { AL_FORMAT_BFORMAT2D_16, UserFmtBFormat2D, UserFmtShort }, { AL_FORMAT_BFORMAT2D_FLOAT32, UserFmtBFormat2D, UserFmtFloat }, { AL_FORMAT_BFORMAT2D_MULAW, UserFmtBFormat2D, UserFmtMulaw }, { AL_FORMAT_BFORMAT3D_8, UserFmtBFormat3D, UserFmtUByte }, { AL_FORMAT_BFORMAT3D_16, UserFmtBFormat3D, UserFmtShort }, { AL_FORMAT_BFORMAT3D_FLOAT32, UserFmtBFormat3D, UserFmtFloat }, { AL_FORMAT_BFORMAT3D_MULAW, UserFmtBFormat3D, UserFmtMulaw }, }; ALuint i; for(i = 0;i < COUNTOF(list);i++) { if(list[i].format == format) { *chans = list[i].channels; *type = list[i].type; return AL_TRUE; } } return AL_FALSE; } ALsizei BytesFromFmt(enum FmtType type) { switch(type) { case FmtUByte: return sizeof(ALubyte); case FmtShort: return sizeof(ALshort); case FmtFloat: return sizeof(ALfloat); case FmtDouble: return sizeof(ALdouble); case FmtMulaw: return sizeof(ALubyte); case FmtAlaw: return sizeof(ALubyte); } return 0; } ALsizei ChannelsFromFmt(enum FmtChannels chans) { switch(chans) { case FmtMono: return 1; case FmtStereo: return 2; case FmtRear: return 2; case FmtQuad: return 4; case FmtX51: return 6; case FmtX61: return 7; case FmtX71: return 8; case FmtBFormat2D: return 3; case FmtBFormat3D: return 4; } return 0; } static ALsizei SanitizeAlignment(enum UserFmtType type, ALsizei align) { if(align < 0) return 0; if(align == 0) { if(type == UserFmtIMA4) { /* Here is where things vary: * nVidia and Apple use 64+1 sample frames per block -> block_size=36 bytes per channel * Most PC sound software uses 2040+1 sample frames per block -> block_size=1024 bytes per channel */ return 65; } if(type == UserFmtMSADPCM) return 64; return 1; } if(type == UserFmtIMA4) { /* IMA4 block alignment must be a multiple of 8, plus 1. */ if((align&7) == 1) return align; return 0; } if(type == UserFmtMSADPCM) { /* MSADPCM block alignment must be a multiple of 2. */ if((align&1) == 0) return align; return 0; } return align; } static ALbuffer *AllocBuffer(ALCcontext *context) { ALCdevice *device = context->Device; BufferSubList *sublist, *subend; ALbuffer *buffer = NULL; ALsizei lidx = 0; ALsizei slidx; almtx_lock(&device->BufferLock); sublist = VECTOR_BEGIN(device->BufferList); subend = VECTOR_END(device->BufferList); for(;sublist != subend;++sublist) { if(sublist->FreeMask) { slidx = CTZ64(sublist->FreeMask); buffer = sublist->Buffers + slidx; break; } ++lidx; } if(UNLIKELY(!buffer)) { const BufferSubList empty_sublist = { 0, NULL }; /* Don't allocate so many list entries that the 32-bit ID could * overflow... */ if(UNLIKELY(VECTOR_SIZE(device->BufferList) >= 1<<25)) { almtx_unlock(&device->BufferLock); alSetError(context, AL_OUT_OF_MEMORY, "Too many buffers allocated"); return NULL; } lidx = (ALsizei)VECTOR_SIZE(device->BufferList); VECTOR_PUSH_BACK(device->BufferList, empty_sublist); sublist = &VECTOR_BACK(device->BufferList); sublist->FreeMask = ~U64(0); sublist->Buffers = al_calloc(16, sizeof(ALbuffer)*64); if(UNLIKELY(!sublist->Buffers)) { VECTOR_POP_BACK(device->BufferList); almtx_unlock(&device->BufferLock); alSetError(context, AL_OUT_OF_MEMORY, "Failed to allocate buffer batch"); return NULL; } slidx = 0; buffer = sublist->Buffers + slidx; } memset(buffer, 0, sizeof(*buffer)); /* Add 1 to avoid buffer ID 0. */ buffer->id = ((lidx<<6) | slidx) + 1; sublist->FreeMask &= ~(U64(1)<BufferLock); return buffer; } static void FreeBuffer(ALCdevice *device, ALbuffer *buffer) { ALuint id = buffer->id - 1; ALsizei lidx = id >> 6; ALsizei slidx = id & 0x3f; al_free(buffer->data); memset(buffer, 0, sizeof(*buffer)); VECTOR_ELEM(device->BufferList, lidx).FreeMask |= U64(1) << slidx; } /* * ReleaseALBuffers() * * INTERNAL: Called to destroy any buffers that still exist on the device */ ALvoid ReleaseALBuffers(ALCdevice *device) { BufferSubList *sublist = VECTOR_BEGIN(device->BufferList); BufferSubList *subend = VECTOR_END(device->BufferList); size_t leftover = 0; for(;sublist != subend;++sublist) { ALuint64 usemask = ~sublist->FreeMask; while(usemask) { ALsizei idx = CTZ64(usemask); ALbuffer *buffer = sublist->Buffers + idx; al_free(buffer->data); memset(buffer, 0, sizeof(*buffer)); ++leftover; usemask &= ~(U64(1) << idx); } sublist->FreeMask = ~usemask; } if(leftover > 0) WARN("(%p) Deleted "SZFMT" Buffer%s\n", device, leftover, (leftover==1)?"":"s"); } openal-soft-openal-soft-1.19.1/OpenAL32/alEffect.c000066400000000000000000000662771335774445300214600ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include "AL/al.h" #include "AL/alc.h" #include "alMain.h" #include "alEffect.h" #include "alError.h" extern inline void LockEffectList(ALCdevice *device); extern inline void UnlockEffectList(ALCdevice *device); extern inline ALboolean IsReverbEffect(ALenum type); const struct EffectList EffectList[EFFECTLIST_SIZE] = { { "eaxreverb", EAXREVERB_EFFECT, AL_EFFECT_EAXREVERB }, { "reverb", REVERB_EFFECT, AL_EFFECT_REVERB }, { "autowah", AUTOWAH_EFFECT, AL_EFFECT_AUTOWAH }, { "chorus", CHORUS_EFFECT, AL_EFFECT_CHORUS }, { "compressor", COMPRESSOR_EFFECT, AL_EFFECT_COMPRESSOR }, { "distortion", DISTORTION_EFFECT, AL_EFFECT_DISTORTION }, { "echo", ECHO_EFFECT, AL_EFFECT_ECHO }, { "equalizer", EQUALIZER_EFFECT, AL_EFFECT_EQUALIZER }, { "flanger", FLANGER_EFFECT, AL_EFFECT_FLANGER }, { "fshifter", FSHIFTER_EFFECT, AL_EFFECT_FREQUENCY_SHIFTER }, { "modulator", MODULATOR_EFFECT, AL_EFFECT_RING_MODULATOR }, { "pshifter", PSHIFTER_EFFECT, AL_EFFECT_PITCH_SHIFTER }, { "dedicated", DEDICATED_EFFECT, AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT }, { "dedicated", DEDICATED_EFFECT, AL_EFFECT_DEDICATED_DIALOGUE }, }; ALboolean DisabledEffects[MAX_EFFECTS]; static ALeffect *AllocEffect(ALCcontext *context); static void FreeEffect(ALCdevice *device, ALeffect *effect); static void InitEffectParams(ALeffect *effect, ALenum type); static inline ALeffect *LookupEffect(ALCdevice *device, ALuint id) { EffectSubList *sublist; ALuint lidx = (id-1) >> 6; ALsizei slidx = (id-1) & 0x3f; if(UNLIKELY(lidx >= VECTOR_SIZE(device->EffectList))) return NULL; sublist = &VECTOR_ELEM(device->EffectList, lidx); if(UNLIKELY(sublist->FreeMask & (U64(1)<Effects + slidx; } AL_API ALvoid AL_APIENTRY alGenEffects(ALsizei n, ALuint *effects) { ALCcontext *context; ALsizei cur; context = GetContextRef(); if(!context) return; if(!(n >= 0)) alSetError(context, AL_INVALID_VALUE, "Generating %d effects", n); else for(cur = 0;cur < n;cur++) { ALeffect *effect = AllocEffect(context); if(!effect) { alDeleteEffects(cur, effects); break; } effects[cur] = effect->id; } ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alDeleteEffects(ALsizei n, const ALuint *effects) { ALCdevice *device; ALCcontext *context; ALeffect *effect; ALsizei i; context = GetContextRef(); if(!context) return; device = context->Device; LockEffectList(device); if(!(n >= 0)) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Deleting %d effects", n); for(i = 0;i < n;i++) { if(effects[i] && LookupEffect(device, effects[i]) == NULL) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect ID %u", effects[i]); } for(i = 0;i < n;i++) { if((effect=LookupEffect(device, effects[i])) != NULL) FreeEffect(device, effect); } done: UnlockEffectList(device); ALCcontext_DecRef(context); } AL_API ALboolean AL_APIENTRY alIsEffect(ALuint effect) { ALCcontext *Context; ALboolean result; Context = GetContextRef(); if(!Context) return AL_FALSE; LockEffectList(Context->Device); result = ((!effect || LookupEffect(Context->Device, effect)) ? AL_TRUE : AL_FALSE); UnlockEffectList(Context->Device); ALCcontext_DecRef(Context); return result; } AL_API ALvoid AL_APIENTRY alEffecti(ALuint effect, ALenum param, ALint value) { ALCcontext *Context; ALCdevice *Device; ALeffect *ALEffect; Context = GetContextRef(); if(!Context) return; Device = Context->Device; LockEffectList(Device); if((ALEffect=LookupEffect(Device, effect)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect); else { if(param == AL_EFFECT_TYPE) { ALboolean isOk = (value == AL_EFFECT_NULL); ALint i; for(i = 0;!isOk && i < EFFECTLIST_SIZE;i++) { if(value == EffectList[i].val && !DisabledEffects[EffectList[i].type]) isOk = AL_TRUE; } if(isOk) InitEffectParams(ALEffect, value); else alSetError(Context, AL_INVALID_VALUE, "Effect type 0x%04x not supported", value); } else { /* Call the appropriate handler */ ALeffect_setParami(ALEffect, Context, param, value); } } UnlockEffectList(Device); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alEffectiv(ALuint effect, ALenum param, const ALint *values) { ALCcontext *Context; ALCdevice *Device; ALeffect *ALEffect; switch(param) { case AL_EFFECT_TYPE: alEffecti(effect, param, values[0]); return; } Context = GetContextRef(); if(!Context) return; Device = Context->Device; LockEffectList(Device); if((ALEffect=LookupEffect(Device, effect)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect); else { /* Call the appropriate handler */ ALeffect_setParamiv(ALEffect, Context, param, values); } UnlockEffectList(Device); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alEffectf(ALuint effect, ALenum param, ALfloat value) { ALCcontext *Context; ALCdevice *Device; ALeffect *ALEffect; Context = GetContextRef(); if(!Context) return; Device = Context->Device; LockEffectList(Device); if((ALEffect=LookupEffect(Device, effect)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect); else { /* Call the appropriate handler */ ALeffect_setParamf(ALEffect, Context, param, value); } UnlockEffectList(Device); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alEffectfv(ALuint effect, ALenum param, const ALfloat *values) { ALCcontext *Context; ALCdevice *Device; ALeffect *ALEffect; Context = GetContextRef(); if(!Context) return; Device = Context->Device; LockEffectList(Device); if((ALEffect=LookupEffect(Device, effect)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect); else { /* Call the appropriate handler */ ALeffect_setParamfv(ALEffect, Context, param, values); } UnlockEffectList(Device); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alGetEffecti(ALuint effect, ALenum param, ALint *value) { ALCcontext *Context; ALCdevice *Device; ALeffect *ALEffect; Context = GetContextRef(); if(!Context) return; Device = Context->Device; LockEffectList(Device); if((ALEffect=LookupEffect(Device, effect)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect); else { if(param == AL_EFFECT_TYPE) *value = ALEffect->type; else { /* Call the appropriate handler */ ALeffect_getParami(ALEffect, Context, param, value); } } UnlockEffectList(Device); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alGetEffectiv(ALuint effect, ALenum param, ALint *values) { ALCcontext *Context; ALCdevice *Device; ALeffect *ALEffect; switch(param) { case AL_EFFECT_TYPE: alGetEffecti(effect, param, values); return; } Context = GetContextRef(); if(!Context) return; Device = Context->Device; LockEffectList(Device); if((ALEffect=LookupEffect(Device, effect)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect); else { /* Call the appropriate handler */ ALeffect_getParamiv(ALEffect, Context, param, values); } UnlockEffectList(Device); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alGetEffectf(ALuint effect, ALenum param, ALfloat *value) { ALCcontext *Context; ALCdevice *Device; ALeffect *ALEffect; Context = GetContextRef(); if(!Context) return; Device = Context->Device; LockEffectList(Device); if((ALEffect=LookupEffect(Device, effect)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect); else { /* Call the appropriate handler */ ALeffect_getParamf(ALEffect, Context, param, value); } UnlockEffectList(Device); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alGetEffectfv(ALuint effect, ALenum param, ALfloat *values) { ALCcontext *Context; ALCdevice *Device; ALeffect *ALEffect; Context = GetContextRef(); if(!Context) return; Device = Context->Device; LockEffectList(Device); if((ALEffect=LookupEffect(Device, effect)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect); else { /* Call the appropriate handler */ ALeffect_getParamfv(ALEffect, Context, param, values); } UnlockEffectList(Device); ALCcontext_DecRef(Context); } void InitEffect(ALeffect *effect) { InitEffectParams(effect, AL_EFFECT_NULL); } static ALeffect *AllocEffect(ALCcontext *context) { ALCdevice *device = context->Device; EffectSubList *sublist, *subend; ALeffect *effect = NULL; ALsizei lidx = 0; ALsizei slidx; almtx_lock(&device->EffectLock); sublist = VECTOR_BEGIN(device->EffectList); subend = VECTOR_END(device->EffectList); for(;sublist != subend;++sublist) { if(sublist->FreeMask) { slidx = CTZ64(sublist->FreeMask); effect = sublist->Effects + slidx; break; } ++lidx; } if(UNLIKELY(!effect)) { const EffectSubList empty_sublist = { 0, NULL }; /* Don't allocate so many list entries that the 32-bit ID could * overflow... */ if(UNLIKELY(VECTOR_SIZE(device->EffectList) >= 1<<25)) { almtx_unlock(&device->EffectLock); alSetError(context, AL_OUT_OF_MEMORY, "Too many effects allocated"); return NULL; } lidx = (ALsizei)VECTOR_SIZE(device->EffectList); VECTOR_PUSH_BACK(device->EffectList, empty_sublist); sublist = &VECTOR_BACK(device->EffectList); sublist->FreeMask = ~U64(0); sublist->Effects = al_calloc(16, sizeof(ALeffect)*64); if(UNLIKELY(!sublist->Effects)) { VECTOR_POP_BACK(device->EffectList); almtx_unlock(&device->EffectLock); alSetError(context, AL_OUT_OF_MEMORY, "Failed to allocate effect batch"); return NULL; } slidx = 0; effect = sublist->Effects + slidx; } memset(effect, 0, sizeof(*effect)); InitEffectParams(effect, AL_EFFECT_NULL); /* Add 1 to avoid effect ID 0. */ effect->id = ((lidx<<6) | slidx) + 1; sublist->FreeMask &= ~(U64(1)<EffectLock); return effect; } static void FreeEffect(ALCdevice *device, ALeffect *effect) { ALuint id = effect->id - 1; ALsizei lidx = id >> 6; ALsizei slidx = id & 0x3f; memset(effect, 0, sizeof(*effect)); VECTOR_ELEM(device->EffectList, lidx).FreeMask |= U64(1) << slidx; } void ReleaseALEffects(ALCdevice *device) { EffectSubList *sublist = VECTOR_BEGIN(device->EffectList); EffectSubList *subend = VECTOR_END(device->EffectList); size_t leftover = 0; for(;sublist != subend;++sublist) { ALuint64 usemask = ~sublist->FreeMask; while(usemask) { ALsizei idx = CTZ64(usemask); ALeffect *effect = sublist->Effects + idx; memset(effect, 0, sizeof(*effect)); ++leftover; usemask &= ~(U64(1) << idx); } sublist->FreeMask = ~usemask; } if(leftover > 0) WARN("(%p) Deleted "SZFMT" Effect%s\n", device, leftover, (leftover==1)?"":"s"); } static void InitEffectParams(ALeffect *effect, ALenum type) { switch(type) { case AL_EFFECT_EAXREVERB: effect->Props.Reverb.Density = AL_EAXREVERB_DEFAULT_DENSITY; effect->Props.Reverb.Diffusion = AL_EAXREVERB_DEFAULT_DIFFUSION; effect->Props.Reverb.Gain = AL_EAXREVERB_DEFAULT_GAIN; effect->Props.Reverb.GainHF = AL_EAXREVERB_DEFAULT_GAINHF; effect->Props.Reverb.GainLF = AL_EAXREVERB_DEFAULT_GAINLF; effect->Props.Reverb.DecayTime = AL_EAXREVERB_DEFAULT_DECAY_TIME; effect->Props.Reverb.DecayHFRatio = AL_EAXREVERB_DEFAULT_DECAY_HFRATIO; effect->Props.Reverb.DecayLFRatio = AL_EAXREVERB_DEFAULT_DECAY_LFRATIO; effect->Props.Reverb.ReflectionsGain = AL_EAXREVERB_DEFAULT_REFLECTIONS_GAIN; effect->Props.Reverb.ReflectionsDelay = AL_EAXREVERB_DEFAULT_REFLECTIONS_DELAY; effect->Props.Reverb.ReflectionsPan[0] = AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ; effect->Props.Reverb.ReflectionsPan[1] = AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ; effect->Props.Reverb.ReflectionsPan[2] = AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ; effect->Props.Reverb.LateReverbGain = AL_EAXREVERB_DEFAULT_LATE_REVERB_GAIN; effect->Props.Reverb.LateReverbDelay = AL_EAXREVERB_DEFAULT_LATE_REVERB_DELAY; effect->Props.Reverb.LateReverbPan[0] = AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ; effect->Props.Reverb.LateReverbPan[1] = AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ; effect->Props.Reverb.LateReverbPan[2] = AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ; effect->Props.Reverb.EchoTime = AL_EAXREVERB_DEFAULT_ECHO_TIME; effect->Props.Reverb.EchoDepth = AL_EAXREVERB_DEFAULT_ECHO_DEPTH; effect->Props.Reverb.ModulationTime = AL_EAXREVERB_DEFAULT_MODULATION_TIME; effect->Props.Reverb.ModulationDepth = AL_EAXREVERB_DEFAULT_MODULATION_DEPTH; effect->Props.Reverb.AirAbsorptionGainHF = AL_EAXREVERB_DEFAULT_AIR_ABSORPTION_GAINHF; effect->Props.Reverb.HFReference = AL_EAXREVERB_DEFAULT_HFREFERENCE; effect->Props.Reverb.LFReference = AL_EAXREVERB_DEFAULT_LFREFERENCE; effect->Props.Reverb.RoomRolloffFactor = AL_EAXREVERB_DEFAULT_ROOM_ROLLOFF_FACTOR; effect->Props.Reverb.DecayHFLimit = AL_EAXREVERB_DEFAULT_DECAY_HFLIMIT; effect->vtab = &ALeaxreverb_vtable; break; case AL_EFFECT_REVERB: effect->Props.Reverb.Density = AL_REVERB_DEFAULT_DENSITY; effect->Props.Reverb.Diffusion = AL_REVERB_DEFAULT_DIFFUSION; effect->Props.Reverb.Gain = AL_REVERB_DEFAULT_GAIN; effect->Props.Reverb.GainHF = AL_REVERB_DEFAULT_GAINHF; effect->Props.Reverb.GainLF = 1.0f; effect->Props.Reverb.DecayTime = AL_REVERB_DEFAULT_DECAY_TIME; effect->Props.Reverb.DecayHFRatio = AL_REVERB_DEFAULT_DECAY_HFRATIO; effect->Props.Reverb.DecayLFRatio = 1.0f; effect->Props.Reverb.ReflectionsGain = AL_REVERB_DEFAULT_REFLECTIONS_GAIN; effect->Props.Reverb.ReflectionsDelay = AL_REVERB_DEFAULT_REFLECTIONS_DELAY; effect->Props.Reverb.ReflectionsPan[0] = 0.0f; effect->Props.Reverb.ReflectionsPan[1] = 0.0f; effect->Props.Reverb.ReflectionsPan[2] = 0.0f; effect->Props.Reverb.LateReverbGain = AL_REVERB_DEFAULT_LATE_REVERB_GAIN; effect->Props.Reverb.LateReverbDelay = AL_REVERB_DEFAULT_LATE_REVERB_DELAY; effect->Props.Reverb.LateReverbPan[0] = 0.0f; effect->Props.Reverb.LateReverbPan[1] = 0.0f; effect->Props.Reverb.LateReverbPan[2] = 0.0f; effect->Props.Reverb.EchoTime = 0.25f; effect->Props.Reverb.EchoDepth = 0.0f; effect->Props.Reverb.ModulationTime = 0.25f; effect->Props.Reverb.ModulationDepth = 0.0f; effect->Props.Reverb.AirAbsorptionGainHF = AL_REVERB_DEFAULT_AIR_ABSORPTION_GAINHF; effect->Props.Reverb.HFReference = 5000.0f; effect->Props.Reverb.LFReference = 250.0f; effect->Props.Reverb.RoomRolloffFactor = AL_REVERB_DEFAULT_ROOM_ROLLOFF_FACTOR; effect->Props.Reverb.DecayHFLimit = AL_REVERB_DEFAULT_DECAY_HFLIMIT; effect->vtab = &ALreverb_vtable; break; case AL_EFFECT_AUTOWAH: effect->Props.Autowah.AttackTime = AL_AUTOWAH_DEFAULT_ATTACK_TIME; effect->Props.Autowah.ReleaseTime = AL_AUTOWAH_DEFAULT_RELEASE_TIME; effect->Props.Autowah.Resonance = AL_AUTOWAH_DEFAULT_RESONANCE; effect->Props.Autowah.PeakGain = AL_AUTOWAH_DEFAULT_PEAK_GAIN; effect->vtab = &ALautowah_vtable; break; case AL_EFFECT_CHORUS: effect->Props.Chorus.Waveform = AL_CHORUS_DEFAULT_WAVEFORM; effect->Props.Chorus.Phase = AL_CHORUS_DEFAULT_PHASE; effect->Props.Chorus.Rate = AL_CHORUS_DEFAULT_RATE; effect->Props.Chorus.Depth = AL_CHORUS_DEFAULT_DEPTH; effect->Props.Chorus.Feedback = AL_CHORUS_DEFAULT_FEEDBACK; effect->Props.Chorus.Delay = AL_CHORUS_DEFAULT_DELAY; effect->vtab = &ALchorus_vtable; break; case AL_EFFECT_COMPRESSOR: effect->Props.Compressor.OnOff = AL_COMPRESSOR_DEFAULT_ONOFF; effect->vtab = &ALcompressor_vtable; break; case AL_EFFECT_DISTORTION: effect->Props.Distortion.Edge = AL_DISTORTION_DEFAULT_EDGE; effect->Props.Distortion.Gain = AL_DISTORTION_DEFAULT_GAIN; effect->Props.Distortion.LowpassCutoff = AL_DISTORTION_DEFAULT_LOWPASS_CUTOFF; effect->Props.Distortion.EQCenter = AL_DISTORTION_DEFAULT_EQCENTER; effect->Props.Distortion.EQBandwidth = AL_DISTORTION_DEFAULT_EQBANDWIDTH; effect->vtab = &ALdistortion_vtable; break; case AL_EFFECT_ECHO: effect->Props.Echo.Delay = AL_ECHO_DEFAULT_DELAY; effect->Props.Echo.LRDelay = AL_ECHO_DEFAULT_LRDELAY; effect->Props.Echo.Damping = AL_ECHO_DEFAULT_DAMPING; effect->Props.Echo.Feedback = AL_ECHO_DEFAULT_FEEDBACK; effect->Props.Echo.Spread = AL_ECHO_DEFAULT_SPREAD; effect->vtab = &ALecho_vtable; break; case AL_EFFECT_EQUALIZER: effect->Props.Equalizer.LowCutoff = AL_EQUALIZER_DEFAULT_LOW_CUTOFF; effect->Props.Equalizer.LowGain = AL_EQUALIZER_DEFAULT_LOW_GAIN; effect->Props.Equalizer.Mid1Center = AL_EQUALIZER_DEFAULT_MID1_CENTER; effect->Props.Equalizer.Mid1Gain = AL_EQUALIZER_DEFAULT_MID1_GAIN; effect->Props.Equalizer.Mid1Width = AL_EQUALIZER_DEFAULT_MID1_WIDTH; effect->Props.Equalizer.Mid2Center = AL_EQUALIZER_DEFAULT_MID2_CENTER; effect->Props.Equalizer.Mid2Gain = AL_EQUALIZER_DEFAULT_MID2_GAIN; effect->Props.Equalizer.Mid2Width = AL_EQUALIZER_DEFAULT_MID2_WIDTH; effect->Props.Equalizer.HighCutoff = AL_EQUALIZER_DEFAULT_HIGH_CUTOFF; effect->Props.Equalizer.HighGain = AL_EQUALIZER_DEFAULT_HIGH_GAIN; effect->vtab = &ALequalizer_vtable; break; case AL_EFFECT_FLANGER: effect->Props.Chorus.Waveform = AL_FLANGER_DEFAULT_WAVEFORM; effect->Props.Chorus.Phase = AL_FLANGER_DEFAULT_PHASE; effect->Props.Chorus.Rate = AL_FLANGER_DEFAULT_RATE; effect->Props.Chorus.Depth = AL_FLANGER_DEFAULT_DEPTH; effect->Props.Chorus.Feedback = AL_FLANGER_DEFAULT_FEEDBACK; effect->Props.Chorus.Delay = AL_FLANGER_DEFAULT_DELAY; effect->vtab = &ALflanger_vtable; break; case AL_EFFECT_FREQUENCY_SHIFTER: effect->Props.Fshifter.Frequency = AL_FREQUENCY_SHIFTER_DEFAULT_FREQUENCY; effect->Props.Fshifter.LeftDirection = AL_FREQUENCY_SHIFTER_DEFAULT_LEFT_DIRECTION; effect->Props.Fshifter.RightDirection = AL_FREQUENCY_SHIFTER_DEFAULT_RIGHT_DIRECTION; effect->vtab = &ALfshifter_vtable; break; case AL_EFFECT_RING_MODULATOR: effect->Props.Modulator.Frequency = AL_RING_MODULATOR_DEFAULT_FREQUENCY; effect->Props.Modulator.HighPassCutoff = AL_RING_MODULATOR_DEFAULT_HIGHPASS_CUTOFF; effect->Props.Modulator.Waveform = AL_RING_MODULATOR_DEFAULT_WAVEFORM; effect->vtab = &ALmodulator_vtable; break; case AL_EFFECT_PITCH_SHIFTER: effect->Props.Pshifter.CoarseTune = AL_PITCH_SHIFTER_DEFAULT_COARSE_TUNE; effect->Props.Pshifter.FineTune = AL_PITCH_SHIFTER_DEFAULT_FINE_TUNE; effect->vtab = &ALpshifter_vtable; break; case AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT: case AL_EFFECT_DEDICATED_DIALOGUE: effect->Props.Dedicated.Gain = 1.0f; effect->vtab = &ALdedicated_vtable; break; default: effect->vtab = &ALnull_vtable; break; } effect->type = type; } #include "AL/efx-presets.h" #define DECL(x) { #x, EFX_REVERB_PRESET_##x } static const struct { const char name[32]; EFXEAXREVERBPROPERTIES props; } reverblist[] = { DECL(GENERIC), DECL(PADDEDCELL), DECL(ROOM), DECL(BATHROOM), DECL(LIVINGROOM), DECL(STONEROOM), DECL(AUDITORIUM), DECL(CONCERTHALL), DECL(CAVE), DECL(ARENA), DECL(HANGAR), DECL(CARPETEDHALLWAY), DECL(HALLWAY), DECL(STONECORRIDOR), DECL(ALLEY), DECL(FOREST), DECL(CITY), DECL(MOUNTAINS), DECL(QUARRY), DECL(PLAIN), DECL(PARKINGLOT), DECL(SEWERPIPE), DECL(UNDERWATER), DECL(DRUGGED), DECL(DIZZY), DECL(PSYCHOTIC), DECL(CASTLE_SMALLROOM), DECL(CASTLE_SHORTPASSAGE), DECL(CASTLE_MEDIUMROOM), DECL(CASTLE_LARGEROOM), DECL(CASTLE_LONGPASSAGE), DECL(CASTLE_HALL), DECL(CASTLE_CUPBOARD), DECL(CASTLE_COURTYARD), DECL(CASTLE_ALCOVE), DECL(FACTORY_SMALLROOM), DECL(FACTORY_SHORTPASSAGE), DECL(FACTORY_MEDIUMROOM), DECL(FACTORY_LARGEROOM), DECL(FACTORY_LONGPASSAGE), DECL(FACTORY_HALL), DECL(FACTORY_CUPBOARD), DECL(FACTORY_COURTYARD), DECL(FACTORY_ALCOVE), DECL(ICEPALACE_SMALLROOM), DECL(ICEPALACE_SHORTPASSAGE), DECL(ICEPALACE_MEDIUMROOM), DECL(ICEPALACE_LARGEROOM), DECL(ICEPALACE_LONGPASSAGE), DECL(ICEPALACE_HALL), DECL(ICEPALACE_CUPBOARD), DECL(ICEPALACE_COURTYARD), DECL(ICEPALACE_ALCOVE), DECL(SPACESTATION_SMALLROOM), DECL(SPACESTATION_SHORTPASSAGE), DECL(SPACESTATION_MEDIUMROOM), DECL(SPACESTATION_LARGEROOM), DECL(SPACESTATION_LONGPASSAGE), DECL(SPACESTATION_HALL), DECL(SPACESTATION_CUPBOARD), DECL(SPACESTATION_ALCOVE), DECL(WOODEN_SMALLROOM), DECL(WOODEN_SHORTPASSAGE), DECL(WOODEN_MEDIUMROOM), DECL(WOODEN_LARGEROOM), DECL(WOODEN_LONGPASSAGE), DECL(WOODEN_HALL), DECL(WOODEN_CUPBOARD), DECL(WOODEN_COURTYARD), DECL(WOODEN_ALCOVE), DECL(SPORT_EMPTYSTADIUM), DECL(SPORT_SQUASHCOURT), DECL(SPORT_SMALLSWIMMINGPOOL), DECL(SPORT_LARGESWIMMINGPOOL), DECL(SPORT_GYMNASIUM), DECL(SPORT_FULLSTADIUM), DECL(SPORT_STADIUMTANNOY), DECL(PREFAB_WORKSHOP), DECL(PREFAB_SCHOOLROOM), DECL(PREFAB_PRACTISEROOM), DECL(PREFAB_OUTHOUSE), DECL(PREFAB_CARAVAN), DECL(DOME_TOMB), DECL(PIPE_SMALL), DECL(DOME_SAINTPAULS), DECL(PIPE_LONGTHIN), DECL(PIPE_LARGE), DECL(PIPE_RESONANT), DECL(OUTDOORS_BACKYARD), DECL(OUTDOORS_ROLLINGPLAINS), DECL(OUTDOORS_DEEPCANYON), DECL(OUTDOORS_CREEK), DECL(OUTDOORS_VALLEY), DECL(MOOD_HEAVEN), DECL(MOOD_HELL), DECL(MOOD_MEMORY), DECL(DRIVING_COMMENTATOR), DECL(DRIVING_PITGARAGE), DECL(DRIVING_INCAR_RACER), DECL(DRIVING_INCAR_SPORTS), DECL(DRIVING_INCAR_LUXURY), DECL(DRIVING_FULLGRANDSTAND), DECL(DRIVING_EMPTYGRANDSTAND), DECL(DRIVING_TUNNEL), DECL(CITY_STREETS), DECL(CITY_SUBWAY), DECL(CITY_MUSEUM), DECL(CITY_LIBRARY), DECL(CITY_UNDERPASS), DECL(CITY_ABANDONED), DECL(DUSTYROOM), DECL(CHAPEL), DECL(SMALLWATERROOM), }; #undef DECL void LoadReverbPreset(const char *name, ALeffect *effect) { size_t i; if(strcasecmp(name, "NONE") == 0) { InitEffectParams(effect, AL_EFFECT_NULL); TRACE("Loading reverb '%s'\n", "NONE"); return; } if(!DisabledEffects[EAXREVERB_EFFECT]) InitEffectParams(effect, AL_EFFECT_EAXREVERB); else if(!DisabledEffects[REVERB_EFFECT]) InitEffectParams(effect, AL_EFFECT_REVERB); else InitEffectParams(effect, AL_EFFECT_NULL); for(i = 0;i < COUNTOF(reverblist);i++) { const EFXEAXREVERBPROPERTIES *props; if(strcasecmp(name, reverblist[i].name) != 0) continue; TRACE("Loading reverb '%s'\n", reverblist[i].name); props = &reverblist[i].props; effect->Props.Reverb.Density = props->flDensity; effect->Props.Reverb.Diffusion = props->flDiffusion; effect->Props.Reverb.Gain = props->flGain; effect->Props.Reverb.GainHF = props->flGainHF; effect->Props.Reverb.GainLF = props->flGainLF; effect->Props.Reverb.DecayTime = props->flDecayTime; effect->Props.Reverb.DecayHFRatio = props->flDecayHFRatio; effect->Props.Reverb.DecayLFRatio = props->flDecayLFRatio; effect->Props.Reverb.ReflectionsGain = props->flReflectionsGain; effect->Props.Reverb.ReflectionsDelay = props->flReflectionsDelay; effect->Props.Reverb.ReflectionsPan[0] = props->flReflectionsPan[0]; effect->Props.Reverb.ReflectionsPan[1] = props->flReflectionsPan[1]; effect->Props.Reverb.ReflectionsPan[2] = props->flReflectionsPan[2]; effect->Props.Reverb.LateReverbGain = props->flLateReverbGain; effect->Props.Reverb.LateReverbDelay = props->flLateReverbDelay; effect->Props.Reverb.LateReverbPan[0] = props->flLateReverbPan[0]; effect->Props.Reverb.LateReverbPan[1] = props->flLateReverbPan[1]; effect->Props.Reverb.LateReverbPan[2] = props->flLateReverbPan[2]; effect->Props.Reverb.EchoTime = props->flEchoTime; effect->Props.Reverb.EchoDepth = props->flEchoDepth; effect->Props.Reverb.ModulationTime = props->flModulationTime; effect->Props.Reverb.ModulationDepth = props->flModulationDepth; effect->Props.Reverb.AirAbsorptionGainHF = props->flAirAbsorptionGainHF; effect->Props.Reverb.HFReference = props->flHFReference; effect->Props.Reverb.LFReference = props->flLFReference; effect->Props.Reverb.RoomRolloffFactor = props->flRoomRolloffFactor; effect->Props.Reverb.DecayHFLimit = props->iDecayHFLimit; return; } WARN("Reverb preset '%s' not found\n", name); } openal-soft-openal-soft-1.19.1/OpenAL32/alError.c000066400000000000000000000063021335774445300213340ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2000 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #ifdef HAVE_WINDOWS_H #define WIN32_LEAN_AND_MEAN #include #endif #include "alMain.h" #include "AL/alc.h" #include "alError.h" ALboolean TrapALError = AL_FALSE; void alSetError(ALCcontext *context, ALenum errorCode, const char *msg, ...) { ALenum curerr = AL_NO_ERROR; char message[1024] = { 0 }; va_list args; int msglen; va_start(args, msg); msglen = vsnprintf(message, sizeof(message), msg, args); va_end(args); if(msglen < 0 || (size_t)msglen >= sizeof(message)) { message[sizeof(message)-1] = 0; msglen = (int)strlen(message); } if(msglen > 0) msg = message; else { msg = ""; msglen = (int)strlen(msg); } WARN("Error generated on context %p, code 0x%04x, \"%s\"\n", context, errorCode, message); if(TrapALError) { #ifdef _WIN32 /* DebugBreak will cause an exception if there is no debugger */ if(IsDebuggerPresent()) DebugBreak(); #elif defined(SIGTRAP) raise(SIGTRAP); #endif } ATOMIC_COMPARE_EXCHANGE_STRONG_SEQ(&context->LastError, &curerr, errorCode); if((ATOMIC_LOAD(&context->EnabledEvts, almemory_order_relaxed)&EventType_Error)) { ALbitfieldSOFT enabledevts; almtx_lock(&context->EventCbLock); enabledevts = ATOMIC_LOAD(&context->EnabledEvts, almemory_order_relaxed); if((enabledevts&EventType_Error) && context->EventCb) (*context->EventCb)(AL_EVENT_TYPE_ERROR_SOFT, 0, errorCode, msglen, msg, context->EventParam); almtx_unlock(&context->EventCbLock); } } AL_API ALenum AL_APIENTRY alGetError(void) { ALCcontext *context; ALenum errorCode; context = GetContextRef(); if(!context) { const ALenum deferror = AL_INVALID_OPERATION; WARN("Querying error state on null context (implicitly 0x%04x)\n", deferror); if(TrapALError) { #ifdef _WIN32 if(IsDebuggerPresent()) DebugBreak(); #elif defined(SIGTRAP) raise(SIGTRAP); #endif } return deferror; } errorCode = ATOMIC_EXCHANGE_SEQ(&context->LastError, AL_NO_ERROR); ALCcontext_DecRef(context); return errorCode; } openal-soft-openal-soft-1.19.1/OpenAL32/alExtension.c000066400000000000000000000043351335774445300222230ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include "alError.h" #include "alMain.h" #include "alFilter.h" #include "alEffect.h" #include "alAuxEffectSlot.h" #include "alSource.h" #include "alBuffer.h" #include "AL/al.h" #include "AL/alc.h" AL_API ALboolean AL_APIENTRY alIsExtensionPresent(const ALchar *extName) { ALboolean ret = AL_FALSE; ALCcontext *context; const char *ptr; size_t len; context = GetContextRef(); if(!context) return AL_FALSE; if(!extName) SETERR_GOTO(context, AL_INVALID_VALUE, done, "NULL pointer"); len = strlen(extName); ptr = context->ExtensionList; while(ptr && *ptr) { if(strncasecmp(ptr, extName, len) == 0 && (ptr[len] == '\0' || isspace(ptr[len]))) { ret = AL_TRUE; break; } if((ptr=strchr(ptr, ' ')) != NULL) { do { ++ptr; } while(isspace(*ptr)); } } done: ALCcontext_DecRef(context); return ret; } AL_API ALvoid* AL_APIENTRY alGetProcAddress(const ALchar *funcName) { if(!funcName) return NULL; return alcGetProcAddress(NULL, funcName); } AL_API ALenum AL_APIENTRY alGetEnumValue(const ALchar *enumName) { if(!enumName) return (ALenum)0; return alcGetEnumValue(NULL, enumName); } openal-soft-openal-soft-1.19.1/OpenAL32/alFilter.c000066400000000000000000000537251335774445300215030ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include "alMain.h" #include "alu.h" #include "alFilter.h" #include "alError.h" #define FILTER_MIN_GAIN 0.0f #define FILTER_MAX_GAIN 4.0f /* +12dB */ extern inline void LockFilterList(ALCdevice *device); extern inline void UnlockFilterList(ALCdevice *device); static ALfilter *AllocFilter(ALCcontext *context); static void FreeFilter(ALCdevice *device, ALfilter *filter); static void InitFilterParams(ALfilter *filter, ALenum type); static inline ALfilter *LookupFilter(ALCdevice *device, ALuint id) { FilterSubList *sublist; ALuint lidx = (id-1) >> 6; ALsizei slidx = (id-1) & 0x3f; if(UNLIKELY(lidx >= VECTOR_SIZE(device->FilterList))) return NULL; sublist = &VECTOR_ELEM(device->FilterList, lidx); if(UNLIKELY(sublist->FreeMask & (U64(1)<Filters + slidx; } AL_API ALvoid AL_APIENTRY alGenFilters(ALsizei n, ALuint *filters) { ALCcontext *context; ALsizei cur = 0; context = GetContextRef(); if(!context) return; if(!(n >= 0)) alSetError(context, AL_INVALID_VALUE, "Generating %d filters", n); else for(cur = 0;cur < n;cur++) { ALfilter *filter = AllocFilter(context); if(!filter) { alDeleteFilters(cur, filters); break; } filters[cur] = filter->id; } ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alDeleteFilters(ALsizei n, const ALuint *filters) { ALCdevice *device; ALCcontext *context; ALfilter *filter; ALsizei i; context = GetContextRef(); if(!context) return; device = context->Device; LockFilterList(device); if(!(n >= 0)) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Deleting %d filters", n); for(i = 0;i < n;i++) { if(filters[i] && LookupFilter(device, filters[i]) == NULL) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid filter ID %u", filters[i]); } for(i = 0;i < n;i++) { if((filter=LookupFilter(device, filters[i])) != NULL) FreeFilter(device, filter); } done: UnlockFilterList(device); ALCcontext_DecRef(context); } AL_API ALboolean AL_APIENTRY alIsFilter(ALuint filter) { ALCcontext *Context; ALboolean result; Context = GetContextRef(); if(!Context) return AL_FALSE; LockFilterList(Context->Device); result = ((!filter || LookupFilter(Context->Device, filter)) ? AL_TRUE : AL_FALSE); UnlockFilterList(Context->Device); ALCcontext_DecRef(Context); return result; } AL_API ALvoid AL_APIENTRY alFilteri(ALuint filter, ALenum param, ALint value) { ALCcontext *Context; ALCdevice *Device; ALfilter *ALFilter; Context = GetContextRef(); if(!Context) return; Device = Context->Device; LockFilterList(Device); if((ALFilter=LookupFilter(Device, filter)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter); else { if(param == AL_FILTER_TYPE) { if(value == AL_FILTER_NULL || value == AL_FILTER_LOWPASS || value == AL_FILTER_HIGHPASS || value == AL_FILTER_BANDPASS) InitFilterParams(ALFilter, value); else alSetError(Context, AL_INVALID_VALUE, "Invalid filter type 0x%04x", value); } else { /* Call the appropriate handler */ ALfilter_setParami(ALFilter, Context, param, value); } } UnlockFilterList(Device); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alFilteriv(ALuint filter, ALenum param, const ALint *values) { ALCcontext *Context; ALCdevice *Device; ALfilter *ALFilter; switch(param) { case AL_FILTER_TYPE: alFilteri(filter, param, values[0]); return; } Context = GetContextRef(); if(!Context) return; Device = Context->Device; LockFilterList(Device); if((ALFilter=LookupFilter(Device, filter)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter); else { /* Call the appropriate handler */ ALfilter_setParamiv(ALFilter, Context, param, values); } UnlockFilterList(Device); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alFilterf(ALuint filter, ALenum param, ALfloat value) { ALCcontext *Context; ALCdevice *Device; ALfilter *ALFilter; Context = GetContextRef(); if(!Context) return; Device = Context->Device; LockFilterList(Device); if((ALFilter=LookupFilter(Device, filter)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter); else { /* Call the appropriate handler */ ALfilter_setParamf(ALFilter, Context, param, value); } UnlockFilterList(Device); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alFilterfv(ALuint filter, ALenum param, const ALfloat *values) { ALCcontext *Context; ALCdevice *Device; ALfilter *ALFilter; Context = GetContextRef(); if(!Context) return; Device = Context->Device; LockFilterList(Device); if((ALFilter=LookupFilter(Device, filter)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter); else { /* Call the appropriate handler */ ALfilter_setParamfv(ALFilter, Context, param, values); } UnlockFilterList(Device); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alGetFilteri(ALuint filter, ALenum param, ALint *value) { ALCcontext *Context; ALCdevice *Device; ALfilter *ALFilter; Context = GetContextRef(); if(!Context) return; Device = Context->Device; LockFilterList(Device); if((ALFilter=LookupFilter(Device, filter)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter); else { if(param == AL_FILTER_TYPE) *value = ALFilter->type; else { /* Call the appropriate handler */ ALfilter_getParami(ALFilter, Context, param, value); } } UnlockFilterList(Device); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alGetFilteriv(ALuint filter, ALenum param, ALint *values) { ALCcontext *Context; ALCdevice *Device; ALfilter *ALFilter; switch(param) { case AL_FILTER_TYPE: alGetFilteri(filter, param, values); return; } Context = GetContextRef(); if(!Context) return; Device = Context->Device; LockFilterList(Device); if((ALFilter=LookupFilter(Device, filter)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter); else { /* Call the appropriate handler */ ALfilter_getParamiv(ALFilter, Context, param, values); } UnlockFilterList(Device); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alGetFilterf(ALuint filter, ALenum param, ALfloat *value) { ALCcontext *Context; ALCdevice *Device; ALfilter *ALFilter; Context = GetContextRef(); if(!Context) return; Device = Context->Device; LockFilterList(Device); if((ALFilter=LookupFilter(Device, filter)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter); else { /* Call the appropriate handler */ ALfilter_getParamf(ALFilter, Context, param, value); } UnlockFilterList(Device); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alGetFilterfv(ALuint filter, ALenum param, ALfloat *values) { ALCcontext *Context; ALCdevice *Device; ALfilter *ALFilter; Context = GetContextRef(); if(!Context) return; Device = Context->Device; LockFilterList(Device); if((ALFilter=LookupFilter(Device, filter)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter); else { /* Call the appropriate handler */ ALfilter_getParamfv(ALFilter, Context, param, values); } UnlockFilterList(Device); ALCcontext_DecRef(Context); } static void ALlowpass_setParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid low-pass integer property 0x%04x", param); } static void ALlowpass_setParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, const ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid low-pass integer-vector property 0x%04x", param); } static void ALlowpass_setParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val) { switch(param) { case AL_LOWPASS_GAIN: if(!(val >= FILTER_MIN_GAIN && val <= FILTER_MAX_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Low-pass gain %f out of range", val); filter->Gain = val; break; case AL_LOWPASS_GAINHF: if(!(val >= AL_LOWPASS_MIN_GAINHF && val <= AL_LOWPASS_MAX_GAINHF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Low-pass gainhf %f out of range", val); filter->GainHF = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid low-pass float property 0x%04x", param); } } static void ALlowpass_setParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals) { ALlowpass_setParamf(filter, context, param, vals[0]); } static void ALlowpass_getParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid low-pass integer property 0x%04x", param); } static void ALlowpass_getParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid low-pass integer-vector property 0x%04x", param); } static void ALlowpass_getParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val) { switch(param) { case AL_LOWPASS_GAIN: *val = filter->Gain; break; case AL_LOWPASS_GAINHF: *val = filter->GainHF; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid low-pass float property 0x%04x", param); } } static void ALlowpass_getParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals) { ALlowpass_getParamf(filter, context, param, vals); } DEFINE_ALFILTER_VTABLE(ALlowpass); static void ALhighpass_setParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid high-pass integer property 0x%04x", param); } static void ALhighpass_setParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, const ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid high-pass integer-vector property 0x%04x", param); } static void ALhighpass_setParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val) { switch(param) { case AL_HIGHPASS_GAIN: if(!(val >= FILTER_MIN_GAIN && val <= FILTER_MAX_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "High-pass gain out of range"); filter->Gain = val; break; case AL_HIGHPASS_GAINLF: if(!(val >= AL_HIGHPASS_MIN_GAINLF && val <= AL_HIGHPASS_MAX_GAINLF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "High-pass gainlf out of range"); filter->GainLF = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid high-pass float property 0x%04x", param); } } static void ALhighpass_setParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals) { ALhighpass_setParamf(filter, context, param, vals[0]); } static void ALhighpass_getParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid high-pass integer property 0x%04x", param); } static void ALhighpass_getParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid high-pass integer-vector property 0x%04x", param); } static void ALhighpass_getParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val) { switch(param) { case AL_HIGHPASS_GAIN: *val = filter->Gain; break; case AL_HIGHPASS_GAINLF: *val = filter->GainLF; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid high-pass float property 0x%04x", param); } } static void ALhighpass_getParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals) { ALhighpass_getParamf(filter, context, param, vals); } DEFINE_ALFILTER_VTABLE(ALhighpass); static void ALbandpass_setParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid band-pass integer property 0x%04x", param); } static void ALbandpass_setParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, const ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid band-pass integer-vector property 0x%04x", param); } static void ALbandpass_setParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val) { switch(param) { case AL_BANDPASS_GAIN: if(!(val >= FILTER_MIN_GAIN && val <= FILTER_MAX_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Band-pass gain out of range"); filter->Gain = val; break; case AL_BANDPASS_GAINHF: if(!(val >= AL_BANDPASS_MIN_GAINHF && val <= AL_BANDPASS_MAX_GAINHF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Band-pass gainhf out of range"); filter->GainHF = val; break; case AL_BANDPASS_GAINLF: if(!(val >= AL_BANDPASS_MIN_GAINLF && val <= AL_BANDPASS_MAX_GAINLF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Band-pass gainlf out of range"); filter->GainLF = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid band-pass float property 0x%04x", param); } } static void ALbandpass_setParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals) { ALbandpass_setParamf(filter, context, param, vals[0]); } static void ALbandpass_getParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid band-pass integer property 0x%04x", param); } static void ALbandpass_getParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid band-pass integer-vector property 0x%04x", param); } static void ALbandpass_getParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val) { switch(param) { case AL_BANDPASS_GAIN: *val = filter->Gain; break; case AL_BANDPASS_GAINHF: *val = filter->GainHF; break; case AL_BANDPASS_GAINLF: *val = filter->GainLF; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid band-pass float property 0x%04x", param); } } static void ALbandpass_getParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals) { ALbandpass_getParamf(filter, context, param, vals); } DEFINE_ALFILTER_VTABLE(ALbandpass); static void ALnullfilter_setParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); } static void ALnullfilter_setParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, const ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); } static void ALnullfilter_setParamf(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALfloat UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); } static void ALnullfilter_setParamfv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, const ALfloat *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); } static void ALnullfilter_getParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); } static void ALnullfilter_getParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); } static void ALnullfilter_getParamf(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALfloat *UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); } static void ALnullfilter_getParamfv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALfloat *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); } DEFINE_ALFILTER_VTABLE(ALnullfilter); static ALfilter *AllocFilter(ALCcontext *context) { ALCdevice *device = context->Device; FilterSubList *sublist, *subend; ALfilter *filter = NULL; ALsizei lidx = 0; ALsizei slidx; almtx_lock(&device->FilterLock); sublist = VECTOR_BEGIN(device->FilterList); subend = VECTOR_END(device->FilterList); for(;sublist != subend;++sublist) { if(sublist->FreeMask) { slidx = CTZ64(sublist->FreeMask); filter = sublist->Filters + slidx; break; } ++lidx; } if(UNLIKELY(!filter)) { const FilterSubList empty_sublist = { 0, NULL }; /* Don't allocate so many list entries that the 32-bit ID could * overflow... */ if(UNLIKELY(VECTOR_SIZE(device->FilterList) >= 1<<25)) { almtx_unlock(&device->FilterLock); alSetError(context, AL_OUT_OF_MEMORY, "Too many filters allocated"); return NULL; } lidx = (ALsizei)VECTOR_SIZE(device->FilterList); VECTOR_PUSH_BACK(device->FilterList, empty_sublist); sublist = &VECTOR_BACK(device->FilterList); sublist->FreeMask = ~U64(0); sublist->Filters = al_calloc(16, sizeof(ALfilter)*64); if(UNLIKELY(!sublist->Filters)) { VECTOR_POP_BACK(device->FilterList); almtx_unlock(&device->FilterLock); alSetError(context, AL_OUT_OF_MEMORY, "Failed to allocate filter batch"); return NULL; } slidx = 0; filter = sublist->Filters + slidx; } memset(filter, 0, sizeof(*filter)); InitFilterParams(filter, AL_FILTER_NULL); /* Add 1 to avoid filter ID 0. */ filter->id = ((lidx<<6) | slidx) + 1; sublist->FreeMask &= ~(U64(1)<FilterLock); return filter; } static void FreeFilter(ALCdevice *device, ALfilter *filter) { ALuint id = filter->id - 1; ALsizei lidx = id >> 6; ALsizei slidx = id & 0x3f; memset(filter, 0, sizeof(*filter)); VECTOR_ELEM(device->FilterList, lidx).FreeMask |= U64(1) << slidx; } void ReleaseALFilters(ALCdevice *device) { FilterSubList *sublist = VECTOR_BEGIN(device->FilterList); FilterSubList *subend = VECTOR_END(device->FilterList); size_t leftover = 0; for(;sublist != subend;++sublist) { ALuint64 usemask = ~sublist->FreeMask; while(usemask) { ALsizei idx = CTZ64(usemask); ALfilter *filter = sublist->Filters + idx; memset(filter, 0, sizeof(*filter)); ++leftover; usemask &= ~(U64(1) << idx); } sublist->FreeMask = ~usemask; } if(leftover > 0) WARN("(%p) Deleted "SZFMT" Filter%s\n", device, leftover, (leftover==1)?"":"s"); } static void InitFilterParams(ALfilter *filter, ALenum type) { if(type == AL_FILTER_LOWPASS) { filter->Gain = AL_LOWPASS_DEFAULT_GAIN; filter->GainHF = AL_LOWPASS_DEFAULT_GAINHF; filter->HFReference = LOWPASSFREQREF; filter->GainLF = 1.0f; filter->LFReference = HIGHPASSFREQREF; filter->vtab = &ALlowpass_vtable; } else if(type == AL_FILTER_HIGHPASS) { filter->Gain = AL_HIGHPASS_DEFAULT_GAIN; filter->GainHF = 1.0f; filter->HFReference = LOWPASSFREQREF; filter->GainLF = AL_HIGHPASS_DEFAULT_GAINLF; filter->LFReference = HIGHPASSFREQREF; filter->vtab = &ALhighpass_vtable; } else if(type == AL_FILTER_BANDPASS) { filter->Gain = AL_BANDPASS_DEFAULT_GAIN; filter->GainHF = AL_BANDPASS_DEFAULT_GAINHF; filter->HFReference = LOWPASSFREQREF; filter->GainLF = AL_BANDPASS_DEFAULT_GAINLF; filter->LFReference = HIGHPASSFREQREF; filter->vtab = &ALbandpass_vtable; } else { filter->Gain = 1.0f; filter->GainHF = 1.0f; filter->HFReference = LOWPASSFREQREF; filter->GainLF = 1.0f; filter->LFReference = HIGHPASSFREQREF; filter->vtab = &ALnullfilter_vtable; } filter->type = type; } openal-soft-openal-soft-1.19.1/OpenAL32/alListener.c000066400000000000000000000333221335774445300220320ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2000 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include "alMain.h" #include "alu.h" #include "alError.h" #include "alListener.h" #include "alSource.h" #define DO_UPDATEPROPS() do { \ if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire)) \ UpdateListenerProps(context); \ else \ ATOMIC_FLAG_CLEAR(&listener->PropsClean, almemory_order_release); \ } while(0) AL_API ALvoid AL_APIENTRY alListenerf(ALenum param, ALfloat value) { ALlistener *listener; ALCcontext *context; context = GetContextRef(); if(!context) return; listener = context->Listener; almtx_lock(&context->PropLock); switch(param) { case AL_GAIN: if(!(value >= 0.0f && isfinite(value))) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Listener gain out of range"); listener->Gain = value; DO_UPDATEPROPS(); break; case AL_METERS_PER_UNIT: if(!(value >= AL_MIN_METERS_PER_UNIT && value <= AL_MAX_METERS_PER_UNIT)) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Listener meters per unit out of range"); context->MetersPerUnit = value; if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire)) UpdateContextProps(context); else ATOMIC_FLAG_CLEAR(&context->PropsClean, almemory_order_release); break; default: alSetError(context, AL_INVALID_ENUM, "Invalid listener float property"); } done: almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alListener3f(ALenum param, ALfloat value1, ALfloat value2, ALfloat value3) { ALlistener *listener; ALCcontext *context; context = GetContextRef(); if(!context) return; listener = context->Listener; almtx_lock(&context->PropLock); switch(param) { case AL_POSITION: if(!(isfinite(value1) && isfinite(value2) && isfinite(value3))) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Listener position out of range"); listener->Position[0] = value1; listener->Position[1] = value2; listener->Position[2] = value3; DO_UPDATEPROPS(); break; case AL_VELOCITY: if(!(isfinite(value1) && isfinite(value2) && isfinite(value3))) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Listener velocity out of range"); listener->Velocity[0] = value1; listener->Velocity[1] = value2; listener->Velocity[2] = value3; DO_UPDATEPROPS(); break; default: alSetError(context, AL_INVALID_ENUM, "Invalid listener 3-float property"); } done: almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alListenerfv(ALenum param, const ALfloat *values) { ALlistener *listener; ALCcontext *context; if(values) { switch(param) { case AL_GAIN: case AL_METERS_PER_UNIT: alListenerf(param, values[0]); return; case AL_POSITION: case AL_VELOCITY: alListener3f(param, values[0], values[1], values[2]); return; } } context = GetContextRef(); if(!context) return; listener = context->Listener; almtx_lock(&context->PropLock); if(!values) SETERR_GOTO(context, AL_INVALID_VALUE, done, "NULL pointer"); switch(param) { case AL_ORIENTATION: if(!(isfinite(values[0]) && isfinite(values[1]) && isfinite(values[2]) && isfinite(values[3]) && isfinite(values[4]) && isfinite(values[5]))) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Listener orientation out of range"); /* AT then UP */ listener->Forward[0] = values[0]; listener->Forward[1] = values[1]; listener->Forward[2] = values[2]; listener->Up[0] = values[3]; listener->Up[1] = values[4]; listener->Up[2] = values[5]; DO_UPDATEPROPS(); break; default: alSetError(context, AL_INVALID_ENUM, "Invalid listener float-vector property"); } done: almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alListeneri(ALenum param, ALint UNUSED(value)) { ALCcontext *context; context = GetContextRef(); if(!context) return; almtx_lock(&context->PropLock); switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid listener integer property"); } almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alListener3i(ALenum param, ALint value1, ALint value2, ALint value3) { ALCcontext *context; switch(param) { case AL_POSITION: case AL_VELOCITY: alListener3f(param, (ALfloat)value1, (ALfloat)value2, (ALfloat)value3); return; } context = GetContextRef(); if(!context) return; almtx_lock(&context->PropLock); switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid listener 3-integer property"); } almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alListeneriv(ALenum param, const ALint *values) { ALCcontext *context; if(values) { ALfloat fvals[6]; switch(param) { case AL_POSITION: case AL_VELOCITY: alListener3f(param, (ALfloat)values[0], (ALfloat)values[1], (ALfloat)values[2]); return; case AL_ORIENTATION: fvals[0] = (ALfloat)values[0]; fvals[1] = (ALfloat)values[1]; fvals[2] = (ALfloat)values[2]; fvals[3] = (ALfloat)values[3]; fvals[4] = (ALfloat)values[4]; fvals[5] = (ALfloat)values[5]; alListenerfv(param, fvals); return; } } context = GetContextRef(); if(!context) return; almtx_lock(&context->PropLock); if(!values) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); else switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid listener integer-vector property"); } almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alGetListenerf(ALenum param, ALfloat *value) { ALCcontext *context; context = GetContextRef(); if(!context) return; almtx_lock(&context->PropLock); if(!value) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); else switch(param) { case AL_GAIN: *value = context->Listener->Gain; break; case AL_METERS_PER_UNIT: *value = context->MetersPerUnit; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid listener float property"); } almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alGetListener3f(ALenum param, ALfloat *value1, ALfloat *value2, ALfloat *value3) { ALCcontext *context; context = GetContextRef(); if(!context) return; almtx_lock(&context->PropLock); if(!value1 || !value2 || !value3) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); else switch(param) { case AL_POSITION: *value1 = context->Listener->Position[0]; *value2 = context->Listener->Position[1]; *value3 = context->Listener->Position[2]; break; case AL_VELOCITY: *value1 = context->Listener->Velocity[0]; *value2 = context->Listener->Velocity[1]; *value3 = context->Listener->Velocity[2]; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid listener 3-float property"); } almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alGetListenerfv(ALenum param, ALfloat *values) { ALCcontext *context; switch(param) { case AL_GAIN: case AL_METERS_PER_UNIT: alGetListenerf(param, values); return; case AL_POSITION: case AL_VELOCITY: alGetListener3f(param, values+0, values+1, values+2); return; } context = GetContextRef(); if(!context) return; almtx_lock(&context->PropLock); if(!values) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); else switch(param) { case AL_ORIENTATION: // AT then UP values[0] = context->Listener->Forward[0]; values[1] = context->Listener->Forward[1]; values[2] = context->Listener->Forward[2]; values[3] = context->Listener->Up[0]; values[4] = context->Listener->Up[1]; values[5] = context->Listener->Up[2]; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid listener float-vector property"); } almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alGetListeneri(ALenum param, ALint *value) { ALCcontext *context; context = GetContextRef(); if(!context) return; almtx_lock(&context->PropLock); if(!value) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); else switch(param) { default: alSetError(context, AL_INVALID_ENUM, "Invalid listener integer property"); } almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alGetListener3i(ALenum param, ALint *value1, ALint *value2, ALint *value3) { ALCcontext *context; context = GetContextRef(); if(!context) return; almtx_lock(&context->PropLock); if(!value1 || !value2 || !value3) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); else switch(param) { case AL_POSITION: *value1 = (ALint)context->Listener->Position[0]; *value2 = (ALint)context->Listener->Position[1]; *value3 = (ALint)context->Listener->Position[2]; break; case AL_VELOCITY: *value1 = (ALint)context->Listener->Velocity[0]; *value2 = (ALint)context->Listener->Velocity[1]; *value3 = (ALint)context->Listener->Velocity[2]; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid listener 3-integer property"); } almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alGetListeneriv(ALenum param, ALint* values) { ALCcontext *context; switch(param) { case AL_POSITION: case AL_VELOCITY: alGetListener3i(param, values+0, values+1, values+2); return; } context = GetContextRef(); if(!context) return; almtx_lock(&context->PropLock); if(!values) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); else switch(param) { case AL_ORIENTATION: // AT then UP values[0] = (ALint)context->Listener->Forward[0]; values[1] = (ALint)context->Listener->Forward[1]; values[2] = (ALint)context->Listener->Forward[2]; values[3] = (ALint)context->Listener->Up[0]; values[4] = (ALint)context->Listener->Up[1]; values[5] = (ALint)context->Listener->Up[2]; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid listener integer-vector property"); } almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); } void UpdateListenerProps(ALCcontext *context) { ALlistener *listener = context->Listener; struct ALlistenerProps *props; /* Get an unused proprty container, or allocate a new one as needed. */ props = ATOMIC_LOAD(&context->FreeListenerProps, almemory_order_acquire); if(!props) props = al_calloc(16, sizeof(*props)); else { struct ALlistenerProps *next; do { next = ATOMIC_LOAD(&props->next, almemory_order_relaxed); } while(ATOMIC_COMPARE_EXCHANGE_PTR_WEAK(&context->FreeListenerProps, &props, next, almemory_order_seq_cst, almemory_order_acquire) == 0); } /* Copy in current property values. */ props->Position[0] = listener->Position[0]; props->Position[1] = listener->Position[1]; props->Position[2] = listener->Position[2]; props->Velocity[0] = listener->Velocity[0]; props->Velocity[1] = listener->Velocity[1]; props->Velocity[2] = listener->Velocity[2]; props->Forward[0] = listener->Forward[0]; props->Forward[1] = listener->Forward[1]; props->Forward[2] = listener->Forward[2]; props->Up[0] = listener->Up[0]; props->Up[1] = listener->Up[1]; props->Up[2] = listener->Up[2]; props->Gain = listener->Gain; /* Set the new container for updating internal parameters. */ props = ATOMIC_EXCHANGE_PTR(&listener->Update, props, almemory_order_acq_rel); if(props) { /* If there was an unused update container, put it back in the * freelist. */ ATOMIC_REPLACE_HEAD(struct ALlistenerProps*, &context->FreeListenerProps, props); } } openal-soft-openal-soft-1.19.1/OpenAL32/alSource.c000066400000000000000000003535051335774445300215150ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include #include "AL/al.h" #include "AL/alc.h" #include "alMain.h" #include "alError.h" #include "alSource.h" #include "alBuffer.h" #include "alFilter.h" #include "alAuxEffectSlot.h" #include "ringbuffer.h" #include "backends/base.h" #include "threads.h" #include "almalloc.h" static ALsource *AllocSource(ALCcontext *context); static void FreeSource(ALCcontext *context, ALsource *source); static void InitSourceParams(ALsource *Source, ALsizei num_sends); static void DeinitSource(ALsource *source, ALsizei num_sends); static void UpdateSourceProps(ALsource *source, ALvoice *voice, ALsizei num_sends, ALCcontext *context); static ALint64 GetSourceSampleOffset(ALsource *Source, ALCcontext *context, ALuint64 *clocktime); static ALdouble GetSourceSecOffset(ALsource *Source, ALCcontext *context, ALuint64 *clocktime); static ALdouble GetSourceOffset(ALsource *Source, ALenum name, ALCcontext *context); static ALboolean GetSampleOffset(ALsource *Source, ALuint *offset, ALsizei *frac); static ALboolean ApplyOffset(ALsource *Source, ALvoice *voice); static inline void LockSourceList(ALCcontext *context) { almtx_lock(&context->SourceLock); } static inline void UnlockSourceList(ALCcontext *context) { almtx_unlock(&context->SourceLock); } static inline ALsource *LookupSource(ALCcontext *context, ALuint id) { SourceSubList *sublist; ALuint lidx = (id-1) >> 6; ALsizei slidx = (id-1) & 0x3f; if(UNLIKELY(lidx >= VECTOR_SIZE(context->SourceList))) return NULL; sublist = &VECTOR_ELEM(context->SourceList, lidx); if(UNLIKELY(sublist->FreeMask & (U64(1)<Sources + slidx; } static inline ALbuffer *LookupBuffer(ALCdevice *device, ALuint id) { BufferSubList *sublist; ALuint lidx = (id-1) >> 6; ALsizei slidx = (id-1) & 0x3f; if(UNLIKELY(lidx >= VECTOR_SIZE(device->BufferList))) return NULL; sublist = &VECTOR_ELEM(device->BufferList, lidx); if(UNLIKELY(sublist->FreeMask & (U64(1)<Buffers + slidx; } static inline ALfilter *LookupFilter(ALCdevice *device, ALuint id) { FilterSubList *sublist; ALuint lidx = (id-1) >> 6; ALsizei slidx = (id-1) & 0x3f; if(UNLIKELY(lidx >= VECTOR_SIZE(device->FilterList))) return NULL; sublist = &VECTOR_ELEM(device->FilterList, lidx); if(UNLIKELY(sublist->FreeMask & (U64(1)<Filters + slidx; } static inline ALeffectslot *LookupEffectSlot(ALCcontext *context, ALuint id) { id--; if(UNLIKELY(id >= VECTOR_SIZE(context->EffectSlotList))) return NULL; return VECTOR_ELEM(context->EffectSlotList, id); } typedef enum SourceProp { srcPitch = AL_PITCH, srcGain = AL_GAIN, srcMinGain = AL_MIN_GAIN, srcMaxGain = AL_MAX_GAIN, srcMaxDistance = AL_MAX_DISTANCE, srcRolloffFactor = AL_ROLLOFF_FACTOR, srcDopplerFactor = AL_DOPPLER_FACTOR, srcConeOuterGain = AL_CONE_OUTER_GAIN, srcSecOffset = AL_SEC_OFFSET, srcSampleOffset = AL_SAMPLE_OFFSET, srcByteOffset = AL_BYTE_OFFSET, srcConeInnerAngle = AL_CONE_INNER_ANGLE, srcConeOuterAngle = AL_CONE_OUTER_ANGLE, srcRefDistance = AL_REFERENCE_DISTANCE, srcPosition = AL_POSITION, srcVelocity = AL_VELOCITY, srcDirection = AL_DIRECTION, srcSourceRelative = AL_SOURCE_RELATIVE, srcLooping = AL_LOOPING, srcBuffer = AL_BUFFER, srcSourceState = AL_SOURCE_STATE, srcBuffersQueued = AL_BUFFERS_QUEUED, srcBuffersProcessed = AL_BUFFERS_PROCESSED, srcSourceType = AL_SOURCE_TYPE, /* ALC_EXT_EFX */ srcConeOuterGainHF = AL_CONE_OUTER_GAINHF, srcAirAbsorptionFactor = AL_AIR_ABSORPTION_FACTOR, srcRoomRolloffFactor = AL_ROOM_ROLLOFF_FACTOR, srcDirectFilterGainHFAuto = AL_DIRECT_FILTER_GAINHF_AUTO, srcAuxSendFilterGainAuto = AL_AUXILIARY_SEND_FILTER_GAIN_AUTO, srcAuxSendFilterGainHFAuto = AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO, srcDirectFilter = AL_DIRECT_FILTER, srcAuxSendFilter = AL_AUXILIARY_SEND_FILTER, /* AL_SOFT_direct_channels */ srcDirectChannelsSOFT = AL_DIRECT_CHANNELS_SOFT, /* AL_EXT_source_distance_model */ srcDistanceModel = AL_DISTANCE_MODEL, /* AL_SOFT_source_latency */ srcSampleOffsetLatencySOFT = AL_SAMPLE_OFFSET_LATENCY_SOFT, srcSecOffsetLatencySOFT = AL_SEC_OFFSET_LATENCY_SOFT, /* AL_EXT_STEREO_ANGLES */ srcAngles = AL_STEREO_ANGLES, /* AL_EXT_SOURCE_RADIUS */ srcRadius = AL_SOURCE_RADIUS, /* AL_EXT_BFORMAT */ srcOrientation = AL_ORIENTATION, /* AL_SOFT_source_resampler */ srcResampler = AL_SOURCE_RESAMPLER_SOFT, /* AL_SOFT_source_spatialize */ srcSpatialize = AL_SOURCE_SPATIALIZE_SOFT, /* ALC_SOFT_device_clock */ srcSampleOffsetClockSOFT = AL_SAMPLE_OFFSET_CLOCK_SOFT, srcSecOffsetClockSOFT = AL_SEC_OFFSET_CLOCK_SOFT, } SourceProp; static ALboolean SetSourcefv(ALsource *Source, ALCcontext *Context, SourceProp prop, const ALfloat *values); static ALboolean SetSourceiv(ALsource *Source, ALCcontext *Context, SourceProp prop, const ALint *values); static ALboolean SetSourcei64v(ALsource *Source, ALCcontext *Context, SourceProp prop, const ALint64SOFT *values); static ALboolean GetSourcedv(ALsource *Source, ALCcontext *Context, SourceProp prop, ALdouble *values); static ALboolean GetSourceiv(ALsource *Source, ALCcontext *Context, SourceProp prop, ALint *values); static ALboolean GetSourcei64v(ALsource *Source, ALCcontext *Context, SourceProp prop, ALint64 *values); static inline ALvoice *GetSourceVoice(ALsource *source, ALCcontext *context) { ALint idx = source->VoiceIdx; if(idx >= 0 && idx < context->VoiceCount) { ALvoice *voice = context->Voices[idx]; if(ATOMIC_LOAD(&voice->Source, almemory_order_acquire) == source) return voice; } source->VoiceIdx = -1; return NULL; } /** * Returns if the last known state for the source was playing or paused. Does * not sync with the mixer voice. */ static inline bool IsPlayingOrPaused(ALsource *source) { return source->state == AL_PLAYING || source->state == AL_PAUSED; } /** * Returns an updated source state using the matching voice's status (or lack * thereof). */ static inline ALenum GetSourceState(ALsource *source, ALvoice *voice) { if(!voice && source->state == AL_PLAYING) source->state = AL_STOPPED; return source->state; } /** * Returns if the source should specify an update, given the context's * deferring state and the source's last known state. */ static inline bool SourceShouldUpdate(ALsource *source, ALCcontext *context) { return !ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire) && IsPlayingOrPaused(source); } /** Can only be called while the mixer is locked! */ static void SendStateChangeEvent(ALCcontext *context, ALuint id, ALenum state) { AsyncEvent evt = ASYNC_EVENT(EventType_SourceStateChange); ALbitfieldSOFT enabledevt; enabledevt = ATOMIC_LOAD(&context->EnabledEvts, almemory_order_acquire); if(!(enabledevt&EventType_SourceStateChange)) return; evt.u.user.type = AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT; evt.u.user.id = id; evt.u.user.param = state; snprintf(evt.u.user.msg, sizeof(evt.u.user.msg), "Source ID %u state changed to %s", id, (state==AL_INITIAL) ? "AL_INITIAL" : (state==AL_PLAYING) ? "AL_PLAYING" : (state==AL_PAUSED) ? "AL_PAUSED" : (state==AL_STOPPED) ? "AL_STOPPED" : "" ); /* The mixer may have queued a state change that's not yet been processed, * and we don't want state change messages to occur out of order, so send * it through the async queue to ensure proper ordering. */ if(ll_ringbuffer_write(context->AsyncEvents, (const char*)&evt, 1) == 1) alsem_post(&context->EventSem); } static ALint FloatValsByProp(ALenum prop) { if(prop != (ALenum)((SourceProp)prop)) return 0; switch((SourceProp)prop) { case AL_PITCH: case AL_GAIN: case AL_MIN_GAIN: case AL_MAX_GAIN: case AL_MAX_DISTANCE: case AL_ROLLOFF_FACTOR: case AL_DOPPLER_FACTOR: case AL_CONE_OUTER_GAIN: case AL_SEC_OFFSET: case AL_SAMPLE_OFFSET: case AL_BYTE_OFFSET: case AL_CONE_INNER_ANGLE: case AL_CONE_OUTER_ANGLE: case AL_REFERENCE_DISTANCE: case AL_CONE_OUTER_GAINHF: case AL_AIR_ABSORPTION_FACTOR: case AL_ROOM_ROLLOFF_FACTOR: case AL_DIRECT_FILTER_GAINHF_AUTO: case AL_AUXILIARY_SEND_FILTER_GAIN_AUTO: case AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO: case AL_DIRECT_CHANNELS_SOFT: case AL_DISTANCE_MODEL: case AL_SOURCE_RELATIVE: case AL_LOOPING: case AL_SOURCE_STATE: case AL_BUFFERS_QUEUED: case AL_BUFFERS_PROCESSED: case AL_SOURCE_TYPE: case AL_SOURCE_RADIUS: case AL_SOURCE_RESAMPLER_SOFT: case AL_SOURCE_SPATIALIZE_SOFT: return 1; case AL_STEREO_ANGLES: return 2; case AL_POSITION: case AL_VELOCITY: case AL_DIRECTION: return 3; case AL_ORIENTATION: return 6; case AL_SEC_OFFSET_LATENCY_SOFT: case AL_SEC_OFFSET_CLOCK_SOFT: break; /* Double only */ case AL_BUFFER: case AL_DIRECT_FILTER: case AL_AUXILIARY_SEND_FILTER: break; /* i/i64 only */ case AL_SAMPLE_OFFSET_LATENCY_SOFT: case AL_SAMPLE_OFFSET_CLOCK_SOFT: break; /* i64 only */ } return 0; } static ALint DoubleValsByProp(ALenum prop) { if(prop != (ALenum)((SourceProp)prop)) return 0; switch((SourceProp)prop) { case AL_PITCH: case AL_GAIN: case AL_MIN_GAIN: case AL_MAX_GAIN: case AL_MAX_DISTANCE: case AL_ROLLOFF_FACTOR: case AL_DOPPLER_FACTOR: case AL_CONE_OUTER_GAIN: case AL_SEC_OFFSET: case AL_SAMPLE_OFFSET: case AL_BYTE_OFFSET: case AL_CONE_INNER_ANGLE: case AL_CONE_OUTER_ANGLE: case AL_REFERENCE_DISTANCE: case AL_CONE_OUTER_GAINHF: case AL_AIR_ABSORPTION_FACTOR: case AL_ROOM_ROLLOFF_FACTOR: case AL_DIRECT_FILTER_GAINHF_AUTO: case AL_AUXILIARY_SEND_FILTER_GAIN_AUTO: case AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO: case AL_DIRECT_CHANNELS_SOFT: case AL_DISTANCE_MODEL: case AL_SOURCE_RELATIVE: case AL_LOOPING: case AL_SOURCE_STATE: case AL_BUFFERS_QUEUED: case AL_BUFFERS_PROCESSED: case AL_SOURCE_TYPE: case AL_SOURCE_RADIUS: case AL_SOURCE_RESAMPLER_SOFT: case AL_SOURCE_SPATIALIZE_SOFT: return 1; case AL_SEC_OFFSET_LATENCY_SOFT: case AL_SEC_OFFSET_CLOCK_SOFT: case AL_STEREO_ANGLES: return 2; case AL_POSITION: case AL_VELOCITY: case AL_DIRECTION: return 3; case AL_ORIENTATION: return 6; case AL_BUFFER: case AL_DIRECT_FILTER: case AL_AUXILIARY_SEND_FILTER: break; /* i/i64 only */ case AL_SAMPLE_OFFSET_LATENCY_SOFT: case AL_SAMPLE_OFFSET_CLOCK_SOFT: break; /* i64 only */ } return 0; } static ALint IntValsByProp(ALenum prop) { if(prop != (ALenum)((SourceProp)prop)) return 0; switch((SourceProp)prop) { case AL_PITCH: case AL_GAIN: case AL_MIN_GAIN: case AL_MAX_GAIN: case AL_MAX_DISTANCE: case AL_ROLLOFF_FACTOR: case AL_DOPPLER_FACTOR: case AL_CONE_OUTER_GAIN: case AL_SEC_OFFSET: case AL_SAMPLE_OFFSET: case AL_BYTE_OFFSET: case AL_CONE_INNER_ANGLE: case AL_CONE_OUTER_ANGLE: case AL_REFERENCE_DISTANCE: case AL_CONE_OUTER_GAINHF: case AL_AIR_ABSORPTION_FACTOR: case AL_ROOM_ROLLOFF_FACTOR: case AL_DIRECT_FILTER_GAINHF_AUTO: case AL_AUXILIARY_SEND_FILTER_GAIN_AUTO: case AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO: case AL_DIRECT_CHANNELS_SOFT: case AL_DISTANCE_MODEL: case AL_SOURCE_RELATIVE: case AL_LOOPING: case AL_BUFFER: case AL_SOURCE_STATE: case AL_BUFFERS_QUEUED: case AL_BUFFERS_PROCESSED: case AL_SOURCE_TYPE: case AL_DIRECT_FILTER: case AL_SOURCE_RADIUS: case AL_SOURCE_RESAMPLER_SOFT: case AL_SOURCE_SPATIALIZE_SOFT: return 1; case AL_POSITION: case AL_VELOCITY: case AL_DIRECTION: case AL_AUXILIARY_SEND_FILTER: return 3; case AL_ORIENTATION: return 6; case AL_SAMPLE_OFFSET_LATENCY_SOFT: case AL_SAMPLE_OFFSET_CLOCK_SOFT: break; /* i64 only */ case AL_SEC_OFFSET_LATENCY_SOFT: case AL_SEC_OFFSET_CLOCK_SOFT: break; /* Double only */ case AL_STEREO_ANGLES: break; /* Float/double only */ } return 0; } static ALint Int64ValsByProp(ALenum prop) { if(prop != (ALenum)((SourceProp)prop)) return 0; switch((SourceProp)prop) { case AL_PITCH: case AL_GAIN: case AL_MIN_GAIN: case AL_MAX_GAIN: case AL_MAX_DISTANCE: case AL_ROLLOFF_FACTOR: case AL_DOPPLER_FACTOR: case AL_CONE_OUTER_GAIN: case AL_SEC_OFFSET: case AL_SAMPLE_OFFSET: case AL_BYTE_OFFSET: case AL_CONE_INNER_ANGLE: case AL_CONE_OUTER_ANGLE: case AL_REFERENCE_DISTANCE: case AL_CONE_OUTER_GAINHF: case AL_AIR_ABSORPTION_FACTOR: case AL_ROOM_ROLLOFF_FACTOR: case AL_DIRECT_FILTER_GAINHF_AUTO: case AL_AUXILIARY_SEND_FILTER_GAIN_AUTO: case AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO: case AL_DIRECT_CHANNELS_SOFT: case AL_DISTANCE_MODEL: case AL_SOURCE_RELATIVE: case AL_LOOPING: case AL_BUFFER: case AL_SOURCE_STATE: case AL_BUFFERS_QUEUED: case AL_BUFFERS_PROCESSED: case AL_SOURCE_TYPE: case AL_DIRECT_FILTER: case AL_SOURCE_RADIUS: case AL_SOURCE_RESAMPLER_SOFT: case AL_SOURCE_SPATIALIZE_SOFT: return 1; case AL_SAMPLE_OFFSET_LATENCY_SOFT: case AL_SAMPLE_OFFSET_CLOCK_SOFT: return 2; case AL_POSITION: case AL_VELOCITY: case AL_DIRECTION: case AL_AUXILIARY_SEND_FILTER: return 3; case AL_ORIENTATION: return 6; case AL_SEC_OFFSET_LATENCY_SOFT: case AL_SEC_OFFSET_CLOCK_SOFT: break; /* Double only */ case AL_STEREO_ANGLES: break; /* Float/double only */ } return 0; } #define CHECKVAL(x) do { \ if(!(x)) \ { \ alSetError(Context, AL_INVALID_VALUE, "Value out of range"); \ return AL_FALSE; \ } \ } while(0) #define DO_UPDATEPROPS() do { \ ALvoice *voice; \ if(SourceShouldUpdate(Source, Context) && \ (voice=GetSourceVoice(Source, Context)) != NULL) \ UpdateSourceProps(Source, voice, device->NumAuxSends, Context); \ else \ ATOMIC_FLAG_CLEAR(&Source->PropsClean, almemory_order_release); \ } while(0) static ALboolean SetSourcefv(ALsource *Source, ALCcontext *Context, SourceProp prop, const ALfloat *values) { ALCdevice *device = Context->Device; ALint ival; switch(prop) { case AL_SEC_OFFSET_LATENCY_SOFT: case AL_SEC_OFFSET_CLOCK_SOFT: /* Query only */ SETERR_RETURN(Context, AL_INVALID_OPERATION, AL_FALSE, "Setting read-only source property 0x%04x", prop); case AL_PITCH: CHECKVAL(*values >= 0.0f); Source->Pitch = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_CONE_INNER_ANGLE: CHECKVAL(*values >= 0.0f && *values <= 360.0f); Source->InnerAngle = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_CONE_OUTER_ANGLE: CHECKVAL(*values >= 0.0f && *values <= 360.0f); Source->OuterAngle = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_GAIN: CHECKVAL(*values >= 0.0f); Source->Gain = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_MAX_DISTANCE: CHECKVAL(*values >= 0.0f); Source->MaxDistance = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_ROLLOFF_FACTOR: CHECKVAL(*values >= 0.0f); Source->RolloffFactor = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_REFERENCE_DISTANCE: CHECKVAL(*values >= 0.0f); Source->RefDistance = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_MIN_GAIN: CHECKVAL(*values >= 0.0f); Source->MinGain = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_MAX_GAIN: CHECKVAL(*values >= 0.0f); Source->MaxGain = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_CONE_OUTER_GAIN: CHECKVAL(*values >= 0.0f && *values <= 1.0f); Source->OuterGain = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_CONE_OUTER_GAINHF: CHECKVAL(*values >= 0.0f && *values <= 1.0f); Source->OuterGainHF = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_AIR_ABSORPTION_FACTOR: CHECKVAL(*values >= 0.0f && *values <= 10.0f); Source->AirAbsorptionFactor = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_ROOM_ROLLOFF_FACTOR: CHECKVAL(*values >= 0.0f && *values <= 10.0f); Source->RoomRolloffFactor = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_DOPPLER_FACTOR: CHECKVAL(*values >= 0.0f && *values <= 1.0f); Source->DopplerFactor = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_SEC_OFFSET: case AL_SAMPLE_OFFSET: case AL_BYTE_OFFSET: CHECKVAL(*values >= 0.0f); Source->OffsetType = prop; Source->Offset = *values; if(IsPlayingOrPaused(Source)) { ALvoice *voice; ALCdevice_Lock(Context->Device); /* Double-check that the source is still playing while we have * the lock. */ voice = GetSourceVoice(Source, Context); if(voice) { if(ApplyOffset(Source, voice) == AL_FALSE) { ALCdevice_Unlock(Context->Device); SETERR_RETURN(Context, AL_INVALID_VALUE, AL_FALSE, "Invalid offset"); } } ALCdevice_Unlock(Context->Device); } return AL_TRUE; case AL_SOURCE_RADIUS: CHECKVAL(*values >= 0.0f && isfinite(*values)); Source->Radius = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_STEREO_ANGLES: CHECKVAL(isfinite(values[0]) && isfinite(values[1])); Source->StereoPan[0] = values[0]; Source->StereoPan[1] = values[1]; DO_UPDATEPROPS(); return AL_TRUE; case AL_POSITION: CHECKVAL(isfinite(values[0]) && isfinite(values[1]) && isfinite(values[2])); Source->Position[0] = values[0]; Source->Position[1] = values[1]; Source->Position[2] = values[2]; DO_UPDATEPROPS(); return AL_TRUE; case AL_VELOCITY: CHECKVAL(isfinite(values[0]) && isfinite(values[1]) && isfinite(values[2])); Source->Velocity[0] = values[0]; Source->Velocity[1] = values[1]; Source->Velocity[2] = values[2]; DO_UPDATEPROPS(); return AL_TRUE; case AL_DIRECTION: CHECKVAL(isfinite(values[0]) && isfinite(values[1]) && isfinite(values[2])); Source->Direction[0] = values[0]; Source->Direction[1] = values[1]; Source->Direction[2] = values[2]; DO_UPDATEPROPS(); return AL_TRUE; case AL_ORIENTATION: CHECKVAL(isfinite(values[0]) && isfinite(values[1]) && isfinite(values[2]) && isfinite(values[3]) && isfinite(values[4]) && isfinite(values[5])); Source->Orientation[0][0] = values[0]; Source->Orientation[0][1] = values[1]; Source->Orientation[0][2] = values[2]; Source->Orientation[1][0] = values[3]; Source->Orientation[1][1] = values[4]; Source->Orientation[1][2] = values[5]; DO_UPDATEPROPS(); return AL_TRUE; case AL_SOURCE_RELATIVE: case AL_LOOPING: case AL_SOURCE_STATE: case AL_SOURCE_TYPE: case AL_DISTANCE_MODEL: case AL_DIRECT_FILTER_GAINHF_AUTO: case AL_AUXILIARY_SEND_FILTER_GAIN_AUTO: case AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO: case AL_DIRECT_CHANNELS_SOFT: case AL_SOURCE_RESAMPLER_SOFT: case AL_SOURCE_SPATIALIZE_SOFT: ival = (ALint)values[0]; return SetSourceiv(Source, Context, prop, &ival); case AL_BUFFERS_QUEUED: case AL_BUFFERS_PROCESSED: ival = (ALint)((ALuint)values[0]); return SetSourceiv(Source, Context, prop, &ival); case AL_BUFFER: case AL_DIRECT_FILTER: case AL_AUXILIARY_SEND_FILTER: case AL_SAMPLE_OFFSET_LATENCY_SOFT: case AL_SAMPLE_OFFSET_CLOCK_SOFT: break; } ERR("Unexpected property: 0x%04x\n", prop); SETERR_RETURN(Context, AL_INVALID_ENUM, AL_FALSE, "Invalid source float property 0x%04x", prop); } static ALboolean SetSourceiv(ALsource *Source, ALCcontext *Context, SourceProp prop, const ALint *values) { ALCdevice *device = Context->Device; ALbuffer *buffer = NULL; ALfilter *filter = NULL; ALeffectslot *slot = NULL; ALbufferlistitem *oldlist; ALfloat fvals[6]; switch(prop) { case AL_SOURCE_STATE: case AL_SOURCE_TYPE: case AL_BUFFERS_QUEUED: case AL_BUFFERS_PROCESSED: /* Query only */ SETERR_RETURN(Context, AL_INVALID_OPERATION, AL_FALSE, "Setting read-only source property 0x%04x", prop); case AL_SOURCE_RELATIVE: CHECKVAL(*values == AL_FALSE || *values == AL_TRUE); Source->HeadRelative = (ALboolean)*values; DO_UPDATEPROPS(); return AL_TRUE; case AL_LOOPING: CHECKVAL(*values == AL_FALSE || *values == AL_TRUE); Source->Looping = (ALboolean)*values; if(IsPlayingOrPaused(Source)) { ALvoice *voice = GetSourceVoice(Source, Context); if(voice) { if(Source->Looping) ATOMIC_STORE(&voice->loop_buffer, Source->queue, almemory_order_release); else ATOMIC_STORE(&voice->loop_buffer, NULL, almemory_order_release); /* If the source is playing, wait for the current mix to finish * to ensure it isn't currently looping back or reaching the * end. */ while((ATOMIC_LOAD(&device->MixCount, almemory_order_acquire)&1)) althrd_yield(); } } return AL_TRUE; case AL_BUFFER: LockBufferList(device); if(!(*values == 0 || (buffer=LookupBuffer(device, *values)) != NULL)) { UnlockBufferList(device); SETERR_RETURN(Context, AL_INVALID_VALUE, AL_FALSE, "Invalid buffer ID %u", *values); } if(buffer && buffer->MappedAccess != 0 && !(buffer->MappedAccess&AL_MAP_PERSISTENT_BIT_SOFT)) { UnlockBufferList(device); SETERR_RETURN(Context, AL_INVALID_OPERATION, AL_FALSE, "Setting non-persistently mapped buffer %u", buffer->id); } else { ALenum state = GetSourceState(Source, GetSourceVoice(Source, Context)); if(state == AL_PLAYING || state == AL_PAUSED) { UnlockBufferList(device); SETERR_RETURN(Context, AL_INVALID_OPERATION, AL_FALSE, "Setting buffer on playing or paused source %u", Source->id); } } oldlist = Source->queue; if(buffer != NULL) { /* Add the selected buffer to a one-item queue */ ALbufferlistitem *newlist = al_calloc(DEF_ALIGN, FAM_SIZE(ALbufferlistitem, buffers, 1)); ATOMIC_INIT(&newlist->next, NULL); newlist->max_samples = buffer->SampleLen; newlist->num_buffers = 1; newlist->buffers[0] = buffer; IncrementRef(&buffer->ref); /* Source is now Static */ Source->SourceType = AL_STATIC; Source->queue = newlist; } else { /* Source is now Undetermined */ Source->SourceType = AL_UNDETERMINED; Source->queue = NULL; } UnlockBufferList(device); /* Delete all elements in the previous queue */ while(oldlist != NULL) { ALsizei i; ALbufferlistitem *temp = oldlist; oldlist = ATOMIC_LOAD(&temp->next, almemory_order_relaxed); for(i = 0;i < temp->num_buffers;i++) { if(temp->buffers[i]) DecrementRef(&temp->buffers[i]->ref); } al_free(temp); } return AL_TRUE; case AL_SEC_OFFSET: case AL_SAMPLE_OFFSET: case AL_BYTE_OFFSET: CHECKVAL(*values >= 0); Source->OffsetType = prop; Source->Offset = *values; if(IsPlayingOrPaused(Source)) { ALvoice *voice; ALCdevice_Lock(Context->Device); voice = GetSourceVoice(Source, Context); if(voice) { if(ApplyOffset(Source, voice) == AL_FALSE) { ALCdevice_Unlock(Context->Device); SETERR_RETURN(Context, AL_INVALID_VALUE, AL_FALSE, "Invalid source offset"); } } ALCdevice_Unlock(Context->Device); } return AL_TRUE; case AL_DIRECT_FILTER: LockFilterList(device); if(!(*values == 0 || (filter=LookupFilter(device, *values)) != NULL)) { UnlockFilterList(device); SETERR_RETURN(Context, AL_INVALID_VALUE, AL_FALSE, "Invalid filter ID %u", *values); } if(!filter) { Source->Direct.Gain = 1.0f; Source->Direct.GainHF = 1.0f; Source->Direct.HFReference = LOWPASSFREQREF; Source->Direct.GainLF = 1.0f; Source->Direct.LFReference = HIGHPASSFREQREF; } else { Source->Direct.Gain = filter->Gain; Source->Direct.GainHF = filter->GainHF; Source->Direct.HFReference = filter->HFReference; Source->Direct.GainLF = filter->GainLF; Source->Direct.LFReference = filter->LFReference; } UnlockFilterList(device); DO_UPDATEPROPS(); return AL_TRUE; case AL_DIRECT_FILTER_GAINHF_AUTO: CHECKVAL(*values == AL_FALSE || *values == AL_TRUE); Source->DryGainHFAuto = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_AUXILIARY_SEND_FILTER_GAIN_AUTO: CHECKVAL(*values == AL_FALSE || *values == AL_TRUE); Source->WetGainAuto = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO: CHECKVAL(*values == AL_FALSE || *values == AL_TRUE); Source->WetGainHFAuto = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_DIRECT_CHANNELS_SOFT: CHECKVAL(*values == AL_FALSE || *values == AL_TRUE); Source->DirectChannels = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_DISTANCE_MODEL: CHECKVAL(*values == AL_NONE || *values == AL_INVERSE_DISTANCE || *values == AL_INVERSE_DISTANCE_CLAMPED || *values == AL_LINEAR_DISTANCE || *values == AL_LINEAR_DISTANCE_CLAMPED || *values == AL_EXPONENT_DISTANCE || *values == AL_EXPONENT_DISTANCE_CLAMPED); Source->DistanceModel = *values; if(Context->SourceDistanceModel) DO_UPDATEPROPS(); return AL_TRUE; case AL_SOURCE_RESAMPLER_SOFT: CHECKVAL(*values >= 0 && *values <= ResamplerMax); Source->Resampler = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_SOURCE_SPATIALIZE_SOFT: CHECKVAL(*values >= AL_FALSE && *values <= AL_AUTO_SOFT); Source->Spatialize = *values; DO_UPDATEPROPS(); return AL_TRUE; case AL_AUXILIARY_SEND_FILTER: LockEffectSlotList(Context); if(!(values[0] == 0 || (slot=LookupEffectSlot(Context, values[0])) != NULL)) { UnlockEffectSlotList(Context); SETERR_RETURN(Context, AL_INVALID_VALUE, AL_FALSE, "Invalid effect ID %u", values[0]); } if(!((ALuint)values[1] < (ALuint)device->NumAuxSends)) { UnlockEffectSlotList(Context); SETERR_RETURN(Context, AL_INVALID_VALUE, AL_FALSE, "Invalid send %u", values[1]); } LockFilterList(device); if(!(values[2] == 0 || (filter=LookupFilter(device, values[2])) != NULL)) { UnlockFilterList(device); UnlockEffectSlotList(Context); SETERR_RETURN(Context, AL_INVALID_VALUE, AL_FALSE, "Invalid filter ID %u", values[2]); } if(!filter) { /* Disable filter */ Source->Send[values[1]].Gain = 1.0f; Source->Send[values[1]].GainHF = 1.0f; Source->Send[values[1]].HFReference = LOWPASSFREQREF; Source->Send[values[1]].GainLF = 1.0f; Source->Send[values[1]].LFReference = HIGHPASSFREQREF; } else { Source->Send[values[1]].Gain = filter->Gain; Source->Send[values[1]].GainHF = filter->GainHF; Source->Send[values[1]].HFReference = filter->HFReference; Source->Send[values[1]].GainLF = filter->GainLF; Source->Send[values[1]].LFReference = filter->LFReference; } UnlockFilterList(device); if(slot != Source->Send[values[1]].Slot && IsPlayingOrPaused(Source)) { ALvoice *voice; /* Add refcount on the new slot, and release the previous slot */ if(slot) IncrementRef(&slot->ref); if(Source->Send[values[1]].Slot) DecrementRef(&Source->Send[values[1]].Slot->ref); Source->Send[values[1]].Slot = slot; /* We must force an update if the auxiliary slot changed on an * active source, in case the slot is about to be deleted. */ if((voice=GetSourceVoice(Source, Context)) != NULL) UpdateSourceProps(Source, voice, device->NumAuxSends, Context); else ATOMIC_FLAG_CLEAR(&Source->PropsClean, almemory_order_release); } else { if(slot) IncrementRef(&slot->ref); if(Source->Send[values[1]].Slot) DecrementRef(&Source->Send[values[1]].Slot->ref); Source->Send[values[1]].Slot = slot; DO_UPDATEPROPS(); } UnlockEffectSlotList(Context); return AL_TRUE; /* 1x float */ case AL_CONE_INNER_ANGLE: case AL_CONE_OUTER_ANGLE: case AL_PITCH: case AL_GAIN: case AL_MIN_GAIN: case AL_MAX_GAIN: case AL_REFERENCE_DISTANCE: case AL_ROLLOFF_FACTOR: case AL_CONE_OUTER_GAIN: case AL_MAX_DISTANCE: case AL_DOPPLER_FACTOR: case AL_CONE_OUTER_GAINHF: case AL_AIR_ABSORPTION_FACTOR: case AL_ROOM_ROLLOFF_FACTOR: case AL_SOURCE_RADIUS: fvals[0] = (ALfloat)*values; return SetSourcefv(Source, Context, (int)prop, fvals); /* 3x float */ case AL_POSITION: case AL_VELOCITY: case AL_DIRECTION: fvals[0] = (ALfloat)values[0]; fvals[1] = (ALfloat)values[1]; fvals[2] = (ALfloat)values[2]; return SetSourcefv(Source, Context, (int)prop, fvals); /* 6x float */ case AL_ORIENTATION: fvals[0] = (ALfloat)values[0]; fvals[1] = (ALfloat)values[1]; fvals[2] = (ALfloat)values[2]; fvals[3] = (ALfloat)values[3]; fvals[4] = (ALfloat)values[4]; fvals[5] = (ALfloat)values[5]; return SetSourcefv(Source, Context, (int)prop, fvals); case AL_SAMPLE_OFFSET_LATENCY_SOFT: case AL_SEC_OFFSET_LATENCY_SOFT: case AL_SEC_OFFSET_CLOCK_SOFT: case AL_SAMPLE_OFFSET_CLOCK_SOFT: case AL_STEREO_ANGLES: break; } ERR("Unexpected property: 0x%04x\n", prop); SETERR_RETURN(Context, AL_INVALID_ENUM, AL_FALSE, "Invalid source integer property 0x%04x", prop); } static ALboolean SetSourcei64v(ALsource *Source, ALCcontext *Context, SourceProp prop, const ALint64SOFT *values) { ALfloat fvals[6]; ALint ivals[3]; switch(prop) { case AL_SOURCE_TYPE: case AL_BUFFERS_QUEUED: case AL_BUFFERS_PROCESSED: case AL_SOURCE_STATE: case AL_SAMPLE_OFFSET_LATENCY_SOFT: case AL_SAMPLE_OFFSET_CLOCK_SOFT: /* Query only */ SETERR_RETURN(Context, AL_INVALID_OPERATION, AL_FALSE, "Setting read-only source property 0x%04x", prop); /* 1x int */ case AL_SOURCE_RELATIVE: case AL_LOOPING: case AL_SEC_OFFSET: case AL_SAMPLE_OFFSET: case AL_BYTE_OFFSET: case AL_DIRECT_FILTER_GAINHF_AUTO: case AL_AUXILIARY_SEND_FILTER_GAIN_AUTO: case AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO: case AL_DIRECT_CHANNELS_SOFT: case AL_DISTANCE_MODEL: case AL_SOURCE_RESAMPLER_SOFT: case AL_SOURCE_SPATIALIZE_SOFT: CHECKVAL(*values <= INT_MAX && *values >= INT_MIN); ivals[0] = (ALint)*values; return SetSourceiv(Source, Context, (int)prop, ivals); /* 1x uint */ case AL_BUFFER: case AL_DIRECT_FILTER: CHECKVAL(*values <= UINT_MAX && *values >= 0); ivals[0] = (ALuint)*values; return SetSourceiv(Source, Context, (int)prop, ivals); /* 3x uint */ case AL_AUXILIARY_SEND_FILTER: CHECKVAL(values[0] <= UINT_MAX && values[0] >= 0 && values[1] <= UINT_MAX && values[1] >= 0 && values[2] <= UINT_MAX && values[2] >= 0); ivals[0] = (ALuint)values[0]; ivals[1] = (ALuint)values[1]; ivals[2] = (ALuint)values[2]; return SetSourceiv(Source, Context, (int)prop, ivals); /* 1x float */ case AL_CONE_INNER_ANGLE: case AL_CONE_OUTER_ANGLE: case AL_PITCH: case AL_GAIN: case AL_MIN_GAIN: case AL_MAX_GAIN: case AL_REFERENCE_DISTANCE: case AL_ROLLOFF_FACTOR: case AL_CONE_OUTER_GAIN: case AL_MAX_DISTANCE: case AL_DOPPLER_FACTOR: case AL_CONE_OUTER_GAINHF: case AL_AIR_ABSORPTION_FACTOR: case AL_ROOM_ROLLOFF_FACTOR: case AL_SOURCE_RADIUS: fvals[0] = (ALfloat)*values; return SetSourcefv(Source, Context, (int)prop, fvals); /* 3x float */ case AL_POSITION: case AL_VELOCITY: case AL_DIRECTION: fvals[0] = (ALfloat)values[0]; fvals[1] = (ALfloat)values[1]; fvals[2] = (ALfloat)values[2]; return SetSourcefv(Source, Context, (int)prop, fvals); /* 6x float */ case AL_ORIENTATION: fvals[0] = (ALfloat)values[0]; fvals[1] = (ALfloat)values[1]; fvals[2] = (ALfloat)values[2]; fvals[3] = (ALfloat)values[3]; fvals[4] = (ALfloat)values[4]; fvals[5] = (ALfloat)values[5]; return SetSourcefv(Source, Context, (int)prop, fvals); case AL_SEC_OFFSET_LATENCY_SOFT: case AL_SEC_OFFSET_CLOCK_SOFT: case AL_STEREO_ANGLES: break; } ERR("Unexpected property: 0x%04x\n", prop); SETERR_RETURN(Context, AL_INVALID_ENUM, AL_FALSE, "Invalid source integer64 property 0x%04x", prop); } #undef CHECKVAL static ALboolean GetSourcedv(ALsource *Source, ALCcontext *Context, SourceProp prop, ALdouble *values) { ALCdevice *device = Context->Device; ClockLatency clocktime; ALuint64 srcclock; ALint ivals[3]; ALboolean err; switch(prop) { case AL_GAIN: *values = Source->Gain; return AL_TRUE; case AL_PITCH: *values = Source->Pitch; return AL_TRUE; case AL_MAX_DISTANCE: *values = Source->MaxDistance; return AL_TRUE; case AL_ROLLOFF_FACTOR: *values = Source->RolloffFactor; return AL_TRUE; case AL_REFERENCE_DISTANCE: *values = Source->RefDistance; return AL_TRUE; case AL_CONE_INNER_ANGLE: *values = Source->InnerAngle; return AL_TRUE; case AL_CONE_OUTER_ANGLE: *values = Source->OuterAngle; return AL_TRUE; case AL_MIN_GAIN: *values = Source->MinGain; return AL_TRUE; case AL_MAX_GAIN: *values = Source->MaxGain; return AL_TRUE; case AL_CONE_OUTER_GAIN: *values = Source->OuterGain; return AL_TRUE; case AL_SEC_OFFSET: case AL_SAMPLE_OFFSET: case AL_BYTE_OFFSET: *values = GetSourceOffset(Source, prop, Context); return AL_TRUE; case AL_CONE_OUTER_GAINHF: *values = Source->OuterGainHF; return AL_TRUE; case AL_AIR_ABSORPTION_FACTOR: *values = Source->AirAbsorptionFactor; return AL_TRUE; case AL_ROOM_ROLLOFF_FACTOR: *values = Source->RoomRolloffFactor; return AL_TRUE; case AL_DOPPLER_FACTOR: *values = Source->DopplerFactor; return AL_TRUE; case AL_SOURCE_RADIUS: *values = Source->Radius; return AL_TRUE; case AL_STEREO_ANGLES: values[0] = Source->StereoPan[0]; values[1] = Source->StereoPan[1]; return AL_TRUE; case AL_SEC_OFFSET_LATENCY_SOFT: /* Get the source offset with the clock time first. Then get the * clock time with the device latency. Order is important. */ values[0] = GetSourceSecOffset(Source, Context, &srcclock); almtx_lock(&device->BackendLock); clocktime = GetClockLatency(device); almtx_unlock(&device->BackendLock); if(srcclock == (ALuint64)clocktime.ClockTime) values[1] = (ALdouble)clocktime.Latency / 1000000000.0; else { /* If the clock time incremented, reduce the latency by that * much since it's that much closer to the source offset it got * earlier. */ ALuint64 diff = clocktime.ClockTime - srcclock; values[1] = (ALdouble)(clocktime.Latency - minu64(clocktime.Latency, diff)) / 1000000000.0; } return AL_TRUE; case AL_SEC_OFFSET_CLOCK_SOFT: values[0] = GetSourceSecOffset(Source, Context, &srcclock); values[1] = srcclock / 1000000000.0; return AL_TRUE; case AL_POSITION: values[0] = Source->Position[0]; values[1] = Source->Position[1]; values[2] = Source->Position[2]; return AL_TRUE; case AL_VELOCITY: values[0] = Source->Velocity[0]; values[1] = Source->Velocity[1]; values[2] = Source->Velocity[2]; return AL_TRUE; case AL_DIRECTION: values[0] = Source->Direction[0]; values[1] = Source->Direction[1]; values[2] = Source->Direction[2]; return AL_TRUE; case AL_ORIENTATION: values[0] = Source->Orientation[0][0]; values[1] = Source->Orientation[0][1]; values[2] = Source->Orientation[0][2]; values[3] = Source->Orientation[1][0]; values[4] = Source->Orientation[1][1]; values[5] = Source->Orientation[1][2]; return AL_TRUE; /* 1x int */ case AL_SOURCE_RELATIVE: case AL_LOOPING: case AL_SOURCE_STATE: case AL_BUFFERS_QUEUED: case AL_BUFFERS_PROCESSED: case AL_SOURCE_TYPE: case AL_DIRECT_FILTER_GAINHF_AUTO: case AL_AUXILIARY_SEND_FILTER_GAIN_AUTO: case AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO: case AL_DIRECT_CHANNELS_SOFT: case AL_DISTANCE_MODEL: case AL_SOURCE_RESAMPLER_SOFT: case AL_SOURCE_SPATIALIZE_SOFT: if((err=GetSourceiv(Source, Context, (int)prop, ivals)) != AL_FALSE) *values = (ALdouble)ivals[0]; return err; case AL_BUFFER: case AL_DIRECT_FILTER: case AL_AUXILIARY_SEND_FILTER: case AL_SAMPLE_OFFSET_LATENCY_SOFT: case AL_SAMPLE_OFFSET_CLOCK_SOFT: break; } ERR("Unexpected property: 0x%04x\n", prop); SETERR_RETURN(Context, AL_INVALID_ENUM, AL_FALSE, "Invalid source double property 0x%04x", prop); } static ALboolean GetSourceiv(ALsource *Source, ALCcontext *Context, SourceProp prop, ALint *values) { ALbufferlistitem *BufferList; ALdouble dvals[6]; ALboolean err; switch(prop) { case AL_SOURCE_RELATIVE: *values = Source->HeadRelative; return AL_TRUE; case AL_LOOPING: *values = Source->Looping; return AL_TRUE; case AL_BUFFER: BufferList = (Source->SourceType == AL_STATIC) ? Source->queue : NULL; *values = (BufferList && BufferList->num_buffers >= 1 && BufferList->buffers[0]) ? BufferList->buffers[0]->id : 0; return AL_TRUE; case AL_SOURCE_STATE: *values = GetSourceState(Source, GetSourceVoice(Source, Context)); return AL_TRUE; case AL_BUFFERS_QUEUED: if(!(BufferList=Source->queue)) *values = 0; else { ALsizei count = 0; do { count += BufferList->num_buffers; BufferList = ATOMIC_LOAD(&BufferList->next, almemory_order_relaxed); } while(BufferList != NULL); *values = count; } return AL_TRUE; case AL_BUFFERS_PROCESSED: if(Source->Looping || Source->SourceType != AL_STREAMING) { /* Buffers on a looping source are in a perpetual state of * PENDING, so don't report any as PROCESSED */ *values = 0; } else { const ALbufferlistitem *BufferList = Source->queue; const ALbufferlistitem *Current = NULL; ALsizei played = 0; ALvoice *voice; if((voice=GetSourceVoice(Source, Context)) != NULL) Current = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed); else if(Source->state == AL_INITIAL) Current = BufferList; while(BufferList && BufferList != Current) { played += BufferList->num_buffers; BufferList = ATOMIC_LOAD(&CONST_CAST(ALbufferlistitem*,BufferList)->next, almemory_order_relaxed); } *values = played; } return AL_TRUE; case AL_SOURCE_TYPE: *values = Source->SourceType; return AL_TRUE; case AL_DIRECT_FILTER_GAINHF_AUTO: *values = Source->DryGainHFAuto; return AL_TRUE; case AL_AUXILIARY_SEND_FILTER_GAIN_AUTO: *values = Source->WetGainAuto; return AL_TRUE; case AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO: *values = Source->WetGainHFAuto; return AL_TRUE; case AL_DIRECT_CHANNELS_SOFT: *values = Source->DirectChannels; return AL_TRUE; case AL_DISTANCE_MODEL: *values = Source->DistanceModel; return AL_TRUE; case AL_SOURCE_RESAMPLER_SOFT: *values = Source->Resampler; return AL_TRUE; case AL_SOURCE_SPATIALIZE_SOFT: *values = Source->Spatialize; return AL_TRUE; /* 1x float/double */ case AL_CONE_INNER_ANGLE: case AL_CONE_OUTER_ANGLE: case AL_PITCH: case AL_GAIN: case AL_MIN_GAIN: case AL_MAX_GAIN: case AL_REFERENCE_DISTANCE: case AL_ROLLOFF_FACTOR: case AL_CONE_OUTER_GAIN: case AL_MAX_DISTANCE: case AL_SEC_OFFSET: case AL_SAMPLE_OFFSET: case AL_BYTE_OFFSET: case AL_DOPPLER_FACTOR: case AL_AIR_ABSORPTION_FACTOR: case AL_ROOM_ROLLOFF_FACTOR: case AL_CONE_OUTER_GAINHF: case AL_SOURCE_RADIUS: if((err=GetSourcedv(Source, Context, prop, dvals)) != AL_FALSE) *values = (ALint)dvals[0]; return err; /* 3x float/double */ case AL_POSITION: case AL_VELOCITY: case AL_DIRECTION: if((err=GetSourcedv(Source, Context, prop, dvals)) != AL_FALSE) { values[0] = (ALint)dvals[0]; values[1] = (ALint)dvals[1]; values[2] = (ALint)dvals[2]; } return err; /* 6x float/double */ case AL_ORIENTATION: if((err=GetSourcedv(Source, Context, prop, dvals)) != AL_FALSE) { values[0] = (ALint)dvals[0]; values[1] = (ALint)dvals[1]; values[2] = (ALint)dvals[2]; values[3] = (ALint)dvals[3]; values[4] = (ALint)dvals[4]; values[5] = (ALint)dvals[5]; } return err; case AL_SAMPLE_OFFSET_LATENCY_SOFT: case AL_SAMPLE_OFFSET_CLOCK_SOFT: break; /* i64 only */ case AL_SEC_OFFSET_LATENCY_SOFT: case AL_SEC_OFFSET_CLOCK_SOFT: break; /* Double only */ case AL_STEREO_ANGLES: break; /* Float/double only */ case AL_DIRECT_FILTER: case AL_AUXILIARY_SEND_FILTER: break; /* ??? */ } ERR("Unexpected property: 0x%04x\n", prop); SETERR_RETURN(Context, AL_INVALID_ENUM, AL_FALSE, "Invalid source integer property 0x%04x", prop); } static ALboolean GetSourcei64v(ALsource *Source, ALCcontext *Context, SourceProp prop, ALint64 *values) { ALCdevice *device = Context->Device; ClockLatency clocktime; ALuint64 srcclock; ALdouble dvals[6]; ALint ivals[3]; ALboolean err; switch(prop) { case AL_SAMPLE_OFFSET_LATENCY_SOFT: /* Get the source offset with the clock time first. Then get the * clock time with the device latency. Order is important. */ values[0] = GetSourceSampleOffset(Source, Context, &srcclock); almtx_lock(&device->BackendLock); clocktime = GetClockLatency(device); almtx_unlock(&device->BackendLock); if(srcclock == (ALuint64)clocktime.ClockTime) values[1] = clocktime.Latency; else { /* If the clock time incremented, reduce the latency by that * much since it's that much closer to the source offset it got * earlier. */ ALuint64 diff = clocktime.ClockTime - srcclock; values[1] = clocktime.Latency - minu64(clocktime.Latency, diff); } return AL_TRUE; case AL_SAMPLE_OFFSET_CLOCK_SOFT: values[0] = GetSourceSampleOffset(Source, Context, &srcclock); values[1] = srcclock; return AL_TRUE; /* 1x float/double */ case AL_CONE_INNER_ANGLE: case AL_CONE_OUTER_ANGLE: case AL_PITCH: case AL_GAIN: case AL_MIN_GAIN: case AL_MAX_GAIN: case AL_REFERENCE_DISTANCE: case AL_ROLLOFF_FACTOR: case AL_CONE_OUTER_GAIN: case AL_MAX_DISTANCE: case AL_SEC_OFFSET: case AL_SAMPLE_OFFSET: case AL_BYTE_OFFSET: case AL_DOPPLER_FACTOR: case AL_AIR_ABSORPTION_FACTOR: case AL_ROOM_ROLLOFF_FACTOR: case AL_CONE_OUTER_GAINHF: case AL_SOURCE_RADIUS: if((err=GetSourcedv(Source, Context, prop, dvals)) != AL_FALSE) *values = (ALint64)dvals[0]; return err; /* 3x float/double */ case AL_POSITION: case AL_VELOCITY: case AL_DIRECTION: if((err=GetSourcedv(Source, Context, prop, dvals)) != AL_FALSE) { values[0] = (ALint64)dvals[0]; values[1] = (ALint64)dvals[1]; values[2] = (ALint64)dvals[2]; } return err; /* 6x float/double */ case AL_ORIENTATION: if((err=GetSourcedv(Source, Context, prop, dvals)) != AL_FALSE) { values[0] = (ALint64)dvals[0]; values[1] = (ALint64)dvals[1]; values[2] = (ALint64)dvals[2]; values[3] = (ALint64)dvals[3]; values[4] = (ALint64)dvals[4]; values[5] = (ALint64)dvals[5]; } return err; /* 1x int */ case AL_SOURCE_RELATIVE: case AL_LOOPING: case AL_SOURCE_STATE: case AL_BUFFERS_QUEUED: case AL_BUFFERS_PROCESSED: case AL_SOURCE_TYPE: case AL_DIRECT_FILTER_GAINHF_AUTO: case AL_AUXILIARY_SEND_FILTER_GAIN_AUTO: case AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO: case AL_DIRECT_CHANNELS_SOFT: case AL_DISTANCE_MODEL: case AL_SOURCE_RESAMPLER_SOFT: case AL_SOURCE_SPATIALIZE_SOFT: if((err=GetSourceiv(Source, Context, prop, ivals)) != AL_FALSE) *values = ivals[0]; return err; /* 1x uint */ case AL_BUFFER: case AL_DIRECT_FILTER: if((err=GetSourceiv(Source, Context, prop, ivals)) != AL_FALSE) *values = (ALuint)ivals[0]; return err; /* 3x uint */ case AL_AUXILIARY_SEND_FILTER: if((err=GetSourceiv(Source, Context, prop, ivals)) != AL_FALSE) { values[0] = (ALuint)ivals[0]; values[1] = (ALuint)ivals[1]; values[2] = (ALuint)ivals[2]; } return err; case AL_SEC_OFFSET_LATENCY_SOFT: case AL_SEC_OFFSET_CLOCK_SOFT: break; /* Double only */ case AL_STEREO_ANGLES: break; /* Float/double only */ } ERR("Unexpected property: 0x%04x\n", prop); SETERR_RETURN(Context, AL_INVALID_ENUM, AL_FALSE, "Invalid source integer64 property 0x%04x", prop); } AL_API ALvoid AL_APIENTRY alGenSources(ALsizei n, ALuint *sources) { ALCcontext *context; ALsizei cur = 0; context = GetContextRef(); if(!context) return; if(!(n >= 0)) alSetError(context, AL_INVALID_VALUE, "Generating %d sources", n); else for(cur = 0;cur < n;cur++) { ALsource *source = AllocSource(context); if(!source) { alDeleteSources(cur, sources); break; } sources[cur] = source->id; } ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alDeleteSources(ALsizei n, const ALuint *sources) { ALCcontext *context; ALsource *Source; ALsizei i; context = GetContextRef(); if(!context) return; LockSourceList(context); if(!(n >= 0)) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Deleting %d sources", n); /* Check that all Sources are valid */ for(i = 0;i < n;i++) { if(LookupSource(context, sources[i]) == NULL) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid source ID %u", sources[i]); } for(i = 0;i < n;i++) { if((Source=LookupSource(context, sources[i])) != NULL) FreeSource(context, Source); } done: UnlockSourceList(context); ALCcontext_DecRef(context); } AL_API ALboolean AL_APIENTRY alIsSource(ALuint source) { ALCcontext *context; ALboolean ret; context = GetContextRef(); if(!context) return AL_FALSE; LockSourceList(context); ret = (LookupSource(context, source) ? AL_TRUE : AL_FALSE); UnlockSourceList(context); ALCcontext_DecRef(context); return ret; } AL_API ALvoid AL_APIENTRY alSourcef(ALuint source, ALenum param, ALfloat value) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; almtx_lock(&Context->PropLock); LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!(FloatValsByProp(param) == 1)) alSetError(Context, AL_INVALID_ENUM, "Invalid float property 0x%04x", param); else SetSourcefv(Source, Context, param, &value); UnlockSourceList(Context); almtx_unlock(&Context->PropLock); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alSource3f(ALuint source, ALenum param, ALfloat value1, ALfloat value2, ALfloat value3) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; almtx_lock(&Context->PropLock); LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!(FloatValsByProp(param) == 3)) alSetError(Context, AL_INVALID_ENUM, "Invalid 3-float property 0x%04x", param); else { ALfloat fvals[3] = { value1, value2, value3 }; SetSourcefv(Source, Context, param, fvals); } UnlockSourceList(Context); almtx_unlock(&Context->PropLock); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alSourcefv(ALuint source, ALenum param, const ALfloat *values) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; almtx_lock(&Context->PropLock); LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!values) alSetError(Context, AL_INVALID_VALUE, "NULL pointer"); else if(!(FloatValsByProp(param) > 0)) alSetError(Context, AL_INVALID_ENUM, "Invalid float-vector property 0x%04x", param); else SetSourcefv(Source, Context, param, values); UnlockSourceList(Context); almtx_unlock(&Context->PropLock); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alSourcedSOFT(ALuint source, ALenum param, ALdouble value) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; almtx_lock(&Context->PropLock); LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!(DoubleValsByProp(param) == 1)) alSetError(Context, AL_INVALID_ENUM, "Invalid double property 0x%04x", param); else { ALfloat fval = (ALfloat)value; SetSourcefv(Source, Context, param, &fval); } UnlockSourceList(Context); almtx_unlock(&Context->PropLock); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alSource3dSOFT(ALuint source, ALenum param, ALdouble value1, ALdouble value2, ALdouble value3) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; almtx_lock(&Context->PropLock); LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!(DoubleValsByProp(param) == 3)) alSetError(Context, AL_INVALID_ENUM, "Invalid 3-double property 0x%04x", param); else { ALfloat fvals[3] = { (ALfloat)value1, (ALfloat)value2, (ALfloat)value3 }; SetSourcefv(Source, Context, param, fvals); } UnlockSourceList(Context); almtx_unlock(&Context->PropLock); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alSourcedvSOFT(ALuint source, ALenum param, const ALdouble *values) { ALCcontext *Context; ALsource *Source; ALint count; Context = GetContextRef(); if(!Context) return; almtx_lock(&Context->PropLock); LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!values) alSetError(Context, AL_INVALID_VALUE, "NULL pointer"); else if(!((count=DoubleValsByProp(param)) > 0 && count <= 6)) alSetError(Context, AL_INVALID_ENUM, "Invalid double-vector property 0x%04x", param); else { ALfloat fvals[6]; ALint i; for(i = 0;i < count;i++) fvals[i] = (ALfloat)values[i]; SetSourcefv(Source, Context, param, fvals); } UnlockSourceList(Context); almtx_unlock(&Context->PropLock); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alSourcei(ALuint source, ALenum param, ALint value) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; almtx_lock(&Context->PropLock); LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!(IntValsByProp(param) == 1)) alSetError(Context, AL_INVALID_ENUM, "Invalid integer property 0x%04x", param); else SetSourceiv(Source, Context, param, &value); UnlockSourceList(Context); almtx_unlock(&Context->PropLock); ALCcontext_DecRef(Context); } AL_API void AL_APIENTRY alSource3i(ALuint source, ALenum param, ALint value1, ALint value2, ALint value3) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; almtx_lock(&Context->PropLock); LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!(IntValsByProp(param) == 3)) alSetError(Context, AL_INVALID_ENUM, "Invalid 3-integer property 0x%04x", param); else { ALint ivals[3] = { value1, value2, value3 }; SetSourceiv(Source, Context, param, ivals); } UnlockSourceList(Context); almtx_unlock(&Context->PropLock); ALCcontext_DecRef(Context); } AL_API void AL_APIENTRY alSourceiv(ALuint source, ALenum param, const ALint *values) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; almtx_lock(&Context->PropLock); LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!values) alSetError(Context, AL_INVALID_VALUE, "NULL pointer"); else if(!(IntValsByProp(param) > 0)) alSetError(Context, AL_INVALID_ENUM, "Invalid integer-vector property 0x%04x", param); else SetSourceiv(Source, Context, param, values); UnlockSourceList(Context); almtx_unlock(&Context->PropLock); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alSourcei64SOFT(ALuint source, ALenum param, ALint64SOFT value) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; almtx_lock(&Context->PropLock); LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!(Int64ValsByProp(param) == 1)) alSetError(Context, AL_INVALID_ENUM, "Invalid integer64 property 0x%04x", param); else SetSourcei64v(Source, Context, param, &value); UnlockSourceList(Context); almtx_unlock(&Context->PropLock); ALCcontext_DecRef(Context); } AL_API void AL_APIENTRY alSource3i64SOFT(ALuint source, ALenum param, ALint64SOFT value1, ALint64SOFT value2, ALint64SOFT value3) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; almtx_lock(&Context->PropLock); LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!(Int64ValsByProp(param) == 3)) alSetError(Context, AL_INVALID_ENUM, "Invalid 3-integer64 property 0x%04x", param); else { ALint64SOFT i64vals[3] = { value1, value2, value3 }; SetSourcei64v(Source, Context, param, i64vals); } UnlockSourceList(Context); almtx_unlock(&Context->PropLock); ALCcontext_DecRef(Context); } AL_API void AL_APIENTRY alSourcei64vSOFT(ALuint source, ALenum param, const ALint64SOFT *values) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; almtx_lock(&Context->PropLock); LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!values) alSetError(Context, AL_INVALID_VALUE, "NULL pointer"); else if(!(Int64ValsByProp(param) > 0)) alSetError(Context, AL_INVALID_ENUM, "Invalid integer64-vector property 0x%04x", param); else SetSourcei64v(Source, Context, param, values); UnlockSourceList(Context); almtx_unlock(&Context->PropLock); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alGetSourcef(ALuint source, ALenum param, ALfloat *value) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!value) alSetError(Context, AL_INVALID_VALUE, "NULL pointer"); else if(!(FloatValsByProp(param) == 1)) alSetError(Context, AL_INVALID_ENUM, "Invalid float property 0x%04x", param); else { ALdouble dval; if(GetSourcedv(Source, Context, param, &dval)) *value = (ALfloat)dval; } UnlockSourceList(Context); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alGetSource3f(ALuint source, ALenum param, ALfloat *value1, ALfloat *value2, ALfloat *value3) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!(value1 && value2 && value3)) alSetError(Context, AL_INVALID_VALUE, "NULL pointer"); else if(!(FloatValsByProp(param) == 3)) alSetError(Context, AL_INVALID_ENUM, "Invalid 3-float property 0x%04x", param); else { ALdouble dvals[3]; if(GetSourcedv(Source, Context, param, dvals)) { *value1 = (ALfloat)dvals[0]; *value2 = (ALfloat)dvals[1]; *value3 = (ALfloat)dvals[2]; } } UnlockSourceList(Context); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alGetSourcefv(ALuint source, ALenum param, ALfloat *values) { ALCcontext *Context; ALsource *Source; ALint count; Context = GetContextRef(); if(!Context) return; LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!values) alSetError(Context, AL_INVALID_VALUE, "NULL pointer"); else if(!((count=FloatValsByProp(param)) > 0 && count <= 6)) alSetError(Context, AL_INVALID_ENUM, "Invalid float-vector property 0x%04x", param); else { ALdouble dvals[6]; if(GetSourcedv(Source, Context, param, dvals)) { ALint i; for(i = 0;i < count;i++) values[i] = (ALfloat)dvals[i]; } } UnlockSourceList(Context); ALCcontext_DecRef(Context); } AL_API void AL_APIENTRY alGetSourcedSOFT(ALuint source, ALenum param, ALdouble *value) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!value) alSetError(Context, AL_INVALID_VALUE, "NULL pointer"); else if(!(DoubleValsByProp(param) == 1)) alSetError(Context, AL_INVALID_ENUM, "Invalid double property 0x%04x", param); else GetSourcedv(Source, Context, param, value); UnlockSourceList(Context); ALCcontext_DecRef(Context); } AL_API void AL_APIENTRY alGetSource3dSOFT(ALuint source, ALenum param, ALdouble *value1, ALdouble *value2, ALdouble *value3) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!(value1 && value2 && value3)) alSetError(Context, AL_INVALID_VALUE, "NULL pointer"); else if(!(DoubleValsByProp(param) == 3)) alSetError(Context, AL_INVALID_ENUM, "Invalid 3-double property 0x%04x", param); else { ALdouble dvals[3]; if(GetSourcedv(Source, Context, param, dvals)) { *value1 = dvals[0]; *value2 = dvals[1]; *value3 = dvals[2]; } } UnlockSourceList(Context); ALCcontext_DecRef(Context); } AL_API void AL_APIENTRY alGetSourcedvSOFT(ALuint source, ALenum param, ALdouble *values) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!values) alSetError(Context, AL_INVALID_VALUE, "NULL pointer"); else if(!(DoubleValsByProp(param) > 0)) alSetError(Context, AL_INVALID_ENUM, "Invalid double-vector property 0x%04x", param); else GetSourcedv(Source, Context, param, values); UnlockSourceList(Context); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alGetSourcei(ALuint source, ALenum param, ALint *value) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!value) alSetError(Context, AL_INVALID_VALUE, "NULL pointer"); else if(!(IntValsByProp(param) == 1)) alSetError(Context, AL_INVALID_ENUM, "Invalid integer property 0x%04x", param); else GetSourceiv(Source, Context, param, value); UnlockSourceList(Context); ALCcontext_DecRef(Context); } AL_API void AL_APIENTRY alGetSource3i(ALuint source, ALenum param, ALint *value1, ALint *value2, ALint *value3) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!(value1 && value2 && value3)) alSetError(Context, AL_INVALID_VALUE, "NULL pointer"); else if(!(IntValsByProp(param) == 3)) alSetError(Context, AL_INVALID_ENUM, "Invalid 3-integer property 0x%04x", param); else { ALint ivals[3]; if(GetSourceiv(Source, Context, param, ivals)) { *value1 = ivals[0]; *value2 = ivals[1]; *value3 = ivals[2]; } } UnlockSourceList(Context); ALCcontext_DecRef(Context); } AL_API void AL_APIENTRY alGetSourceiv(ALuint source, ALenum param, ALint *values) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!values) alSetError(Context, AL_INVALID_VALUE, "NULL pointer"); else if(!(IntValsByProp(param) > 0)) alSetError(Context, AL_INVALID_ENUM, "Invalid integer-vector property 0x%04x", param); else GetSourceiv(Source, Context, param, values); UnlockSourceList(Context); ALCcontext_DecRef(Context); } AL_API void AL_APIENTRY alGetSourcei64SOFT(ALuint source, ALenum param, ALint64SOFT *value) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!value) alSetError(Context, AL_INVALID_VALUE, "NULL pointer"); else if(!(Int64ValsByProp(param) == 1)) alSetError(Context, AL_INVALID_ENUM, "Invalid integer64 property 0x%04x", param); else GetSourcei64v(Source, Context, param, value); UnlockSourceList(Context); ALCcontext_DecRef(Context); } AL_API void AL_APIENTRY alGetSource3i64SOFT(ALuint source, ALenum param, ALint64SOFT *value1, ALint64SOFT *value2, ALint64SOFT *value3) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!(value1 && value2 && value3)) alSetError(Context, AL_INVALID_VALUE, "NULL pointer"); else if(!(Int64ValsByProp(param) == 3)) alSetError(Context, AL_INVALID_ENUM, "Invalid 3-integer64 property 0x%04x", param); else { ALint64 i64vals[3]; if(GetSourcei64v(Source, Context, param, i64vals)) { *value1 = i64vals[0]; *value2 = i64vals[1]; *value3 = i64vals[2]; } } UnlockSourceList(Context); ALCcontext_DecRef(Context); } AL_API void AL_APIENTRY alGetSourcei64vSOFT(ALuint source, ALenum param, ALint64SOFT *values) { ALCcontext *Context; ALsource *Source; Context = GetContextRef(); if(!Context) return; LockSourceList(Context); if((Source=LookupSource(Context, source)) == NULL) alSetError(Context, AL_INVALID_NAME, "Invalid source ID %u", source); else if(!values) alSetError(Context, AL_INVALID_VALUE, "NULL pointer"); else if(!(Int64ValsByProp(param) > 0)) alSetError(Context, AL_INVALID_ENUM, "Invalid integer64-vector property 0x%04x", param); else GetSourcei64v(Source, Context, param, values); UnlockSourceList(Context); ALCcontext_DecRef(Context); } AL_API ALvoid AL_APIENTRY alSourcePlay(ALuint source) { alSourcePlayv(1, &source); } AL_API ALvoid AL_APIENTRY alSourcePlayv(ALsizei n, const ALuint *sources) { ALCcontext *context; ALCdevice *device; ALsource *source; ALvoice *voice; ALsizei i, j; context = GetContextRef(); if(!context) return; LockSourceList(context); if(!(n >= 0)) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Playing %d sources", n); for(i = 0;i < n;i++) { if(!LookupSource(context, sources[i])) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid source ID %u", sources[i]); } device = context->Device; ALCdevice_Lock(device); /* If the device is disconnected, go right to stopped. */ if(!ATOMIC_LOAD(&device->Connected, almemory_order_acquire)) { /* TODO: Send state change event? */ for(i = 0;i < n;i++) { source = LookupSource(context, sources[i]); source->OffsetType = AL_NONE; source->Offset = 0.0; source->state = AL_STOPPED; } ALCdevice_Unlock(device); goto done; } while(n > context->MaxVoices-context->VoiceCount) { ALsizei newcount = context->MaxVoices << 1; if(context->MaxVoices >= newcount) { ALCdevice_Unlock(device); SETERR_GOTO(context, AL_OUT_OF_MEMORY, done, "Overflow increasing voice count %d -> %d", context->MaxVoices, newcount); } AllocateVoices(context, newcount, device->NumAuxSends); } for(i = 0;i < n;i++) { ALbufferlistitem *BufferList; bool start_fading = false; ALint vidx = -1; source = LookupSource(context, sources[i]); /* Check that there is a queue containing at least one valid, non zero * length buffer. */ BufferList = source->queue; while(BufferList && BufferList->max_samples == 0) BufferList = ATOMIC_LOAD(&BufferList->next, almemory_order_relaxed); /* If there's nothing to play, go right to stopped. */ if(UNLIKELY(!BufferList)) { /* NOTE: A source without any playable buffers should not have an * ALvoice since it shouldn't be in a playing or paused state. So * there's no need to look up its voice and clear the source. */ ALenum oldstate = GetSourceState(source, NULL); source->OffsetType = AL_NONE; source->Offset = 0.0; if(oldstate != AL_STOPPED) { source->state = AL_STOPPED; SendStateChangeEvent(context, source->id, AL_STOPPED); } continue; } voice = GetSourceVoice(source, context); switch(GetSourceState(source, voice)) { case AL_PLAYING: assert(voice != NULL); /* A source that's already playing is restarted from the beginning. */ ATOMIC_STORE(&voice->current_buffer, BufferList, almemory_order_relaxed); ATOMIC_STORE(&voice->position, 0, almemory_order_relaxed); ATOMIC_STORE(&voice->position_fraction, 0, almemory_order_release); continue; case AL_PAUSED: assert(voice != NULL); /* A source that's paused simply resumes. */ ATOMIC_STORE(&voice->Playing, true, almemory_order_release); source->state = AL_PLAYING; SendStateChangeEvent(context, source->id, AL_PLAYING); continue; default: break; } /* Look for an unused voice to play this source with. */ assert(voice == NULL); for(j = 0;j < context->VoiceCount;j++) { if(ATOMIC_LOAD(&context->Voices[j]->Source, almemory_order_acquire) == NULL) { vidx = j; break; } } if(vidx == -1) vidx = context->VoiceCount++; voice = context->Voices[vidx]; ATOMIC_STORE(&voice->Playing, false, almemory_order_release); ATOMIC_FLAG_TEST_AND_SET(&source->PropsClean, almemory_order_acquire); UpdateSourceProps(source, voice, device->NumAuxSends, context); /* A source that's not playing or paused has any offset applied when it * starts playing. */ if(source->Looping) ATOMIC_STORE(&voice->loop_buffer, source->queue, almemory_order_relaxed); else ATOMIC_STORE(&voice->loop_buffer, NULL, almemory_order_relaxed); ATOMIC_STORE(&voice->current_buffer, BufferList, almemory_order_relaxed); ATOMIC_STORE(&voice->position, 0, almemory_order_relaxed); ATOMIC_STORE(&voice->position_fraction, 0, almemory_order_relaxed); if(ApplyOffset(source, voice) != AL_FALSE) start_fading = ATOMIC_LOAD(&voice->position, almemory_order_relaxed) != 0 || ATOMIC_LOAD(&voice->position_fraction, almemory_order_relaxed) != 0 || ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed) != BufferList; for(j = 0;j < BufferList->num_buffers;j++) { ALbuffer *buffer = BufferList->buffers[j]; if(buffer) { voice->NumChannels = ChannelsFromFmt(buffer->FmtChannels); voice->SampleSize = BytesFromFmt(buffer->FmtType); break; } } /* Clear previous samples. */ memset(voice->PrevSamples, 0, sizeof(voice->PrevSamples)); /* Clear the stepping value so the mixer knows not to mix this until * the update gets applied. */ voice->Step = 0; voice->Flags = start_fading ? VOICE_IS_FADING : 0; if(source->SourceType == AL_STATIC) voice->Flags |= VOICE_IS_STATIC; memset(voice->Direct.Params, 0, sizeof(voice->Direct.Params[0])*voice->NumChannels); for(j = 0;j < device->NumAuxSends;j++) memset(voice->Send[j].Params, 0, sizeof(voice->Send[j].Params[0])*voice->NumChannels); if(device->AvgSpeakerDist > 0.0f) { ALfloat w1 = SPEEDOFSOUNDMETRESPERSEC / (device->AvgSpeakerDist * device->Frequency); for(j = 0;j < voice->NumChannels;j++) NfcFilterCreate(&voice->Direct.Params[j].NFCtrlFilter, 0.0f, w1); } ATOMIC_STORE(&voice->Source, source, almemory_order_relaxed); ATOMIC_STORE(&voice->Playing, true, almemory_order_release); source->state = AL_PLAYING; source->VoiceIdx = vidx; SendStateChangeEvent(context, source->id, AL_PLAYING); } ALCdevice_Unlock(device); done: UnlockSourceList(context); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alSourcePause(ALuint source) { alSourcePausev(1, &source); } AL_API ALvoid AL_APIENTRY alSourcePausev(ALsizei n, const ALuint *sources) { ALCcontext *context; ALCdevice *device; ALsource *source; ALvoice *voice; ALsizei i; context = GetContextRef(); if(!context) return; LockSourceList(context); if(!(n >= 0)) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Pausing %d sources", n); for(i = 0;i < n;i++) { if(!LookupSource(context, sources[i])) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid source ID %u", sources[i]); } device = context->Device; ALCdevice_Lock(device); for(i = 0;i < n;i++) { source = LookupSource(context, sources[i]); if((voice=GetSourceVoice(source, context)) != NULL) ATOMIC_STORE(&voice->Playing, false, almemory_order_release); if(GetSourceState(source, voice) == AL_PLAYING) { source->state = AL_PAUSED; SendStateChangeEvent(context, source->id, AL_PAUSED); } } ALCdevice_Unlock(device); done: UnlockSourceList(context); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alSourceStop(ALuint source) { alSourceStopv(1, &source); } AL_API ALvoid AL_APIENTRY alSourceStopv(ALsizei n, const ALuint *sources) { ALCcontext *context; ALCdevice *device; ALsource *source; ALvoice *voice; ALsizei i; context = GetContextRef(); if(!context) return; LockSourceList(context); if(!(n >= 0)) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Stopping %d sources", n); for(i = 0;i < n;i++) { if(!LookupSource(context, sources[i])) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid source ID %u", sources[i]); } device = context->Device; ALCdevice_Lock(device); for(i = 0;i < n;i++) { ALenum oldstate; source = LookupSource(context, sources[i]); if((voice=GetSourceVoice(source, context)) != NULL) { ATOMIC_STORE(&voice->Source, NULL, almemory_order_relaxed); ATOMIC_STORE(&voice->Playing, false, almemory_order_release); voice = NULL; } oldstate = GetSourceState(source, voice); if(oldstate != AL_INITIAL && oldstate != AL_STOPPED) { source->state = AL_STOPPED; SendStateChangeEvent(context, source->id, AL_STOPPED); } source->OffsetType = AL_NONE; source->Offset = 0.0; } ALCdevice_Unlock(device); done: UnlockSourceList(context); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alSourceRewind(ALuint source) { alSourceRewindv(1, &source); } AL_API ALvoid AL_APIENTRY alSourceRewindv(ALsizei n, const ALuint *sources) { ALCcontext *context; ALCdevice *device; ALsource *source; ALvoice *voice; ALsizei i; context = GetContextRef(); if(!context) return; LockSourceList(context); if(!(n >= 0)) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Rewinding %d sources", n); for(i = 0;i < n;i++) { if(!LookupSource(context, sources[i])) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid source ID %u", sources[i]); } device = context->Device; ALCdevice_Lock(device); for(i = 0;i < n;i++) { source = LookupSource(context, sources[i]); if((voice=GetSourceVoice(source, context)) != NULL) { ATOMIC_STORE(&voice->Source, NULL, almemory_order_relaxed); ATOMIC_STORE(&voice->Playing, false, almemory_order_release); voice = NULL; } if(GetSourceState(source, voice) != AL_INITIAL) { source->state = AL_INITIAL; SendStateChangeEvent(context, source->id, AL_INITIAL); } source->OffsetType = AL_NONE; source->Offset = 0.0; } ALCdevice_Unlock(device); done: UnlockSourceList(context); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alSourceQueueBuffers(ALuint src, ALsizei nb, const ALuint *buffers) { ALCdevice *device; ALCcontext *context; ALsource *source; ALsizei i; ALbufferlistitem *BufferListStart; ALbufferlistitem *BufferList; ALbuffer *BufferFmt = NULL; if(nb == 0) return; context = GetContextRef(); if(!context) return; device = context->Device; LockSourceList(context); if(!(nb >= 0)) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Queueing %d buffers", nb); if((source=LookupSource(context, src)) == NULL) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid source ID %u", src); if(source->SourceType == AL_STATIC) { /* Can't queue on a Static Source */ SETERR_GOTO(context, AL_INVALID_OPERATION, done, "Queueing onto static source %u", src); } /* Check for a valid Buffer, for its frequency and format */ BufferList = source->queue; while(BufferList) { for(i = 0;i < BufferList->num_buffers;i++) { if((BufferFmt=BufferList->buffers[i]) != NULL) break; } if(BufferFmt) break; BufferList = ATOMIC_LOAD(&BufferList->next, almemory_order_relaxed); } LockBufferList(device); BufferListStart = NULL; BufferList = NULL; for(i = 0;i < nb;i++) { ALbuffer *buffer = NULL; if(buffers[i] && (buffer=LookupBuffer(device, buffers[i])) == NULL) SETERR_GOTO(context, AL_INVALID_NAME, buffer_error, "Queueing invalid buffer ID %u", buffers[i]); if(!BufferListStart) { BufferListStart = al_calloc(DEF_ALIGN, FAM_SIZE(ALbufferlistitem, buffers, 1)); BufferList = BufferListStart; } else { ALbufferlistitem *item = al_calloc(DEF_ALIGN, FAM_SIZE(ALbufferlistitem, buffers, 1)); ATOMIC_STORE(&BufferList->next, item, almemory_order_relaxed); BufferList = item; } ATOMIC_INIT(&BufferList->next, NULL); BufferList->max_samples = buffer ? buffer->SampleLen : 0; BufferList->num_buffers = 1; BufferList->buffers[0] = buffer; if(!buffer) continue; IncrementRef(&buffer->ref); if(buffer->MappedAccess != 0 && !(buffer->MappedAccess&AL_MAP_PERSISTENT_BIT_SOFT)) SETERR_GOTO(context, AL_INVALID_OPERATION, buffer_error, "Queueing non-persistently mapped buffer %u", buffer->id); if(BufferFmt == NULL) BufferFmt = buffer; else if(BufferFmt->Frequency != buffer->Frequency || BufferFmt->FmtChannels != buffer->FmtChannels || BufferFmt->OriginalType != buffer->OriginalType) { alSetError(context, AL_INVALID_OPERATION, "Queueing buffer with mismatched format"); buffer_error: /* A buffer failed (invalid ID or format), so unlock and release * each buffer we had. */ while(BufferListStart) { ALbufferlistitem *next = ATOMIC_LOAD(&BufferListStart->next, almemory_order_relaxed); for(i = 0;i < BufferListStart->num_buffers;i++) { if((buffer=BufferListStart->buffers[i]) != NULL) DecrementRef(&buffer->ref); } al_free(BufferListStart); BufferListStart = next; } UnlockBufferList(device); goto done; } } /* All buffers good. */ UnlockBufferList(device); /* Source is now streaming */ source->SourceType = AL_STREAMING; if(!(BufferList=source->queue)) source->queue = BufferListStart; else { ALbufferlistitem *next; while((next=ATOMIC_LOAD(&BufferList->next, almemory_order_relaxed)) != NULL) BufferList = next; ATOMIC_STORE(&BufferList->next, BufferListStart, almemory_order_release); } done: UnlockSourceList(context); ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alSourceQueueBufferLayersSOFT(ALuint src, ALsizei nb, const ALuint *buffers) { ALCdevice *device; ALCcontext *context; ALbufferlistitem *BufferListStart; ALbufferlistitem *BufferList; ALbuffer *BufferFmt = NULL; ALsource *source; ALsizei i; if(nb == 0) return; context = GetContextRef(); if(!context) return; device = context->Device; LockSourceList(context); if(!(nb >= 0 && nb < 16)) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Queueing %d buffer layers", nb); if((source=LookupSource(context, src)) == NULL) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid source ID %u", src); if(source->SourceType == AL_STATIC) { /* Can't queue on a Static Source */ SETERR_GOTO(context, AL_INVALID_OPERATION, done, "Queueing onto static source %u", src); } /* Check for a valid Buffer, for its frequency and format */ BufferList = source->queue; while(BufferList) { for(i = 0;i < BufferList->num_buffers;i++) { if((BufferFmt=BufferList->buffers[i]) != NULL) break; } if(BufferFmt) break; BufferList = ATOMIC_LOAD(&BufferList->next, almemory_order_relaxed); } LockBufferList(device); BufferListStart = al_calloc(DEF_ALIGN, FAM_SIZE(ALbufferlistitem, buffers, nb)); BufferList = BufferListStart; ATOMIC_INIT(&BufferList->next, NULL); BufferList->max_samples = 0; BufferList->num_buffers = 0; for(i = 0;i < nb;i++) { ALbuffer *buffer = NULL; if(buffers[i] && (buffer=LookupBuffer(device, buffers[i])) == NULL) SETERR_GOTO(context, AL_INVALID_NAME, buffer_error, "Queueing invalid buffer ID %u", buffers[i]); BufferList->buffers[BufferList->num_buffers++] = buffer; if(!buffer) continue; IncrementRef(&buffer->ref); BufferList->max_samples = maxi(BufferList->max_samples, buffer->SampleLen); if(buffer->MappedAccess != 0 && !(buffer->MappedAccess&AL_MAP_PERSISTENT_BIT_SOFT)) SETERR_GOTO(context, AL_INVALID_OPERATION, buffer_error, "Queueing non-persistently mapped buffer %u", buffer->id); if(BufferFmt == NULL) BufferFmt = buffer; else if(BufferFmt->Frequency != buffer->Frequency || BufferFmt->FmtChannels != buffer->FmtChannels || BufferFmt->OriginalType != buffer->OriginalType) { alSetError(context, AL_INVALID_OPERATION, "Queueing buffer with mismatched format"); buffer_error: /* A buffer failed (invalid ID or format), so unlock and release * each buffer we had. */ while(BufferListStart) { ALbufferlistitem *next = ATOMIC_LOAD(&BufferListStart->next, almemory_order_relaxed); for(i = 0;i < BufferListStart->num_buffers;i++) { if((buffer=BufferListStart->buffers[i]) != NULL) DecrementRef(&buffer->ref); } al_free(BufferListStart); BufferListStart = next; } UnlockBufferList(device); goto done; } } /* All buffers good. */ UnlockBufferList(device); /* Source is now streaming */ source->SourceType = AL_STREAMING; if(!(BufferList=source->queue)) source->queue = BufferListStart; else { ALbufferlistitem *next; while((next=ATOMIC_LOAD(&BufferList->next, almemory_order_relaxed)) != NULL) BufferList = next; ATOMIC_STORE(&BufferList->next, BufferListStart, almemory_order_release); } done: UnlockSourceList(context); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alSourceUnqueueBuffers(ALuint src, ALsizei nb, ALuint *buffers) { ALCcontext *context; ALsource *source; ALbufferlistitem *BufferList; ALbufferlistitem *Current; ALvoice *voice; ALsizei i; context = GetContextRef(); if(!context) return; LockSourceList(context); if(!(nb >= 0)) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Unqueueing %d buffers", nb); if((source=LookupSource(context, src)) == NULL) SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid source ID %u", src); /* Nothing to unqueue. */ if(nb == 0) goto done; if(source->Looping) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Unqueueing from looping source %u", src); if(source->SourceType != AL_STREAMING) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Unqueueing from a non-streaming source %u", src); /* Make sure enough buffers have been processed to unqueue. */ BufferList = source->queue; Current = NULL; if((voice=GetSourceVoice(source, context)) != NULL) Current = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed); else if(source->state == AL_INITIAL) Current = BufferList; if(BufferList == Current) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Unqueueing pending buffers"); i = BufferList->num_buffers; while(i < nb) { /* If the next bufferlist to check is NULL or is the current one, it's * trying to unqueue pending buffers. */ ALbufferlistitem *next = ATOMIC_LOAD(&BufferList->next, almemory_order_relaxed); if(!next || next == Current) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Unqueueing pending buffers"); BufferList = next; i += BufferList->num_buffers; } while(nb > 0) { ALbufferlistitem *head = source->queue; ALbufferlistitem *next = ATOMIC_LOAD(&head->next, almemory_order_relaxed); for(i = 0;i < head->num_buffers && nb > 0;i++,nb--) { ALbuffer *buffer = head->buffers[i]; if(!buffer) *(buffers++) = 0; else { *(buffers++) = buffer->id; DecrementRef(&buffer->ref); } } if(i < head->num_buffers) { /* This head has some buffers left over, so move them to the front * and update the sample and buffer count. */ ALsizei max_length = 0; ALsizei j = 0; while(i < head->num_buffers) { ALbuffer *buffer = head->buffers[i++]; if(buffer) max_length = maxi(max_length, buffer->SampleLen); head->buffers[j++] = buffer; } head->max_samples = max_length; head->num_buffers = j; break; } /* Otherwise, free this item and set the source queue head to the next * one. */ al_free(head); source->queue = next; } done: UnlockSourceList(context); ALCcontext_DecRef(context); } static void InitSourceParams(ALsource *Source, ALsizei num_sends) { ALsizei i; Source->InnerAngle = 360.0f; Source->OuterAngle = 360.0f; Source->Pitch = 1.0f; Source->Position[0] = 0.0f; Source->Position[1] = 0.0f; Source->Position[2] = 0.0f; Source->Velocity[0] = 0.0f; Source->Velocity[1] = 0.0f; Source->Velocity[2] = 0.0f; Source->Direction[0] = 0.0f; Source->Direction[1] = 0.0f; Source->Direction[2] = 0.0f; Source->Orientation[0][0] = 0.0f; Source->Orientation[0][1] = 0.0f; Source->Orientation[0][2] = -1.0f; Source->Orientation[1][0] = 0.0f; Source->Orientation[1][1] = 1.0f; Source->Orientation[1][2] = 0.0f; Source->RefDistance = 1.0f; Source->MaxDistance = FLT_MAX; Source->RolloffFactor = 1.0f; Source->Gain = 1.0f; Source->MinGain = 0.0f; Source->MaxGain = 1.0f; Source->OuterGain = 0.0f; Source->OuterGainHF = 1.0f; Source->DryGainHFAuto = AL_TRUE; Source->WetGainAuto = AL_TRUE; Source->WetGainHFAuto = AL_TRUE; Source->AirAbsorptionFactor = 0.0f; Source->RoomRolloffFactor = 0.0f; Source->DopplerFactor = 1.0f; Source->HeadRelative = AL_FALSE; Source->Looping = AL_FALSE; Source->DistanceModel = DefaultDistanceModel; Source->Resampler = ResamplerDefault; Source->DirectChannels = AL_FALSE; Source->Spatialize = SpatializeAuto; Source->StereoPan[0] = DEG2RAD( 30.0f); Source->StereoPan[1] = DEG2RAD(-30.0f); Source->Radius = 0.0f; Source->Direct.Gain = 1.0f; Source->Direct.GainHF = 1.0f; Source->Direct.HFReference = LOWPASSFREQREF; Source->Direct.GainLF = 1.0f; Source->Direct.LFReference = HIGHPASSFREQREF; Source->Send = al_calloc(16, num_sends*sizeof(Source->Send[0])); for(i = 0;i < num_sends;i++) { Source->Send[i].Slot = NULL; Source->Send[i].Gain = 1.0f; Source->Send[i].GainHF = 1.0f; Source->Send[i].HFReference = LOWPASSFREQREF; Source->Send[i].GainLF = 1.0f; Source->Send[i].LFReference = HIGHPASSFREQREF; } Source->Offset = 0.0; Source->OffsetType = AL_NONE; Source->SourceType = AL_UNDETERMINED; Source->state = AL_INITIAL; Source->queue = NULL; /* No way to do an 'init' here, so just test+set with relaxed ordering and * ignore the test. */ ATOMIC_FLAG_TEST_AND_SET(&Source->PropsClean, almemory_order_relaxed); Source->VoiceIdx = -1; } static void DeinitSource(ALsource *source, ALsizei num_sends) { ALbufferlistitem *BufferList; ALsizei i; BufferList = source->queue; while(BufferList != NULL) { ALbufferlistitem *next = ATOMIC_LOAD(&BufferList->next, almemory_order_relaxed); for(i = 0;i < BufferList->num_buffers;i++) { if(BufferList->buffers[i] != NULL) DecrementRef(&BufferList->buffers[i]->ref); } al_free(BufferList); BufferList = next; } source->queue = NULL; if(source->Send) { for(i = 0;i < num_sends;i++) { if(source->Send[i].Slot) DecrementRef(&source->Send[i].Slot->ref); source->Send[i].Slot = NULL; } al_free(source->Send); source->Send = NULL; } } static void UpdateSourceProps(ALsource *source, ALvoice *voice, ALsizei num_sends, ALCcontext *context) { struct ALvoiceProps *props; ALsizei i; /* Get an unused property container, or allocate a new one as needed. */ props = ATOMIC_LOAD(&context->FreeVoiceProps, almemory_order_acquire); if(!props) props = al_calloc(16, FAM_SIZE(struct ALvoiceProps, Send, num_sends)); else { struct ALvoiceProps *next; do { next = ATOMIC_LOAD(&props->next, almemory_order_relaxed); } while(ATOMIC_COMPARE_EXCHANGE_PTR_WEAK(&context->FreeVoiceProps, &props, next, almemory_order_acq_rel, almemory_order_acquire) == 0); } /* Copy in current property values. */ props->Pitch = source->Pitch; props->Gain = source->Gain; props->OuterGain = source->OuterGain; props->MinGain = source->MinGain; props->MaxGain = source->MaxGain; props->InnerAngle = source->InnerAngle; props->OuterAngle = source->OuterAngle; props->RefDistance = source->RefDistance; props->MaxDistance = source->MaxDistance; props->RolloffFactor = source->RolloffFactor; for(i = 0;i < 3;i++) props->Position[i] = source->Position[i]; for(i = 0;i < 3;i++) props->Velocity[i] = source->Velocity[i]; for(i = 0;i < 3;i++) props->Direction[i] = source->Direction[i]; for(i = 0;i < 2;i++) { ALsizei j; for(j = 0;j < 3;j++) props->Orientation[i][j] = source->Orientation[i][j]; } props->HeadRelative = source->HeadRelative; props->DistanceModel = source->DistanceModel; props->Resampler = source->Resampler; props->DirectChannels = source->DirectChannels; props->SpatializeMode = source->Spatialize; props->DryGainHFAuto = source->DryGainHFAuto; props->WetGainAuto = source->WetGainAuto; props->WetGainHFAuto = source->WetGainHFAuto; props->OuterGainHF = source->OuterGainHF; props->AirAbsorptionFactor = source->AirAbsorptionFactor; props->RoomRolloffFactor = source->RoomRolloffFactor; props->DopplerFactor = source->DopplerFactor; props->StereoPan[0] = source->StereoPan[0]; props->StereoPan[1] = source->StereoPan[1]; props->Radius = source->Radius; props->Direct.Gain = source->Direct.Gain; props->Direct.GainHF = source->Direct.GainHF; props->Direct.HFReference = source->Direct.HFReference; props->Direct.GainLF = source->Direct.GainLF; props->Direct.LFReference = source->Direct.LFReference; for(i = 0;i < num_sends;i++) { props->Send[i].Slot = source->Send[i].Slot; props->Send[i].Gain = source->Send[i].Gain; props->Send[i].GainHF = source->Send[i].GainHF; props->Send[i].HFReference = source->Send[i].HFReference; props->Send[i].GainLF = source->Send[i].GainLF; props->Send[i].LFReference = source->Send[i].LFReference; } /* Set the new container for updating internal parameters. */ props = ATOMIC_EXCHANGE_PTR(&voice->Update, props, almemory_order_acq_rel); if(props) { /* If there was an unused update container, put it back in the * freelist. */ ATOMIC_REPLACE_HEAD(struct ALvoiceProps*, &context->FreeVoiceProps, props); } } void UpdateAllSourceProps(ALCcontext *context) { ALsizei num_sends = context->Device->NumAuxSends; ALsizei pos; for(pos = 0;pos < context->VoiceCount;pos++) { ALvoice *voice = context->Voices[pos]; ALsource *source = ATOMIC_LOAD(&voice->Source, almemory_order_acquire); if(source && !ATOMIC_FLAG_TEST_AND_SET(&source->PropsClean, almemory_order_acq_rel)) UpdateSourceProps(source, voice, num_sends, context); } } /* GetSourceSampleOffset * * Gets the current read offset for the given Source, in 32.32 fixed-point * samples. The offset is relative to the start of the queue (not the start of * the current buffer). */ static ALint64 GetSourceSampleOffset(ALsource *Source, ALCcontext *context, ALuint64 *clocktime) { ALCdevice *device = context->Device; const ALbufferlistitem *Current; ALuint64 readPos; ALuint refcount; ALvoice *voice; do { Current = NULL; readPos = 0; while(((refcount=ATOMIC_LOAD(&device->MixCount, almemory_order_acquire))&1)) althrd_yield(); *clocktime = GetDeviceClockTime(device); voice = GetSourceVoice(Source, context); if(voice) { Current = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed); readPos = (ALuint64)ATOMIC_LOAD(&voice->position, almemory_order_relaxed) << 32; readPos |= (ALuint64)ATOMIC_LOAD(&voice->position_fraction, almemory_order_relaxed) << (32-FRACTIONBITS); } ATOMIC_THREAD_FENCE(almemory_order_acquire); } while(refcount != ATOMIC_LOAD(&device->MixCount, almemory_order_relaxed)); if(voice) { const ALbufferlistitem *BufferList = Source->queue; while(BufferList && BufferList != Current) { readPos += (ALuint64)BufferList->max_samples << 32; BufferList = ATOMIC_LOAD(&CONST_CAST(ALbufferlistitem*,BufferList)->next, almemory_order_relaxed); } readPos = minu64(readPos, U64(0x7fffffffffffffff)); } return (ALint64)readPos; } /* GetSourceSecOffset * * Gets the current read offset for the given Source, in seconds. The offset is * relative to the start of the queue (not the start of the current buffer). */ static ALdouble GetSourceSecOffset(ALsource *Source, ALCcontext *context, ALuint64 *clocktime) { ALCdevice *device = context->Device; const ALbufferlistitem *Current; ALuint64 readPos; ALuint refcount; ALdouble offset; ALvoice *voice; do { Current = NULL; readPos = 0; while(((refcount=ATOMIC_LOAD(&device->MixCount, almemory_order_acquire))&1)) althrd_yield(); *clocktime = GetDeviceClockTime(device); voice = GetSourceVoice(Source, context); if(voice) { Current = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed); readPos = (ALuint64)ATOMIC_LOAD(&voice->position, almemory_order_relaxed) << FRACTIONBITS; readPos |= ATOMIC_LOAD(&voice->position_fraction, almemory_order_relaxed); } ATOMIC_THREAD_FENCE(almemory_order_acquire); } while(refcount != ATOMIC_LOAD(&device->MixCount, almemory_order_relaxed)); offset = 0.0; if(voice) { const ALbufferlistitem *BufferList = Source->queue; const ALbuffer *BufferFmt = NULL; while(BufferList && BufferList != Current) { ALsizei i = 0; while(!BufferFmt && i < BufferList->num_buffers) BufferFmt = BufferList->buffers[i++]; readPos += (ALuint64)BufferList->max_samples << FRACTIONBITS; BufferList = ATOMIC_LOAD(&CONST_CAST(ALbufferlistitem*,BufferList)->next, almemory_order_relaxed); } while(BufferList && !BufferFmt) { ALsizei i = 0; while(!BufferFmt && i < BufferList->num_buffers) BufferFmt = BufferList->buffers[i++]; BufferList = ATOMIC_LOAD(&CONST_CAST(ALbufferlistitem*,BufferList)->next, almemory_order_relaxed); } assert(BufferFmt != NULL); offset = (ALdouble)readPos / (ALdouble)FRACTIONONE / (ALdouble)BufferFmt->Frequency; } return offset; } /* GetSourceOffset * * Gets the current read offset for the given Source, in the appropriate format * (Bytes, Samples or Seconds). The offset is relative to the start of the * queue (not the start of the current buffer). */ static ALdouble GetSourceOffset(ALsource *Source, ALenum name, ALCcontext *context) { ALCdevice *device = context->Device; const ALbufferlistitem *Current; ALuint readPos; ALsizei readPosFrac; ALuint refcount; ALdouble offset; ALvoice *voice; do { Current = NULL; readPos = readPosFrac = 0; while(((refcount=ATOMIC_LOAD(&device->MixCount, almemory_order_acquire))&1)) althrd_yield(); voice = GetSourceVoice(Source, context); if(voice) { Current = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed); readPos = ATOMIC_LOAD(&voice->position, almemory_order_relaxed); readPosFrac = ATOMIC_LOAD(&voice->position_fraction, almemory_order_relaxed); } ATOMIC_THREAD_FENCE(almemory_order_acquire); } while(refcount != ATOMIC_LOAD(&device->MixCount, almemory_order_relaxed)); offset = 0.0; if(voice) { const ALbufferlistitem *BufferList = Source->queue; const ALbuffer *BufferFmt = NULL; ALboolean readFin = AL_FALSE; ALuint totalBufferLen = 0; while(BufferList != NULL) { ALsizei i = 0; while(!BufferFmt && i < BufferList->num_buffers) BufferFmt = BufferList->buffers[i++]; readFin |= (BufferList == Current); totalBufferLen += BufferList->max_samples; if(!readFin) readPos += BufferList->max_samples; BufferList = ATOMIC_LOAD(&CONST_CAST(ALbufferlistitem*,BufferList)->next, almemory_order_relaxed); } assert(BufferFmt != NULL); if(Source->Looping) readPos %= totalBufferLen; else { /* Wrap back to 0 */ if(readPos >= totalBufferLen) readPos = readPosFrac = 0; } offset = 0.0; switch(name) { case AL_SEC_OFFSET: offset = (readPos + (ALdouble)readPosFrac/FRACTIONONE) / BufferFmt->Frequency; break; case AL_SAMPLE_OFFSET: offset = readPos + (ALdouble)readPosFrac/FRACTIONONE; break; case AL_BYTE_OFFSET: if(BufferFmt->OriginalType == UserFmtIMA4) { ALsizei align = (BufferFmt->OriginalAlign-1)/2 + 4; ALuint BlockSize = align * ChannelsFromFmt(BufferFmt->FmtChannels); ALuint FrameBlockSize = BufferFmt->OriginalAlign; /* Round down to nearest ADPCM block */ offset = (ALdouble)(readPos / FrameBlockSize * BlockSize); } else if(BufferFmt->OriginalType == UserFmtMSADPCM) { ALsizei align = (BufferFmt->OriginalAlign-2)/2 + 7; ALuint BlockSize = align * ChannelsFromFmt(BufferFmt->FmtChannels); ALuint FrameBlockSize = BufferFmt->OriginalAlign; /* Round down to nearest ADPCM block */ offset = (ALdouble)(readPos / FrameBlockSize * BlockSize); } else { ALuint FrameSize = FrameSizeFromFmt(BufferFmt->FmtChannels, BufferFmt->FmtType); offset = (ALdouble)(readPos * FrameSize); } break; } } return offset; } /* ApplyOffset * * Apply the stored playback offset to the Source. This function will update * the number of buffers "played" given the stored offset. */ static ALboolean ApplyOffset(ALsource *Source, ALvoice *voice) { ALbufferlistitem *BufferList; ALuint totalBufferLen; ALuint offset = 0; ALsizei frac = 0; /* Get sample frame offset */ if(!GetSampleOffset(Source, &offset, &frac)) return AL_FALSE; totalBufferLen = 0; BufferList = Source->queue; while(BufferList && totalBufferLen <= offset) { if((ALuint)BufferList->max_samples > offset-totalBufferLen) { /* Offset is in this buffer */ ATOMIC_STORE(&voice->position, offset - totalBufferLen, almemory_order_relaxed); ATOMIC_STORE(&voice->position_fraction, frac, almemory_order_relaxed); ATOMIC_STORE(&voice->current_buffer, BufferList, almemory_order_release); return AL_TRUE; } totalBufferLen += BufferList->max_samples; BufferList = ATOMIC_LOAD(&BufferList->next, almemory_order_relaxed); } /* Offset is out of range of the queue */ return AL_FALSE; } /* GetSampleOffset * * Retrieves the sample offset into the Source's queue (from the Sample, Byte * or Second offset supplied by the application). This takes into account the * fact that the buffer format may have been modifed since. */ static ALboolean GetSampleOffset(ALsource *Source, ALuint *offset, ALsizei *frac) { const ALbuffer *BufferFmt = NULL; const ALbufferlistitem *BufferList; ALdouble dbloff, dblfrac; /* Find the first valid Buffer in the Queue */ BufferList = Source->queue; while(BufferList) { ALsizei i; for(i = 0;i < BufferList->num_buffers && !BufferFmt;i++) BufferFmt = BufferList->buffers[i]; if(BufferFmt) break; BufferList = ATOMIC_LOAD(&CONST_CAST(ALbufferlistitem*,BufferList)->next, almemory_order_relaxed); } if(!BufferFmt) { Source->OffsetType = AL_NONE; Source->Offset = 0.0; return AL_FALSE; } switch(Source->OffsetType) { case AL_BYTE_OFFSET: /* Determine the ByteOffset (and ensure it is block aligned) */ *offset = (ALuint)Source->Offset; if(BufferFmt->OriginalType == UserFmtIMA4) { ALsizei align = (BufferFmt->OriginalAlign-1)/2 + 4; *offset /= align * ChannelsFromFmt(BufferFmt->FmtChannels); *offset *= BufferFmt->OriginalAlign; } else if(BufferFmt->OriginalType == UserFmtMSADPCM) { ALsizei align = (BufferFmt->OriginalAlign-2)/2 + 7; *offset /= align * ChannelsFromFmt(BufferFmt->FmtChannels); *offset *= BufferFmt->OriginalAlign; } else *offset /= FrameSizeFromFmt(BufferFmt->FmtChannels, BufferFmt->FmtType); *frac = 0; break; case AL_SAMPLE_OFFSET: dblfrac = modf(Source->Offset, &dbloff); *offset = (ALuint)mind(dbloff, UINT_MAX); *frac = (ALsizei)mind(dblfrac*FRACTIONONE, FRACTIONONE-1.0); break; case AL_SEC_OFFSET: dblfrac = modf(Source->Offset*BufferFmt->Frequency, &dbloff); *offset = (ALuint)mind(dbloff, UINT_MAX); *frac = (ALsizei)mind(dblfrac*FRACTIONONE, FRACTIONONE-1.0); break; } Source->OffsetType = AL_NONE; Source->Offset = 0.0; return AL_TRUE; } static ALsource *AllocSource(ALCcontext *context) { ALCdevice *device = context->Device; SourceSubList *sublist, *subend; ALsource *source = NULL; ALsizei lidx = 0; ALsizei slidx; almtx_lock(&context->SourceLock); if(context->NumSources >= device->SourcesMax) { almtx_unlock(&context->SourceLock); alSetError(context, AL_OUT_OF_MEMORY, "Exceeding %u source limit", device->SourcesMax); return NULL; } sublist = VECTOR_BEGIN(context->SourceList); subend = VECTOR_END(context->SourceList); for(;sublist != subend;++sublist) { if(sublist->FreeMask) { slidx = CTZ64(sublist->FreeMask); source = sublist->Sources + slidx; break; } ++lidx; } if(UNLIKELY(!source)) { const SourceSubList empty_sublist = { 0, NULL }; /* Don't allocate so many list entries that the 32-bit ID could * overflow... */ if(UNLIKELY(VECTOR_SIZE(context->SourceList) >= 1<<25)) { almtx_unlock(&device->BufferLock); alSetError(context, AL_OUT_OF_MEMORY, "Too many sources allocated"); return NULL; } lidx = (ALsizei)VECTOR_SIZE(context->SourceList); VECTOR_PUSH_BACK(context->SourceList, empty_sublist); sublist = &VECTOR_BACK(context->SourceList); sublist->FreeMask = ~U64(0); sublist->Sources = al_calloc(16, sizeof(ALsource)*64); if(UNLIKELY(!sublist->Sources)) { VECTOR_POP_BACK(context->SourceList); almtx_unlock(&context->SourceLock); alSetError(context, AL_OUT_OF_MEMORY, "Failed to allocate source batch"); return NULL; } slidx = 0; source = sublist->Sources + slidx; } memset(source, 0, sizeof(*source)); InitSourceParams(source, device->NumAuxSends); /* Add 1 to avoid source ID 0. */ source->id = ((lidx<<6) | slidx) + 1; context->NumSources++; sublist->FreeMask &= ~(U64(1)<SourceLock); return source; } static void FreeSource(ALCcontext *context, ALsource *source) { ALCdevice *device = context->Device; ALuint id = source->id - 1; ALsizei lidx = id >> 6; ALsizei slidx = id & 0x3f; ALvoice *voice; ALCdevice_Lock(device); if((voice=GetSourceVoice(source, context)) != NULL) { ATOMIC_STORE(&voice->Source, NULL, almemory_order_relaxed); ATOMIC_STORE(&voice->Playing, false, almemory_order_release); } ALCdevice_Unlock(device); DeinitSource(source, device->NumAuxSends); memset(source, 0, sizeof(*source)); VECTOR_ELEM(context->SourceList, lidx).FreeMask |= U64(1) << slidx; context->NumSources--; } /* ReleaseALSources * * Destroys all sources in the source map. */ ALvoid ReleaseALSources(ALCcontext *context) { ALCdevice *device = context->Device; SourceSubList *sublist = VECTOR_BEGIN(context->SourceList); SourceSubList *subend = VECTOR_END(context->SourceList); size_t leftover = 0; for(;sublist != subend;++sublist) { ALuint64 usemask = ~sublist->FreeMask; while(usemask) { ALsizei idx = CTZ64(usemask); ALsource *source = sublist->Sources + idx; DeinitSource(source, device->NumAuxSends); memset(source, 0, sizeof(*source)); ++leftover; usemask &= ~(U64(1) << idx); } sublist->FreeMask = ~usemask; } if(leftover > 0) WARN("(%p) Deleted "SZFMT" Source%s\n", device, leftover, (leftover==1)?"":"s"); } openal-soft-openal-soft-1.19.1/OpenAL32/alState.c000066400000000000000000000546611335774445300213360ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2000 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include "version.h" #include #include "alMain.h" #include "AL/alc.h" #include "AL/al.h" #include "AL/alext.h" #include "alError.h" #include "alListener.h" #include "alSource.h" #include "alAuxEffectSlot.h" #include "backends/base.h" static const ALchar alVendor[] = "OpenAL Community"; static const ALchar alVersion[] = "1.1 ALSOFT "ALSOFT_VERSION; static const ALchar alRenderer[] = "OpenAL Soft"; // Error Messages static const ALchar alNoError[] = "No Error"; static const ALchar alErrInvalidName[] = "Invalid Name"; static const ALchar alErrInvalidEnum[] = "Invalid Enum"; static const ALchar alErrInvalidValue[] = "Invalid Value"; static const ALchar alErrInvalidOp[] = "Invalid Operation"; static const ALchar alErrOutOfMemory[] = "Out of Memory"; /* Resampler strings */ static const ALchar alPointResampler[] = "Nearest"; static const ALchar alLinearResampler[] = "Linear"; static const ALchar alCubicResampler[] = "Cubic"; static const ALchar alBSinc12Resampler[] = "11th order Sinc"; static const ALchar alBSinc24Resampler[] = "23rd order Sinc"; /* WARNING: Non-standard export! Not part of any extension, or exposed in the * alcFunctions list. */ AL_API const ALchar* AL_APIENTRY alsoft_get_version(void) { const char *spoof = getenv("ALSOFT_SPOOF_VERSION"); if(spoof && spoof[0] != '\0') return spoof; return ALSOFT_VERSION; } #define DO_UPDATEPROPS() do { \ if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire)) \ UpdateContextProps(context); \ else \ ATOMIC_FLAG_CLEAR(&context->PropsClean, almemory_order_release); \ } while(0) AL_API ALvoid AL_APIENTRY alEnable(ALenum capability) { ALCcontext *context; context = GetContextRef(); if(!context) return; almtx_lock(&context->PropLock); switch(capability) { case AL_SOURCE_DISTANCE_MODEL: context->SourceDistanceModel = AL_TRUE; DO_UPDATEPROPS(); break; default: alSetError(context, AL_INVALID_VALUE, "Invalid enable property 0x%04x", capability); } almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alDisable(ALenum capability) { ALCcontext *context; context = GetContextRef(); if(!context) return; almtx_lock(&context->PropLock); switch(capability) { case AL_SOURCE_DISTANCE_MODEL: context->SourceDistanceModel = AL_FALSE; DO_UPDATEPROPS(); break; default: alSetError(context, AL_INVALID_VALUE, "Invalid disable property 0x%04x", capability); } almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); } AL_API ALboolean AL_APIENTRY alIsEnabled(ALenum capability) { ALCcontext *context; ALboolean value=AL_FALSE; context = GetContextRef(); if(!context) return AL_FALSE; almtx_lock(&context->PropLock); switch(capability) { case AL_SOURCE_DISTANCE_MODEL: value = context->SourceDistanceModel; break; default: alSetError(context, AL_INVALID_VALUE, "Invalid is enabled property 0x%04x", capability); } almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); return value; } AL_API ALboolean AL_APIENTRY alGetBoolean(ALenum pname) { ALCcontext *context; ALboolean value=AL_FALSE; context = GetContextRef(); if(!context) return AL_FALSE; almtx_lock(&context->PropLock); switch(pname) { case AL_DOPPLER_FACTOR: if(context->DopplerFactor != 0.0f) value = AL_TRUE; break; case AL_DOPPLER_VELOCITY: if(context->DopplerVelocity != 0.0f) value = AL_TRUE; break; case AL_DISTANCE_MODEL: if(context->DistanceModel == AL_INVERSE_DISTANCE_CLAMPED) value = AL_TRUE; break; case AL_SPEED_OF_SOUND: if(context->SpeedOfSound != 0.0f) value = AL_TRUE; break; case AL_DEFERRED_UPDATES_SOFT: if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire)) value = AL_TRUE; break; case AL_GAIN_LIMIT_SOFT: if(GAIN_MIX_MAX/context->GainBoost != 0.0f) value = AL_TRUE; break; case AL_NUM_RESAMPLERS_SOFT: /* Always non-0. */ value = AL_TRUE; break; case AL_DEFAULT_RESAMPLER_SOFT: value = ResamplerDefault ? AL_TRUE : AL_FALSE; break; default: alSetError(context, AL_INVALID_VALUE, "Invalid boolean property 0x%04x", pname); } almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); return value; } AL_API ALdouble AL_APIENTRY alGetDouble(ALenum pname) { ALCcontext *context; ALdouble value = 0.0; context = GetContextRef(); if(!context) return 0.0; almtx_lock(&context->PropLock); switch(pname) { case AL_DOPPLER_FACTOR: value = (ALdouble)context->DopplerFactor; break; case AL_DOPPLER_VELOCITY: value = (ALdouble)context->DopplerVelocity; break; case AL_DISTANCE_MODEL: value = (ALdouble)context->DistanceModel; break; case AL_SPEED_OF_SOUND: value = (ALdouble)context->SpeedOfSound; break; case AL_DEFERRED_UPDATES_SOFT: if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire)) value = (ALdouble)AL_TRUE; break; case AL_GAIN_LIMIT_SOFT: value = (ALdouble)GAIN_MIX_MAX/context->GainBoost; break; case AL_NUM_RESAMPLERS_SOFT: value = (ALdouble)(ResamplerMax + 1); break; case AL_DEFAULT_RESAMPLER_SOFT: value = (ALdouble)ResamplerDefault; break; default: alSetError(context, AL_INVALID_VALUE, "Invalid double property 0x%04x", pname); } almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); return value; } AL_API ALfloat AL_APIENTRY alGetFloat(ALenum pname) { ALCcontext *context; ALfloat value = 0.0f; context = GetContextRef(); if(!context) return 0.0f; almtx_lock(&context->PropLock); switch(pname) { case AL_DOPPLER_FACTOR: value = context->DopplerFactor; break; case AL_DOPPLER_VELOCITY: value = context->DopplerVelocity; break; case AL_DISTANCE_MODEL: value = (ALfloat)context->DistanceModel; break; case AL_SPEED_OF_SOUND: value = context->SpeedOfSound; break; case AL_DEFERRED_UPDATES_SOFT: if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire)) value = (ALfloat)AL_TRUE; break; case AL_GAIN_LIMIT_SOFT: value = GAIN_MIX_MAX/context->GainBoost; break; case AL_NUM_RESAMPLERS_SOFT: value = (ALfloat)(ResamplerMax + 1); break; case AL_DEFAULT_RESAMPLER_SOFT: value = (ALfloat)ResamplerDefault; break; default: alSetError(context, AL_INVALID_VALUE, "Invalid float property 0x%04x", pname); } almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); return value; } AL_API ALint AL_APIENTRY alGetInteger(ALenum pname) { ALCcontext *context; ALint value = 0; context = GetContextRef(); if(!context) return 0; almtx_lock(&context->PropLock); switch(pname) { case AL_DOPPLER_FACTOR: value = (ALint)context->DopplerFactor; break; case AL_DOPPLER_VELOCITY: value = (ALint)context->DopplerVelocity; break; case AL_DISTANCE_MODEL: value = (ALint)context->DistanceModel; break; case AL_SPEED_OF_SOUND: value = (ALint)context->SpeedOfSound; break; case AL_DEFERRED_UPDATES_SOFT: if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire)) value = (ALint)AL_TRUE; break; case AL_GAIN_LIMIT_SOFT: value = (ALint)(GAIN_MIX_MAX/context->GainBoost); break; case AL_NUM_RESAMPLERS_SOFT: value = ResamplerMax + 1; break; case AL_DEFAULT_RESAMPLER_SOFT: value = ResamplerDefault; break; default: alSetError(context, AL_INVALID_VALUE, "Invalid integer property 0x%04x", pname); } almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); return value; } AL_API ALint64SOFT AL_APIENTRY alGetInteger64SOFT(ALenum pname) { ALCcontext *context; ALint64SOFT value = 0; context = GetContextRef(); if(!context) return 0; almtx_lock(&context->PropLock); switch(pname) { case AL_DOPPLER_FACTOR: value = (ALint64SOFT)context->DopplerFactor; break; case AL_DOPPLER_VELOCITY: value = (ALint64SOFT)context->DopplerVelocity; break; case AL_DISTANCE_MODEL: value = (ALint64SOFT)context->DistanceModel; break; case AL_SPEED_OF_SOUND: value = (ALint64SOFT)context->SpeedOfSound; break; case AL_DEFERRED_UPDATES_SOFT: if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire)) value = (ALint64SOFT)AL_TRUE; break; case AL_GAIN_LIMIT_SOFT: value = (ALint64SOFT)(GAIN_MIX_MAX/context->GainBoost); break; case AL_NUM_RESAMPLERS_SOFT: value = (ALint64SOFT)(ResamplerMax + 1); break; case AL_DEFAULT_RESAMPLER_SOFT: value = (ALint64SOFT)ResamplerDefault; break; default: alSetError(context, AL_INVALID_VALUE, "Invalid integer64 property 0x%04x", pname); } almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); return value; } AL_API void* AL_APIENTRY alGetPointerSOFT(ALenum pname) { ALCcontext *context; void *value = NULL; context = GetContextRef(); if(!context) return NULL; almtx_lock(&context->PropLock); switch(pname) { case AL_EVENT_CALLBACK_FUNCTION_SOFT: value = context->EventCb; break; case AL_EVENT_CALLBACK_USER_PARAM_SOFT: value = context->EventParam; break; default: alSetError(context, AL_INVALID_VALUE, "Invalid pointer property 0x%04x", pname); } almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); return value; } AL_API ALvoid AL_APIENTRY alGetBooleanv(ALenum pname, ALboolean *values) { ALCcontext *context; if(values) { switch(pname) { case AL_DOPPLER_FACTOR: case AL_DOPPLER_VELOCITY: case AL_DISTANCE_MODEL: case AL_SPEED_OF_SOUND: case AL_DEFERRED_UPDATES_SOFT: case AL_GAIN_LIMIT_SOFT: case AL_NUM_RESAMPLERS_SOFT: case AL_DEFAULT_RESAMPLER_SOFT: values[0] = alGetBoolean(pname); return; } } context = GetContextRef(); if(!context) return; if(!values) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); switch(pname) { default: alSetError(context, AL_INVALID_VALUE, "Invalid boolean-vector property 0x%04x", pname); } ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alGetDoublev(ALenum pname, ALdouble *values) { ALCcontext *context; if(values) { switch(pname) { case AL_DOPPLER_FACTOR: case AL_DOPPLER_VELOCITY: case AL_DISTANCE_MODEL: case AL_SPEED_OF_SOUND: case AL_DEFERRED_UPDATES_SOFT: case AL_GAIN_LIMIT_SOFT: case AL_NUM_RESAMPLERS_SOFT: case AL_DEFAULT_RESAMPLER_SOFT: values[0] = alGetDouble(pname); return; } } context = GetContextRef(); if(!context) return; if(!values) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); switch(pname) { default: alSetError(context, AL_INVALID_VALUE, "Invalid double-vector property 0x%04x", pname); } ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alGetFloatv(ALenum pname, ALfloat *values) { ALCcontext *context; if(values) { switch(pname) { case AL_DOPPLER_FACTOR: case AL_DOPPLER_VELOCITY: case AL_DISTANCE_MODEL: case AL_SPEED_OF_SOUND: case AL_DEFERRED_UPDATES_SOFT: case AL_GAIN_LIMIT_SOFT: case AL_NUM_RESAMPLERS_SOFT: case AL_DEFAULT_RESAMPLER_SOFT: values[0] = alGetFloat(pname); return; } } context = GetContextRef(); if(!context) return; if(!values) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); switch(pname) { default: alSetError(context, AL_INVALID_VALUE, "Invalid float-vector property 0x%04x", pname); } ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alGetIntegerv(ALenum pname, ALint *values) { ALCcontext *context; if(values) { switch(pname) { case AL_DOPPLER_FACTOR: case AL_DOPPLER_VELOCITY: case AL_DISTANCE_MODEL: case AL_SPEED_OF_SOUND: case AL_DEFERRED_UPDATES_SOFT: case AL_GAIN_LIMIT_SOFT: case AL_NUM_RESAMPLERS_SOFT: case AL_DEFAULT_RESAMPLER_SOFT: values[0] = alGetInteger(pname); return; } } context = GetContextRef(); if(!context) return; if(!values) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); switch(pname) { default: alSetError(context, AL_INVALID_VALUE, "Invalid integer-vector property 0x%04x", pname); } ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alGetInteger64vSOFT(ALenum pname, ALint64SOFT *values) { ALCcontext *context; if(values) { switch(pname) { case AL_DOPPLER_FACTOR: case AL_DOPPLER_VELOCITY: case AL_DISTANCE_MODEL: case AL_SPEED_OF_SOUND: case AL_DEFERRED_UPDATES_SOFT: case AL_GAIN_LIMIT_SOFT: case AL_NUM_RESAMPLERS_SOFT: case AL_DEFAULT_RESAMPLER_SOFT: values[0] = alGetInteger64SOFT(pname); return; } } context = GetContextRef(); if(!context) return; if(!values) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); switch(pname) { default: alSetError(context, AL_INVALID_VALUE, "Invalid integer64-vector property 0x%04x", pname); } ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alGetPointervSOFT(ALenum pname, void **values) { ALCcontext *context; if(values) { switch(pname) { case AL_EVENT_CALLBACK_FUNCTION_SOFT: case AL_EVENT_CALLBACK_USER_PARAM_SOFT: values[0] = alGetPointerSOFT(pname); return; } } context = GetContextRef(); if(!context) return; if(!values) alSetError(context, AL_INVALID_VALUE, "NULL pointer"); switch(pname) { default: alSetError(context, AL_INVALID_VALUE, "Invalid pointer-vector property 0x%04x", pname); } ALCcontext_DecRef(context); } AL_API const ALchar* AL_APIENTRY alGetString(ALenum pname) { const ALchar *value = NULL; ALCcontext *context; context = GetContextRef(); if(!context) return NULL; switch(pname) { case AL_VENDOR: value = alVendor; break; case AL_VERSION: value = alVersion; break; case AL_RENDERER: value = alRenderer; break; case AL_EXTENSIONS: value = context->ExtensionList; break; case AL_NO_ERROR: value = alNoError; break; case AL_INVALID_NAME: value = alErrInvalidName; break; case AL_INVALID_ENUM: value = alErrInvalidEnum; break; case AL_INVALID_VALUE: value = alErrInvalidValue; break; case AL_INVALID_OPERATION: value = alErrInvalidOp; break; case AL_OUT_OF_MEMORY: value = alErrOutOfMemory; break; default: alSetError(context, AL_INVALID_VALUE, "Invalid string property 0x%04x", pname); } ALCcontext_DecRef(context); return value; } AL_API ALvoid AL_APIENTRY alDopplerFactor(ALfloat value) { ALCcontext *context; context = GetContextRef(); if(!context) return; if(!(value >= 0.0f && isfinite(value))) alSetError(context, AL_INVALID_VALUE, "Doppler factor %f out of range", value); else { almtx_lock(&context->PropLock); context->DopplerFactor = value; DO_UPDATEPROPS(); almtx_unlock(&context->PropLock); } ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alDopplerVelocity(ALfloat value) { ALCcontext *context; context = GetContextRef(); if(!context) return; if((ATOMIC_LOAD(&context->EnabledEvts, almemory_order_relaxed)&EventType_Deprecated)) { static const ALCchar msg[] = "alDopplerVelocity is deprecated in AL1.1, use alSpeedOfSound"; const ALsizei msglen = (ALsizei)strlen(msg); ALbitfieldSOFT enabledevts; almtx_lock(&context->EventCbLock); enabledevts = ATOMIC_LOAD(&context->EnabledEvts, almemory_order_relaxed); if((enabledevts&EventType_Deprecated) && context->EventCb) (*context->EventCb)(AL_EVENT_TYPE_DEPRECATED_SOFT, 0, 0, msglen, msg, context->EventParam); almtx_unlock(&context->EventCbLock); } if(!(value >= 0.0f && isfinite(value))) alSetError(context, AL_INVALID_VALUE, "Doppler velocity %f out of range", value); else { almtx_lock(&context->PropLock); context->DopplerVelocity = value; DO_UPDATEPROPS(); almtx_unlock(&context->PropLock); } ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alSpeedOfSound(ALfloat value) { ALCcontext *context; context = GetContextRef(); if(!context) return; if(!(value > 0.0f && isfinite(value))) alSetError(context, AL_INVALID_VALUE, "Speed of sound %f out of range", value); else { almtx_lock(&context->PropLock); context->SpeedOfSound = value; DO_UPDATEPROPS(); almtx_unlock(&context->PropLock); } ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alDistanceModel(ALenum value) { ALCcontext *context; context = GetContextRef(); if(!context) return; if(!(value == AL_INVERSE_DISTANCE || value == AL_INVERSE_DISTANCE_CLAMPED || value == AL_LINEAR_DISTANCE || value == AL_LINEAR_DISTANCE_CLAMPED || value == AL_EXPONENT_DISTANCE || value == AL_EXPONENT_DISTANCE_CLAMPED || value == AL_NONE)) alSetError(context, AL_INVALID_VALUE, "Distance model 0x%04x out of range", value); else { almtx_lock(&context->PropLock); context->DistanceModel = value; if(!context->SourceDistanceModel) DO_UPDATEPROPS(); almtx_unlock(&context->PropLock); } ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alDeferUpdatesSOFT(void) { ALCcontext *context; context = GetContextRef(); if(!context) return; ALCcontext_DeferUpdates(context); ALCcontext_DecRef(context); } AL_API ALvoid AL_APIENTRY alProcessUpdatesSOFT(void) { ALCcontext *context; context = GetContextRef(); if(!context) return; ALCcontext_ProcessUpdates(context); ALCcontext_DecRef(context); } AL_API const ALchar* AL_APIENTRY alGetStringiSOFT(ALenum pname, ALsizei index) { const char *ResamplerNames[] = { alPointResampler, alLinearResampler, alCubicResampler, alBSinc12Resampler, alBSinc24Resampler, }; const ALchar *value = NULL; ALCcontext *context; static_assert(COUNTOF(ResamplerNames) == ResamplerMax+1, "Incorrect ResamplerNames list"); context = GetContextRef(); if(!context) return NULL; switch(pname) { case AL_RESAMPLER_NAME_SOFT: if(index < 0 || (size_t)index >= COUNTOF(ResamplerNames)) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Resampler name index %d out of range", index); value = ResamplerNames[index]; break; default: alSetError(context, AL_INVALID_VALUE, "Invalid string indexed property"); } done: ALCcontext_DecRef(context); return value; } void UpdateContextProps(ALCcontext *context) { struct ALcontextProps *props; /* Get an unused proprty container, or allocate a new one as needed. */ props = ATOMIC_LOAD(&context->FreeContextProps, almemory_order_acquire); if(!props) props = al_calloc(16, sizeof(*props)); else { struct ALcontextProps *next; do { next = ATOMIC_LOAD(&props->next, almemory_order_relaxed); } while(ATOMIC_COMPARE_EXCHANGE_PTR_WEAK(&context->FreeContextProps, &props, next, almemory_order_seq_cst, almemory_order_acquire) == 0); } /* Copy in current property values. */ props->MetersPerUnit = context->MetersPerUnit; props->DopplerFactor = context->DopplerFactor; props->DopplerVelocity = context->DopplerVelocity; props->SpeedOfSound = context->SpeedOfSound; props->SourceDistanceModel = context->SourceDistanceModel; props->DistanceModel = context->DistanceModel; /* Set the new container for updating internal parameters. */ props = ATOMIC_EXCHANGE_PTR(&context->Update, props, almemory_order_acq_rel); if(props) { /* If there was an unused update container, put it back in the * freelist. */ ATOMIC_REPLACE_HEAD(struct ALcontextProps*, &context->FreeContextProps, props); } } openal-soft-openal-soft-1.19.1/OpenAL32/event.c000066400000000000000000000076711335774445300210610ustar00rootroot00000000000000 #include "config.h" #include "AL/alc.h" #include "AL/al.h" #include "AL/alext.h" #include "alMain.h" #include "alError.h" #include "alAuxEffectSlot.h" #include "ringbuffer.h" int EventThread(void *arg) { ALCcontext *context = arg; bool quitnow = false; while(!quitnow) { ALbitfieldSOFT enabledevts; AsyncEvent evt; if(ll_ringbuffer_read(context->AsyncEvents, (char*)&evt, 1) == 0) { alsem_wait(&context->EventSem); continue; } almtx_lock(&context->EventCbLock); do { quitnow = evt.EnumType == EventType_KillThread; if(quitnow) break; if(evt.EnumType == EventType_ReleaseEffectState) { ALeffectState_DecRef(evt.u.EffectState); continue; } enabledevts = ATOMIC_LOAD(&context->EnabledEvts, almemory_order_acquire); if(context->EventCb && (enabledevts&evt.EnumType) == evt.EnumType) context->EventCb(evt.u.user.type, evt.u.user.id, evt.u.user.param, (ALsizei)strlen(evt.u.user.msg), evt.u.user.msg, context->EventParam ); } while(ll_ringbuffer_read(context->AsyncEvents, (char*)&evt, 1) != 0); almtx_unlock(&context->EventCbLock); } return 0; } AL_API void AL_APIENTRY alEventControlSOFT(ALsizei count, const ALenum *types, ALboolean enable) { ALCcontext *context; ALbitfieldSOFT enabledevts; ALbitfieldSOFT flags = 0; ALsizei i; context = GetContextRef(); if(!context) return; if(count < 0) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Controlling %d events", count); if(count == 0) goto done; if(!types) SETERR_GOTO(context, AL_INVALID_VALUE, done, "NULL pointer"); for(i = 0;i < count;i++) { if(types[i] == AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT) flags |= EventType_BufferCompleted; else if(types[i] == AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT) flags |= EventType_SourceStateChange; else if(types[i] == AL_EVENT_TYPE_ERROR_SOFT) flags |= EventType_Error; else if(types[i] == AL_EVENT_TYPE_PERFORMANCE_SOFT) flags |= EventType_Performance; else if(types[i] == AL_EVENT_TYPE_DEPRECATED_SOFT) flags |= EventType_Deprecated; else if(types[i] == AL_EVENT_TYPE_DISCONNECTED_SOFT) flags |= EventType_Disconnected; else SETERR_GOTO(context, AL_INVALID_ENUM, done, "Invalid event type 0x%04x", types[i]); } if(enable) { enabledevts = ATOMIC_LOAD(&context->EnabledEvts, almemory_order_relaxed); while(ATOMIC_COMPARE_EXCHANGE_WEAK(&context->EnabledEvts, &enabledevts, enabledevts|flags, almemory_order_acq_rel, almemory_order_acquire) == 0) { /* enabledevts is (re-)filled with the current value on failure, so * just try again. */ } } else { enabledevts = ATOMIC_LOAD(&context->EnabledEvts, almemory_order_relaxed); while(ATOMIC_COMPARE_EXCHANGE_WEAK(&context->EnabledEvts, &enabledevts, enabledevts&~flags, almemory_order_acq_rel, almemory_order_acquire) == 0) { } /* Wait to ensure the event handler sees the changed flags before * returning. */ almtx_lock(&context->EventCbLock); almtx_unlock(&context->EventCbLock); } done: ALCcontext_DecRef(context); } AL_API void AL_APIENTRY alEventCallbackSOFT(ALEVENTPROCSOFT callback, void *userParam) { ALCcontext *context; context = GetContextRef(); if(!context) return; almtx_lock(&context->PropLock); almtx_lock(&context->EventCbLock); context->EventCb = callback; context->EventParam = userParam; almtx_unlock(&context->EventCbLock); almtx_unlock(&context->PropLock); ALCcontext_DecRef(context); } openal-soft-openal-soft-1.19.1/OpenAL32/sample_cvt.c000066400000000000000000000227561335774445300220760ustar00rootroot00000000000000 #include "config.h" #include "sample_cvt.h" #include "AL/al.h" #include "alu.h" #include "alBuffer.h" /* IMA ADPCM Stepsize table */ static const int IMAStep_size[89] = { 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, 9493,10442, 11487,12635,13899,15289,16818,18500,20350,22358,24633,27086,29794, 32767 }; /* IMA4 ADPCM Codeword decode table */ static const int IMA4Codeword[16] = { 1, 3, 5, 7, 9, 11, 13, 15, -1,-3,-5,-7,-9,-11,-13,-15, }; /* IMA4 ADPCM Step index adjust decode table */ static const int IMA4Index_adjust[16] = { -1,-1,-1,-1, 2, 4, 6, 8, -1,-1,-1,-1, 2, 4, 6, 8 }; /* MSADPCM Adaption table */ static const int MSADPCMAdaption[16] = { 230, 230, 230, 230, 307, 409, 512, 614, 768, 614, 512, 409, 307, 230, 230, 230 }; /* MSADPCM Adaption Coefficient tables */ static const int MSADPCMAdaptionCoeff[7][2] = { { 256, 0 }, { 512, -256 }, { 0, 0 }, { 192, 64 }, { 240, 0 }, { 460, -208 }, { 392, -232 } }; /* A quick'n'dirty lookup table to decode a muLaw-encoded byte sample into a * signed 16-bit sample */ const ALshort muLawDecompressionTable[256] = { -32124,-31100,-30076,-29052,-28028,-27004,-25980,-24956, -23932,-22908,-21884,-20860,-19836,-18812,-17788,-16764, -15996,-15484,-14972,-14460,-13948,-13436,-12924,-12412, -11900,-11388,-10876,-10364, -9852, -9340, -8828, -8316, -7932, -7676, -7420, -7164, -6908, -6652, -6396, -6140, -5884, -5628, -5372, -5116, -4860, -4604, -4348, -4092, -3900, -3772, -3644, -3516, -3388, -3260, -3132, -3004, -2876, -2748, -2620, -2492, -2364, -2236, -2108, -1980, -1884, -1820, -1756, -1692, -1628, -1564, -1500, -1436, -1372, -1308, -1244, -1180, -1116, -1052, -988, -924, -876, -844, -812, -780, -748, -716, -684, -652, -620, -588, -556, -524, -492, -460, -428, -396, -372, -356, -340, -324, -308, -292, -276, -260, -244, -228, -212, -196, -180, -164, -148, -132, -120, -112, -104, -96, -88, -80, -72, -64, -56, -48, -40, -32, -24, -16, -8, 0, 32124, 31100, 30076, 29052, 28028, 27004, 25980, 24956, 23932, 22908, 21884, 20860, 19836, 18812, 17788, 16764, 15996, 15484, 14972, 14460, 13948, 13436, 12924, 12412, 11900, 11388, 10876, 10364, 9852, 9340, 8828, 8316, 7932, 7676, 7420, 7164, 6908, 6652, 6396, 6140, 5884, 5628, 5372, 5116, 4860, 4604, 4348, 4092, 3900, 3772, 3644, 3516, 3388, 3260, 3132, 3004, 2876, 2748, 2620, 2492, 2364, 2236, 2108, 1980, 1884, 1820, 1756, 1692, 1628, 1564, 1500, 1436, 1372, 1308, 1244, 1180, 1116, 1052, 988, 924, 876, 844, 812, 780, 748, 716, 684, 652, 620, 588, 556, 524, 492, 460, 428, 396, 372, 356, 340, 324, 308, 292, 276, 260, 244, 228, 212, 196, 180, 164, 148, 132, 120, 112, 104, 96, 88, 80, 72, 64, 56, 48, 40, 32, 24, 16, 8, 0 }; /* A quick'n'dirty lookup table to decode an aLaw-encoded byte sample into a * signed 16-bit sample */ const ALshort aLawDecompressionTable[256] = { -5504, -5248, -6016, -5760, -4480, -4224, -4992, -4736, -7552, -7296, -8064, -7808, -6528, -6272, -7040, -6784, -2752, -2624, -3008, -2880, -2240, -2112, -2496, -2368, -3776, -3648, -4032, -3904, -3264, -3136, -3520, -3392, -22016,-20992,-24064,-23040,-17920,-16896,-19968,-18944, -30208,-29184,-32256,-31232,-26112,-25088,-28160,-27136, -11008,-10496,-12032,-11520, -8960, -8448, -9984, -9472, -15104,-14592,-16128,-15616,-13056,-12544,-14080,-13568, -344, -328, -376, -360, -280, -264, -312, -296, -472, -456, -504, -488, -408, -392, -440, -424, -88, -72, -120, -104, -24, -8, -56, -40, -216, -200, -248, -232, -152, -136, -184, -168, -1376, -1312, -1504, -1440, -1120, -1056, -1248, -1184, -1888, -1824, -2016, -1952, -1632, -1568, -1760, -1696, -688, -656, -752, -720, -560, -528, -624, -592, -944, -912, -1008, -976, -816, -784, -880, -848, 5504, 5248, 6016, 5760, 4480, 4224, 4992, 4736, 7552, 7296, 8064, 7808, 6528, 6272, 7040, 6784, 2752, 2624, 3008, 2880, 2240, 2112, 2496, 2368, 3776, 3648, 4032, 3904, 3264, 3136, 3520, 3392, 22016, 20992, 24064, 23040, 17920, 16896, 19968, 18944, 30208, 29184, 32256, 31232, 26112, 25088, 28160, 27136, 11008, 10496, 12032, 11520, 8960, 8448, 9984, 9472, 15104, 14592, 16128, 15616, 13056, 12544, 14080, 13568, 344, 328, 376, 360, 280, 264, 312, 296, 472, 456, 504, 488, 408, 392, 440, 424, 88, 72, 120, 104, 24, 8, 56, 40, 216, 200, 248, 232, 152, 136, 184, 168, 1376, 1312, 1504, 1440, 1120, 1056, 1248, 1184, 1888, 1824, 2016, 1952, 1632, 1568, 1760, 1696, 688, 656, 752, 720, 560, 528, 624, 592, 944, 912, 1008, 976, 816, 784, 880, 848 }; static void DecodeIMA4Block(ALshort *dst, const ALubyte *src, ALint numchans, ALsizei align) { ALint sample[MAX_INPUT_CHANNELS] = { 0 }; ALint index[MAX_INPUT_CHANNELS] = { 0 }; ALuint code[MAX_INPUT_CHANNELS] = { 0 }; ALsizei c, i; for(c = 0;c < numchans;c++) { sample[c] = *(src++); sample[c] |= *(src++) << 8; sample[c] = (sample[c]^0x8000) - 32768; index[c] = *(src++); index[c] |= *(src++) << 8; index[c] = (index[c]^0x8000) - 32768; index[c] = clampi(index[c], 0, 88); dst[c] = sample[c]; } for(i = 1;i < align;i++) { if((i&7) == 1) { for(c = 0;c < numchans;c++) { code[c] = *(src++); code[c] |= *(src++) << 8; code[c] |= *(src++) << 16; code[c] |= *(src++) << 24; } } for(c = 0;c < numchans;c++) { int nibble = code[c]&0xf; code[c] >>= 4; sample[c] += IMA4Codeword[nibble] * IMAStep_size[index[c]] / 8; sample[c] = clampi(sample[c], -32768, 32767); index[c] += IMA4Index_adjust[nibble]; index[c] = clampi(index[c], 0, 88); *(dst++) = sample[c]; } } } static void DecodeMSADPCMBlock(ALshort *dst, const ALubyte *src, ALint numchans, ALsizei align) { ALubyte blockpred[MAX_INPUT_CHANNELS] = { 0 }; ALint delta[MAX_INPUT_CHANNELS] = { 0 }; ALshort samples[MAX_INPUT_CHANNELS][2] = { { 0, 0 } }; ALint c, i; for(c = 0;c < numchans;c++) { blockpred[c] = *(src++); blockpred[c] = minu(blockpred[c], 6); } for(c = 0;c < numchans;c++) { delta[c] = *(src++); delta[c] |= *(src++) << 8; delta[c] = (delta[c]^0x8000) - 32768; } for(c = 0;c < numchans;c++) { samples[c][0] = *(src++); samples[c][0] |= *(src++) << 8; samples[c][0] = (samples[c][0]^0x8000) - 32768; } for(c = 0;c < numchans;c++) { samples[c][1] = *(src++); samples[c][1] |= *(src++) << 8; samples[c][1] = (samples[c][1]^0x8000) - 0x8000; } /* Second sample is written first. */ for(c = 0;c < numchans;c++) *(dst++) = samples[c][1]; for(c = 0;c < numchans;c++) *(dst++) = samples[c][0]; for(i = 2;i < align;i++) { for(c = 0;c < numchans;c++) { const ALint num = (i*numchans) + c; ALint nibble, pred; /* Read the nibble (first is in the upper bits). */ if(!(num&1)) nibble = (*src>>4)&0x0f; else nibble = (*(src++))&0x0f; pred = (samples[c][0]*MSADPCMAdaptionCoeff[blockpred[c]][0] + samples[c][1]*MSADPCMAdaptionCoeff[blockpred[c]][1]) / 256; pred += ((nibble^0x08) - 0x08) * delta[c]; pred = clampi(pred, -32768, 32767); samples[c][1] = samples[c][0]; samples[c][0] = pred; delta[c] = (MSADPCMAdaption[nibble] * delta[c]) / 256; delta[c] = maxi(16, delta[c]); *(dst++) = pred; } } } void Convert_ALshort_ALima4(ALshort *dst, const ALubyte *src, ALsizei numchans, ALsizei len, ALsizei align) { ALsizei byte_align = ((align-1)/2 + 4) * numchans; ALsizei i; assert(align > 0 && (len%align) == 0); for(i = 0;i < len;i += align) { DecodeIMA4Block(dst, src, numchans, align); src += byte_align; dst += align*numchans; } } void Convert_ALshort_ALmsadpcm(ALshort *dst, const ALubyte *src, ALsizei numchans, ALsizei len, ALsizei align) { ALsizei byte_align = ((align-2)/2 + 7) * numchans; ALsizei i; assert(align > 1 && (len%align) == 0); for(i = 0;i < len;i += align) { DecodeMSADPCMBlock(dst, src, numchans, align); src += byte_align; dst += align*numchans; } } openal-soft-openal-soft-1.19.1/README.md000066400000000000000000000055011335774445300175560ustar00rootroot00000000000000OpenAL soft =========== `master` branch CI status : [![Build Status](https://travis-ci.org/kcat/openal-soft.svg?branch=master)](https://travis-ci.org/kcat/openal-soft) [![Windows Build Status](https://ci.appveyor.com/api/projects/status/github/kcat/openal-soft?branch=master&svg=true)](https://ci.appveyor.com/api/projects/status/github/kcat/openal-soft?branch=master&svg=true) OpenAL Soft is an LGPL-licensed, cross-platform, software implementation of the OpenAL 3D audio API. It's forked from the open-sourced Windows version available originally from openal.org's SVN repository (now defunct). OpenAL provides capabilities for playing audio in a virtual 3D environment. Distance attenuation, doppler shift, and directional sound emitters are among the features handled by the API. More advanced effects, including air absorption, occlusion, and environmental reverb, are available through the EFX extension. It also facilitates streaming audio, multi-channel buffers, and audio capture. More information is available on the [official website](http://openal-soft.org/) Source Install ------------- To install OpenAL Soft, use your favorite shell to go into the build/ directory, and run: ```bash cmake .. ``` Assuming configuration went well, you can then build it, typically using GNU Make (KDevelop, MSVC, and others are possible depending on your system setup and CMake configuration). Please Note: Double check that the appropriate backends were detected. Often, complaints of no sound, crashing, and missing devices can be solved by making sure the correct backends are being used. CMake's output will identify which backends were enabled. For most systems, you will likely want to make sure ALSA, OSS, and PulseAudio were detected (if your target system uses them). For Windows, make sure DirectSound was detected. Utilities --------- The source package comes with an informational utility, openal-info, and is built by default. It prints out information provided by the ALC and AL sub- systems, including discovered devices, version information, and extensions. Configuration ------------- OpenAL Soft can be configured on a per-user and per-system basis. This allows users and sysadmins to control information provided to applications, as well as application-agnostic behavior of the library. See alsoftrc.sample for available settings. Acknowledgements ---------------- Special thanks go to: - Creative Labs for the original source code this is based off of. - Christopher Fitzgerald for the current reverb effect implementation, and helping with the low-pass and HRTF filters. - Christian Borss for the 3D panning code previous versions used as a base. - Ben Davis for the idea behind a previous version of the click-removal code. - Richard Furse for helping with my understanding of Ambisonics that is used by the various parts of the library. openal-soft-openal-soft-1.19.1/XCompile-Android.txt000066400000000000000000000032541335774445300221410ustar00rootroot00000000000000# Cross-compiling requires CMake 2.6 or newer. Example: # cmake .. -DCMAKE_TOOLCHAIN_FILE=../XCompile-Android.txt -DHOST=arm-linux-androideabi # Where 'arm-linux-androideabi' is the host prefix for the cross-compiler. If # you already have a toolchain file setup, you may use that instead of this # file. Make sure to set CMAKE_FIND_ROOT_PATH to where the NDK toolchain was # installed (e.g. "$ENV{HOME}/toolchains/arm-linux-androideabi-r10c-21"). # the name of the target operating system SET(CMAKE_SYSTEM_NAME Linux) # which compilers to use for C and C++ SET(CMAKE_C_COMPILER "${HOST}-gcc") SET(CMAKE_CXX_COMPILER "${HOST}-g++") SET(CMAKE_RC_COMPILER "${HOST}-windres") # here is the target environment located SET(CMAKE_FIND_ROOT_PATH "SET THIS TO THE NDK TOOLCHAIN'S INSTALL PATH") # here is where stuff gets installed to SET(CMAKE_INSTALL_PREFIX "${CMAKE_FIND_ROOT_PATH}" CACHE STRING "Install path prefix, prepended onto install directories." FORCE) # adjust the default behaviour of the FIND_XXX() commands: # search headers and libraries in the target environment, search # programs in the host environment set(CMAKE_FIND_ROOT_PATH_MODE_PROGRAM NEVER) set(CMAKE_FIND_ROOT_PATH_MODE_LIBRARY ONLY) set(CMAKE_FIND_ROOT_PATH_MODE_INCLUDE ONLY) # set env vars so that pkg-config will look in the appropriate directory for # .pc files (as there seems to be no way to force using ${HOST}-pkg-config) set(ENV{PKG_CONFIG_LIBDIR} "${CMAKE_INSTALL_PREFIX}/lib/pkgconfig") set(ENV{PKG_CONFIG_PATH} "") # Qt4 tools SET(QT_QMAKE_EXECUTABLE ${HOST}-qmake) SET(QT_MOC_EXECUTABLE ${HOST}-moc) SET(QT_RCC_EXECUTABLE ${HOST}-rcc) SET(QT_UIC_EXECUTABLE ${HOST}-uic) SET(QT_LRELEASE_EXECUTABLE ${HOST}-lrelease) openal-soft-openal-soft-1.19.1/XCompile.txt000066400000000000000000000027541335774445300205670ustar00rootroot00000000000000# Cross-compiling requires CMake 2.6 or newer. Example: # cmake .. -DCMAKE_TOOLCHAIN_FILE=../XCompile.txt -DHOST=i686-w64-mingw32 # Where 'i686-w64-mingw32' is the host prefix for your cross-compiler. If you # already have a toolchain file setup, you may use that instead of this file. # the name of the target operating system SET(CMAKE_SYSTEM_NAME Windows) # which compilers to use for C and C++ SET(CMAKE_C_COMPILER "${HOST}-gcc") SET(CMAKE_CXX_COMPILER "${HOST}-g++") SET(CMAKE_RC_COMPILER "${HOST}-windres") # here is the target environment located SET(CMAKE_FIND_ROOT_PATH "/usr/${HOST}") # here is where stuff gets installed to SET(CMAKE_INSTALL_PREFIX "${CMAKE_FIND_ROOT_PATH}" CACHE STRING "Install path prefix, prepended onto install directories." FORCE) # adjust the default behaviour of the FIND_XXX() commands: # search headers and libraries in the target environment, search # programs in the host environment set(CMAKE_FIND_ROOT_PATH_MODE_PROGRAM NEVER) set(CMAKE_FIND_ROOT_PATH_MODE_LIBRARY ONLY) set(CMAKE_FIND_ROOT_PATH_MODE_INCLUDE ONLY) # set env vars so that pkg-config will look in the appropriate directory for # .pc files (as there seems to be no way to force using ${HOST}-pkg-config) set(ENV{PKG_CONFIG_LIBDIR} "${CMAKE_INSTALL_PREFIX}/lib/pkgconfig") set(ENV{PKG_CONFIG_PATH} "") # Qt4 tools SET(QT_QMAKE_EXECUTABLE ${HOST}-qmake) SET(QT_MOC_EXECUTABLE ${HOST}-moc) SET(QT_RCC_EXECUTABLE ${HOST}-rcc) SET(QT_UIC_EXECUTABLE ${HOST}-uic) SET(QT_LRELEASE_EXECUTABLE ${HOST}-lrelease) openal-soft-openal-soft-1.19.1/alsoftrc.sample000066400000000000000000000446261335774445300213320ustar00rootroot00000000000000# OpenAL config file. # # Option blocks may appear multiple times, and duplicated options will take the # last value specified. Environment variables may be specified within option # values, and are automatically substituted when the config file is loaded. # Environment variable names may only contain alpha-numeric characters (a-z, # A-Z, 0-9) and underscores (_), and are prefixed with $. For example, # specifying "$HOME/file.ext" would typically result in something like # "/home/user/file.ext". To specify an actual "$" character, use "$$". # # Device-specific values may be specified by including the device name in the # block name, with "general" replaced by the device name. That is, general # options for the device "Name of Device" would be in the [Name of Device] # block, while ALSA options would be in the [alsa/Name of Device] block. # Options marked as "(global)" are not influenced by the device. # # The system-wide settings can be put in /etc/openal/alsoft.conf and user- # specific override settings in $HOME/.alsoftrc. # For Windows, these settings should go into $AppData\alsoft.ini # # Option and block names are case-senstive. The supplied values are only hints # and may not be honored (though generally it'll try to get as close as # possible). Note: options that are left unset may default to app- or system- # specified values. These are the current available settings: ## ## General stuff ## [general] ## disable-cpu-exts: (global) # Disables use of specialized methods that use specific CPU intrinsics. # Certain methods may utilize CPU extensions for improved performance, and # this option is useful for preventing some or all of those methods from being # used. The available extensions are: sse, sse2, sse3, sse4.1, and neon. # Specifying 'all' disables use of all such specialized methods. #disable-cpu-exts = ## drivers: (global) # Sets the backend driver list order, comma-seperated. Unknown backends and # duplicated names are ignored. Unlisted backends won't be considered for use # unless the list is ended with a comma (e.g. 'oss,' will try OSS first before # other backends, while 'oss' will try OSS only). Backends prepended with - # won't be considered for use (e.g. '-oss,' will try all available backends # except OSS). An empty list means to try all backends. #drivers = ## channels: # Sets the output channel configuration. If left unspecified, one will try to # be detected from the system, and defaulting to stereo. The available values # are: mono, stereo, quad, surround51, surround51rear, surround61, surround71, # ambi1, ambi2, ambi3. Note that the ambi* configurations provide ambisonic # channels of the given order (using ACN ordering and SN3D normalization by # default), which need to be decoded to play correctly on speakers. #channels = ## sample-type: # Sets the output sample type. Currently, all mixing is done with 32-bit float # and converted to the output sample type as needed. Available values are: # int8 - signed 8-bit int # uint8 - unsigned 8-bit int # int16 - signed 16-bit int # uint16 - unsigned 16-bit int # int32 - signed 32-bit int # uint32 - unsigned 32-bit int # float32 - 32-bit float #sample-type = float32 ## frequency: # Sets the output frequency. If left unspecified it will try to detect a # default from the system, otherwise it will default to 44100. #frequency = ## period_size: # Sets the update period size, in frames. This is the number of frames needed # for each mixing update. Acceptable values range between 64 and 8192. #period_size = 1024 ## periods: # Sets the number of update periods. Higher values create a larger mix ahead, # which helps protect against skips when the CPU is under load, but increases # the delay between a sound getting mixed and being heard. Acceptable values # range between 2 and 16. #periods = 3 ## stereo-mode: # Specifies if stereo output is treated as being headphones or speakers. With # headphones, HRTF or crossfeed filters may be used for better audio quality. # Valid settings are auto, speakers, and headphones. #stereo-mode = auto ## stereo-encoding: # Specifies the encoding method for non-HRTF stereo output. 'panpot' (default) # uses standard amplitude panning (aka pair-wise, stereo pair, etc) between # -30 and +30 degrees, while 'uhj' creates stereo-compatible two-channel UHJ # output, which encodes some surround sound information into stereo output # that can be decoded with a surround sound receiver. If crossfeed filters are # used, UHJ is disabled. #stereo-encoding = panpot ## ambi-format: # Specifies the channel order and normalization for the "ambi*" set of channel # configurations. Valid settings are: fuma, acn+sn3d, acn+n3d #ambi-format = acn+sn3d ## hrtf: # Controls HRTF processing. These filters provide better spatialization of # sounds while using headphones, but do require a bit more CPU power. The # default filters will only work with 44100hz or 48000hz stereo output. While # HRTF is used, the cf_level option is ignored. Setting this to auto (default) # will allow HRTF to be used when headphones are detected or the app requests # it, while setting true or false will forcefully enable or disable HRTF # respectively. #hrtf = auto ## default-hrtf: # Specifies the default HRTF to use. When multiple HRTFs are available, this # determines the preferred one to use if none are specifically requested. Note # that this is the enumerated HRTF name, not necessarily the filename. #default-hrtf = ## hrtf-paths: # Specifies a comma-separated list of paths containing HRTF data sets. The # format of the files are described in docs/hrtf.txt. The files within the # directories must have the .mhr file extension to be recognized. By default, # OS-dependent data paths will be used. They will also be used if the list # ends with a comma. On Windows this is: # $AppData\openal\hrtf # And on other systems, it's (in order): # $XDG_DATA_HOME/openal/hrtf (defaults to $HOME/.local/share/openal/hrtf) # $XDG_DATA_DIRS/openal/hrtf (defaults to /usr/local/share/openal/hrtf and # /usr/share/openal/hrtf) #hrtf-paths = ## cf_level: # Sets the crossfeed level for stereo output. Valid values are: # 0 - No crossfeed # 1 - Low crossfeed # 2 - Middle crossfeed # 3 - High crossfeed (virtual speakers are closer to itself) # 4 - Low easy crossfeed # 5 - Middle easy crossfeed # 6 - High easy crossfeed # Users of headphones may want to try various settings. Has no effect on non- # stereo modes. #cf_level = 0 ## resampler: (global) # Selects the resampler used when mixing sources. Valid values are: # point - nearest sample, no interpolation # linear - extrapolates samples using a linear slope between samples # cubic - extrapolates samples using a Catmull-Rom spline # bsinc12 - extrapolates samples using a band-limited Sinc filter (varying # between 12 and 24 points, with anti-aliasing) # bsinc24 - extrapolates samples using a band-limited Sinc filter (varying # between 24 and 48 points, with anti-aliasing) #resampler = linear ## rt-prio: (global) # Sets real-time priority for the mixing thread. Not all drivers may use this # (eg. PortAudio) as they already control the priority of the mixing thread. # 0 and negative values will disable it. Note that this may constitute a # security risk since a real-time priority thread can indefinitely block # normal-priority threads if it fails to wait. As such, the default is # disabled. #rt-prio = 0 ## sources: # Sets the maximum number of allocatable sources. Lower values may help for # systems with apps that try to play more sounds than the CPU can handle. #sources = 256 ## slots: # Sets the maximum number of Auxiliary Effect Slots an app can create. A slot # can use a non-negligible amount of CPU time if an effect is set on it even # if no sources are feeding it, so this may help when apps use more than the # system can handle. #slots = 64 ## sends: # Limits the number of auxiliary sends allowed per source. Setting this higher # than the default has no effect. #sends = 16 ## front-stablizer: # Applies filters to "stablize" front sound imaging. A psychoacoustic method # is used to generate a front-center channel signal from the front-left and # front-right channels, improving the front response by reducing the combing # artifacts and phase errors. Consequently, it will only work with channel # configurations that include front-left, front-right, and front-center. #front-stablizer = false ## output-limiter: # Applies a gain limiter on the final mixed output. This reduces the volume # when the output samples would otherwise clamp, avoiding excessive clipping # noise. #output-limiter = true ## dither: # Applies dithering on the final mix, for 8- and 16-bit output by default. # This replaces the distortion created by nearest-value quantization with low- # level whitenoise. #dither = true ## dither-depth: # Quantization bit-depth for dithered output. A value of 0 (or less) will # match the output sample depth. For int32, uint32, and float32 output, 0 will # disable dithering because they're at or beyond the rendered precision. The # maximum dither depth is 24. #dither-depth = 0 ## volume-adjust: # A global volume adjustment for source output, expressed in decibels. The # value is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will # be a scale of 4x, etc. Similarly, -6 will be x1/2, and -12 is about x1/4. A # value of 0 means no change. #volume-adjust = 0 ## excludefx: (global) # Sets which effects to exclude, preventing apps from using them. This can # help for apps that try to use effects which are too CPU intensive for the # system to handle. Available effects are: eaxreverb,reverb,autowah,chorus, # compressor,distortion,echo,equalizer,flanger,modulator,dedicated,pshifter, # fshifter #excludefx = ## default-reverb: (global) # A reverb preset that applies by default to all sources on send 0 # (applications that set their own slots on send 0 will override this). # Available presets are: None, Generic, PaddedCell, Room, Bathroom, # Livingroom, Stoneroom, Auditorium, ConcertHall, Cave, Arena, Hangar, # CarpetedHallway, Hallway, StoneCorridor, Alley, Forest, City, Moutains, # Quarry, Plain, ParkingLot, SewerPipe, Underwater, Drugged, Dizzy, Psychotic. #default-reverb = ## trap-alc-error: (global) # Generates a SIGTRAP signal when an ALC device error is generated, on systems # that support it. This helps when debugging, while trying to find the cause # of a device error. On Windows, a breakpoint exception is generated. #trap-alc-error = false ## trap-al-error: (global) # Generates a SIGTRAP signal when an AL context error is generated, on systems # that support it. This helps when debugging, while trying to find the cause # of a context error. On Windows, a breakpoint exception is generated. #trap-al-error = false ## ## Ambisonic decoder stuff ## [decoder] ## hq-mode: # Enables a high-quality ambisonic decoder. This mode is capable of frequency- # dependent processing, creating a better reproduction of 3D sound rendering # over surround sound speakers. Enabling this also requires specifying decoder # configuration files for the appropriate speaker configuration you intend to # use (see the quad, surround51, etc options below). Currently, up to third- # order decoding is supported. hq-mode = false ## distance-comp: # Enables compensation for the speakers' relative distances to the listener. # This applies the necessary delays and attenuation to make the speakers # behave as though they are all equidistant, which is important for proper # playback of 3D sound rendering. Requires the proper distances to be # specified in the decoder configuration file. distance-comp = true ## nfc: # Enables near-field control filters. This simulates and compensates for low- # frequency effects caused by the curvature of nearby sound-waves, which # creates a more realistic perception of sound distance. Note that the effect # may be stronger or weaker than intended if the application doesn't use or # specify an appropriate unit scale, or if incorrect speaker distances are set # in the decoder configuration file. Requires hq-mode to be enabled. nfc = true ## nfc-ref-delay # Specifies the reference delay value for ambisonic output. When channels is # set to one of the ambi* formats, this option enables NFC-HOA output with the # specified Reference Delay parameter. The specified value can then be shared # with an appropriate NFC-HOA decoder to reproduce correct near-field effects. # Keep in mind that despite being designed for higher-order ambisonics, this # applies to first-order output all the same. When left unset, normal output # is created with no near-field simulation. nfc-ref-delay = ## quad: # Decoder configuration file for Quadraphonic channel output. See # docs/ambdec.txt for a description of the file format. quad = ## surround51: # Decoder configuration file for 5.1 Surround (Side and Rear) channel output. # See docs/ambdec.txt for a description of the file format. surround51 = ## surround61: # Decoder configuration file for 6.1 Surround channel output. See # docs/ambdec.txt for a description of the file format. surround61 = ## surround71: # Decoder configuration file for 7.1 Surround channel output. See # docs/ambdec.txt for a description of the file format. Note: This can be used # to enable 3D7.1 with the appropriate configuration and speaker placement, # see docs/3D7.1.txt. surround71 = ## ## Reverb effect stuff (includes EAX reverb) ## [reverb] ## boost: (global) # A global amplification for reverb output, expressed in decibels. The value # is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will be a # scale of 4x, etc. Similarly, -6 will be about half, and -12 about 1/4th. A # value of 0 means no change. #boost = 0 ## ## PulseAudio backend stuff ## [pulse] ## spawn-server: (global) # Attempts to autospawn a PulseAudio server whenever needed (initializing the # backend, enumerating devices, etc). Setting autospawn to false in Pulse's # client.conf will still prevent autospawning even if this is set to true. #spawn-server = true ## allow-moves: (global) # Allows PulseAudio to move active streams to different devices. Note that the # device specifier (seen by applications) will not be updated when this # occurs, and neither will the AL device configuration (sample rate, format, # etc). #allow-moves = false ## fix-rate: # Specifies whether to match the playback stream's sample rate to the device's # sample rate. Enabling this forces OpenAL Soft to mix sources and effects # directly to the actual output rate, avoiding a second resample pass by the # PulseAudio server. #fix-rate = false ## ## ALSA backend stuff ## [alsa] ## device: (global) # Sets the device name for the default playback device. #device = default ## device-prefix: (global) # Sets the prefix used by the discovered (non-default) playback devices. This # will be appended with "CARD=c,DEV=d", where c is the card id and d is the # device index for the requested device name. #device-prefix = plughw: ## device-prefix-*: (global) # Card- and device-specific prefixes may be used to override the device-prefix # option. The option may specify the card id (eg, device-prefix-NVidia), or # the card id and device index (eg, device-prefix-NVidia-0). The card id is # case-sensitive. #device-prefix- = ## capture: (global) # Sets the device name for the default capture device. #capture = default ## capture-prefix: (global) # Sets the prefix used by the discovered (non-default) capture devices. This # will be appended with "CARD=c,DEV=d", where c is the card id and d is the # device number for the requested device name. #capture-prefix = plughw: ## capture-prefix-*: (global) # Card- and device-specific prefixes may be used to override the # capture-prefix option. The option may specify the card id (eg, # capture-prefix-NVidia), or the card id and device index (eg, # capture-prefix-NVidia-0). The card id is case-sensitive. #capture-prefix- = ## mmap: # Sets whether to try using mmap mode (helps reduce latencies and CPU # consumption). If mmap isn't available, it will automatically fall back to # non-mmap mode. True, yes, on, and non-0 values will attempt to use mmap. 0 # and anything else will force mmap off. #mmap = true ## allow-resampler: # Specifies whether to allow ALSA's built-in resampler. Enabling this will # allow the playback device to be set to a different sample rate than the # actual output, causing ALSA to apply its own resampling pass after OpenAL # Soft resamples and mixes the sources and effects for output. #allow-resampler = false ## ## OSS backend stuff ## [oss] ## device: (global) # Sets the device name for OSS output. #device = /dev/dsp ## capture: (global) # Sets the device name for OSS capture. #capture = /dev/dsp ## ## Solaris backend stuff ## [solaris] ## device: (global) # Sets the device name for Solaris output. #device = /dev/audio ## ## QSA backend stuff ## [qsa] ## ## JACK backend stuff ## [jack] ## spawn-server: (global) # Attempts to autospawn a JACK server whenever needed (initializing the # backend, opening devices, etc). #spawn-server = false ## buffer-size: # Sets the update buffer size, in samples, that the backend will keep buffered # to handle the server's real-time processing requests. This value must be a # power of 2, or else it will be rounded up to the next power of 2. If it is # less than JACK's buffer update size, it will be clamped. This option may # be useful in case the server's update size is too small and doesn't give the # mixer time to keep enough audio available for the processing requests. #buffer-size = 0 ## ## WASAPI backend stuff ## [wasapi] ## ## DirectSound backend stuff ## [dsound] ## ## Windows Multimedia backend stuff ## [winmm] ## ## PortAudio backend stuff ## [port] ## device: (global) # Sets the device index for output. Negative values will use the default as # given by PortAudio itself. #device = -1 ## capture: (global) # Sets the device index for capture. Negative values will use the default as # given by PortAudio itself. #capture = -1 ## ## Wave File Writer stuff ## [wave] ## file: (global) # Sets the filename of the wave file to write to. An empty name prevents the # backend from opening, even when explicitly requested. # THIS WILL OVERWRITE EXISTING FILES WITHOUT QUESTION! #file = ## bformat: (global) # Creates AMB format files using first-order ambisonics instead of a standard # single- or multi-channel .wav file. #bformat = false openal-soft-openal-soft-1.19.1/appveyor.yml000066400000000000000000000012221335774445300206630ustar00rootroot00000000000000version: 1.19.0.{build} environment: matrix: - GEN: "Visual Studio 14 2015" CFG: Release - GEN: "Visual Studio 14 2015 Win64" CFG: Release install: # Remove the VS Xamarin targets to reduce AppVeyor specific noise in build # logs. See also http://help.appveyor.com/discussions/problems/4569 - del "C:\Program Files (x86)\MSBuild\14.0\Microsoft.Common.targets\ImportAfter\Xamarin.Common.targets" build_script: - cd build - cmake -G"%GEN%" -DALSOFT_REQUIRE_WINMM=ON -DALSOFT_REQUIRE_DSOUND=ON -DALSOFT_REQUIRE_WASAPI=ON -DALSOFT_EMBED_HRTF_DATA=YES .. - cmake --build . --config %CFG% --clean-first openal-soft-openal-soft-1.19.1/build/000077500000000000000000000000001335774445300173755ustar00rootroot00000000000000openal-soft-openal-soft-1.19.1/build/.empty000066400000000000000000000000001335774445300205220ustar00rootroot00000000000000openal-soft-openal-soft-1.19.1/cmake/000077500000000000000000000000001335774445300173565ustar00rootroot00000000000000openal-soft-openal-soft-1.19.1/cmake/CheckFileOffsetBits.c000066400000000000000000000002551335774445300233320ustar00rootroot00000000000000#include #define KB ((off_t)(1024)) #define MB ((off_t)(KB*1024)) #define GB ((off_t)(MB*1024)) int tb[((GB+GB+GB) > GB) ? 1 : -1]; int main() { return 0; } openal-soft-openal-soft-1.19.1/cmake/CheckFileOffsetBits.cmake000066400000000000000000000034031335774445300241660ustar00rootroot00000000000000# - Check if the _FILE_OFFSET_BITS macro is needed for large files # CHECK_FILE_OFFSET_BITS() # # The following variables may be set before calling this macro to # modify the way the check is run: # # CMAKE_REQUIRED_FLAGS = string of compile command line flags # CMAKE_REQUIRED_DEFINITIONS = list of macros to define (-DFOO=bar) # CMAKE_REQUIRED_INCLUDES = list of include directories # Copyright (c) 2009, Chris Robinson # # Redistribution and use is allowed according to the terms of the LGPL license. MACRO(CHECK_FILE_OFFSET_BITS) IF(NOT DEFINED _FILE_OFFSET_BITS) MESSAGE(STATUS "Checking _FILE_OFFSET_BITS for large files") TRY_COMPILE(__WITHOUT_FILE_OFFSET_BITS_64 ${CMAKE_CURRENT_BINARY_DIR} ${CMAKE_CURRENT_SOURCE_DIR}/cmake/CheckFileOffsetBits.c COMPILE_DEFINITIONS ${CMAKE_REQUIRED_DEFINITIONS}) IF(NOT __WITHOUT_FILE_OFFSET_BITS_64) TRY_COMPILE(__WITH_FILE_OFFSET_BITS_64 ${CMAKE_CURRENT_BINARY_DIR} ${CMAKE_CURRENT_SOURCE_DIR}/cmake/CheckFileOffsetBits.c COMPILE_DEFINITIONS ${CMAKE_REQUIRED_DEFINITIONS} -D_FILE_OFFSET_BITS=64) ENDIF(NOT __WITHOUT_FILE_OFFSET_BITS_64) IF(NOT __WITHOUT_FILE_OFFSET_BITS_64 AND __WITH_FILE_OFFSET_BITS_64) SET(_FILE_OFFSET_BITS 64 CACHE INTERNAL "_FILE_OFFSET_BITS macro needed for large files") MESSAGE(STATUS "Checking _FILE_OFFSET_BITS for large files - 64") ELSE(NOT __WITHOUT_FILE_OFFSET_BITS_64 AND __WITH_FILE_OFFSET_BITS_64) SET(_FILE_OFFSET_BITS "" CACHE INTERNAL "_FILE_OFFSET_BITS macro needed for large files") MESSAGE(STATUS "Checking _FILE_OFFSET_BITS for large files - not needed") ENDIF(NOT __WITHOUT_FILE_OFFSET_BITS_64 AND __WITH_FILE_OFFSET_BITS_64) ENDIF(NOT DEFINED _FILE_OFFSET_BITS) ENDMACRO(CHECK_FILE_OFFSET_BITS)openal-soft-openal-soft-1.19.1/cmake/CheckSharedFunctionExists.cmake000066400000000000000000000110401335774445300254260ustar00rootroot00000000000000# - Check if a symbol exists as a function, variable, or macro # CHECK_SYMBOL_EXISTS( ) # # Check that the is available after including given header # and store the result in a . Specify the list # of files in one argument as a semicolon-separated list. # # If the header files define the symbol as a macro it is considered # available and assumed to work. If the header files declare the # symbol as a function or variable then the symbol must also be # available for linking. If the symbol is a type or enum value # it will not be recognized (consider using CheckTypeSize or # CheckCSourceCompiles). # # The following variables may be set before calling this macro to # modify the way the check is run: # # CMAKE_REQUIRED_FLAGS = string of compile command line flags # CMAKE_REQUIRED_DEFINITIONS = list of macros to define (-DFOO=bar) # CMAKE_REQUIRED_INCLUDES = list of include directories # CMAKE_REQUIRED_LIBRARIES = list of libraries to link #============================================================================= # Copyright 2003-2011 Kitware, Inc. # # Distributed under the OSI-approved BSD License (the "License"); # see accompanying file Copyright.txt for details. # # This software is distributed WITHOUT ANY WARRANTY; without even the # implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. # See the License for more information. #============================================================================= # (To distribute this file outside of CMake, substitute the full # License text for the above reference.) MACRO(CHECK_SHARED_FUNCTION_EXISTS SYMBOL FILES LIBRARY LOCATION VARIABLE) IF(NOT DEFINED "${VARIABLE}" OR "x${${VARIABLE}}" STREQUAL "x${VARIABLE}") SET(CMAKE_CONFIGURABLE_FILE_CONTENT "/* */\n") SET(MACRO_CHECK_SYMBOL_EXISTS_FLAGS ${CMAKE_REQUIRED_FLAGS}) IF(CMAKE_REQUIRED_LIBRARIES) SET(CHECK_SYMBOL_EXISTS_LIBS "-DLINK_LIBRARIES:STRING=${CMAKE_REQUIRED_LIBRARIES};${LIBRARY}") ELSE(CMAKE_REQUIRED_LIBRARIES) SET(CHECK_SYMBOL_EXISTS_LIBS "-DLINK_LIBRARIES:STRING=${LIBRARY}") ENDIF(CMAKE_REQUIRED_LIBRARIES) IF(CMAKE_REQUIRED_INCLUDES) SET(CMAKE_SYMBOL_EXISTS_INCLUDES "-DINCLUDE_DIRECTORIES:STRING=${CMAKE_REQUIRED_INCLUDES}") ELSE(CMAKE_REQUIRED_INCLUDES) SET(CMAKE_SYMBOL_EXISTS_INCLUDES) ENDIF(CMAKE_REQUIRED_INCLUDES) FOREACH(FILE ${FILES}) SET(CMAKE_CONFIGURABLE_FILE_CONTENT "${CMAKE_CONFIGURABLE_FILE_CONTENT}#include <${FILE}>\n") ENDFOREACH(FILE) SET(CMAKE_CONFIGURABLE_FILE_CONTENT "${CMAKE_CONFIGURABLE_FILE_CONTENT}\nvoid cmakeRequireSymbol(int dummy,...){(void)dummy;}\nint main()\n{\n cmakeRequireSymbol(0,&${SYMBOL});\n return 0;\n}\n") CONFIGURE_FILE("${CMAKE_ROOT}/Modules/CMakeConfigurableFile.in" "${CMAKE_CURRENT_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeTmp/CheckSymbolExists.c" @ONLY) MESSAGE(STATUS "Looking for ${SYMBOL} in ${LIBRARY}") TRY_COMPILE(${VARIABLE} ${CMAKE_CURRENT_BINARY_DIR} ${CMAKE_CURRENT_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeTmp/CheckSymbolExists.c COMPILE_DEFINITIONS ${CMAKE_REQUIRED_DEFINITIONS} CMAKE_FLAGS -DCOMPILE_DEFINITIONS:STRING=${MACRO_CHECK_SYMBOL_EXISTS_FLAGS} -DLINK_DIRECTORIES:STRING=${LOCATION} "${CHECK_SYMBOL_EXISTS_LIBS}" "${CMAKE_SYMBOL_EXISTS_INCLUDES}" OUTPUT_VARIABLE OUTPUT) IF(${VARIABLE}) MESSAGE(STATUS "Looking for ${SYMBOL} in ${LIBRARY} - found") SET(${VARIABLE} 1 CACHE INTERNAL "Have symbol ${SYMBOL} in ${LIBRARY}") FILE(APPEND ${CMAKE_CURRENT_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeOutput.log "Determining if the ${SYMBOL} " "exist in ${LIBRARY} passed with the following output:\n" "${OUTPUT}\nFile ${CMAKE_CURRENT_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeTmp/CheckSymbolExists.c:\n" "${CMAKE_CONFIGURABLE_FILE_CONTENT}\n") ELSE(${VARIABLE}) MESSAGE(STATUS "Looking for ${SYMBOL} in ${LIBRARY} - not found.") SET(${VARIABLE} "" CACHE INTERNAL "Have symbol ${SYMBOL} in ${LIBRARY}") FILE(APPEND ${CMAKE_CURRENT_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeError.log "Determining if the ${SYMBOL} " "exist in ${LIBRARY} failed with the following output:\n" "${OUTPUT}\nFile ${CMAKE_CURRENT_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeTmp/CheckSymbolExists.c:\n" "${CMAKE_CONFIGURABLE_FILE_CONTENT}\n") ENDIF(${VARIABLE}) ENDIF(NOT DEFINED "${VARIABLE}" OR "x${${VARIABLE}}" STREQUAL "x${VARIABLE}") ENDMACRO(CHECK_SHARED_FUNCTION_EXISTS) openal-soft-openal-soft-1.19.1/cmake/FindALSA.cmake000066400000000000000000000063211335774445300217030ustar00rootroot00000000000000# - Find alsa # Find the alsa libraries (asound) # # This module defines the following variables: # ALSA_FOUND - True if ALSA_INCLUDE_DIR & ALSA_LIBRARY are found # ALSA_LIBRARIES - Set when ALSA_LIBRARY is found # ALSA_INCLUDE_DIRS - Set when ALSA_INCLUDE_DIR is found # # ALSA_INCLUDE_DIR - where to find asoundlib.h, etc. # ALSA_LIBRARY - the asound library # ALSA_VERSION_STRING - the version of alsa found (since CMake 2.8.8) # #============================================================================= # Copyright 2009-2011 Kitware, Inc. # Copyright 2009-2011 Philip Lowman # # Redistribution and use in source and binary forms, with or without # modification, are permitted provided that the following conditions are # met: # # * Redistributions of source code must retain the above copyright notice, # this list of conditions and the following disclaimer. # # * Redistributions in binary form must reproduce the above copyright notice, # this list of conditions and the following disclaimer in the documentation # and/or other materials provided with the distribution. # # * The names of Kitware, Inc., the Insight Consortium, or the names of # any consortium members, or of any contributors, may not be used to # endorse or promote products derived from this software without # specific prior written permission. # # THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDER AND CONTRIBUTORS ``AS IS'' # AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE # IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE # ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHORS OR CONTRIBUTORS BE LIABLE FOR # ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL # DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR # SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER # CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, # OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE # OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. #============================================================================= find_path(ALSA_INCLUDE_DIR NAMES alsa/asoundlib.h DOC "The ALSA (asound) include directory" ) find_library(ALSA_LIBRARY NAMES asound DOC "The ALSA (asound) library" ) if(ALSA_INCLUDE_DIR AND EXISTS "${ALSA_INCLUDE_DIR}/alsa/version.h") file(STRINGS "${ALSA_INCLUDE_DIR}/alsa/version.h" alsa_version_str REGEX "^#define[\t ]+SND_LIB_VERSION_STR[\t ]+\".*\"") string(REGEX REPLACE "^.*SND_LIB_VERSION_STR[\t ]+\"([^\"]*)\".*$" "\\1" ALSA_VERSION_STRING "${alsa_version_str}") unset(alsa_version_str) endif() # handle the QUIETLY and REQUIRED arguments and set ALSA_FOUND to TRUE if # all listed variables are TRUE include(FindPackageHandleStandardArgs) find_package_handle_standard_args(ALSA REQUIRED_VARS ALSA_LIBRARY ALSA_INCLUDE_DIR VERSION_VAR ALSA_VERSION_STRING) if(ALSA_FOUND) set( ALSA_LIBRARIES ${ALSA_LIBRARY} ) set( ALSA_INCLUDE_DIRS ${ALSA_INCLUDE_DIR} ) endif() mark_as_advanced(ALSA_INCLUDE_DIR ALSA_LIBRARY) openal-soft-openal-soft-1.19.1/cmake/FindAudioIO.cmake000066400000000000000000000011051335774445300224470ustar00rootroot00000000000000# - Find AudioIO includes and libraries # # AUDIOIO_FOUND - True if AUDIOIO_INCLUDE_DIR is found # AUDIOIO_INCLUDE_DIRS - Set when AUDIOIO_INCLUDE_DIR is found # # AUDIOIO_INCLUDE_DIR - where to find sys/audioio.h, etc. # find_path(AUDIOIO_INCLUDE_DIR NAMES sys/audioio.h DOC "The AudioIO include directory" ) include(FindPackageHandleStandardArgs) find_package_handle_standard_args(AudioIO REQUIRED_VARS AUDIOIO_INCLUDE_DIR) if(AUDIOIO_FOUND) set(AUDIOIO_INCLUDE_DIRS ${AUDIOIO_INCLUDE_DIR}) endif() mark_as_advanced(AUDIOIO_INCLUDE_DIR) openal-soft-openal-soft-1.19.1/cmake/FindDSound.cmake000066400000000000000000000024551335774445300223630ustar00rootroot00000000000000# - Find DirectSound includes and libraries # # DSOUND_FOUND - True if DSOUND_INCLUDE_DIR & DSOUND_LIBRARY are found # DSOUND_LIBRARIES - Set when DSOUND_LIBRARY is found # DSOUND_INCLUDE_DIRS - Set when DSOUND_INCLUDE_DIR is found # # DSOUND_INCLUDE_DIR - where to find dsound.h, etc. # DSOUND_LIBRARY - the dsound library # if (WIN32) include(FindWindowsSDK) if (WINDOWSSDK_FOUND) get_windowssdk_library_dirs(${WINDOWSSDK_PREFERRED_DIR} WINSDK_LIB_DIRS) get_windowssdk_include_dirs(${WINDOWSSDK_PREFERRED_DIR} WINSDK_INCLUDE_DIRS) endif() endif() # DSOUND_INCLUDE_DIR find_path(DSOUND_INCLUDE_DIR NAMES "dsound.h" PATHS "${DXSDK_DIR}" ${WINSDK_INCLUDE_DIRS} PATH_SUFFIXES include DOC "The DirectSound include directory") # DSOUND_LIBRARY find_library(DSOUND_LIBRARY NAMES dsound PATHS "${DXSDK_DIR}" ${WINSDK_LIB_DIRS} PATH_SUFFIXES lib lib/x86 lib/x64 DOC "The DirectSound library") include(FindPackageHandleStandardArgs) find_package_handle_standard_args(DSound REQUIRED_VARS DSOUND_LIBRARY DSOUND_INCLUDE_DIR) if(DSOUND_FOUND) set(DSOUND_LIBRARIES ${DSOUND_LIBRARY}) set(DSOUND_INCLUDE_DIRS ${DSOUND_INCLUDE_DIR}) endif() mark_as_advanced(DSOUND_INCLUDE_DIR DSOUND_LIBRARY) openal-soft-openal-soft-1.19.1/cmake/FindFFmpeg.cmake000066400000000000000000000155221335774445300223320ustar00rootroot00000000000000# vim: ts=2 sw=2 # - Try to find the required ffmpeg components(default: AVFORMAT, AVUTIL, AVCODEC) # # Once done this will define # FFMPEG_FOUND - System has the all required components. # FFMPEG_INCLUDE_DIRS - Include directory necessary for using the required components headers. # FFMPEG_LIBRARIES - Link these to use the required ffmpeg components. # FFMPEG_DEFINITIONS - Compiler switches required for using the required ffmpeg components. # # For each of the components it will additionaly set. # - AVCODEC # - AVDEVICE # - AVFORMAT # - AVUTIL # - POSTPROC # - SWSCALE # - SWRESAMPLE # the following variables will be defined # _FOUND - System has # _INCLUDE_DIRS - Include directory necessary for using the headers # _LIBRARIES - Link these to use # _DEFINITIONS - Compiler switches required for using # _VERSION - The components version # # Copyright (c) 2006, Matthias Kretz, # Copyright (c) 2008, Alexander Neundorf, # Copyright (c) 2011, Michael Jansen, # # Redistribution and use is allowed according to the terms of the BSD license. include(FindPackageHandleStandardArgs) if(NOT FFmpeg_FIND_COMPONENTS) set(FFmpeg_FIND_COMPONENTS AVFORMAT AVCODEC AVUTIL) endif() # ### Macro: set_component_found # # Marks the given component as found if both *_LIBRARIES AND *_INCLUDE_DIRS is present. # macro(set_component_found _component) if(${_component}_LIBRARIES AND ${_component}_INCLUDE_DIRS) # message(STATUS " - ${_component} found.") set(${_component}_FOUND TRUE) else() # message(STATUS " - ${_component} not found.") endif() endmacro() # ### Macro: find_component # # Checks for the given component by invoking pkgconfig and then looking up the libraries and # include directories. # macro(find_component _component _pkgconfig _library _header) if(NOT WIN32) # use pkg-config to get the directories and then use these values # in the FIND_PATH() and FIND_LIBRARY() calls find_package(PkgConfig) if(PKG_CONFIG_FOUND) pkg_check_modules(PC_${_component} ${_pkgconfig}) endif() endif() find_path(${_component}_INCLUDE_DIRS ${_header} HINTS ${FFMPEGSDK_INC} ${PC_LIB${_component}_INCLUDEDIR} ${PC_LIB${_component}_INCLUDE_DIRS} PATH_SUFFIXES ffmpeg ) find_library(${_component}_LIBRARIES NAMES ${_library} HINTS ${FFMPEGSDK_LIB} ${PC_LIB${_component}_LIBDIR} ${PC_LIB${_component}_LIBRARY_DIRS} ) STRING(REGEX REPLACE "/.*" "/version.h" _ver_header ${_header}) if(EXISTS "${${_component}_INCLUDE_DIRS}/${_ver_header}") file(STRINGS "${${_component}_INCLUDE_DIRS}/${_ver_header}" version_str REGEX "^#define[\t ]+LIB${_component}_VERSION_M.*") foreach(_str "${version_str}") if(NOT version_maj) string(REGEX REPLACE "^.*LIB${_component}_VERSION_MAJOR[\t ]+([0-9]*).*$" "\\1" version_maj "${_str}") endif() if(NOT version_min) string(REGEX REPLACE "^.*LIB${_component}_VERSION_MINOR[\t ]+([0-9]*).*$" "\\1" version_min "${_str}") endif() if(NOT version_mic) string(REGEX REPLACE "^.*LIB${_component}_VERSION_MICRO[\t ]+([0-9]*).*$" "\\1" version_mic "${_str}") endif() endforeach() unset(version_str) set(${_component}_VERSION "${version_maj}.${version_min}.${version_mic}" CACHE STRING "The ${_component} version number.") unset(version_maj) unset(version_min) unset(version_mic) endif(EXISTS "${${_component}_INCLUDE_DIRS}/${_ver_header}") set(${_component}_VERSION ${PC_${_component}_VERSION} CACHE STRING "The ${_component} version number.") set(${_component}_DEFINITIONS ${PC_${_component}_CFLAGS_OTHER} CACHE STRING "The ${_component} CFLAGS.") set_component_found(${_component}) mark_as_advanced( ${_component}_INCLUDE_DIRS ${_component}_LIBRARIES ${_component}_DEFINITIONS ${_component}_VERSION) endmacro() set(FFMPEGSDK $ENV{FFMPEG_HOME}) if(FFMPEGSDK) set(FFMPEGSDK_INC "${FFMPEGSDK}/include") set(FFMPEGSDK_LIB "${FFMPEGSDK}/lib") endif() # Check for all possible components. find_component(AVCODEC libavcodec avcodec libavcodec/avcodec.h) find_component(AVFORMAT libavformat avformat libavformat/avformat.h) find_component(AVDEVICE libavdevice avdevice libavdevice/avdevice.h) find_component(AVUTIL libavutil avutil libavutil/avutil.h) find_component(SWSCALE libswscale swscale libswscale/swscale.h) find_component(SWRESAMPLE libswresample swresample libswresample/swresample.h) find_component(POSTPROC libpostproc postproc libpostproc/postprocess.h) # Check if the required components were found and add their stuff to the FFMPEG_* vars. foreach(_component ${FFmpeg_FIND_COMPONENTS}) if(${_component}_FOUND) # message(STATUS "Required component ${_component} present.") set(FFMPEG_LIBRARIES ${FFMPEG_LIBRARIES} ${${_component}_LIBRARIES}) set(FFMPEG_DEFINITIONS ${FFMPEG_DEFINITIONS} ${${_component}_DEFINITIONS}) list(APPEND FFMPEG_INCLUDE_DIRS ${${_component}_INCLUDE_DIRS}) else() # message(STATUS "Required component ${_component} missing.") endif() endforeach() # Add libz if it exists (needed for static ffmpeg builds) find_library(_FFmpeg_HAVE_LIBZ NAMES z) if(_FFmpeg_HAVE_LIBZ) set(FFMPEG_LIBRARIES ${FFMPEG_LIBRARIES} ${_FFmpeg_HAVE_LIBZ}) endif() # Build the include path and library list with duplicates removed. if(FFMPEG_INCLUDE_DIRS) list(REMOVE_DUPLICATES FFMPEG_INCLUDE_DIRS) endif() if(FFMPEG_LIBRARIES) list(REMOVE_DUPLICATES FFMPEG_LIBRARIES) endif() # cache the vars. set(FFMPEG_INCLUDE_DIRS ${FFMPEG_INCLUDE_DIRS} CACHE STRING "The FFmpeg include directories." FORCE) set(FFMPEG_LIBRARIES ${FFMPEG_LIBRARIES} CACHE STRING "The FFmpeg libraries." FORCE) set(FFMPEG_DEFINITIONS ${FFMPEG_DEFINITIONS} CACHE STRING "The FFmpeg cflags." FORCE) mark_as_advanced(FFMPEG_INCLUDE_DIRS FFMPEG_LIBRARIES FFMPEG_DEFINITIONS) # Now set the noncached _FOUND vars for the components. foreach(_component AVCODEC AVDEVICE AVFORMAT AVUTIL POSTPROCESS SWRESAMPLE SWSCALE) set_component_found(${_component}) endforeach () # Compile the list of required vars set(_FFmpeg_REQUIRED_VARS FFMPEG_LIBRARIES FFMPEG_INCLUDE_DIRS) foreach(_component ${FFmpeg_FIND_COMPONENTS}) list(APPEND _FFmpeg_REQUIRED_VARS ${_component}_LIBRARIES ${_component}_INCLUDE_DIRS) endforeach() # Give a nice error message if some of the required vars are missing. find_package_handle_standard_args(FFmpeg DEFAULT_MSG ${_FFmpeg_REQUIRED_VARS}) openal-soft-openal-soft-1.19.1/cmake/FindJACK.cmake000066400000000000000000000050531335774445300216740ustar00rootroot00000000000000# - Find JACK # Find the JACK libraries # # This module defines the following variables: # JACK_FOUND - True if JACK_INCLUDE_DIR & JACK_LIBRARY are found # JACK_INCLUDE_DIRS - where to find jack.h, etc. # JACK_LIBRARIES - the jack library # #============================================================================= # Copyright 2009-2011 Kitware, Inc. # Copyright 2009-2011 Philip Lowman # # Redistribution and use in source and binary forms, with or without # modification, are permitted provided that the following conditions are # met: # # * Redistributions of source code must retain the above copyright notice, # this list of conditions and the following disclaimer. # # * Redistributions in binary form must reproduce the above copyright notice, # this list of conditions and the following disclaimer in the documentation # and/or other materials provided with the distribution. # # * The names of Kitware, Inc., the Insight Consortium, or the names of # any consortium members, or of any contributors, may not be used to # endorse or promote products derived from this software without # specific prior written permission. # # THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDER AND CONTRIBUTORS ``AS IS'' # AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE # IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE # ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHORS OR CONTRIBUTORS BE LIABLE FOR # ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL # DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR # SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER # CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, # OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE # OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. #============================================================================= find_path(JACK_INCLUDE_DIR NAMES jack/jack.h DOC "The JACK include directory" ) find_library(JACK_LIBRARY NAMES jack DOC "The JACK library" ) # handle the QUIETLY and REQUIRED arguments and set JACK_FOUND to TRUE if # all listed variables are TRUE include(FindPackageHandleStandardArgs) find_package_handle_standard_args(JACK REQUIRED_VARS JACK_LIBRARY JACK_INCLUDE_DIR) if(JACK_FOUND) set(JACK_LIBRARIES ${JACK_LIBRARY}) set(JACK_INCLUDE_DIRS ${JACK_INCLUDE_DIR}) endif() mark_as_advanced(JACK_INCLUDE_DIR JACK_LIBRARY) openal-soft-openal-soft-1.19.1/cmake/FindOSS.cmake000066400000000000000000000015221335774445300216250ustar00rootroot00000000000000# - Find OSS includes # # OSS_FOUND - True if OSS_INCLUDE_DIR is found # OSS_INCLUDE_DIRS - Set when OSS_INCLUDE_DIR is found # OSS_LIBRARIES - Set when OSS_LIBRARY is found # # OSS_INCLUDE_DIR - where to find sys/soundcard.h, etc. # OSS_LIBRARY - where to find libossaudio (optional). # find_path(OSS_INCLUDE_DIR NAMES sys/soundcard.h DOC "The OSS include directory" ) find_library(OSS_LIBRARY NAMES ossaudio DOC "Optional OSS library" ) include(FindPackageHandleStandardArgs) find_package_handle_standard_args(OSS REQUIRED_VARS OSS_INCLUDE_DIR) if(OSS_FOUND) set(OSS_INCLUDE_DIRS ${OSS_INCLUDE_DIR}) if(OSS_LIBRARY) set(OSS_LIBRARIES ${OSS_LIBRARY}) else() unset(OSS_LIBRARIES) endif() endif() mark_as_advanced(OSS_INCLUDE_DIR OSS_LIBRARY) openal-soft-openal-soft-1.19.1/cmake/FindPortAudio.cmake000066400000000000000000000017041335774445300230710ustar00rootroot00000000000000# - Find PortAudio includes and libraries # # PORTAUDIO_FOUND - True if PORTAUDIO_INCLUDE_DIR & PORTAUDIO_LIBRARY # are found # PORTAUDIO_LIBRARIES - Set when PORTAUDIO_LIBRARY is found # PORTAUDIO_INCLUDE_DIRS - Set when PORTAUDIO_INCLUDE_DIR is found # # PORTAUDIO_INCLUDE_DIR - where to find portaudio.h, etc. # PORTAUDIO_LIBRARY - the portaudio library # find_path(PORTAUDIO_INCLUDE_DIR NAMES portaudio.h DOC "The PortAudio include directory" ) find_library(PORTAUDIO_LIBRARY NAMES portaudio DOC "The PortAudio library" ) include(FindPackageHandleStandardArgs) find_package_handle_standard_args(PortAudio REQUIRED_VARS PORTAUDIO_LIBRARY PORTAUDIO_INCLUDE_DIR ) if(PORTAUDIO_FOUND) set(PORTAUDIO_LIBRARIES ${PORTAUDIO_LIBRARY}) set(PORTAUDIO_INCLUDE_DIRS ${PORTAUDIO_INCLUDE_DIR}) endif() mark_as_advanced(PORTAUDIO_INCLUDE_DIR PORTAUDIO_LIBRARY) openal-soft-openal-soft-1.19.1/cmake/FindPulseAudio.cmake000066400000000000000000000027751335774445300232460ustar00rootroot00000000000000# - Find PulseAudio includes and libraries # # PULSEAUDIO_FOUND - True if PULSEAUDIO_INCLUDE_DIR & # PULSEAUDIO_LIBRARY are found # PULSEAUDIO_LIBRARIES - Set when PULSEAUDIO_LIBRARY is found # PULSEAUDIO_INCLUDE_DIRS - Set when PULSEAUDIO_INCLUDE_DIR is found # # PULSEAUDIO_INCLUDE_DIR - where to find pulse/pulseaudio.h, etc. # PULSEAUDIO_LIBRARY - the pulse library # PULSEAUDIO_VERSION_STRING - the version of PulseAudio found # find_path(PULSEAUDIO_INCLUDE_DIR NAMES pulse/pulseaudio.h DOC "The PulseAudio include directory" ) find_library(PULSEAUDIO_LIBRARY NAMES pulse DOC "The PulseAudio library" ) if(PULSEAUDIO_INCLUDE_DIR AND EXISTS "${PULSEAUDIO_INCLUDE_DIR}/pulse/version.h") file(STRINGS "${PULSEAUDIO_INCLUDE_DIR}/pulse/version.h" pulse_version_str REGEX "^#define[\t ]+pa_get_headers_version\\(\\)[\t ]+\\(\".*\"\\)") string(REGEX REPLACE "^.*pa_get_headers_version\\(\\)[\t ]+\\(\"([^\"]*)\"\\).*$" "\\1" PULSEAUDIO_VERSION_STRING "${pulse_version_str}") unset(pulse_version_str) endif() include(FindPackageHandleStandardArgs) find_package_handle_standard_args(PulseAudio REQUIRED_VARS PULSEAUDIO_LIBRARY PULSEAUDIO_INCLUDE_DIR VERSION_VAR PULSEAUDIO_VERSION_STRING ) if(PULSEAUDIO_FOUND) set(PULSEAUDIO_LIBRARIES ${PULSEAUDIO_LIBRARY}) set(PULSEAUDIO_INCLUDE_DIRS ${PULSEAUDIO_INCLUDE_DIR}) endif() mark_as_advanced(PULSEAUDIO_INCLUDE_DIR PULSEAUDIO_LIBRARY) openal-soft-openal-soft-1.19.1/cmake/FindQSA.cmake000066400000000000000000000016521335774445300216110ustar00rootroot00000000000000# - Find QSA includes and libraries # # QSA_FOUND - True if QSA_INCLUDE_DIR & QSA_LIBRARY are found # QSA_LIBRARIES - Set when QSA_LIBRARY is found # QSA_INCLUDE_DIRS - Set when QSA_INCLUDE_DIR is found # # QSA_INCLUDE_DIR - where to find sys/asoundlib.h, etc. # QSA_LIBRARY - the asound library # # Only check for QSA on QNX, because it conflicts with ALSA. if("${CMAKE_C_PLATFORM_ID}" STREQUAL "QNX") find_path(QSA_INCLUDE_DIR NAMES sys/asoundlib.h DOC "The QSA include directory" ) find_library(QSA_LIBRARY NAMES asound DOC "The QSA library" ) endif() include(FindPackageHandleStandardArgs) find_package_handle_standard_args(QSA REQUIRED_VARS QSA_LIBRARY QSA_INCLUDE_DIR ) if(QSA_FOUND) set(QSA_LIBRARIES ${QSA_LIBRARY}) set(QSA_INCLUDE_DIRS ${QSA_INCLUDE_DIR}) endif() mark_as_advanced(QSA_INCLUDE_DIR QSA_LIBRARY) openal-soft-openal-soft-1.19.1/cmake/FindSDL2.cmake000066400000000000000000000154441335774445300216750ustar00rootroot00000000000000# Locate SDL2 library # This module defines # SDL2_LIBRARY, the name of the library to link against # SDL2_FOUND, if false, do not try to link to SDL2 # SDL2_INCLUDE_DIR, where to find SDL.h # # This module responds to the the flag: # SDL2_BUILDING_LIBRARY # If this is defined, then no SDL2_main will be linked in because # only applications need main(). # Otherwise, it is assumed you are building an application and this # module will attempt to locate and set the the proper link flags # as part of the returned SDL2_LIBRARY variable. # # Don't forget to include SDL2main.h and SDL2main.m your project for the # OS X framework based version. (Other versions link to -lSDL2main which # this module will try to find on your behalf.) Also for OS X, this # module will automatically add the -framework Cocoa on your behalf. # # # Additional Note: If you see an empty SDL2_CORE_LIBRARY in your configuration # and no SDL2_LIBRARY, it means CMake did not find your SDL2 library # (SDL2.dll, libsdl2.so, SDL2.framework, etc). # Set SDL2_CORE_LIBRARY to point to your SDL2 library, and configure again. # Similarly, if you see an empty SDL2MAIN_LIBRARY, you should set this value # as appropriate. These values are used to generate the final SDL2_LIBRARY # variable, but when these values are unset, SDL2_LIBRARY does not get created. # # # $SDL2DIR is an environment variable that would # correspond to the ./configure --prefix=$SDL2DIR # used in building SDL2. # l.e.galup 9-20-02 # # Modified by Eric Wing. # Added code to assist with automated building by using environmental variables # and providing a more controlled/consistent search behavior. # Added new modifications to recognize OS X frameworks and # additional Unix paths (FreeBSD, etc). # Also corrected the header search path to follow "proper" SDL2 guidelines. # Added a search for SDL2main which is needed by some platforms. # Added a search for threads which is needed by some platforms. # Added needed compile switches for MinGW. # # On OSX, this will prefer the Framework version (if found) over others. # People will have to manually change the cache values of # SDL2_LIBRARY to override this selection or set the CMake environment # CMAKE_INCLUDE_PATH to modify the search paths. # # Note that the header path has changed from SDL2/SDL.h to just SDL.h # This needed to change because "proper" SDL2 convention # is #include "SDL.h", not . This is done for portability # reasons because not all systems place things in SDL2/ (see FreeBSD). # # Ported by Johnny Patterson. This is a literal port for SDL2 of the FindSDL.cmake # module with the minor edit of changing "SDL" to "SDL2" where necessary. This # was not created for redistribution, and exists temporarily pending official # SDL2 CMake modules. #============================================================================= # Copyright 2003-2009 Kitware, Inc. # # Distributed under the OSI-approved BSD License (the "License"); # see accompanying file Copyright.txt for details. # # This software is distributed WITHOUT ANY WARRANTY; without even the # implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. # See the License for more information. #============================================================================= # (To distribute this file outside of CMake, substitute the full # License text for the above reference.) FIND_PATH(SDL2_INCLUDE_DIR SDL.h HINTS $ENV{SDL2DIR} PATH_SUFFIXES include/SDL2 include PATHS ~/Library/Frameworks /Library/Frameworks /usr/local/include/SDL2 /usr/include/SDL2 /sw # Fink /opt/local # DarwinPorts /opt/csw # Blastwave /opt ) #MESSAGE("SDL2_INCLUDE_DIR is ${SDL2_INCLUDE_DIR}") FIND_LIBRARY(SDL2_CORE_LIBRARY NAMES SDL2 HINTS $ENV{SDL2DIR} PATH_SUFFIXES lib64 lib PATHS /sw /opt/local /opt/csw /opt ) #MESSAGE("SDL2_CORE_LIBRARY is ${SDL2_CORE_LIBRARY}") IF(NOT SDL2_BUILDING_LIBRARY) IF(NOT ${SDL2_INCLUDE_DIR} MATCHES ".framework") # Non-OS X framework versions expect you to also dynamically link to # SDL2main. This is mainly for Windows and OS X. Other (Unix) platforms # seem to provide SDL2main for compatibility even though they don't # necessarily need it. FIND_LIBRARY(SDL2MAIN_LIBRARY NAMES SDL2main HINTS $ENV{SDL2DIR} PATH_SUFFIXES lib64 lib PATHS /sw /opt/local /opt/csw /opt ) ENDIF(NOT ${SDL2_INCLUDE_DIR} MATCHES ".framework") ENDIF(NOT SDL2_BUILDING_LIBRARY) # SDL2 may require threads on your system. # The Apple build may not need an explicit flag because one of the # frameworks may already provide it. # But for non-OSX systems, I will use the CMake Threads package. IF(NOT APPLE) FIND_PACKAGE(Threads) ENDIF(NOT APPLE) # MinGW needs an additional library, mwindows # It's total link flags should look like -lmingw32 -lSDL2main -lSDL2 -lmwindows # (Actually on second look, I think it only needs one of the m* libraries.) IF(MINGW) SET(MINGW32_LIBRARY mingw32 CACHE STRING "mwindows for MinGW") ENDIF(MINGW) SET(SDL2_FOUND "NO") IF(SDL2_CORE_LIBRARY) SET(SDL2_LIBRARY_TEMP ${SDL2_CORE_LIBRARY}) # For SDL2main IF(NOT SDL2_BUILDING_LIBRARY) IF(SDL2MAIN_LIBRARY) SET(SDL2_LIBRARY_TEMP ${SDL2MAIN_LIBRARY} ${SDL2_LIBRARY_TEMP}) ENDIF(SDL2MAIN_LIBRARY) ENDIF(NOT SDL2_BUILDING_LIBRARY) # For OS X, SDL2 uses Cocoa as a backend so it must link to Cocoa. # CMake doesn't display the -framework Cocoa string in the UI even # though it actually is there if I modify a pre-used variable. # I think it has something to do with the CACHE STRING. # So I use a temporary variable until the end so I can set the # "real" variable in one-shot. IF(APPLE) SET(SDL2_LIBRARY_TEMP ${SDL2_LIBRARY_TEMP} "-framework Cocoa") ENDIF(APPLE) # For threads, as mentioned Apple doesn't need this. # In fact, there seems to be a problem if I used the Threads package # and try using this line, so I'm just skipping it entirely for OS X. IF(NOT APPLE) SET(SDL2_LIBRARY_TEMP ${SDL2_LIBRARY_TEMP} ${CMAKE_THREAD_LIBS_INIT}) ENDIF(NOT APPLE) # For MinGW library IF(MINGW) SET(SDL2_LIBRARY_TEMP ${MINGW32_LIBRARY} ${SDL2_LIBRARY_TEMP}) ENDIF(MINGW) IF(WIN32) SET(SDL2_LIBRARY_TEMP winmm imm32 version msimg32 ${SDL2_LIBRARY_TEMP}) ENDIF(WIN32) # Set the final string here so the GUI reflects the final state. SET(SDL2_LIBRARY ${SDL2_LIBRARY_TEMP}) SET(SDL2_FOUND "YES") ENDIF(SDL2_CORE_LIBRARY) INCLUDE(FindPackageHandleStandardArgs) FIND_PACKAGE_HANDLE_STANDARD_ARGS(SDL2 REQUIRED_VARS SDL2_LIBRARY SDL2_INCLUDE_DIR) IF(SDL2_STATIC) if (UNIX AND NOT APPLE) EXECUTE_PROCESS(COMMAND sdl2-config --static-libs OUTPUT_VARIABLE SDL2_LINK_FLAGS) STRING(REGEX REPLACE "(\r?\n)+$" "" SDL2_LINK_FLAGS "${SDL2_LINK_FLAGS}") SET(SDL2_LIBRARY ${SDL2_LINK_FLAGS}) ENDIF() ENDIF(SDL2_STATIC) openal-soft-openal-soft-1.19.1/cmake/FindSDL_sound.cmake000066400000000000000000000460561335774445300230260ustar00rootroot00000000000000# - Locates the SDL_sound library # # This module depends on SDL being found and # must be called AFTER FindSDL.cmake or FindSDL2.cmake is called. # # This module defines # SDL_SOUND_INCLUDE_DIR, where to find SDL_sound.h # SDL_SOUND_FOUND, if false, do not try to link to SDL_sound # SDL_SOUND_LIBRARIES, this contains the list of libraries that you need # to link against. This is a read-only variable and is marked INTERNAL. # SDL_SOUND_EXTRAS, this is an optional variable for you to add your own # flags to SDL_SOUND_LIBRARIES. This is prepended to SDL_SOUND_LIBRARIES. # This is available mostly for cases this module failed to anticipate for # and you must add additional flags. This is marked as ADVANCED. # SDL_SOUND_VERSION_STRING, human-readable string containing the version of SDL_sound # # This module also defines (but you shouldn't need to use directly) # SDL_SOUND_LIBRARY, the name of just the SDL_sound library you would link # against. Use SDL_SOUND_LIBRARIES for you link instructions and not this one. # And might define the following as needed # MIKMOD_LIBRARY # MODPLUG_LIBRARY # OGG_LIBRARY # VORBIS_LIBRARY # SMPEG_LIBRARY # FLAC_LIBRARY # SPEEX_LIBRARY # # Typically, you should not use these variables directly, and you should use # SDL_SOUND_LIBRARIES which contains SDL_SOUND_LIBRARY and the other audio libraries # (if needed) to successfully compile on your system. # # Created by Eric Wing. # This module is a bit more complicated than the other FindSDL* family modules. # The reason is that SDL_sound can be compiled in a large variety of different ways # which are independent of platform. SDL_sound may dynamically link against other 3rd # party libraries to get additional codec support, such as Ogg Vorbis, SMPEG, ModPlug, # MikMod, FLAC, Speex, and potentially others. # Under some circumstances which I don't fully understand, # there seems to be a requirement # that dependent libraries of libraries you use must also be explicitly # linked against in order to successfully compile. SDL_sound does not currently # have any system in place to know how it was compiled. # So this CMake module does the hard work in trying to discover which 3rd party # libraries are required for building (if any). # This module uses a brute force approach to create a test program that uses SDL_sound, # and then tries to build it. If the build fails, it parses the error output for # known symbol names to figure out which libraries are needed. # # Responds to the $SDLDIR and $SDLSOUNDDIR environmental variable that would # correspond to the ./configure --prefix=$SDLDIR used in building SDL. # # On OSX, this will prefer the Framework version (if found) over others. # People will have to manually change the cache values of # SDL_LIBRARY or SDL2_LIBRARY to override this selection or set the CMake # environment CMAKE_INCLUDE_PATH to modify the search paths. #============================================================================= # Copyright 2005-2009 Kitware, Inc. # Copyright 2012 Benjamin Eikel # # Distributed under the OSI-approved BSD License (the "License"); # see accompanying file Copyright.txt for details. # # This software is distributed WITHOUT ANY WARRANTY; without even the # implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. # See the License for more information. #============================================================================= # (To distribute this file outside of CMake, substitute the full # License text for the above reference.) set(SDL_SOUND_EXTRAS "" CACHE STRING "SDL_sound extra flags") mark_as_advanced(SDL_SOUND_EXTRAS) # Find SDL_sound.h find_path(SDL_SOUND_INCLUDE_DIR SDL_sound.h HINTS ENV SDLSOUNDDIR ENV SDLDIR PATH_SUFFIXES SDL SDL12 SDL11 ) find_library(SDL_SOUND_LIBRARY NAMES SDL_sound HINTS ENV SDLSOUNDDIR ENV SDLDIR ) if(SDL2_FOUND OR SDL_FOUND) if(SDL_SOUND_INCLUDE_DIR AND SDL_SOUND_LIBRARY) # CMake is giving me problems using TRY_COMPILE with the CMAKE_FLAGS # for the :STRING syntax if I have multiple values contained in a # single variable. This is a problem for the SDL2_LIBRARY variable # because it does just that. When I feed this variable to the command, # only the first value gets the appropriate modifier (e.g. -I) and # the rest get dropped. # To get multiple single variables to work, I must separate them with a "\;" # I could go back and modify the FindSDL2.cmake module, but that's kind of painful. # The solution would be to try something like: # set(SDL2_TRY_COMPILE_LIBRARY_LIST "${SDL2_TRY_COMPILE_LIBRARY_LIST}\;${CMAKE_THREAD_LIBS_INIT}") # Instead, it was suggested on the mailing list to write a temporary CMakeLists.txt # with a temporary test project and invoke that with TRY_COMPILE. # See message thread "Figuring out dependencies for a library in order to build" # 2005-07-16 # try_compile( # MY_RESULT # ${CMAKE_BINARY_DIR} # ${PROJECT_SOURCE_DIR}/DetermineSoundLibs.c # CMAKE_FLAGS # -DINCLUDE_DIRECTORIES:STRING=${SDL2_INCLUDE_DIR}\;${SDL_SOUND_INCLUDE_DIR} # -DLINK_LIBRARIES:STRING=${SDL_SOUND_LIBRARY}\;${SDL2_LIBRARY} # OUTPUT_VARIABLE MY_OUTPUT # ) # To minimize external dependencies, create a sdlsound test program # which will be used to figure out if additional link dependencies are # required for the link phase. file(WRITE ${PROJECT_BINARY_DIR}/CMakeTmp/DetermineSoundLibs.c "#include \"SDL_sound.h\" #include \"SDL.h\" int main(int argc, char* argv[]) { Sound_AudioInfo desired; Sound_Sample* sample; SDL_Init(0); Sound_Init(); /* This doesn't actually have to work, but Init() is a no-op * for some of the decoders, so this should force more symbols * to be pulled in. */ sample = Sound_NewSampleFromFile(argv[1], &desired, 4096); Sound_Quit(); SDL_Quit(); return 0; }" ) # Calling # target_link_libraries(DetermineSoundLibs "${SDL_SOUND_LIBRARY} ${SDL2_LIBRARY}) # causes problems when SDL2_LIBRARY looks like # /Library/Frameworks/SDL2.framework;-framework Cocoa # The ;-framework Cocoa seems to be confusing CMake once the OS X # framework support was added. I was told that breaking up the list # would fix the problem. set(TMP_LIBS "") if(SDL2_FOUND) set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARY} ${SDL2_LIBRARY}) foreach(lib ${SDL_SOUND_LIBRARY} ${SDL2_LIBRARY}) set(TMP_LIBS "${TMP_LIBS} \"${lib}\"") endforeach() set(TMP_INCLUDE_DIRS ${SDL2_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR}) else() set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARY} ${SDL_LIBRARY}) foreach(lib ${SDL_SOUND_LIBRARY} ${SDL_LIBRARY}) set(TMP_LIBS "${TMP_LIBS} \"${lib}\"") endforeach() set(TMP_INCLUDE_DIRS ${SDL_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR}) endif() # Keep trying to build a temp project until we find all missing libs. set(TRY_AGAIN TRUE) WHILE(TRY_AGAIN) set(TRY_AGAIN FALSE) # message("TMP_TRY_LIBS ${TMP_TRY_LIBS}") # Write the CMakeLists.txt and test project # Weird, this is still sketchy. If I don't quote the variables # in the TARGET_LINK_LIBRARIES, I seem to loose everything # in the SDL2_LIBRARY string after the "-framework". # But if I quote the stuff in INCLUDE_DIRECTORIES, it doesn't work. file(WRITE ${PROJECT_BINARY_DIR}/CMakeTmp/CMakeLists.txt "cmake_minimum_required(VERSION 2.8) project(DetermineSoundLibs C) include_directories(${TMP_INCLUDE_DIRS}) add_executable(DetermineSoundLibs DetermineSoundLibs.c) target_link_libraries(DetermineSoundLibs ${TMP_LIBS})" ) try_compile( MY_RESULT ${PROJECT_BINARY_DIR}/CMakeTmp ${PROJECT_BINARY_DIR}/CMakeTmp DetermineSoundLibs OUTPUT_VARIABLE MY_OUTPUT ) # message("${MY_RESULT}") # message(${MY_OUTPUT}) if(NOT MY_RESULT) # I expect that MPGLIB, VOC, WAV, AIFF, and SHN are compiled in statically. # I think Timidity is also compiled in statically. # I've never had to explcitly link against Quicktime, so I'll skip that for now. # Find libmath if("${MY_OUTPUT}" MATCHES "cos@@GLIBC") find_library(MATH_LIBRARY NAMES m) if(MATH_LIBRARY) set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${MATH_LIBRARY}) set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${MATH_LIBRARY}\"") set(TRY_AGAIN TRUE) endif(MATH_LIBRARY) endif("${MY_OUTPUT}" MATCHES "cos@@GLIBC") # Find MikMod if("${MY_OUTPUT}" MATCHES "MikMod_") find_library(MIKMOD_LIBRARY NAMES libmikmod-coreaudio mikmod PATHS ENV MIKMODDIR ENV SDLSOUNDDIR ENV SDLDIR /sw /opt/local /opt/csw /opt PATH_SUFFIXES lib ) if(MIKMOD_LIBRARY) set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${MIKMOD_LIBRARY}) set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${MIKMOD_LIBRARY}\"") set(TRY_AGAIN TRUE) endif(MIKMOD_LIBRARY) endif("${MY_OUTPUT}" MATCHES "MikMod_") # Find ModPlug if("${MY_OUTPUT}" MATCHES "MODPLUG_") find_library(MODPLUG_LIBRARY NAMES modplug PATHS ENV MODPLUGDIR ENV SDLSOUNDDIR ENV SDLDIR /sw /opt/local /opt/csw /opt PATH_SUFFIXES lib ) if(MODPLUG_LIBRARY) set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${MODPLUG_LIBRARY}) set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${MODPLUG_LIBRARY}\"") set(TRY_AGAIN TRUE) endif() endif() # Find Ogg and Vorbis if("${MY_OUTPUT}" MATCHES "ov_") find_library(VORBISFILE_LIBRARY NAMES vorbisfile VorbisFile VORBISFILE PATHS ENV VORBISDIR ENV OGGDIR ENV SDLSOUNDDIR ENV SDLDIR /sw /opt/local /opt/csw /opt PATH_SUFFIXES lib ) if(VORBISFILE_LIBRARY) set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${VORBISFILE_LIBRARY}) set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${VORBISFILE_LIBRARY}\"") set(TRY_AGAIN TRUE) endif() find_library(VORBIS_LIBRARY NAMES vorbis Vorbis VORBIS PATHS ENV OGGDIR ENV VORBISDIR ENV SDLSOUNDDIR ENV SDLDIR /sw /opt/local /opt/csw /opt PATH_SUFFIXES lib ) if(VORBIS_LIBRARY) set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${VORBIS_LIBRARY}) set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${VORBIS_LIBRARY}\"") set(TRY_AGAIN TRUE) endif() find_library(OGG_LIBRARY NAMES ogg Ogg OGG PATHS ENV OGGDIR ENV VORBISDIR ENV SDLSOUNDDIR ENV SDLDIR /sw /opt/local /opt/csw /opt PATH_SUFFIXES lib ) if(OGG_LIBRARY) set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${OGG_LIBRARY}) set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${OGG_LIBRARY}\"") set(TRY_AGAIN TRUE) endif() endif() # Find SMPEG if("${MY_OUTPUT}" MATCHES "SMPEG_") find_library(SMPEG_LIBRARY NAMES smpeg SMPEG Smpeg SMpeg PATHS ENV SMPEGDIR ENV SDLSOUNDDIR ENV SDLDIR /sw /opt/local /opt/csw /opt PATH_SUFFIXES lib ) if(SMPEG_LIBRARY) set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${SMPEG_LIBRARY}) set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${SMPEG_LIBRARY}\"") set(TRY_AGAIN TRUE) endif() endif() # Find FLAC if("${MY_OUTPUT}" MATCHES "FLAC_") find_library(FLAC_LIBRARY NAMES flac FLAC PATHS ENV FLACDIR ENV SDLSOUNDDIR ENV SDLDIR /sw /opt/local /opt/csw /opt PATH_SUFFIXES lib ) if(FLAC_LIBRARY) set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${FLAC_LIBRARY}) set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${FLAC_LIBRARY}\"") set(TRY_AGAIN TRUE) endif() endif() # Hmmm...Speex seems to depend on Ogg. This might be a problem if # the TRY_COMPILE attempt gets blocked at SPEEX before it can pull # in the Ogg symbols. I'm not sure if I should duplicate the ogg stuff # above for here or if two ogg entries will screw up things. if("${MY_OUTPUT}" MATCHES "speex_") find_library(SPEEX_LIBRARY NAMES speex SPEEX PATHS ENV SPEEXDIR ENV SDLSOUNDDIR ENV SDLDIR /sw /opt/local /opt/csw /opt PATH_SUFFIXES lib ) if(SPEEX_LIBRARY) set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${SPEEX_LIBRARY}) set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${SPEEX_LIBRARY}\"") set(TRY_AGAIN TRUE) endif() # Find OGG (needed for Speex) # We might have already found Ogg for Vorbis, so skip it if so. if(NOT OGG_LIBRARY) find_library(OGG_LIBRARY NAMES ogg Ogg OGG PATHS ENV OGGDIR ENV VORBISDIR ENV SPEEXDIR ENV SDLSOUNDDIR ENV SDLDIR /sw /opt/local /opt/csw /opt PATH_SUFFIXES lib ) if(OGG_LIBRARY) set(SDL_SOUND_LIBRARIES_TMP ${SDL_SOUND_LIBRARIES_TMP} ${OGG_LIBRARY}) set(TMP_LIBS "${SDL_SOUND_LIBRARIES_TMP} \"${OGG_LIBRARY}\"") set(TRY_AGAIN TRUE) endif() endif() endif() endif() ENDWHILE() unset(TMP_INCLUDE_DIRS) unset(TMP_LIBS) set(SDL_SOUND_LIBRARIES ${SDL_SOUND_EXTRAS} ${SDL_SOUND_LIBRARIES_TMP} CACHE INTERNAL "SDL_sound and dependent libraries") endif() endif() if(SDL_SOUND_INCLUDE_DIR AND EXISTS "${SDL_SOUND_INCLUDE_DIR}/SDL_sound.h") file(STRINGS "${SDL_SOUND_INCLUDE_DIR}/SDL_sound.h" SDL_SOUND_VERSION_MAJOR_LINE REGEX "^#define[ \t]+SOUND_VER_MAJOR[ \t]+[0-9]+$") file(STRINGS "${SDL_SOUND_INCLUDE_DIR}/SDL_sound.h" SDL_SOUND_VERSION_MINOR_LINE REGEX "^#define[ \t]+SOUND_VER_MINOR[ \t]+[0-9]+$") file(STRINGS "${SDL_SOUND_INCLUDE_DIR}/SDL_sound.h" SDL_SOUND_VERSION_PATCH_LINE REGEX "^#define[ \t]+SOUND_VER_PATCH[ \t]+[0-9]+$") string(REGEX REPLACE "^#define[ \t]+SOUND_VER_MAJOR[ \t]+([0-9]+)$" "\\1" SDL_SOUND_VERSION_MAJOR "${SDL_SOUND_VERSION_MAJOR_LINE}") string(REGEX REPLACE "^#define[ \t]+SOUND_VER_MINOR[ \t]+([0-9]+)$" "\\1" SDL_SOUND_VERSION_MINOR "${SDL_SOUND_VERSION_MINOR_LINE}") string(REGEX REPLACE "^#define[ \t]+SOUND_VER_PATCH[ \t]+([0-9]+)$" "\\1" SDL_SOUND_VERSION_PATCH "${SDL_SOUND_VERSION_PATCH_LINE}") set(SDL_SOUND_VERSION_STRING ${SDL_SOUND_VERSION_MAJOR}.${SDL_SOUND_VERSION_MINOR}.${SDL_SOUND_VERSION_PATCH}) unset(SDL_SOUND_VERSION_MAJOR_LINE) unset(SDL_SOUND_VERSION_MINOR_LINE) unset(SDL_SOUND_VERSION_PATCH_LINE) unset(SDL_SOUND_VERSION_MAJOR) unset(SDL_SOUND_VERSION_MINOR) unset(SDL_SOUND_VERSION_PATCH) endif() include(FindPackageHandleStandardArgs) FIND_PACKAGE_HANDLE_STANDARD_ARGS(SDL_sound REQUIRED_VARS SDL_SOUND_LIBRARIES SDL_SOUND_INCLUDE_DIR VERSION_VAR SDL_SOUND_VERSION_STRING) openal-soft-openal-soft-1.19.1/cmake/FindSoundIO.cmake000066400000000000000000000016121335774445300225010ustar00rootroot00000000000000# - Find SoundIO (sndio) includes and libraries # # SOUNDIO_FOUND - True if SOUNDIO_INCLUDE_DIR & SOUNDIO_LIBRARY are # found # SOUNDIO_LIBRARIES - Set when SOUNDIO_LIBRARY is found # SOUNDIO_INCLUDE_DIRS - Set when SOUNDIO_INCLUDE_DIR is found # # SOUNDIO_INCLUDE_DIR - where to find sndio.h, etc. # SOUNDIO_LIBRARY - the sndio library # find_path(SOUNDIO_INCLUDE_DIR NAMES sndio.h DOC "The SoundIO include directory" ) find_library(SOUNDIO_LIBRARY NAMES sndio DOC "The SoundIO library" ) include(FindPackageHandleStandardArgs) find_package_handle_standard_args(SoundIO REQUIRED_VARS SOUNDIO_LIBRARY SOUNDIO_INCLUDE_DIR ) if(SOUNDIO_FOUND) set(SOUNDIO_LIBRARIES ${SOUNDIO_LIBRARY}) set(SOUNDIO_INCLUDE_DIRS ${SOUNDIO_INCLUDE_DIR}) endif() mark_as_advanced(SOUNDIO_INCLUDE_DIR SOUNDIO_LIBRARY) openal-soft-openal-soft-1.19.1/cmake/FindWindowsSDK.cmake000066400000000000000000000567721335774445300231760ustar00rootroot00000000000000# - Find the Windows SDK aka Platform SDK # # Relevant Wikipedia article: http://en.wikipedia.org/wiki/Microsoft_Windows_SDK # # Pass "COMPONENTS tools" to ignore Visual Studio version checks: in case # you just want the tool binaries to run, rather than the libraries and headers # for compiling. # # Variables: # WINDOWSSDK_FOUND - if any version of the windows or platform SDK was found that is usable with the current version of visual studio # WINDOWSSDK_LATEST_DIR # WINDOWSSDK_LATEST_NAME # WINDOWSSDK_FOUND_PREFERENCE - if we found an entry indicating a "preferred" SDK listed for this visual studio version # WINDOWSSDK_PREFERRED_DIR # WINDOWSSDK_PREFERRED_NAME # # WINDOWSSDK_DIRS - contains no duplicates, ordered most recent first. # WINDOWSSDK_PREFERRED_FIRST_DIRS - contains no duplicates, ordered with preferred first, followed by the rest in descending recency # # Functions: # windowssdk_name_lookup( ) - Find the name corresponding with the SDK directory you pass in, or # NOTFOUND if not recognized. Your directory must be one of WINDOWSSDK_DIRS for this to work. # # windowssdk_build_lookup( ) - Find the build version number corresponding with the SDK directory you pass in, or # NOTFOUND if not recognized. Your directory must be one of WINDOWSSDK_DIRS for this to work. # # get_windowssdk_from_component( ) - Given a library or include dir, # find the Windows SDK root dir corresponding to it, or NOTFOUND if unrecognized. # # get_windowssdk_library_dirs( ) - Find the architecture-appropriate # library directories corresponding to the SDK directory you pass in (or NOTFOUND if none) # # get_windowssdk_library_dirs_multiple( ...) - Find the architecture-appropriate # library directories corresponding to the SDK directories you pass in, in order, skipping those not found. NOTFOUND if none at all. # Good for passing WINDOWSSDK_DIRS or WINDOWSSDK_DIRS to if you really just want a file and don't care where from. # # get_windowssdk_include_dirs( ) - Find the # include directories corresponding to the SDK directory you pass in (or NOTFOUND if none) # # get_windowssdk_include_dirs_multiple( ...) - Find the # include directories corresponding to the SDK directories you pass in, in order, skipping those not found. NOTFOUND if none at all. # Good for passing WINDOWSSDK_DIRS or WINDOWSSDK_DIRS to if you really just want a file and don't care where from. # # Requires these CMake modules: # FindPackageHandleStandardArgs (known included with CMake >=2.6.2) # # Original Author: # 2012 Ryan Pavlik # http://academic.cleardefinition.com # Iowa State University HCI Graduate Program/VRAC # # Copyright Iowa State University 2012. # Distributed under the Boost Software License, Version 1.0. # (See accompanying file LICENSE_1_0.txt or copy at # http://www.boost.org/LICENSE_1_0.txt) set(_preferred_sdk_dirs) # pre-output set(_win_sdk_dirs) # pre-output set(_win_sdk_versanddirs) # pre-output set(_win_sdk_buildsanddirs) # pre-output set(_winsdk_vistaonly) # search parameters set(_winsdk_kits) # search parameters set(_WINDOWSSDK_ANNOUNCE OFF) if(NOT WINDOWSSDK_FOUND AND (NOT WindowsSDK_FIND_QUIETLY)) set(_WINDOWSSDK_ANNOUNCE ON) endif() macro(_winsdk_announce) if(_WINSDK_ANNOUNCE) message(STATUS ${ARGN}) endif() endmacro() set(_winsdk_win10vers 10.0.14393.0 # Redstone aka Win10 1607 "Anniversary Update" 10.0.10586.0 # TH2 aka Win10 1511 10.0.10240.0 # Win10 RTM 10.0.10150.0 # just ucrt 10.0.10056.0 ) if(WindowsSDK_FIND_COMPONENTS MATCHES "tools") set(_WINDOWSSDK_IGNOREMSVC ON) _winsdk_announce("Checking for tools from Windows/Platform SDKs...") else() set(_WINDOWSSDK_IGNOREMSVC OFF) _winsdk_announce("Checking for Windows/Platform SDKs...") endif() # Appends to the three main pre-output lists used only if the path exists # and is not already in the list. function(_winsdk_conditional_append _vername _build _path) if(("${_path}" MATCHES "registry") OR (NOT EXISTS "${_path}")) # Path invalid - do not add return() endif() list(FIND _win_sdk_dirs "${_path}" _win_sdk_idx) if(_win_sdk_idx GREATER -1) # Path already in list - do not add return() endif() _winsdk_announce( " - ${_vername}, Build ${_build} @ ${_path}") # Not yet in the list, so we'll add it list(APPEND _win_sdk_dirs "${_path}") set(_win_sdk_dirs "${_win_sdk_dirs}" CACHE INTERNAL "" FORCE) list(APPEND _win_sdk_versanddirs "${_vername}" "${_path}") set(_win_sdk_versanddirs "${_win_sdk_versanddirs}" CACHE INTERNAL "" FORCE) list(APPEND _win_sdk_buildsanddirs "${_build}" "${_path}") set(_win_sdk_buildsanddirs "${_win_sdk_buildsanddirs}" CACHE INTERNAL "" FORCE) endfunction() # Appends to the "preferred SDK" lists only if the path exists function(_winsdk_conditional_append_preferred _info _path) if(("${_path}" MATCHES "registry") OR (NOT EXISTS "${_path}")) # Path invalid - do not add return() endif() get_filename_component(_path "${_path}" ABSOLUTE) list(FIND _win_sdk_preferred_sdk_dirs "${_path}" _win_sdk_idx) if(_win_sdk_idx GREATER -1) # Path already in list - do not add return() endif() _winsdk_announce( " - Found \"preferred\" SDK ${_info} @ ${_path}") # Not yet in the list, so we'll add it list(APPEND _win_sdk_preferred_sdk_dirs "${_path}") set(_win_sdk_preferred_sdk_dirs "${_win_sdk_dirs}" CACHE INTERNAL "" FORCE) # Just in case we somehow missed it: _winsdk_conditional_append("${_info}" "" "${_path}") endfunction() # Given a version like v7.0A, looks for an SDK in the registry under "Microsoft SDKs". # If the given version might be in both HKEY_LOCAL_MACHINE\\SOFTWARE\\Microsoft\\Microsoft SDKs\\Windows # and HKEY_LOCAL_MACHINE\\SOFTWARE\\Microsoft\\Windows Kits\\Installed Roots aka "Windows Kits", # use this macro first, since these registry keys usually have more information. # # Pass a "default" build number as an extra argument in case we can't find it. function(_winsdk_check_microsoft_sdks_registry _winsdkver) set(SDKKEY "HKEY_LOCAL_MACHINE\\SOFTWARE\\Microsoft\\Microsoft SDKs\\Windows\\${_winsdkver}") get_filename_component(_sdkdir "[${SDKKEY};InstallationFolder]" ABSOLUTE) set(_sdkname "Windows SDK ${_winsdkver}") # Default build number passed as extra argument set(_build ${ARGN}) # See if the registry holds a Microsoft-mutilated, err, designated, product name # (just using get_filename_component to execute the registry lookup) get_filename_component(_sdkproductname "[${SDKKEY};ProductName]" NAME) if(NOT "${_sdkproductname}" MATCHES "registry") # Got a product name set(_sdkname "${_sdkname} (${_sdkproductname})") endif() # try for a version to augment our name # (just using get_filename_component to execute the registry lookup) get_filename_component(_sdkver "[${SDKKEY};ProductVersion]" NAME) if(NOT "${_sdkver}" MATCHES "registry" AND NOT MATCHES) # Got a version if(NOT "${_sdkver}" MATCHES "\\.\\.") # and it's not an invalid one with two dots in it: # use to override the default build set(_build ${_sdkver}) if(NOT "${_sdkname}" MATCHES "${_sdkver}") # Got a version that's not already in the name, let's use it to improve our name. set(_sdkname "${_sdkname} (${_sdkver})") endif() endif() endif() _winsdk_conditional_append("${_sdkname}" "${_build}" "${_sdkdir}") endfunction() # Given a name for identification purposes, the build number, and a key (technically a "value name") # corresponding to a Windows SDK packaged as a "Windows Kit", look for it # in HKEY_LOCAL_MACHINE\\SOFTWARE\\Microsoft\\Windows Kits\\Installed Roots # Note that the key or "value name" tends to be something weird like KitsRoot81 - # no easy way to predict, just have to observe them in the wild. # Doesn't hurt to also try _winsdk_check_microsoft_sdks_registry for these: # sometimes you get keys in both parts of the registry (in the wow64 portion especially), # and the non-"Windows Kits" location is often more descriptive. function(_winsdk_check_windows_kits_registry _winkit_name _winkit_build _winkit_key) get_filename_component(_sdkdir "[HKEY_LOCAL_MACHINE\\SOFTWARE\\Microsoft\\Windows Kits\\Installed Roots;${_winkit_key}]" ABSOLUTE) _winsdk_conditional_append("${_winkit_name}" "${_winkit_build}" "${_sdkdir}") endfunction() # Given a name for identification purposes and the build number # corresponding to a Windows 10 SDK packaged as a "Windows Kit", look for it # in HKEY_LOCAL_MACHINE\\SOFTWARE\\Microsoft\\Windows Kits\\Installed Roots # Doesn't hurt to also try _winsdk_check_microsoft_sdks_registry for these: # sometimes you get keys in both parts of the registry (in the wow64 portion especially), # and the non-"Windows Kits" location is often more descriptive. function(_winsdk_check_win10_kits _winkit_build) get_filename_component(_sdkdir "[HKEY_LOCAL_MACHINE\\SOFTWARE\\Microsoft\\Windows Kits\\Installed Roots;KitsRoot10]" ABSOLUTE) if(("${_sdkdir}" MATCHES "registry") OR (NOT EXISTS "${_sdkdir}")) return() # not found endif() if(EXISTS "${_sdkdir}/Include/${_winkit_build}/um") _winsdk_conditional_append("Windows Kits 10 (Build ${_winkit_build})" "${_winkit_build}" "${_sdkdir}") endif() endfunction() # Given a name for indentification purposes, the build number, and the associated package GUID, # look in the registry under both HKLM and HKCU in \\SOFTWARE\\Microsoft\\MicrosoftSDK\\InstalledSDKs\\ # for that guid and the SDK it points to. function(_winsdk_check_platformsdk_registry _platformsdkname _build _platformsdkguid) foreach(_winsdk_hive HKEY_LOCAL_MACHINE HKEY_CURRENT_USER) get_filename_component(_sdkdir "[${_winsdk_hive}\\SOFTWARE\\Microsoft\\MicrosoftSDK\\InstalledSDKs\\${_platformsdkguid};Install Dir]" ABSOLUTE) _winsdk_conditional_append("${_platformsdkname} (${_build})" "${_build}" "${_sdkdir}") endforeach() endfunction() ### # Detect toolchain information: to know whether it's OK to use Vista+ only SDKs ### set(_winsdk_vistaonly_ok OFF) if(MSVC AND NOT _WINDOWSSDK_IGNOREMSVC) # VC 10 and older has broad target support if(MSVC_VERSION LESS 1700) # VC 11 by default targets Vista and later only, so we can add a few more SDKs that (might?) only work on vista+ elseif("${CMAKE_VS_PLATFORM_TOOLSET}" MATCHES "_xp") # This is the XP-compatible v110+ toolset elseif("${CMAKE_VS_PLATFORM_TOOLSET}" STREQUAL "v100" OR "${CMAKE_VS_PLATFORM_TOOLSET}" STREQUAL "v90") # This is the VS2010/VS2008 toolset else() # OK, we're VC11 or newer and not using a backlevel or XP-compatible toolset. # These versions have no XP (and possibly Vista pre-SP1) support set(_winsdk_vistaonly_ok ON) if(_WINDOWSSDK_ANNOUNCE AND NOT _WINDOWSSDK_VISTAONLY_PESTERED) set(_WINDOWSSDK_VISTAONLY_PESTERED ON CACHE INTERNAL "" FORCE) message(STATUS "FindWindowsSDK: Detected Visual Studio 2012 or newer, not using the _xp toolset variant: including SDK versions that drop XP support in search!") endif() endif() endif() if(_WINDOWSSDK_IGNOREMSVC) set(_winsdk_vistaonly_ok ON) endif() ### # MSVC version checks - keeps messy conditionals in one place # (messy because of _WINDOWSSDK_IGNOREMSVC) ### set(_winsdk_msvc_greater_1200 OFF) if(_WINDOWSSDK_IGNOREMSVC OR (MSVC AND (MSVC_VERSION GREATER 1200))) set(_winsdk_msvc_greater_1200 ON) endif() # Newer than VS .NET/VS Toolkit 2003 set(_winsdk_msvc_greater_1310 OFF) if(_WINDOWSSDK_IGNOREMSVC OR (MSVC AND (MSVC_VERSION GREATER 1310))) set(_winsdk_msvc_greater_1310 ON) endif() # VS2005/2008 set(_winsdk_msvc_less_1600 OFF) if(_WINDOWSSDK_IGNOREMSVC OR (MSVC AND (MSVC_VERSION LESS 1600))) set(_winsdk_msvc_less_1600 ON) endif() # VS2013+ set(_winsdk_msvc_not_less_1800 OFF) if(_WINDOWSSDK_IGNOREMSVC OR (MSVC AND (NOT MSVC_VERSION LESS 1800))) set(_winsdk_msvc_not_less_1800 ON) endif() ### # START body of find module ### if(_winsdk_msvc_greater_1310) # Newer than VS .NET/VS Toolkit 2003 ### # Look for "preferred" SDKs ### # Environment variable for SDK dir if(EXISTS "$ENV{WindowsSDKDir}" AND (NOT "$ENV{WindowsSDKDir}" STREQUAL "")) _winsdk_conditional_append_preferred("WindowsSDKDir environment variable" "$ENV{WindowsSDKDir}") endif() if(_winsdk_msvc_less_1600) # Per-user current Windows SDK for VS2005/2008 get_filename_component(_sdkdir "[HKEY_CURRENT_USER\\Software\\Microsoft\\Microsoft SDKs\\Windows;CurrentInstallFolder]" ABSOLUTE) _winsdk_conditional_append_preferred("Per-user current Windows SDK" "${_sdkdir}") # System-wide current Windows SDK for VS2005/2008 get_filename_component(_sdkdir "[HKEY_LOCAL_MACHINE\\SOFTWARE\\Microsoft\\Microsoft SDKs\\Windows;CurrentInstallFolder]" ABSOLUTE) _winsdk_conditional_append_preferred("System-wide current Windows SDK" "${_sdkdir}") endif() ### # Begin the massive list of SDK searching! ### if(_winsdk_vistaonly_ok AND _winsdk_msvc_not_less_1800) # These require at least Visual Studio 2013 (VC12) _winsdk_check_microsoft_sdks_registry(v10.0A) # Windows Software Development Kit (SDK) for Windows 10 # Several different versions living in the same directory - if nothing else we can assume RTM (10240) _winsdk_check_microsoft_sdks_registry(v10.0 10.0.10240.0) foreach(_win10build ${_winsdk_win10vers}) _winsdk_check_win10_kits(${_win10build}) endforeach() endif() # vista-only and 2013+ # Included in Visual Studio 2013 # Includes the v120_xp toolset _winsdk_check_microsoft_sdks_registry(v8.1A 8.1.51636) if(_winsdk_vistaonly_ok AND _winsdk_msvc_not_less_1800) # Windows Software Development Kit (SDK) for Windows 8.1 # http://msdn.microsoft.com/en-gb/windows/desktop/bg162891 _winsdk_check_microsoft_sdks_registry(v8.1 8.1.25984.0) _winsdk_check_windows_kits_registry("Windows Kits 8.1" 8.1.25984.0 KitsRoot81) endif() # vista-only and 2013+ if(_winsdk_vistaonly_ok) # Included in Visual Studio 2012 _winsdk_check_microsoft_sdks_registry(v8.0A 8.0.50727) # Microsoft Windows SDK for Windows 8 and .NET Framework 4.5 # This is the first version to also include the DirectX SDK # http://msdn.microsoft.com/en-US/windows/desktop/hh852363.aspx _winsdk_check_microsoft_sdks_registry(v8.0 6.2.9200.16384) _winsdk_check_windows_kits_registry("Windows Kits 8.0" 6.2.9200.16384 KitsRoot) endif() # vista-only # Included with VS 2012 Update 1 or later # Introduces v110_xp toolset _winsdk_check_microsoft_sdks_registry(v7.1A 7.1.51106) if(_winsdk_vistaonly_ok) # Microsoft Windows SDK for Windows 7 and .NET Framework 4 # http://www.microsoft.com/downloads/en/details.aspx?FamilyID=6b6c21d2-2006-4afa-9702-529fa782d63b _winsdk_check_microsoft_sdks_registry(v7.1 7.1.7600.0.30514) endif() # vista-only # Included with VS 2010 _winsdk_check_microsoft_sdks_registry(v7.0A 6.1.7600.16385) # Windows SDK for Windows 7 and .NET Framework 3.5 SP1 # Works with VC9 # http://www.microsoft.com/en-us/download/details.aspx?id=18950 _winsdk_check_microsoft_sdks_registry(v7.0 6.1.7600.16385) # Two versions call themselves "v6.1": # Older: # Windows Vista Update & .NET 3.0 SDK # http://www.microsoft.com/en-us/download/details.aspx?id=14477 # Newer: # Windows Server 2008 & .NET 3.5 SDK # may have broken VS9SP1? they recommend v7.0 instead, or a KB... # http://www.microsoft.com/en-us/download/details.aspx?id=24826 _winsdk_check_microsoft_sdks_registry(v6.1 6.1.6000.16384.10) # Included in VS 2008 _winsdk_check_microsoft_sdks_registry(v6.0A 6.1.6723.1) # Microsoft Windows Software Development Kit for Windows Vista and .NET Framework 3.0 Runtime Components # http://blogs.msdn.com/b/stanley/archive/2006/11/08/microsoft-windows-software-development-kit-for-windows-vista-and-net-framework-3-0-runtime-components.aspx _winsdk_check_microsoft_sdks_registry(v6.0 6.0.6000.16384) endif() # Let's not forget the Platform SDKs, which sometimes are useful! if(_winsdk_msvc_greater_1200) _winsdk_check_platformsdk_registry("Microsoft Platform SDK for Windows Server 2003 R2" "5.2.3790.2075.51" "D2FF9F89-8AA2-4373-8A31-C838BF4DBBE1") _winsdk_check_platformsdk_registry("Microsoft Platform SDK for Windows Server 2003 SP1" "5.2.3790.1830.15" "8F9E5EF3-A9A5-491B-A889-C58EFFECE8B3") endif() ### # Finally, look for "preferred" SDKs ### if(_winsdk_msvc_greater_1310) # Newer than VS .NET/VS Toolkit 2003 # Environment variable for SDK dir if(EXISTS "$ENV{WindowsSDKDir}" AND (NOT "$ENV{WindowsSDKDir}" STREQUAL "")) _winsdk_conditional_append_preferred("WindowsSDKDir environment variable" "$ENV{WindowsSDKDir}") endif() if(_winsdk_msvc_less_1600) # Per-user current Windows SDK for VS2005/2008 get_filename_component(_sdkdir "[HKEY_CURRENT_USER\\Software\\Microsoft\\Microsoft SDKs\\Windows;CurrentInstallFolder]" ABSOLUTE) _winsdk_conditional_append_preferred("Per-user current Windows SDK" "${_sdkdir}") # System-wide current Windows SDK for VS2005/2008 get_filename_component(_sdkdir "[HKEY_LOCAL_MACHINE\\SOFTWARE\\Microsoft\\Microsoft SDKs\\Windows;CurrentInstallFolder]" ABSOLUTE) _winsdk_conditional_append_preferred("System-wide current Windows SDK" "${_sdkdir}") endif() endif() function(windowssdk_name_lookup _dir _outvar) list(FIND _win_sdk_versanddirs "${_dir}" _diridx) math(EXPR _idx "${_diridx} - 1") if(${_idx} GREATER -1) list(GET _win_sdk_versanddirs ${_idx} _ret) else() set(_ret "NOTFOUND") endif() set(${_outvar} "${_ret}" PARENT_SCOPE) endfunction() function(windowssdk_build_lookup _dir _outvar) list(FIND _win_sdk_buildsanddirs "${_dir}" _diridx) math(EXPR _idx "${_diridx} - 1") if(${_idx} GREATER -1) list(GET _win_sdk_buildsanddirs ${_idx} _ret) else() set(_ret "NOTFOUND") endif() set(${_outvar} "${_ret}" PARENT_SCOPE) endfunction() # If we found something... if(_win_sdk_dirs) list(GET _win_sdk_dirs 0 WINDOWSSDK_LATEST_DIR) windowssdk_name_lookup("${WINDOWSSDK_LATEST_DIR}" WINDOWSSDK_LATEST_NAME) set(WINDOWSSDK_DIRS ${_win_sdk_dirs}) # Fallback, in case no preference found. set(WINDOWSSDK_PREFERRED_DIR "${WINDOWSSDK_LATEST_DIR}") set(WINDOWSSDK_PREFERRED_NAME "${WINDOWSSDK_LATEST_NAME}") set(WINDOWSSDK_PREFERRED_FIRST_DIRS ${WINDOWSSDK_DIRS}) set(WINDOWSSDK_FOUND_PREFERENCE OFF) endif() # If we found indications of a user preference... if(_win_sdk_preferred_sdk_dirs) list(GET _win_sdk_preferred_sdk_dirs 0 WINDOWSSDK_PREFERRED_DIR) windowssdk_name_lookup("${WINDOWSSDK_PREFERRED_DIR}" WINDOWSSDK_PREFERRED_NAME) set(WINDOWSSDK_PREFERRED_FIRST_DIRS ${_win_sdk_preferred_sdk_dirs} ${_win_sdk_dirs}) list(REMOVE_DUPLICATES WINDOWSSDK_PREFERRED_FIRST_DIRS) set(WINDOWSSDK_FOUND_PREFERENCE ON) endif() include(FindPackageHandleStandardArgs) find_package_handle_standard_args(WindowsSDK "No compatible version of the Windows SDK or Platform SDK found." WINDOWSSDK_DIRS) if(WINDOWSSDK_FOUND) # Internal: Architecture-appropriate library directory names. if("${CMAKE_VS_PLATFORM_NAME}" STREQUAL "ARM") if(CMAKE_SIZEOF_VOID_P MATCHES "8") # Only supported in Win10 SDK and up. set(_winsdk_arch8 arm64) # what the WDK for Win8+ calls this architecture else() set(_winsdk_archbare /arm) # what the architecture used to be called in oldest SDKs set(_winsdk_arch arm) # what the architecture used to be called set(_winsdk_arch8 arm) # what the WDK for Win8+ calls this architecture endif() else() if(CMAKE_SIZEOF_VOID_P MATCHES "8") set(_winsdk_archbare /x64) # what the architecture used to be called in oldest SDKs set(_winsdk_arch amd64) # what the architecture used to be called set(_winsdk_arch8 x64) # what the WDK for Win8+ calls this architecture else() set(_winsdk_archbare ) # what the architecture used to be called in oldest SDKs set(_winsdk_arch i386) # what the architecture used to be called set(_winsdk_arch8 x86) # what the WDK for Win8+ calls this architecture endif() endif() function(get_windowssdk_from_component _component _var) get_filename_component(_component "${_component}" ABSOLUTE) file(TO_CMAKE_PATH "${_component}" _component) foreach(_sdkdir ${WINDOWSSDK_DIRS}) get_filename_component(_sdkdir "${_sdkdir}" ABSOLUTE) string(LENGTH "${_sdkdir}" _sdklen) file(RELATIVE_PATH _rel "${_sdkdir}" "${_component}") # If we don't have any "parent directory" items... if(NOT "${_rel}" MATCHES "[.][.]") set(${_var} "${_sdkdir}" PARENT_SCOPE) return() endif() endforeach() # Fail. set(${_var} "NOTFOUND" PARENT_SCOPE) endfunction() function(get_windowssdk_library_dirs _winsdk_dir _var) set(_dirs) set(_suffixes "lib${_winsdk_archbare}" # SDKs like 7.1A "lib/${_winsdk_arch}" # just because some SDKs have x86 dir and root dir "lib/w2k/${_winsdk_arch}" # Win2k min requirement "lib/wxp/${_winsdk_arch}" # WinXP min requirement "lib/wnet/${_winsdk_arch}" # Win Server 2003 min requirement "lib/wlh/${_winsdk_arch}" "lib/wlh/um/${_winsdk_arch8}" # Win Vista ("Long Horn") min requirement "lib/win7/${_winsdk_arch}" "lib/win7/um/${_winsdk_arch8}" # Win 7 min requirement ) foreach(_ver wlh # Win Vista ("Long Horn") min requirement win7 # Win 7 min requirement win8 # Win 8 min requirement winv6.3 # Win 8.1 min requirement ) list(APPEND _suffixes "lib/${_ver}/${_winsdk_arch}" "lib/${_ver}/um/${_winsdk_arch8}" "lib/${_ver}/km/${_winsdk_arch8}" ) endforeach() # Look for WDF libraries in Win10+ SDK foreach(_mode umdf kmdf) file(GLOB _wdfdirs RELATIVE "${_winsdk_dir}" "${_winsdk_dir}/lib/wdf/${_mode}/${_winsdk_arch8}/*") if(_wdfdirs) list(APPEND _suffixes ${_wdfdirs}) endif() endforeach() # Look in each Win10+ SDK version for the components foreach(_win10ver ${_winsdk_win10vers}) foreach(_component um km ucrt mmos) list(APPEND _suffixes "lib/${_win10ver}/${_component}/${_winsdk_arch8}") endforeach() endforeach() foreach(_suffix ${_suffixes}) # Check to see if a library actually exists here. file(GLOB _libs "${_winsdk_dir}/${_suffix}/*.lib") if(_libs) list(APPEND _dirs "${_winsdk_dir}/${_suffix}") endif() endforeach() if("${_dirs}" STREQUAL "") set(_dirs NOTFOUND) else() list(REMOVE_DUPLICATES _dirs) endif() set(${_var} ${_dirs} PARENT_SCOPE) endfunction() function(get_windowssdk_include_dirs _winsdk_dir _var) set(_dirs) set(_subdirs shared um winrt km wdf mmos ucrt) set(_suffixes Include) foreach(_dir ${_subdirs}) list(APPEND _suffixes "Include/${_dir}") endforeach() foreach(_ver ${_winsdk_win10vers}) foreach(_dir ${_subdirs}) list(APPEND _suffixes "Include/${_ver}/${_dir}") endforeach() endforeach() foreach(_suffix ${_suffixes}) # Check to see if a header file actually exists here. file(GLOB _headers "${_winsdk_dir}/${_suffix}/*.h") if(_headers) list(APPEND _dirs "${_winsdk_dir}/${_suffix}") endif() endforeach() if("${_dirs}" STREQUAL "") set(_dirs NOTFOUND) else() list(REMOVE_DUPLICATES _dirs) endif() set(${_var} ${_dirs} PARENT_SCOPE) endfunction() function(get_windowssdk_library_dirs_multiple _var) set(_dirs) foreach(_sdkdir ${ARGN}) get_windowssdk_library_dirs("${_sdkdir}" _current_sdk_libdirs) if(_current_sdk_libdirs) list(APPEND _dirs ${_current_sdk_libdirs}) endif() endforeach() if("${_dirs}" STREQUAL "") set(_dirs NOTFOUND) else() list(REMOVE_DUPLICATES _dirs) endif() set(${_var} ${_dirs} PARENT_SCOPE) endfunction() function(get_windowssdk_include_dirs_multiple _var) set(_dirs) foreach(_sdkdir ${ARGN}) get_windowssdk_include_dirs("${_sdkdir}" _current_sdk_incdirs) if(_current_sdk_libdirs) list(APPEND _dirs ${_current_sdk_incdirs}) endif() endforeach() if("${_dirs}" STREQUAL "") set(_dirs NOTFOUND) else() list(REMOVE_DUPLICATES _dirs) endif() set(${_var} ${_dirs} PARENT_SCOPE) endfunction() endif() openal-soft-openal-soft-1.19.1/common/000077500000000000000000000000001335774445300175665ustar00rootroot00000000000000openal-soft-openal-soft-1.19.1/common/alcomplex.c000066400000000000000000000042131335774445300217160ustar00rootroot00000000000000 #include "config.h" #include "alcomplex.h" #include "math_defs.h" extern inline ALcomplex complex_add(ALcomplex a, ALcomplex b); extern inline ALcomplex complex_sub(ALcomplex a, ALcomplex b); extern inline ALcomplex complex_mult(ALcomplex a, ALcomplex b); void complex_fft(ALcomplex *FFTBuffer, ALsizei FFTSize, ALdouble Sign) { ALsizei i, j, k, mask, step, step2; ALcomplex temp, u, w; ALdouble arg; /* Bit-reversal permutation applied to a sequence of FFTSize items */ for(i = 1;i < FFTSize-1;i++) { for(mask = 0x1, j = 0;mask < FFTSize;mask <<= 1) { if((i&mask) != 0) j++; j <<= 1; } j >>= 1; if(i < j) { temp = FFTBuffer[i]; FFTBuffer[i] = FFTBuffer[j]; FFTBuffer[j] = temp; } } /* Iterative form of Danielson–Lanczos lemma */ for(i = 1, step = 2;i < FFTSize;i<<=1, step<<=1) { step2 = step >> 1; arg = M_PI / step2; w.Real = cos(arg); w.Imag = sin(arg) * Sign; u.Real = 1.0; u.Imag = 0.0; for(j = 0;j < step2;j++) { for(k = j;k < FFTSize;k+=step) { temp = complex_mult(FFTBuffer[k+step2], u); FFTBuffer[k+step2] = complex_sub(FFTBuffer[k], temp); FFTBuffer[k] = complex_add(FFTBuffer[k], temp); } u = complex_mult(u, w); } } } void complex_hilbert(ALcomplex *Buffer, ALsizei size) { const ALdouble inverse_size = 1.0/(ALdouble)size; ALsizei todo, i; for(i = 0;i < size;i++) Buffer[i].Imag = 0.0; complex_fft(Buffer, size, 1.0); todo = size >> 1; Buffer[0].Real *= inverse_size; Buffer[0].Imag *= inverse_size; for(i = 1;i < todo;i++) { Buffer[i].Real *= 2.0*inverse_size; Buffer[i].Imag *= 2.0*inverse_size; } Buffer[i].Real *= inverse_size; Buffer[i].Imag *= inverse_size; i++; for(;i < size;i++) { Buffer[i].Real = 0.0; Buffer[i].Imag = 0.0; } complex_fft(Buffer, size, -1.0); } openal-soft-openal-soft-1.19.1/common/alcomplex.h000066400000000000000000000032731335774445300217300ustar00rootroot00000000000000#ifndef ALCOMPLEX_H #define ALCOMPLEX_H #include "AL/al.h" #ifdef __cplusplus extern "C" { #endif typedef struct ALcomplex { ALdouble Real; ALdouble Imag; } ALcomplex; /** Addition of two complex numbers. */ inline ALcomplex complex_add(ALcomplex a, ALcomplex b) { ALcomplex result; result.Real = a.Real + b.Real; result.Imag = a.Imag + b.Imag; return result; } /** Subtraction of two complex numbers. */ inline ALcomplex complex_sub(ALcomplex a, ALcomplex b) { ALcomplex result; result.Real = a.Real - b.Real; result.Imag = a.Imag - b.Imag; return result; } /** Multiplication of two complex numbers. */ inline ALcomplex complex_mult(ALcomplex a, ALcomplex b) { ALcomplex result; result.Real = a.Real*b.Real - a.Imag*b.Imag; result.Imag = a.Imag*b.Real + a.Real*b.Imag; return result; } /** * Iterative implementation of 2-radix FFT (In-place algorithm). Sign = -1 is * FFT and 1 is iFFT (inverse). Fills FFTBuffer[0...FFTSize-1] with the * Discrete Fourier Transform (DFT) of the time domain data stored in * FFTBuffer[0...FFTSize-1]. FFTBuffer is an array of complex numbers, FFTSize * MUST BE power of two. */ void complex_fft(ALcomplex *FFTBuffer, ALsizei FFTSize, ALdouble Sign); /** * Calculate the complex helical sequence (discrete-time analytical signal) of * the given input using the discrete Hilbert transform (In-place algorithm). * Fills Buffer[0...size-1] with the discrete-time analytical signal stored in * Buffer[0...size-1]. Buffer is an array of complex numbers, size MUST BE * power of two. */ void complex_hilbert(ALcomplex *Buffer, ALsizei size); #ifdef __cplusplus } // extern "C" #endif #endif /* ALCOMPLEX_H */ openal-soft-openal-soft-1.19.1/common/align.h000066400000000000000000000010571335774445300210340ustar00rootroot00000000000000#ifndef AL_ALIGN_H #define AL_ALIGN_H #if defined(HAVE_STDALIGN_H) && defined(HAVE_C11_ALIGNAS) #include #endif #ifndef alignas #if defined(IN_IDE_PARSER) /* KDevelop has problems with our align macro, so just use nothing for parsing. */ #define alignas(x) #elif defined(HAVE_C11_ALIGNAS) #define alignas _Alignas #else /* NOTE: Our custom ALIGN macro can't take a type name like alignas can. For * maximum compatibility, only provide constant integer values to alignas. */ #define alignas(_x) ALIGN(_x) #endif #endif #endif /* AL_ALIGN_H */ openal-soft-openal-soft-1.19.1/common/almalloc.c000066400000000000000000000042751335774445300215260ustar00rootroot00000000000000 #include "config.h" #include "almalloc.h" #include #include #ifdef HAVE_MALLOC_H #include #endif #ifdef HAVE_WINDOWS_H #include #else #include #endif #ifdef __GNUC__ #define LIKELY(x) __builtin_expect(!!(x), !0) #define UNLIKELY(x) __builtin_expect(!!(x), 0) #else #define LIKELY(x) (!!(x)) #define UNLIKELY(x) (!!(x)) #endif void *al_malloc(size_t alignment, size_t size) { #if defined(HAVE_ALIGNED_ALLOC) size = (size+(alignment-1))&~(alignment-1); return aligned_alloc(alignment, size); #elif defined(HAVE_POSIX_MEMALIGN) void *ret; if(posix_memalign(&ret, alignment, size) == 0) return ret; return NULL; #elif defined(HAVE__ALIGNED_MALLOC) return _aligned_malloc(size, alignment); #else char *ret = malloc(size+alignment); if(ret != NULL) { *(ret++) = 0x00; while(((ptrdiff_t)ret&(alignment-1)) != 0) *(ret++) = 0x55; } return ret; #endif } void *al_calloc(size_t alignment, size_t size) { void *ret = al_malloc(alignment, size); if(ret) memset(ret, 0, size); return ret; } void al_free(void *ptr) { #if defined(HAVE_ALIGNED_ALLOC) || defined(HAVE_POSIX_MEMALIGN) free(ptr); #elif defined(HAVE__ALIGNED_MALLOC) _aligned_free(ptr); #else if(ptr != NULL) { char *finder = ptr; do { --finder; } while(*finder == 0x55); free(finder); } #endif } size_t al_get_page_size(void) { static size_t psize = 0; if(UNLIKELY(!psize)) { #ifdef HAVE_SYSCONF #if defined(_SC_PAGESIZE) if(!psize) psize = sysconf(_SC_PAGESIZE); #elif defined(_SC_PAGE_SIZE) if(!psize) psize = sysconf(_SC_PAGE_SIZE); #endif #endif /* HAVE_SYSCONF */ #ifdef _WIN32 if(!psize) { SYSTEM_INFO sysinfo; memset(&sysinfo, 0, sizeof(sysinfo)); GetSystemInfo(&sysinfo); psize = sysinfo.dwPageSize; } #endif if(!psize) psize = DEF_ALIGN; } return psize; } int al_is_sane_alignment_allocator(void) { #if defined(HAVE_ALIGNED_ALLOC) || defined(HAVE_POSIX_MEMALIGN) || defined(HAVE__ALIGNED_MALLOC) return 1; #else return 0; #endif } openal-soft-openal-soft-1.19.1/common/almalloc.h000066400000000000000000000011531335774445300215230ustar00rootroot00000000000000#ifndef AL_MALLOC_H #define AL_MALLOC_H #include #ifdef __cplusplus extern "C" { #endif /* Minimum alignment required by posix_memalign. */ #define DEF_ALIGN sizeof(void*) void *al_malloc(size_t alignment, size_t size); void *al_calloc(size_t alignment, size_t size); void al_free(void *ptr); size_t al_get_page_size(void); /** * Returns non-0 if the allocation function has direct alignment handling. * Otherwise, the standard malloc is used with an over-allocation and pointer * offset strategy. */ int al_is_sane_alignment_allocator(void); #ifdef __cplusplus } #endif #endif /* AL_MALLOC_H */ openal-soft-openal-soft-1.19.1/common/atomic.c000066400000000000000000000003561335774445300212120ustar00rootroot00000000000000 #include "config.h" #include "atomic.h" extern inline void InitRef(RefCount *ptr, uint value); extern inline uint ReadRef(RefCount *ptr); extern inline uint IncrementRef(RefCount *ptr); extern inline uint DecrementRef(RefCount *ptr); openal-soft-openal-soft-1.19.1/common/atomic.h000066400000000000000000000437311335774445300212230ustar00rootroot00000000000000#ifndef AL_ATOMIC_H #define AL_ATOMIC_H #include "static_assert.h" #include "bool.h" #ifdef __GNUC__ /* This helps cast away the const-ness of a pointer without accidentally * changing the pointer type. This is necessary due to Clang's inability to use * atomic_load on a const _Atomic variable. */ #define CONST_CAST(T, V) __extension__({ \ const T _tmp = (V); \ (T)_tmp; \ }) #else #define CONST_CAST(T, V) ((T)(V)) #endif #ifdef __cplusplus extern "C" { #endif /* Atomics using C11 */ #ifdef HAVE_C11_ATOMIC #include #define almemory_order memory_order #define almemory_order_relaxed memory_order_relaxed #define almemory_order_consume memory_order_consume #define almemory_order_acquire memory_order_acquire #define almemory_order_release memory_order_release #define almemory_order_acq_rel memory_order_acq_rel #define almemory_order_seq_cst memory_order_seq_cst #define ATOMIC(T) T _Atomic #define ATOMIC_FLAG atomic_flag #define ATOMIC_INIT atomic_init #define ATOMIC_INIT_STATIC ATOMIC_VAR_INIT /*#define ATOMIC_FLAG_INIT ATOMIC_FLAG_INIT*/ #define ATOMIC_LOAD atomic_load_explicit #define ATOMIC_STORE atomic_store_explicit #define ATOMIC_ADD atomic_fetch_add_explicit #define ATOMIC_SUB atomic_fetch_sub_explicit #define ATOMIC_EXCHANGE atomic_exchange_explicit #define ATOMIC_COMPARE_EXCHANGE_STRONG atomic_compare_exchange_strong_explicit #define ATOMIC_COMPARE_EXCHANGE_WEAK atomic_compare_exchange_weak_explicit #define ATOMIC_FLAG_TEST_AND_SET atomic_flag_test_and_set_explicit #define ATOMIC_FLAG_CLEAR atomic_flag_clear_explicit #define ATOMIC_THREAD_FENCE atomic_thread_fence /* Atomics using GCC intrinsics */ #elif defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 1)) && !defined(__QNXNTO__) enum almemory_order { almemory_order_relaxed, almemory_order_consume, almemory_order_acquire, almemory_order_release, almemory_order_acq_rel, almemory_order_seq_cst }; #define ATOMIC(T) struct { T volatile value; } #define ATOMIC_FLAG ATOMIC(int) #define ATOMIC_INIT(_val, _newval) do { (_val)->value = (_newval); } while(0) #define ATOMIC_INIT_STATIC(_newval) {(_newval)} #define ATOMIC_FLAG_INIT ATOMIC_INIT_STATIC(0) #define ATOMIC_LOAD(_val, _MO) __extension__({ \ __typeof((_val)->value) _r = (_val)->value; \ __asm__ __volatile__("" ::: "memory"); \ _r; \ }) #define ATOMIC_STORE(_val, _newval, _MO) do { \ __asm__ __volatile__("" ::: "memory"); \ (_val)->value = (_newval); \ } while(0) #define ATOMIC_ADD(_val, _incr, _MO) __sync_fetch_and_add(&(_val)->value, (_incr)) #define ATOMIC_SUB(_val, _decr, _MO) __sync_fetch_and_sub(&(_val)->value, (_decr)) #define ATOMIC_EXCHANGE(_val, _newval, _MO) __extension__({ \ __asm__ __volatile__("" ::: "memory"); \ __sync_lock_test_and_set(&(_val)->value, (_newval)); \ }) #define ATOMIC_COMPARE_EXCHANGE_STRONG(_val, _oldval, _newval, _MO1, _MO2) __extension__({ \ __typeof(*(_oldval)) _o = *(_oldval); \ *(_oldval) = __sync_val_compare_and_swap(&(_val)->value, _o, (_newval)); \ *(_oldval) == _o; \ }) #define ATOMIC_FLAG_TEST_AND_SET(_val, _MO) __extension__({ \ __asm__ __volatile__("" ::: "memory"); \ __sync_lock_test_and_set(&(_val)->value, 1); \ }) #define ATOMIC_FLAG_CLEAR(_val, _MO) __extension__({ \ __sync_lock_release(&(_val)->value); \ __asm__ __volatile__("" ::: "memory"); \ }) #define ATOMIC_THREAD_FENCE(order) do { \ enum { must_be_constant = (order) }; \ const int _o = must_be_constant; \ if(_o > almemory_order_relaxed) \ __asm__ __volatile__("" ::: "memory"); \ } while(0) /* Atomics using x86/x86-64 GCC inline assembly */ #elif defined(__GNUC__) && (defined(__i386__) || defined(__x86_64__)) #define WRAP_ADD(S, ret, dest, incr) __asm__ __volatile__( \ "lock; xadd"S" %0,(%1)" \ : "=r" (ret) \ : "r" (dest), "0" (incr) \ : "memory" \ ) #define WRAP_SUB(S, ret, dest, decr) __asm__ __volatile__( \ "lock; xadd"S" %0,(%1)" \ : "=r" (ret) \ : "r" (dest), "0" (-(decr)) \ : "memory" \ ) #define WRAP_XCHG(S, ret, dest, newval) __asm__ __volatile__( \ "lock; xchg"S" %0,(%1)" \ : "=r" (ret) \ : "r" (dest), "0" (newval) \ : "memory" \ ) #define WRAP_CMPXCHG(S, ret, dest, oldval, newval) __asm__ __volatile__( \ "lock; cmpxchg"S" %2,(%1)" \ : "=a" (ret) \ : "r" (dest), "r" (newval), "0" (oldval) \ : "memory" \ ) enum almemory_order { almemory_order_relaxed, almemory_order_consume, almemory_order_acquire, almemory_order_release, almemory_order_acq_rel, almemory_order_seq_cst }; #define ATOMIC(T) struct { T volatile value; } #define ATOMIC_INIT(_val, _newval) do { (_val)->value = (_newval); } while(0) #define ATOMIC_INIT_STATIC(_newval) {(_newval)} #define ATOMIC_LOAD(_val, _MO) __extension__({ \ __typeof((_val)->value) _r = (_val)->value; \ __asm__ __volatile__("" ::: "memory"); \ _r; \ }) #define ATOMIC_STORE(_val, _newval, _MO) do { \ __asm__ __volatile__("" ::: "memory"); \ (_val)->value = (_newval); \ } while(0) #define ATOMIC_ADD(_val, _incr, _MO) __extension__({ \ static_assert(sizeof((_val)->value)==4 || sizeof((_val)->value)==8, "Unsupported size!"); \ __typeof((_val)->value) _r; \ if(sizeof((_val)->value) == 4) WRAP_ADD("l", _r, &(_val)->value, _incr); \ else if(sizeof((_val)->value) == 8) WRAP_ADD("q", _r, &(_val)->value, _incr); \ _r; \ }) #define ATOMIC_SUB(_val, _decr, _MO) __extension__({ \ static_assert(sizeof((_val)->value)==4 || sizeof((_val)->value)==8, "Unsupported size!"); \ __typeof((_val)->value) _r; \ if(sizeof((_val)->value) == 4) WRAP_SUB("l", _r, &(_val)->value, _decr); \ else if(sizeof((_val)->value) == 8) WRAP_SUB("q", _r, &(_val)->value, _decr); \ _r; \ }) #define ATOMIC_EXCHANGE(_val, _newval, _MO) __extension__({ \ __typeof((_val)->value) _r; \ if(sizeof((_val)->value) == 4) WRAP_XCHG("l", _r, &(_val)->value, (_newval)); \ else if(sizeof((_val)->value) == 8) WRAP_XCHG("q", _r, &(_val)->value, (_newval)); \ _r; \ }) #define ATOMIC_COMPARE_EXCHANGE_STRONG(_val, _oldval, _newval, _MO1, _MO2) __extension__({ \ __typeof(*(_oldval)) _old = *(_oldval); \ if(sizeof((_val)->value) == 4) WRAP_CMPXCHG("l", *(_oldval), &(_val)->value, _old, (_newval)); \ else if(sizeof((_val)->value) == 8) WRAP_CMPXCHG("q", *(_oldval), &(_val)->value, _old, (_newval)); \ *(_oldval) == _old; \ }) #define ATOMIC_EXCHANGE_PTR(_val, _newval, _MO) __extension__({ \ void *_r; \ if(sizeof(void*) == 4) WRAP_XCHG("l", _r, &(_val)->value, (_newval)); \ else if(sizeof(void*) == 8) WRAP_XCHG("q", _r, &(_val)->value, (_newval));\ _r; \ }) #define ATOMIC_COMPARE_EXCHANGE_PTR_STRONG(_val, _oldval, _newval, _MO1, _MO2) __extension__({ \ void *_old = *(_oldval); \ if(sizeof(void*) == 4) WRAP_CMPXCHG("l", *(_oldval), &(_val)->value, _old, (_newval)); \ else if(sizeof(void*) == 8) WRAP_CMPXCHG("q", *(_oldval), &(_val)->value, _old, (_newval)); \ *(_oldval) == _old; \ }) #define ATOMIC_THREAD_FENCE(order) do { \ enum { must_be_constant = (order) }; \ const int _o = must_be_constant; \ if(_o > almemory_order_relaxed) \ __asm__ __volatile__("" ::: "memory"); \ } while(0) /* Atomics using Windows methods */ #elif defined(_WIN32) #define WIN32_LEAN_AND_MEAN #include /* NOTE: This mess is *extremely* touchy. It lacks quite a bit of safety * checking due to the lack of multi-statement expressions, typeof(), and C99 * compound literals. It is incapable of properly exchanging floats, which get * casted to LONG/int, and could cast away potential warnings. * * Unfortunately, it's the only semi-safe way that doesn't rely on C99 (because * MSVC). */ inline LONG AtomicAdd32(volatile LONG *dest, LONG incr) { return InterlockedExchangeAdd(dest, incr); } inline LONGLONG AtomicAdd64(volatile LONGLONG *dest, LONGLONG incr) { return InterlockedExchangeAdd64(dest, incr); } inline LONG AtomicSub32(volatile LONG *dest, LONG decr) { return InterlockedExchangeAdd(dest, -decr); } inline LONGLONG AtomicSub64(volatile LONGLONG *dest, LONGLONG decr) { return InterlockedExchangeAdd64(dest, -decr); } inline LONG AtomicSwap32(volatile LONG *dest, LONG newval) { return InterlockedExchange(dest, newval); } inline LONGLONG AtomicSwap64(volatile LONGLONG *dest, LONGLONG newval) { return InterlockedExchange64(dest, newval); } inline void *AtomicSwapPtr(void *volatile *dest, void *newval) { return InterlockedExchangePointer(dest, newval); } inline bool CompareAndSwap32(volatile LONG *dest, LONG newval, LONG *oldval) { LONG old = *oldval; *oldval = InterlockedCompareExchange(dest, newval, *oldval); return old == *oldval; } inline bool CompareAndSwap64(volatile LONGLONG *dest, LONGLONG newval, LONGLONG *oldval) { LONGLONG old = *oldval; *oldval = InterlockedCompareExchange64(dest, newval, *oldval); return old == *oldval; } inline bool CompareAndSwapPtr(void *volatile *dest, void *newval, void **oldval) { void *old = *oldval; *oldval = InterlockedCompareExchangePointer(dest, newval, *oldval); return old == *oldval; } #define WRAP_ADDSUB(T, _func, _ptr, _amnt) _func((T volatile*)(_ptr), (_amnt)) #define WRAP_XCHG(T, _func, _ptr, _newval) _func((T volatile*)(_ptr), (_newval)) #define WRAP_CMPXCHG(T, _func, _ptr, _newval, _oldval) _func((T volatile*)(_ptr), (_newval), (T*)(_oldval)) enum almemory_order { almemory_order_relaxed, almemory_order_consume, almemory_order_acquire, almemory_order_release, almemory_order_acq_rel, almemory_order_seq_cst }; #define ATOMIC(T) struct { T volatile value; } #define ATOMIC_INIT(_val, _newval) do { (_val)->value = (_newval); } while(0) #define ATOMIC_INIT_STATIC(_newval) {(_newval)} #define ATOMIC_LOAD(_val, _MO) ((_val)->value) #define ATOMIC_STORE(_val, _newval, _MO) do { \ (_val)->value = (_newval); \ } while(0) int _al_invalid_atomic_size(); /* not defined */ void *_al_invalid_atomic_ptr_size(); /* not defined */ #define ATOMIC_ADD(_val, _incr, _MO) \ ((sizeof((_val)->value)==4) ? WRAP_ADDSUB(LONG, AtomicAdd32, &(_val)->value, (_incr)) : \ (sizeof((_val)->value)==8) ? WRAP_ADDSUB(LONGLONG, AtomicAdd64, &(_val)->value, (_incr)) : \ _al_invalid_atomic_size()) #define ATOMIC_SUB(_val, _decr, _MO) \ ((sizeof((_val)->value)==4) ? WRAP_ADDSUB(LONG, AtomicSub32, &(_val)->value, (_decr)) : \ (sizeof((_val)->value)==8) ? WRAP_ADDSUB(LONGLONG, AtomicSub64, &(_val)->value, (_decr)) : \ _al_invalid_atomic_size()) #define ATOMIC_EXCHANGE(_val, _newval, _MO) \ ((sizeof((_val)->value)==4) ? WRAP_XCHG(LONG, AtomicSwap32, &(_val)->value, (_newval)) : \ (sizeof((_val)->value)==8) ? WRAP_XCHG(LONGLONG, AtomicSwap64, &(_val)->value, (_newval)) : \ (LONG)_al_invalid_atomic_size()) #define ATOMIC_COMPARE_EXCHANGE_STRONG(_val, _oldval, _newval, _MO1, _MO2) \ ((sizeof((_val)->value)==4) ? WRAP_CMPXCHG(LONG, CompareAndSwap32, &(_val)->value, (_newval), (_oldval)) : \ (sizeof((_val)->value)==8) ? WRAP_CMPXCHG(LONGLONG, CompareAndSwap64, &(_val)->value, (_newval), (_oldval)) : \ (bool)_al_invalid_atomic_size()) #define ATOMIC_EXCHANGE_PTR(_val, _newval, _MO) \ ((sizeof((_val)->value)==sizeof(void*)) ? AtomicSwapPtr((void*volatile*)&(_val)->value, (_newval)) : \ _al_invalid_atomic_ptr_size()) #define ATOMIC_COMPARE_EXCHANGE_PTR_STRONG(_val, _oldval, _newval, _MO1, _MO2)\ ((sizeof((_val)->value)==sizeof(void*)) ? CompareAndSwapPtr((void*volatile*)&(_val)->value, (_newval), (void**)(_oldval)) : \ (bool)_al_invalid_atomic_size()) #define ATOMIC_THREAD_FENCE(order) do { \ enum { must_be_constant = (order) }; \ const int _o = must_be_constant; \ if(_o > almemory_order_relaxed) \ _ReadWriteBarrier(); \ } while(0) #else #error "No atomic functions available on this platform!" #define ATOMIC(T) T #define ATOMIC_INIT(_val, _newval) ((void)0) #define ATOMIC_INIT_STATIC(_newval) (0) #define ATOMIC_LOAD(...) (0) #define ATOMIC_STORE(...) ((void)0) #define ATOMIC_ADD(...) (0) #define ATOMIC_SUB(...) (0) #define ATOMIC_EXCHANGE(...) (0) #define ATOMIC_COMPARE_EXCHANGE_STRONG(...) (0) #define ATOMIC_THREAD_FENCE(...) ((void)0) #endif /* If no PTR xchg variants are provided, the normal ones can handle it. */ #ifndef ATOMIC_EXCHANGE_PTR #define ATOMIC_EXCHANGE_PTR ATOMIC_EXCHANGE #define ATOMIC_COMPARE_EXCHANGE_PTR_STRONG ATOMIC_COMPARE_EXCHANGE_STRONG #define ATOMIC_COMPARE_EXCHANGE_PTR_WEAK ATOMIC_COMPARE_EXCHANGE_WEAK #endif /* If no weak cmpxchg is provided (not all systems will have one), substitute a * strong cmpxchg. */ #ifndef ATOMIC_COMPARE_EXCHANGE_WEAK #define ATOMIC_COMPARE_EXCHANGE_WEAK ATOMIC_COMPARE_EXCHANGE_STRONG #endif #ifndef ATOMIC_COMPARE_EXCHANGE_PTR_WEAK #define ATOMIC_COMPARE_EXCHANGE_PTR_WEAK ATOMIC_COMPARE_EXCHANGE_PTR_STRONG #endif /* If no ATOMIC_FLAG is defined, simulate one with an atomic int using exchange * and store ops. */ #ifndef ATOMIC_FLAG #define ATOMIC_FLAG ATOMIC(int) #define ATOMIC_FLAG_INIT ATOMIC_INIT_STATIC(0) #define ATOMIC_FLAG_TEST_AND_SET(_val, _MO) ATOMIC_EXCHANGE(_val, 1, _MO) #define ATOMIC_FLAG_CLEAR(_val, _MO) ATOMIC_STORE(_val, 0, _MO) #endif #define ATOMIC_LOAD_SEQ(_val) ATOMIC_LOAD(_val, almemory_order_seq_cst) #define ATOMIC_STORE_SEQ(_val, _newval) ATOMIC_STORE(_val, _newval, almemory_order_seq_cst) #define ATOMIC_ADD_SEQ(_val, _incr) ATOMIC_ADD(_val, _incr, almemory_order_seq_cst) #define ATOMIC_SUB_SEQ(_val, _decr) ATOMIC_SUB(_val, _decr, almemory_order_seq_cst) #define ATOMIC_EXCHANGE_SEQ(_val, _newval) ATOMIC_EXCHANGE(_val, _newval, almemory_order_seq_cst) #define ATOMIC_COMPARE_EXCHANGE_STRONG_SEQ(_val, _oldval, _newval) \ ATOMIC_COMPARE_EXCHANGE_STRONG(_val, _oldval, _newval, almemory_order_seq_cst, almemory_order_seq_cst) #define ATOMIC_COMPARE_EXCHANGE_WEAK_SEQ(_val, _oldval, _newval) \ ATOMIC_COMPARE_EXCHANGE_WEAK(_val, _oldval, _newval, almemory_order_seq_cst, almemory_order_seq_cst) #define ATOMIC_EXCHANGE_PTR_SEQ(_val, _newval) ATOMIC_EXCHANGE_PTR(_val, _newval, almemory_order_seq_cst) #define ATOMIC_COMPARE_EXCHANGE_PTR_STRONG_SEQ(_val, _oldval, _newval) \ ATOMIC_COMPARE_EXCHANGE_PTR_STRONG(_val, _oldval, _newval, almemory_order_seq_cst, almemory_order_seq_cst) #define ATOMIC_COMPARE_EXCHANGE_PTR_WEAK_SEQ(_val, _oldval, _newval) \ ATOMIC_COMPARE_EXCHANGE_PTR_WEAK(_val, _oldval, _newval, almemory_order_seq_cst, almemory_order_seq_cst) typedef unsigned int uint; typedef ATOMIC(uint) RefCount; inline void InitRef(RefCount *ptr, uint value) { ATOMIC_INIT(ptr, value); } inline uint ReadRef(RefCount *ptr) { return ATOMIC_LOAD(ptr, almemory_order_acquire); } inline uint IncrementRef(RefCount *ptr) { return ATOMIC_ADD(ptr, 1, almemory_order_acq_rel)+1; } inline uint DecrementRef(RefCount *ptr) { return ATOMIC_SUB(ptr, 1, almemory_order_acq_rel)-1; } /* WARNING: A livelock is theoretically possible if another thread keeps * changing the head without giving this a chance to actually swap in the new * one (practically impossible with this little code, but...). */ #define ATOMIC_REPLACE_HEAD(T, _head, _entry) do { \ T _first = ATOMIC_LOAD(_head, almemory_order_acquire); \ do { \ ATOMIC_STORE(&(_entry)->next, _first, almemory_order_relaxed); \ } while(ATOMIC_COMPARE_EXCHANGE_PTR_WEAK(_head, &_first, _entry, \ almemory_order_acq_rel, almemory_order_acquire) == 0); \ } while(0) #ifdef __cplusplus } #endif #endif /* AL_ATOMIC_H */ openal-soft-openal-soft-1.19.1/common/bool.h000066400000000000000000000003511335774445300206710ustar00rootroot00000000000000#ifndef AL_BOOL_H #define AL_BOOL_H #ifdef HAVE_STDBOOL_H #include #endif #ifndef bool #ifdef HAVE_C99_BOOL #define bool _Bool #else #define bool int #endif #define false 0 #define true 1 #endif #endif /* AL_BOOL_H */ openal-soft-openal-soft-1.19.1/common/math_defs.h000066400000000000000000000024011335774445300216660ustar00rootroot00000000000000#ifndef AL_MATH_DEFS_H #define AL_MATH_DEFS_H #include #ifdef HAVE_FLOAT_H #include #endif #ifndef M_PI #define M_PI (3.14159265358979323846) #endif #define F_PI (3.14159265358979323846f) #define F_PI_2 (1.57079632679489661923f) #define F_TAU (6.28318530717958647692f) #ifndef FLT_EPSILON #define FLT_EPSILON (1.19209290e-07f) #endif #define SQRT_2 1.41421356237309504880 #define SQRT_3 1.73205080756887719318 #define SQRTF_2 1.41421356237309504880f #define SQRTF_3 1.73205080756887719318f #ifndef HUGE_VALF static const union msvc_inf_hack { unsigned char b[4]; float f; } msvc_inf_union = {{ 0x00, 0x00, 0x80, 0x7F }}; #define HUGE_VALF (msvc_inf_union.f) #endif #ifndef HAVE_LOG2F static inline float log2f(float f) { return logf(f) / logf(2.0f); } #endif #ifndef HAVE_CBRTF static inline float cbrtf(float f) { return powf(f, 1.0f/3.0f); } #endif #ifndef HAVE_COPYSIGNF static inline float copysignf(float x, float y) { union { float f; unsigned int u; } ux = { x }, uy = { y }; ux.u &= 0x7fffffffu; ux.u |= (uy.u&0x80000000u); return ux.f; } #endif #define DEG2RAD(x) ((float)(x) * (float)(M_PI/180.0)) #define RAD2DEG(x) ((float)(x) * (float)(180.0/M_PI)) #endif /* AL_MATH_DEFS_H */ openal-soft-openal-soft-1.19.1/common/rwlock.c000066400000000000000000000032241335774445300212340ustar00rootroot00000000000000 #include "config.h" #include "rwlock.h" #include "bool.h" #include "atomic.h" #include "threads.h" /* A simple spinlock. Yield the thread while the given integer is set by * another. Could probably be improved... */ #define LOCK(l) do { \ while(ATOMIC_FLAG_TEST_AND_SET(&(l), almemory_order_acq_rel) == true) \ althrd_yield(); \ } while(0) #define UNLOCK(l) ATOMIC_FLAG_CLEAR(&(l), almemory_order_release) void RWLockInit(RWLock *lock) { InitRef(&lock->read_count, 0); InitRef(&lock->write_count, 0); ATOMIC_FLAG_CLEAR(&lock->read_lock, almemory_order_relaxed); ATOMIC_FLAG_CLEAR(&lock->read_entry_lock, almemory_order_relaxed); ATOMIC_FLAG_CLEAR(&lock->write_lock, almemory_order_relaxed); } void ReadLock(RWLock *lock) { LOCK(lock->read_entry_lock); LOCK(lock->read_lock); /* NOTE: ATOMIC_ADD returns the *old* value! */ if(ATOMIC_ADD(&lock->read_count, 1, almemory_order_acq_rel) == 0) LOCK(lock->write_lock); UNLOCK(lock->read_lock); UNLOCK(lock->read_entry_lock); } void ReadUnlock(RWLock *lock) { /* NOTE: ATOMIC_SUB returns the *old* value! */ if(ATOMIC_SUB(&lock->read_count, 1, almemory_order_acq_rel) == 1) UNLOCK(lock->write_lock); } void WriteLock(RWLock *lock) { if(ATOMIC_ADD(&lock->write_count, 1, almemory_order_acq_rel) == 0) LOCK(lock->read_lock); LOCK(lock->write_lock); } void WriteUnlock(RWLock *lock) { UNLOCK(lock->write_lock); if(ATOMIC_SUB(&lock->write_count, 1, almemory_order_acq_rel) == 1) UNLOCK(lock->read_lock); } openal-soft-openal-soft-1.19.1/common/rwlock.h000066400000000000000000000012441335774445300212410ustar00rootroot00000000000000#ifndef AL_RWLOCK_H #define AL_RWLOCK_H #include "bool.h" #include "atomic.h" #ifdef __cplusplus extern "C" { #endif typedef struct { RefCount read_count; RefCount write_count; ATOMIC_FLAG read_lock; ATOMIC_FLAG read_entry_lock; ATOMIC_FLAG write_lock; } RWLock; #define RWLOCK_STATIC_INITIALIZE { ATOMIC_INIT_STATIC(0), ATOMIC_INIT_STATIC(0), \ ATOMIC_FLAG_INIT, ATOMIC_FLAG_INIT, ATOMIC_FLAG_INIT } void RWLockInit(RWLock *lock); void ReadLock(RWLock *lock); void ReadUnlock(RWLock *lock); void WriteLock(RWLock *lock); void WriteUnlock(RWLock *lock); #ifdef __cplusplus } #endif #endif /* AL_RWLOCK_H */ openal-soft-openal-soft-1.19.1/common/static_assert.h000066400000000000000000000011421335774445300226050ustar00rootroot00000000000000#ifndef AL_STATIC_ASSERT_H #define AL_STATIC_ASSERT_H #include #ifndef static_assert #ifdef HAVE_C11_STATIC_ASSERT #define static_assert _Static_assert #else #define CTASTR2(_pre,_post) _pre##_post #define CTASTR(_pre,_post) CTASTR2(_pre,_post) #if defined(__COUNTER__) #define static_assert(_cond, _msg) typedef struct { int CTASTR(static_assert_failed_at_line_,__LINE__) : !!(_cond); } CTASTR(static_assertion_,__COUNTER__) #else #define static_assert(_cond, _msg) struct { int CTASTR(static_assert_failed_at_line_,__LINE__) : !!(_cond); } #endif #endif #endif #endif /* AL_STATIC_ASSERT_H */ openal-soft-openal-soft-1.19.1/common/threads.c000066400000000000000000000403651335774445300213740ustar00rootroot00000000000000/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include "threads.h" #include #include #include #include "uintmap.h" extern inline althrd_t althrd_current(void); extern inline int althrd_equal(althrd_t thr0, althrd_t thr1); extern inline void althrd_exit(int res); extern inline void althrd_yield(void); extern inline int almtx_lock(almtx_t *mtx); extern inline int almtx_unlock(almtx_t *mtx); extern inline int almtx_trylock(almtx_t *mtx); extern inline void *altss_get(altss_t tss_id); extern inline int altss_set(altss_t tss_id, void *val); #ifndef UNUSED #if defined(__cplusplus) #define UNUSED(x) #elif defined(__GNUC__) #define UNUSED(x) UNUSED_##x __attribute__((unused)) #elif defined(__LCLINT__) #define UNUSED(x) /*@unused@*/ x #else #define UNUSED(x) x #endif #endif #define THREAD_STACK_SIZE (2*1024*1024) /* 2MB */ #ifdef _WIN32 #define WIN32_LEAN_AND_MEAN #include #include /* An associative map of uint:void* pairs. The key is the unique Thread ID and * the value is the thread HANDLE. The thread ID is passed around as the * althrd_t since there is only one ID per thread, whereas a thread may be * referenced by multiple different HANDLEs. This map allows retrieving the * original handle which is needed to join the thread and get its return value. */ static UIntMap ThrdIdHandle = UINTMAP_STATIC_INITIALIZE; /* An associative map of uint:void* pairs. The key is the TLS index (given by * TlsAlloc), and the value is the altss_dtor_t callback. When a thread exits, * we iterate over the TLS indices for their thread-local value and call the * destructor function with it if they're both not NULL. */ static UIntMap TlsDestructors = UINTMAP_STATIC_INITIALIZE; void althrd_setname(althrd_t thr, const char *name) { #if defined(_MSC_VER) #define MS_VC_EXCEPTION 0x406D1388 #pragma pack(push,8) struct { DWORD dwType; // Must be 0x1000. LPCSTR szName; // Pointer to name (in user addr space). DWORD dwThreadID; // Thread ID (-1=caller thread). DWORD dwFlags; // Reserved for future use, must be zero. } info; #pragma pack(pop) info.dwType = 0x1000; info.szName = name; info.dwThreadID = thr; info.dwFlags = 0; __try { RaiseException(MS_VC_EXCEPTION, 0, sizeof(info)/sizeof(ULONG_PTR), (ULONG_PTR*)&info); } __except(EXCEPTION_CONTINUE_EXECUTION) { } #undef MS_VC_EXCEPTION #else (void)thr; (void)name; #endif } typedef struct thread_cntr { althrd_start_t func; void *arg; } thread_cntr; static DWORD WINAPI althrd_starter(void *arg) { thread_cntr cntr; memcpy(&cntr, arg, sizeof(cntr)); free(arg); return (DWORD)((*cntr.func)(cntr.arg)); } int althrd_create(althrd_t *thr, althrd_start_t func, void *arg) { thread_cntr *cntr; DWORD thrid; HANDLE hdl; cntr = malloc(sizeof(*cntr)); if(!cntr) return althrd_nomem; cntr->func = func; cntr->arg = arg; hdl = CreateThread(NULL, THREAD_STACK_SIZE, althrd_starter, cntr, 0, &thrid); if(!hdl) { free(cntr); return althrd_error; } InsertUIntMapEntry(&ThrdIdHandle, thrid, hdl); *thr = thrid; return althrd_success; } int althrd_detach(althrd_t thr) { HANDLE hdl = RemoveUIntMapKey(&ThrdIdHandle, thr); if(!hdl) return althrd_error; CloseHandle(hdl); return althrd_success; } int althrd_join(althrd_t thr, int *res) { DWORD code; HANDLE hdl = RemoveUIntMapKey(&ThrdIdHandle, thr); if(!hdl) return althrd_error; WaitForSingleObject(hdl, INFINITE); GetExitCodeThread(hdl, &code); CloseHandle(hdl); if(res != NULL) *res = (int)code; return althrd_success; } int althrd_sleep(const struct timespec *ts, struct timespec* UNUSED(rem)) { DWORD msec; if(ts->tv_sec < 0 || ts->tv_sec >= (0x7fffffff / 1000) || ts->tv_nsec < 0 || ts->tv_nsec >= 1000000000) return -2; msec = (DWORD)(ts->tv_sec * 1000); msec += (DWORD)((ts->tv_nsec+999999) / 1000000); Sleep(msec); return 0; } int almtx_init(almtx_t *mtx, int type) { if(!mtx) return althrd_error; type &= ~almtx_recursive; if(type != almtx_plain) return althrd_error; InitializeCriticalSection(mtx); return althrd_success; } void almtx_destroy(almtx_t *mtx) { DeleteCriticalSection(mtx); } #if defined(_WIN32_WINNT) && _WIN32_WINNT >= 0x0600 int alcnd_init(alcnd_t *cond) { InitializeConditionVariable(cond); return althrd_success; } int alcnd_signal(alcnd_t *cond) { WakeConditionVariable(cond); return althrd_success; } int alcnd_broadcast(alcnd_t *cond) { WakeAllConditionVariable(cond); return althrd_success; } int alcnd_wait(alcnd_t *cond, almtx_t *mtx) { if(SleepConditionVariableCS(cond, mtx, INFINITE) != 0) return althrd_success; return althrd_error; } void alcnd_destroy(alcnd_t* UNUSED(cond)) { /* Nothing to delete? */ } #else /* WARNING: This is a rather poor implementation of condition variables, with * known problems. However, it's simple, efficient, and good enough for now to * not require Vista. Based on "Strategies for Implementing POSIX Condition * Variables" by Douglas C. Schmidt and Irfan Pyarali: * http://www.cs.wustl.edu/~schmidt/win32-cv-1.html */ /* A better solution may be using Wine's implementation. It requires internals * (NtCreateKeyedEvent, NtReleaseKeyedEvent, and NtWaitForKeyedEvent) from * ntdll, and implemention of exchange and compare-exchange for RefCounts. */ typedef struct { RefCount wait_count; HANDLE events[2]; } _int_alcnd_t; enum { SIGNAL = 0, BROADCAST = 1 }; int alcnd_init(alcnd_t *cond) { _int_alcnd_t *icond = calloc(1, sizeof(*icond)); if(!icond) return althrd_nomem; InitRef(&icond->wait_count, 0); icond->events[SIGNAL] = CreateEventW(NULL, FALSE, FALSE, NULL); icond->events[BROADCAST] = CreateEventW(NULL, TRUE, FALSE, NULL); if(!icond->events[SIGNAL] || !icond->events[BROADCAST]) { if(icond->events[SIGNAL]) CloseHandle(icond->events[SIGNAL]); if(icond->events[BROADCAST]) CloseHandle(icond->events[BROADCAST]); free(icond); return althrd_error; } cond->Ptr = icond; return althrd_success; } int alcnd_signal(alcnd_t *cond) { _int_alcnd_t *icond = cond->Ptr; if(ReadRef(&icond->wait_count) > 0) SetEvent(icond->events[SIGNAL]); return althrd_success; } int alcnd_broadcast(alcnd_t *cond) { _int_alcnd_t *icond = cond->Ptr; if(ReadRef(&icond->wait_count) > 0) SetEvent(icond->events[BROADCAST]); return althrd_success; } int alcnd_wait(alcnd_t *cond, almtx_t *mtx) { _int_alcnd_t *icond = cond->Ptr; int res; IncrementRef(&icond->wait_count); LeaveCriticalSection(mtx); res = WaitForMultipleObjects(2, icond->events, FALSE, INFINITE); if(DecrementRef(&icond->wait_count) == 0 && res == WAIT_OBJECT_0+BROADCAST) ResetEvent(icond->events[BROADCAST]); EnterCriticalSection(mtx); return althrd_success; } void alcnd_destroy(alcnd_t *cond) { _int_alcnd_t *icond = cond->Ptr; CloseHandle(icond->events[SIGNAL]); CloseHandle(icond->events[BROADCAST]); free(icond); } #endif /* defined(_WIN32_WINNT) && _WIN32_WINNT >= 0x0600 */ int alsem_init(alsem_t *sem, unsigned int initial) { *sem = CreateSemaphore(NULL, initial, INT_MAX, NULL); if(*sem != NULL) return althrd_success; return althrd_error; } void alsem_destroy(alsem_t *sem) { CloseHandle(*sem); } int alsem_post(alsem_t *sem) { DWORD ret = ReleaseSemaphore(*sem, 1, NULL); if(ret) return althrd_success; return althrd_error; } int alsem_wait(alsem_t *sem) { DWORD ret = WaitForSingleObject(*sem, INFINITE); if(ret == WAIT_OBJECT_0) return althrd_success; return althrd_error; } int alsem_trywait(alsem_t *sem) { DWORD ret = WaitForSingleObject(*sem, 0); if(ret == WAIT_OBJECT_0) return althrd_success; if(ret == WAIT_TIMEOUT) return althrd_busy; return althrd_error; } int altss_create(altss_t *tss_id, altss_dtor_t callback) { DWORD key = TlsAlloc(); if(key == TLS_OUT_OF_INDEXES) return althrd_error; *tss_id = key; if(callback != NULL) InsertUIntMapEntry(&TlsDestructors, key, callback); return althrd_success; } void altss_delete(altss_t tss_id) { RemoveUIntMapKey(&TlsDestructors, tss_id); TlsFree(tss_id); } int altimespec_get(struct timespec *ts, int base) { static_assert(sizeof(FILETIME) == sizeof(ULARGE_INTEGER), "Size of FILETIME does not match ULARGE_INTEGER"); if(base == AL_TIME_UTC) { union { FILETIME ftime; ULARGE_INTEGER ulint; } systime; GetSystemTimeAsFileTime(&systime.ftime); /* FILETIME is in 100-nanosecond units, or 1/10th of a microsecond. */ ts->tv_sec = systime.ulint.QuadPart/10000000; ts->tv_nsec = (systime.ulint.QuadPart%10000000) * 100; return base; } return 0; } void alcall_once(alonce_flag *once, void (*callback)(void)) { LONG ret; while((ret=InterlockedExchange(once, 1)) == 1) althrd_yield(); if(ret == 0) (*callback)(); InterlockedExchange(once, 2); } void althrd_deinit(void) { ResetUIntMap(&ThrdIdHandle); ResetUIntMap(&TlsDestructors); } void althrd_thread_detach(void) { ALsizei i; LockUIntMapRead(&TlsDestructors); for(i = 0;i < TlsDestructors.size;i++) { void *ptr = altss_get(TlsDestructors.keys[i]); altss_dtor_t callback = (altss_dtor_t)TlsDestructors.values[i]; if(ptr) { if(callback) callback(ptr); altss_set(TlsDestructors.keys[i], NULL); } } UnlockUIntMapRead(&TlsDestructors); } #else #include #include #include #ifdef HAVE_PTHREAD_NP_H #include #endif extern inline int althrd_sleep(const struct timespec *ts, struct timespec *rem); extern inline void alcall_once(alonce_flag *once, void (*callback)(void)); extern inline void althrd_deinit(void); extern inline void althrd_thread_detach(void); void althrd_setname(althrd_t thr, const char *name) { #if defined(HAVE_PTHREAD_SETNAME_NP) #if defined(PTHREAD_SETNAME_NP_ONE_PARAM) if(althrd_equal(thr, althrd_current())) pthread_setname_np(name); #elif defined(PTHREAD_SETNAME_NP_THREE_PARAMS) pthread_setname_np(thr, "%s", (void*)name); #else pthread_setname_np(thr, name); #endif #elif defined(HAVE_PTHREAD_SET_NAME_NP) pthread_set_name_np(thr, name); #else (void)thr; (void)name; #endif } typedef struct thread_cntr { althrd_start_t func; void *arg; } thread_cntr; static void *althrd_starter(void *arg) { thread_cntr cntr; memcpy(&cntr, arg, sizeof(cntr)); free(arg); return (void*)(intptr_t)((*cntr.func)(cntr.arg)); } int althrd_create(althrd_t *thr, althrd_start_t func, void *arg) { thread_cntr *cntr; pthread_attr_t attr; size_t stackmult = 1; int err; cntr = malloc(sizeof(*cntr)); if(!cntr) return althrd_nomem; if(pthread_attr_init(&attr) != 0) { free(cntr); return althrd_error; } retry_stacksize: if(pthread_attr_setstacksize(&attr, THREAD_STACK_SIZE*stackmult) != 0) { pthread_attr_destroy(&attr); free(cntr); return althrd_error; } cntr->func = func; cntr->arg = arg; if((err=pthread_create(thr, &attr, althrd_starter, cntr)) == 0) { pthread_attr_destroy(&attr); return althrd_success; } if(err == EINVAL) { /* If an invalid stack size, try increasing it (limit x4, 8MB). */ if(stackmult < 4) { stackmult *= 2; goto retry_stacksize; } /* If still nothing, try defaults and hope they're good enough. */ if(pthread_create(thr, NULL, althrd_starter, cntr) == 0) { pthread_attr_destroy(&attr); return althrd_success; } } pthread_attr_destroy(&attr); free(cntr); return althrd_error; } int althrd_detach(althrd_t thr) { if(pthread_detach(thr) != 0) return althrd_error; return althrd_success; } int althrd_join(althrd_t thr, int *res) { void *code; if(pthread_join(thr, &code) != 0) return althrd_error; if(res != NULL) *res = (int)(intptr_t)code; return althrd_success; } int almtx_init(almtx_t *mtx, int type) { int ret; if(!mtx) return althrd_error; if((type&~almtx_recursive) != 0) return althrd_error; if(type == almtx_plain) ret = pthread_mutex_init(mtx, NULL); else { pthread_mutexattr_t attr; ret = pthread_mutexattr_init(&attr); if(ret) return althrd_error; if(type == almtx_recursive) { ret = pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE); #ifdef HAVE_PTHREAD_MUTEXATTR_SETKIND_NP if(ret != 0) ret = pthread_mutexattr_setkind_np(&attr, PTHREAD_MUTEX_RECURSIVE); #endif } else ret = 1; if(ret == 0) ret = pthread_mutex_init(mtx, &attr); pthread_mutexattr_destroy(&attr); } return ret ? althrd_error : althrd_success; } void almtx_destroy(almtx_t *mtx) { pthread_mutex_destroy(mtx); } int alcnd_init(alcnd_t *cond) { if(pthread_cond_init(cond, NULL) == 0) return althrd_success; return althrd_error; } int alcnd_signal(alcnd_t *cond) { if(pthread_cond_signal(cond) == 0) return althrd_success; return althrd_error; } int alcnd_broadcast(alcnd_t *cond) { if(pthread_cond_broadcast(cond) == 0) return althrd_success; return althrd_error; } int alcnd_wait(alcnd_t *cond, almtx_t *mtx) { if(pthread_cond_wait(cond, mtx) == 0) return althrd_success; return althrd_error; } void alcnd_destroy(alcnd_t *cond) { pthread_cond_destroy(cond); } int alsem_init(alsem_t *sem, unsigned int initial) { if(sem_init(sem, 0, initial) == 0) return althrd_success; return althrd_error; } void alsem_destroy(alsem_t *sem) { sem_destroy(sem); } int alsem_post(alsem_t *sem) { if(sem_post(sem) == 0) return althrd_success; return althrd_error; } int alsem_wait(alsem_t *sem) { if(sem_wait(sem) == 0) return althrd_success; if(errno == EINTR) return -2; return althrd_error; } int alsem_trywait(alsem_t *sem) { if(sem_trywait(sem) == 0) return althrd_success; if(errno == EWOULDBLOCK) return althrd_busy; if(errno == EINTR) return -2; return althrd_error; } int altss_create(altss_t *tss_id, altss_dtor_t callback) { if(pthread_key_create(tss_id, callback) != 0) return althrd_error; return althrd_success; } void altss_delete(altss_t tss_id) { pthread_key_delete(tss_id); } int altimespec_get(struct timespec *ts, int base) { if(base == AL_TIME_UTC) { int ret; #if _POSIX_TIMERS > 0 ret = clock_gettime(CLOCK_REALTIME, ts); if(ret == 0) return base; #else /* _POSIX_TIMERS > 0 */ struct timeval tv; ret = gettimeofday(&tv, NULL); if(ret == 0) { ts->tv_sec = tv.tv_sec; ts->tv_nsec = tv.tv_usec * 1000; return base; } #endif } return 0; } #endif void al_nssleep(unsigned long nsec) { struct timespec ts, rem; ts.tv_sec = nsec / 1000000000ul; ts.tv_nsec = nsec % 1000000000ul; while(althrd_sleep(&ts, &rem) == -1) ts = rem; } openal-soft-openal-soft-1.19.1/common/threads.h000066400000000000000000000116311335774445300213730ustar00rootroot00000000000000#ifndef AL_THREADS_H #define AL_THREADS_H #include #if defined(__GNUC__) && defined(__i386__) /* force_align_arg_pointer is required for proper function arguments aligning * when SSE code is used. Some systems (Windows, QNX) do not guarantee our * thread functions will be properly aligned on the stack, even though GCC may * generate code with the assumption that it is. */ #define FORCE_ALIGN __attribute__((force_align_arg_pointer)) #else #define FORCE_ALIGN #endif #ifdef __cplusplus extern "C" { #endif enum { althrd_success = 0, althrd_error, althrd_nomem, althrd_timedout, althrd_busy }; enum { almtx_plain = 0, almtx_recursive = 1, }; typedef int (*althrd_start_t)(void*); typedef void (*altss_dtor_t)(void*); #define AL_TIME_UTC 1 #ifdef _WIN32 #define WIN32_LEAN_AND_MEAN #include #ifndef HAVE_STRUCT_TIMESPEC struct timespec { time_t tv_sec; long tv_nsec; }; #endif typedef DWORD althrd_t; typedef CRITICAL_SECTION almtx_t; #if defined(_WIN32_WINNT) && _WIN32_WINNT >= 0x0600 typedef CONDITION_VARIABLE alcnd_t; #else typedef struct { void *Ptr; } alcnd_t; #endif typedef HANDLE alsem_t; typedef DWORD altss_t; typedef LONG alonce_flag; #define AL_ONCE_FLAG_INIT 0 int althrd_sleep(const struct timespec *ts, struct timespec *rem); void alcall_once(alonce_flag *once, void (*callback)(void)); void althrd_deinit(void); void althrd_thread_detach(void); inline althrd_t althrd_current(void) { return GetCurrentThreadId(); } inline int althrd_equal(althrd_t thr0, althrd_t thr1) { return thr0 == thr1; } inline void althrd_exit(int res) { ExitThread(res); } inline void althrd_yield(void) { SwitchToThread(); } inline int almtx_lock(almtx_t *mtx) { if(!mtx) return althrd_error; EnterCriticalSection(mtx); return althrd_success; } inline int almtx_unlock(almtx_t *mtx) { if(!mtx) return althrd_error; LeaveCriticalSection(mtx); return althrd_success; } inline int almtx_trylock(almtx_t *mtx) { if(!mtx) return althrd_error; if(!TryEnterCriticalSection(mtx)) return althrd_busy; return althrd_success; } inline void *altss_get(altss_t tss_id) { return TlsGetValue(tss_id); } inline int altss_set(altss_t tss_id, void *val) { if(TlsSetValue(tss_id, val) == 0) return althrd_error; return althrd_success; } #else #include #include #include #include typedef pthread_t althrd_t; typedef pthread_mutex_t almtx_t; typedef pthread_cond_t alcnd_t; typedef sem_t alsem_t; typedef pthread_key_t altss_t; typedef pthread_once_t alonce_flag; #define AL_ONCE_FLAG_INIT PTHREAD_ONCE_INIT inline althrd_t althrd_current(void) { return pthread_self(); } inline int althrd_equal(althrd_t thr0, althrd_t thr1) { return pthread_equal(thr0, thr1); } inline void althrd_exit(int res) { pthread_exit((void*)(intptr_t)res); } inline void althrd_yield(void) { sched_yield(); } inline int althrd_sleep(const struct timespec *ts, struct timespec *rem) { int ret = nanosleep(ts, rem); if(ret != 0) { ret = ((errno==EINTR) ? -1 : -2); errno = 0; } return ret; } inline int almtx_lock(almtx_t *mtx) { if(pthread_mutex_lock(mtx) != 0) return althrd_error; return althrd_success; } inline int almtx_unlock(almtx_t *mtx) { if(pthread_mutex_unlock(mtx) != 0) return althrd_error; return althrd_success; } inline int almtx_trylock(almtx_t *mtx) { int ret = pthread_mutex_trylock(mtx); switch(ret) { case 0: return althrd_success; case EBUSY: return althrd_busy; } return althrd_error; } inline void *altss_get(altss_t tss_id) { return pthread_getspecific(tss_id); } inline int altss_set(altss_t tss_id, void *val) { if(pthread_setspecific(tss_id, val) != 0) return althrd_error; return althrd_success; } inline void alcall_once(alonce_flag *once, void (*callback)(void)) { pthread_once(once, callback); } inline void althrd_deinit(void) { } inline void althrd_thread_detach(void) { } #endif int althrd_create(althrd_t *thr, althrd_start_t func, void *arg); int althrd_detach(althrd_t thr); int althrd_join(althrd_t thr, int *res); void althrd_setname(althrd_t thr, const char *name); int almtx_init(almtx_t *mtx, int type); void almtx_destroy(almtx_t *mtx); int alcnd_init(alcnd_t *cond); int alcnd_signal(alcnd_t *cond); int alcnd_broadcast(alcnd_t *cond); int alcnd_wait(alcnd_t *cond, almtx_t *mtx); void alcnd_destroy(alcnd_t *cond); int alsem_init(alsem_t *sem, unsigned int initial); void alsem_destroy(alsem_t *sem); int alsem_post(alsem_t *sem); int alsem_wait(alsem_t *sem); int alsem_trywait(alsem_t *sem); int altss_create(altss_t *tss_id, altss_dtor_t callback); void altss_delete(altss_t tss_id); int altimespec_get(struct timespec *ts, int base); void al_nssleep(unsigned long nsec); #ifdef __cplusplus } #endif #endif /* AL_THREADS_H */ openal-soft-openal-soft-1.19.1/common/uintmap.c000066400000000000000000000111061335774445300214060ustar00rootroot00000000000000 #include "config.h" #include "uintmap.h" #include #include #include "almalloc.h" extern inline void LockUIntMapRead(UIntMap *map); extern inline void UnlockUIntMapRead(UIntMap *map); extern inline void LockUIntMapWrite(UIntMap *map); extern inline void UnlockUIntMapWrite(UIntMap *map); void InitUIntMap(UIntMap *map, ALsizei limit) { map->keys = NULL; map->values = NULL; map->size = 0; map->capacity = 0; map->limit = limit; RWLockInit(&map->lock); } void ResetUIntMap(UIntMap *map) { WriteLock(&map->lock); al_free(map->keys); map->keys = NULL; map->values = NULL; map->size = 0; map->capacity = 0; WriteUnlock(&map->lock); } ALenum InsertUIntMapEntry(UIntMap *map, ALuint key, ALvoid *value) { ALsizei pos = 0; WriteLock(&map->lock); if(map->size > 0) { ALsizei count = map->size; do { ALsizei step = count>>1; ALsizei i = pos+step; if(!(map->keys[i] < key)) count = step; else { pos = i+1; count -= step+1; } } while(count > 0); } if(pos == map->size || map->keys[pos] != key) { if(map->size >= map->limit) { WriteUnlock(&map->lock); return AL_OUT_OF_MEMORY; } if(map->size == map->capacity) { ALuint *keys = NULL; ALvoid **values; ALsizei newcap, keylen; newcap = (map->capacity ? (map->capacity<<1) : 4); if(map->limit > 0 && newcap > map->limit) newcap = map->limit; if(newcap > map->capacity) { /* Round the memory size for keys up to a multiple of the * pointer size. */ keylen = newcap * sizeof(map->keys[0]); keylen += sizeof(map->values[0]) - 1; keylen -= keylen%sizeof(map->values[0]); keys = al_malloc(16, keylen + newcap*sizeof(map->values[0])); } if(!keys) { WriteUnlock(&map->lock); return AL_OUT_OF_MEMORY; } values = (ALvoid**)((ALbyte*)keys + keylen); if(map->keys) { memcpy(keys, map->keys, map->size*sizeof(map->keys[0])); memcpy(values, map->values, map->size*sizeof(map->values[0])); } al_free(map->keys); map->keys = keys; map->values = values; map->capacity = newcap; } if(pos < map->size) { memmove(&map->keys[pos+1], &map->keys[pos], (map->size-pos)*sizeof(map->keys[0])); memmove(&map->values[pos+1], &map->values[pos], (map->size-pos)*sizeof(map->values[0])); } map->size++; } map->keys[pos] = key; map->values[pos] = value; WriteUnlock(&map->lock); return AL_NO_ERROR; } ALvoid *RemoveUIntMapKey(UIntMap *map, ALuint key) { ALvoid *ptr = NULL; WriteLock(&map->lock); if(map->size > 0) { ALsizei pos = 0; ALsizei count = map->size; do { ALsizei step = count>>1; ALsizei i = pos+step; if(!(map->keys[i] < key)) count = step; else { pos = i+1; count -= step+1; } } while(count > 0); if(pos < map->size && map->keys[pos] == key) { ptr = map->values[pos]; if(pos < map->size-1) { memmove(&map->keys[pos], &map->keys[pos+1], (map->size-1-pos)*sizeof(map->keys[0])); memmove(&map->values[pos], &map->values[pos+1], (map->size-1-pos)*sizeof(map->values[0])); } map->size--; } } WriteUnlock(&map->lock); return ptr; } ALvoid *LookupUIntMapKey(UIntMap *map, ALuint key) { ALvoid *ptr = NULL; ReadLock(&map->lock); if(map->size > 0) { ALsizei pos = 0; ALsizei count = map->size; do { ALsizei step = count>>1; ALsizei i = pos+step; if(!(map->keys[i] < key)) count = step; else { pos = i+1; count -= step+1; } } while(count > 0); if(pos < map->size && map->keys[pos] == key) ptr = map->values[pos]; } ReadUnlock(&map->lock); return ptr; } openal-soft-openal-soft-1.19.1/common/uintmap.h000066400000000000000000000021011335774445300214060ustar00rootroot00000000000000#ifndef AL_UINTMAP_H #define AL_UINTMAP_H #include #include "AL/al.h" #include "rwlock.h" #ifdef __cplusplus extern "C" { #endif typedef struct UIntMap { ALuint *keys; /* Shares memory with keys. */ ALvoid **values; ALsizei size; ALsizei capacity; ALsizei limit; RWLock lock; } UIntMap; #define UINTMAP_STATIC_INITIALIZE_N(_n) { NULL, NULL, 0, 0, (_n), RWLOCK_STATIC_INITIALIZE } #define UINTMAP_STATIC_INITIALIZE UINTMAP_STATIC_INITIALIZE_N(INT_MAX) void InitUIntMap(UIntMap *map, ALsizei limit); void ResetUIntMap(UIntMap *map); ALenum InsertUIntMapEntry(UIntMap *map, ALuint key, ALvoid *value); ALvoid *RemoveUIntMapKey(UIntMap *map, ALuint key); ALvoid *LookupUIntMapKey(UIntMap *map, ALuint key); inline void LockUIntMapRead(UIntMap *map) { ReadLock(&map->lock); } inline void UnlockUIntMapRead(UIntMap *map) { ReadUnlock(&map->lock); } inline void LockUIntMapWrite(UIntMap *map) { WriteLock(&map->lock); } inline void UnlockUIntMapWrite(UIntMap *map) { WriteUnlock(&map->lock); } #ifdef __cplusplus } #endif #endif /* AL_UINTMAP_H */ openal-soft-openal-soft-1.19.1/common/win_main_utf8.h000066400000000000000000000046341335774445300225150ustar00rootroot00000000000000#ifndef WIN_MAIN_UTF8_H #define WIN_MAIN_UTF8_H /* For Windows systems this provides a way to get UTF-8 encoded argv strings, * and also overrides fopen to accept UTF-8 filenames. Working with wmain * directly complicates cross-platform compatibility, while normal main() in * Windows uses the current codepage (which has limited availability of * characters). * * For MinGW, you must link with -municode */ #ifdef _WIN32 #define WIN32_LEAN_AND_MEAN #include #include static FILE *my_fopen(const char *fname, const char *mode) { WCHAR *wname=NULL, *wmode=NULL; int namelen, modelen; FILE *file = NULL; errno_t err; namelen = MultiByteToWideChar(CP_UTF8, 0, fname, -1, NULL, 0); modelen = MultiByteToWideChar(CP_UTF8, 0, mode, -1, NULL, 0); if(namelen <= 0 || modelen <= 0) { fprintf(stderr, "Failed to convert UTF-8 fname \"%s\", mode \"%s\"\n", fname, mode); return NULL; } wname = calloc(sizeof(WCHAR), namelen+modelen); wmode = wname + namelen; MultiByteToWideChar(CP_UTF8, 0, fname, -1, wname, namelen); MultiByteToWideChar(CP_UTF8, 0, mode, -1, wmode, modelen); err = _wfopen_s(&file, wname, wmode); if(err) { errno = err; file = NULL; } free(wname); return file; } #define fopen my_fopen static char **arglist; static void cleanup_arglist(void) { free(arglist); } static void GetUnicodeArgs(int *argc, char ***argv) { size_t total; wchar_t **args; int nargs, i; args = CommandLineToArgvW(GetCommandLineW(), &nargs); if(!args) { fprintf(stderr, "Failed to get command line args: %ld\n", GetLastError()); exit(EXIT_FAILURE); } total = sizeof(**argv) * nargs; for(i = 0;i < nargs;i++) total += WideCharToMultiByte(CP_UTF8, 0, args[i], -1, NULL, 0, NULL, NULL); atexit(cleanup_arglist); arglist = *argv = calloc(1, total); (*argv)[0] = (char*)(*argv + nargs); for(i = 0;i < nargs-1;i++) { int len = WideCharToMultiByte(CP_UTF8, 0, args[i], -1, (*argv)[i], 65535, NULL, NULL); (*argv)[i+1] = (*argv)[i] + len; } WideCharToMultiByte(CP_UTF8, 0, args[i], -1, (*argv)[i], 65535, NULL, NULL); *argc = nargs; LocalFree(args); } #define GET_UNICODE_ARGS(argc, argv) GetUnicodeArgs(argc, argv) #else /* Do nothing. */ #define GET_UNICODE_ARGS(argc, argv) #endif #endif /* WIN_MAIN_UTF8_H */ openal-soft-openal-soft-1.19.1/config.h.in000066400000000000000000000130131335774445300203170ustar00rootroot00000000000000/* API declaration export attribute */ #define AL_API ${EXPORT_DECL} #define ALC_API ${EXPORT_DECL} /* Define any available alignment declaration */ #define ALIGN(x) ${ALIGN_DECL} /* Define a built-in call indicating an aligned data pointer */ #define ASSUME_ALIGNED(x, y) ${ASSUME_ALIGNED_DECL} /* Define if HRTF data is embedded in the library */ #cmakedefine ALSOFT_EMBED_HRTF_DATA /* Define if we have the sysconf function */ #cmakedefine HAVE_SYSCONF /* Define if we have the C11 aligned_alloc function */ #cmakedefine HAVE_ALIGNED_ALLOC /* Define if we have the posix_memalign function */ #cmakedefine HAVE_POSIX_MEMALIGN /* Define if we have the _aligned_malloc function */ #cmakedefine HAVE__ALIGNED_MALLOC /* Define if we have the proc_pidpath function */ #cmakedefine HAVE_PROC_PIDPATH /* Define if we have the getopt function */ #cmakedefine HAVE_GETOPT /* Define if we have SSE CPU extensions */ #cmakedefine HAVE_SSE #cmakedefine HAVE_SSE2 #cmakedefine HAVE_SSE3 #cmakedefine HAVE_SSE4_1 /* Define if we have ARM Neon CPU extensions */ #cmakedefine HAVE_NEON /* Define if we have the ALSA backend */ #cmakedefine HAVE_ALSA /* Define if we have the OSS backend */ #cmakedefine HAVE_OSS /* Define if we have the Solaris backend */ #cmakedefine HAVE_SOLARIS /* Define if we have the SndIO backend */ #cmakedefine HAVE_SNDIO /* Define if we have the QSA backend */ #cmakedefine HAVE_QSA /* Define if we have the WASAPI backend */ #cmakedefine HAVE_WASAPI /* Define if we have the DSound backend */ #cmakedefine HAVE_DSOUND /* Define if we have the Windows Multimedia backend */ #cmakedefine HAVE_WINMM /* Define if we have the PortAudio backend */ #cmakedefine HAVE_PORTAUDIO /* Define if we have the PulseAudio backend */ #cmakedefine HAVE_PULSEAUDIO /* Define if we have the JACK backend */ #cmakedefine HAVE_JACK /* Define if we have the CoreAudio backend */ #cmakedefine HAVE_COREAUDIO /* Define if we have the OpenSL backend */ #cmakedefine HAVE_OPENSL /* Define if we have the Wave Writer backend */ #cmakedefine HAVE_WAVE /* Define if we have the SDL2 backend */ #cmakedefine HAVE_SDL2 /* Define if we have the stat function */ #cmakedefine HAVE_STAT /* Define if we have the lrintf function */ #cmakedefine HAVE_LRINTF /* Define if we have the modff function */ #cmakedefine HAVE_MODFF /* Define if we have the log2f function */ #cmakedefine HAVE_LOG2F /* Define if we have the cbrtf function */ #cmakedefine HAVE_CBRTF /* Define if we have the copysignf function */ #cmakedefine HAVE_COPYSIGNF /* Define if we have the strtof function */ #cmakedefine HAVE_STRTOF /* Define if we have the strnlen function */ #cmakedefine HAVE_STRNLEN /* Define if we have the __int64 type */ #cmakedefine HAVE___INT64 /* Define to the size of a long int type */ #cmakedefine SIZEOF_LONG ${SIZEOF_LONG} /* Define to the size of a long long int type */ #cmakedefine SIZEOF_LONG_LONG ${SIZEOF_LONG_LONG} /* Define if we have C99 _Bool support */ #cmakedefine HAVE_C99_BOOL /* Define if we have C11 _Static_assert support */ #cmakedefine HAVE_C11_STATIC_ASSERT /* Define if we have C11 _Alignas support */ #cmakedefine HAVE_C11_ALIGNAS /* Define if we have C11 _Atomic support */ #cmakedefine HAVE_C11_ATOMIC /* Define if we have GCC's destructor attribute */ #cmakedefine HAVE_GCC_DESTRUCTOR /* Define if we have GCC's format attribute */ #cmakedefine HAVE_GCC_FORMAT /* Define if we have stdint.h */ #cmakedefine HAVE_STDINT_H /* Define if we have stdbool.h */ #cmakedefine HAVE_STDBOOL_H /* Define if we have stdalign.h */ #cmakedefine HAVE_STDALIGN_H /* Define if we have windows.h */ #cmakedefine HAVE_WINDOWS_H /* Define if we have dlfcn.h */ #cmakedefine HAVE_DLFCN_H /* Define if we have pthread_np.h */ #cmakedefine HAVE_PTHREAD_NP_H /* Define if we have malloc.h */ #cmakedefine HAVE_MALLOC_H /* Define if we have dirent.h */ #cmakedefine HAVE_DIRENT_H /* Define if we have strings.h */ #cmakedefine HAVE_STRINGS_H /* Define if we have cpuid.h */ #cmakedefine HAVE_CPUID_H /* Define if we have intrin.h */ #cmakedefine HAVE_INTRIN_H /* Define if we have sys/sysconf.h */ #cmakedefine HAVE_SYS_SYSCONF_H /* Define if we have guiddef.h */ #cmakedefine HAVE_GUIDDEF_H /* Define if we have initguid.h */ #cmakedefine HAVE_INITGUID_H /* Define if we have ieeefp.h */ #cmakedefine HAVE_IEEEFP_H /* Define if we have float.h */ #cmakedefine HAVE_FLOAT_H /* Define if we have fenv.h */ #cmakedefine HAVE_FENV_H /* Define if we have GCC's __get_cpuid() */ #cmakedefine HAVE_GCC_GET_CPUID /* Define if we have the __cpuid() intrinsic */ #cmakedefine HAVE_CPUID_INTRINSIC /* Define if we have the _BitScanForward64() intrinsic */ #cmakedefine HAVE_BITSCANFORWARD64_INTRINSIC /* Define if we have the _BitScanForward() intrinsic */ #cmakedefine HAVE_BITSCANFORWARD_INTRINSIC /* Define if we have _controlfp() */ #cmakedefine HAVE__CONTROLFP /* Define if we have __control87_2() */ #cmakedefine HAVE___CONTROL87_2 /* Define if we have pthread_setschedparam() */ #cmakedefine HAVE_PTHREAD_SETSCHEDPARAM /* Define if we have pthread_setname_np() */ #cmakedefine HAVE_PTHREAD_SETNAME_NP /* Define if pthread_setname_np() only accepts one parameter */ #cmakedefine PTHREAD_SETNAME_NP_ONE_PARAM /* Define if pthread_setname_np() accepts three parameters */ #cmakedefine PTHREAD_SETNAME_NP_THREE_PARAMS /* Define if we have pthread_set_name_np() */ #cmakedefine HAVE_PTHREAD_SET_NAME_NP /* Define if we have pthread_mutexattr_setkind_np() */ #cmakedefine HAVE_PTHREAD_MUTEXATTR_SETKIND_NP /* Define if we have pthread_mutex_timedlock() */ #cmakedefine HAVE_PTHREAD_MUTEX_TIMEDLOCK openal-soft-openal-soft-1.19.1/docs/000077500000000000000000000000001335774445300172265ustar00rootroot00000000000000openal-soft-openal-soft-1.19.1/docs/3D7.1.txt000066400000000000000000000102771335774445300204720ustar00rootroot00000000000000Overview ======== 3D7.1 is a custom speaker layout designed by Simon Goodwin at Codemasters[1]. Typical surround sound setups, like quad, 5.1, 6.1, and 7.1, only produce audio on a 2D horizontal plane with no verticality, which means the envelopment of "surround" sound is limited to left, right, front, and back panning. Sounds that should come from above or below will still only play in 2D since there is no height difference in the speaker array. To work around this, 3D7.1 was designed so that some speakers are placed higher than the listener while others are lower, in a particular configuration that tries to provide balanced output and maintain some compatibility with existing audio content and software. Software that recognizes this setup, or can be configured for it, can then take advantage of the height difference and increase the perception of verticality for true 3D audio. The result is that sounds can be perceived as coming from left, right, front, and back, as well as up and down. [1] http://www.codemasters.com/research/3D_sound_for_3D_games.pdf Hardware Setup ============== Setting up 3D7.1 requires an audio device capable of raw 8-channel or 7.1 output, along with a 7.1 speaker kit. The speakers should be hooked up to the device in the usual way, with front-left and front-right output going to the front-left and front-right speakers, etc. The placement of the speakers should be set up according to the table below. Azimuth is the horizontal angle in degrees, with 0 directly in front and positive values go /left/, and elevation is the vertical angle in degrees, with 0 at head level and positive values go /up/. ------------------------------------------------------------ - Speaker label | Azimuth | Elevation | New label - ------------------------------------------------------------ - Front left | 51 | 24 | Upper front left - - Front right | -51 | 24 | Upper front right - - Front center | 0 | 0 | Front center - - Subwoofer/LFE | N/A | N/A | Subwoofer/LFE - - Side left | 129 | -24 | Lower back left - - Side right | -129 | -24 | Lower back right - - Back left | 180 | 55 | Upper back center - - Back right | 0 | -55 | Lower front center - ------------------------------------------------------------ Note that this speaker layout *IS NOT* compatible with standard 7.1 content. Audio that should be played from the back will come out at the wrong location since the back speakers are placed in the lower front and upper back positions. However, this speaker layout *IS* more or less compatible with standard 5.1 content. Though slightly tilted, to a listener sitting a bit further back from the center, the front and side speakers will be close enough to their intended locations that the output won't be too off. Software Setup ============== To enable 3D7.1 on OpenAL Soft, first make sure the audio device is configured for 7.1 output. Then in the alsoft-config utility, under the Renderer tab, select the 3D7.1.ambdec preset for the 7.1 Surround decoder configuration. And that's it. Any applications using OpenAL Soft can take advantage of fully 3D audio, and multi-channel sounds will be properly remixed for the speaker layout. Playback can be improved by (copying and) modifying the 3D7.1.ambdec preset, changing the specified speaker distances to match the the real distance (in meters) from the center of the speaker array, then enable High Quality Mode in alsoft-config. That will improve the quality when the speakers are not all equidistant. Note that care must be taken that the audio device is not treated as a "true" 7.1 device by non-3D7.1-capable applications. In particular, the audio server should not try to upmix stereo and 5.1 content to "fill out" the back speakers, and non-3D7.1 apps should be set to either stereo or 5.1 output. As such, if your system is capable of it, it may be useful to define a virtual 5.1 device that maps the front, side, and LFE channels to the main device for output and disables upmixing, then use that virtual 5.1 device for apps that do normal stereo or surround sound output, and use the main device for apps that understand 3D7.1 output. openal-soft-openal-soft-1.19.1/docs/ambdec.txt000066400000000000000000000210161335774445300212020ustar00rootroot00000000000000AmbDec Configuration Files ========================== AmbDec configuration files were developed by Fons Adriaensen as part of the AmbDec program . The file works by specifying a decoder matrix or matrices which transform ambisonic channels into speaker feeds. Single-band decoders specify a single matrix that transforms all frequencies, while dual-band decoders specifies two matrices where one transforms low frequency sounds and the other transforms high frequency sounds. See docs/ambisonics.txt for more general information about ambisonics. Starting with OpenAL Soft 1.18, version 3 of the file format is supported as a means of specifying custom surround sound speaker layouts. These configuration files are also used to enable the high-quality ambisonic decoder. File Format =========== As of this writing, there is no official documentation of the .ambdec file format. However, the format as OpenAL Soft sees it is as follows: The file is plain text. Comments start with a hash/pound character (#). There may be any amount of whitespace in between the option and parameter values. Strings are *not* enclosed in quotation marks. /description Specifies a text description of the configuration. Ignored by OpenAL Soft. /version Declares the format version used by the configuration file. OpenAL Soft currently only supports version 3. /dec/chan_mask Specifies a hexadecimal mask value of ambisonic input channels used by this decoder. Counting up from the least significant bit, bit 0 maps to Ambisonic Channel Number (ACN) 0, bit 1 maps to ACN 1, etc. As an example, a value of 'b' enables bits 0, 1, and 3 (1011 in binary), which correspond to ACN 0, 1, and 3 (first-order horizontal). /dec/freq_bands Specifies the number of frequency bands used by the decoder. This must be 1 for single-band or 2 for dual-band. /dec/speakers Specifies the number of output speakers to decode to. /dec/coeff_scale Specifies the scaling used by the decoder coefficients. Currently recognized types are fuma, sn3d, and n3d, for Furse-Malham (FuMa), semi-normalized (SN3D), and fully normalized (N3D) scaling, respectively. /opt/input_scale Specifies the scaling used by the ambisonic input data. As OpenAL Soft renders the data itself and knows the scaling, this is ignored. /opt/nfeff_comp Specifies whether near-field effect compensation is off (not applied at all), applied on input (faster, less accurate with varying speaker distances) or output (slower, more accurate with varying speaker distances). Ignored by OpenAL Soft. /opt/delay_comp Specifies whether delay compensation is applied for output. This is used to correct for time variations caused by different speaker distances. As OpenAL Soft has its own config option for this, this is ignored. /opt/level_comp Specifies whether gain compensation is applied for output. This is used to correct for volume variations caused by different speaker distances. As OpenAL Soft has its own config option for this, this is ignored. /opt/xover_freq Specifies the crossover frequency for dual-band decoders. Frequencies less than this are fed to the low-frequency matrix, and frequencies greater than this are fed to the high-frequency matrix. Unused for single-band decoders. /opt/xover_ratio Specifies the volume ratio between the frequency bands. Values greater than 0 decrease the low-frequency output by half the specified value and increase the high-frequency output by half the specified value, while values less than 0 increase the low-frequency output and decrease the high-frequency output to similar effect. Unused for single-band decoders. /speakers/{ Begins the output speaker definitions. A speaker is defined using the add_spkr command, and there must be a matching number of speaker definitions as the specified speaker count. The definitions are ended with a "/}". add_spkr Defines an output speaker. The ID is a string identifier for the output speaker (see Speaker IDs below). The distance is in meters from the center-point of the physical speaker array. The azimuth is the horizontal angle of the speaker, in degrees, where 0 is directly front and positive values go left. The elevation is the vertical angle of the speaker, in degrees, where 0 is directly front and positive goes upward. The connection string is the JACK port name the speaker should connect to. Currently, OpenAL Soft uses the ID and distance, and ignores the rest. /lfmatrix/{ Begins the low-frequency decoder matrix definition. The definition should include an order_gain command to specify the base gain for the ambisonic orders. Each matrix row is defined using the add_row command, and there must be a matching number of rows as the number of speakers. Additionally the row definitions are in the same order as the speaker definitions. The definitions are ended with a "/}". Only valid for dual-band decoders. /hfmatrix/{ Begins the high-frequency decoder matrix definition. The definition should include an order_gain command to specify the base gain for the ambisonic orders. Each matrix row is defined using the add_row command, and there must be a matching number of rows as the number of speakers, Additionally the row definitions are in the same order as the speaker definitions. The definitions are ended with a "/}". Only valid for dual-band decoders. /matrix/{ Begins the decoder matrix definition. The definition should include an order_gain command to specify the base gain for the ambisonic orders. Each matrix row is defined using the add_row command, and there must be a matching number of rows as the number of speakers. Additionally the row definitions are in the same order as the speaker definitions. The definitions are ended with a "/}". Only valid for single-band decoders. order_gain Specifies the base gain for the zeroth-, first-, second-, and third-order coefficients in the given matrix, automatically scaling the related coefficients. This should be specified at the beginning of the matrix definition. add_row ... Specifies a row of coefficients for the matrix. There should be one coefficient for each enabled bit in the channel mask, and corresponds to the matching ACN channel. /end Marks the end of the configuration file. Speaker IDs =========== The AmbDec program uses the speaker ID as a label to display in its config dialog, but does not otherwise use it for any particular purpose. However, since OpenAL Soft needs to match a speaker definition to an output channel, the speaker ID is used to identify what output channel it correspond to. Therefore, OpenAL Soft requires these channel labels to be recognized: LF = Front left RF = Front right LS = Side left RS = Side right LB = Back left RB = Back right CE = Front center CB = Back center Additionally, configuration files for surround51 will acknowledge back speakers for side channels, and surround51rear will acknowledge side speakers for back channels, to avoid issues with a configuration expecting 5.1 to use the side channels when the device is configured for back, or vice-versa. Furthermore, OpenAL Soft does not require a speaker definition for each output channel the configuration is used with. So for example a 5.1 configuration may omit a front center speaker definition, in which case the front center output channel will not contribute to the ambisonic decode (though OpenAL Soft will still use it in certain scenarios, such as the AL_EFFECT_DEDICATED_DIALOGUE effect). Creating Configuration Files ============================ Configuration files can be created or modified by hand in a text editor. The AmbDec program also has a GUI for creating and editing them. However, these methods rely on you having the coefficients to fill in... they won't be generated for you. Another option is to use the Ambisonic Decoder Toolbox . This is a collection of MATLAB and GNU Octave scripts that can generate AmbDec configuration files from an array of speaker definitions (labels and positions). If you're familiar with using MATLAB or GNU Octave, this may be a good option. There are plans for OpenAL Soft to include a utility to generate coefficients and make configuration files. However, calculating proper coefficients for anything other than regular or semi-regular speaker setups is somewhat of a black art, so may take some time. openal-soft-openal-soft-1.19.1/docs/ambisonics.txt000066400000000000000000000160761335774445300221300ustar00rootroot00000000000000OpenAL Soft's renderer has advanced quite a bit since its start with panned stereo output. Among these advancements is support for surround sound output, using psychoacoustic modeling and more accurate plane wave reconstruction. The concepts in use may not be immediately obvious to people just getting into 3D audio, or people who only have more indirect experience through the use of 3D audio APIs, so this document aims to introduce the ideas and purpose of Ambisonics as used by OpenAL Soft. What Is It? =========== Originally developed in the 1970s by Michael Gerzon and a team others, Ambisonics was created as a means of recording and playing back 3D sound. Taking advantage of the way sound waves propogate, it is possible to record a fully 3D soundfield using as few as 4 channels (or even just 3, if you don't mind dropping down to 2 dimensions like many surround sound systems are). This representation is called B-Format. It was designed to handle audio independent of any specific speaker layout, so with a proper decoder the same recording can be played back on a variety of speaker setups, from quadraphonic and hexagonal to cubic and other periphonic (with height) layouts. Although it was developed decades ago, various factors held ambisonics back from really taking hold in the consumer market. However, given the solid theories backing it, as well as the potential and practical benefits on offer, it continued to be a topic of research over the years, with improvements being made over the original design. One of the improvements made is the use of Spherical Harmonics to increase the number of channels for greater spatial definition. Where the original 4-channel design is termed as "First-Order Ambisonics", or FOA, the increased channel count through the use of Spherical Harmonics is termed as "Higher-Order Ambisonics", or HOA. The details of higher order ambisonics are out of the scope of this document, but know that the added channels are still independent of any speaker layout, and aim to further improve the spatial detail for playback. Today, the processing power available on even low-end computers means real-time Ambisonics processing is possible. Not only can decoders be implemented in software, but so can encoders, synthesizing a soundfield using multiple panned sources, thus taking advantage of what ambisonics offers in a virtual audio environment. How Does It Help? ================= Positional sound has come a long way from pan-pot stereo (aka pair-wise). Although useful at the time, the issues became readily apparent when trying to extend it for surround sound. Pan-pot doesn't work as well for depth (front- back) or vertical panning, it has a rather small "sweet spot" (the area the head needs to be in to perceive the sound in its intended direction), and it misses key distance-related details of sound waves. Ambisonics takes a different approach. It uses all available speakers to help localize a sound, and it also takes into account how the brain localizes low frequency sounds compared to high frequency ones -- a so-called psychoacoustic model. It may seem counter-intuitive (if a sound is coming from the front-left, surely just play it on the front-left speaker?), but to properly model a sound coming from where a speaker doesn't exist, more needs to be done to construct a proper sound wave that's perceived to come from the intended direction. Doing this creates a larger sweet spot, allowing the perceived sound direction to remain correct over a larger area around the center of the speakers. In addition, Ambisonics can encode the near-field effect of sounds, effectively capturing the sound distance. The near-field effect is a subtle low-frequency boost as a result of wave-front curvature, and properly compensating for this occuring with the output speakers (as well as emulating it with a synthesized soundfield) can create an improved sense of distance for sounds that move near or far. How Is It Used? =============== As a 3D audio API, OpenAL is tasked with playing 3D sound as best it can with the speaker setup the user has. Since the OpenAL API does not explicitly handle the output channel configuration, it has a lot of leeway in how to deal with the audio before it's played back for the user to hear. Consequently, OpenAL Soft (or any other OpenAL implementation that wishes to) can render using Ambisonics and decode the ambisonic mix for a high level of accuracy over what simple pan-pot could provide. This is effectively what the high-quality mode option does, when given an appropriate decoder configuation for the playback channel layout. 3D rendering and effect mixing is done to an ambisonic buffer, which is later decoded for output utilizing the benefits available to ambisonic processing. The basic, non-high-quality, renderer uses similar principles, however it skips the frequency-dependent processing (so low frequency sounds are treated the same as high frequency sounds) and does some creative manipulation of the involved math to skip the intermediate ambisonic buffer, rendering more directly to the output while still taking advantage of all the available speakers to reconstruct the sound wave. This method trades away some playback quality for less memory and processor usage. In addition to providing good support for surround sound playback, Ambisonics also has benefits with stereo output. 2-channel UHJ is a stereo-compatible format that encodes some surround sound information using a wide-band 90-degree phase shift filter. It works by taking a B-Format signal, and deriving a frontal stereo mix with the rear sounds attenuated and filtered in with it. Although the result is not as good as 3-channel (2D) B-Format, it has the distinct advantage of only using 2 channels and being compatible with stereo output. This means it will sound just fine when played as-is through a normal stereo device, or it may optionally be fed to a properly configured surround sound receiver which can extract the encoded information and restore some of the original surround sound signal. What Are Its Limitations? ========================= As good as Ambisonics is, it's not a magic bullet that can overcome all problems. One of the bigger issues it has is dealing with irregular speaker setups, such as 5.1 surround sound. The problem mainly lies in the imbalanced speaker positioning -- there are three speakers within the front 60-degree area (meaning only 30-degree gaps in between each of the three speakers), while only two speakers cover the back 140-degree area, leaving 80-degree gaps on the sides. It should be noted that this problem is inherent to the speaker layout itself; there isn't much that can be done to get an optimal surround sound response, with ambisonics or not. It will do the best it can, but there are trade-offs between detail and accuracy. Another issue lies with HRTF. While it's certainly possible to play an ambisonic mix using HRTF and retain a sense of 3D sound, doing so with a high degree of spatial detail requires a fair amount of resources, in both memory and processing time. And even with it, mixing sounds with HRTF directly will still be better for positional accuracy. openal-soft-openal-soft-1.19.1/docs/env-vars.txt000066400000000000000000000102651335774445300215340ustar00rootroot00000000000000Useful Environment Variables Below is a list of environment variables that can be set to aid with running or debugging apps that use OpenAL Soft. They should be set before the app is run. *** Logging *** ALSOFT_LOGLEVEL Specifies the amount of logging OpenAL Soft will write out: 0 - Effectively disables all logging 1 - Prints out errors only 2 - Prints out warnings and errors 3 - Prints out additional information, as well as warnings and errors 4 - Same as 3, but also device and context reference count changes. This will print out *a lot* of info, and is generally not useful unless you're trying to track a reference leak within the library. ALSOFT_LOGFILE Specifies a filename that logged output will be written to. Note that the file will be first cleared when logging is initialized. *** Overrides *** ALSOFT_CONF Specifies an additional configuration file to load settings from. These settings will take precedence over the global and user configs, but not other environment variable settings. ALSOFT_DRIVERS Overrides the drivers config option. This specifies which backend drivers to consider or not consider for use. Please see the drivers option in alsoftrc.sample for a list of available drivers. ALSOFT_DEFAULT_REVERB Specifies the default reverb preset to apply to sources. Please see the default-reverb option in alsoftrc.sample for additional information and a list of available presets. ALSOFT_TRAP_AL_ERROR Set to "true" or "1" to force trapping AL errors. Like the trap-al-error config option, this will raise a SIGTRAP signal (or a breakpoint exception under Windows) when a context-level error is generated. Useful when run under a debugger as it will break execution right when the error occurs, making it easier to track the cause. ALSOFT_TRAP_ALC_ERROR Set to "true" or "1" to force trapping ALC errors. Like the trap-alc-error config option, this will raise a SIGTRAP signal (or a breakpoint exception under Windows) when a device-level error is generated. Useful when run under a debugger as it will break execution right when the error occurs, making it easier to track the cause. ALSOFT_TRAP_ERROR Set to "true" or "1" to force trapping both ALC and AL errors. *** Compatibility *** __ALSOFT_HALF_ANGLE_CONES Older versions of OpenAL Soft incorrectly calculated the cone angles to range between 0 and 180 degrees, instead of the expected range of 0 to 360 degrees. Setting this to "true" or "1" restores the old buggy behavior, for apps that were written to expect the incorrect range. __ALSOFT_REVERSE_Z Applications that don't natively use OpenAL's coordinate system have to convert to it before passing in 3D coordinates. Depending on how exactly this is done, it can cause correct output for stereo but incorrect Z panning for surround sound (i.e., sounds that are supposed to be behind you sound like they're in front, and vice-versa). Setting this to "true" or "1" will negate the localized Z coordinate to attempt to fix output for apps that have incorrect front/back panning. __ALSOFT_SUSPEND_CONTEXT Due to the OpenAL spec not being very clear about them, behavior of the alcSuspendContext and alcProcessContext methods has varied, and because of that, previous versions of OpenAL Soft had them no-op. Creative's hardware drivers and the Rapture3D driver, however, use these methods to batch changes, which some applications make use of to protect against partial updates. In an attempt to standardize on that behavior, OpenAL Soft has changed those methods accordingly. Setting this to "ignore" restores the previous no-op behavior for applications that interact poorly with the new behavior. __ALSOFT_REVERB_IGNORES_SOUND_SPEED Older versions of OpenAL Soft ignored the app-specified speed of sound when calculating distance-related reverb decays and always assumed the default 343.3m/s. Now, both of the AL_SPEED_OF_SOUND and AL_METERS_PER_UNIT properties are taken into account for speed of sound adjustments to have an appropriate affect on the reverb decay. Consequently, applications that use reverb but don't set these properties properly may find the reverb decay too strong. Setting this to "true" or "1" will revert to the old behavior for those apps and assume the default speed of sound for reverb. openal-soft-openal-soft-1.19.1/docs/hrtf.txt000066400000000000000000000075521335774445300207430ustar00rootroot00000000000000HRTF Support ============ Starting with OpenAL Soft 1.14, HRTFs can be used to enable enhanced spatialization for both 3D (mono) and multi-channel sources, when used with headphones/stereo output. This can be enabled using the 'hrtf' config option. For multi-channel sources this creates a virtual speaker effect, making it sound as if speakers provide a discrete position for each channel around the listener. For mono sources this provides much more versatility in the perceived placement of sounds, making it seem as though they are coming from all around, including above and below the listener, instead of just to the front, back, and sides. The default data set is based on the KEMAR HRTF data provided by MIT, which can be found at . It's only available when using 44100hz or 48000hz playback. Custom HRTF Data Sets ===================== OpenAL Soft also provides an option to use user-specified data sets, in addition to or in place of the default set. This allows users to provide their own data sets, which could be better suited for their heads, or to work with stereo speakers instead of headphones, or to support more playback sample rates, for example. The file format is specified below. It uses little-endian byte order. == ALchar magic[8] = "MinPHR02"; ALuint sampleRate; ALubyte sampleType; /* Can be 0 (16-bit) or 1 (24-bit). */ ALubyte channelType; /* Can be 0 (mono) or 1 (stereo). */ ALubyte hrirSize; /* Can be 8 to 128 in steps of 8. */ ALubyte fdCount; /* Can be 1 to 16. */ struct { ALushort distance; /* Can be 50mm to 2500mm. */ ALubyte evCount; /* Can be 5 to 128. */ ALubyte azCount[evCount]; /* Each can be 1 to 128. */ } fields[fdCount]; /* NOTE: ALtype can be ALshort (16-bit) or ALbyte[3] (24-bit) depending on * sampleType, * hrirCount is the sum of all azCounts. * channels can be 1 (mono) or 2 (stereo) depending on channelType. */ ALtype coefficients[hrirCount][hrirSize][channels]; ALubyte delays[hrirCount][channels]; /* Each can be 0 to 63. */ == The data is described as thus: The file first starts with the 8-byte marker, "MinPHR02", to identify it as an HRTF data set. This is followed by an unsigned 32-bit integer, specifying the sample rate the data set is designed for (OpenAL Soft will not use it if the output device's playback rate doesn't match). Afterward, an unsigned 8-bit integer specifies how many sample points (or finite impulse response filter coefficients) make up each HRIR. The following unsigned 8-bit integer specifies the number of fields used by the data set. Then for each field an unsigned 16-bit short specifies the distance for that field (in millimeters), followed by an 8-bit integer for the number of elevations. These elevations start at the bottom (-90 degrees), and increment upwards. Following this is an array of unsigned 8-bit integers, one for each elevation which specifies the number of azimuths (and thus HRIRs) that make up each elevation. Azimuths start clockwise from the front, constructing a full circle. Mono HRTFs use the same HRIRs for both ears by reversing the azimuth calculation (ie. left = angle, right = 360-angle). The actual coefficients follow. Each coefficient is a signed 16-bit or 24-bit sample. Stereo HRTFs interleave left/right ear coefficients. The HRIRs must be minimum-phase. This allows the use of a smaller filter length, reducing computation. For reference, the default data set uses a 32-point filter while even the smallest data set provided by MIT used a 128-sample filter (a 4x reduction by applying minimum-phase reconstruction). After the coefficients is an array of unsigned 8-bit delay values, one for each HRIR (with stereo HRTFs interleaving left/right ear delays). This is the propagation delay (in samples) a signal must wait before being convolved with the corresponding minimum-phase HRIR filter. openal-soft-openal-soft-1.19.1/examples/000077500000000000000000000000001335774445300201145ustar00rootroot00000000000000openal-soft-openal-soft-1.19.1/examples/alffplay.cpp000066400000000000000000001710621335774445300224250ustar00rootroot00000000000000/* * An example showing how to play a stream sync'd to video, using ffmpeg. * * Requires C++11. */ #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include extern "C" { #include "libavcodec/avcodec.h" #include "libavformat/avformat.h" #include "libavformat/avio.h" #include "libavutil/time.h" #include "libavutil/pixfmt.h" #include "libavutil/avstring.h" #include "libavutil/channel_layout.h" #include "libswscale/swscale.h" #include "libswresample/swresample.h" } #include "SDL.h" #include "AL/alc.h" #include "AL/al.h" #include "AL/alext.h" #include "common/alhelpers.h" extern "C" { /* Undefine this to disable use of experimental extensions. Don't use for * production code! Interfaces and behavior may change prior to being * finalized. */ #define ALLOW_EXPERIMENTAL_EXTS #ifdef ALLOW_EXPERIMENTAL_EXTS #ifndef AL_SOFT_map_buffer #define AL_SOFT_map_buffer 1 typedef unsigned int ALbitfieldSOFT; #define AL_MAP_READ_BIT_SOFT 0x00000001 #define AL_MAP_WRITE_BIT_SOFT 0x00000002 #define AL_MAP_PERSISTENT_BIT_SOFT 0x00000004 #define AL_PRESERVE_DATA_BIT_SOFT 0x00000008 typedef void (AL_APIENTRY*LPALBUFFERSTORAGESOFT)(ALuint buffer, ALenum format, const ALvoid *data, ALsizei size, ALsizei freq, ALbitfieldSOFT flags); typedef void* (AL_APIENTRY*LPALMAPBUFFERSOFT)(ALuint buffer, ALsizei offset, ALsizei length, ALbitfieldSOFT access); typedef void (AL_APIENTRY*LPALUNMAPBUFFERSOFT)(ALuint buffer); typedef void (AL_APIENTRY*LPALFLUSHMAPPEDBUFFERSOFT)(ALuint buffer, ALsizei offset, ALsizei length); #endif #ifndef AL_SOFT_events #define AL_SOFT_events 1 #define AL_EVENT_CALLBACK_FUNCTION_SOFT 0x1220 #define AL_EVENT_CALLBACK_USER_PARAM_SOFT 0x1221 #define AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT 0x1222 #define AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT 0x1223 #define AL_EVENT_TYPE_ERROR_SOFT 0x1224 #define AL_EVENT_TYPE_PERFORMANCE_SOFT 0x1225 #define AL_EVENT_TYPE_DEPRECATED_SOFT 0x1226 #define AL_EVENT_TYPE_DISCONNECTED_SOFT 0x1227 typedef void (AL_APIENTRY*ALEVENTPROCSOFT)(ALenum eventType, ALuint object, ALuint param, ALsizei length, const ALchar *message, void *userParam); typedef void (AL_APIENTRY*LPALEVENTCONTROLSOFT)(ALsizei count, const ALenum *types, ALboolean enable); typedef void (AL_APIENTRY*LPALEVENTCALLBACKSOFT)(ALEVENTPROCSOFT callback, void *userParam); typedef void* (AL_APIENTRY*LPALGETPOINTERSOFT)(ALenum pname); typedef void (AL_APIENTRY*LPALGETPOINTERVSOFT)(ALenum pname, void **values); #endif #endif /* ALLOW_EXPERIMENTAL_EXTS */ } namespace { #ifndef M_PI #define M_PI (3.14159265358979323846) #endif using nanoseconds = std::chrono::nanoseconds; using microseconds = std::chrono::microseconds; using milliseconds = std::chrono::milliseconds; using seconds = std::chrono::seconds; using seconds_d64 = std::chrono::duration; const std::string AppName("alffplay"); bool EnableDirectOut = false; bool EnableWideStereo = false; LPALGETSOURCEI64VSOFT alGetSourcei64vSOFT; LPALCGETINTEGER64VSOFT alcGetInteger64vSOFT; #ifdef AL_SOFT_map_buffer LPALBUFFERSTORAGESOFT alBufferStorageSOFT; LPALMAPBUFFERSOFT alMapBufferSOFT; LPALUNMAPBUFFERSOFT alUnmapBufferSOFT; #endif #ifdef AL_SOFT_events LPALEVENTCONTROLSOFT alEventControlSOFT; LPALEVENTCALLBACKSOFT alEventCallbackSOFT; #endif const seconds AVNoSyncThreshold(10); const milliseconds VideoSyncThreshold(10); #define VIDEO_PICTURE_QUEUE_SIZE 16 const seconds_d64 AudioSyncThreshold(0.03); const milliseconds AudioSampleCorrectionMax(50); /* Averaging filter coefficient for audio sync. */ #define AUDIO_DIFF_AVG_NB 20 const double AudioAvgFilterCoeff = std::pow(0.01, 1.0/AUDIO_DIFF_AVG_NB); /* Per-buffer size, in time */ const milliseconds AudioBufferTime(20); /* Buffer total size, in time (should be divisible by the buffer time) */ const milliseconds AudioBufferTotalTime(800); #define MAX_QUEUE_SIZE (15 * 1024 * 1024) /* Bytes of compressed data to keep queued */ enum { FF_UPDATE_EVENT = SDL_USEREVENT, FF_REFRESH_EVENT, FF_MOVIE_DONE_EVENT }; enum class SyncMaster { Audio, Video, External, Default = External }; inline microseconds get_avtime() { return microseconds(av_gettime()); } /* Define unique_ptrs to auto-cleanup associated ffmpeg objects. */ struct AVIOContextDeleter { void operator()(AVIOContext *ptr) { avio_closep(&ptr); } }; using AVIOContextPtr = std::unique_ptr; struct AVFormatCtxDeleter { void operator()(AVFormatContext *ptr) { avformat_close_input(&ptr); } }; using AVFormatCtxPtr = std::unique_ptr; struct AVCodecCtxDeleter { void operator()(AVCodecContext *ptr) { avcodec_free_context(&ptr); } }; using AVCodecCtxPtr = std::unique_ptr; struct AVFrameDeleter { void operator()(AVFrame *ptr) { av_frame_free(&ptr); } }; using AVFramePtr = std::unique_ptr; struct SwrContextDeleter { void operator()(SwrContext *ptr) { swr_free(&ptr); } }; using SwrContextPtr = std::unique_ptr; struct SwsContextDeleter { void operator()(SwsContext *ptr) { sws_freeContext(ptr); } }; using SwsContextPtr = std::unique_ptr; class PacketQueue { std::deque mPackets; size_t mTotalSize{0}; public: ~PacketQueue() { clear(); } bool empty() const noexcept { return mPackets.empty(); } size_t totalSize() const noexcept { return mTotalSize; } void put(const AVPacket *pkt) { mPackets.push_back(AVPacket{}); if(av_packet_ref(&mPackets.back(), pkt) != 0) mPackets.pop_back(); else mTotalSize += mPackets.back().size; } AVPacket *front() noexcept { return &mPackets.front(); } void pop() { AVPacket *pkt = &mPackets.front(); mTotalSize -= pkt->size; av_packet_unref(pkt); mPackets.pop_front(); } void clear() { for(AVPacket &pkt : mPackets) av_packet_unref(&pkt); mPackets.clear(); mTotalSize = 0; } }; struct MovieState; struct AudioState { MovieState &mMovie; AVStream *mStream{nullptr}; AVCodecCtxPtr mCodecCtx; std::mutex mQueueMtx; std::condition_variable mQueueCond; /* Used for clock difference average computation */ seconds_d64 mClockDiffAvg{0}; /* Time of the next sample to be buffered */ nanoseconds mCurrentPts{0}; /* Device clock time that the stream started at. */ nanoseconds mDeviceStartTime{nanoseconds::min()}; /* Decompressed sample frame, and swresample context for conversion */ AVFramePtr mDecodedFrame; SwrContextPtr mSwresCtx; /* Conversion format, for what gets fed to OpenAL */ int mDstChanLayout{0}; AVSampleFormat mDstSampleFmt{AV_SAMPLE_FMT_NONE}; /* Storage of converted samples */ uint8_t *mSamples{nullptr}; int mSamplesLen{0}; /* In samples */ int mSamplesPos{0}; int mSamplesMax{0}; /* OpenAL format */ ALenum mFormat{AL_NONE}; ALsizei mFrameSize{0}; std::mutex mSrcMutex; std::condition_variable mSrcCond; std::atomic_flag mConnected; ALuint mSource{0}; std::vector mBuffers; ALsizei mBufferIdx{0}; AudioState(MovieState &movie) : mMovie(movie) { mConnected.test_and_set(std::memory_order_relaxed); } ~AudioState() { if(mSource) alDeleteSources(1, &mSource); if(!mBuffers.empty()) alDeleteBuffers(mBuffers.size(), mBuffers.data()); av_freep(&mSamples); } #ifdef AL_SOFT_events static void AL_APIENTRY EventCallback(ALenum eventType, ALuint object, ALuint param, ALsizei length, const ALchar *message, void *userParam); #endif nanoseconds getClockNoLock(); nanoseconds getClock() { std::lock_guard lock(mSrcMutex); return getClockNoLock(); } bool isBufferFilled(); void startPlayback(); int getSync(); int decodeFrame(); bool readAudio(uint8_t *samples, int length); int handler(); }; struct VideoState { MovieState &mMovie; AVStream *mStream{nullptr}; AVCodecCtxPtr mCodecCtx; std::mutex mQueueMtx; std::condition_variable mQueueCond; nanoseconds mClock{0}; nanoseconds mFrameTimer{0}; nanoseconds mFrameLastPts{0}; nanoseconds mFrameLastDelay{0}; nanoseconds mCurrentPts{0}; /* time (av_gettime) at which we updated mCurrentPts - used to have running video pts */ microseconds mCurrentPtsTime{0}; /* Decompressed video frame, and swscale context for conversion */ AVFramePtr mDecodedFrame; SwsContextPtr mSwscaleCtx; struct Picture { SDL_Texture *mImage{nullptr}; int mWidth{0}, mHeight{0}; /* Logical image size (actual size may be larger) */ std::atomic mUpdated{false}; nanoseconds mPts{0}; ~Picture() { if(mImage) SDL_DestroyTexture(mImage); mImage = nullptr; } }; std::array mPictQ; size_t mPictQSize{0}, mPictQRead{0}, mPictQWrite{0}; std::mutex mPictQMutex; std::condition_variable mPictQCond; bool mFirstUpdate{true}; std::atomic mEOS{false}; std::atomic mFinalUpdate{false}; VideoState(MovieState &movie) : mMovie(movie) { } nanoseconds getClock(); bool isBufferFilled(); static Uint32 SDLCALL sdl_refresh_timer_cb(Uint32 interval, void *opaque); void schedRefresh(milliseconds delay); void display(SDL_Window *screen, SDL_Renderer *renderer); void refreshTimer(SDL_Window *screen, SDL_Renderer *renderer); void updatePicture(SDL_Window *screen, SDL_Renderer *renderer); int queuePicture(nanoseconds pts); int handler(); }; struct MovieState { AVIOContextPtr mIOContext; AVFormatCtxPtr mFormatCtx; SyncMaster mAVSyncType{SyncMaster::Default}; microseconds mClockBase{0}; std::atomic mPlaying{false}; std::mutex mSendMtx; std::condition_variable mSendCond; /* NOTE: false/clear = need data, true/set = no data needed */ std::atomic_flag mSendDataGood; std::atomic mQuit{false}; AudioState mAudio; VideoState mVideo; std::thread mParseThread; std::thread mAudioThread; std::thread mVideoThread; std::string mFilename; MovieState(std::string fname) : mAudio(*this), mVideo(*this), mFilename(std::move(fname)) { } ~MovieState() { mQuit = true; if(mParseThread.joinable()) mParseThread.join(); } static int decode_interrupt_cb(void *ctx); bool prepare(); void setTitle(SDL_Window *window); nanoseconds getClock(); nanoseconds getMasterClock(); nanoseconds getDuration(); int streamComponentOpen(int stream_index); int parse_handler(); }; nanoseconds AudioState::getClockNoLock() { // The audio clock is the timestamp of the sample currently being heard. if(alcGetInteger64vSOFT) { // If device start time = min, we aren't playing yet. if(mDeviceStartTime == nanoseconds::min()) return nanoseconds::zero(); // Get the current device clock time and latency. auto device = alcGetContextsDevice(alcGetCurrentContext()); ALCint64SOFT devtimes[2] = {0,0}; alcGetInteger64vSOFT(device, ALC_DEVICE_CLOCK_LATENCY_SOFT, 2, devtimes); auto latency = nanoseconds(devtimes[1]); auto device_time = nanoseconds(devtimes[0]); // The clock is simply the current device time relative to the recorded // start time. We can also subtract the latency to get more a accurate // position of where the audio device actually is in the output stream. return device_time - mDeviceStartTime - latency; } /* The source-based clock is based on 4 components: * 1 - The timestamp of the next sample to buffer (mCurrentPts) * 2 - The length of the source's buffer queue * (AudioBufferTime*AL_BUFFERS_QUEUED) * 3 - The offset OpenAL is currently at in the source (the first value * from AL_SAMPLE_OFFSET_LATENCY_SOFT) * 4 - The latency between OpenAL and the DAC (the second value from * AL_SAMPLE_OFFSET_LATENCY_SOFT) * * Subtracting the length of the source queue from the next sample's * timestamp gives the timestamp of the sample at the start of the source * queue. Adding the source offset to that results in the timestamp for the * sample at OpenAL's current position, and subtracting the source latency * from that gives the timestamp of the sample currently at the DAC. */ nanoseconds pts = mCurrentPts; if(mSource) { ALint64SOFT offset[2]; ALint queued; ALint status; /* NOTE: The source state must be checked last, in case an underrun * occurs and the source stops between retrieving the offset+latency * and getting the state. */ if(alGetSourcei64vSOFT) alGetSourcei64vSOFT(mSource, AL_SAMPLE_OFFSET_LATENCY_SOFT, offset); else { ALint ioffset; alGetSourcei(mSource, AL_SAMPLE_OFFSET, &ioffset); offset[0] = (ALint64SOFT)ioffset << 32; offset[1] = 0; } alGetSourcei(mSource, AL_BUFFERS_QUEUED, &queued); alGetSourcei(mSource, AL_SOURCE_STATE, &status); /* If the source is AL_STOPPED, then there was an underrun and all * buffers are processed, so ignore the source queue. The audio thread * will put the source into an AL_INITIAL state and clear the queue * when it starts recovery. */ if(status != AL_STOPPED) { using fixed32 = std::chrono::duration>; pts -= AudioBufferTime*queued; pts += std::chrono::duration_cast( fixed32(offset[0] / mCodecCtx->sample_rate) ); } /* Don't offset by the latency if the source isn't playing. */ if(status == AL_PLAYING) pts -= nanoseconds(offset[1]); } return std::max(pts, nanoseconds::zero()); } bool AudioState::isBufferFilled() { /* All of OpenAL's buffer queueing happens under the mSrcMutex lock, as * does the source gen. So when we're able to grab the lock and the source * is valid, the queue must be full. */ std::lock_guard lock(mSrcMutex); return mSource != 0; } void AudioState::startPlayback() { alSourcePlay(mSource); if(alcGetInteger64vSOFT) { using fixed32 = std::chrono::duration>; // Subtract the total buffer queue time from the current pts to get the // pts of the start of the queue. nanoseconds startpts = mCurrentPts - AudioBufferTotalTime; int64_t srctimes[2]={0,0}; alGetSourcei64vSOFT(mSource, AL_SAMPLE_OFFSET_CLOCK_SOFT, srctimes); auto device_time = nanoseconds(srctimes[1]); auto src_offset = std::chrono::duration_cast(fixed32(srctimes[0])) / mCodecCtx->sample_rate; // The mixer may have ticked and incremented the device time and sample // offset, so subtract the source offset from the device time to get // the device time the source started at. Also subtract startpts to get // the device time the stream would have started at to reach where it // is now. mDeviceStartTime = device_time - src_offset - startpts; } } int AudioState::getSync() { if(mMovie.mAVSyncType == SyncMaster::Audio) return 0; auto ref_clock = mMovie.getMasterClock(); auto diff = ref_clock - getClockNoLock(); if(!(diff < AVNoSyncThreshold && diff > -AVNoSyncThreshold)) { /* Difference is TOO big; reset accumulated average */ mClockDiffAvg = seconds_d64::zero(); return 0; } /* Accumulate the diffs */ mClockDiffAvg = mClockDiffAvg*AudioAvgFilterCoeff + diff; auto avg_diff = mClockDiffAvg*(1.0 - AudioAvgFilterCoeff); if(avg_diff < AudioSyncThreshold/2.0 && avg_diff > -AudioSyncThreshold) return 0; /* Constrain the per-update difference to avoid exceedingly large skips */ diff = std::min(std::max(diff, -AudioSampleCorrectionMax), AudioSampleCorrectionMax); return (int)std::chrono::duration_cast(diff*mCodecCtx->sample_rate).count(); } int AudioState::decodeFrame() { while(!mMovie.mQuit.load(std::memory_order_relaxed)) { std::unique_lock lock(mQueueMtx); int ret = avcodec_receive_frame(mCodecCtx.get(), mDecodedFrame.get()); if(ret == AVERROR(EAGAIN)) { mMovie.mSendDataGood.clear(std::memory_order_relaxed); std::unique_lock(mMovie.mSendMtx).unlock(); mMovie.mSendCond.notify_one(); do { mQueueCond.wait(lock); ret = avcodec_receive_frame(mCodecCtx.get(), mDecodedFrame.get()); } while(ret == AVERROR(EAGAIN)); } lock.unlock(); if(ret == AVERROR_EOF) break; mMovie.mSendDataGood.clear(std::memory_order_relaxed); mMovie.mSendCond.notify_one(); if(ret < 0) { std::cerr<< "Failed to decode frame: "<nb_samples <= 0) { av_frame_unref(mDecodedFrame.get()); continue; } /* If provided, update w/ pts */ if(mDecodedFrame->best_effort_timestamp != AV_NOPTS_VALUE) mCurrentPts = std::chrono::duration_cast( seconds_d64(av_q2d(mStream->time_base)*mDecodedFrame->best_effort_timestamp) ); if(mDecodedFrame->nb_samples > mSamplesMax) { av_freep(&mSamples); av_samples_alloc( &mSamples, nullptr, mCodecCtx->channels, mDecodedFrame->nb_samples, mDstSampleFmt, 0 ); mSamplesMax = mDecodedFrame->nb_samples; } /* Return the amount of sample frames converted */ int data_size = swr_convert(mSwresCtx.get(), &mSamples, mDecodedFrame->nb_samples, (const uint8_t**)mDecodedFrame->data, mDecodedFrame->nb_samples ); av_frame_unref(mDecodedFrame.get()); return data_size; } return 0; } /* Duplicates the sample at in to out, count times. The frame size is a * multiple of the template type size. */ template static void sample_dup(uint8_t *out, const uint8_t *in, int count, int frame_size) { const T *sample = reinterpret_cast(in); T *dst = reinterpret_cast(out); if(frame_size == sizeof(T)) std::fill_n(dst, count, *sample); else { /* NOTE: frame_size is a multiple of sizeof(T). */ int type_mult = frame_size / sizeof(T); int i = 0; std::generate_n(dst, count*type_mult, [sample,type_mult,&i]() -> T { T ret = sample[i]; i = (i+1)%type_mult; return ret; } ); } } bool AudioState::readAudio(uint8_t *samples, int length) { int sample_skip = getSync(); int audio_size = 0; /* Read the next chunk of data, refill the buffer, and queue it * on the source */ length /= mFrameSize; while(audio_size < length) { if(mSamplesLen <= 0 || mSamplesPos >= mSamplesLen) { int frame_len = decodeFrame(); if(frame_len <= 0) break; mSamplesLen = frame_len; mSamplesPos = std::min(mSamplesLen, sample_skip); sample_skip -= mSamplesPos; // Adjust the device start time and current pts by the amount we're // skipping/duplicating, so that the clock remains correct for the // current stream position. auto skip = nanoseconds(seconds(mSamplesPos)) / mCodecCtx->sample_rate; mDeviceStartTime -= skip; mCurrentPts += skip; continue; } int rem = length - audio_size; if(mSamplesPos >= 0) { int len = mSamplesLen - mSamplesPos; if(rem > len) rem = len; memcpy(samples, mSamples + mSamplesPos*mFrameSize, rem*mFrameSize); } else { rem = std::min(rem, -mSamplesPos); /* Add samples by copying the first sample */ if((mFrameSize&7) == 0) sample_dup(samples, mSamples, rem, mFrameSize); else if((mFrameSize&3) == 0) sample_dup(samples, mSamples, rem, mFrameSize); else if((mFrameSize&1) == 0) sample_dup(samples, mSamples, rem, mFrameSize); else sample_dup(samples, mSamples, rem, mFrameSize); } mSamplesPos += rem; mCurrentPts += nanoseconds(seconds(rem)) / mCodecCtx->sample_rate; samples += rem*mFrameSize; audio_size += rem; } if(audio_size <= 0) return false; if(audio_size < length) { int rem = length - audio_size; std::fill_n(samples, rem*mFrameSize, (mDstSampleFmt == AV_SAMPLE_FMT_U8) ? 0x80 : 0x00); mCurrentPts += nanoseconds(seconds(rem)) / mCodecCtx->sample_rate; audio_size += rem; } return true; } #ifdef AL_SOFT_events void AL_APIENTRY AudioState::EventCallback(ALenum eventType, ALuint object, ALuint param, ALsizei length, const ALchar *message, void *userParam) { AudioState *self = reinterpret_cast(userParam); if(eventType == AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT) { /* Temporarily lock the source mutex to ensure it's not between * checking the processed count and going to sleep. */ std::unique_lock(self->mSrcMutex).unlock(); self->mSrcCond.notify_one(); return; } std::cout<< "\n---- AL Event on AudioState "< lock(self->mSrcMutex); self->mConnected.clear(std::memory_order_release); } std::unique_lock(self->mSrcMutex).unlock(); self->mSrcCond.notify_one(); } } #endif int AudioState::handler() { std::unique_lock lock(mSrcMutex); milliseconds sleep_time = AudioBufferTime / 3; ALenum fmt; #ifdef AL_SOFT_events const std::array evt_types{{ AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT, AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT, AL_EVENT_TYPE_ERROR_SOFT, AL_EVENT_TYPE_PERFORMANCE_SOFT, AL_EVENT_TYPE_DEPRECATED_SOFT, AL_EVENT_TYPE_DISCONNECTED_SOFT }}; if(alEventControlSOFT) { alEventControlSOFT(evt_types.size(), evt_types.data(), AL_TRUE); alEventCallbackSOFT(EventCallback, this); sleep_time = AudioBufferTotalTime; } #endif /* Find a suitable format for OpenAL. */ mDstChanLayout = 0; if(mCodecCtx->sample_fmt == AV_SAMPLE_FMT_U8 || mCodecCtx->sample_fmt == AV_SAMPLE_FMT_U8P) { mDstSampleFmt = AV_SAMPLE_FMT_U8; mFrameSize = 1; if(mCodecCtx->channel_layout == AV_CH_LAYOUT_7POINT1 && alIsExtensionPresent("AL_EXT_MCFORMATS") && (fmt=alGetEnumValue("AL_FORMAT_71CHN8")) != AL_NONE && fmt != -1) { mDstChanLayout = mCodecCtx->channel_layout; mFrameSize *= 8; mFormat = fmt; } if((mCodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1 || mCodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK) && alIsExtensionPresent("AL_EXT_MCFORMATS") && (fmt=alGetEnumValue("AL_FORMAT_51CHN8")) != AL_NONE && fmt != -1) { mDstChanLayout = mCodecCtx->channel_layout; mFrameSize *= 6; mFormat = fmt; } if(mCodecCtx->channel_layout == AV_CH_LAYOUT_MONO) { mDstChanLayout = mCodecCtx->channel_layout; mFrameSize *= 1; mFormat = AL_FORMAT_MONO8; } if(!mDstChanLayout) { mDstChanLayout = AV_CH_LAYOUT_STEREO; mFrameSize *= 2; mFormat = AL_FORMAT_STEREO8; } } if((mCodecCtx->sample_fmt == AV_SAMPLE_FMT_FLT || mCodecCtx->sample_fmt == AV_SAMPLE_FMT_FLTP) && alIsExtensionPresent("AL_EXT_FLOAT32")) { mDstSampleFmt = AV_SAMPLE_FMT_FLT; mFrameSize = 4; if(mCodecCtx->channel_layout == AV_CH_LAYOUT_7POINT1 && alIsExtensionPresent("AL_EXT_MCFORMATS") && (fmt=alGetEnumValue("AL_FORMAT_71CHN32")) != AL_NONE && fmt != -1) { mDstChanLayout = mCodecCtx->channel_layout; mFrameSize *= 8; mFormat = fmt; } if((mCodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1 || mCodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK) && alIsExtensionPresent("AL_EXT_MCFORMATS") && (fmt=alGetEnumValue("AL_FORMAT_51CHN32")) != AL_NONE && fmt != -1) { mDstChanLayout = mCodecCtx->channel_layout; mFrameSize *= 6; mFormat = fmt; } if(mCodecCtx->channel_layout == AV_CH_LAYOUT_MONO) { mDstChanLayout = mCodecCtx->channel_layout; mFrameSize *= 1; mFormat = AL_FORMAT_MONO_FLOAT32; } if(!mDstChanLayout) { mDstChanLayout = AV_CH_LAYOUT_STEREO; mFrameSize *= 2; mFormat = AL_FORMAT_STEREO_FLOAT32; } } if(!mDstChanLayout) { mDstSampleFmt = AV_SAMPLE_FMT_S16; mFrameSize = 2; if(mCodecCtx->channel_layout == AV_CH_LAYOUT_7POINT1 && alIsExtensionPresent("AL_EXT_MCFORMATS") && (fmt=alGetEnumValue("AL_FORMAT_71CHN16")) != AL_NONE && fmt != -1) { mDstChanLayout = mCodecCtx->channel_layout; mFrameSize *= 8; mFormat = fmt; } if((mCodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1 || mCodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK) && alIsExtensionPresent("AL_EXT_MCFORMATS") && (fmt=alGetEnumValue("AL_FORMAT_51CHN16")) != AL_NONE && fmt != -1) { mDstChanLayout = mCodecCtx->channel_layout; mFrameSize *= 6; mFormat = fmt; } if(mCodecCtx->channel_layout == AV_CH_LAYOUT_MONO) { mDstChanLayout = mCodecCtx->channel_layout; mFrameSize *= 1; mFormat = AL_FORMAT_MONO16; } if(!mDstChanLayout) { mDstChanLayout = AV_CH_LAYOUT_STEREO; mFrameSize *= 2; mFormat = AL_FORMAT_STEREO16; } } void *samples = nullptr; ALsizei buffer_len = std::chrono::duration_cast>( mCodecCtx->sample_rate * AudioBufferTime).count() * mFrameSize; mSamples = NULL; mSamplesMax = 0; mSamplesPos = 0; mSamplesLen = 0; mDecodedFrame.reset(av_frame_alloc()); if(!mDecodedFrame) { std::cerr<< "Failed to allocate audio frame" <sample_rate, mCodecCtx->channel_layout ? mCodecCtx->channel_layout : (uint64_t)av_get_default_channel_layout(mCodecCtx->channels), mCodecCtx->sample_fmt, mCodecCtx->sample_rate, 0, nullptr )); if(!mSwresCtx || swr_init(mSwresCtx.get()) != 0) { std::cerr<< "Failed to initialize audio converter" <sample_rate, AL_MAP_WRITE_BIT_SOFT); if(alGetError() != AL_NO_ERROR) { fprintf(stderr, "Failed to use mapped buffers\n"); samples = av_malloc(buffer_len); } } else #endif samples = av_malloc(buffer_len); while(alGetError() == AL_NO_ERROR && !mMovie.mQuit.load(std::memory_order_relaxed) && mConnected.test_and_set(std::memory_order_relaxed)) { /* First remove any processed buffers. */ ALint processed; alGetSourcei(mSource, AL_BUFFERS_PROCESSED, &processed); while(processed > 0) { std::array bids; alSourceUnqueueBuffers(mSource, std::min(bids.size(), processed), bids.data()); processed -= std::min(bids.size(), processed); } /* Refill the buffer queue. */ ALint queued; alGetSourcei(mSource, AL_BUFFERS_QUEUED, &queued); while((ALuint)queued < mBuffers.size()) { ALuint bufid = mBuffers[mBufferIdx]; uint8_t *ptr = reinterpret_cast(samples #ifdef AL_SOFT_map_buffer ? samples : alMapBufferSOFT(bufid, 0, buffer_len, AL_MAP_WRITE_BIT_SOFT) #endif ); if(!ptr) break; /* Read the next chunk of data, filling the buffer, and queue it on * the source */ bool got_audio = readAudio(ptr, buffer_len); #ifdef AL_SOFT_map_buffer if(!samples) alUnmapBufferSOFT(bufid); #endif if(!got_audio) break; if(samples) alBufferData(bufid, mFormat, samples, buffer_len, mCodecCtx->sample_rate); alSourceQueueBuffers(mSource, 1, &bufid); mBufferIdx = (mBufferIdx+1) % mBuffers.size(); ++queued; } if(queued == 0) break; /* Check that the source is playing. */ ALint state; alGetSourcei(mSource, AL_SOURCE_STATE, &state); if(state == AL_STOPPED) { /* AL_STOPPED means there was an underrun. Clear the buffer queue * since this likely means we're late, and rewind the source to get * it back into an AL_INITIAL state. */ alSourceRewind(mSource); alSourcei(mSource, AL_BUFFER, 0); continue; } /* (re)start the source if needed, and wait for a buffer to finish */ if(state != AL_PLAYING && state != AL_PAUSED && mMovie.mPlaying.load(std::memory_order_relaxed)) startPlayback(); mSrcCond.wait_for(lock, sleep_time); } alSourceRewind(mSource); alSourcei(mSource, AL_BUFFER, 0); finish: av_freep(&samples); #ifdef AL_SOFT_events if(alEventControlSOFT) { alEventControlSOFT(evt_types.size(), evt_types.data(), AL_FALSE); alEventCallbackSOFT(nullptr, nullptr); } #endif return 0; } nanoseconds VideoState::getClock() { /* NOTE: This returns incorrect times while not playing. */ auto delta = get_avtime() - mCurrentPtsTime; return mCurrentPts + delta; } bool VideoState::isBufferFilled() { std::unique_lock lock(mPictQMutex); return mPictQSize >= mPictQ.size(); } Uint32 SDLCALL VideoState::sdl_refresh_timer_cb(Uint32 /*interval*/, void *opaque) { SDL_Event evt{}; evt.user.type = FF_REFRESH_EVENT; evt.user.data1 = opaque; SDL_PushEvent(&evt); return 0; /* 0 means stop timer */ } /* Schedules an FF_REFRESH_EVENT event to occur in 'delay' ms. */ void VideoState::schedRefresh(milliseconds delay) { SDL_AddTimer(delay.count(), sdl_refresh_timer_cb, this); } /* Called by VideoState::refreshTimer to display the next video frame. */ void VideoState::display(SDL_Window *screen, SDL_Renderer *renderer) { Picture *vp = &mPictQ[mPictQRead]; if(!vp->mImage) return; float aspect_ratio; int win_w, win_h; int w, h, x, y; if(mCodecCtx->sample_aspect_ratio.num == 0) aspect_ratio = 0.0f; else { aspect_ratio = av_q2d(mCodecCtx->sample_aspect_ratio) * mCodecCtx->width / mCodecCtx->height; } if(aspect_ratio <= 0.0f) aspect_ratio = (float)mCodecCtx->width / (float)mCodecCtx->height; SDL_GetWindowSize(screen, &win_w, &win_h); h = win_h; w = ((int)rint(h * aspect_ratio) + 3) & ~3; if(w > win_w) { w = win_w; h = ((int)rint(w / aspect_ratio) + 3) & ~3; } x = (win_w - w) / 2; y = (win_h - h) / 2; SDL_Rect src_rect{ 0, 0, vp->mWidth, vp->mHeight }; SDL_Rect dst_rect{ x, y, w, h }; SDL_RenderCopy(renderer, vp->mImage, &src_rect, &dst_rect); SDL_RenderPresent(renderer); } /* FF_REFRESH_EVENT handler called on the main thread where the SDL_Renderer * was created. It handles the display of the next decoded video frame (if not * falling behind), and sets up the timer for the following video frame. */ void VideoState::refreshTimer(SDL_Window *screen, SDL_Renderer *renderer) { if(!mStream) { if(mEOS) { mFinalUpdate = true; std::unique_lock(mPictQMutex).unlock(); mPictQCond.notify_all(); return; } schedRefresh(milliseconds(100)); return; } if(!mMovie.mPlaying.load(std::memory_order_relaxed)) { schedRefresh(milliseconds(1)); return; } std::unique_lock lock(mPictQMutex); retry: if(mPictQSize == 0) { if(mEOS) mFinalUpdate = true; else schedRefresh(milliseconds(1)); lock.unlock(); mPictQCond.notify_all(); return; } Picture *vp = &mPictQ[mPictQRead]; mCurrentPts = vp->mPts; mCurrentPtsTime = get_avtime(); /* Get delay using the frame pts and the pts from last frame. */ auto delay = vp->mPts - mFrameLastPts; if(delay <= seconds::zero() || delay >= seconds(1)) { /* If incorrect delay, use previous one. */ delay = mFrameLastDelay; } /* Save for next frame. */ mFrameLastDelay = delay; mFrameLastPts = vp->mPts; /* Update delay to sync to clock if not master source. */ if(mMovie.mAVSyncType != SyncMaster::Video) { auto ref_clock = mMovie.getMasterClock(); auto diff = vp->mPts - ref_clock; /* Skip or repeat the frame. Take delay into account. */ auto sync_threshold = std::min(delay, VideoSyncThreshold); if(!(diff < AVNoSyncThreshold && diff > -AVNoSyncThreshold)) { if(diff <= -sync_threshold) delay = nanoseconds::zero(); else if(diff >= sync_threshold) delay *= 2; } } mFrameTimer += delay; /* Compute the REAL delay. */ auto actual_delay = mFrameTimer - get_avtime(); if(!(actual_delay >= VideoSyncThreshold)) { /* We don't have time to handle this picture, just skip to the next one. */ mPictQRead = (mPictQRead+1)%mPictQ.size(); mPictQSize--; goto retry; } schedRefresh(std::chrono::duration_cast(actual_delay)); /* Show the picture! */ display(screen, renderer); /* Update queue for next picture. */ mPictQRead = (mPictQRead+1)%mPictQ.size(); mPictQSize--; lock.unlock(); mPictQCond.notify_all(); } /* FF_UPDATE_EVENT handler, updates the picture's texture. It's called on the * main thread where the renderer was created. */ void VideoState::updatePicture(SDL_Window *screen, SDL_Renderer *renderer) { Picture *vp = &mPictQ[mPictQWrite]; bool fmt_updated = false; /* allocate or resize the buffer! */ if(!vp->mImage || vp->mWidth != mCodecCtx->width || vp->mHeight != mCodecCtx->height) { fmt_updated = true; if(vp->mImage) SDL_DestroyTexture(vp->mImage); vp->mImage = SDL_CreateTexture( renderer, SDL_PIXELFORMAT_IYUV, SDL_TEXTUREACCESS_STREAMING, mCodecCtx->coded_width, mCodecCtx->coded_height ); if(!vp->mImage) std::cerr<< "Failed to create YV12 texture!" <mWidth = mCodecCtx->width; vp->mHeight = mCodecCtx->height; if(mFirstUpdate && vp->mWidth > 0 && vp->mHeight > 0) { /* For the first update, set the window size to the video size. */ mFirstUpdate = false; int w = vp->mWidth; int h = vp->mHeight; if(mCodecCtx->sample_aspect_ratio.den != 0) { double aspect_ratio = av_q2d(mCodecCtx->sample_aspect_ratio); if(aspect_ratio >= 1.0) w = (int)(w*aspect_ratio + 0.5); else if(aspect_ratio > 0.0) h = (int)(h/aspect_ratio + 0.5); } SDL_SetWindowSize(screen, w, h); } } if(vp->mImage) { AVFrame *frame = mDecodedFrame.get(); void *pixels = nullptr; int pitch = 0; if(mCodecCtx->pix_fmt == AV_PIX_FMT_YUV420P) SDL_UpdateYUVTexture(vp->mImage, nullptr, frame->data[0], frame->linesize[0], frame->data[1], frame->linesize[1], frame->data[2], frame->linesize[2] ); else if(SDL_LockTexture(vp->mImage, nullptr, &pixels, &pitch) != 0) std::cerr<< "Failed to lock texture" <coded_width; int coded_h = mCodecCtx->coded_height; int w = mCodecCtx->width; int h = mCodecCtx->height; if(!mSwscaleCtx || fmt_updated) { mSwscaleCtx.reset(sws_getContext( w, h, mCodecCtx->pix_fmt, w, h, AV_PIX_FMT_YUV420P, 0, nullptr, nullptr, nullptr )); } /* point pict at the queue */ uint8_t *pict_data[3]; pict_data[0] = reinterpret_cast(pixels); pict_data[1] = pict_data[0] + coded_w*coded_h; pict_data[2] = pict_data[1] + coded_w*coded_h/4; int pict_linesize[3]; pict_linesize[0] = pitch; pict_linesize[1] = pitch / 2; pict_linesize[2] = pitch / 2; sws_scale(mSwscaleCtx.get(), (const uint8_t**)frame->data, frame->linesize, 0, h, pict_data, pict_linesize); SDL_UnlockTexture(vp->mImage); } } vp->mUpdated.store(true, std::memory_order_release); std::unique_lock(mPictQMutex).unlock(); mPictQCond.notify_one(); } int VideoState::queuePicture(nanoseconds pts) { /* Wait until we have space for a new pic */ std::unique_lock lock(mPictQMutex); while(mPictQSize >= mPictQ.size() && !mMovie.mQuit.load(std::memory_order_relaxed)) mPictQCond.wait(lock); lock.unlock(); if(mMovie.mQuit.load(std::memory_order_relaxed)) return -1; Picture *vp = &mPictQ[mPictQWrite]; /* We have to create/update the picture in the main thread */ vp->mUpdated.store(false, std::memory_order_relaxed); SDL_Event evt{}; evt.user.type = FF_UPDATE_EVENT; evt.user.data1 = this; SDL_PushEvent(&evt); /* Wait until the picture is updated. */ lock.lock(); while(!vp->mUpdated.load(std::memory_order_relaxed)) { if(mMovie.mQuit.load(std::memory_order_relaxed)) return -1; mPictQCond.wait(lock); } if(mMovie.mQuit.load(std::memory_order_relaxed)) return -1; vp->mPts = pts; mPictQWrite = (mPictQWrite+1)%mPictQ.size(); mPictQSize++; lock.unlock(); return 0; } int VideoState::handler() { mDecodedFrame.reset(av_frame_alloc()); while(!mMovie.mQuit.load(std::memory_order_relaxed)) { std::unique_lock lock(mQueueMtx); /* Decode video frame */ int ret = avcodec_receive_frame(mCodecCtx.get(), mDecodedFrame.get()); if(ret == AVERROR(EAGAIN)) { mMovie.mSendDataGood.clear(std::memory_order_relaxed); std::unique_lock(mMovie.mSendMtx).unlock(); mMovie.mSendCond.notify_one(); do { mQueueCond.wait(lock); ret = avcodec_receive_frame(mCodecCtx.get(), mDecodedFrame.get()); } while(ret == AVERROR(EAGAIN)); } lock.unlock(); if(ret == AVERROR_EOF) break; mMovie.mSendDataGood.clear(std::memory_order_relaxed); mMovie.mSendCond.notify_one(); if(ret < 0) { std::cerr<< "Failed to decode frame: "<best_effort_timestamp != AV_NOPTS_VALUE) mClock = std::chrono::duration_cast( seconds_d64(av_q2d(mStream->time_base)*mDecodedFrame->best_effort_timestamp) ); pts = mClock; /* Update the video clock to the next expected PTS. */ auto frame_delay = av_q2d(mCodecCtx->time_base); frame_delay += mDecodedFrame->repeat_pict * (frame_delay * 0.5); mClock += std::chrono::duration_cast(seconds_d64(frame_delay)); if(queuePicture(pts) < 0) break; av_frame_unref(mDecodedFrame.get()); } mEOS = true; std::unique_lock lock(mPictQMutex); if(mMovie.mQuit.load(std::memory_order_relaxed)) { mPictQRead = 0; mPictQWrite = 0; mPictQSize = 0; } while(!mFinalUpdate) mPictQCond.wait(lock); return 0; } int MovieState::decode_interrupt_cb(void *ctx) { return reinterpret_cast(ctx)->mQuit.load(std::memory_order_relaxed); } bool MovieState::prepare() { AVIOContext *avioctx = nullptr; AVIOInterruptCB intcb = { decode_interrupt_cb, this }; if(avio_open2(&avioctx, mFilename.c_str(), AVIO_FLAG_READ, &intcb, nullptr)) { std::cerr<< "Failed to open "<pb = mIOContext.get(); fmtctx->interrupt_callback = intcb; if(avformat_open_input(&fmtctx, mFilename.c_str(), nullptr, nullptr) != 0) { std::cerr<< "Failed to open "<>(mFormatCtx->duration); } int MovieState::streamComponentOpen(int stream_index) { if(stream_index < 0 || (unsigned int)stream_index >= mFormatCtx->nb_streams) return -1; /* Get a pointer to the codec context for the stream, and open the * associated codec. */ AVCodecCtxPtr avctx(avcodec_alloc_context3(nullptr)); if(!avctx) return -1; if(avcodec_parameters_to_context(avctx.get(), mFormatCtx->streams[stream_index]->codecpar)) return -1; AVCodec *codec = avcodec_find_decoder(avctx->codec_id); if(!codec || avcodec_open2(avctx.get(), codec, nullptr) < 0) { std::cerr<< "Unsupported codec: "<codec_id) << " (0x"<codec_id<codec_type) { case AVMEDIA_TYPE_AUDIO: mAudio.mStream = mFormatCtx->streams[stream_index]; mAudio.mCodecCtx = std::move(avctx); mAudioThread = std::thread(std::mem_fn(&AudioState::handler), &mAudio); break; case AVMEDIA_TYPE_VIDEO: mVideo.mStream = mFormatCtx->streams[stream_index]; mVideo.mCodecCtx = std::move(avctx); mVideoThread = std::thread(std::mem_fn(&VideoState::handler), &mVideo); break; default: return -1; } return stream_index; } int MovieState::parse_handler() { int video_index = -1; int audio_index = -1; /* Dump information about file onto standard error */ av_dump_format(mFormatCtx.get(), 0, mFilename.c_str(), 0); /* Find the first video and audio streams */ for(unsigned int i = 0;i < mFormatCtx->nb_streams;i++) { auto codecpar = mFormatCtx->streams[i]->codecpar; if(codecpar->codec_type == AVMEDIA_TYPE_VIDEO && video_index < 0) video_index = streamComponentOpen(i); else if(codecpar->codec_type == AVMEDIA_TYPE_AUDIO && audio_index < 0) audio_index = streamComponentOpen(i); } if(video_index < 0 && audio_index < 0) { std::cerr<< mFilename<<": could not open codecs" < lock(mAudio.mQueueMtx); int ret; do { ret = avcodec_send_packet(mAudio.mCodecCtx.get(), audio_queue.front()); if(ret != AVERROR(EAGAIN)) audio_queue.pop(); } while(ret != AVERROR(EAGAIN) && !audio_queue.empty()); lock.unlock(); mAudio.mQueueCond.notify_one(); } if(!video_queue.empty()) { std::unique_lock lock(mVideo.mQueueMtx); int ret; do { ret = avcodec_send_packet(mVideo.mCodecCtx.get(), video_queue.front()); if(ret != AVERROR(EAGAIN)) video_queue.pop(); } while(ret != AVERROR(EAGAIN) && !video_queue.empty()); lock.unlock(); mVideo.mQueueCond.notify_one(); } /* If the queues are completely empty, or it's not full and there's * more input to read, go get more. */ size_t queue_size = audio_queue.totalSize() + video_queue.totalSize(); if(queue_size == 0 || (queue_size < MAX_QUEUE_SIZE && !input_finished)) break; if(!mPlaying.load(std::memory_order_relaxed)) { if((!mAudio.mCodecCtx || mAudio.isBufferFilled()) && (!mVideo.mCodecCtx || mVideo.isBufferFilled())) { /* Set the base time 50ms ahead of the current av time. */ mClockBase = get_avtime() + milliseconds(50); mVideo.mCurrentPtsTime = mClockBase; mVideo.mFrameTimer = mVideo.mCurrentPtsTime; mAudio.startPlayback(); mPlaying.store(std::memory_order_release); } } /* Nothing to send or get for now, wait a bit and try again. */ { std::unique_lock lock(mSendMtx); if(mSendDataGood.test_and_set(std::memory_order_relaxed)) mSendCond.wait_for(lock, milliseconds(10)); } } while(!mQuit.load(std::memory_order_relaxed)); } /* Pass a null packet to finish the send buffers (the receive functions * will get AVERROR_EOF when emptied). */ if(mVideo.mCodecCtx) { { std::lock_guard lock(mVideo.mQueueMtx); avcodec_send_packet(mVideo.mCodecCtx.get(), nullptr); } mVideo.mQueueCond.notify_one(); } if(mAudio.mCodecCtx) { { std::lock_guard lock(mAudio.mQueueMtx); avcodec_send_packet(mAudio.mCodecCtx.get(), nullptr); } mAudio.mQueueCond.notify_one(); } video_queue.clear(); audio_queue.clear(); /* all done - wait for it */ if(mVideoThread.joinable()) mVideoThread.join(); if(mAudioThread.joinable()) mAudioThread.join(); mVideo.mEOS = true; std::unique_lock lock(mVideo.mPictQMutex); while(!mVideo.mFinalUpdate) mVideo.mPictQCond.wait(lock); lock.unlock(); SDL_Event evt{}; evt.user.type = FF_MOVIE_DONE_EVENT; SDL_PushEvent(&evt); return 0; } // Helper class+method to print the time with human-readable formatting. struct PrettyTime { seconds mTime; }; inline std::ostream &operator<<(std::ostream &os, const PrettyTime &rhs) { using hours = std::chrono::hours; using minutes = std::chrono::minutes; using std::chrono::duration_cast; seconds t = rhs.mTime; if(t.count() < 0) { os << '-'; t *= -1; } // Only handle up to hour formatting if(t >= hours(1)) os << duration_cast(t).count() << 'h' << std::setfill('0') << std::setw(2) << (duration_cast(t).count() % 60) << 'm'; else os << duration_cast(t).count() << 'm' << std::setfill('0'); os << std::setw(2) << (duration_cast(t).count() % 60) << 's' << std::setw(0) << std::setfill(' '); return os; } } // namespace int main(int argc, char *argv[]) { std::unique_ptr movState; if(argc < 2) { std::cerr<< "Usage: "<] [-direct] " <( alcGetProcAddress(device, "alcGetInteger64vSOFT") ); } } if(alIsExtensionPresent("AL_SOFT_source_latency")) { std::cout<< "Found AL_SOFT_source_latency" <( alGetProcAddress("alGetSourcei64vSOFT") ); } #ifdef AL_SOFT_map_buffer if(alIsExtensionPresent("AL_SOFTX_map_buffer")) { std::cout<< "Found AL_SOFT_map_buffer" <( alGetProcAddress("alBufferStorageSOFT")); alMapBufferSOFT = reinterpret_cast( alGetProcAddress("alMapBufferSOFT")); alUnmapBufferSOFT = reinterpret_cast( alGetProcAddress("alUnmapBufferSOFT")); } #endif #ifdef AL_SOFT_events if(alIsExtensionPresent("AL_SOFTX_events")) { std::cout<< "Found AL_SOFT_events" <( alGetProcAddress("alEventControlSOFT")); alEventCallbackSOFT = reinterpret_cast( alGetProcAddress("alEventCallbackSOFT")); } #endif int fileidx = 0; for(;fileidx < argc;++fileidx) { if(strcmp(argv[fileidx], "-direct") == 0) { if(!alIsExtensionPresent("AL_SOFT_direct_channels")) std::cerr<< "AL_SOFT_direct_channels not supported for direct output" <(new MovieState(argv[fileidx++])); if(!movState->prepare()) movState = nullptr; } if(!movState) { std::cerr<< "Could not start a video" <setTitle(screen); /* Default to going to the next movie at the end of one. */ enum class EomAction { Next, Quit } eom_action = EomAction::Next; seconds last_time(-1); SDL_Event event; while(1) { int have_evt = SDL_WaitEventTimeout(&event, 10); auto cur_time = std::chrono::duration_cast(movState->getMasterClock()); if(cur_time != last_time) { auto end_time = std::chrono::duration_cast(movState->getDuration()); std::cout<< "\r "<mQuit = true; eom_action = EomAction::Quit; break; case SDLK_n: movState->mQuit = true; eom_action = EomAction::Next; break; default: break; } break; case SDL_WINDOWEVENT: switch(event.window.event) { case SDL_WINDOWEVENT_RESIZED: SDL_SetRenderDrawColor(renderer, 0, 0, 0, 255); SDL_RenderFillRect(renderer, nullptr); break; default: break; } break; case SDL_QUIT: movState->mQuit = true; eom_action = EomAction::Quit; break; case FF_UPDATE_EVENT: reinterpret_cast(event.user.data1)->updatePicture( screen, renderer ); break; case FF_REFRESH_EVENT: reinterpret_cast(event.user.data1)->refreshTimer( screen, renderer ); break; case FF_MOVIE_DONE_EVENT: std::cout<<'\n'; last_time = seconds(-1); if(eom_action != EomAction::Quit) { movState = nullptr; while(fileidx < argc && !movState) { movState = std::unique_ptr(new MovieState(argv[fileidx++])); if(!movState->prepare()) movState = nullptr; } if(movState) { movState->setTitle(screen); break; } } /* Nothing more to play. Shut everything down and quit. */ movState = nullptr; CloseAL(); SDL_DestroyRenderer(renderer); renderer = nullptr; SDL_DestroyWindow(screen); screen = nullptr; SDL_Quit(); exit(0); default: break; } } std::cerr<< "SDL_WaitEvent error - "< * * Permission is hereby granted, free of charge, to any person obtaining a copy * of this software and associated documentation files (the "Software"), to deal * in the Software without restriction, including without limitation the rights * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell * copies of the Software, and to permit persons to whom the Software is * furnished to do so, subject to the following conditions: * * The above copyright notice and this permission notice shall be included in * all copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN * THE SOFTWARE. */ /* This file contains an example for selecting an HRTF. */ #include #include #include #include #include "AL/al.h" #include "AL/alc.h" #include "AL/alext.h" #include "common/alhelpers.h" #ifndef M_PI #define M_PI (3.14159265358979323846) #endif static LPALCGETSTRINGISOFT alcGetStringiSOFT; static LPALCRESETDEVICESOFT alcResetDeviceSOFT; /* LoadBuffer loads the named audio file into an OpenAL buffer object, and * returns the new buffer ID. */ static ALuint LoadSound(const char *filename) { Sound_Sample *sample; ALenum err, format; ALuint buffer; Uint32 slen; /* Open the audio file */ sample = Sound_NewSampleFromFile(filename, NULL, 65536); if(!sample) { fprintf(stderr, "Could not open audio in %s\n", filename); return 0; } /* Get the sound format, and figure out the OpenAL format */ if(sample->actual.channels == 1) { if(sample->actual.format == AUDIO_U8) format = AL_FORMAT_MONO8; else if(sample->actual.format == AUDIO_S16SYS) format = AL_FORMAT_MONO16; else { fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format); Sound_FreeSample(sample); return 0; } } else if(sample->actual.channels == 2) { if(sample->actual.format == AUDIO_U8) format = AL_FORMAT_STEREO8; else if(sample->actual.format == AUDIO_S16SYS) format = AL_FORMAT_STEREO16; else { fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format); Sound_FreeSample(sample); return 0; } } else { fprintf(stderr, "Unsupported channel count: %d\n", sample->actual.channels); Sound_FreeSample(sample); return 0; } /* Decode the whole audio stream to a buffer. */ slen = Sound_DecodeAll(sample); if(!sample->buffer || slen == 0) { fprintf(stderr, "Failed to read audio from %s\n", filename); Sound_FreeSample(sample); return 0; } /* Buffer the audio data into a new buffer object, then free the data and * close the file. */ buffer = 0; alGenBuffers(1, &buffer); alBufferData(buffer, format, sample->buffer, slen, sample->actual.rate); Sound_FreeSample(sample); /* Check if an error occured, and clean up if so. */ err = alGetError(); if(err != AL_NO_ERROR) { fprintf(stderr, "OpenAL Error: %s\n", alGetString(err)); if(buffer && alIsBuffer(buffer)) alDeleteBuffers(1, &buffer); return 0; } return buffer; } int main(int argc, char **argv) { ALCdevice *device; ALboolean has_angle_ext; ALuint source, buffer; const char *soundname; const char *hrtfname; ALCint hrtf_state; ALCint num_hrtf; ALdouble angle; ALenum state; /* Print out usage if no arguments were specified */ if(argc < 2) { fprintf(stderr, "Usage: %s [-device ] [-hrtf ] \n", argv[0]); return 1; } /* Initialize OpenAL, and check for HRTF support. */ argv++; argc--; if(InitAL(&argv, &argc) != 0) return 1; device = alcGetContextsDevice(alcGetCurrentContext()); if(!alcIsExtensionPresent(device, "ALC_SOFT_HRTF")) { fprintf(stderr, "Error: ALC_SOFT_HRTF not supported\n"); CloseAL(); return 1; } /* Define a macro to help load the function pointers. */ #define LOAD_PROC(d, x) ((x) = alcGetProcAddress((d), #x)) LOAD_PROC(device, alcGetStringiSOFT); LOAD_PROC(device, alcResetDeviceSOFT); #undef LOAD_PROC /* Check for the AL_EXT_STEREO_ANGLES extension to be able to also rotate * stereo sources. */ has_angle_ext = alIsExtensionPresent("AL_EXT_STEREO_ANGLES"); printf("AL_EXT_STEREO_ANGLES%s found\n", has_angle_ext?"":" not"); /* Check for user-preferred HRTF */ if(strcmp(argv[0], "-hrtf") == 0) { hrtfname = argv[1]; soundname = argv[2]; } else { hrtfname = NULL; soundname = argv[0]; } /* Enumerate available HRTFs, and reset the device using one. */ alcGetIntegerv(device, ALC_NUM_HRTF_SPECIFIERS_SOFT, 1, &num_hrtf); if(!num_hrtf) printf("No HRTFs found\n"); else { ALCint attr[5]; ALCint index = -1; ALCint i; printf("Available HRTFs:\n"); for(i = 0;i < num_hrtf;i++) { const ALCchar *name = alcGetStringiSOFT(device, ALC_HRTF_SPECIFIER_SOFT, i); printf(" %d: %s\n", i, name); /* Check if this is the HRTF the user requested. */ if(hrtfname && strcmp(name, hrtfname) == 0) index = i; } i = 0; attr[i++] = ALC_HRTF_SOFT; attr[i++] = ALC_TRUE; if(index == -1) { if(hrtfname) printf("HRTF \"%s\" not found\n", hrtfname); printf("Using default HRTF...\n"); } else { printf("Selecting HRTF %d...\n", index); attr[i++] = ALC_HRTF_ID_SOFT; attr[i++] = index; } attr[i] = 0; if(!alcResetDeviceSOFT(device, attr)) printf("Failed to reset device: %s\n", alcGetString(device, alcGetError(device))); } /* Check if HRTF is enabled, and show which is being used. */ alcGetIntegerv(device, ALC_HRTF_SOFT, 1, &hrtf_state); if(!hrtf_state) printf("HRTF not enabled!\n"); else { const ALchar *name = alcGetString(device, ALC_HRTF_SPECIFIER_SOFT); printf("HRTF enabled, using %s\n", name); } fflush(stdout); /* Initialize SDL_sound. */ Sound_Init(); /* Load the sound into a buffer. */ buffer = LoadSound(soundname); if(!buffer) { Sound_Quit(); CloseAL(); return 1; } /* Create the source to play the sound with. */ source = 0; alGenSources(1, &source); alSourcei(source, AL_SOURCE_RELATIVE, AL_TRUE); alSource3f(source, AL_POSITION, 0.0f, 0.0f, -1.0f); alSourcei(source, AL_BUFFER, buffer); assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source"); /* Play the sound until it finishes. */ angle = 0.0; alSourcePlay(source); do { al_nssleep(10000000); /* Rotate the source around the listener by about 1/4 cycle per second, * and keep it within -pi...+pi. */ angle += 0.01 * M_PI * 0.5; if(angle > M_PI) angle -= M_PI*2.0; /* This only rotates mono sounds. */ alSource3f(source, AL_POSITION, (ALfloat)sin(angle), 0.0f, -(ALfloat)cos(angle)); if(has_angle_ext) { /* This rotates stereo sounds with the AL_EXT_STEREO_ANGLES * extension. Angles are specified counter-clockwise in radians. */ ALfloat angles[2] = { (ALfloat)(M_PI/6.0 - angle), (ALfloat)(-M_PI/6.0 - angle) }; alSourcefv(source, AL_STEREO_ANGLES, angles); } alGetSourcei(source, AL_SOURCE_STATE, &state); } while(alGetError() == AL_NO_ERROR && state == AL_PLAYING); /* All done. Delete resources, and close down SDL_sound and OpenAL. */ alDeleteSources(1, &source); alDeleteBuffers(1, &buffer); Sound_Quit(); CloseAL(); return 0; } openal-soft-openal-soft-1.19.1/examples/allatency.c000066400000000000000000000150371335774445300222420ustar00rootroot00000000000000/* * OpenAL Source Latency Example * * Copyright (c) 2012 by Chris Robinson * * Permission is hereby granted, free of charge, to any person obtaining a copy * of this software and associated documentation files (the "Software"), to deal * in the Software without restriction, including without limitation the rights * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell * copies of the Software, and to permit persons to whom the Software is * furnished to do so, subject to the following conditions: * * The above copyright notice and this permission notice shall be included in * all copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN * THE SOFTWARE. */ /* This file contains an example for checking the latency of a sound. */ #include #include #include #include "AL/al.h" #include "AL/alc.h" #include "AL/alext.h" #include "common/alhelpers.h" static LPALSOURCEDSOFT alSourcedSOFT; static LPALSOURCE3DSOFT alSource3dSOFT; static LPALSOURCEDVSOFT alSourcedvSOFT; static LPALGETSOURCEDSOFT alGetSourcedSOFT; static LPALGETSOURCE3DSOFT alGetSource3dSOFT; static LPALGETSOURCEDVSOFT alGetSourcedvSOFT; static LPALSOURCEI64SOFT alSourcei64SOFT; static LPALSOURCE3I64SOFT alSource3i64SOFT; static LPALSOURCEI64VSOFT alSourcei64vSOFT; static LPALGETSOURCEI64SOFT alGetSourcei64SOFT; static LPALGETSOURCE3I64SOFT alGetSource3i64SOFT; static LPALGETSOURCEI64VSOFT alGetSourcei64vSOFT; /* LoadBuffer loads the named audio file into an OpenAL buffer object, and * returns the new buffer ID. */ static ALuint LoadSound(const char *filename) { Sound_Sample *sample; ALenum err, format; ALuint buffer; Uint32 slen; /* Open the audio file */ sample = Sound_NewSampleFromFile(filename, NULL, 65536); if(!sample) { fprintf(stderr, "Could not open audio in %s\n", filename); return 0; } /* Get the sound format, and figure out the OpenAL format */ if(sample->actual.channels == 1) { if(sample->actual.format == AUDIO_U8) format = AL_FORMAT_MONO8; else if(sample->actual.format == AUDIO_S16SYS) format = AL_FORMAT_MONO16; else { fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format); Sound_FreeSample(sample); return 0; } } else if(sample->actual.channels == 2) { if(sample->actual.format == AUDIO_U8) format = AL_FORMAT_STEREO8; else if(sample->actual.format == AUDIO_S16SYS) format = AL_FORMAT_STEREO16; else { fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format); Sound_FreeSample(sample); return 0; } } else { fprintf(stderr, "Unsupported channel count: %d\n", sample->actual.channels); Sound_FreeSample(sample); return 0; } /* Decode the whole audio stream to a buffer. */ slen = Sound_DecodeAll(sample); if(!sample->buffer || slen == 0) { fprintf(stderr, "Failed to read audio from %s\n", filename); Sound_FreeSample(sample); return 0; } /* Buffer the audio data into a new buffer object, then free the data and * close the file. */ buffer = 0; alGenBuffers(1, &buffer); alBufferData(buffer, format, sample->buffer, slen, sample->actual.rate); Sound_FreeSample(sample); /* Check if an error occured, and clean up if so. */ err = alGetError(); if(err != AL_NO_ERROR) { fprintf(stderr, "OpenAL Error: %s\n", alGetString(err)); if(buffer && alIsBuffer(buffer)) alDeleteBuffers(1, &buffer); return 0; } return buffer; } int main(int argc, char **argv) { ALuint source, buffer; ALdouble offsets[2]; ALenum state; /* Print out usage if no arguments were specified */ if(argc < 2) { fprintf(stderr, "Usage: %s [-device ] \n", argv[0]); return 1; } /* Initialize OpenAL, and check for source_latency support. */ argv++; argc--; if(InitAL(&argv, &argc) != 0) return 1; if(!alIsExtensionPresent("AL_SOFT_source_latency")) { fprintf(stderr, "Error: AL_SOFT_source_latency not supported\n"); CloseAL(); return 1; } /* Define a macro to help load the function pointers. */ #define LOAD_PROC(x) ((x) = alGetProcAddress(#x)) LOAD_PROC(alSourcedSOFT); LOAD_PROC(alSource3dSOFT); LOAD_PROC(alSourcedvSOFT); LOAD_PROC(alGetSourcedSOFT); LOAD_PROC(alGetSource3dSOFT); LOAD_PROC(alGetSourcedvSOFT); LOAD_PROC(alSourcei64SOFT); LOAD_PROC(alSource3i64SOFT); LOAD_PROC(alSourcei64vSOFT); LOAD_PROC(alGetSourcei64SOFT); LOAD_PROC(alGetSource3i64SOFT); LOAD_PROC(alGetSourcei64vSOFT); #undef LOAD_PROC /* Initialize SDL_sound. */ Sound_Init(); /* Load the sound into a buffer. */ buffer = LoadSound(argv[0]); if(!buffer) { Sound_Quit(); CloseAL(); return 1; } /* Create the source to play the sound with. */ source = 0; alGenSources(1, &source); alSourcei(source, AL_BUFFER, buffer); assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source"); /* Play the sound until it finishes. */ alSourcePlay(source); do { al_nssleep(10000000); alGetSourcei(source, AL_SOURCE_STATE, &state); /* Get the source offset and latency. AL_SEC_OFFSET_LATENCY_SOFT will * place the offset (in seconds) in offsets[0], and the time until that * offset will be heard (in seconds) in offsets[1]. */ alGetSourcedvSOFT(source, AL_SEC_OFFSET_LATENCY_SOFT, offsets); printf("\rOffset: %f - Latency:%3u ms ", offsets[0], (ALuint)(offsets[1]*1000)); fflush(stdout); } while(alGetError() == AL_NO_ERROR && state == AL_PLAYING); printf("\n"); /* All done. Delete resources, and close down SDL_sound and OpenAL. */ alDeleteSources(1, &source); alDeleteBuffers(1, &buffer); Sound_Quit(); CloseAL(); return 0; } openal-soft-openal-soft-1.19.1/examples/alloopback.c000066400000000000000000000202261335774445300223710ustar00rootroot00000000000000/* * OpenAL Loopback Example * * Copyright (c) 2013 by Chris Robinson * * Permission is hereby granted, free of charge, to any person obtaining a copy * of this software and associated documentation files (the "Software"), to deal * in the Software without restriction, including without limitation the rights * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell * copies of the Software, and to permit persons to whom the Software is * furnished to do so, subject to the following conditions: * * The above copyright notice and this permission notice shall be included in * all copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN * THE SOFTWARE. */ /* This file contains an example for using the loopback device for custom * output handling. */ #include #include #include #include #include "AL/al.h" #include "AL/alc.h" #include "AL/alext.h" #include "common/alhelpers.h" #ifndef SDL_AUDIO_MASK_BITSIZE #define SDL_AUDIO_MASK_BITSIZE (0xFF) #endif #ifndef SDL_AUDIO_BITSIZE #define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE) #endif #ifndef M_PI #define M_PI (3.14159265358979323846) #endif typedef struct { ALCdevice *Device; ALCcontext *Context; ALCsizei FrameSize; } PlaybackInfo; static LPALCLOOPBACKOPENDEVICESOFT alcLoopbackOpenDeviceSOFT; static LPALCISRENDERFORMATSUPPORTEDSOFT alcIsRenderFormatSupportedSOFT; static LPALCRENDERSAMPLESSOFT alcRenderSamplesSOFT; void SDLCALL RenderSDLSamples(void *userdata, Uint8 *stream, int len) { PlaybackInfo *playback = (PlaybackInfo*)userdata; alcRenderSamplesSOFT(playback->Device, stream, len/playback->FrameSize); } static const char *ChannelsName(ALCenum chans) { switch(chans) { case ALC_MONO_SOFT: return "Mono"; case ALC_STEREO_SOFT: return "Stereo"; case ALC_QUAD_SOFT: return "Quadraphonic"; case ALC_5POINT1_SOFT: return "5.1 Surround"; case ALC_6POINT1_SOFT: return "6.1 Surround"; case ALC_7POINT1_SOFT: return "7.1 Surround"; } return "Unknown Channels"; } static const char *TypeName(ALCenum type) { switch(type) { case ALC_BYTE_SOFT: return "S8"; case ALC_UNSIGNED_BYTE_SOFT: return "U8"; case ALC_SHORT_SOFT: return "S16"; case ALC_UNSIGNED_SHORT_SOFT: return "U16"; case ALC_INT_SOFT: return "S32"; case ALC_UNSIGNED_INT_SOFT: return "U32"; case ALC_FLOAT_SOFT: return "Float32"; } return "Unknown Type"; } /* Creates a one second buffer containing a sine wave, and returns the new * buffer ID. */ static ALuint CreateSineWave(void) { ALshort data[44100*4]; ALuint buffer; ALenum err; ALuint i; for(i = 0;i < 44100*4;i++) data[i] = (ALshort)(sin(i/44100.0 * 1000.0 * 2.0*M_PI) * 32767.0); /* Buffer the audio data into a new buffer object. */ buffer = 0; alGenBuffers(1, &buffer); alBufferData(buffer, AL_FORMAT_MONO16, data, sizeof(data), 44100); /* Check if an error occured, and clean up if so. */ err = alGetError(); if(err != AL_NO_ERROR) { fprintf(stderr, "OpenAL Error: %s\n", alGetString(err)); if(alIsBuffer(buffer)) alDeleteBuffers(1, &buffer); return 0; } return buffer; } int main(int argc, char *argv[]) { PlaybackInfo playback = { NULL, NULL, 0 }; SDL_AudioSpec desired, obtained; ALuint source, buffer; ALCint attrs[16]; ALenum state; (void)argc; (void)argv; /* Print out error if extension is missing. */ if(!alcIsExtensionPresent(NULL, "ALC_SOFT_loopback")) { fprintf(stderr, "Error: ALC_SOFT_loopback not supported!\n"); return 1; } /* Define a macro to help load the function pointers. */ #define LOAD_PROC(x) ((x) = alcGetProcAddress(NULL, #x)) LOAD_PROC(alcLoopbackOpenDeviceSOFT); LOAD_PROC(alcIsRenderFormatSupportedSOFT); LOAD_PROC(alcRenderSamplesSOFT); #undef LOAD_PROC if(SDL_Init(SDL_INIT_AUDIO) == -1) { fprintf(stderr, "Failed to init SDL audio: %s\n", SDL_GetError()); return 1; } /* Set up SDL audio with our requested format and callback. */ desired.channels = 2; desired.format = AUDIO_S16SYS; desired.freq = 44100; desired.padding = 0; desired.samples = 4096; desired.callback = RenderSDLSamples; desired.userdata = &playback; if(SDL_OpenAudio(&desired, &obtained) != 0) { SDL_Quit(); fprintf(stderr, "Failed to open SDL audio: %s\n", SDL_GetError()); return 1; } /* Set up our OpenAL attributes based on what we got from SDL. */ attrs[0] = ALC_FORMAT_CHANNELS_SOFT; if(obtained.channels == 1) attrs[1] = ALC_MONO_SOFT; else if(obtained.channels == 2) attrs[1] = ALC_STEREO_SOFT; else { fprintf(stderr, "Unhandled SDL channel count: %d\n", obtained.channels); goto error; } attrs[2] = ALC_FORMAT_TYPE_SOFT; if(obtained.format == AUDIO_U8) attrs[3] = ALC_UNSIGNED_BYTE_SOFT; else if(obtained.format == AUDIO_S8) attrs[3] = ALC_BYTE_SOFT; else if(obtained.format == AUDIO_U16SYS) attrs[3] = ALC_UNSIGNED_SHORT_SOFT; else if(obtained.format == AUDIO_S16SYS) attrs[3] = ALC_SHORT_SOFT; else { fprintf(stderr, "Unhandled SDL format: 0x%04x\n", obtained.format); goto error; } attrs[4] = ALC_FREQUENCY; attrs[5] = obtained.freq; attrs[6] = 0; /* end of list */ playback.FrameSize = obtained.channels * SDL_AUDIO_BITSIZE(obtained.format) / 8; /* Initialize OpenAL loopback device, using our format attributes. */ playback.Device = alcLoopbackOpenDeviceSOFT(NULL); if(!playback.Device) { fprintf(stderr, "Failed to open loopback device!\n"); goto error; } /* Make sure the format is supported before setting them on the device. */ if(alcIsRenderFormatSupportedSOFT(playback.Device, attrs[5], attrs[1], attrs[3]) == ALC_FALSE) { fprintf(stderr, "Render format not supported: %s, %s, %dhz\n", ChannelsName(attrs[1]), TypeName(attrs[3]), attrs[5]); goto error; } playback.Context = alcCreateContext(playback.Device, attrs); if(!playback.Context || alcMakeContextCurrent(playback.Context) == ALC_FALSE) { fprintf(stderr, "Failed to set an OpenAL audio context\n"); goto error; } /* Start SDL playing. Our callback (thus alcRenderSamplesSOFT) will now * start being called regularly to update the AL playback state. */ SDL_PauseAudio(0); /* Load the sound into a buffer. */ buffer = CreateSineWave(); if(!buffer) { SDL_CloseAudio(); alcDestroyContext(playback.Context); alcCloseDevice(playback.Device); SDL_Quit(); return 1; } /* Create the source to play the sound with. */ source = 0; alGenSources(1, &source); alSourcei(source, AL_BUFFER, buffer); assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source"); /* Play the sound until it finishes. */ alSourcePlay(source); do { al_nssleep(10000000); alGetSourcei(source, AL_SOURCE_STATE, &state); } while(alGetError() == AL_NO_ERROR && state == AL_PLAYING); /* All done. Delete resources, and close OpenAL. */ alDeleteSources(1, &source); alDeleteBuffers(1, &buffer); /* Stop SDL playing. */ SDL_PauseAudio(1); /* Close up OpenAL and SDL. */ SDL_CloseAudio(); alcDestroyContext(playback.Context); alcCloseDevice(playback.Device); SDL_Quit(); return 0; error: SDL_CloseAudio(); if(playback.Context) alcDestroyContext(playback.Context); if(playback.Device) alcCloseDevice(playback.Device); SDL_Quit(); return 1; } openal-soft-openal-soft-1.19.1/examples/almultireverb.c000066400000000000000000000633161335774445300231460ustar00rootroot00000000000000/* * OpenAL Multi-Zone Reverb Example * * Copyright (c) 2018 by Chris Robinson * * Permission is hereby granted, free of charge, to any person obtaining a copy * of this software and associated documentation files (the "Software"), to deal * in the Software without restriction, including without limitation the rights * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell * copies of the Software, and to permit persons to whom the Software is * furnished to do so, subject to the following conditions: * * The above copyright notice and this permission notice shall be included in * all copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN * THE SOFTWARE. */ /* This file contains an example for controlling multiple reverb zones to * smoothly transition between reverb environments. The general concept is to * extend single-reverb by also tracking the closest adjacent environment, and * utilize EAX Reverb's panning vectors to position them relative to the * listener. */ #include #include #include #include #include "AL/al.h" #include "AL/alc.h" #include "AL/alext.h" #include "AL/efx-presets.h" #include "common/alhelpers.h" #ifndef M_PI #define M_PI 3.14159265358979323846 #endif /* Filter object functions */ static LPALGENFILTERS alGenFilters; static LPALDELETEFILTERS alDeleteFilters; static LPALISFILTER alIsFilter; static LPALFILTERI alFilteri; static LPALFILTERIV alFilteriv; static LPALFILTERF alFilterf; static LPALFILTERFV alFilterfv; static LPALGETFILTERI alGetFilteri; static LPALGETFILTERIV alGetFilteriv; static LPALGETFILTERF alGetFilterf; static LPALGETFILTERFV alGetFilterfv; /* Effect object functions */ static LPALGENEFFECTS alGenEffects; static LPALDELETEEFFECTS alDeleteEffects; static LPALISEFFECT alIsEffect; static LPALEFFECTI alEffecti; static LPALEFFECTIV alEffectiv; static LPALEFFECTF alEffectf; static LPALEFFECTFV alEffectfv; static LPALGETEFFECTI alGetEffecti; static LPALGETEFFECTIV alGetEffectiv; static LPALGETEFFECTF alGetEffectf; static LPALGETEFFECTFV alGetEffectfv; /* Auxiliary Effect Slot object functions */ static LPALGENAUXILIARYEFFECTSLOTS alGenAuxiliaryEffectSlots; static LPALDELETEAUXILIARYEFFECTSLOTS alDeleteAuxiliaryEffectSlots; static LPALISAUXILIARYEFFECTSLOT alIsAuxiliaryEffectSlot; static LPALAUXILIARYEFFECTSLOTI alAuxiliaryEffectSloti; static LPALAUXILIARYEFFECTSLOTIV alAuxiliaryEffectSlotiv; static LPALAUXILIARYEFFECTSLOTF alAuxiliaryEffectSlotf; static LPALAUXILIARYEFFECTSLOTFV alAuxiliaryEffectSlotfv; static LPALGETAUXILIARYEFFECTSLOTI alGetAuxiliaryEffectSloti; static LPALGETAUXILIARYEFFECTSLOTIV alGetAuxiliaryEffectSlotiv; static LPALGETAUXILIARYEFFECTSLOTF alGetAuxiliaryEffectSlotf; static LPALGETAUXILIARYEFFECTSLOTFV alGetAuxiliaryEffectSlotfv; /* LoadEffect loads the given initial reverb properties into the given OpenAL * effect object, and returns non-zero on success. */ static int LoadEffect(ALuint effect, const EFXEAXREVERBPROPERTIES *reverb) { ALenum err; alGetError(); /* Prepare the effect for EAX Reverb (standard reverb doesn't contain * the needed panning vectors). */ alEffecti(effect, AL_EFFECT_TYPE, AL_EFFECT_EAXREVERB); if((err=alGetError()) != AL_NO_ERROR) { fprintf(stderr, "Failed to set EAX Reverb: %s (0x%04x)\n", alGetString(err), err); return 0; } /* Load the reverb properties. */ alEffectf(effect, AL_EAXREVERB_DENSITY, reverb->flDensity); alEffectf(effect, AL_EAXREVERB_DIFFUSION, reverb->flDiffusion); alEffectf(effect, AL_EAXREVERB_GAIN, reverb->flGain); alEffectf(effect, AL_EAXREVERB_GAINHF, reverb->flGainHF); alEffectf(effect, AL_EAXREVERB_GAINLF, reverb->flGainLF); alEffectf(effect, AL_EAXREVERB_DECAY_TIME, reverb->flDecayTime); alEffectf(effect, AL_EAXREVERB_DECAY_HFRATIO, reverb->flDecayHFRatio); alEffectf(effect, AL_EAXREVERB_DECAY_LFRATIO, reverb->flDecayLFRatio); alEffectf(effect, AL_EAXREVERB_REFLECTIONS_GAIN, reverb->flReflectionsGain); alEffectf(effect, AL_EAXREVERB_REFLECTIONS_DELAY, reverb->flReflectionsDelay); alEffectfv(effect, AL_EAXREVERB_REFLECTIONS_PAN, reverb->flReflectionsPan); alEffectf(effect, AL_EAXREVERB_LATE_REVERB_GAIN, reverb->flLateReverbGain); alEffectf(effect, AL_EAXREVERB_LATE_REVERB_DELAY, reverb->flLateReverbDelay); alEffectfv(effect, AL_EAXREVERB_LATE_REVERB_PAN, reverb->flLateReverbPan); alEffectf(effect, AL_EAXREVERB_ECHO_TIME, reverb->flEchoTime); alEffectf(effect, AL_EAXREVERB_ECHO_DEPTH, reverb->flEchoDepth); alEffectf(effect, AL_EAXREVERB_MODULATION_TIME, reverb->flModulationTime); alEffectf(effect, AL_EAXREVERB_MODULATION_DEPTH, reverb->flModulationDepth); alEffectf(effect, AL_EAXREVERB_AIR_ABSORPTION_GAINHF, reverb->flAirAbsorptionGainHF); alEffectf(effect, AL_EAXREVERB_HFREFERENCE, reverb->flHFReference); alEffectf(effect, AL_EAXREVERB_LFREFERENCE, reverb->flLFReference); alEffectf(effect, AL_EAXREVERB_ROOM_ROLLOFF_FACTOR, reverb->flRoomRolloffFactor); alEffecti(effect, AL_EAXREVERB_DECAY_HFLIMIT, reverb->iDecayHFLimit); /* Check if an error occured, and return failure if so. */ if((err=alGetError()) != AL_NO_ERROR) { fprintf(stderr, "Error setting up reverb: %s\n", alGetString(err)); return 0; } return 1; } /* LoadBuffer loads the named audio file into an OpenAL buffer object, and * returns the new buffer ID. */ static ALuint LoadSound(const char *filename) { Sound_Sample *sample; ALenum err, format; ALuint buffer; Uint32 slen; /* Open the audio file */ sample = Sound_NewSampleFromFile(filename, NULL, 65536); if(!sample) { fprintf(stderr, "Could not open audio in %s\n", filename); return 0; } /* Get the sound format, and figure out the OpenAL format */ if(sample->actual.channels == 1) { if(sample->actual.format == AUDIO_U8) format = AL_FORMAT_MONO8; else if(sample->actual.format == AUDIO_S16SYS) format = AL_FORMAT_MONO16; else { fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format); Sound_FreeSample(sample); return 0; } } else if(sample->actual.channels == 2) { if(sample->actual.format == AUDIO_U8) format = AL_FORMAT_STEREO8; else if(sample->actual.format == AUDIO_S16SYS) format = AL_FORMAT_STEREO16; else { fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format); Sound_FreeSample(sample); return 0; } } else { fprintf(stderr, "Unsupported channel count: %d\n", sample->actual.channels); Sound_FreeSample(sample); return 0; } /* Decode the whole audio stream to a buffer. */ slen = Sound_DecodeAll(sample); if(!sample->buffer || slen == 0) { fprintf(stderr, "Failed to read audio from %s\n", filename); Sound_FreeSample(sample); return 0; } /* Buffer the audio data into a new buffer object, then free the data and * close the file. */ buffer = 0; alGenBuffers(1, &buffer); alBufferData(buffer, format, sample->buffer, slen, sample->actual.rate); Sound_FreeSample(sample); /* Check if an error occured, and clean up if so. */ err = alGetError(); if(err != AL_NO_ERROR) { fprintf(stderr, "OpenAL Error: %s\n", alGetString(err)); if(buffer && alIsBuffer(buffer)) alDeleteBuffers(1, &buffer); return 0; } return buffer; } /* Helper to calculate the dot-product of the two given vectors. */ static ALfloat dot_product(const ALfloat vec0[3], const ALfloat vec1[3]) { return vec0[0]*vec1[0] + vec0[1]*vec1[1] + vec0[2]*vec1[2]; } /* Helper to normalize a given vector. */ static void normalize(ALfloat vec[3]) { ALfloat mag = sqrtf(dot_product(vec, vec)); if(mag > 0.00001f) { vec[0] /= mag; vec[1] /= mag; vec[2] /= mag; } else { vec[0] = 0.0f; vec[1] = 0.0f; vec[2] = 0.0f; } } /* The main update function to update the listener and environment effects. */ static void UpdateListenerAndEffects(float timediff, const ALuint slots[2], const ALuint effects[2], const EFXEAXREVERBPROPERTIES reverbs[2]) { static const ALfloat listener_move_scale = 10.0f; /* Individual reverb zones are connected via "portals". Each portal has a * position (center point of the connecting area), a normal (facing * direction), and a radius (approximate size of the connecting area). */ const ALfloat portal_pos[3] = { 0.0f, 0.0f, 0.0f }; const ALfloat portal_norm[3] = { sqrtf(0.5f), 0.0f, -sqrtf(0.5f) }; const ALfloat portal_radius = 2.5f; ALfloat other_dir[3], this_dir[3]; ALfloat listener_pos[3]; ALfloat local_norm[3]; ALfloat local_dir[3]; ALfloat near_edge[3]; ALfloat far_edge[3]; ALfloat dist, edist; /* Update the listener position for the amount of time passed. This uses a * simple triangular LFO to offset the position (moves along the X axis * between -listener_move_scale and +listener_move_scale for each * transition). */ listener_pos[0] = (fabsf(2.0f - timediff/2.0f) - 1.0f) * listener_move_scale; listener_pos[1] = 0.0f; listener_pos[2] = 0.0f; alListenerfv(AL_POSITION, listener_pos); /* Calculate local_dir, which represents the listener-relative point to the * adjacent zone (should also include orientation). Because EAX Reverb uses * left-handed coordinates instead of right-handed like the rest of OpenAL, * negate Z for the local values. */ local_dir[0] = portal_pos[0] - listener_pos[0]; local_dir[1] = portal_pos[1] - listener_pos[1]; local_dir[2] = -(portal_pos[2] - listener_pos[2]); /* A normal application would also rotate the portal's normal given the * listener orientation, to get the listener-relative normal. */ local_norm[0] = portal_norm[0]; local_norm[1] = portal_norm[1]; local_norm[2] = -portal_norm[2]; /* Calculate the distance from the listener to the portal, and ensure it's * far enough away to not suffer severe floating-point precision issues. */ dist = sqrtf(dot_product(local_dir, local_dir)); if(dist > 0.00001f) { const EFXEAXREVERBPROPERTIES *other_reverb, *this_reverb; ALuint other_effect, this_effect; ALfloat magnitude, dir_dot_norm; /* Normalize the direction to the portal. */ local_dir[0] /= dist; local_dir[1] /= dist; local_dir[2] /= dist; /* Calculate the dot product of the portal's local direction and local * normal, which is used for angular and side checks later on. */ dir_dot_norm = dot_product(local_dir, local_norm); /* Figure out which zone we're in. */ if(dir_dot_norm <= 0.0f) { /* We're in front of the portal, so we're in Zone 0. */ this_effect = effects[0]; other_effect = effects[1]; this_reverb = &reverbs[0]; other_reverb = &reverbs[1]; } else { /* We're behind the portal, so we're in Zone 1. */ this_effect = effects[1]; other_effect = effects[0]; this_reverb = &reverbs[1]; other_reverb = &reverbs[0]; } /* Calculate the listener-relative extents of the portal. */ /* First, project the listener-to-portal vector onto the portal's plane * to get the portal-relative direction along the plane that goes away * from the listener (toward the farthest edge of the portal). */ far_edge[0] = local_dir[0] - local_norm[0]*dir_dot_norm; far_edge[1] = local_dir[1] - local_norm[1]*dir_dot_norm; far_edge[2] = local_dir[2] - local_norm[2]*dir_dot_norm; edist = sqrtf(dot_product(far_edge, far_edge)); if(edist > 0.0001f) { /* Rescale the portal-relative vector to be at the radius edge. */ ALfloat mag = portal_radius / edist; far_edge[0] *= mag; far_edge[1] *= mag; far_edge[2] *= mag; /* Calculate the closest edge of the portal by negating the * farthest, and add an offset to make them both relative to the * listener. */ near_edge[0] = local_dir[0]*dist - far_edge[0]; near_edge[1] = local_dir[1]*dist - far_edge[1]; near_edge[2] = local_dir[2]*dist - far_edge[2]; far_edge[0] += local_dir[0]*dist; far_edge[1] += local_dir[1]*dist; far_edge[2] += local_dir[2]*dist; /* Normalize the listener-relative extents of the portal, then * calculate the panning magnitude for the other zone given the * apparent size of the opening. The panning magnitude affects the * envelopment of the environment, with 1 being a point, 0.5 being * half coverage around the listener, and 0 being full coverage. */ normalize(far_edge); normalize(near_edge); magnitude = 1.0f - acosf(dot_product(far_edge, near_edge))/(float)(M_PI*2.0); /* Recalculate the panning direction, to be directly between the * direction of the two extents. */ local_dir[0] = far_edge[0] + near_edge[0]; local_dir[1] = far_edge[1] + near_edge[1]; local_dir[2] = far_edge[2] + near_edge[2]; normalize(local_dir); } else { /* If we get here, the listener is directly in front of or behind * the center of the portal, making all aperture edges effectively * equidistant. Calculating the panning magnitude is simplified, * using the arctangent of the radius and distance. */ magnitude = 1.0f - (atan2f(portal_radius, dist) / (float)M_PI); } /* Scale the other zone's panning vector. */ other_dir[0] = local_dir[0] * magnitude; other_dir[1] = local_dir[1] * magnitude; other_dir[2] = local_dir[2] * magnitude; /* Pan the current zone to the opposite direction of the portal, and * take the remaining percentage of the portal's magnitude. */ this_dir[0] = local_dir[0] * (magnitude-1.0f); this_dir[1] = local_dir[1] * (magnitude-1.0f); this_dir[2] = local_dir[2] * (magnitude-1.0f); /* Now set the effects' panning vectors and gain. Energy is shared * between environments, so attenuate according to each zone's * contribution (note: gain^2 = energy). */ alEffectf(this_effect, AL_EAXREVERB_REFLECTIONS_GAIN, this_reverb->flReflectionsGain * sqrtf(magnitude)); alEffectf(this_effect, AL_EAXREVERB_LATE_REVERB_GAIN, this_reverb->flLateReverbGain * sqrtf(magnitude)); alEffectfv(this_effect, AL_EAXREVERB_REFLECTIONS_PAN, this_dir); alEffectfv(this_effect, AL_EAXREVERB_LATE_REVERB_PAN, this_dir); alEffectf(other_effect, AL_EAXREVERB_REFLECTIONS_GAIN, other_reverb->flReflectionsGain * sqrtf(1.0f-magnitude)); alEffectf(other_effect, AL_EAXREVERB_LATE_REVERB_GAIN, other_reverb->flLateReverbGain * sqrtf(1.0f-magnitude)); alEffectfv(other_effect, AL_EAXREVERB_REFLECTIONS_PAN, other_dir); alEffectfv(other_effect, AL_EAXREVERB_LATE_REVERB_PAN, other_dir); } else { /* We're practically in the center of the portal. Give the panning * vectors a 50/50 split, with Zone 0 covering the half in front of * the normal, and Zone 1 covering the half behind. */ this_dir[0] = local_norm[0] / 2.0f; this_dir[1] = local_norm[1] / 2.0f; this_dir[2] = local_norm[2] / 2.0f; other_dir[0] = local_norm[0] / -2.0f; other_dir[1] = local_norm[1] / -2.0f; other_dir[2] = local_norm[2] / -2.0f; alEffectf(effects[0], AL_EAXREVERB_REFLECTIONS_GAIN, reverbs[0].flReflectionsGain * sqrtf(0.5f)); alEffectf(effects[0], AL_EAXREVERB_LATE_REVERB_GAIN, reverbs[0].flLateReverbGain * sqrtf(0.5f)); alEffectfv(effects[0], AL_EAXREVERB_REFLECTIONS_PAN, this_dir); alEffectfv(effects[0], AL_EAXREVERB_LATE_REVERB_PAN, this_dir); alEffectf(effects[1], AL_EAXREVERB_REFLECTIONS_GAIN, reverbs[1].flReflectionsGain * sqrtf(0.5f)); alEffectf(effects[1], AL_EAXREVERB_LATE_REVERB_GAIN, reverbs[1].flLateReverbGain * sqrtf(0.5f)); alEffectfv(effects[1], AL_EAXREVERB_REFLECTIONS_PAN, other_dir); alEffectfv(effects[1], AL_EAXREVERB_LATE_REVERB_PAN, other_dir); } /* Finally, update the effect slots with the updated effect parameters. */ alAuxiliaryEffectSloti(slots[0], AL_EFFECTSLOT_EFFECT, effects[0]); alAuxiliaryEffectSloti(slots[1], AL_EFFECTSLOT_EFFECT, effects[1]); } int main(int argc, char **argv) { static const int MaxTransitions = 8; EFXEAXREVERBPROPERTIES reverbs[2] = { EFX_REVERB_PRESET_CARPETEDHALLWAY, EFX_REVERB_PRESET_BATHROOM }; struct timespec basetime; ALCdevice *device = NULL; ALCcontext *context = NULL; ALuint effects[2] = { 0, 0 }; ALuint slots[2] = { 0, 0 }; ALuint direct_filter = 0; ALuint buffer = 0; ALuint source = 0; ALCint num_sends = 0; ALenum state = AL_INITIAL; ALfloat direct_gain = 1.0f; int loops = 0; /* Print out usage if no arguments were specified */ if(argc < 2) { fprintf(stderr, "Usage: %s [-device ] [options] \n\n" "Options:\n" "\t-nodirect\tSilence direct path output (easier to hear reverb)\n\n", argv[0]); return 1; } /* Initialize OpenAL, and check for EFX support with at least 2 auxiliary * sends (if multiple sends are supported, 2 are provided by default; if * you want more, you have to request it through alcCreateContext). */ argv++; argc--; if(InitAL(&argv, &argc) != 0) return 1; while(argc > 0) { if(strcmp(argv[0], "-nodirect") == 0) direct_gain = 0.0f; else break; argv++; argc--; } if(argc < 1) { fprintf(stderr, "No filename spacified.\n"); CloseAL(); return 1; } context = alcGetCurrentContext(); device = alcGetContextsDevice(context); if(!alcIsExtensionPresent(device, "ALC_EXT_EFX")) { fprintf(stderr, "Error: EFX not supported\n"); CloseAL(); return 1; } num_sends = 0; alcGetIntegerv(device, ALC_MAX_AUXILIARY_SENDS, 1, &num_sends); if(alcGetError(device) != ALC_NO_ERROR || num_sends < 2) { fprintf(stderr, "Error: Device does not support multiple sends (got %d, need 2)\n", num_sends); CloseAL(); return 1; } /* Define a macro to help load the function pointers. */ #define LOAD_PROC(x) ((x) = alGetProcAddress(#x)) LOAD_PROC(alGenFilters); LOAD_PROC(alDeleteFilters); LOAD_PROC(alIsFilter); LOAD_PROC(alFilteri); LOAD_PROC(alFilteriv); LOAD_PROC(alFilterf); LOAD_PROC(alFilterfv); LOAD_PROC(alGetFilteri); LOAD_PROC(alGetFilteriv); LOAD_PROC(alGetFilterf); LOAD_PROC(alGetFilterfv); LOAD_PROC(alGenEffects); LOAD_PROC(alDeleteEffects); LOAD_PROC(alIsEffect); LOAD_PROC(alEffecti); LOAD_PROC(alEffectiv); LOAD_PROC(alEffectf); LOAD_PROC(alEffectfv); LOAD_PROC(alGetEffecti); LOAD_PROC(alGetEffectiv); LOAD_PROC(alGetEffectf); LOAD_PROC(alGetEffectfv); LOAD_PROC(alGenAuxiliaryEffectSlots); LOAD_PROC(alDeleteAuxiliaryEffectSlots); LOAD_PROC(alIsAuxiliaryEffectSlot); LOAD_PROC(alAuxiliaryEffectSloti); LOAD_PROC(alAuxiliaryEffectSlotiv); LOAD_PROC(alAuxiliaryEffectSlotf); LOAD_PROC(alAuxiliaryEffectSlotfv); LOAD_PROC(alGetAuxiliaryEffectSloti); LOAD_PROC(alGetAuxiliaryEffectSlotiv); LOAD_PROC(alGetAuxiliaryEffectSlotf); LOAD_PROC(alGetAuxiliaryEffectSlotfv); #undef LOAD_PROC /* Initialize SDL_sound. */ Sound_Init(); /* Load the sound into a buffer. */ buffer = LoadSound(argv[0]); if(!buffer) { CloseAL(); Sound_Quit(); return 1; } /* Generate two effects for two "zones", and load a reverb into each one. * Note that unlike single-zone reverb, where you can store one effect per * preset, for multi-zone reverb you should have one effect per environment * instance, or one per audible zone. This is because we'll be changing the * effects' properties in real-time based on the environment instance * relative to the listener. */ alGenEffects(2, effects); if(!LoadEffect(effects[0], &reverbs[0]) || !LoadEffect(effects[1], &reverbs[1])) { alDeleteEffects(2, effects); alDeleteBuffers(1, &buffer); Sound_Quit(); CloseAL(); return 1; } /* Create the effect slot objects, one for each "active" effect. */ alGenAuxiliaryEffectSlots(2, slots); /* Tell the effect slots to use the loaded effect objects, with slot 0 for * Zone 0 and slot 1 for Zone 1. Note that this effectively copies the * effect properties. Modifying or deleting the effect object afterward * won't directly affect the effect slot until they're reapplied like this. */ alAuxiliaryEffectSloti(slots[0], AL_EFFECTSLOT_EFFECT, effects[0]); alAuxiliaryEffectSloti(slots[1], AL_EFFECTSLOT_EFFECT, effects[1]); assert(alGetError()==AL_NO_ERROR && "Failed to set effect slot"); /* For the purposes of this example, prepare a filter that optionally * silences the direct path which allows us to hear just the reverberation. * A filter like this is normally used for obstruction, where the path * directly between the listener and source is blocked (the exact * properties depending on the type and thickness of the obstructing * material). */ alGenFilters(1, &direct_filter); alFilteri(direct_filter, AL_FILTER_TYPE, AL_FILTER_LOWPASS); alFilterf(direct_filter, AL_LOWPASS_GAIN, direct_gain); assert(alGetError()==AL_NO_ERROR && "Failed to set direct filter"); /* Create the source to play the sound with, place it in front of the * listener's path in the left zone. */ source = 0; alGenSources(1, &source); alSourcei(source, AL_LOOPING, AL_TRUE); alSource3f(source, AL_POSITION, -5.0f, 0.0f, -2.0f); alSourcei(source, AL_DIRECT_FILTER, direct_filter); alSourcei(source, AL_BUFFER, buffer); /* Connect the source to the effect slots. Here, we connect source send 0 * to Zone 0's slot, and send 1 to Zone 1's slot. Filters can be specified * to occlude the source from each zone by varying amounts; for example, a * source within a particular zone would be unfiltered, while a source that * can only see a zone through a window or thin wall may be attenuated for * that zone. */ alSource3i(source, AL_AUXILIARY_SEND_FILTER, slots[0], 0, AL_FILTER_NULL); alSource3i(source, AL_AUXILIARY_SEND_FILTER, slots[1], 1, AL_FILTER_NULL); assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source"); /* Get the current time as the base for timing in the main loop. */ altimespec_get(&basetime, AL_TIME_UTC); loops = 0; printf("Transition %d of %d...\n", loops+1, MaxTransitions); /* Play the sound for a while. */ alSourcePlay(source); do { struct timespec curtime; ALfloat timediff; /* Start a batch update, to ensure all changes apply simultaneously. */ alcSuspendContext(context); /* Get the current time to track the amount of time that passed. * Convert the difference to seconds. */ altimespec_get(&curtime, AL_TIME_UTC); timediff = (ALfloat)(curtime.tv_sec - basetime.tv_sec); timediff += (ALfloat)(curtime.tv_nsec - basetime.tv_nsec) / 1000000000.0f; /* Avoid negative time deltas, in case of non-monotonic clocks. */ if(timediff < 0.0f) timediff = 0.0f; else while(timediff >= 4.0f*((loops&1)+1)) { /* For this example, each transition occurs over 4 seconds, and * there's 2 transitions per cycle. */ if(++loops < MaxTransitions) printf("Transition %d of %d...\n", loops+1, MaxTransitions); if(!(loops&1)) { /* Cycle completed. Decrease the delta and increase the base * time to start a new cycle. */ timediff -= 8.0f; basetime.tv_sec += 8; } } /* Update the listener and effects, and finish the batch. */ UpdateListenerAndEffects(timediff, slots, effects, reverbs); alcProcessContext(context); al_nssleep(10000000); alGetSourcei(source, AL_SOURCE_STATE, &state); } while(alGetError() == AL_NO_ERROR && state == AL_PLAYING && loops < MaxTransitions); /* All done. Delete resources, and close down SDL_sound and OpenAL. */ alDeleteSources(1, &source); alDeleteAuxiliaryEffectSlots(2, slots); alDeleteEffects(2, effects); alDeleteFilters(1, &direct_filter); alDeleteBuffers(1, &buffer); Sound_Quit(); CloseAL(); return 0; } openal-soft-openal-soft-1.19.1/examples/alplay.c000066400000000000000000000117771335774445300215570ustar00rootroot00000000000000/* * OpenAL Source Play Example * * Copyright (c) 2017 by Chris Robinson * * Permission is hereby granted, free of charge, to any person obtaining a copy * of this software and associated documentation files (the "Software"), to deal * in the Software without restriction, including without limitation the rights * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell * copies of the Software, and to permit persons to whom the Software is * furnished to do so, subject to the following conditions: * * The above copyright notice and this permission notice shall be included in * all copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN * THE SOFTWARE. */ /* This file contains an example for playing a sound buffer. */ #include #include #include #include "AL/al.h" #include "AL/alc.h" #include "common/alhelpers.h" /* LoadBuffer loads the named audio file into an OpenAL buffer object, and * returns the new buffer ID. */ static ALuint LoadSound(const char *filename) { Sound_Sample *sample; ALenum err, format; ALuint buffer; Uint32 slen; /* Open the audio file */ sample = Sound_NewSampleFromFile(filename, NULL, 65536); if(!sample) { fprintf(stderr, "Could not open audio in %s\n", filename); return 0; } /* Get the sound format, and figure out the OpenAL format */ if(sample->actual.channels == 1) { if(sample->actual.format == AUDIO_U8) format = AL_FORMAT_MONO8; else if(sample->actual.format == AUDIO_S16SYS) format = AL_FORMAT_MONO16; else { fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format); Sound_FreeSample(sample); return 0; } } else if(sample->actual.channels == 2) { if(sample->actual.format == AUDIO_U8) format = AL_FORMAT_STEREO8; else if(sample->actual.format == AUDIO_S16SYS) format = AL_FORMAT_STEREO16; else { fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format); Sound_FreeSample(sample); return 0; } } else { fprintf(stderr, "Unsupported channel count: %d\n", sample->actual.channels); Sound_FreeSample(sample); return 0; } /* Decode the whole audio stream to a buffer. */ slen = Sound_DecodeAll(sample); if(!sample->buffer || slen == 0) { fprintf(stderr, "Failed to read audio from %s\n", filename); Sound_FreeSample(sample); return 0; } /* Buffer the audio data into a new buffer object, then free the data and * close the file. */ buffer = 0; alGenBuffers(1, &buffer); alBufferData(buffer, format, sample->buffer, slen, sample->actual.rate); Sound_FreeSample(sample); /* Check if an error occured, and clean up if so. */ err = alGetError(); if(err != AL_NO_ERROR) { fprintf(stderr, "OpenAL Error: %s\n", alGetString(err)); if(buffer && alIsBuffer(buffer)) alDeleteBuffers(1, &buffer); return 0; } return buffer; } int main(int argc, char **argv) { ALuint source, buffer; ALfloat offset; ALenum state; /* Print out usage if no arguments were specified */ if(argc < 2) { fprintf(stderr, "Usage: %s [-device ] \n", argv[0]); return 1; } /* Initialize OpenAL. */ argv++; argc--; if(InitAL(&argv, &argc) != 0) return 1; /* Initialize SDL_sound. */ Sound_Init(); /* Load the sound into a buffer. */ buffer = LoadSound(argv[0]); if(!buffer) { Sound_Quit(); CloseAL(); return 1; } /* Create the source to play the sound with. */ source = 0; alGenSources(1, &source); alSourcei(source, AL_BUFFER, buffer); assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source"); /* Play the sound until it finishes. */ alSourcePlay(source); do { al_nssleep(10000000); alGetSourcei(source, AL_SOURCE_STATE, &state); /* Get the source offset. */ alGetSourcef(source, AL_SEC_OFFSET, &offset); printf("\rOffset: %f ", offset); fflush(stdout); } while(alGetError() == AL_NO_ERROR && state == AL_PLAYING); printf("\n"); /* All done. Delete resources, and close down SDL_sound and OpenAL. */ alDeleteSources(1, &source); alDeleteBuffers(1, &buffer); Sound_Quit(); CloseAL(); return 0; } openal-soft-openal-soft-1.19.1/examples/alrecord.c000066400000000000000000000313001335774445300220500ustar00rootroot00000000000000/* * OpenAL Recording Example * * Copyright (c) 2017 by Chris Robinson * * Permission is hereby granted, free of charge, to any person obtaining a copy * of this software and associated documentation files (the "Software"), to deal * in the Software without restriction, including without limitation the rights * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell * copies of the Software, and to permit persons to whom the Software is * furnished to do so, subject to the following conditions: * * The above copyright notice and this permission notice shall be included in * all copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN * THE SOFTWARE. */ /* This file contains a relatively simple recorder. */ #include #include #include #include #include "AL/al.h" #include "AL/alc.h" #include "AL/alext.h" #include "common/alhelpers.h" #if defined(_WIN64) #define SZFMT "%I64u" #elif defined(_WIN32) #define SZFMT "%u" #else #define SZFMT "%zu" #endif #if defined(_MSC_VER) && (_MSC_VER < 1900) static float msvc_strtof(const char *str, char **end) { return (float)strtod(str, end); } #define strtof msvc_strtof #endif static void fwrite16le(ALushort val, FILE *f) { ALubyte data[2] = { val&0xff, (val>>8)&0xff }; fwrite(data, 1, 2, f); } static void fwrite32le(ALuint val, FILE *f) { ALubyte data[4] = { val&0xff, (val>>8)&0xff, (val>>16)&0xff, (val>>24)&0xff }; fwrite(data, 1, 4, f); } typedef struct Recorder { ALCdevice *mDevice; FILE *mFile; long mDataSizeOffset; ALuint mDataSize; float mRecTime; int mChannels; int mBits; int mSampleRate; ALuint mFrameSize; ALbyte *mBuffer; ALsizei mBufferSize; } Recorder; int main(int argc, char **argv) { static const char optlist[] = " --channels/-c Set channel count (1 or 2)\n" " --bits/-b Set channel count (8, 16, or 32)\n" " --rate/-r Set sample rate (8000 to 96000)\n" " --time/-t