/*
* Free codec.
*/
static pj_status_t ilbc_dealloc_codec( pjmedia_codec_factory *factory,
pjmedia_codec *codec )
{
struct ilbc_codec *ilbc_codec;
PJ_ASSERT_RETURN(factory && codec, PJ_EINVAL);
PJ_UNUSED_ARG(factory);
PJ_ASSERT_RETURN(factory == &ilbc_factory.base, PJ_EINVAL);
ilbc_codec = (struct ilbc_codec*) codec;
#if defined(PJMEDIA_ILBC_CODEC_USE_COREAUDIO)&& PJMEDIA_ILBC_CODEC_USE_COREAUDIO
if (ilbc_codec->enc) {
AudioConverterDispose(ilbc_codec->enc);
ilbc_codec->enc = NULL;
}
if (ilbc_codec->dec) {
AudioConverterDispose(ilbc_codec->dec);
ilbc_codec->dec = NULL;
}
#endif
pj_pool_release(ilbc_codec->pool);
return PJ_SUCCESS;
}
static GVBool gviHardwareInitCapture(GVIDevice * device)
{
GVIHardwareData * data = (GVIHardwareData *)device->m_data;
UInt32 size;
OSStatus result;
GVICapturedFrame * frame;
int numCaptureBufferBytes;
int numCaptureBufferFrames;
int i;
// get the capture format
size = sizeof(AudioStreamBasicDescription);
result = AudioDeviceGetProperty(device->m_deviceID, 0, true, kAudioDevicePropertyStreamFormat, &size, &data->m_captureStreamDescriptor);
if(result != noErr)
return GVFalse;
// create a converter from the capture format to the GV format
result = AudioConverterNew(&data->m_captureStreamDescriptor, &GVIVoiceFormat, &data->m_captureConverter);
if(result != noErr)
return GVFalse;
// allocate a capture buffer
data->m_captureBuffer = (GVSample *)gsimalloc(GVIBytesPerFrame);
if(!data->m_captureBuffer)
{
AudioConverterDispose(data->m_captureConverter);
return GVFalse;
}
// allocate space for holding captured frames
numCaptureBufferBytes = gviMultiplyByBytesPerMillisecond(GVI_CAPTURE_BUFFER_MILLISECONDS);
numCaptureBufferBytes = gviRoundUpToNearestMultiple(numCaptureBufferBytes, GVIBytesPerFrame);
numCaptureBufferFrames = (numCaptureBufferBytes / GVIBytesPerFrame);
for(i = 0 ; i < numCaptureBufferFrames ; i++)
{
frame = (GVICapturedFrame *)gsimalloc(sizeof(GVICapturedFrame) + GVIBytesPerFrame - sizeof(GVSample));
if(!frame)
{
gviFreeCapturedFrames(&data->m_captureAvailableFrames);
gsifree(data->m_captureBuffer);
AudioConverterDispose(data->m_captureConverter);
return GVFalse;
}
gviPushFirstFrame(&data->m_captureAvailableFrames, frame);
}
// init the last crossed time
data->m_captureLastCrossedThresholdTime = (data->m_captureClock - GVI_HOLD_THRESHOLD_FRAMES - 1);
// add property listener
AudioDeviceAddPropertyListener(device->m_deviceID, 0, true, kAudioDevicePropertyDeviceIsAlive, gviPropertyListener, device);
#if GVI_VOLUME_IN_SOFTWARE
// init volume
data->m_captureVolume = (GVScalar)1.0;
#endif
return GVTrue;
}
void
AACEncode::setBitrate(int bitrate)
{
if(m_bitrate != bitrate) {
m_converterMutex.lock();
UInt32 br = bitrate;
AudioConverterDispose(m_audioConverter);
const OSType subtype = kAudioFormatMPEG4AAC;
AudioClassDescription requestedCodecs[2] = {
{
kAudioEncoderComponentType,
subtype,
kAppleSoftwareAudioCodecManufacturer
},
{
kAudioEncoderComponentType,
subtype,
kAppleHardwareAudioCodecManufacturer
}
};
AudioConverterNewSpecific(&m_in, &m_out, 2,requestedCodecs, &m_audioConverter);
OSStatus result = AudioConverterSetProperty(m_audioConverter, kAudioConverterEncodeBitRate, sizeof(br), &br);
UInt32 propSize = sizeof(br);
if(result == noErr) {
AudioConverterGetProperty(m_audioConverter, kAudioConverterEncodeBitRate, &propSize, &br);
m_bitrate = br;
}
m_converterMutex.unlock();
}
}
static int close_coreaudio(audio_output_t *ao)
{
mpg123_coreaudio_t* ca = (mpg123_coreaudio_t*)ao->userptr;
if (ca) {
ca->decode_done = 1;
while(!ca->play_done && ca->play) usleep(10000);
/* No matter the error code, we want to close it (by brute force if necessary) */
AudioConverterDispose(ca->converter);
AudioOutputUnitStop(ca->outputUnit);
AudioUnitUninitialize(ca->outputUnit);
CloseComponent(ca->outputUnit);
/* Free the ring buffer */
sfifo_close( &ca->fifo );
/* Free the conversion buffer */
if (ca->buffer) {
free( ca->buffer );
ca->buffer = NULL;
}
}
return 0;
}
void ca_close(phastream_t *as) {
ca_dev *cadev = (ca_dev *) as->drvinfo;
DBG_DYNA_AUDIO_DRV("** Closing audio stream\n");
verify_noerr(AudioDeviceStop(get_audiodeviceid(cadev->inputID), input_proc));
clean_input_device(as);
verify_noerr(AudioOutputUnitStop(cadev->outputAU));
verify_noerr(AudioUnitUninitialize (cadev->outputAU));
printf("phad_coreaudio:ca_close:cleaning cadev\n"); // power pc hack 2/2
if (cadev) {
if (cadev->convertedInputBuffer) {
free(cadev->convertedInputBuffer);
}
cadev->convertedInputBuffer = NULL;
if (cadev->inputConverter) {
AudioConverterDispose(cadev->inputConverter);
cadev->inputConverter = NULL;
}
free(cadev);
as->drvinfo = 0;
}
}
Audio_Stream::~Audio_Stream()
{
if (m_defaultContentType) {
CFRelease(m_defaultContentType), m_defaultContentType = NULL;
}
if (m_contentType) {
CFRelease(m_contentType), m_contentType = NULL;
}
close();
delete [] m_outputBuffer, m_outputBuffer = 0;
m_httpStream->m_delegate = 0;
delete m_httpStream, m_httpStream = 0;
if (m_audioConverter) {
AudioConverterDispose(m_audioConverter), m_audioConverter = 0;
}
if (m_fileOutput) {
delete m_fileOutput, m_fileOutput = 0;
}
}
static void gviCleanupCapture(GVIHardwareData * data)
{
if(data->m_captureConverter)
AudioConverterDispose(data->m_captureConverter);
gviFreeCapturedFrames(&data->m_captureAvailableFrames);
gviFreeCapturedFrames(&data->m_capturedFrames);
gsifree(data->m_captureBuffer);
}
static void gviCleanupPlayback(GVIHardwareData * data)
{
if(data->m_playbackSources)
gviFreeSourceList(data->m_playbackSources);
if(data->m_playbackConverter)
AudioConverterDispose(data->m_playbackConverter);
gsifree(data->m_playbackBuffer);
}
RageSound_CA::~RageSound_CA()
{
if( mOutputDevice != NULL )
mOutputDevice->StopIOProc( GetData );
delete mOutputDevice;
if( mConverter != NULL )
AudioConverterDispose( mConverter );
}
// _______________________________________________________________________________________
//
void CAAudioFile::CloseConverter()
{
if (mConverter) {
#if VERBOSE_CONVERTER
printf("CAAudioFile %p : CloseConverter\n", this);
#endif
AudioConverterDispose(mConverter);
mConverter = NULL;
}
}
/***********************************************************************
* Close
***********************************************************************
*
**********************************************************************/
void encCoreAudioClose( hb_work_object_t * w )
{
hb_work_private_t * pv = w->private_data;
if( pv->converter )
{
AudioConverterDispose( pv->converter );
hb_list_empty( &pv->list );
free( pv->obuf );
free( pv->buf );
free( pv );
w->private_data = NULL;
}
}
static void ca_close_capture(ALCdevice *device)
{
ca_data *data = (ca_data*)device->ExtraData;
DestroyRingBuffer(data->ring);
free(data->resampleBuffer);
destroy_buffer_list(data->bufferList);
AudioConverterDispose(data->audioConverter);
CloseComponent(data->audioUnit);
free(data);
device->ExtraData = NULL;
}
static void aq_stop_w(MSFilter * f)
{
AQData *d = (AQData *) f->data;
if (d->write_started == TRUE) {
ms_mutex_lock(&d->mutex);
d->write_started = FALSE; /* avoid a deadlock related to buffer conversion in callback */
ms_mutex_unlock(&d->mutex);
#if 0
AudioConverterDispose(d->writeAudioConverter);
#endif
AudioQueueStop(d->writeQueue, true);
AudioQueueDispose(d->writeQueue, true);
}
}
bool SFB::Audio::Converter::Close(CFErrorRef *error)
{
if(!IsOpen()) {
LOGGER_WARNING("org.sbooth.AudioEngine.AudioConverter", "Close() called on an AudioConverter that hasn't been opened");
return true;
}
mConverterState.reset();
mDecoder.reset();
if(mConverter)
AudioConverterDispose(mConverter), mConverter = nullptr;
mIsOpen = false;
return true;
}
static void ca_stop_w(CAData *d){
OSErr err;
if(d->write_started == TRUE) {
if(AudioOutputUnitStop(d->caOutAudioUnit) == noErr)
d->write_started=FALSE;
}
if (d->caOutConverter!=NULL)
{
AudioConverterDispose(d->caOutConverter);
d->caOutConverter=NULL;
}
if (d->caOutAudioUnit!=NULL)
{
AudioUnitUninitialize(d->caOutAudioUnit);
d->caOutAudioUnit=NULL;
}
}
Audio_Stream::~Audio_Stream()
{
close();
delete [] m_outputBuffer, m_outputBuffer = 0;
m_httpStream->m_delegate = 0;
delete m_httpStream, m_httpStream = 0;
if (m_audioConverter) {
AudioConverterDispose(m_audioConverter), m_audioConverter = 0;
}
if (m_fileOutput) {
delete m_fileOutput, m_fileOutput = 0;
}
}
static void aq_stop_w(MSFilter * f)
{
AQData *d = (AQData *) f->data;
if (d->write_started == TRUE) {
ms_mutex_lock(&d->mutex);
d->write_started = FALSE; /* avoid a deadlock related to buffer conversion in callback */
ms_mutex_unlock(&d->mutex);
#if 0
AudioConverterDispose(d->writeAudioConverter);
#endif
AudioQueueStop(d->writeQueue, true);
AudioQueueDispose(d->writeQueue, true);
#if TARGET_OS_IPHONE
check_aqresult(AudioSessionSetActive(false),"AudioSessionSetActive(false)");
#endif
}
}
nsresult
AppleATDecoder::Shutdown()
{
LOG("Shutdown: Apple AudioToolbox AAC decoder");
OSStatus rv1 = AudioConverterDispose(mConverter);
if (rv1) {
LOG("error %d disposing of AudioConverter", rv1);
} else {
mConverter = nullptr;
}
OSStatus rv2 = AudioFileStreamClose(mStream);
if (rv2) {
LOG("error %d closing AudioFileStream", rv2);
} else {
mStream = nullptr;
}
return (rv1 && rv2) ? NS_OK : NS_ERROR_FAILURE;
}
nsresult
AppleATDecoder::Shutdown()
{
LOG("Shutdown: Apple AudioToolbox AAC decoder");
mQueuedSamples.Clear();
OSStatus rv = AudioConverterDispose(mConverter);
if (rv) {
LOG("error %d disposing of AudioConverter", rv);
return NS_ERROR_FAILURE;
}
mConverter = nullptr;
if (mStream) {
rv = AudioFileStreamClose(mStream);
if (rv) {
LOG("error %d disposing of AudioFileStream", rv);
return NS_ERROR_FAILURE;
}
mStream = nullptr;
}
return NS_OK;
}
void
AppleATDecoder::ProcessShutdown()
{
MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
if (mStream) {
OSStatus rv = AudioFileStreamClose(mStream);
if (rv) {
LOG("error %d disposing of AudioFileStream", rv);
return;
}
mStream = nullptr;
}
if (mConverter) {
LOG("Shutdown: Apple AudioToolbox AAC decoder");
OSStatus rv = AudioConverterDispose(mConverter);
if (rv) {
LOG("error %d disposing of AudioConverter", rv);
}
mConverter = nullptr;
}
}
/***********************************************************************
* Close
***********************************************************************
*
**********************************************************************/
void encCoreAudioClose(hb_work_object_t *w)
{
hb_work_private_t *pv = w->private_data;
if (pv != NULL)
{
if (pv->converter)
{
AudioConverterDispose(pv->converter);
}
if (pv->buf != NULL)
{
free(pv->buf);
}
if (pv->remap != NULL)
{
hb_audio_remap_free(pv->remap);
}
hb_list_empty(&pv->list);
free(pv);
w->private_data = NULL;
}
}
static void ca_stop_r(CAData *d){
OSErr err;
if(d->read_started == TRUE) {
if(AudioOutputUnitStop(d->caInAudioUnit) == noErr)
d->read_started=FALSE;
}
if (d->caInConverter!=NULL)
{
AudioConverterDispose(d->caInConverter);
d->caInConverter=NULL;
}
if (d->caInAudioUnit!=NULL)
{
AudioUnitUninitialize(d->caInAudioUnit);
d->caInAudioUnit=NULL;
}
if (d->fAudioBuffer)
DestroyAudioBufferList(d->fAudioBuffer);
d->fAudioBuffer=NULL;
if (d->fMSBuffer)
DestroyAudioBufferList(d->fMSBuffer);
d->fMSBuffer=NULL;
}
//-----------------------------------------------------------------------------
bool AudioFile::Load()
{
#if MAC
AudioFileID mAudioFileID;
AudioStreamBasicDescription fileDescription, outputFormat;
SInt64 dataSize64;
UInt32 dataSize;
OSStatus err;
UInt32 size;
// ファイルを開く
FSRef ref;
Boolean isDirectory=false;
FSPathMakeRef((const UInt8*)GetFilePath(), &ref, &isDirectory);
err = AudioFileOpen(&ref, fsRdPerm, 0, &mAudioFileID);
if (err) {
//NSLog(@"AudioFileOpen failed");
return false;
}
// 開いたファイルの基本情報を fileDescription へ
size = sizeof(AudioStreamBasicDescription);
err = AudioFileGetProperty(mAudioFileID, kAudioFilePropertyDataFormat,
&size, &fileDescription);
if (err) {
//NSLog(@"AudioFileGetProperty failed");
AudioFileClose(mAudioFileID);
return false;
}
// 開いたファイルのデータ部のバイト数を dataSize へ
size = sizeof(SInt64);
err = AudioFileGetProperty(mAudioFileID, kAudioFilePropertyAudioDataByteCount,
&size, &dataSize64);
if (err) {
//NSLog(@"AudioFileGetProperty failed");
AudioFileClose(mAudioFileID);
return false;
}
dataSize = static_cast<UInt32>(dataSize64);
AudioFileTypeID fileTypeID;
size = sizeof( AudioFileTypeID );
err = AudioFileGetProperty(mAudioFileID, kAudioFilePropertyFileFormat, &size, &fileTypeID);
if (err) {
//NSLog(@"AudioFileGetProperty failed");
AudioFileClose(mAudioFileID);
return false;
}
// Instrument情報を初期化
mInstData.basekey = 60;
mInstData.lowkey = 0;
mInstData.highkey = 127;
mInstData.loop = 0;
//ループポイントの取得
Float64 st_point=0.0,end_point=0.0;
if ( fileTypeID == kAudioFileAIFFType || fileTypeID == kAudioFileAIFCType ) {
//INSTチャンクの取得
AudioFileGetUserDataSize(mAudioFileID, 'INST', 0, &size);
if ( size > 4 ) {
UInt8 *instChunk = new UInt8[size];
AudioFileGetUserData(mAudioFileID, 'INST', 0, &size, instChunk);
//MIDI情報の取得
mInstData.basekey = instChunk[0];
mInstData.lowkey = instChunk[2];
mInstData.highkey = instChunk[3];
if ( instChunk[9] > 0 ) { //ループフラグを確認
//マーカーの取得
UInt32 writable;
err = AudioFileGetPropertyInfo(mAudioFileID, kAudioFilePropertyMarkerList,
&size, &writable);
if (err) {
//NSLog(@"AudioFileGetPropertyInfo failed");
AudioFileClose(mAudioFileID);
return NULL;
}
UInt8 *markersBuffer = new UInt8[size];
AudioFileMarkerList *markers = reinterpret_cast<AudioFileMarkerList*>(markersBuffer);
err = AudioFileGetProperty(mAudioFileID, kAudioFilePropertyMarkerList,
&size, markers);
if (err) {
//NSLog(@"AudioFileGetProperty failed");
AudioFileClose(mAudioFileID);
return NULL;
}
//ループポイントの設定
for (unsigned int i=0; i<markers->mNumberMarkers; i++) {
if (markers->mMarkers[i].mMarkerID == instChunk[11] ) {
st_point = markers->mMarkers[i].mFramePosition;
}
else if (markers->mMarkers[i].mMarkerID == instChunk[13] ) {
end_point = markers->mMarkers[i].mFramePosition;
}
CFRelease(markers->mMarkers[i].mName);
}
if ( st_point < end_point ) {
mInstData.loop = 1;
}
delete [] markersBuffer;
}
delete [] instChunk;
}
}
else if ( fileTypeID == kAudioFileWAVEType ) {
//smplチャンクの取得
AudioFileGetUserDataSize( mAudioFileID, 'smpl', 0, &size );
if ( size >= sizeof(WAV_smpl) ) {
UInt8 *smplChunk = new UInt8[size];
AudioFileGetUserData( mAudioFileID, 'smpl', 0, &size, smplChunk );
WAV_smpl *smpl = (WAV_smpl *)smplChunk;
smpl->loops = EndianU32_LtoN( smpl->loops );
if ( smpl->loops > 0 ) {
mInstData.loop = true;
mInstData.basekey = EndianU32_LtoN( smpl->note );
st_point = EndianU32_LtoN( smpl->start );
end_point = EndianU32_LtoN( smpl->end ) + 1; //SoundForge等では最終ポイントを含める解釈
//end_point = EndianU32_LtoN( smpl->end ); //PeakではなぜかAIFFと同じ
}
else {
mInstData.basekey = EndianU32_LtoN( smpl->note );
}
delete [] smplChunk;
}
}
//容量の制限
SInt64 dataSamples = dataSize / fileDescription.mBytesPerFrame;
if ( dataSamples > MAXIMUM_SAMPLES ) {
dataSize = MAXIMUM_SAMPLES * fileDescription.mBytesPerFrame;
}
if ( st_point > MAXIMUM_SAMPLES ) {
st_point = MAXIMUM_SAMPLES;
}
if ( end_point > MAXIMUM_SAMPLES ) {
end_point = MAXIMUM_SAMPLES;
}
// 波形一時読み込み用メモリを確保
char *fileBuffer;
unsigned int fileBufferSize;
if (mInstData.loop) {
fileBufferSize = dataSize+EXPAND_BUFFER*fileDescription.mBytesPerFrame;
}
else {
fileBufferSize = dataSize;
}
fileBuffer = new char[fileBufferSize];
memset(fileBuffer, 0, fileBufferSize);
// ファイルから波形データの読み込み
err = AudioFileReadBytes(mAudioFileID, false, 0, &dataSize, fileBuffer);
if (err) {
//NSLog(@"AudioFileReadBytes failed");
AudioFileClose(mAudioFileID);
delete [] fileBuffer;
return false;
}
AudioFileClose(mAudioFileID);
//ループを展開する
Float64 adjustment = 1.0;
outputFormat=fileDescription;
if (mInstData.loop) {
UInt32 plusalpha=0, framestocopy;
while (plusalpha < EXPAND_BUFFER) {
framestocopy =
(end_point-st_point)>(EXPAND_BUFFER-plusalpha)?(EXPAND_BUFFER-plusalpha):end_point-st_point;
memcpy(fileBuffer+((int)end_point+plusalpha)*fileDescription.mBytesPerFrame,
fileBuffer+(int)st_point*fileDescription.mBytesPerFrame,
framestocopy*fileDescription.mBytesPerFrame);
plusalpha += framestocopy;
}
dataSize += plusalpha*fileDescription.mBytesPerFrame;
//16サンプル境界にFIXする
adjustment = ( (long long)((end_point-st_point)/16) ) / ((end_point-st_point)/16.0);
st_point *= adjustment;
end_point *= adjustment;
}
outputFormat.mFormatID = kAudioFormatLinearPCM;
outputFormat.mFormatFlags = kAudioFormatFlagIsFloat | kAudioFormatFlagsNativeEndian;
outputFormat.mChannelsPerFrame = 1;
outputFormat.mBytesPerFrame = sizeof(float);
outputFormat.mBitsPerChannel = 32;
outputFormat.mBytesPerPacket = outputFormat.mBytesPerFrame;
// バイトオーダー変換用のコンバータを用意
AudioConverterRef converter;
err = AudioConverterNew(&fileDescription, &outputFormat, &converter);
if (err) {
//NSLog(@"AudioConverterNew failed");
delete [] fileBuffer;
return false;
}
//サンプリングレート変換の質を最高に設定
// if (fileDescription.mSampleRate != outputFormat.mSampleRate) {
// size = sizeof(UInt32);
// UInt32 setProp = kAudioConverterQuality_Max;
// AudioConverterSetProperty(converter, kAudioConverterSampleRateConverterQuality,
// size, &setProp);
//
// size = sizeof(UInt32);
// setProp = kAudioConverterSampleRateConverterComplexity_Mastering;
// AudioConverterSetProperty(converter, kAudioConverterSampleRateConverterComplexity,
// size, &setProp);
//
// }
//出力に必要十分なバッファサイズを得る
UInt32 outputSize = dataSize;
size = sizeof(UInt32);
err = AudioConverterGetProperty(converter, kAudioConverterPropertyCalculateOutputBufferSize,
&size, &outputSize);
if (err) {
//NSLog(@"AudioConverterGetProperty failed");
delete [] fileBuffer;
AudioConverterDispose(converter);
return false;
}
UInt32 monoSamples = outputSize/sizeof(float);
// バイトオーダー変換
float *monoData = new float[monoSamples];
AudioConverterConvertBuffer(converter, dataSize, fileBuffer,
&outputSize, monoData);
if(outputSize == 0) {
//NSLog(@"AudioConverterConvertBuffer failed");
delete [] fileBuffer;
AudioConverterDispose(converter);
return false;
}
//ループ長が16の倍数でない場合はサンプリングレート変換
Float64 inputSampleRate = fileDescription.mSampleRate;
Float64 outputSampleRate = fileDescription.mSampleRate * adjustment;
int outSamples = monoSamples;
if ( outputSampleRate == inputSampleRate ) {
m_pAudioData = new short[monoSamples];
for (int i=0; i<monoSamples; i++) {
m_pAudioData[i] = static_cast<short>(monoData[i] * 32768);
}
}
else {
outSamples = static_cast<int>(monoSamples / (inputSampleRate / outputSampleRate));
m_pAudioData = new short[outSamples];
resampling(monoData, monoSamples, inputSampleRate,
m_pAudioData, &outSamples, outputSampleRate);
}
// 後始末
delete [] monoData;
delete [] fileBuffer;
AudioConverterDispose(converter);
//Instデータの設定
if ( st_point > MAXIMUM_SAMPLES ) {
mInstData.lp = MAXIMUM_SAMPLES;
}
else {
mInstData.lp = st_point;
}
if ( end_point > MAXIMUM_SAMPLES ) {
mInstData.lp_end = MAXIMUM_SAMPLES;
}
else {
mInstData.lp_end = end_point;
}
mInstData.srcSamplerate = outputSampleRate;
mLoadedSamples = outSamples;
mIsLoaded = true;
return true;
#else
//Windowsのオーディオファイル読み込み処理
// ファイルを開く
HMMIO hmio = NULL;
MMRESULT err;
DWORD size;
hmio = mmioOpen( mPath, NULL, MMIO_READ );
if ( !hmio ) {
return false;
}
// RIFFチャンクを探す
MMCKINFO riffChunkInfo;
riffChunkInfo.fccType = mmioFOURCC('W', 'A', 'V', 'E');
err = mmioDescend( hmio, &riffChunkInfo, NULL, MMIO_FINDRIFF );
if ( err != MMSYSERR_NOERROR ) {
mmioClose( hmio, 0 );
return false;
}
if ( (riffChunkInfo.ckid != FOURCC_RIFF) || (riffChunkInfo.fccType != mmioFOURCC('W', 'A', 'V', 'E') ) ) {
mmioClose( hmio, 0 );
return false;
}
// フォーマットチャンクを探す
MMCKINFO formatChunkInfo;
formatChunkInfo.ckid = mmioFOURCC('f', 'm', 't', ' ');
err = mmioDescend( hmio, &formatChunkInfo, &riffChunkInfo, MMIO_FINDCHUNK );
if ( err != MMSYSERR_NOERROR ) {
mmioClose( hmio, 0 );
return false;
}
if ( formatChunkInfo.cksize < sizeof(PCMWAVEFORMAT) ) {
mmioClose( hmio, 0 );
return false;
}
//フォーマット情報を取得
WAVEFORMATEX pcmWaveFormat;
DWORD fmsize = (formatChunkInfo.cksize > sizeof(WAVEFORMATEX)) ? sizeof(WAVEFORMATEX):formatChunkInfo.cksize;
size = mmioRead( hmio, (HPSTR)&pcmWaveFormat, fmsize );
if ( size != fmsize ) {
mmioClose( hmio, 0 );
return false;
}
if ( pcmWaveFormat.wFormatTag != WAVE_FORMAT_PCM ) {
mmioClose( hmio, 0 );
return false;
}
mmioAscend(hmio, &formatChunkInfo, 0);
// Instrument情報を初期化
mInstData.basekey = 60;
mInstData.lowkey = 0;
mInstData.highkey = 127;
mInstData.loop = 0;
//smplチャンクを探す
MMCKINFO smplChunkInfo;
smplChunkInfo.ckid = mmioFOURCC('s', 'm', 'p', 'l');
err = mmioDescend( hmio, &smplChunkInfo, &riffChunkInfo, MMIO_FINDCHUNK );
if ( err != MMSYSERR_NOERROR ) {
smplChunkInfo.cksize = 0;
}
double st_point=0.0;
double end_point=0.0;
if ( smplChunkInfo.cksize >= sizeof(WAV_smpl) ) {
//ループポイントの取得
unsigned char *smplChunk = new unsigned char[smplChunkInfo.cksize];
size = mmioRead(hmio,(HPSTR)smplChunk, smplChunkInfo.cksize);
WAV_smpl *smpl = (WAV_smpl *)smplChunk;
if ( smpl->loops > 0 ) {
mInstData.loop = 1;
mInstData.basekey = smpl->note;
st_point = smpl->start;
end_point = smpl->end + 1; //SoundForge等では最終ポイントを含める解釈
}
else {
mInstData.basekey = smpl->note;
}
delete [] smplChunk;
}
mmioAscend(hmio, &formatChunkInfo, 0);
//dataチャンクを探す
MMCKINFO dataChunkInfo;
dataChunkInfo.ckid = mmioFOURCC('d', 'a', 't', 'a');
err = mmioDescend( hmio, &dataChunkInfo, &riffChunkInfo, MMIO_FINDCHUNK );
if( err != MMSYSERR_NOERROR ) {
mmioClose( hmio, 0 );
return false;
}
// 波形一時読み込み用メモリを確保
unsigned int dataSize = dataChunkInfo.cksize;
int bytesPerSample = pcmWaveFormat.nBlockAlign;
char *fileBuffer;
unsigned int fileBufferSize;
//容量制限
int dataSamples = dataSize / pcmWaveFormat.nBlockAlign;
if ( dataSamples > MAXIMUM_SAMPLES ) {
dataSize = MAXIMUM_SAMPLES * pcmWaveFormat.nBlockAlign;
}
if ( st_point > MAXIMUM_SAMPLES ) {
st_point = MAXIMUM_SAMPLES;
}
if ( end_point > MAXIMUM_SAMPLES ) {
end_point = MAXIMUM_SAMPLES;
}
if (mInstData.loop) {
fileBufferSize = dataSize+EXPAND_BUFFER*bytesPerSample;
}
else {
fileBufferSize = dataSize;
}
fileBuffer = new char[fileBufferSize];
memset(fileBuffer, 0, fileBufferSize);
// ファイルから波形データの読み込み
size = mmioRead(hmio, (HPSTR)fileBuffer, dataSize);
if ( size != dataSize ) {
mmioClose( hmio, 0 );
return false;
}
mmioClose(hmio,0);
//ループを展開する
double inputSampleRate = pcmWaveFormat.nSamplesPerSec;
double outputSampleRate = inputSampleRate;
if (mInstData.loop) {
unsigned int plusalpha=0;
double framestocopy;
while (plusalpha < EXPAND_BUFFER) {
framestocopy =
(end_point-st_point)>(EXPAND_BUFFER-plusalpha)?(EXPAND_BUFFER-plusalpha):end_point-st_point;
memcpy(fileBuffer+((int)end_point+plusalpha)*bytesPerSample,
fileBuffer+(int)st_point*bytesPerSample,
static_cast<size_t>(framestocopy*bytesPerSample));
plusalpha += static_cast<unsigned int>(framestocopy);
}
dataSize += plusalpha*bytesPerSample;
//16サンプル境界にFIXする
double adjustment = ( (long long)((end_point-st_point)/16) ) / ((end_point-st_point)/16.0);
outputSampleRate *= adjustment;
st_point *= adjustment;
end_point *= adjustment;
}
//一旦floatモノラルデータに変換
int bytesPerChannel = bytesPerSample / pcmWaveFormat.nChannels;
unsigned int inputPtr = 0;
unsigned int outputPtr = 0;
int monoSamples = dataSize / bytesPerSample;
float range = static_cast<float>((1<<(bytesPerChannel*8-1)) * pcmWaveFormat.nChannels);
float *monoData = new float[monoSamples];
while (inputPtr < dataSize) {
int frameSum = 0;
for (int ch=0; ch<pcmWaveFormat.nChannels; ch++) {
for (int i=0; i<bytesPerChannel; i++) {
if (i<bytesPerChannel-1) {
frameSum += (unsigned char)fileBuffer[inputPtr] << (8*i);
}
else {
frameSum += fileBuffer[inputPtr] << (8*i);
}
inputPtr++;
}
}
monoData[outputPtr] = frameSum / range;
outputPtr++;
}
//ループ長が16の倍数でない場合はサンプリングレート変換
int outSamples = monoSamples;
if ( outputSampleRate == inputSampleRate ) {
m_pAudioData = new short[monoSamples];
for (int i=0; i<monoSamples; i++) {
m_pAudioData[i] = static_cast<short>(monoData[i] * 32768);
}
}
else {
outSamples = static_cast<int>(monoSamples / (inputSampleRate / outputSampleRate));
m_pAudioData = new short[outSamples];
resampling(monoData, monoSamples, inputSampleRate,
m_pAudioData, &outSamples, outputSampleRate);
}
// 後始末
delete [] fileBuffer;
delete [] monoData;
//Instデータの設定
mInstData.lp = static_cast<int>(st_point);
mInstData.lp_end = static_cast<int>(end_point);
mInstData.srcSamplerate = outputSampleRate;
mLoadedSamples = outSamples;
mIsLoaded = true;
return true;
#endif
}
AACEncode::~AACEncode() {
AudioConverterDispose(m_audioConverter);
}
static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
{
AudioStreamBasicDescription requestedFormat; // The application requested format
AudioStreamBasicDescription hardwareFormat; // The hardware format
AudioStreamBasicDescription outputFormat; // The AudioUnit output format
AURenderCallbackStruct input;
ComponentDescription desc;
AudioDeviceID inputDevice;
UInt32 outputFrameCount;
UInt32 propertySize;
UInt32 enableIO;
Component comp;
ca_data *data;
OSStatus err;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_HALOutput;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
// Search for component with given description
comp = FindNextComponent(NULL, &desc);
if(comp == NULL)
{
ERR("FindNextComponent failed\n");
return ALC_INVALID_VALUE;
}
data = calloc(1, sizeof(*data));
device->ExtraData = data;
// Open the component
err = OpenAComponent(comp, &data->audioUnit);
if(err != noErr)
{
ERR("OpenAComponent failed\n");
goto error;
}
// Turn off AudioUnit output
enableIO = 0;
err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
goto error;
}
// Turn on AudioUnit input
enableIO = 1;
err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
goto error;
}
// Get the default input device
propertySize = sizeof(AudioDeviceID);
err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, &propertySize, &inputDevice);
if(err != noErr)
{
ERR("AudioHardwareGetProperty failed\n");
goto error;
}
if(inputDevice == kAudioDeviceUnknown)
{
ERR("No input device found\n");
goto error;
}
// Track the input device
err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
goto error;
}
// set capture callback
input.inputProc = ca_capture_callback;
input.inputProcRefCon = device;
err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
goto error;
}
// Initialize the device
err = AudioUnitInitialize(data->audioUnit);
if(err != noErr)
{
ERR("AudioUnitInitialize failed\n");
goto error;
}
// Get the hardware format
propertySize = sizeof(AudioStreamBasicDescription);
err = AudioUnitGetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize);
if(err != noErr || propertySize != sizeof(AudioStreamBasicDescription))
{
ERR("AudioUnitGetProperty failed\n");
goto error;
}
// Set up the requested format description
switch(device->FmtType)
{
case DevFmtUByte:
requestedFormat.mBitsPerChannel = 8;
requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked;
break;
case DevFmtShort:
requestedFormat.mBitsPerChannel = 16;
requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
break;
case DevFmtInt:
requestedFormat.mBitsPerChannel = 32;
requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
break;
case DevFmtFloat:
requestedFormat.mBitsPerChannel = 32;
requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked;
break;
case DevFmtByte:
case DevFmtUShort:
case DevFmtUInt:
ERR("%s samples not supported\n", DevFmtTypeString(device->FmtType));
goto error;
}
switch(device->FmtChans)
{
case DevFmtMono:
requestedFormat.mChannelsPerFrame = 1;
break;
case DevFmtStereo:
requestedFormat.mChannelsPerFrame = 2;
break;
case DevFmtQuad:
case DevFmtX51:
case DevFmtX51Side:
case DevFmtX61:
case DevFmtX71:
ERR("%s not supported\n", DevFmtChannelsString(device->FmtChans));
goto error;
}
requestedFormat.mBytesPerFrame = requestedFormat.mChannelsPerFrame * requestedFormat.mBitsPerChannel / 8;
requestedFormat.mBytesPerPacket = requestedFormat.mBytesPerFrame;
requestedFormat.mSampleRate = device->Frequency;
requestedFormat.mFormatID = kAudioFormatLinearPCM;
requestedFormat.mReserved = 0;
requestedFormat.mFramesPerPacket = 1;
// save requested format description for later use
data->format = requestedFormat;
data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
// Use intermediate format for sample rate conversion (outputFormat)
// Set sample rate to the same as hardware for resampling later
outputFormat = requestedFormat;
outputFormat.mSampleRate = hardwareFormat.mSampleRate;
// Determine sample rate ratio for resampling
data->sampleRateRatio = outputFormat.mSampleRate / device->Frequency;
// The output format should be the requested format, but using the hardware sample rate
// This is because the AudioUnit will automatically scale other properties, except for sample rate
err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
goto error;
}
// Set the AudioUnit output format frame count
outputFrameCount = device->UpdateSize * data->sampleRateRatio;
err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Output, 0, &outputFrameCount, sizeof(outputFrameCount));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed: %d\n", err);
goto error;
}
// Set up sample converter
err = AudioConverterNew(&outputFormat, &requestedFormat, &data->audioConverter);
if(err != noErr)
{
ERR("AudioConverterNew failed: %d\n", err);
goto error;
}
// Create a buffer for use in the resample callback
data->resampleBuffer = malloc(device->UpdateSize * data->frameSize * data->sampleRateRatio);
// Allocate buffer for the AudioUnit output
data->bufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * data->frameSize * data->sampleRateRatio);
if(data->bufferList == NULL)
goto error;
data->ring = CreateRingBuffer(data->frameSize, (device->UpdateSize * data->sampleRateRatio) * device->NumUpdates);
if(data->ring == NULL)
goto error;
al_string_copy_cstr(&device->DeviceName, deviceName);
return ALC_NO_ERROR;
error:
DestroyRingBuffer(data->ring);
free(data->resampleBuffer);
destroy_buffer_list(data->bufferList);
if(data->audioConverter)
AudioConverterDispose(data->audioConverter);
if(data->audioUnit)
CloseComponent(data->audioUnit);
free(data);
device->ExtraData = NULL;
return ALC_INVALID_VALUE;
}
AudioMixer::~AudioMixer()
{
for ( auto & it : m_converters ) {
AudioConverterDispose(it.second.converter);
}
}
/* This is called by audio file stream parser when it finds property values */
void Audio_Stream::propertyValueCallback(void *inClientData, AudioFileStreamID inAudioFileStream, AudioFileStreamPropertyID inPropertyID, UInt32 *ioFlags)
{
AS_TRACE("%s\n", __PRETTY_FUNCTION__);
Audio_Stream *THIS = static_cast<Audio_Stream*>(inClientData);
if (!THIS->m_audioStreamParserRunning) {
AS_TRACE("%s: stray callback detected!\n", __PRETTY_FUNCTION__);
return;
}
switch (inPropertyID) {
case kAudioFileStreamProperty_DataOffset: {
SInt64 offset;
UInt32 offsetSize = sizeof(offset);
OSStatus result = AudioFileStreamGetProperty(inAudioFileStream, kAudioFileStreamProperty_DataOffset, &offsetSize, &offset);
if (result == 0) {
THIS->m_dataOffset = offset;
} else {
AS_TRACE("%s: reading kAudioFileStreamProperty_DataOffset property failed\n", __PRETTY_FUNCTION__);
}
break;
}
case kAudioFileStreamProperty_AudioDataByteCount: {
UInt32 byteCountSize = sizeof(UInt64);
OSStatus err = AudioFileStreamGetProperty(inAudioFileStream,
kAudioFileStreamProperty_AudioDataByteCount,
&byteCountSize, &THIS->m_audioDataByteCount);
if (err) {
THIS->m_audioDataByteCount = 0;
}
break;
}
case kAudioFileStreamProperty_ReadyToProducePackets: {
memset(&(THIS->m_srcFormat), 0, sizeof m_srcFormat);
UInt32 asbdSize = sizeof(m_srcFormat);
OSStatus err = AudioFileStreamGetProperty(inAudioFileStream, kAudioFileStreamProperty_DataFormat, &asbdSize, &(THIS->m_srcFormat));
if (err) {
AS_TRACE("Unable to set the src format\n");
break;
}
THIS->m_packetDuration = THIS->m_srcFormat.mFramesPerPacket / THIS->m_srcFormat.mSampleRate;
AS_TRACE("srcFormat, bytes per packet %i\n", (unsigned int)THIS->m_srcFormat.mBytesPerPacket);
if (THIS->m_audioConverter) {
AudioConverterDispose(THIS->m_audioConverter);
}
err = AudioConverterNew(&(THIS->m_srcFormat),
&(THIS->m_dstFormat),
&(THIS->m_audioConverter));
if (err) {
AS_TRACE("Error in creating an audio converter, error %i\n", err);
THIS->m_initializationError = err;
}
THIS->setCookiesForStream(inAudioFileStream);
THIS->audioQueue()->handlePropertyChange(inAudioFileStream, inPropertyID, ioFlags);
break;
}
default: {
THIS->audioQueue()->handlePropertyChange(inAudioFileStream, inPropertyID, ioFlags);
break;
}
}
}
OSStatus DoConvertFile(CFURLRef sourceURL, CFURLRef destinationURL, OSType outputFormat, Float64 outputSampleRate, UInt32 outputBitRate)
{
AudioFileID sourceFileID = 0;
AudioFileID destinationFileID = 0;
AudioConverterRef converter = NULL;
Boolean canResumeFromInterruption = true; // we can continue unless told otherwise
CAStreamBasicDescription srcFormat, dstFormat;
AudioFileIO afio = {};
char *outputBuffer = NULL;
AudioStreamPacketDescription *outputPacketDescriptions = NULL;
OSStatus error = noErr;
// in this sample we should never be on the main thread here
assert(![NSThread isMainThread]);
// transition thread state to kStateRunning before continuing
printf("\nDoConvertFile\n");
try {
// get the source file
XThrowIfError(AudioFileOpenURL(sourceURL, kAudioFileReadPermission, 0, &sourceFileID), "AudioFileOpenURL failed");
// get the source data format
UInt32 size = sizeof(srcFormat);
XThrowIfError(AudioFileGetProperty(sourceFileID, kAudioFilePropertyDataFormat, &size, &srcFormat), "couldn't get source data format");
// setup the output file format
dstFormat.mSampleRate = (outputSampleRate == 0 ? srcFormat.mSampleRate : outputSampleRate); // set sample rate
if (outputFormat == kAudioFormatLinearPCM) {
// if the output format is PC create a 16-bit int PCM file format description as an example
dstFormat.mFormatID = outputFormat;
dstFormat.mChannelsPerFrame = srcFormat.NumberChannels();
dstFormat.mBitsPerChannel = 16;
dstFormat.mBytesPerPacket = dstFormat.mBytesPerFrame = 2 * dstFormat.mChannelsPerFrame;
dstFormat.mFramesPerPacket = 1;
dstFormat.mFormatFlags = kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsSignedInteger; // little-endian
} else {
// compressed format - need to set at least format, sample rate and channel fields for kAudioFormatProperty_FormatInfo
dstFormat.mFormatID = outputFormat;
dstFormat.mChannelsPerFrame = (outputFormat == kAudioFormatiLBC ? 1 : srcFormat.NumberChannels()); // for iLBC num channels must be 1
// use AudioFormat API to fill out the rest of the description
size = sizeof(dstFormat);
XThrowIfError(AudioFormatGetProperty(kAudioFormatProperty_FormatInfo, 0, NULL, &size, &dstFormat), "couldn't create destination data format");
}
printf("Source File format: "); srcFormat.Print();
printf("Destination format: "); dstFormat.Print();
// create the AudioConverter
XThrowIfError(AudioConverterNew(&srcFormat, &dstFormat, &converter), "AudioConverterNew failed!");
// if the source has a cookie, get it and set it on the Audio Converter
ReadCookie(sourceFileID, converter);
// get the actual formats back from the Audio Converter
size = sizeof(srcFormat);
XThrowIfError(AudioConverterGetProperty(converter, kAudioConverterCurrentInputStreamDescription, &size, &srcFormat), "AudioConverterGetProperty kAudioConverterCurrentInputStreamDescription failed!");
size = sizeof(dstFormat);
XThrowIfError(AudioConverterGetProperty(converter, kAudioConverterCurrentOutputStreamDescription, &size, &dstFormat), "AudioConverterGetProperty kAudioConverterCurrentOutputStreamDescription failed!");
printf("Formats returned from AudioConverter:\n");
printf(" Source format: "); srcFormat.Print();
printf(" Destination File format: "); dstFormat.Print();
// if encoding to AAC set the bitrate
// kAudioConverterEncodeBitRate is a UInt32 value containing the number of bits per second to aim for when encoding data
// when you explicitly set the bit rate and the sample rate, this tells the encoder to stick with both bit rate and sample rate
// but there are combinations (also depending on the number of channels) which will not be allowed
// if you do not explicitly set a bit rate the encoder will pick the correct value for you depending on samplerate and number of channels
// bit rate also scales with the number of channels, therefore one bit rate per sample rate can be used for mono cases
// and if you have stereo or more, you can multiply that number by the number of channels.
if (outputBitRate == 0) {
outputBitRate = 192000; // 192kbs
}
if (dstFormat.mFormatID == kAudioFormatMPEG4AAC) {
UInt32 propSize = sizeof(outputBitRate);
// set the bit rate depending on the samplerate chosen
XThrowIfError(AudioConverterSetProperty(converter, kAudioConverterEncodeBitRate, propSize, &outputBitRate),
"AudioConverterSetProperty kAudioConverterEncodeBitRate failed!");
// get it back and print it out
AudioConverterGetProperty(converter, kAudioConverterEncodeBitRate, &propSize, &outputBitRate);
printf ("AAC Encode Bitrate: %u\n", (unsigned int)outputBitRate);
}
// can the Audio Converter resume conversion after an interruption?
// this property may be queried at any time after construction of the Audio Converter after setting its output format
// there's no clear reason to prefer construction time, interruption time, or potential resumption time but we prefer
// construction time since it means less code to execute during or after interruption time
UInt32 canResume = 0;
size = sizeof(canResume);
error = AudioConverterGetProperty(converter, kAudioConverterPropertyCanResumeFromInterruption, &size, &canResume);
if (noErr == error) {
// we recieved a valid return value from the GetProperty call
// if the property's value is 1, then the codec CAN resume work following an interruption
// if the property's value is 0, then interruptions destroy the codec's state and we're done
if (0 == canResume) canResumeFromInterruption = false;
printf("Audio Converter %s continue after interruption!\n", (canResumeFromInterruption == 0 ? "CANNOT" : "CAN"));
} else {
// if the property is unimplemented (kAudioConverterErr_PropertyNotSupported, or paramErr returned in the case of PCM),
// then the codec being used is not a hardware codec so we're not concerned about codec state
// we are always going to be able to resume conversion after an interruption
if (kAudioConverterErr_PropertyNotSupported == error) {
printf("kAudioConverterPropertyCanResumeFromInterruption property not supported - see comments in source for more info.\n");
} else {
printf("AudioConverterGetProperty kAudioConverterPropertyCanResumeFromInterruption result %ld, paramErr is OK if PCM\n", error);
}
error = noErr;
}
// create the destination file
XThrowIfError(AudioFileCreateWithURL(destinationURL, kAudioFileCAFType, &dstFormat, kAudioFileFlags_EraseFile, &destinationFileID), "AudioFileCreateWithURL failed!");
// set up source buffers and data proc info struct
afio.srcFileID = sourceFileID;
afio.srcBufferSize = 32768;
afio.srcBuffer = new char [afio.srcBufferSize];
afio.srcFilePos = 0;
afio.srcFormat = srcFormat;
if (srcFormat.mBytesPerPacket == 0) {
// if the source format is VBR, we need to get the maximum packet size
// use kAudioFilePropertyPacketSizeUpperBound which returns the theoretical maximum packet size
// in the file (without actually scanning the whole file to find the largest packet,
// as may happen with kAudioFilePropertyMaximumPacketSize)
size = sizeof(afio.srcSizePerPacket);
XThrowIfError(AudioFileGetProperty(sourceFileID, kAudioFilePropertyPacketSizeUpperBound, &size, &afio.srcSizePerPacket), "AudioFileGetProperty kAudioFilePropertyPacketSizeUpperBound failed!");
// how many packets can we read for our buffer size?
afio.numPacketsPerRead = afio.srcBufferSize / afio.srcSizePerPacket;
// allocate memory for the PacketDescription structures describing the layout of each packet
afio.packetDescriptions = new AudioStreamPacketDescription [afio.numPacketsPerRead];
} else {
// CBR source format
afio.srcSizePerPacket = srcFormat.mBytesPerPacket;
afio.numPacketsPerRead = afio.srcBufferSize / afio.srcSizePerPacket;
afio.packetDescriptions = NULL;
}
// set up output buffers
UInt32 outputSizePerPacket = dstFormat.mBytesPerPacket; // this will be non-zero if the format is CBR
UInt32 theOutputBufSize = 32768;
outputBuffer = new char[theOutputBufSize];
if (outputSizePerPacket == 0) {
// if the destination format is VBR, we need to get max size per packet from the converter
size = sizeof(outputSizePerPacket);
XThrowIfError(AudioConverterGetProperty(converter, kAudioConverterPropertyMaximumOutputPacketSize, &size, &outputSizePerPacket), "AudioConverterGetProperty kAudioConverterPropertyMaximumOutputPacketSize failed!");
// allocate memory for the PacketDescription structures describing the layout of each packet
outputPacketDescriptions = new AudioStreamPacketDescription [theOutputBufSize / outputSizePerPacket];
}
UInt32 numOutputPackets = theOutputBufSize / outputSizePerPacket;
// if the destination format has a cookie, get it and set it on the output file
WriteCookie(converter, destinationFileID);
// write destination channel layout
if (srcFormat.mChannelsPerFrame > 2) {
WriteDestinationChannelLayout(converter, sourceFileID, destinationFileID);
}
UInt64 totalOutputFrames = 0; // used for debgging printf
SInt64 outputFilePos = 0;
// loop to convert data
printf("Converting...\n");
while (1) {
// set up output buffer list
AudioBufferList fillBufList;
fillBufList.mNumberBuffers = 1;
fillBufList.mBuffers[0].mNumberChannels = dstFormat.mChannelsPerFrame;
fillBufList.mBuffers[0].mDataByteSize = theOutputBufSize;
fillBufList.mBuffers[0].mData = outputBuffer;
// this will block if we're interrupted
Boolean wasInterrupted = NO;
if ((error || wasInterrupted) && (false == canResumeFromInterruption)) {
// this is our interruption termination condition
// an interruption has occured but the Audio Converter cannot continue
error = kMyAudioConverterErr_CannotResumeFromInterruptionError;
break;
}
// convert data
UInt32 ioOutputDataPackets = numOutputPackets;
printf("AudioConverterFillComplexBuffer...\n");
error = AudioConverterFillComplexBuffer(converter, EncoderDataProc, &afio, &ioOutputDataPackets, &fillBufList, outputPacketDescriptions);
// if interrupted in the process of the conversion call, we must handle the error appropriately
if (error) {
if (kAudioConverterErr_HardwareInUse == error) {
printf("Audio Converter returned kAudioConverterErr_HardwareInUse!\n");
} else {
XThrowIfError(error, "AudioConverterFillComplexBuffer error!");
}
} else {
if (ioOutputDataPackets == 0) {
// this is the EOF conditon
error = noErr;
break;
}
}
if (noErr == error) {
// write to output file
UInt32 inNumBytes = fillBufList.mBuffers[0].mDataByteSize;
XThrowIfError(AudioFileWritePackets(destinationFileID, false, inNumBytes, outputPacketDescriptions, outputFilePos, &ioOutputDataPackets, outputBuffer), "AudioFileWritePackets failed!");
printf("Convert Output: Write %lu packets at position %lld, size: %ld\n", ioOutputDataPackets, outputFilePos, inNumBytes);
// advance output file packet position
outputFilePos += ioOutputDataPackets;
if (dstFormat.mFramesPerPacket) {
// the format has constant frames per packet
totalOutputFrames += (ioOutputDataPackets * dstFormat.mFramesPerPacket);
} else if (outputPacketDescriptions != NULL) {
// variable frames per packet require doing this for each packet (adding up the number of sample frames of data in each packet)
for (UInt32 i = 0; i < ioOutputDataPackets; ++i)
totalOutputFrames += outputPacketDescriptions[i].mVariableFramesInPacket;
}
}
} // while
if (noErr == error) {
// write out any of the leading and trailing frames for compressed formats only
if (dstFormat.mBitsPerChannel == 0) {
// our output frame count should jive with
printf("Total number of output frames counted: %lld\n", totalOutputFrames);
WritePacketTableInfo(converter, destinationFileID);
}
// write the cookie again - sometimes codecs will update cookies at the end of a conversion
WriteCookie(converter, destinationFileID);
}
}
catch (CAXException e) {
char buf[256];
fprintf(stderr, "Error: %s (%s)\n", e.mOperation, e.FormatError(buf));
error = e.mError;
}
// cleanup
if (converter) AudioConverterDispose(converter);
if (destinationFileID) AudioFileClose(destinationFileID);
if (sourceFileID) AudioFileClose(sourceFileID);
if (afio.srcBuffer) delete [] afio.srcBuffer;
if (afio.packetDescriptions) delete [] afio.packetDescriptions;
if (outputBuffer) delete [] outputBuffer;
if (outputPacketDescriptions) delete [] outputPacketDescriptions;
return error;
}
bool FCoreAudioSoundSource::DetachFromAUGraph()
{
AURenderCallbackStruct Input;
Input.inputProc = NULL;
Input.inputProcRefCon = NULL;
SAFE_CA_CALL( AudioUnitSetProperty( SourceUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &Input, sizeof( Input ) ) );
if( StreamSplitterNode )
{
SAFE_CA_CALL( AUGraphDisconnectNodeInput( AudioDevice->GetAudioUnitGraph(), StreamSplitterNode, 0 ) );
}
if( ReverbNode )
{
SAFE_CA_CALL( AUGraphDisconnectNodeInput( AudioDevice->GetAudioUnitGraph(), ReverbNode, 0 ) );
}
if( RadioNode )
{
SAFE_CA_CALL( AUGraphDisconnectNodeInput( AudioDevice->GetAudioUnitGraph(), RadioNode, 0 ) );
}
if( EQNode )
{
SAFE_CA_CALL( AUGraphDisconnectNodeInput( AudioDevice->GetAudioUnitGraph(), EQNode, 0 ) );
}
if( StreamMergerNode )
{
SAFE_CA_CALL( AUGraphDisconnectNodeInput( AudioDevice->GetAudioUnitGraph(), StreamMergerNode, 0 ) );
}
if( AudioChannel )
{
if( Buffer->NumChannels < 3 )
{
SAFE_CA_CALL( AUGraphDisconnectNodeInput( AudioDevice->GetAudioUnitGraph(), AudioDevice->GetMixer3DNode(), MixerInputNumber ) );
AudioDevice->SetFreeMixer3DInput( MixerInputNumber );
}
else
{
SAFE_CA_CALL( AUGraphDisconnectNodeInput( AudioDevice->GetAudioUnitGraph(), AudioDevice->GetMatrixMixerNode(), MixerInputNumber ) );
AudioDevice->SetFreeMatrixMixerInput( MixerInputNumber );
}
}
if( StreamMergerNode )
{
SAFE_CA_CALL( AUGraphRemoveNode( AudioDevice->GetAudioUnitGraph(), StreamMergerNode ) );
}
if( EQNode )
{
SAFE_CA_CALL( AUGraphRemoveNode( AudioDevice->GetAudioUnitGraph(), EQNode ) );
}
if( RadioNode )
{
SAFE_CA_CALL( AUGraphRemoveNode( AudioDevice->GetAudioUnitGraph(), RadioNode ) );
}
if( ReverbNode )
{
SAFE_CA_CALL( AUGraphRemoveNode( AudioDevice->GetAudioUnitGraph(), ReverbNode ) );
}
if( StreamSplitterNode )
{
SAFE_CA_CALL( AUGraphRemoveNode( AudioDevice->GetAudioUnitGraph(), StreamSplitterNode ) );
}
if( AudioChannel )
{
SAFE_CA_CALL( AUGraphRemoveNode( AudioDevice->GetAudioUnitGraph(), SourceNode ) );
}
SAFE_CA_CALL( AUGraphUpdate( AudioDevice->GetAudioUnitGraph(), NULL ) );
AudioConverterDispose( CoreAudioConverter );
CoreAudioConverter = NULL;
StreamMergerNode = 0;
StreamMergerUnit = NULL;
EQNode = 0;
EQUnit = NULL;
RadioNode = 0;
RadioUnit = NULL;
ReverbNode = 0;
ReverbUnit = NULL;
StreamSplitterNode = 0;
StreamSplitterUnit = NULL;
SourceNode = 0;
SourceUnit = NULL;
MixerInputNumber = -1;
GAudioChannels[AudioChannel] = NULL;
AudioChannel = 0;
return true;
}
static GVBool gviHardwareInitPlayback(GVIDevice * device)
{
GVIHardwareData * data = (GVIHardwareData *)device->m_data;
UInt32 size;
OSStatus result;
UInt32 primeMethod;
SInt32 channelMap[100];
int i;
// create the array of sources
data->m_playbackSources = gviNewSourceList();
if(!data->m_playbackSources)
return GVFalse;
// get the playback format
size = sizeof(AudioStreamBasicDescription);
result = AudioDeviceGetProperty(device->m_deviceID, 0, false, kAudioDevicePropertyStreamFormat, &size, &data->m_playbackStreamDescriptor);
if(result != noErr)
{
gviFreeSourceList(data->m_playbackSources);
return GVFalse;
}
// create a converter from the GV format to the playback format
result = AudioConverterNew(&GVIVoiceFormat, &data->m_playbackStreamDescriptor, &data->m_playbackConverter);
if(result != noErr)
{
gviFreeSourceList(data->m_playbackSources);
return GVFalse;
}
// set it to do no priming
primeMethod = kConverterPrimeMethod_None;
result = AudioConverterSetProperty(data->m_playbackConverter, kAudioConverterPrimeMethod, sizeof(UInt32), &primeMethod);
if(result != noErr)
{
AudioConverterDispose(data->m_playbackConverter);
gviFreeSourceList(data->m_playbackSources);
return GVFalse;
}
// setup the converter to map the input channel to all output channels
result = AudioConverterGetPropertyInfo(data->m_playbackConverter, kAudioConverterChannelMap, &size, NULL);
if(result == noErr)
{
result = AudioConverterGetProperty(data->m_playbackConverter, kAudioConverterChannelMap, &size, channelMap);
if(result == noErr)
{
for(i = 0 ; i < (size / sizeof(SInt32)) ; i++)
channelMap[i] = 0;
AudioConverterSetProperty(data->m_playbackConverter, kAudioConverterChannelMap, size, channelMap);
}
}
// allocate the playback buffer
data->m_playbackBuffer = (GVSample *)gsimalloc(GVIBytesPerFrame);
if(!data->m_playbackBuffer)
{
AudioConverterDispose(data->m_playbackConverter);
gviFreeSourceList(data->m_playbackSources);
return GVFalse;
}
// add property listener
AudioDeviceAddPropertyListener(device->m_deviceID, 0, false, kAudioDevicePropertyDeviceIsAlive, gviPropertyListener, device);
#if GVI_VOLUME_IN_SOFTWARE
// init volume
data->m_playbackVolume = (GVScalar)1.0;
#endif
return GVTrue;
}