pax_global_header00006660000000000000000000000064147265700740014526gustar00rootroot0000000000000052 comment=e907571bad7dee296fb56a409a0f7c5a0c57d38a pcaudiolib-1.3/000077500000000000000000000000001472657007400135045ustar00rootroot00000000000000pcaudiolib-1.3/.gitignore000066400000000000000000000006031472657007400154730ustar00rootroot00000000000000# build artefacts: *.o *.lo # vi/vim editor files: *~ .*.swp # build output: src/libpcaudio.a src/libpcaudio.la pcaudiolib-* # autotools output: Makefile Makefile.in m4/ .libs/ .deps/ .dirstamp README NEWS INSTALL aclocal.m4 autom4te.cache/ ChangeLog compile config.h config.h.in config.h.in~ config.log config.* configure depcomp install-sh missing stamp-h1 libtool ltmain.sh pcaudiolib-1.3/.gitmodules000066400000000000000000000000001472657007400156470ustar00rootroot00000000000000pcaudiolib-1.3/AUTHORS000066400000000000000000000000161472657007400145510ustar00rootroot00000000000000Reece H. Dunn pcaudiolib-1.3/COPYING000066400000000000000000001045131472657007400145430ustar00rootroot00000000000000 GNU GENERAL PUBLIC LICENSE Version 3, 29 June 2007 Copyright (C) 2007 Free Software Foundation, Inc. Everyone is permitted to copy and distribute verbatim copies of this license document, but changing it is not allowed. Preamble The GNU General Public License is a free, copyleft license for software and other kinds of works. The licenses for most software and other practical works are designed to take away your freedom to share and change the works. By contrast, the GNU General Public License is intended to guarantee your freedom to share and change all versions of a program--to make sure it remains free software for all its users. We, the Free Software Foundation, use the GNU General Public License for most of our software; it applies also to any other work released this way by its authors. You can apply it to your programs, too. 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The hypothetical commands `show w' and `show c' should show the appropriate parts of the General Public License. Of course, your program's commands might be different; for a GUI interface, you would use an "about box". You should also get your employer (if you work as a programmer) or school, if any, to sign a "copyright disclaimer" for the program, if necessary. For more information on this, and how to apply and follow the GNU GPL, see . The GNU General Public License does not permit incorporating your program into proprietary programs. If your program is a subroutine library, you may consider it more useful to permit linking proprietary applications with the library. If this is what you want to do, use the GNU Lesser General Public License instead of this License. But first, please read . pcaudiolib-1.3/ChangeLog.md000066400000000000000000000012121472657007400156510ustar00rootroot00000000000000# Change Log ## 1.2 - \[18 Aug 2021\] * Fix cancellation snappiness * Fix build on non-linux * Alsa: fixed sample_size calculation, multiply with channel count. * Fix some typos ## 1.1 - \[27 Feb 2018\] * Fix handling EBADFD, EPIPE and short write when writing to the ALSA device. * Fix audio echos when using `snd_pcm_drop` in ALSA devices. * Experimental support for Mac OS CoreAudio. ## 1.0 \[10 Sep 2016\] * Provide a consistent API for accessing the different audio interfaces. * Support for PulseAudio using the simple API. * Support for ALSA. * Support for OSS version 3 and later. * Support for QSA. * Support for XAudio2. pcaudiolib-1.3/Makefile.am000066400000000000000000000036651472657007400155520ustar00rootroot00000000000000AUTOMAKE_OPTIONS = subdir-objects localedir = $(datadir)/locale AM_CFLAGS = \ -Isrc/include ACLOCAL_AMFLAGS = -I m4 lib_LTLIBRARIES = EXTRA_DIST = CLEANFILES = ############################# libtool ######################################### EXTRA_DIST += config.guess config.sub ltmain.sh # Increment if the interface has changed and is not backward compatible CURRENT=0 # Increment if source files have changed # Reset to 0 if the interface has changed REVISION=1 # Increment if the interface is backward compatible (superset) # Reset to 0 if the interface is not backward compatible AGE=0 LIBPCAUDIO_VERSION=$(CURRENT):$(REVISION):$(AGE) ############################# ChangeLog ####################################### ChangeLog: git log > ChangeLog dist-hook: ChangeLog .PHONY: ChangeLog EXTRA_DIST += ChangeLog ############################# libpcaudio ###################################### EXTRA_DIST += \ autogen.sh \ ChangeLog.md \ README.md libpcaudio_includedir = $(includedir)/pcaudiolib libpcaudio_include_HEADERS = \ src/include/pcaudiolib/audio.h lib_LTLIBRARIES += src/libpcaudio.la src_libpcaudio_la_LDFLAGS = -version-info $(LIBPCAUDIO_VERSION) \ ${ALSA_LIBS} \ ${PULSEAUDIO_LIBS} \ ${QSA_LIBS} \ ${COREAUDIO_LIBS} src_libpcaudio_la_CFLAGS = ${AM_CFLAGS} \ ${ALSA_CFLAGS} \ ${PULSEAUDIO_CFLAGS} \ ${COREAUDIO_CFLAGS} src_libpcaudio_la_SOURCES = \ src/alsa.c \ src/qsa.c \ src/oss.c \ src/pulseaudio.c \ src/audio_priv.h \ src/audio.c # Windows audio support EXTRA_DIST += \ src/windows.c \ src/xaudio2.cpp # Mac OSX audio support if HAVE_COREAUDIO src_libpcaudio_la_SOURCES += \ src/coreaudio.c \ src/TPCircularBuffer/TPCircularBuffer.c \ src/TPCircularBuffer/TPCircularBuffer.h \ src/TPCircularBuffer/TPCircularBuffer+AudioBufferList.c \ src/TPCircularBuffer/TPCircularBuffer+AudioBufferList.h EXTRA_DIST += \ src/TPCircularBuffer/README.markdown \ src/TPCircularBuffer/TPCircularBuffer.podspec endif pcaudiolib-1.3/README.md000066400000000000000000000050151472657007400147640ustar00rootroot00000000000000# Portable C Audio Library 1.3 - [Build Dependencies](#build-dependencies) - [Debian](#debian) - [Mac OS](#mac-os) - [Building](#building) - [Bugs](#bugs) - [License Information](#license-information) ---------- The Portable C Audio Library (pcaudiolib) provides a C API to different audio devices. It supports: | Audio Framework | Operating System | |-----------------|------------------| | ALSA | Linux | | CoreAudio | Mac OS | | OSS | POSIX | | PulseAudio | Linux | | QSA | QNX | | XAudio2 | Windows | See the [ChangeLog](ChangeLog.md) for a description of the changes in the various releases. ## Build Dependencies In order to build the project, you need: 1. an autotools system (`make`, `autoconf`, `automake`, `libtool` and `pkg-config`); 2. a C compiler (e.g. gcc or clang). Optionally, you need: 1. the alsa development libraries to enable alsa audio output; 2. the pulseaudio development library to enable pulseaudio output. ### Debian Core dependencies: | Dependency | Install | |------------|------------------------------------------------------------------| | autotools | `sudo apt-get install make autoconf automake libtool pkg-config` | | c compiler | `sudo apt-get install gcc` | Optional Libraries: | Dependency | Install | |----------------|--------------------------------------------| | alsa | `sudo apt-get install libasound2-dev` | | pulseaudio | `sudo apt-get install libpulse-dev` | ### Mac OS To enable CoreAudio output support you need to have the coreaudio framework on your system. Installing XCode along with the Mac OS SDK is the recommended way of getting it. For Xcode on OS X 10.9 and earlier C11 compiler or newer is needed. ### Windows Some of the Windows SDK APIs (e.g. the XAudio2 headers) don't compile with a C compiler, so a C++ compiler is needed to compile them. ## Building This project uses a standard autotools build system with an `autogen.sh` script. You can thus build the project using: ./autogen.sh ./configure --prefix=/usr make and install it using: sudo make install ## Bugs Report bugs to the [pcaudiolib issues](https://github.com/espeak-ng/pcaudiolib/issues) page on GitHub. ## License Information The Portable C Audio Library is released under the [GPL version 3](COPYING) or later license. pcaudiolib-1.3/autogen.sh000077500000000000000000000004761472657007400155140ustar00rootroot00000000000000#!/bin/sh LIBTOOLIZE=`which libtoolize` if ! test -f "$LIBTOOLIZE" ; then # Mac OSX support ... LIBTOOLIZE=`which glibtoolize` fi mkdir -p m4 ln -sf README.md README ln -sf ChangeLog.md NEWS aclocal -I m4 || exit 1 ${LIBTOOLIZE} || exit 1 autoheader || exit 1 automake --add-missing || exit 1 autoconf || exit 1 pcaudiolib-1.3/configure.ac000066400000000000000000000107601472657007400157760ustar00rootroot00000000000000AC_PREREQ([2.65]) AC_INIT([pcaudiolib], [1.3], [https://github.com/espeak-ng/pcaudiolib/issues], [], [https://github.com/espeak-ng/pcaudiolib]) AM_INIT_AUTOMAKE() m4_ifdef([AM_SILENT_RULES],[AM_SILENT_RULES]) AM_SILENT_RULES([yes]) AC_CONFIG_SRCDIR([src]) AC_CONFIG_MACRO_DIR([m4]) AC_CONFIG_HEADERS([config.h]) AC_CANONICAL_HOST dnl ================================================================ dnl Program checks. dnl ================================================================ AC_PROG_CC AC_PROG_MAKE_SET AC_PROG_LIBTOOL dnl ================================================================ dnl PulseAudio checks. dnl ================================================================ AC_ARG_WITH([pulseaudio], [AS_HELP_STRING([--with-pulseaudio], [support for PulseAudio output @<:@default=yes@:>@])], []) AS_IF([test "x$with_pulseaudio" = "xno"], [ echo "Disabling PulseAudio output support"; have_pulseaudio=no ], [ PKG_CHECK_MODULES(PULSEAUDIO, [libpulse-simple >= 0.9], [ AC_DEFINE(HAVE_PULSE_SIMPLE_H, [], [Do we have pulse/simple.h]) have_pulseaudio=yes ],[ have_pulseaudio=no ]) ]) AC_SUBST(PULSEAUDIO_CFLAGS) AC_SUBST(PULSEAUDIO_LIBS) dnl ================================================================ dnl ALSA checks. dnl ================================================================ AC_ARG_WITH([alsa], [AS_HELP_STRING([--with-alsa], [support for ALSA audio output @<:@default=yes@:>@])], []) AS_IF([test "x$with_alsa" = "xno"], [ echo "Disabling ALSA audio output support"; have_alsa=no ], [ PKG_CHECK_MODULES(ALSA, [alsa], [ AC_DEFINE(HAVE_ALSA_ASOUNDLIB_H, [], [Do we have ALSA]) have_alsa=yes ],[ have_alsa=no ])]) AC_SUBST(ALSA_CFLAGS) AC_SUBST(ALSA_LIBS) dnl ================================================================ dnl QSA checks. dnl ================================================================ AC_ARG_WITH([qsa], [AS_HELP_STRING([--with-qsa], [support for QSA audio output @<:@default=yes@:>@])], []) if test "$with_qsa" = "no"; then echo "Disabling QSA audio output support" have_qsa=no else AC_CHECK_HEADERS([sys/asoundlib.h],[ AC_CHECK_HEADERS([sys/asound.h],[ have_qsa=yes QSA_LIBS=-lasound ],[ have_qsa=no ]) ],[ have_qsa=no ]) fi AC_SUBST(QSA_LIBS) dnl ================================================================ dnl coreaudio checks. dnl ================================================================ AC_ARG_WITH([coreaudio], [AS_HELP_STRING([--with-coreaudio], [support for coreaudio audio output on Mac OS @<:@default=yes@:>@])], []) if test "$with_coreaudio" = "no";then echo "Disabling coreaudio audio output support" have_coreaudio=no else case $host_os in *darwin*) COREAUDIO_CFLAGS="-dynamiclib -fvisibility=default" COREAUDIO_LIBS="-framework CoreAudio -framework AudioToolbox -framework AudioUnit" have_coreaudio=yes AC_DEFINE(HAVE_COREAUDIO_H, [], [Do we have coreaudio?]) ;; *) have_coreaudio=no ;; esac fi AC_SUBST(COREAUDIO_CFLAGS) AC_SUBST(COREAUDIO_LIBS) AM_CONDITIONAL([HAVE_COREAUDIO], [test "x${have_coreaudio}" = "xyes"]) dnl ================================================================ dnl OSS checks. dnl ================================================================ AC_ARG_WITH([oss], [AS_HELP_STRING([--with-oss], [support for OSS audio output @<:@default=yes@:>@])], []) if test "$with_oss" = "no"; then echo "Disabling OSS audio output support" have_oss=no else AC_CHECK_HEADERS([sys/soundcard.h],[ AC_CHECK_LIB(ossaudio, _oss_ioctl) have_oss=yes ],[ have_oss=no ]) fi dnl ================================================================ dnl Generate output. dnl ================================================================ AC_CONFIG_FILES([Makefile]) AC_OUTPUT AC_MSG_NOTICE([ Configuration for Portable C Audio Library complete. Source code location: ${srcdir} Compiler: ${CC} Compiler flags: ${CFLAGS} PulseAudio support: ${have_pulseaudio} ALSA support: ${have_alsa} QSA support: ${have_qsa} Coreaudio support: ${have_coreaudio} OSS support: ${have_oss} ]) pcaudiolib-1.3/src/000077500000000000000000000000001472657007400142735ustar00rootroot00000000000000pcaudiolib-1.3/src/TPCircularBuffer/000077500000000000000000000000001472657007400174355ustar00rootroot00000000000000pcaudiolib-1.3/src/TPCircularBuffer/README.markdown000066400000000000000000000053511472657007400221420ustar00rootroot00000000000000A simple, fast circular buffer implementation for audio processing ================================================================== A simple C implementation for a circular (ring) buffer. Thread-safe with a single producer and a single consumer, using OSAtomic.h primitives, and avoids any need for buffer wrapping logic by using a virtual memory map technique to place a virtual copy of the buffer straight after the end of the real buffer. Usage ----- Initialisation and cleanup: `TPCircularBufferInit` and `TPCircularBufferCleanup` to allocate and free resources. Producing: Use `TPCircularBufferHead` to get a pointer to write to the buffer, followed by `TPCircularBufferProduce` to submit the written data. `TPCircularBufferProduceBytes` is a convenience routine for writing data straight to the buffer. Consuming: Use `TPCircularBufferTail` to get a pointer to the next data to read, followed by `TPCircularBufferConsume` to free up the space once processed. TPCircularBuffer+AudioBufferList.(c,h) contain helper functions to queue and dequeue AudioBufferList structures. These will automatically adjust the mData fields of each buffer to point to 16-byte aligned regions within the circular buffer. Thread safety ------------- As long as you restrict multithreaded access to just one producer, and just one consumer, this utility should be thread safe. Only one shared variable is used (the buffer fill count), and OSAtomic primitives are used to write to this value to ensure atomicity. License ------- Copyright (C) 2012-2013 A Tasty Pixel This software is provided 'as-is', without any express or implied warranty. In no event will the authors be held liable for any damages arising from the use of this software. Permission is granted to anyone to use this software for any purpose, including commercial applications, and to alter it and redistribute it freely, subject to the following restrictions: 1. The origin of this software must not be misrepresented; you must not claim that you wrote the original software. If you use this software in a product, an acknowledgment in the product documentation would be appreciated but is not required. 2. Altered source versions must be plainly marked as such, and must not be misrepresented as being the original software. 3. This notice may not be removed or altered from any source distribution. ----------------------------------------------------- Virtual memory technique originally proposed by [Philip Howard](http://vrb.slashusr.org/), and [adapted to Darwin](http://www.snoize.com/Code/PlayBufferedSoundFile.tar.gz) by [Kurt Revis](http://www.snoize.com) See more info at [atastypixel.com](http://atastypixel.com/blog/a-simple-fast-circular-buffer-implementation-for-audio-processing/) pcaudiolib-1.3/src/TPCircularBuffer/TPCircularBuffer+AudioBufferList.c000066400000000000000000000335371472657007400260010ustar00rootroot00000000000000// // TPCircularBuffer+AudioBufferList.c // Circular/Ring buffer implementation // // https://github.com/michaeltyson/TPCircularBuffer // // Created by Michael Tyson on 20/03/2012. // // Copyright (C) 2012-2013 A Tasty Pixel // // This software is provided 'as-is', without any express or implied // warranty. In no event will the authors be held liable for any damages // arising from the use of this software. // // Permission is granted to anyone to use this software for any purpose, // including commercial applications, and to alter it and redistribute it // freely, subject to the following restrictions: // // 1. The origin of this software must not be misrepresented; you must not // claim that you wrote the original software. If you use this software // in a product, an acknowledgment in the product documentation would be // appreciated but is not required. // // 2. Altered source versions must be plainly marked as such, and must not be // misrepresented as being the original software. // // 3. This notice may not be removed or altered from any source distribution. // #include "TPCircularBuffer+AudioBufferList.h" #include static double __secondsToHostTicks = 0.0; static inline long align16byte(long val) { if ( val & (16-1) ) { return val + (16 - (val & (16-1))); } return val; } static inline long min(long a, long b) { return a > b ? b : a; } AudioBufferList *TPCircularBufferPrepareEmptyAudioBufferList(TPCircularBuffer *buffer, int numberOfBuffers, int bytesPerBuffer, const AudioTimeStamp *inTimestamp) { int32_t availableBytes; TPCircularBufferABLBlockHeader *block = (TPCircularBufferABLBlockHeader*)TPCircularBufferHead(buffer, &availableBytes); if ( !block || availableBytes < sizeof(TPCircularBufferABLBlockHeader)+((numberOfBuffers-1)*sizeof(AudioBuffer))+(numberOfBuffers*bytesPerBuffer) ) return NULL; #ifdef DEBUG assert(!((unsigned long)block & 0xF) /* Beware unaligned accesses */); #endif if ( inTimestamp ) { memcpy(&block->timestamp, inTimestamp, sizeof(AudioTimeStamp)); } else { memset(&block->timestamp, 0, sizeof(AudioTimeStamp)); } memset(&block->bufferList, 0, sizeof(AudioBufferList)+((numberOfBuffers-1)*sizeof(AudioBuffer))); block->bufferList.mNumberBuffers = numberOfBuffers; char *dataPtr = (char*)&block->bufferList + sizeof(AudioBufferList)+((numberOfBuffers-1)*sizeof(AudioBuffer)); for ( int i=0; i availableBytes ) { return NULL; } block->bufferList.mBuffers[i].mData = dataPtr; block->bufferList.mBuffers[i].mDataByteSize = bytesPerBuffer; block->bufferList.mBuffers[i].mNumberChannels = 1; dataPtr += bytesPerBuffer; } // Make sure whole buffer (including timestamp and length value) is 16-byte aligned in length block->totalLength = (UInt32)align16byte(dataPtr - (char*)block); if ( block->totalLength > availableBytes ) { return NULL; } return &block->bufferList; } AudioBufferList *TPCircularBufferPrepareEmptyAudioBufferListWithAudioFormat(TPCircularBuffer *buffer, const AudioStreamBasicDescription *audioFormat, UInt32 frameCount, const AudioTimeStamp *timestamp) { return TPCircularBufferPrepareEmptyAudioBufferList(buffer, (audioFormat->mFormatFlags & kAudioFormatFlagIsNonInterleaved) ? audioFormat->mChannelsPerFrame : 1, audioFormat->mBytesPerFrame * frameCount, timestamp); } void TPCircularBufferProduceAudioBufferList(TPCircularBuffer *buffer, const AudioTimeStamp *inTimestamp) { int32_t availableBytes; TPCircularBufferABLBlockHeader *block = (TPCircularBufferABLBlockHeader*)TPCircularBufferHead(buffer, &availableBytes); assert(block); #ifdef DEBUG assert(!((unsigned long)block & 0xF) /* Beware unaligned accesses */); #endif assert(block->bufferList.mBuffers[0].mDataByteSize > 0); if ( inTimestamp ) { memcpy(&block->timestamp, inTimestamp, sizeof(AudioTimeStamp)); } UInt32 calculatedLength = (UInt32)(((char*)block->bufferList.mBuffers[block->bufferList.mNumberBuffers-1].mData + block->bufferList.mBuffers[block->bufferList.mNumberBuffers-1].mDataByteSize) - (char*)block); // Make sure whole buffer (including timestamp and length value) is 16-byte aligned in length calculatedLength = (UInt32)align16byte(calculatedLength); assert(calculatedLength <= block->totalLength && calculatedLength <= availableBytes); block->totalLength = calculatedLength; TPCircularBufferProduce(buffer, block->totalLength); } bool TPCircularBufferCopyAudioBufferList(TPCircularBuffer *buffer, const AudioBufferList *inBufferList, const AudioTimeStamp *inTimestamp, UInt32 frames, const AudioStreamBasicDescription *audioDescription) { if ( frames == 0 ) return true; int byteCount = inBufferList->mBuffers[0].mDataByteSize; if ( frames != kTPCircularBufferCopyAll ) { byteCount = frames * audioDescription->mBytesPerFrame; assert(byteCount <= inBufferList->mBuffers[0].mDataByteSize); } if ( byteCount == 0 ) return true; AudioBufferList *bufferList = TPCircularBufferPrepareEmptyAudioBufferList(buffer, inBufferList->mNumberBuffers, byteCount, inTimestamp); if ( !bufferList ) return false; for ( int i=0; imNumberBuffers; i++ ) { memcpy(bufferList->mBuffers[i].mData, inBufferList->mBuffers[i].mData, byteCount); } TPCircularBufferProduceAudioBufferList(buffer, NULL); return true; } AudioBufferList *TPCircularBufferNextBufferListAfter(TPCircularBuffer *buffer, const AudioBufferList *bufferList, AudioTimeStamp *outTimestamp) { int32_t availableBytes; void *tail = TPCircularBufferTail(buffer, &availableBytes); void *end = (char*)tail + availableBytes; assert((void*)bufferList > (void*)tail && (void*)bufferList < end); TPCircularBufferABLBlockHeader *originalBlock = (TPCircularBufferABLBlockHeader*)((char*)bufferList - offsetof(TPCircularBufferABLBlockHeader, bufferList)); #ifdef DEBUG assert(!((unsigned long)originalBlock & 0xF) /* Beware unaligned accesses */); #endif TPCircularBufferABLBlockHeader *nextBlock = (TPCircularBufferABLBlockHeader*)((char*)originalBlock + originalBlock->totalLength); if ( (void*)nextBlock >= end ) return NULL; #ifdef DEBUG assert(!((unsigned long)nextBlock & 0xF) /* Beware unaligned accesses */); #endif if ( outTimestamp ) { memcpy(outTimestamp, &nextBlock->timestamp, sizeof(AudioTimeStamp)); } return &nextBlock->bufferList; } void TPCircularBufferConsumeNextBufferListPartial(TPCircularBuffer *buffer, int framesToConsume, const AudioStreamBasicDescription *audioFormat) { assert(framesToConsume >= 0); int32_t dontcare; TPCircularBufferABLBlockHeader *block = (TPCircularBufferABLBlockHeader*)TPCircularBufferTail(buffer, &dontcare); if ( !block ) return; #ifdef DEBUG assert(!((unsigned long)block & 0xF)); // Beware unaligned accesses #endif int bytesToConsume = (int)min(framesToConsume * audioFormat->mBytesPerFrame, block->bufferList.mBuffers[0].mDataByteSize); if ( bytesToConsume == block->bufferList.mBuffers[0].mDataByteSize ) { TPCircularBufferConsumeNextBufferList(buffer); return; } for ( int i=0; ibufferList.mNumberBuffers; i++ ) { assert(bytesToConsume <= block->bufferList.mBuffers[i].mDataByteSize); block->bufferList.mBuffers[i].mData = (char*)block->bufferList.mBuffers[i].mData + bytesToConsume; block->bufferList.mBuffers[i].mDataByteSize -= bytesToConsume; } if ( block->timestamp.mFlags & kAudioTimeStampSampleTimeValid ) { block->timestamp.mSampleTime += framesToConsume; } if ( block->timestamp.mFlags & kAudioTimeStampHostTimeValid ) { if ( !__secondsToHostTicks ) { mach_timebase_info_data_t tinfo; mach_timebase_info(&tinfo); __secondsToHostTicks = 1.0 / (((double)tinfo.numer / tinfo.denom) * 1.0e-9); } block->timestamp.mHostTime += ((double)framesToConsume / audioFormat->mSampleRate) * __secondsToHostTicks; } // Reposition block forward, just before the audio data, ensuring 16-byte alignment TPCircularBufferABLBlockHeader *newBlock = (TPCircularBufferABLBlockHeader*)(((unsigned long)block + bytesToConsume) & ~0xFul); memmove(newBlock, block, sizeof(TPCircularBufferABLBlockHeader) + (block->bufferList.mNumberBuffers-1)*sizeof(AudioBuffer)); intptr_t bytesFreed = (intptr_t)newBlock - (intptr_t)block; newBlock->totalLength -= bytesFreed; TPCircularBufferConsume(buffer, (int32_t)bytesFreed); } void TPCircularBufferDequeueBufferListFrames(TPCircularBuffer *buffer, UInt32 *ioLengthInFrames, const AudioBufferList *outputBufferList, AudioTimeStamp *outTimestamp, const AudioStreamBasicDescription *audioFormat) { bool hasTimestamp = false; UInt32 bytesToGo = *ioLengthInFrames * audioFormat->mBytesPerFrame; UInt32 bytesCopied = 0; while ( bytesToGo > 0 ) { AudioBufferList *bufferList = TPCircularBufferNextBufferList(buffer, !hasTimestamp ? outTimestamp : NULL); if ( !bufferList ) break; hasTimestamp = true; long bytesToCopy = min(bytesToGo, bufferList->mBuffers[0].mDataByteSize); if ( outputBufferList ) { for ( int i=0; imNumberBuffers; i++ ) { assert(bytesCopied + bytesToCopy <= outputBufferList->mBuffers[i].mDataByteSize); memcpy((char*)outputBufferList->mBuffers[i].mData + bytesCopied, bufferList->mBuffers[i].mData, bytesToCopy); } } TPCircularBufferConsumeNextBufferListPartial(buffer, (int)bytesToCopy/audioFormat->mBytesPerFrame, audioFormat); bytesToGo -= bytesToCopy; bytesCopied += bytesToCopy; } *ioLengthInFrames -= bytesToGo / audioFormat->mBytesPerFrame; } UInt32 TPCircularBufferPeekContiguousWrapped(TPCircularBuffer *buffer, AudioTimeStamp *outTimestamp, const AudioStreamBasicDescription *audioFormat, UInt32 contiguousToleranceSampleTime, UInt32 wrapPoint) { int32_t availableBytes; TPCircularBufferABLBlockHeader *block = (TPCircularBufferABLBlockHeader*)TPCircularBufferTail(buffer, &availableBytes); if ( !block ) return 0; #ifdef DEBUG assert(!((unsigned long)block & 0xF) /* Beware unaligned accesses */); #endif if ( outTimestamp ) { memcpy(outTimestamp, &block->timestamp, sizeof(AudioTimeStamp)); } void *end = (char*)block + availableBytes; UInt32 byteCount = 0; while ( 1 ) { byteCount += block->bufferList.mBuffers[0].mDataByteSize; TPCircularBufferABLBlockHeader *nextBlock = (TPCircularBufferABLBlockHeader*)((char*)block + block->totalLength); if ( (void*)nextBlock >= end ) { break; } if ( contiguousToleranceSampleTime != UINT32_MAX ) { UInt32 frames = block->bufferList.mBuffers[0].mDataByteSize / audioFormat->mBytesPerFrame; Float64 nextTime = block->timestamp.mSampleTime + frames; if ( wrapPoint && nextTime > wrapPoint ) nextTime = fmod(nextTime, wrapPoint); Float64 diff = fabs(nextBlock->timestamp.mSampleTime - nextTime); if ( diff > contiguousToleranceSampleTime && (!wrapPoint || fabs(diff-wrapPoint) > contiguousToleranceSampleTime) ) { break; } } #ifdef DEBUG assert(!((unsigned long)nextBlock & 0xF) /* Beware unaligned accesses */); #endif block = nextBlock; } return byteCount / audioFormat->mBytesPerFrame; } UInt32 TPCircularBufferPeek(TPCircularBuffer *buffer, AudioTimeStamp *outTimestamp, const AudioStreamBasicDescription *audioFormat) { return TPCircularBufferPeekContiguousWrapped(buffer, outTimestamp, audioFormat, UINT32_MAX, 0); } UInt32 TPCircularBufferPeekContiguous(TPCircularBuffer *buffer, AudioTimeStamp *outTimestamp, const AudioStreamBasicDescription *audioFormat, UInt32 contiguousToleranceSampleTime) { return TPCircularBufferPeekContiguousWrapped(buffer, outTimestamp, audioFormat, contiguousToleranceSampleTime, 0); } UInt32 TPCircularBufferGetAvailableSpace(TPCircularBuffer *buffer, const AudioStreamBasicDescription *audioFormat) { // Look at buffer head; make sure there's space for the block metadata int32_t availableBytes; TPCircularBufferABLBlockHeader *block = (TPCircularBufferABLBlockHeader*)TPCircularBufferHead(buffer, &availableBytes); if ( !block ) return 0; #ifdef DEBUG assert(!((unsigned long)block & 0xF) /* Beware unaligned accesses */); #endif // Now find out how much 16-byte aligned audio we can store in the space available int numberOfBuffers = audioFormat->mFormatFlags & kAudioFormatFlagIsNonInterleaved ? audioFormat->mChannelsPerFrame : 1; char * endOfBuffer = (char*)block + availableBytes; char * dataPtr = (char*)align16byte((long)(&block->bufferList + sizeof(AudioBufferList)+((numberOfBuffers-1)*sizeof(AudioBuffer)))); if ( dataPtr >= endOfBuffer ) return 0; int32_t availableAudioBytes = (int)(endOfBuffer - dataPtr); int32_t availableAudioBytesPerBuffer = availableAudioBytes / numberOfBuffers; availableAudioBytesPerBuffer -= (availableAudioBytesPerBuffer % (16-1)); return availableAudioBytesPerBuffer > 0 ? availableAudioBytesPerBuffer / audioFormat->mBytesPerFrame : 0; } pcaudiolib-1.3/src/TPCircularBuffer/TPCircularBuffer+AudioBufferList.h000066400000000000000000000261571472657007400260060ustar00rootroot00000000000000// // TPCircularBuffer+AudioBufferList.h // Circular/Ring buffer implementation // // https://github.com/michaeltyson/TPCircularBuffer // // Created by Michael Tyson on 20/03/2012. // // Copyright (C) 2012-2013 A Tasty Pixel // // This software is provided 'as-is', without any express or implied // warranty. In no event will the authors be held liable for any damages // arising from the use of this software. // // Permission is granted to anyone to use this software for any purpose, // including commercial applications, and to alter it and redistribute it // freely, subject to the following restrictions: // // 1. The origin of this software must not be misrepresented; you must not // claim that you wrote the original software. If you use this software // in a product, an acknowledgment in the product documentation would be // appreciated but is not required. // // 2. Altered source versions must be plainly marked as such, and must not be // misrepresented as being the original software. // // 3. This notice may not be removed or altered from any source distribution. // #ifndef TPCircularBuffer_AudioBufferList_h #define TPCircularBuffer_AudioBufferList_h #ifdef __cplusplus extern "C" { #endif #include "TPCircularBuffer.h" #include #define kTPCircularBufferCopyAll UINT32_MAX typedef struct { AudioTimeStamp timestamp; UInt32 totalLength; AudioBufferList bufferList; } TPCircularBufferABLBlockHeader; /*! * Prepare an empty buffer list, stored on the circular buffer * * @param buffer Circular buffer * @param numberOfBuffers The number of buffers to be contained within the buffer list * @param bytesPerBuffer The number of bytes to store for each buffer * @param timestamp The timestamp associated with the buffer, or NULL. Note that you can also pass a timestamp into TPCircularBufferProduceAudioBufferList, to set it there instead. * @return The empty buffer list, or NULL if circular buffer has insufficient space */ AudioBufferList *TPCircularBufferPrepareEmptyAudioBufferList(TPCircularBuffer *buffer, int numberOfBuffers, int bytesPerBuffer, const AudioTimeStamp *timestamp); /*! * Prepare an empty buffer list, stored on the circular buffer, using an audio description to automatically configure buffer * * @param buffer Circular buffer * @param audioFormat The kind of audio that will be stored * @param frameCount The number of frames that will be stored * @param timestamp The timestamp associated with the buffer, or NULL. Note that you can also pass a timestamp into TPCircularBufferProduceAudioBufferList, to set it there instead. * @return The empty buffer list, or NULL if circular buffer has insufficient space */ AudioBufferList *TPCircularBufferPrepareEmptyAudioBufferListWithAudioFormat(TPCircularBuffer *buffer, const AudioStreamBasicDescription *audioFormat, UInt32 frameCount, const AudioTimeStamp *timestamp); /*! * Mark next audio buffer list as ready for reading * * This marks the audio buffer list prepared using TPCircularBufferPrepareEmptyAudioBufferList * as ready for reading. You must not call this function without first calling * TPCircularBufferPrepareEmptyAudioBufferList. * * @param buffer Circular buffer * @param inTimestamp The timestamp associated with the buffer, or NULL to leave as-is. Note that you can also pass a timestamp into TPCircularBufferPrepareEmptyAudioBufferList, to set it there instead. */ void TPCircularBufferProduceAudioBufferList(TPCircularBuffer *buffer, const AudioTimeStamp *inTimestamp); /*! * Copy the audio buffer list onto the buffer * * @param buffer Circular buffer * @param bufferList Buffer list containing audio to copy to buffer * @param timestamp The timestamp associated with the buffer, or NULL * @param frames Length of audio in frames. Specify kTPCircularBufferCopyAll to copy the whole buffer (audioFormat can be NULL, in this case) * @param audioFormat The AudioStreamBasicDescription describing the audio, or NULL if you specify kTPCircularBufferCopyAll to the `frames` argument * @return YES if buffer list was successfully copied; NO if there was insufficient space */ bool TPCircularBufferCopyAudioBufferList(TPCircularBuffer *buffer, const AudioBufferList *bufferList, const AudioTimeStamp *timestamp, UInt32 frames, const AudioStreamBasicDescription *audioFormat); /*! * Get a pointer to the next stored buffer list * * @param buffer Circular buffer * @param outTimestamp On output, if not NULL, the timestamp corresponding to the buffer * @return Pointer to the next buffer list in the buffer */ static __inline__ __attribute__((always_inline)) AudioBufferList *TPCircularBufferNextBufferList(TPCircularBuffer *buffer, AudioTimeStamp *outTimestamp) { int32_t dontcare; // Length of segment is contained within buffer list, so we can ignore this TPCircularBufferABLBlockHeader *block = (TPCircularBufferABLBlockHeader*)TPCircularBufferTail(buffer, &dontcare); if ( !block ) { if ( outTimestamp ) { memset(outTimestamp, 0, sizeof(AudioTimeStamp)); } return NULL; } if ( outTimestamp ) { memcpy(outTimestamp, &block->timestamp, sizeof(AudioTimeStamp)); } return &block->bufferList; } /*! * Get a pointer to the next stored buffer list after the given one * * @param buffer Circular buffer * @param bufferList Preceding buffer list * @param outTimestamp On output, if not NULL, the timestamp corresponding to the buffer * @return Pointer to the next buffer list in the buffer, or NULL */ AudioBufferList *TPCircularBufferNextBufferListAfter(TPCircularBuffer *buffer, const AudioBufferList *bufferList, AudioTimeStamp *outTimestamp); /*! * Consume the next buffer list * * @param buffer Circular buffer */ static __inline__ __attribute__((always_inline)) void TPCircularBufferConsumeNextBufferList(TPCircularBuffer *buffer) { int32_t dontcare; TPCircularBufferABLBlockHeader *block = (TPCircularBufferABLBlockHeader*)TPCircularBufferTail(buffer, &dontcare); if ( !block ) return; TPCircularBufferConsume(buffer, block->totalLength); } /*! * Consume a portion of the next buffer list * * This will also increment the sample time and host time portions of the timestamp of * the buffer list, if present. * * @param buffer Circular buffer * @param framesToConsume The number of frames to consume from the buffer list * @param audioFormat The AudioStreamBasicDescription describing the audio */ void TPCircularBufferConsumeNextBufferListPartial(TPCircularBuffer *buffer, int framesToConsume, const AudioStreamBasicDescription *audioFormat); /*! * Consume a certain number of frames from the buffer, possibly from multiple queued buffer lists * * Copies the given number of frames from the buffer into outputBufferList, of the * given audio description, then consumes the audio buffers. If an audio buffer has * not been entirely consumed, then updates the queued buffer list structure to point * to the unconsumed data only. * * @param buffer Circular buffer * @param ioLengthInFrames On input, the number of frames in the given audio format to consume; on output, the number of frames provided * @param outputBufferList The buffer list to copy audio to, or NULL to discard audio. If not NULL, the structure must be initialised properly, and the mData pointers must not be NULL. * @param outTimestamp On output, if not NULL, the timestamp corresponding to the first audio frame returned * @param audioFormat The format of the audio stored in the buffer */ void TPCircularBufferDequeueBufferListFrames(TPCircularBuffer *buffer, UInt32 *ioLengthInFrames, const AudioBufferList *outputBufferList, AudioTimeStamp *outTimestamp, const AudioStreamBasicDescription *audioFormat); /*! * Determine how many frames of audio are buffered * * Given the provided audio format, determines the frame count of all queued buffers * * Note: This function should only be used on the consumer thread, not the producer thread. * * @param buffer Circular buffer * @param outTimestamp On output, if not NULL, the timestamp corresponding to the first audio frame * @param audioFormat The format of the audio stored in the buffer * @return The number of frames in the given audio format that are in the buffer */ UInt32 TPCircularBufferPeek(TPCircularBuffer *buffer, AudioTimeStamp *outTimestamp, const AudioStreamBasicDescription *audioFormat); /*! * Determine how many contiguous frames of audio are buffered * * Given the provided audio format, determines the frame count of all queued buffers that are contiguous, * given their corresponding timestamps (sample time). * * Note: This function should only be used on the consumer thread, not the producer thread. * * @param buffer Circular buffer * @param outTimestamp On output, if not NULL, the timestamp corresponding to the first audio frame * @param audioFormat The format of the audio stored in the buffer * @param contiguousToleranceSampleTime The number of samples of discrepancy to tolerate * @return The number of frames in the given audio format that are in the buffer */ UInt32 TPCircularBufferPeekContiguous(TPCircularBuffer *buffer, AudioTimeStamp *outTimestamp, const AudioStreamBasicDescription *audioFormat, UInt32 contiguousToleranceSampleTime); /*! * Determine how many contiguous frames of audio are buffered, with wrap around * * Like TPCircularBufferPeekContiguous, determines how many contiguous frames are buffered, * but considers audio that wraps around a region of a given length as also contiguous. This * is good for audio that loops. * * Note: This function should only be used on the consumer thread, not the producer thread. * * @param buffer Circular buffer * @param outTimestamp On output, if not NULL, the timestamp corresponding to the first audio frame * @param audioFormat The format of the audio stored in the buffer * @param contiguousToleranceSampleTime The number of samples of discrepancy to tolerate * @param wrapPoint The point around which the audio may wrap and still be considered contiguous, or 0 to disable * @return The number of frames in the given audio format that are in the buffer */ UInt32 TPCircularBufferPeekContiguousWrapped(TPCircularBuffer *buffer, AudioTimeStamp *outTimestamp, const AudioStreamBasicDescription *audioFormat, UInt32 contiguousToleranceSampleTime, UInt32 wrapPoint); /*! * Determine how many much space there is in the buffer * * Given the provided audio format, determines the number of frames of audio that can be buffered. * * Note: This function should only be used on the producer thread, not the consumer thread. * * @param buffer Circular buffer * @param audioFormat The format of the audio stored in the buffer * @return The number of frames in the given audio format that can be stored in the buffer */ UInt32 TPCircularBufferGetAvailableSpace(TPCircularBuffer *buffer, const AudioStreamBasicDescription *audioFormat); #ifdef __cplusplus } #endif #endif pcaudiolib-1.3/src/TPCircularBuffer/TPCircularBuffer.c000066400000000000000000000135531472657007400227520ustar00rootroot00000000000000// // TPCircularBuffer.c // Circular/Ring buffer implementation // // https://github.com/michaeltyson/TPCircularBuffer // // Created by Michael Tyson on 10/12/2011. // // Copyright (C) 2012-2013 A Tasty Pixel // // This software is provided 'as-is', without any express or implied // warranty. In no event will the authors be held liable for any damages // arising from the use of this software. // // Permission is granted to anyone to use this software for any purpose, // including commercial applications, and to alter it and redistribute it // freely, subject to the following restrictions: // // 1. The origin of this software must not be misrepresented; you must not // claim that you wrote the original software. If you use this software // in a product, an acknowledgment in the product documentation would be // appreciated but is not required. // // 2. Altered source versions must be plainly marked as such, and must not be // misrepresented as being the original software. // // 3. This notice may not be removed or altered from any source distribution. // #include "TPCircularBuffer.h" #include #include #include #define reportResult(result,operation) (_reportResult((result),(operation),strrchr(__FILE__, '/')+1,__LINE__)) static inline bool _reportResult(kern_return_t result, const char *operation, const char* file, int line) { if ( result != ERR_SUCCESS ) { printf("%s:%d: %s: %s\n", file, line, operation, mach_error_string(result)); return false; } return true; } bool _TPCircularBufferInit(TPCircularBuffer *buffer, int32_t length, size_t structSize) { assert(length > 0); if ( structSize != sizeof(TPCircularBuffer) ) { fprintf(stderr, "TPCircularBuffer: Header version mismatch. Check for old versions of TPCircularBuffer in your project\n"); abort(); } // Keep trying until we get our buffer, needed to handle race conditions int retries = 3; while ( true ) { buffer->length = (int32_t)round_page(length); // We need whole page sizes // Temporarily allocate twice the length, so we have the contiguous address space to // support a second instance of the buffer directly after vm_address_t bufferAddress; kern_return_t result = vm_allocate(mach_task_self(), &bufferAddress, buffer->length * 2, VM_FLAGS_ANYWHERE); // allocate anywhere it'll fit if ( result != ERR_SUCCESS ) { if ( retries-- == 0 ) { reportResult(result, "Buffer allocation"); return false; } // Try again if we fail continue; } // Now replace the second half of the allocation with a virtual copy of the first half. Deallocate the second half... result = vm_deallocate(mach_task_self(), bufferAddress + buffer->length, buffer->length); if ( result != ERR_SUCCESS ) { if ( retries-- == 0 ) { reportResult(result, "Buffer deallocation"); return false; } // If this fails somehow, deallocate the whole region and try again vm_deallocate(mach_task_self(), bufferAddress, buffer->length); continue; } // Re-map the buffer to the address space immediately after the buffer vm_address_t virtualAddress = bufferAddress + buffer->length; vm_prot_t cur_prot, max_prot; result = vm_remap(mach_task_self(), &virtualAddress, // mirror target buffer->length, // size of mirror 0, // auto alignment 0, // force remapping to virtualAddress mach_task_self(), // same task bufferAddress, // mirror source 0, // MAP READ-WRITE, NOT COPY &cur_prot, // unused protection struct &max_prot, // unused protection struct VM_INHERIT_DEFAULT); if ( result != ERR_SUCCESS ) { if ( retries-- == 0 ) { reportResult(result, "Remap buffer memory"); return false; } // If this remap failed, we hit a race condition, so deallocate and try again vm_deallocate(mach_task_self(), bufferAddress, buffer->length); continue; } if ( virtualAddress != bufferAddress+buffer->length ) { // If the memory is not contiguous, clean up both allocated buffers and try again if ( retries-- == 0 ) { printf("Couldn't map buffer memory to end of buffer\n"); return false; } vm_deallocate(mach_task_self(), virtualAddress, buffer->length); vm_deallocate(mach_task_self(), bufferAddress, buffer->length); continue; } buffer->buffer = (void*)bufferAddress; buffer->fillCount = 0; buffer->head = buffer->tail = 0; buffer->atomic = true; return true; } return false; } void TPCircularBufferCleanup(TPCircularBuffer *buffer) { vm_deallocate(mach_task_self(), (vm_address_t)buffer->buffer, buffer->length * 2); memset(buffer, 0, sizeof(TPCircularBuffer)); } void TPCircularBufferClear(TPCircularBuffer *buffer) { int32_t fillCount; if ( TPCircularBufferTail(buffer, &fillCount) ) { TPCircularBufferConsume(buffer, fillCount); } } void TPCircularBufferSetAtomic(TPCircularBuffer *buffer, bool atomic) { buffer->atomic = atomic; } pcaudiolib-1.3/src/TPCircularBuffer/TPCircularBuffer.h000066400000000000000000000177101472657007400227560ustar00rootroot00000000000000// // TPCircularBuffer.h // Circular/Ring buffer implementation // // https://github.com/michaeltyson/TPCircularBuffer // // Created by Michael Tyson on 10/12/2011. // // // This implementation makes use of a virtual memory mapping technique that inserts a virtual copy // of the buffer memory directly after the buffer's end, negating the need for any buffer wrap-around // logic. Clients can simply use the returned memory address as if it were contiguous space. // // The implementation is thread-safe in the case of a single producer and single consumer. // // Virtual memory technique originally proposed by Philip Howard (http://vrb.slashusr.org/), and // adapted to Darwin by Kurt Revis (http://www.snoize.com, // http://www.snoize.com/Code/PlayBufferedSoundFile.tar.gz) // // // Copyright (C) 2012-2013 A Tasty Pixel // // This software is provided 'as-is', without any express or implied // warranty. In no event will the authors be held liable for any damages // arising from the use of this software. // // Permission is granted to anyone to use this software for any purpose, // including commercial applications, and to alter it and redistribute it // freely, subject to the following restrictions: // // 1. The origin of this software must not be misrepresented; you must not // claim that you wrote the original software. If you use this software // in a product, an acknowledgment in the product documentation would be // appreciated but is not required. // // 2. Altered source versions must be plainly marked as such, and must not be // misrepresented as being the original software. // // 3. This notice may not be removed or altered from any source distribution. // #ifndef TPCircularBuffer_h #define TPCircularBuffer_h #include #include #include #include #ifdef __APPLE__ #include #ifndef __deprecated_msg #define __deprecated_msg(_msg) __attribute__((__deprecated__(_msg))) #endif #endif #ifdef __cplusplus extern "C++" { #include using namespace std; } #else #include #endif #ifdef __cplusplus extern "C" { #endif typedef struct { void *buffer; int32_t length; int32_t tail; int32_t head; volatile atomic_int fillCount; bool atomic; } TPCircularBuffer; /*! * Initialise buffer * * Note that the length is advisory only: Because of the way the * memory mirroring technique works, the true buffer length will * be multiples of the device page size (e.g. 4096 bytes) * * If you intend to use the AudioBufferList utilities, you should * always allocate a bit more space than you need for pure audio * data, so there's room for the metadata. How much extra is required * depends on how many AudioBufferList structures are used, which is * a function of how many audio frames each buffer holds. A good rule * of thumb is to add 15%, or at least another 2048 bytes or so. * * @param buffer Circular buffer * @param length Length of buffer */ #define TPCircularBufferInit(buffer, length) \ _TPCircularBufferInit(buffer, length, sizeof(*buffer)) bool _TPCircularBufferInit(TPCircularBuffer *buffer, int32_t length, size_t structSize); /*! * Cleanup buffer * * Releases buffer resources. */ void TPCircularBufferCleanup(TPCircularBuffer *buffer); /*! * Clear buffer * * Resets buffer to original, empty state. * * This is safe for use by consumer while producer is accessing * buffer. */ void TPCircularBufferClear(TPCircularBuffer *buffer); /*! * Set the atomicity * * If you set the atomiticy to false using this method, the buffer will * not use atomic operations. This can be used to give the compiler a little * more optimisation opportunities when the buffer is only used on one thread. * * Important note: Only set this to false if you know what you're doing! * * The default value is true (the buffer will use atomic operations) * * @param buffer Circular buffer * @param atomic Whether the buffer is atomic (default true) */ void TPCircularBufferSetAtomic(TPCircularBuffer *buffer, bool atomic); // Reading (consuming) /*! * Access end of buffer * * This gives you a pointer to the end of the buffer, ready * for reading, and the number of available bytes to read. * * @param buffer Circular buffer * @param availableBytes On output, the number of bytes ready for reading * @return Pointer to the first bytes ready for reading, or NULL if buffer is empty */ static __inline__ __attribute__((always_inline)) void* TPCircularBufferTail(TPCircularBuffer *buffer, int32_t* availableBytes) { *availableBytes = buffer->fillCount; if ( *availableBytes == 0 ) return NULL; return (void*)((char*)buffer->buffer + buffer->tail); } /*! * Consume bytes in buffer * * This frees up the just-read bytes, ready for writing again. * * @param buffer Circular buffer * @param amount Number of bytes to consume */ static __inline__ __attribute__((always_inline)) void TPCircularBufferConsume(TPCircularBuffer *buffer, int32_t amount) { buffer->tail = (buffer->tail + amount) % buffer->length; if ( buffer->atomic ) { atomic_fetch_add(&buffer->fillCount, -amount); } else { buffer->fillCount -= amount; } assert(buffer->fillCount >= 0); } /*! * Access front of buffer * * This gives you a pointer to the front of the buffer, ready * for writing, and the number of available bytes to write. * * @param buffer Circular buffer * @param availableBytes On output, the number of bytes ready for writing * @return Pointer to the first bytes ready for writing, or NULL if buffer is full */ static __inline__ __attribute__((always_inline)) void* TPCircularBufferHead(TPCircularBuffer *buffer, int32_t* availableBytes) { *availableBytes = (buffer->length - buffer->fillCount); if ( *availableBytes == 0 ) return NULL; return (void*)((char*)buffer->buffer + buffer->head); } // Writing (producing) /*! * Produce bytes in buffer * * This marks the given section of the buffer ready for reading. * * @param buffer Circular buffer * @param amount Number of bytes to produce */ static __inline__ __attribute__((always_inline)) void TPCircularBufferProduce(TPCircularBuffer *buffer, int32_t amount) { buffer->head = (buffer->head + amount) % buffer->length; if ( buffer->atomic ) { atomic_fetch_add(&buffer->fillCount, amount); } else { buffer->fillCount += amount; } assert(buffer->fillCount <= buffer->length); } /*! * Helper routine to copy bytes to buffer * * This copies the given bytes to the buffer, and marks them ready for reading. * * @param buffer Circular buffer * @param src Source buffer * @param len Number of bytes in source buffer * @return true if bytes copied, false if there was insufficient space */ static __inline__ __attribute__((always_inline)) bool TPCircularBufferProduceBytes(TPCircularBuffer *buffer, const void* src, int32_t len) { int32_t space; void *ptr = TPCircularBufferHead(buffer, &space); if ( space < len ) return false; memcpy(ptr, src, len); TPCircularBufferProduce(buffer, len); return true; } /*! * Deprecated method */ static __inline__ __attribute__((always_inline)) __deprecated_msg("use TPCircularBufferSetAtomic(false) and TPCircularBufferConsume instead") void TPCircularBufferConsumeNoBarrier(TPCircularBuffer *buffer, int32_t amount) { buffer->tail = (buffer->tail + amount) % buffer->length; buffer->fillCount -= amount; assert(buffer->fillCount >= 0); } /*! * Deprecated method */ static __inline__ __attribute__((always_inline)) __deprecated_msg("use TPCircularBufferSetAtomic(false) and TPCircularBufferProduce instead") void TPCircularBufferProduceNoBarrier(TPCircularBuffer *buffer, int32_t amount) { buffer->head = (buffer->head + amount) % buffer->length; buffer->fillCount += amount; assert(buffer->fillCount <= buffer->length); } #ifdef __cplusplus } #endif #endif pcaudiolib-1.3/src/TPCircularBuffer/TPCircularBuffer.podspec000066400000000000000000000032601472657007400241570ustar00rootroot00000000000000license = <<-END 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." END Pod::Spec.new do |s| s.name = "TPCircularBuffer" s.version = '1.4' s.summary = 'A simple, fast circular buffer implementation.' s.homepage = 'https://github.com/michaeltyson/TPCircularBuffer' s.authors = { 'Michael Tyson' => 'michael@atastypixel.com' } s.license = { :type => 'MIT', :text => license } s.source = { :git => 'https://github.com/michaeltyson/TPCircularBuffer.git', :tag => '1.4' } s.source_files = '*.{c,h}' s.requires_arc = false s.frameworks = 'AudioToolbox' s.ios.deployment_target = '4.3' s.osx.deployment_target = '10.8' end pcaudiolib-1.3/src/alsa.c000066400000000000000000000210611472657007400153570ustar00rootroot00000000000000/* ALSA Output. * * Copyright (C) 2016 Reece H. Dunn * * This file is part of pcaudiolib. * * pcaudiolib is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. * * pcaudiolib 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 General Public License for more details. * * You should have received a copy of the GNU General Public License * along with pcaudiolib. If not, see . */ #include "config.h" #include "audio_priv.h" #ifdef HAVE_ALSA_ASOUNDLIB_H #include #include #include struct alsa_object { struct audio_object vtable; snd_pcm_t *handle; uint8_t sample_size; char *device; /* saved audio_object_open parameters */ int is_open; enum audio_object_format format; uint32_t rate; uint8_t channels; }; #define to_alsa_object(object) container_of(object, struct alsa_object, vtable) int alsa_object_open(struct audio_object *object, enum audio_object_format format, uint32_t rate, uint8_t channels) { struct alsa_object *self = to_alsa_object(object); if (self->handle) return -EEXIST; snd_pcm_format_t pcm_format; #define FORMAT(srcfmt, dstfmt, size) case srcfmt: pcm_format = dstfmt; self->sample_size = size*channels; break; switch (format) { FORMAT(AUDIO_OBJECT_FORMAT_ALAW, SND_PCM_FORMAT_A_LAW, 1) FORMAT(AUDIO_OBJECT_FORMAT_ULAW, SND_PCM_FORMAT_MU_LAW, 1) FORMAT(AUDIO_OBJECT_FORMAT_S8, SND_PCM_FORMAT_S8, 1) FORMAT(AUDIO_OBJECT_FORMAT_U8, SND_PCM_FORMAT_U8, 1) FORMAT(AUDIO_OBJECT_FORMAT_S16LE, SND_PCM_FORMAT_S16_LE, 2) FORMAT(AUDIO_OBJECT_FORMAT_S16BE, SND_PCM_FORMAT_S16_BE, 2) FORMAT(AUDIO_OBJECT_FORMAT_U16LE, SND_PCM_FORMAT_U16_LE, 2) FORMAT(AUDIO_OBJECT_FORMAT_U16BE, SND_PCM_FORMAT_U16_BE, 2) FORMAT(AUDIO_OBJECT_FORMAT_S18LE, SND_PCM_FORMAT_S18_3LE, 3) FORMAT(AUDIO_OBJECT_FORMAT_S18BE, SND_PCM_FORMAT_S18_3BE, 3) FORMAT(AUDIO_OBJECT_FORMAT_U18LE, SND_PCM_FORMAT_U18_3LE, 3) FORMAT(AUDIO_OBJECT_FORMAT_U18BE, SND_PCM_FORMAT_U18_3BE, 3) FORMAT(AUDIO_OBJECT_FORMAT_S20LE, SND_PCM_FORMAT_S20_3LE, 3) FORMAT(AUDIO_OBJECT_FORMAT_S20BE, SND_PCM_FORMAT_S20_3BE, 3) FORMAT(AUDIO_OBJECT_FORMAT_U20LE, SND_PCM_FORMAT_U20_3LE, 3) FORMAT(AUDIO_OBJECT_FORMAT_U20BE, SND_PCM_FORMAT_U20_3BE, 3) FORMAT(AUDIO_OBJECT_FORMAT_S24LE, SND_PCM_FORMAT_S24_3LE, 3) FORMAT(AUDIO_OBJECT_FORMAT_S24BE, SND_PCM_FORMAT_S24_3BE, 3) FORMAT(AUDIO_OBJECT_FORMAT_U24LE, SND_PCM_FORMAT_U24_3LE, 3) FORMAT(AUDIO_OBJECT_FORMAT_U24BE, SND_PCM_FORMAT_U24_3BE, 3) FORMAT(AUDIO_OBJECT_FORMAT_S24_32LE, SND_PCM_FORMAT_S24_LE, 4) FORMAT(AUDIO_OBJECT_FORMAT_S24_32BE, SND_PCM_FORMAT_S24_BE, 4) FORMAT(AUDIO_OBJECT_FORMAT_U24_32LE, SND_PCM_FORMAT_U24_LE, 4) FORMAT(AUDIO_OBJECT_FORMAT_U24_32BE, SND_PCM_FORMAT_U24_BE, 4) FORMAT(AUDIO_OBJECT_FORMAT_S32LE, SND_PCM_FORMAT_S32_LE, 4) FORMAT(AUDIO_OBJECT_FORMAT_S32BE, SND_PCM_FORMAT_S32_BE, 4) FORMAT(AUDIO_OBJECT_FORMAT_U32LE, SND_PCM_FORMAT_U32_LE, 4) FORMAT(AUDIO_OBJECT_FORMAT_U32BE, SND_PCM_FORMAT_U32_BE, 4) FORMAT(AUDIO_OBJECT_FORMAT_FLOAT32LE, SND_PCM_FORMAT_FLOAT_LE, 4) FORMAT(AUDIO_OBJECT_FORMAT_FLOAT32BE, SND_PCM_FORMAT_FLOAT_BE, 4) FORMAT(AUDIO_OBJECT_FORMAT_FLOAT64LE, SND_PCM_FORMAT_FLOAT64_LE, 8) FORMAT(AUDIO_OBJECT_FORMAT_FLOAT64BE, SND_PCM_FORMAT_FLOAT64_BE, 8) FORMAT(AUDIO_OBJECT_FORMAT_IEC958LE, SND_PCM_FORMAT_IEC958_SUBFRAME_LE, 1) FORMAT(AUDIO_OBJECT_FORMAT_IEC958BE, SND_PCM_FORMAT_IEC958_SUBFRAME_BE, 1) FORMAT(AUDIO_OBJECT_FORMAT_ADPCM, SND_PCM_FORMAT_IMA_ADPCM, 1) FORMAT(AUDIO_OBJECT_FORMAT_MPEG, SND_PCM_FORMAT_MPEG, 1) FORMAT(AUDIO_OBJECT_FORMAT_GSM, SND_PCM_FORMAT_GSM, 1) default: return -EINVAL; } #undef FORMAT snd_pcm_hw_params_t *params = NULL; snd_pcm_hw_params_malloc(¶ms); unsigned int period_time = LATENCY * 1000; int dir = 0; int err = 0; if ((err = snd_pcm_open(&self->handle, self->device ? self->device : "default", SND_PCM_STREAM_PLAYBACK, 0)) < 0) goto error; if ((err = snd_pcm_hw_params_any(self->handle, params)) < 0) goto error; if ((err = snd_pcm_hw_params_set_access(self->handle, params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) goto error; if ((err = snd_pcm_hw_params_set_format(self->handle, params, pcm_format)) < 0) goto error; if ((err = snd_pcm_hw_params_set_rate_near(self->handle, params, &rate, 0)) < 0) goto error; if ((err = snd_pcm_hw_params_set_channels(self->handle, params, channels)) < 0) goto error; if ((err = snd_pcm_hw_params_set_period_time_near(self->handle, params, &period_time, &dir)) < 0) goto error; if ((err = snd_pcm_hw_params(self->handle, params)) < 0) goto error; if ((err = snd_pcm_prepare(self->handle)) < 0) goto error; self->is_open = 1; self->format = format; self->rate = rate; self->channels = channels; return 0; error: if (params) snd_pcm_hw_params_free(params); if (self->handle) { snd_pcm_close(self->handle); self->handle = NULL; self->is_open = 0; } return err; } void alsa_object_close(struct audio_object *object) { struct alsa_object *self = to_alsa_object(object); if (self->handle) { snd_pcm_close(self->handle); self->handle = NULL; self->is_open = 1; } } void alsa_object_destroy(struct audio_object *object) { struct alsa_object *self = to_alsa_object(object); free(self->device); free(self); } int alsa_object_drain(struct audio_object *object) { struct alsa_object *self = to_alsa_object(object); int ret = 0; if (self->handle) { snd_pcm_drain(self->handle); ret = snd_pcm_prepare(self->handle); } return ret; } int alsa_object_flush(struct audio_object *object) { struct alsa_object *self = to_alsa_object(object); if (!self) return 0; // Using snd_pcm_drop does not discard the audio, so reopen the device // to reset the sound buffer. if (self->is_open) { audio_object_close(object); return audio_object_open(object, self->format, self->rate, self->channels); } return 0; } int alsa_object_write(struct audio_object *object, const void *data, size_t bytes) { struct alsa_object *self = to_alsa_object(object); if (!self->handle) return 0; int err = 0; snd_pcm_uframes_t nToWrite = bytes / self->sample_size; // Number of frames to write. snd_pcm_sframes_t nWritten = 0; // And number alsa actually wrote. while (1) { nWritten = snd_pcm_writei(self->handle, data, nToWrite); if ((nWritten >= 0) && (nWritten < nToWrite)) { // Can happen in case of a signal or underrun. nToWrite -= nWritten; data += nWritten * self->sample_size; // Open question: if a signal caused the short read, should we snd_pcm_prepare? } else if ((nWritten == -EPIPE) #ifdef EBADFD || (nWritten == -EBADFD) #endif ) { // Either there was an underrun or the PCM was in a bad state. err = snd_pcm_prepare(self->handle); if (err != 0) break; } #ifdef ESTRPIPE else if (nWritten == -ESTRPIPE) { // Sound suspended, try to resume. do { err = snd_pcm_resume(self->handle); sleep(1); } while (err == -EAGAIN); if (err == -ENOSYS) { // Hardware doesn't support "fine resume". // So just prepare. err = snd_pcm_prepare(self->handle); } if (err < 0) { break; } } #endif else { err = nWritten; break; } } return err >= 0 ? 0 : err; } const char * alsa_object_strerror(struct audio_object *object, int error) { return snd_strerror(error); } struct audio_object * create_alsa_object(const char *device, const char *application_name, const char *description) { struct alsa_object *self = malloc(sizeof(struct alsa_object)); if (!self) return NULL; self->handle = NULL; self->sample_size = 0; self->device = device ? strdup(device) : NULL; self->is_open = 0; self->vtable.open = alsa_object_open; self->vtable.close = alsa_object_close; self->vtable.destroy = alsa_object_destroy; self->vtable.write = alsa_object_write; self->vtable.drain = alsa_object_drain; self->vtable.flush = alsa_object_flush; self->vtable.strerror = alsa_object_strerror; return &self->vtable; } #else struct audio_object * create_alsa_object(const char *device, const char *application_name, const char *description) { return NULL; } #endif pcaudiolib-1.3/src/audio.c000066400000000000000000000053241472657007400155440ustar00rootroot00000000000000/* Public API Implementation. * * Copyright (C) 2016 Reece H. Dunn * * This file is part of pcaudiolib. * * pcaudiolib is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. * * pcaudiolib 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 General Public License for more details. * * You should have received a copy of the GNU General Public License * along with pcaudiolib. If not, see . */ #include "config.h" #include "audio_priv.h" int audio_object_open(struct audio_object *object, enum audio_object_format format, uint32_t rate, uint8_t channels) { if (object) return object->open(object, format, rate, channels); return 0; } void audio_object_close(struct audio_object *object) { if (object) object->close(object); } void audio_object_destroy(struct audio_object *object) { if (object) object->destroy(object); } int audio_object_write(struct audio_object *object, const void *data, size_t bytes) { if (object) return object->write(object, data, bytes); return 0; } int audio_object_drain(struct audio_object *object) { if (object) return object->drain(object); return 0; } int audio_object_flush(struct audio_object *object) { if (object) return object->flush(object); return 0; } const char * audio_object_strerror(struct audio_object *object, int error) { if (object) return object->strerror(object, error); return NULL; } struct audio_object * create_audio_device_object(const char *device, const char *application_name, const char *description) { struct audio_object *object; #if defined(_WIN32) || defined(_WIN64) if ((object = create_xaudio2_object(device, application_name, description)) != NULL) return object; #else #if defined(__APPLE__) if ((object = create_coreaudio_object(device, application_name, description)) != NULL) return object; #else if ((object = create_pulseaudio_object(device, application_name, description)) != NULL) return object; if ((object = create_alsa_object(device, application_name, description)) != NULL) return object; if ((object = create_qsa_object(device, application_name, description)) != NULL) return object; if ((object = create_oss_object(device, application_name, description)) != NULL) return object; #endif #endif return NULL; } pcaudiolib-1.3/src/audio_priv.h000066400000000000000000000062701472657007400166120ustar00rootroot00000000000000/* Internal Audio API. * * Copyright (C) 2016 Reece H. Dunn * * This file is part of pcaudiolib. * * pcaudiolib is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. * * pcaudiolib 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 General Public License for more details. * * You should have received a copy of the GNU General Public License * along with pcaudiolib. If not, see . */ #ifndef PCAUDIOLIB_AUDIO_PRIV_H #define PCAUDIOLIB_AUDIO_PRIV_H #include #include #ifdef __cplusplus extern "C" { #endif struct audio_object { int (*open)(struct audio_object *object, enum audio_object_format format, uint32_t rate, uint8_t channels); void (*close)(struct audio_object *object); void (*destroy)(struct audio_object *object); int (*write)(struct audio_object *object, const void *data, size_t bytes); int (*drain)(struct audio_object *object); int (*flush)(struct audio_object *object); const char * (*strerror)(struct audio_object *object, int error); }; /* 60ms is the minimum and default buffer size used by eSpeak */ #define LATENCY 60 #if defined(_WIN32) || defined(_WIN64) #include #define container_of(ptr, type, member) ((type *)( (char *)ptr - offsetof(type,member) )) LPWSTR str2wcs(const char *str); HRESULT CreateWaveFormat(enum audio_object_format format, uint32_t rate, uint8_t channels, WAVEFORMATEX **wfmt); const char * windows_hresult_strerror(struct audio_object *object, int error); struct audio_object * create_xaudio2_object(const char *device, const char *application_name, const char *description); #else #define container_of(ptr, type, member) ({ \ const typeof( ((type *)0)->member ) *__mptr = (ptr); \ (type *)( (char *)__mptr - offsetof(type,member) );}) #ifdef __APPLE__ struct audio_object * create_coreaudio_object(const char *device, const char *application_name, const char *description); #else struct audio_object * create_pulseaudio_object(const char *device, const char *application_name, const char *description); struct audio_object * create_alsa_object(const char *device, const char *application_name, const char *description); struct audio_object * create_qsa_object(const char *device, const char *application_name, const char *description); struct audio_object * create_oss_object(const char *device, const char *application_name, const char *description); #endif #endif #ifdef __cplusplus } #endif #endif pcaudiolib-1.3/src/coreaudio.c000066400000000000000000000213021472657007400164070ustar00rootroot00000000000000/* Coreaudio Output. * * Copyright (C) 2016 Rui Batista * * This file is part of pcaudiolib. * * pcaudiolib is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. * * pcaudiolib 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 General Public License for more details. * * You should have received a copy of the GNU General Public License * along with pcaudiolib. If not, see . */ #include "config.h" #include "audio_priv.h" #ifdef HAVE_COREAUDIO_H #include #include #include "TPCircularBuffer/TPCircularBuffer+AudioBufferList.h" #include #include #include #include #define min(a,b) ((a) < (b) ? (a):(b)) #define COREAUDIO_BUFFER_SIZE (1024 * 64) // 4m struct coreaudio_object { struct audio_object vtable; AudioStreamBasicDescription format; AudioUnit outputUnit; TPCircularBuffer circularBuffer; bool initialized; bool running; char *device; char *application_name; char *description; }; #define to_coreaudio_object(object) container_of(object, struct coreaudio_object, vtable) static OSStatus graphRenderProc(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList * ioData) { struct coreaudio_object *self = (struct coreaudio_object *)inRefCon; if(! self->running){ // fill the buffer with sielence, stop was not acnoledged yet memset(ioData->mBuffers[0].mData,0,(self->format).mBytesPerFrame * inNumberFrames); ioData->mBuffers[0].mNumberChannels = 1; ioData->mBuffers[0].mDataByteSize = (self->format).mBytesPerFrame * inNumberFrames; return noErr; } UInt32 available = TPCircularBufferPeek(&(self->circularBuffer), NULL, &(self->format)); if(available == 0) { self->running = FALSE; AudioOutputUnitStop(self->outputUnit); } // copy as much data as we can from the circular buffer TPCircularBufferDequeueBufferListFrames(&(self->circularBuffer), &inNumberFrames, ioData, NULL, &(self->format)); return noErr; } int coreaudio_object_open(struct audio_object *object, enum audio_object_format format, uint32_t rate, uint8_t channels) { struct coreaudio_object *self = to_coreaudio_object(object); OSStatus err = noErr; if (self->initialized) return noErr; memset(&(self->format), 0, sizeof(AudioStreamBasicDescription)); (self->format).mSampleRate = rate; (self->format).mChannelsPerFrame = channels; (self->format).mFramesPerPacket = 1; switch (format) { case AUDIO_OBJECT_FORMAT_S16LE: (self->format).mFormatID = kAudioFormatLinearPCM; (self->format).mBitsPerChannel = 16; (self->format).mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked; break; default: return -1; } // guess the remaining of the AudioBasicStreamDescription structure UInt32 procSize = sizeof(AudioStreamBasicDescription); err = AudioFormatGetProperty(kAudioFormatProperty_FormatInfo, 0, NULL, &procSize, &(self->format)); if(err != noErr) { goto cleanup; } // Find default output AudioComponentDescription outputcd = {0,}; outputcd.componentType = kAudioUnitType_Output; outputcd.componentSubType = kAudioUnitSubType_DefaultOutput; outputcd.componentManufacturer = kAudioUnitManufacturer_Apple; // Create a component AudioComponent outputComponent = AudioComponentFindNext(NULL, &outputcd); err = AudioComponentInstanceNew(outputComponent, &(self->outputUnit)); if(err != noErr) { goto cleanup; } // set render callback AURenderCallbackStruct callbackStruct; callbackStruct.inputProcRefCon = self; callbackStruct.inputProc = graphRenderProc; size_t propSize = sizeof(AURenderCallbackStruct); err = AudioUnitSetProperty(self->outputUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Output, 0, &callbackStruct, propSize); if(err != noErr) { goto cleanup; } // set output unit input format propSize = sizeof(AudioStreamBasicDescription); err = AudioUnitSetProperty(self->outputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &(self->format), propSize); if(err != noErr) { goto cleanup; } err = AudioUnitInitialize(self->outputUnit); if(err != noErr) { goto cleanup; } // create a circular buffer to produce and consume audio TPCircularBufferInit(&(self->circularBuffer), COREAUDIO_BUFFER_SIZE); self->initialized = TRUE; return noErr; cleanup: if(self->outputUnit) { AudioUnitUninitialize(self->outputUnit); AudioComponentInstanceDispose(self->outputUnit); } return err; } void coreaudio_object_close(struct audio_object *object) { struct coreaudio_object *self = to_coreaudio_object(object); if(self->initialized) { AudioUnitUninitialize(self->outputUnit); AudioComponentInstanceDispose(self->outputUnit); TPCircularBufferCleanup(&(self->circularBuffer)); self->initialized = FALSE; } } void coreaudio_object_destroy(struct audio_object *object) { struct coreaudio_object *self = to_coreaudio_object(object); } int coreaudio_object_drain(struct audio_object *object) { struct coreaudio_object *self = to_coreaudio_object(object); if (!self->initialized) return noErr; while(self->running) usleep(10000); return noErr; } int coreaudio_object_flush(struct audio_object *object) { struct coreaudio_object *self = to_coreaudio_object(object); if (!self->initialized) return noErr; OSStatus err = AudioOutputUnitStop(self->outputUnit); TPCircularBufferClear(&(self->circularBuffer)); self->running = FALSE; return err; } int coreaudio_object_write(struct audio_object *object, const void *data, size_t bytes) { struct coreaudio_object *self = to_coreaudio_object(object); if (!self->initialized) return noErr; bool runningAtStart = self->running; if(!runningAtStart) { TPCircularBufferClear(&(self->circularBuffer)); } char *dataPtr = (char*)data; // copy data to our circular buffer // Poll while we don't have space in the buffer. AudioBufferList *bufferList; UInt32 available; while(bytes > 0) { while((!runningAtStart || self->running) && (available = TPCircularBufferGetAvailableSpace(&(self->circularBuffer), &(self->format))) == 0) usleep(10000); // check we didn't stop if(runningAtStart && (!self->running)) return noErr; bufferList = TPCircularBufferPrepareEmptyAudioBufferListWithAudioFormat(&(self->circularBuffer), &(self->format), available, NULL); // we are writing mono/interleaved data so we have one buffer UInt32 bytesToWrite = min(bytes, bufferList->mBuffers[0].mDataByteSize); bufferList->mBuffers[0].mNumberChannels = (self->format).mChannelsPerFrame; bufferList->mBuffers[0].mDataByteSize = bytesToWrite; memcpy(bufferList->mBuffers[0].mData, dataPtr, bytesToWrite); TPCircularBufferProduceAudioBufferList(&(self->circularBuffer), NULL); dataPtr += bytesToWrite; bytes -= bytesToWrite; if(! runningAtStart) { OSStatus err = AudioOutputUnitStart(self->outputUnit); if(err == noErr) { self->running = TRUE; runningAtStart = TRUE; } else { return err; } } } return noErr; } const char * coreaudio_object_strerror(struct audio_object *object, int error) { static char res[40]; sprintf(res, "Runtime error %x", error); return res; } static bool coreaudio_is_available(const char *device, const char *application_name, const char *description) { return true; } struct audio_object * create_coreaudio_object(const char *device, const char *application_name, const char *description) { if (!coreaudio_is_available(device, application_name, description)) return NULL; struct coreaudio_object *self = malloc(sizeof(struct coreaudio_object)); if (!self) return NULL; self->vtable.open = coreaudio_object_open; self->vtable.close = coreaudio_object_close; self->vtable.destroy = coreaudio_object_destroy; self->vtable.write = coreaudio_object_write; self->vtable.drain = coreaudio_object_drain; self->vtable.flush = coreaudio_object_flush; self->vtable.strerror = coreaudio_object_strerror; return &self->vtable; } #else struct audio_object * create_coreaudio_object(const char *device, const char *application_name, const char *description) { return NULL; } #endif pcaudiolib-1.3/src/include/000077500000000000000000000000001472657007400157165ustar00rootroot00000000000000pcaudiolib-1.3/src/include/pcaudiolib/000077500000000000000000000000001472657007400200315ustar00rootroot00000000000000pcaudiolib-1.3/src/include/pcaudiolib/audio.h000066400000000000000000000054471472657007400213150ustar00rootroot00000000000000/* Audio API. * * Copyright (C) 2016 Reece H. Dunn * * This file is part of pcaudiolib. * * pcaudiolib is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. * * pcaudiolib 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 General Public License for more details. * * You should have received a copy of the GNU General Public License * along with pcaudiolib. If not, see . */ #ifndef PCAUDIOLIB_AUDIO_H #define PCAUDIOLIB_AUDIO_H #include #include #ifdef __cplusplus extern "C" { #endif enum audio_object_format { AUDIO_OBJECT_FORMAT_S8, AUDIO_OBJECT_FORMAT_U8, AUDIO_OBJECT_FORMAT_S16LE, AUDIO_OBJECT_FORMAT_S16BE, AUDIO_OBJECT_FORMAT_U16LE, AUDIO_OBJECT_FORMAT_U16BE, AUDIO_OBJECT_FORMAT_S18LE, AUDIO_OBJECT_FORMAT_S18BE, AUDIO_OBJECT_FORMAT_U18LE, AUDIO_OBJECT_FORMAT_U18BE, AUDIO_OBJECT_FORMAT_S20LE, AUDIO_OBJECT_FORMAT_S20BE, AUDIO_OBJECT_FORMAT_U20LE, AUDIO_OBJECT_FORMAT_U20BE, AUDIO_OBJECT_FORMAT_S24LE, AUDIO_OBJECT_FORMAT_S24BE, AUDIO_OBJECT_FORMAT_U24LE, AUDIO_OBJECT_FORMAT_U24BE, AUDIO_OBJECT_FORMAT_S24_32LE, AUDIO_OBJECT_FORMAT_S24_32BE, AUDIO_OBJECT_FORMAT_U24_32LE, AUDIO_OBJECT_FORMAT_U24_32BE, AUDIO_OBJECT_FORMAT_S32LE, AUDIO_OBJECT_FORMAT_S32BE, AUDIO_OBJECT_FORMAT_U32LE, AUDIO_OBJECT_FORMAT_U32BE, AUDIO_OBJECT_FORMAT_FLOAT32LE, AUDIO_OBJECT_FORMAT_FLOAT32BE, AUDIO_OBJECT_FORMAT_FLOAT64LE, AUDIO_OBJECT_FORMAT_FLOAT64BE, AUDIO_OBJECT_FORMAT_IEC958LE, AUDIO_OBJECT_FORMAT_IEC958BE, AUDIO_OBJECT_FORMAT_ALAW, AUDIO_OBJECT_FORMAT_ULAW, AUDIO_OBJECT_FORMAT_ADPCM, AUDIO_OBJECT_FORMAT_MPEG, AUDIO_OBJECT_FORMAT_GSM, AUDIO_OBJECT_FORMAT_AC3, }; struct audio_object; int audio_object_open(struct audio_object *object, enum audio_object_format format, uint32_t rate, uint8_t channels); void audio_object_close(struct audio_object *object); void audio_object_destroy(struct audio_object *object); int audio_object_write(struct audio_object *object, const void *data, size_t bytes); int audio_object_drain(struct audio_object *object); int audio_object_flush(struct audio_object *object); const char * audio_object_strerror(struct audio_object *object, int error); struct audio_object * create_audio_device_object(const char *device, const char *application_name, const char *description); #ifdef __cplusplus } #endif #endif pcaudiolib-1.3/src/oss.c000066400000000000000000000107571472657007400152550ustar00rootroot00000000000000/* OSS Output. * * Copyright (C) 2016 Reece H. Dunn * * This file is part of pcaudiolib. * * pcaudiolib is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. * * pcaudiolib 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 General Public License for more details. * * You should have received a copy of the GNU General Public License * along with pcaudiolib. If not, see . */ #include "config.h" #include "audio_priv.h" #if defined(HAVE_SYS_SOUNDCARD_H) #include #include #include #include #include #include #include #define DEFAULT_OSS_DEVICE "/dev/dsp" struct oss_object { struct audio_object vtable; int fd; char *device; }; #define to_oss_object(object) container_of(object, struct oss_object, vtable) int oss_object_open(struct audio_object *object, enum audio_object_format format, uint32_t rate, uint8_t channels) { struct oss_object *self = to_oss_object(object); if (self->fd != -1) return EEXIST; int oss_format; switch (format) { case AUDIO_OBJECT_FORMAT_ALAW: oss_format = AFMT_A_LAW; break; case AUDIO_OBJECT_FORMAT_ULAW: oss_format = AFMT_MU_LAW; break; case AUDIO_OBJECT_FORMAT_S8: oss_format = AFMT_S8; break; case AUDIO_OBJECT_FORMAT_U8: oss_format = AFMT_U8; break; case AUDIO_OBJECT_FORMAT_S16LE: oss_format = AFMT_S16_LE; break; case AUDIO_OBJECT_FORMAT_S16BE: oss_format = AFMT_S16_BE; break; case AUDIO_OBJECT_FORMAT_U16LE: oss_format = AFMT_U16_LE; break; case AUDIO_OBJECT_FORMAT_U16BE: oss_format = AFMT_U16_BE; break; case AUDIO_OBJECT_FORMAT_ADPCM: oss_format = AFMT_IMA_ADPCM; break; case AUDIO_OBJECT_FORMAT_MPEG: oss_format = AFMT_MPEG; break; #if defined AFMT_AC3 case AUDIO_OBJECT_FORMAT_AC3: oss_format = AFMT_AC3; break; #endif default: return EINVAL; } int data; if ((self->fd = open(self->device ? self->device : DEFAULT_OSS_DEVICE, O_RDWR, 0)) == -1) return errno; if (ioctl(self->fd, SNDCTL_DSP_SETFMT, &oss_format) == -1) goto error; data = rate; if (ioctl(self->fd, SNDCTL_DSP_SPEED, &data) == -1) goto error; data = channels; if (ioctl(self->fd, SNDCTL_DSP_CHANNELS, &data) == -1) goto error; return 0; error: data = errno; close(self->fd); self->fd = -1; return data; } void oss_object_close(struct audio_object *object) { struct oss_object *self = to_oss_object(object); if (self->fd != -1) { close(self->fd); self->fd = -1; } } void oss_object_destroy(struct audio_object *object) { struct oss_object *self = to_oss_object(object); free(self->device); free(self); } int oss_object_drain(struct audio_object *object) { struct oss_object *self = to_oss_object(object); if (ioctl(self->fd, SNDCTL_DSP_SYNC, NULL) == -1) return errno; return 0; } int oss_object_flush(struct audio_object *object) { struct oss_object *self = to_oss_object(object); if (ioctl(self->fd, SNDCTL_DSP_RESET, NULL) == -1) return errno; return 0; } int oss_object_write(struct audio_object *object, const void *data, size_t bytes) { struct oss_object *self = to_oss_object(object); if (write(self->fd, data, bytes) == -1) return errno; return 0; } const char * oss_object_strerror(struct audio_object *object, int error) { return strerror(error); } struct audio_object * create_oss_object(const char *device, const char *application_name, const char *description) { struct oss_object *self = malloc(sizeof(struct oss_object)); if (!self) return NULL; self->fd = -1; self->device = device ? strdup(device) : NULL; self->vtable.open = oss_object_open; self->vtable.close = oss_object_close; self->vtable.destroy = oss_object_destroy; self->vtable.write = oss_object_write; self->vtable.drain = oss_object_drain; self->vtable.flush = oss_object_flush; self->vtable.strerror = oss_object_strerror; return &self->vtable; } #else struct audio_object * create_oss_object(const char *device, const char *application_name, const char *description) { return NULL; } #endif pcaudiolib-1.3/src/pulseaudio.c000066400000000000000000000144421472657007400166160ustar00rootroot00000000000000/* PulseAudio Output. * * Copyright (C) 2016 Reece H. Dunn * * This file is part of pcaudiolib. * * pcaudiolib is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. * * pcaudiolib 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 General Public License for more details. * * You should have received a copy of the GNU General Public License * along with pcaudiolib. If not, see . */ #include "config.h" #include "audio_priv.h" #ifdef HAVE_PULSE_SIMPLE_H #include #include #include #include struct pulseaudio_object { struct audio_object vtable; pa_sample_spec ss; pa_simple *s; char *device; char *application_name; char *description; }; #define to_pulseaudio_object(object) container_of(object, struct pulseaudio_object, vtable) int pulseaudio_object_open(struct audio_object *object, enum audio_object_format format, uint32_t rate, uint8_t channels) { struct pulseaudio_object *self = to_pulseaudio_object(object); if (self->s) return PA_ERR_EXIST; self->ss.rate = rate; self->ss.channels = channels; switch (format) { case AUDIO_OBJECT_FORMAT_ALAW: self->ss.format = PA_SAMPLE_ALAW; break; case AUDIO_OBJECT_FORMAT_ULAW: self->ss.format = PA_SAMPLE_ULAW; break; case AUDIO_OBJECT_FORMAT_U8: self->ss.format = PA_SAMPLE_U8; break; case AUDIO_OBJECT_FORMAT_S16LE: self->ss.format = PA_SAMPLE_S16LE; break; case AUDIO_OBJECT_FORMAT_S16BE: self->ss.format = PA_SAMPLE_S16BE; break; #ifdef PA_SAMPLE_S24LE case AUDIO_OBJECT_FORMAT_S24LE: self->ss.format = PA_SAMPLE_S24LE; break; case AUDIO_OBJECT_FORMAT_S24BE: self->ss.format = PA_SAMPLE_S24BE; break; case AUDIO_OBJECT_FORMAT_S24_32LE: self->ss.format = PA_SAMPLE_S24_32LE; break; case AUDIO_OBJECT_FORMAT_S24_32BE: self->ss.format = PA_SAMPLE_S24_32BE; break; #endif case AUDIO_OBJECT_FORMAT_S32LE: self->ss.format = PA_SAMPLE_S32LE; break; case AUDIO_OBJECT_FORMAT_S32BE: self->ss.format = PA_SAMPLE_S32BE; break; case AUDIO_OBJECT_FORMAT_FLOAT32LE: self->ss.format = PA_SAMPLE_FLOAT32LE; break; case AUDIO_OBJECT_FORMAT_FLOAT32BE: self->ss.format = PA_SAMPLE_FLOAT32BE; break; default: return PA_ERR_INVALID; // Invalid argument. } int error = 0; pa_buffer_attr battr; battr.fragsize = (uint32_t) -1; battr.maxlength = (uint32_t) -1; battr.minreq = (uint32_t) -1; battr.prebuf = (uint32_t) -1; battr.tlength = pa_usec_to_bytes(LATENCY * 1000, &self->ss); self->s = pa_simple_new(NULL, self->application_name, PA_STREAM_PLAYBACK, self->device, self->description, &self->ss, NULL, &battr, &error); return error; } void pulseaudio_object_close(struct audio_object *object) { struct pulseaudio_object *self = to_pulseaudio_object(object); if (self->s) { pa_simple_free(self->s); self->s = NULL; } } void pulseaudio_object_destroy(struct audio_object *object) { struct pulseaudio_object *self = to_pulseaudio_object(object); free(self->device); free(self->application_name); free(self->description); free(self); } int pulseaudio_object_drain(struct audio_object *object) { struct pulseaudio_object *self = to_pulseaudio_object(object); if (!self->s) return 0; int error = 0; pa_simple_drain(self->s, &error); return error; } int pulseaudio_object_flush(struct audio_object *object) { struct pulseaudio_object *self = to_pulseaudio_object(object); if (!self->s) return 0; int error = 0; pa_simple_flush(self->s, &error); return error; } int pulseaudio_object_write(struct audio_object *object, const void *data, size_t bytes) { struct pulseaudio_object *self = to_pulseaudio_object(object); if (!self->s) return 0; int error = 0; pa_simple_write(self->s, data, bytes, &error); return error; } const char * pulseaudio_object_strerror(struct audio_object *object, int error) { return pa_strerror(error); } static bool pulseaudio_is_available(const char *device, const char *application_name, const char *description) { pa_sample_spec ss; ss.format = PA_SAMPLE_S16LE; ss.rate = 44100; ss.channels = 1; pa_simple *s = pa_simple_new(NULL, application_name, PA_STREAM_PLAYBACK, device, description, &ss, NULL, NULL, NULL); if (!s) return false; pa_simple_free(s); return true; } struct audio_object * create_pulseaudio_object(const char *device, const char *application_name, const char *description) { if (!pulseaudio_is_available(device, application_name, description)) return NULL; struct pulseaudio_object *self = malloc(sizeof(struct pulseaudio_object)); if (!self) return NULL; self->s = NULL; self->device = device ? strdup(device) : NULL; self->application_name = application_name ? strdup(application_name) : NULL; self->description = description ? strdup(description) : NULL; self->vtable.open = pulseaudio_object_open; self->vtable.close = pulseaudio_object_close; self->vtable.destroy = pulseaudio_object_destroy; self->vtable.write = pulseaudio_object_write; self->vtable.drain = pulseaudio_object_drain; self->vtable.flush = pulseaudio_object_flush; self->vtable.strerror = pulseaudio_object_strerror; return &self->vtable; } #else struct audio_object * create_pulseaudio_object(const char *device, const char *application_name, const char *description) { return NULL; } #endif pcaudiolib-1.3/src/qsa.c000066400000000000000000000127421472657007400152310ustar00rootroot00000000000000/* QSA Output. * * Copyright (C) 2016 Reece H. Dunn * Copyright (C) 2016 Kaj-Michael Lang * * This file is part of pcaudiolib. * * pcaudiolib is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. * * pcaudiolib 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 General Public License for more details. * * You should have received a copy of the GNU General Public License * along with pcaudiolib. If not, see . */ #include "config.h" #include "audio_priv.h" #if defined(HAVE_SYS_ASOUNDLIB_H) && defined(HAVE_SYS_ASOUND_H) #include #include #include #include struct qsa_object { struct audio_object vtable; snd_pcm_t *handle; uint8_t sample_size; char *device; }; #define to_qsa_object(object) container_of(object, struct qsa_object, vtable) int qsa_object_open(struct audio_object *object, enum audio_object_format format, uint32_t rate, uint8_t channels) { struct qsa_object *self = to_qsa_object(object); if (self->handle) return -EEXIST; int pcm_format; #define FORMAT(srcfmt, dstfmt, size) case srcfmt: pcm_format = dstfmt; self->sample_size = size; break; switch (format) { FORMAT(AUDIO_OBJECT_FORMAT_U8, SND_PCM_SFMT_U8, 1) FORMAT(AUDIO_OBJECT_FORMAT_S8, SND_PCM_SFMT_S8, 1) FORMAT(AUDIO_OBJECT_FORMAT_S16LE, SND_PCM_SFMT_S16_LE, 2) default: return -EINVAL; } #undef FORMAT snd_pcm_info_t pi; snd_pcm_channel_info_t pci; snd_pcm_channel_params_t pp; snd_pcm_channel_setup_t setup; int err = 0; if (self->device) { if ((err = snd_pcm_open_name(&self->handle, self->device, SND_PCM_OPEN_PLAYBACK)) < 0) goto error; } else { if ((err = snd_pcm_open_preferred(&self->handle, NULL, NULL, SND_PCM_OPEN_PLAYBACK)) < 0) goto error; } memset (&pi, 0, sizeof (pi)); if ((err = snd_pcm_info (self->handle, &pi)) < 0) goto error; memset (&pci, 0, sizeof (pci)); pci.channel = SND_PCM_CHANNEL_PLAYBACK; if ((err = snd_pcm_plugin_info (self->handle, &pci)) < 0) goto error; memset (&pp, 0, sizeof (pp)); pp.mode = SND_PCM_MODE_BLOCK; pp.channel = SND_PCM_CHANNEL_PLAYBACK; pp.start_mode = SND_PCM_START_FULL; pp.stop_mode = SND_PCM_STOP_STOP; pp.buf.block.frag_size = pci.max_fragment_size; pp.buf.block.frags_max = 4; // XXX: What should this be? pp.buf.block.frags_min = 1; pp.format.interleave = 1; pp.format.rate = rate; pp.format.voices = channels; pp.format.format = pcm_format; if ((err = snd_pcm_plugin_params (self->handle, &pp)) < 0) goto error; if ((err = snd_pcm_plugin_prepare (self->handle, SND_PCM_CHANNEL_PLAYBACK)) < 0) goto error; return 0; error: if (self->handle) { snd_pcm_close(self->handle); self->handle = NULL; } return err; } void qsa_object_close(struct audio_object *object) { struct qsa_object *self = to_qsa_object(object); if (self->handle) { snd_pcm_close(self->handle); self->handle = NULL; } } void qsa_object_destroy(struct audio_object *object) { struct qsa_object *self = to_qsa_object(object); free(self->device); free(self); } int qsa_object_drain(struct audio_object *object) { struct qsa_object *self = to_qsa_object(object); return snd_pcm_plugin_playback_drain(self->handle); } int qsa_object_flush(struct audio_object *object) { struct qsa_object *self = to_qsa_object(object); return snd_pcm_plugin_flush(self->handle, SND_PCM_CHANNEL_PLAYBACK); } int qsa_object_write(struct audio_object *object, const void *data, size_t bytes) { struct qsa_object *self = to_qsa_object(object); snd_pcm_channel_status_t status; size_t written = snd_pcm_plugin_write(self->handle, data, bytes); if (written < bytes) { int err; memset (&status, 0, sizeof (status)); status.channel = SND_PCM_CHANNEL_PLAYBACK; err=snd_pcm_plugin_status (self->handle, &status); if (err < 0) { return err; } if (status.status == SND_PCM_STATUS_READY || status.status == SND_PCM_STATUS_UNDERRUN) { err = snd_pcm_plugin_prepare(self->handle, SND_PCM_CHANNEL_PLAYBACK); // Try again if we can if (err==0 && written==0) return snd_pcm_plugin_write(self->handle, data, bytes); } return err; } return 0; } const char * qsa_object_strerror(struct audio_object *object, int error) { return snd_strerror(error); } struct audio_object * create_qsa_object(const char *device, const char *application_name, const char *description) { struct qsa_object *self = malloc(sizeof(struct qsa_object)); if (!self) return NULL; self->handle = NULL; self->sample_size = 0; self->device = device ? strdup(device) : NULL; self->vtable.open = qsa_object_open; self->vtable.close = qsa_object_close; self->vtable.destroy = qsa_object_destroy; self->vtable.write = qsa_object_write; self->vtable.drain = qsa_object_drain; self->vtable.flush = qsa_object_flush; self->vtable.strerror = qsa_object_strerror; return &self->vtable; } #else struct audio_object * create_qsa_object(const char *device, const char *application_name, const char *description) { return NULL; } #endif pcaudiolib-1.3/src/windows.c000066400000000000000000000061621472657007400161360ustar00rootroot00000000000000/* Windows Helpers for Writing Windows Audio Objects. * * Copyright (C) 2016 Reece H. Dunn * * This file is part of pcaudiolib. * * pcaudiolib is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. * * pcaudiolib 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 General Public License for more details. * * You should have received a copy of the GNU General Public License * along with pcaudiolib. If not, see . */ #include "config.h" #include "audio_priv.h" #include LPWSTR str2wcs(const char *str) { size_t len = strlen(str); LPWSTR ret = calloc(len + 1, sizeof(wchar_t)); if (!ret) return NULL; mbstowcs(ret, str, len + 1); return ret; } static HRESULT CreateWaveFormatEx(WORD format, uint32_t bits_per_sample, uint32_t rate, uint8_t channels, WAVEFORMATEX **wfmt) { (*wfmt) = CoTaskMemAlloc(sizeof(WAVEFORMATEX)); if (!(*wfmt)) return E_OUTOFMEMORY; (*wfmt)->wFormatTag = format; (*wfmt)->nChannels = channels; (*wfmt)->nBlockAlign = (channels * bits_per_sample) / 8; (*wfmt)->nSamplesPerSec = rate; (*wfmt)->wBitsPerSample = bits_per_sample; (*wfmt)->nAvgBytesPerSec = rate * (*wfmt)->nBlockAlign; (*wfmt)->cbSize = 0; return S_OK; } HRESULT CreateWaveFormat(enum audio_object_format format, uint32_t rate, uint8_t channels, WAVEFORMATEX **wfmt) { if (!wfmt) return E_INVALIDARG; switch (format) { case AUDIO_OBJECT_FORMAT_ALAW: return CreateWaveFormatEx(WAVE_FORMAT_ALAW, 8, rate, channels, wfmt); case AUDIO_OBJECT_FORMAT_ULAW: return CreateWaveFormatEx(WAVE_FORMAT_MULAW, 8, rate, channels, wfmt); case AUDIO_OBJECT_FORMAT_S8: return CreateWaveFormatEx(WAVE_FORMAT_PCM, 8, rate, channels, wfmt); case AUDIO_OBJECT_FORMAT_S16LE: return CreateWaveFormatEx(WAVE_FORMAT_PCM, 16, rate, channels, wfmt); case AUDIO_OBJECT_FORMAT_S32LE: return CreateWaveFormatEx(WAVE_FORMAT_PCM, 32, rate, channels, wfmt); case AUDIO_OBJECT_FORMAT_FLOAT32LE: return CreateWaveFormatEx(WAVE_FORMAT_IEEE_FLOAT, 32, rate, channels, wfmt); case AUDIO_OBJECT_FORMAT_FLOAT64LE: return CreateWaveFormatEx(WAVE_FORMAT_IEEE_FLOAT, 64, rate, channels, wfmt); default: return E_INVALIDARG; } } const char * windows_hresult_strerror(struct audio_object *object, int error) { char *msg = NULL; DWORD result = FormatMessageA(FORMAT_MESSAGE_ALLOCATE_BUFFER | FORMAT_MESSAGE_FROM_SYSTEM | FORMAT_MESSAGE_IGNORE_INSERTS, NULL, error, MAKELANGID(LANG_NEUTRAL, SUBLANG_DEFAULT), &msg, 256, NULL); if (!result) return NULL; size_t len = strlen(msg); while (len > 1 && msg[len - 1] == '\n' || msg[len - 1] == '\r') msg[--len] = 0; char *ret = strdup(msg); LocalFree(msg); return ret; } pcaudiolib-1.3/src/xaudio2.cpp000066400000000000000000000124661472657007400163630ustar00rootroot00000000000000/* Windows xAudio2 API Output. * * Copyright (C) 2016 Reece H. Dunn * * This file is part of pcaudiolib. * * pcaudiolib is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. * * pcaudiolib 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 General Public License for more details. * * You should have received a copy of the GNU General Public License * along with pcaudiolib. If not, see . */ #include "config.h" #include "audio_priv.h" // NOTE: XAudio2.h fails to build with a C compiler #include #pragma comment(lib, "xaudio2.lib") struct xaudio2_object { struct audio_object vtable; IXAudio2 *audio; IXAudio2MasteringVoice *mastering; IXAudio2SourceVoice *source; WAVEFORMATEX *format; LPWSTR devicename; }; class VoiceCallbacks : public IXAudio2VoiceCallback { public: void OnBufferEnd(void* pBufferContext) { if (pBufferContext != NULL) { free((void*)pBufferContext); } } // Stubs for all interface callbacks void OnStreamEnd() { } void OnVoiceProcessingPassEnd() { } void OnVoiceProcessingPassStart(UINT32 SamplesRequired) { } void OnBufferStart(void* pBufferContext) { } void OnLoopEnd(void* pBufferContext) { } void OnVoiceError(void* pBufferContext, HRESULT Error) { } } voiceCallbacks; void xaudio2_object_close(struct audio_object *object); #define to_xaudio2_object(object) container_of(object, struct xaudio2_object, vtable) int xaudio2_object_open(struct audio_object *object, enum audio_object_format format, uint32_t rate, uint8_t channels) { struct xaudio2_object *self = to_xaudio2_object(object); if (self->mastering != NULL) return HRESULT_FROM_WIN32(ERROR_ALREADY_INITIALIZED); HRESULT hr; hr = self->audio->CreateMasteringVoice(&self->mastering); if (FAILED(hr)) goto error; hr = CreateWaveFormat(format, rate, channels, &self->format); if (FAILED(hr)) goto error; hr = self->audio->CreateSourceVoice(&self->source, self->format, 0, 2.0f, &voiceCallbacks); if (FAILED(hr)) goto error; return S_OK; error: xaudio2_object_close(object); return hr; } void xaudio2_object_close(struct audio_object *object) { struct xaudio2_object *self = to_xaudio2_object(object); if (self->source != NULL) { self->source->DestroyVoice(); self->source = NULL; } if (self->format != NULL) { CoTaskMemFree(self->format); self->format = NULL; } if (self->mastering != NULL) { self->mastering->DestroyVoice(); self->mastering = NULL; } } void xaudio2_object_destroy(struct audio_object *object) { struct xaudio2_object *self = to_xaudio2_object(object); self->audio->Release(); free(self->devicename); free(self); CoUninitialize(); } int xaudio2_object_drain(struct audio_object *object) { struct xaudio2_object *self = to_xaudio2_object(object); while (true) { Sleep(10); XAUDIO2_VOICE_STATE state = { 0 }; self->source->GetState(&state); if (state.pCurrentBufferContext == NULL && state.BuffersQueued == 0) break; } return S_OK; } int xaudio2_object_flush(struct audio_object *object) { struct xaudio2_object *self = to_xaudio2_object(object); return S_OK; } int xaudio2_object_write(struct audio_object *object, const void *data, size_t bytes) { struct xaudio2_object *self = to_xaudio2_object(object); BYTE* buf_data = (BYTE *)malloc(bytes); memcpy(buf_data, data, bytes); XAUDIO2_BUFFER buffer = { 0 }; buffer.AudioBytes = bytes; buffer.pAudioData = buf_data; buffer.pContext = buf_data; HRESULT hr = S_OK; if (SUCCEEDED(hr)) hr = self->source->SubmitSourceBuffer(&buffer); XAUDIO2_VOICE_STATE state = { 0 }; self->source->GetState(&state); UINT32 buffersQueued = state.BuffersQueued; while (FAILED(hr)) { Sleep(10); self->source->GetState(&state); if (state.BuffersQueued < buffersQueued) { hr = self->source->SubmitSourceBuffer(&buffer); self->source->GetState(&state); buffersQueued = state.BuffersQueued; } } if (SUCCEEDED(hr)) hr = self->source->Start(0); return hr; } struct audio_object * create_xaudio2_object(const char *device, const char *application_name, const char *description) { CoInitialize(NULL); IXAudio2 *audio = NULL; HRESULT hr = XAudio2Create(&audio, 0, XAUDIO2_DEFAULT_PROCESSOR); if (FAILED(hr) || audio == NULL) { if (audio != NULL) audio->Release(); CoUninitialize(); return NULL; } struct xaudio2_object *self = (struct xaudio2_object *)malloc(sizeof(struct xaudio2_object)); if (!self) return NULL; self->audio = audio; self->mastering = NULL; self->source = NULL; self->format = NULL; self->devicename = device ? str2wcs(device) : NULL; self->vtable.open = xaudio2_object_open; self->vtable.close = xaudio2_object_close; self->vtable.destroy = xaudio2_object_destroy; self->vtable.write = xaudio2_object_write; self->vtable.drain = xaudio2_object_drain; self->vtable.flush = xaudio2_object_flush; self->vtable.strerror = windows_hresult_strerror; return &self->vtable; }