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StAudioQueue.cpp
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StAudioQueue.cpp
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/**
* Copyright © 2009-2017 Kirill Gavrilov <kirill@sview.ru>
*
* StMoviePlayer program 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.
*
* StMoviePlayer program 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 this program.
* If not, see <http://www.gnu.org/licenses/>.
*/
#include "StAudioQueue.h"
#include <StGL/StGLVec.h>
#include <StThreads/StThread.h>
namespace {
/**
* Check OpenAL state.
*/
bool stalCheckErrors(const StString& ST_DEBUG_VAR(theProcedure)) {
ALenum anError = alGetError();
switch(anError) {
case AL_NO_ERROR: return true; // alright
case AL_INVALID_NAME: ST_DEBUG_LOG(theProcedure + ": AL_INVALID_NAME"); return false;
case AL_INVALID_ENUM: ST_DEBUG_LOG(theProcedure + ": AL_INVALID_ENUM"); return false;
case AL_INVALID_VALUE: ST_DEBUG_LOG(theProcedure + ": AL_INVALID_VALUE"); return false;
case AL_INVALID_OPERATION: ST_DEBUG_LOG(theProcedure + ": AL_INVALID_OPERATION"); return false;
case AL_OUT_OF_MEMORY: ST_DEBUG_LOG(theProcedure + ": AL_OUT_OF_MEMORY"); return false;
default: ST_DEBUG_LOG(theProcedure + ": OpenAL unknown error"); return false;
}
}
static const StGLVec3 THE_POSITION_LEFT (-1.0f, 0.0f, 0.0f);
static const StGLVec3 THE_POSITION_RIGHT ( 1.0f, 0.0f, 0.0f);
static const StGLVec3 THE_POSITION_CENTER ( 0.0f, 0.0f, 0.0f);
static const StGLVec3 THE_POSITION_FRONT_LEFT (-1.0f, 0.0f, -1.0f);
static const StGLVec3 THE_POSITION_FRONT_CENTER ( 0.0f, 0.0f, -1.0f);
static const StGLVec3 THE_POSITION_FRONT_RIGHT ( 1.0f, 0.0f, -1.0f);
static const StGLVec3 THE_POSITION_LFE ( 0.0f, 0.0f, 0.0f);
static const StGLVec3 THE_POSITION_REAR_LEFT (-1.0f, 0.0f, 1.0f);
static const StGLVec3 THE_POSITION_REAR_RIGHT ( 1.0f, 0.0f, 1.0f);
static const StGLVec3 THE_POSITION_REAR_LEFT71 (-1.0f, 0.0f, 1.0f);
static const StGLVec3 THE_POSITION_REAR_RIGHT71 ( 1.0f, 0.0f, 1.0f);
static const StGLVec3 THE_POSITION_SIDE_LEFT71 (-1.0f, 0.0f, 0.0f);
static const StGLVec3 THE_POSITION_SIDE_RIGHT71 ( 1.0f, 0.0f, 0.0f);
static const StGLVec3 THE_LISTENER_FORWARD ( 0.0f, 0.0f, -1.0f);
static const StGLVec3 THE_LISTENER_UP ( 0.0f, 1.0f, 0.0f);
}
#if(LIBAVCODEC_VERSION_INT < AV_VERSION_INT(53, 0, 0))
/**
* Check if dynamically linked version of FFmpeg libraries
* is old or not.
*/
inline bool isReoderingNeededInit() {
stAV::Version aVersion = stAV::Version::libavcodec();
// I have no idea which version introduce this feature for ac3 and ogg streams...
// but experimentally detect that FFmpeg 0.5.1 is old and FFmpeg 0.6 include this
// We check libavcodec version here and hopes this will true for most packages
bool isUpToDate = ( aVersion.myMajor > 52) ||
((aVersion.myMajor == 52) && (aVersion.myMinor >= 72));
if(!isUpToDate) {
ST_DEBUG_LOG("Used old FFmpeg, enabled sView channel reorder for multichannel AC3 and OGG Vorbis streams!");
}
return !isUpToDate;
}
inline bool isReoderingNeeded() {
static bool isNeeded = isReoderingNeededInit();
return isNeeded;
}
#endif
void StAudioQueue::stalConfigureSources1() {
alSourcefv(myAlSources[0], AL_POSITION, THE_POSITION_CENTER);
stalCheckErrors("alSource*1.0");
}
void StAudioQueue::stalConfigureSources2_0() {
alSourcefv(myAlSources[0], AL_POSITION, THE_POSITION_LEFT);
alSourcefv(myAlSources[1], AL_POSITION, THE_POSITION_RIGHT);
stalCheckErrors("alSource*2.0");
}
void StAudioQueue::stalConfigureSources3_0() {
alSourcefv(myAlSources[0], AL_POSITION, THE_POSITION_LEFT);
alSourcefv(myAlSources[1], AL_POSITION, THE_POSITION_RIGHT);
alSourcefv(myAlSources[2], AL_POSITION, THE_POSITION_CENTER);
stalCheckErrors("alSource*3.0");
}
void StAudioQueue::stalConfigureSources4_0() {
alSourcefv(myAlSources[0], AL_POSITION, THE_POSITION_FRONT_LEFT);
alSourcefv(myAlSources[1], AL_POSITION, THE_POSITION_FRONT_RIGHT);
alSourcefv(myAlSources[2], AL_POSITION, THE_POSITION_REAR_LEFT);
alSourcefv(myAlSources[3], AL_POSITION, THE_POSITION_REAR_RIGHT);
stalCheckErrors("alSource*4.0");
}
void StAudioQueue::stalConfigureSources5_0() {
alSourcefv(myAlSources[0], AL_POSITION, THE_POSITION_FRONT_LEFT);
alSourcefv(myAlSources[1], AL_POSITION, THE_POSITION_FRONT_RIGHT);
alSourcefv(myAlSources[2], AL_POSITION, THE_POSITION_FRONT_CENTER);
alSourcefv(myAlSources[3], AL_POSITION, THE_POSITION_REAR_LEFT);
alSourcefv(myAlSources[4], AL_POSITION, THE_POSITION_REAR_RIGHT);
stalCheckErrors("alSource*5.0");
}
void StAudioQueue::stalConfigureSources5_1() {
alSourcefv(myAlSources[0], AL_POSITION, THE_POSITION_FRONT_LEFT);
alSourcefv(myAlSources[1], AL_POSITION, THE_POSITION_FRONT_RIGHT);
alSourcefv(myAlSources[2], AL_POSITION, THE_POSITION_FRONT_CENTER);
alSourcefv(myAlSources[3], AL_POSITION, THE_POSITION_LFE);
alSourcefv(myAlSources[4], AL_POSITION, THE_POSITION_REAR_LEFT);
alSourcefv(myAlSources[5], AL_POSITION, THE_POSITION_REAR_RIGHT);
stalCheckErrors("alSource*5.1");
}
void StAudioQueue::stalConfigureSources7_1() {
alSourcefv(myAlSources[0], AL_POSITION, THE_POSITION_FRONT_LEFT);
alSourcefv(myAlSources[1], AL_POSITION, THE_POSITION_FRONT_RIGHT);
alSourcefv(myAlSources[2], AL_POSITION, THE_POSITION_FRONT_CENTER);
alSourcefv(myAlSources[3], AL_POSITION, THE_POSITION_LFE);
alSourcefv(myAlSources[4], AL_POSITION, THE_POSITION_REAR_LEFT71);
alSourcefv(myAlSources[5], AL_POSITION, THE_POSITION_REAR_RIGHT71);
alSourcefv(myAlSources[6], AL_POSITION, THE_POSITION_SIDE_LEFT71);
alSourcefv(myAlSources[7], AL_POSITION, THE_POSITION_SIDE_RIGHT71);
stalCheckErrors("alSource*7.1");
}
bool StAudioQueue::stalInit() {
std::string aDevName;
{
StMutexAuto aLock(mySwitchMutex);
aDevName = myAlDeviceName;
}
if(!myAlCtx.create(aDevName)) {
if(aDevName.empty()
|| !myAlCtx.create("")) {
// retry with default device
return false;
}
}
myAlCtx.makeCurrent();
stalResetHrtf();
{
StMutexAuto aLock(myAlInfoMutex);
myAlInfo.clear();
myAlCtx.fullInfo(myAlInfo);
}
alGetError(); // clear error code
// generate the buffers
for(size_t aSrcId = 0; aSrcId < THE_NUM_AL_SOURCES; ++aSrcId) {
alGenBuffers(THE_NUM_AL_BUFFERS, &myAlBuffers[aSrcId][0]);
stalCheckErrors(StString("alGenBuffers") + aSrcId);
}
// generate the sources
alGenSources(THE_NUM_AL_SOURCES, myAlSources);
stalCheckErrors("alGenSources");
// configure sources
const StGLVec3 aZeroVec(0.0f);
for(size_t aSrcId = 0; aSrcId < THE_NUM_AL_SOURCES; ++aSrcId) {
alSourcefv(myAlSources[aSrcId], AL_POSITION, aZeroVec);
alSourcefv(myAlSources[aSrcId], AL_VELOCITY, aZeroVec);
alSourcefv(myAlSources[aSrcId], AL_DIRECTION, aZeroVec);
alSourcef (myAlSources[aSrcId], AL_ROLLOFF_FACTOR, 0.0f);
//alSourcei(myAlSources[aSrcId], AL_SOURCE_RELATIVE, myToOrientListener ? AL_FALSE : AL_TRUE);
alSourcei (myAlSources[aSrcId], AL_SOURCE_RELATIVE, AL_FALSE);
alSourcef (myAlSources[aSrcId], AL_GAIN, 1.0f);
stalCheckErrors(StString("alSource*") + aSrcId);
}
// configure listener
alListenerfv(AL_POSITION, aZeroVec);
alListenerfv(AL_VELOCITY, aZeroVec);
alListenerf (AL_GAIN, myAlGain); // apply gain to all sources at-once
stalOrientListener();
return true;
}
void StAudioQueue::stalDeinit() {
// clear buffers
stalEmpty();
alSourceStopv(THE_NUM_AL_SOURCES, myAlSources);
alDeleteSources(THE_NUM_AL_SOURCES, myAlSources);
stalCheckErrors("alDeleteSources");
for(size_t aSrcId = 0; aSrcId < THE_NUM_AL_SOURCES; ++aSrcId) {
alDeleteBuffers(THE_NUM_AL_BUFFERS, &myAlBuffers[aSrcId][0]);
stalCheckErrors(StString("alDeleteBuffers") + aSrcId);
}
// close device
myAlCtx.destroy();
}
void StAudioQueue::stalReinitialize() {
stalDeinit(); // release OpenAL context
myIsAlValid = (stalInit() ? ST_AL_INIT_OK : ST_AL_INIT_KO);
myIsDisconnected = false;
myToSwitchDev = false;
}
void StAudioQueue::stalResetHrtf() {
const bool wasChanged = myAlHrtf != myAlHrtfPrev;
myAlHrtfPrev = myAlHrtf;
if(!myAlCtx.hasExtSoftHrtf) {
return;
}
ALCint aHrtfVal = ALC_DONT_CARE_SOFT;
if(myAlHrtf == StAlHrtfRequest_ForceOn) {
aHrtfVal = ALC_TRUE;
} else if(myAlHrtf == StAlHrtfRequest_ForceOff) {
aHrtfVal = ALC_FALSE;
}
ALCint anAttrs[] = {
ALC_HRTF_SOFT, aHrtfVal,
0
};
myAlCtx.alcResetDeviceSOFT(myAlCtx.getAlDevice(), anAttrs);
if(wasChanged) {
(void )wasChanged;
ST_DEBUG_LOG(myAlCtx.toStringExtensions())
}
}
void StAudioQueue::stalOrientListener() {
if(myToOrientListener) {
StGLQuaternion aHeadOrient;
{
StMutexAuto aLock(mySwitchMutex);
aHeadOrient = myHeadOrient;
aHeadOrient.reverse();
aHeadOrient.normalize();
}
const StGLVec3 aFwdVec = aHeadOrient.multiply(THE_LISTENER_FORWARD);
const StGLVec3 anUpVec = aHeadOrient.multiply(THE_LISTENER_UP);
const StGLVec3 aListenerOri[2] = { aFwdVec, anUpVec };
//static StGLVec3 aPrev; if(aFwdVec != aPrev) { aPrev = aFwdVec; ST_DEBUG_LOG("FWD: " + aFwdVec.toString() + "\n[UP: " + anUpVec.toString() + "]\n") }
alListenerfv(AL_ORIENTATION, (const ALfloat* )aListenerOri);
myAlIsListOrient = true;
} else if(myAlIsListOrient) {
const StGLVec3 aListenerOri[2] = { THE_LISTENER_FORWARD, THE_LISTENER_UP };
alListenerfv(AL_ORIENTATION, (const ALfloat* )aListenerOri);
myAlIsListOrient = false;
}
}
void StAudioQueue::stalEmpty() {
alSourceStopv(THE_NUM_AL_SOURCES, myAlSources);
ALint aBufQueued = 0;
ALuint alBuffIdToUnqueue = 0;
for(size_t aSrcId = 0; aSrcId < THE_NUM_AL_SOURCES; ++aSrcId) {
alGetSourcei(myAlSources[aSrcId], AL_BUFFERS_QUEUED, &aBufQueued);
for(ALint aBufIter = 0; aBufIter < aBufQueued; ++aBufIter) {
alSourceUnqueueBuffers(myAlSources[aSrcId], 1, &alBuffIdToUnqueue);
stalCheckErrors(StString("alSourceUnqueueBuffers") + aSrcId);
}
alSourcei(myAlSources[aSrcId], AL_BUFFER, 0);
}
///alSourceRewindv(THE_NUM_AL_SOURCES, myAlSources);
}
ALenum StAudioQueue::stalGetSourceState() {
ALenum aState;
alGetSourcei(myAlSources[0], AL_SOURCE_STATE, &aState);
if(myDbgPrevSrcState != aState) {
switch(aState) {
case AL_INITIAL: ST_DEBUG_LOG("OpenAL source state: INITIAL"); break;
case AL_PLAYING: ST_DEBUG_LOG("OpenAL source state: PLAYING"); break;
case AL_PAUSED: ST_DEBUG_LOG("OpenAL source state: PAUSED") ; break;
case AL_STOPPED: ST_DEBUG_LOG("OpenAL source state: STOPPED"); break;
default: ST_DEBUG_LOG("OpenAL source state: UNKNOWN"); break;
}
}
myDbgPrevSrcState = aState;
return aState;
}
/**
* Simple thread function which just call decodeLoop().
*/
static SV_THREAD_FUNCTION threadFunction(void* audioQueue) {
StAudioQueue* stAudioQueue = (StAudioQueue* )audioQueue;
stAudioQueue->decodeLoop();
return SV_THREAD_RETURN 0;
}
StAudioQueue::StAudioQueue(const std::string& theAlDeviceName,
StAudioQueue::StAlHrtfRequest theAlHrtf)
: StAVPacketQueue(512),
myPlaybackTimer(false),
myDowntimeEvent(true),
myAvSrcFormat(-1),
myAvSampleRate(-1),
myAvNbChannels(-1),
myBufferSrc(StPcmFormat_Int16),
myBufferOut(StPcmFormat_Int16),
myIsAlValid(ST_AL_INIT_NA),
myToSwitchDev(false),
myIsDisconnected(false),
myToOrientListener(false),
myToForceBFormat(false),
myAlDeviceName(theAlDeviceName),
myAlFormat(AL_FORMAT_STEREO16),
myPrevFormat(AL_FORMAT_STEREO16),
myPrevFrequency(0),
myAlGain(1.0f),
myAlGainPrev(1.0f),
myAlSoftLayout(true),
myAlIsListOrient(false),
myAlCanBFormat(false),
myAlIsBFormat(false),
myAlHrtf(theAlHrtf),
myAlHrtfPrev(theAlHrtf),
myDbgPrevQueued(-1),
myDbgPrevSrcState(-1) {
stMemSet(myAlSources, 0, sizeof(myAlSources));
// launch thread parse incoming packets from queue
myThread = new StThread(threadFunction, (void* )this, "StAudioQueue");
}
StAudioQueue::~StAudioQueue() {
myToQuit = true;
pushQuit();
myThread->wait();
myThread.nullify();
deinit();
}
bool StAudioQueue::setupOutMonoFormat() {
switch(myBufferSrc.getFormat()) {
case StPcmFormat_Float64: {
if(myAlCtx.hasExtFloat64) {
// use float64 extension
myAlFormat = alGetEnumValue("AL_FORMAT_MONO_DOUBLE_EXT");
myBufferOut.setFormat(StPcmFormat_Float64);
return true;
}
}
case StPcmFormat_Int32:
case StPcmFormat_Float32: {
if(myAlCtx.hasExtFloat32) {
// use float32 extension to preserve more quality
myAlFormat = alGetEnumValue("AL_FORMAT_MONO_FLOAT32");
myBufferOut.setFormat(StPcmFormat_Float32);
return true;
}
}
case StPcmFormat_Int16: {
// default - int16_t
myAlFormat = AL_FORMAT_MONO16;
myBufferOut.setFormat(StPcmFormat_Int16);
return true;
}
case StPcmFormat_UInt8: {
myAlFormat = AL_FORMAT_MONO8;
myBufferOut.setFormat(StPcmFormat_UInt8);
return true;
}
}
return false;
}
bool StAudioQueue::initOutMono() {
if(!setupOutMonoFormat()) {
return false;
}
myBufferSrc.setupChannels(StChannelMap::CH10, StChannelMap::PCM, 1);
myBufferOut.setupChannels(StChannelMap::CH10, StChannelMap::PCM, 1);
stalConfigureSources1();
return true;
}
bool StAudioQueue::initOut20Soft(const bool theIsPlanar) {
if(!setupOutMonoFormat()) {
return false;
}
myBufferOut.setupChannels(StChannelMap::CH20, StChannelMap::PCM, 2);
myBufferSrc.setupChannels(StChannelMap::CH20, StChannelMap::PCM, theIsPlanar ? myCodecCtx->channels : 1);
stalConfigureSources2_0();
return true;
}
bool StAudioQueue::initOut30Soft(const bool theIsPlanar) {
if(!setupOutMonoFormat()) {
return false;
}
myBufferOut.setupChannels(StChannelMap::CH30, StChannelMap::PCM, 3);
myBufferSrc.setupChannels(StChannelMap::CH30, StChannelMap::PCM, theIsPlanar ? myCodecCtx->channels : 1);
stalConfigureSources3_0();
return true;
}
bool StAudioQueue::initOutStereo(const bool theIsPlanar) {
switch(myBufferSrc.getFormat()) {
case StPcmFormat_Float64: {
if(myAlCtx.hasExtFloat64) {
// use float64 extension
myAlFormat = alGetEnumValue("AL_FORMAT_STEREO_DOUBLE_EXT");
myBufferOut.setFormat(StPcmFormat_Float64);
break;
}
}
case StPcmFormat_Int32:
case StPcmFormat_Float32: {
if(myAlCtx.hasExtFloat32) {
myAlFormat = alGetEnumValue("AL_FORMAT_STEREO_FLOAT32");
myBufferOut.setFormat(StPcmFormat_Float32);
break;
}
}
case StPcmFormat_Int16: {
// default - int16_t
myAlFormat = AL_FORMAT_STEREO16;
myBufferOut.setFormat(StPcmFormat_Int16);
break;
}
case StPcmFormat_UInt8: {
myAlFormat = AL_FORMAT_STEREO8;
myBufferOut.setFormat(StPcmFormat_UInt8);
break;
}
default: return false;
}
myBufferSrc.setupChannels(StChannelMap::CH20, StChannelMap::PCM, theIsPlanar ? myCodecCtx->channels : 1);
myBufferOut.setupChannels(StChannelMap::CH20, StChannelMap::PCM, 1);
stalConfigureSources1();
return true;
}
bool StAudioQueue::initOut40Soft(const bool theIsPlanar) {
if(!setupOutMonoFormat()) {
return false;
}
myBufferOut.setupChannels(StChannelMap::CH40, StChannelMap::PCM, 4);
myBufferSrc.setupChannels(StChannelMap::CH40, StChannelMap::PCM, theIsPlanar ? myCodecCtx->channels : 1);
stalConfigureSources4_0();
return true;
}
bool StAudioQueue::initOut40Ext(const bool theIsPlanar) {
switch(myBufferSrc.getFormat()) {
case StPcmFormat_Int32:
case StPcmFormat_Float32:
case StPcmFormat_Float64: {
myAlFormat = alGetEnumValue("AL_FORMAT_QUAD32");
myBufferOut.setFormat(StPcmFormat_Float32);
break;
}
case StPcmFormat_Int16: {
myAlFormat = alGetEnumValue("AL_FORMAT_QUAD16");
myBufferOut.setFormat(StPcmFormat_Int16);
break;
}
case StPcmFormat_UInt8: {
myAlFormat = alGetEnumValue("AL_FORMAT_QUAD8");
myBufferOut.setFormat(StPcmFormat_UInt8);
break;
}
default: return false;
}
myBufferSrc.setupChannels(StChannelMap::CH40, StChannelMap::PCM, theIsPlanar ? myCodecCtx->channels : 1);
myBufferOut.setupChannels(StChannelMap::CH40, StChannelMap::PCM, 1);
stalConfigureSources1();
return true;
}
bool StAudioQueue::initOut40BFormat(const bool theIsPlanar) {
switch(myBufferSrc.getFormat()) {
case StPcmFormat_Int32:
case StPcmFormat_Float32:
case StPcmFormat_Float64: {
myAlFormat = alGetEnumValue("AL_FORMAT_BFORMAT3D_FLOAT32");
myBufferOut.setFormat(StPcmFormat_Float32);
break;
}
case StPcmFormat_Int16: {
myAlFormat = alGetEnumValue("AL_FORMAT_BFORMAT3D_16");
myBufferOut.setFormat(StPcmFormat_Int16);
break;
}
case StPcmFormat_UInt8: {
myAlFormat = alGetEnumValue("AL_FORMAT_BFORMAT3D_8");
myBufferOut.setFormat(StPcmFormat_UInt8);
break;
}
default: return false;
}
myBufferSrc.setupChannels(StChannelMap::CH40, StChannelMap::WYZX, theIsPlanar ? myCodecCtx->channels : 1);
myBufferOut.setupChannels(StChannelMap::CH40, StChannelMap::PCM, 1);
stalConfigureSources1();
return true;
}
bool StAudioQueue::initOut50Soft(const bool theIsPlanar) {
if(!setupOutMonoFormat()) {
return false;
}
myBufferOut.setupChannels(StChannelMap::CH50, StChannelMap::PCM, 5);
myBufferSrc.setupChannels(StChannelMap::CH50, StChannelMap::PCM, theIsPlanar ? myCodecCtx->channels : 1);
stalConfigureSources5_0();
return true;
}
bool StAudioQueue::initOut51Soft(const bool theIsPlanar) {
if(!setupOutMonoFormat()) {
return false;
}
myBufferOut.setupChannels(StChannelMap::CH51, StChannelMap::PCM, 6);
#if(LIBAVCODEC_VERSION_INT < AV_VERSION_INT(53, 0, 0))
if(isReoderingNeeded()) {
// workaround for old FFmpeg
if(myCodec->id == CODEC_ID_AC3) {
myBufferSrc.setupChannels(StChannelMap::CH51, StChannelMap::AC3, 1);
} else if(myCodec->id == CODEC_ID_VORBIS) {
myBufferSrc.setupChannels(StChannelMap::CH51, StChannelMap::OGG, 1);
} else {
myBufferSrc.setupChannels(StChannelMap::CH51, StChannelMap::PCM, 1);
}
} else
#endif
{
myBufferSrc.setupChannels(StChannelMap::CH51, StChannelMap::PCM, theIsPlanar ? myCodecCtx->channels : 1);
}
stalConfigureSources5_1();
return true;
}
bool StAudioQueue::initOut51Ext(const bool theIsPlanar) {
switch(myBufferSrc.getFormat()) {
case StPcmFormat_Int32:
case StPcmFormat_Float32:
case StPcmFormat_Float64: {
myAlFormat = alGetEnumValue("AL_FORMAT_51CHN32");
myBufferOut.setFormat(StPcmFormat_Float32);
break;
}
case StPcmFormat_Int16: {
myAlFormat = alGetEnumValue("AL_FORMAT_51CHN16");
myBufferOut.setFormat(StPcmFormat_Int16);
break;
}
case StPcmFormat_UInt8: {
myAlFormat = alGetEnumValue("AL_FORMAT_51CHN8");
myBufferOut.setFormat(StPcmFormat_UInt8);
break;
}
default: return false;
}
myBufferOut.setupChannels(StChannelMap::CH51, StChannelMap::PCM, 1);
#if(LIBAVCODEC_VERSION_INT < AV_VERSION_INT(53, 0, 0))
if(isReoderingNeeded()) {
// workaround for old FFmpeg
if(myCodec->id == CODEC_ID_AC3) {
myBufferSrc.setupChannels(StChannelMap::CH51, StChannelMap::AC3, 1);
} else if(myCodec->id == CODEC_ID_VORBIS) {
myBufferSrc.setupChannels(StChannelMap::CH51, StChannelMap::OGG, 1);
} else {
myBufferSrc.setupChannels(StChannelMap::CH51, StChannelMap::PCM, 1);
}
} else
#endif
{
myBufferSrc.setupChannels(StChannelMap::CH51, StChannelMap::PCM, theIsPlanar ? myCodecCtx->channels : 1);
}
stalConfigureSources1();
return true;
}
bool StAudioQueue::initOut71Soft(const bool theIsPlanar) {
if(!setupOutMonoFormat()) {
return false;
}
myBufferOut.setupChannels(StChannelMap::CH71, StChannelMap::PCM, 8);
myBufferSrc.setupChannels(StChannelMap::CH71, StChannelMap::PCM, theIsPlanar ? myCodecCtx->channels : 1);
stalConfigureSources7_1();
return true;
}
bool StAudioQueue::initOut71Ext(const bool theIsPlanar) {
if(!myAlCtx.hasExtMultiChannel
|| myCodecCtx->channels != 8) {
return false;
}
switch(myBufferSrc.getFormat()) {
case StPcmFormat_Int32:
case StPcmFormat_Float32:
case StPcmFormat_Float64: {
myAlFormat = alGetEnumValue("AL_FORMAT_71CHN32");
myBufferOut.setFormat(StPcmFormat_Float32);
break;
}
case StPcmFormat_Int16: {
myAlFormat = alGetEnumValue("AL_FORMAT_71CHN16");
myBufferOut.setFormat(StPcmFormat_Int16);
break;
}
case StPcmFormat_UInt8: {
myAlFormat = alGetEnumValue("AL_FORMAT_71CHN8");
myBufferOut.setFormat(StPcmFormat_UInt8);
break;
}
default: return false;
}
myBufferSrc.setupChannels(StChannelMap::CH71, StChannelMap::PCM, theIsPlanar ? myCodecCtx->channels : 1);
myBufferOut.setupChannels(StChannelMap::CH71, StChannelMap::PCM, 1);
stalConfigureSources1();
return true;
}
bool StAudioQueue::initOutChannels() {
const bool isPlanar =
#if(LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53, 40, 0))
av_sample_fmt_is_planar(myCodecCtx->sample_fmt) != 0;
#else
false;
#endif
myAlCanBFormat = false;
myAlIsBFormat = false;
switch(myCodecCtx->channels) {
case 1: {
myAlSoftLayout = true; // just unsupported
return initOutMono();
}
case 2: {
if(!myToOrientListener) {
myAlSoftLayout = false;
return initOutStereo(isPlanar);
} else {
myAlSoftLayout = true;
return initOut20Soft(isPlanar);
}
}
case 3: {
myAlSoftLayout = true;
return initOut30Soft(isPlanar);
}
case 4: {
myAlCanBFormat = myAlCtx.hasExtBFormat;
if(myToForceBFormat && myAlCtx.hasExtBFormat) {
myAlSoftLayout = true;
myAlIsBFormat = true;
return initOut40BFormat(isPlanar);
}
if(myAlCtx.hasExtMultiChannel && !myToOrientListener) {
myAlSoftLayout = false;
return initOut40Ext(isPlanar);
}
ST_DEBUG_LOG("OpenAL: multichannel extension (AL_FORMAT_QUAD16) is unavailable");
myAlSoftLayout = true;
return initOut40Soft(isPlanar);
}
case 5: {
myAlSoftLayout = true;
return initOut50Soft(isPlanar);
}
case 6: {
if(myAlCtx.hasExtMultiChannel && !myToOrientListener) {
myAlSoftLayout = false;
return initOut51Ext(isPlanar);
}
ST_DEBUG_LOG("OpenAL: multichannel extension (AL_FORMAT_51CHN16) is unavailable");
myAlSoftLayout = true;
return initOut51Soft(isPlanar);
}
case 8: {
if(myAlCtx.hasExtMultiChannel && !myToOrientListener) {
myAlSoftLayout = false;
return initOut71Ext(isPlanar);
}
ST_DEBUG_LOG("OpenAL: multichannel extension (AL_FORMAT_71CHN16) is unavailable");
myAlSoftLayout = true;
return initOut71Soft(isPlanar);
}
default: {
myAlSoftLayout = true;
return false;
}
}
}
bool StAudioQueue::init(AVFormatContext* theFormatCtx,
const unsigned int theStreamId,
const StString& theFileName) {
while(myIsAlValid == ST_AL_INIT_NA) {
StThread::sleep(10);
continue;
}
if(myIsAlValid != ST_AL_INIT_OK) {
signals.onError(stCString("OpenAL: no playback device available"));
deinit();
return false;
}
if(!StAVPacketQueue::init(theFormatCtx, theStreamId, theFileName)
|| myCodecCtx->codec_type != AVMEDIA_TYPE_AUDIO) {
signals.onError(stCString("FFmpeg: invalid stream"));
deinit();
return false;
}
// get AUDIO codec
myCodec = avcodec_find_decoder(myCodecCtx->codec_id);
if(myCodec == NULL) {
signals.onError(stCString("FFmpeg: audio codec not found"));
deinit();
return false;
}
#if(LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53, 8, 0))
if(avcodec_open2(myCodecCtx, myCodec, NULL) < 0) {
#else
if(avcodec_open(myCodecCtx, myCodec) < 0) {
#endif
signals.onError(stCString("FFmpeg: could not open audio codec"));
deinit();
return false;
}
// setup buffers
if(!initBuffers()) {
deinit();
return false;
}
fillCodecInfo(myCodec);
return true;
}
bool StAudioQueue::initBuffers() {
myAvSrcFormat = myCodecCtx->sample_fmt;
myAvSampleRate = myCodecCtx->sample_rate;
myAvNbChannels = myCodecCtx->channels;
if(myCodecCtx->sample_rate < FREQ_5500) {
signals.onError(StString("FFmpeg: wrong audio frequency ") + myCodecCtx->sample_rate);
deinit();
return false;
}
// setup source sample format (bitness)
if(myCodecCtx->sample_fmt == stAV::audio::SAMPLE_FMT::NONE) {
signals.onError(stCString("Invalid audio sample format!"));
deinit();
return false;
} else if(myCodecCtx->sample_fmt == stAV::audio::SAMPLE_FMT::U8
|| myCodecCtx->sample_fmt == stAV::audio::SAMPLE_FMT::U8P) {
myBufferSrc.setFormat(StPcmFormat_UInt8);
} else if(myCodecCtx->sample_fmt == stAV::audio::SAMPLE_FMT::S16
|| myCodecCtx->sample_fmt == stAV::audio::SAMPLE_FMT::S16P) {
myBufferSrc.setFormat(StPcmFormat_Int16);
} else if(myCodecCtx->sample_fmt == stAV::audio::SAMPLE_FMT::S32
|| myCodecCtx->sample_fmt == stAV::audio::SAMPLE_FMT::S32P) {
myBufferSrc.setFormat(StPcmFormat_Int32);
} else if(myCodecCtx->sample_fmt == stAV::audio::SAMPLE_FMT::FLT
|| myCodecCtx->sample_fmt == stAV::audio::SAMPLE_FMT::FLTP) {
myBufferSrc.setFormat(StPcmFormat_Float32);
} else if(myCodecCtx->sample_fmt == stAV::audio::SAMPLE_FMT::DBL
|| myCodecCtx->sample_fmt == stAV::audio::SAMPLE_FMT::DBLP) {
myBufferSrc.setFormat(StPcmFormat_Float64);
} else {
signals.onError(StString("Audio sample format '") + stAV::audio::getSampleFormatString(myCodecCtx)
+ "' not supported");
deinit();
return false;
}
// setup frequency
myBufferSrc.setFreq(myCodecCtx->sample_rate);
myBufferOut.setFreq(myCodecCtx->sample_rate);
// setup channel order
if(!initOutChannels()) {
deinit();
signals.onError(stCString("OpenAL: unsupported format or channels configuration"));
return false;
}
return true;
}
void StAudioQueue::deinit() {
myBufferSrc.clear();
myBufferOut.clear();
myAvSrcFormat = -1;
myAvSampleRate = -1;
myAvNbChannels = -1;
myAlSoftLayout = true;
myAlCanBFormat = false;
myAlIsBFormat = false;
StAVPacketQueue::deinit();
}
bool StAudioQueue::parseEvents() {
double aPtsSeek = 0.0;
bool toResetBuffers = false;
if(myToSwitchDev) {
stalReinitialize();
return true;
} else if(myAlHrtf != myAlHrtfPrev) {
stalResetHrtf();
StMutexAuto aLock(myAlInfoMutex);
myAlInfo.clear();
myAlCtx.fullInfo(myAlInfo);
} else if(myAlIsBFormat != myToForceBFormat
&& myAlCanBFormat) {
toResetBuffers = true;
}
if(myToOrientListener) {
if(!myAlSoftLayout) {
toResetBuffers = true;
} else {
stalOrientListener();
}
} else if(myAlIsListOrient) {
stalOrientListener();
}
if(toResetBuffers) {
myBufferSrc.clear();
myBufferOut.clear();
initBuffers();
return true;
}
if(!stAreEqual(myAlGain, myAlGainPrev, 1.e-7f)) {
ST_DEBUG_LOG("Audio volume changed from " + myAlGainPrev + " to " + myAlGain);
myAlGainPrev = myAlGain;
alListenerf(AL_GAIN, myAlGain); // apply gain to all sources at-once
}
switch(popPlayEvent(aPtsSeek)) {
case ST_PLAYEVENT_PLAY: {
stalEmpty();
playTimerStart(0.0);
playTimerPause();
return false;
}
case ST_PLAYEVENT_STOP: {
playTimerPause();
stalEmpty();
return false;
}
case ST_PLAYEVENT_PAUSE: {
playTimerPause();
alSourcePausev(THE_NUM_AL_SOURCES, myAlSources);
return false;
}
case ST_PLAYEVENT_RESUME: {
playTimerResume();
alSourcePlayv(THE_NUM_AL_SOURCES, myAlSources);
return false;
}
case ST_PLAYEVENT_SEEK: {
stalEmpty();
playTimerStart(aPtsSeek);
playTimerPause();
myBufferSrc.setDataSize(0);
myBufferOut.setDataSize(0);
// return special flag to skip "resume playback from" in loop
return true;
}
case ST_PLAYEVENT_NONE:
default:
return false;
}
}
bool StAudioQueue::stalQueue(const double thePts) {
ALint aQueued = 0;
ALint aProcessed = 0;
ALenum aState = stalGetSourceState();
alGetSourcei(myAlSources[0], AL_BUFFERS_PROCESSED, &aProcessed);
alGetSourcei(myAlSources[0], AL_BUFFERS_QUEUED, &aQueued);
#ifdef ST_DEBUG
if(myDbgPrevQueued != aQueued) {
ST_DEBUG_LOG("OpenAL buffers: " + aQueued + " queued + "
+ aProcessed + " processed from " + THE_NUM_AL_BUFFERS
);
}
myDbgPrevQueued = aQueued;
#else
(void )myDbgPrevQueued;
#endif
if((aState == AL_PLAYING
|| aState == AL_PAUSED)
&& (myPrevFormat != myAlFormat
|| myPrevFrequency != myBufferOut.getFreq()))
{
return false; // wait until tail of previous stream played
}
if(myPrevFormat != myAlFormat
|| myPrevFrequency != myBufferOut.getFreq()
|| (aState == AL_STOPPED
&& aQueued == THE_NUM_AL_BUFFERS)) {
ST_DEBUG_LOG("AL, reinitialize buffers per source , plane size= " + myBufferOut.getPlaneSize()
+ "; freq= " + myBufferOut.getFreq());
stalEmpty();
stalCheckErrors("reset state");
aProcessed = 0;
aQueued = 0;
}
bool toTryToPlay = false;
bool isQueued = false;
if(aProcessed == 0 && aQueued < THE_NUM_AL_BUFFERS) {
if(myBufferOut.isEmpty()) {
ST_DEBUG_LOG(" EMPTY BUFFER ");
return true;
}
stalCheckErrors("reset state");
///ST_DEBUG_LOG("AL, queue more buffers " + aQueued + " / " + NUM_AL_BUFFERS);
myPrevFormat = myAlFormat;
myPrevFrequency = myBufferOut.getFreq();
for(size_t aSrcId = 0; aSrcId < myBufferOut.getPlanesNb(); ++aSrcId) {
alBufferData(myAlBuffers[aSrcId][aQueued], myAlFormat,
myBufferOut.getPlane(aSrcId), (ALsizei )myBufferOut.getPlaneSize(),
myBufferOut.getFreq());
stalCheckErrors("alBufferData1");
alSourceQueueBuffers(myAlSources[aSrcId], 1, &myAlBuffers[aSrcId][aQueued]);
stalCheckErrors("alSourceQueueBuffers");
}
toTryToPlay = ((aQueued + 1) == THE_NUM_AL_BUFFERS);
isQueued = true;
} else if(aProcessed != 0
&& (aState == AL_PLAYING
|| aState == AL_PAUSED)) {
ALuint alBuffIdToFill = 0;
///ST_DEBUG_LOG("queue buffer " + thePts + "; state= " + stalGetSourceState());
if(myBufferOut.isEmpty()) {
ST_DEBUG_LOG(" EMPTY BUFFER ");
return true;
}
myPrevFormat = myAlFormat;
myPrevFrequency = myBufferOut.getFreq();
for(size_t aSrcId = 0; aSrcId < myBufferOut.getPlanesNb(); ++aSrcId) {
// wait other sources for processed buffers
if(aSrcId != 0) {
myLimitTimer.restart();
for(;;) {
alGetSourcei(myAlSources[aSrcId], AL_BUFFERS_PROCESSED, &aProcessed);
if(aProcessed != 0) {
break;
}
if(myLimitTimer.getElapsedTimeInSec() > 2.0) {
// just avoid dead loop - should never happens
return false;
}
StThread::sleep(10);
}
}