OSStatus writeRenderProc(void *inRefCon,
AudioUnitRenderActionFlags *inActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumFrames,
AudioBufferList *ioData)
{
OSStatus err= noErr;
void *inInputDataProcUserData=NULL;
CAData *d=(CAData*)inRefCon;
if (gain_changed_out == true)
{
err = AudioUnitSetParameter(d->caOutAudioUnit, kAudioUnitParameterUnit_LinearGain,
kAudioUnitScope_Global, 0, (Float32)gain_volume_out, 0);
if(err != noErr)
{
ms_error("failed to set output volume %i", err);
}
gain_changed_out = false;
err= noErr;
}
if(d->write_started != FALSE) {
AudioStreamPacketDescription* outPacketDescription = NULL;
err = AudioConverterFillComplexBuffer(d->caOutConverter, writeACInputProc, inRefCon,
&inNumFrames, ioData, outPacketDescription);
if(err != noErr)
ms_error("writeRenderProc:AudioConverterFillComplexBuffer err %08x %d", err, ioData->mNumberBuffers);
}
return err;
}
static ALCenum ca_capture_samples(ALCdevice *device, ALCvoid *buffer, ALCuint samples)
{
ca_data *data = (ca_data*)device->ExtraData;
AudioBufferList *list;
UInt32 frameCount;
OSStatus err;
// If no samples are requested, just return
if(samples == 0)
return ALC_NO_ERROR;
// Allocate a temporary AudioBufferList to use as the return resamples data
list = alloca(sizeof(AudioBufferList) + sizeof(AudioBuffer));
// Point the resampling buffer to the capture buffer
list->mNumberBuffers = 1;
list->mBuffers[0].mNumberChannels = data->format.mChannelsPerFrame;
list->mBuffers[0].mDataByteSize = samples * data->frameSize;
list->mBuffers[0].mData = buffer;
// Resample into another AudioBufferList
frameCount = samples;
err = AudioConverterFillComplexBuffer(data->audioConverter, ca_capture_conversion_callback,
device, &frameCount, list, NULL);
if(err != noErr)
{
ERR("AudioConverterFillComplexBuffer error: %d\n", err);
return ALC_INVALID_VALUE;
}
return ALC_NO_ERROR;
}
uint32_t CoreAudioEncoder::encodeChunk(UInt32 npackets)
{
prepareOutputBuffer(npackets);
AudioBufferList *abl = m_output_abl.get();
AudioStreamPacketDescription *aspd = &m_packet_desc[0];
CHECKCA(AudioConverterFillComplexBuffer(m_converter, staticInputDataProc,
this, &npackets, abl, aspd));
if (samplesRead() == 0)
return false;
if (npackets == 0 && abl->mBuffers[0].mDataByteSize == 0)
return 0;
if (!m_requires_packet_desc) {
writeSamples(abl->mBuffers[0].mData,
abl->mBuffers[0].mDataByteSize, npackets);
} else {
for (uint32_t i = 0; i < npackets; ++i) {
if (aspd[i].mVariableFramesInPacket) m_variable_packet_len = true;
uint32_t nsamples =
m_variable_packet_len ? aspd[i].mVariableFramesInPacket
: m_output_desc.mFramesPerPacket;
if (nsamples) {
uint8_t *p = static_cast<uint8_t*>(abl->mBuffers[0].mData);
writeSamples(p + aspd[i].mStartOffset,
aspd[i].mDataByteSize, nsamples);
}
}
}
return npackets;
}
static OSStatus gviHardwarePlaybackIOProc(AudioDeviceID inDevice,
const AudioTimeStamp * inNow,
const AudioBufferList * inInputData,
const AudioTimeStamp * inInputTime,
AudioBufferList * outOutputData,
const AudioTimeStamp * inOutputTime,
void * inClientData)
{
GVIDevice * device = (GVIDevice *)inClientData;
GVIHardwareData * data = (GVIHardwareData *)device->m_data;
AudioBufferList bufferList;
OSStatus result;
UInt32 size;
// get a lock on the device
if(!gviLockDevice(data))
return (OSStatus)1;
// calculate the number of samples the proc wants
size = (outOutputData->mBuffers[0].mDataByteSize / data->m_playbackStreamDescriptor.mBytesPerFrame);
// setup our own buffer list, pointing at the channel (buffer) we want
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0] = outOutputData->mBuffers[data->m_playbackChannel];
// fill the buffer using the callback
result = AudioConverterFillComplexBuffer(data->m_playbackConverter, gviAudioConverterPlaybackProc, device, &size, &bufferList, NULL);
if(result != noErr)
return (OSStatus)1;
// release the device lock
gviUnlockDevice(data);
return noErr;
}
OSStatus
otMacAudioInputStream::InputReadyCb(void *userdata, AudioUnitRenderActionFlags *actionFlags,
const AudioTimeStamp *timeStamp, UInt32 busNumber,
UInt32 numberFrames, AudioBufferList *data)
{
otMacAudioInputStream *_this = (otMacAudioInputStream*)userdata;
OSStatus err;
DEBUG_DUMP2("InputReadyCb numberFrames = %d, frameEnd = %d", numberFrames, _this->mFrameEnd);
err = AudioUnitRender(_this->mAudioUnit, actionFlags, timeStamp, busNumber,
numberFrames, _this->mBuffer);
if (err == noErr) {
UInt32 bytes = _this->mBuffer->mBuffers[0].mDataByteSize;
char *data = (char*) _this->mBuffer->mBuffers[0].mData;
while (bytes > 0) {
PRUint32 len = PR_MIN(_this->mInputFrameSize - _this->mFrameEnd, bytes);
memcpy(_this->mFrame + _this->mFrameEnd, data, len);
DEBUG_DUMP_N(("InputReadyCbInt bytes = %d frameEnd = %d, inputFrameSize = %d, frameSize = %d", bytes, _this->mFrameEnd, _this->mInputFrameSize, _this->mFrameSize));
data += len;
bytes -= len;
_this->mFrameEnd += len;
if (_this->mFrameEnd < _this->mInputFrameSize)
break;
UInt32 frameSize = _this->mFrameSize/2;
err = AudioConverterFillComplexBuffer(_this->mConverter,
&ConverterCb, _this, &frameSize,
_this->mConvertBuffer, NULL);
if (err != noErr) {
printErrCode(err);
return err;
}
DEBUG_DUMP("SendFrame");
char *data2 = (char*)_this->mConvertBuffer->mBuffers[0].mData;
_this->mFilter->InputData(data2, _this->mFrameSize);
if (_this->mTarget)
_this->mTarget->AcceptData(data2, _this->mFrameSize);
//_this->mFrameEnd = 0;
}
} else {
printErrCode(err);
}
return noErr;
}
UInt32 SFB::Audio::Converter::ConvertAudio(AudioBufferList *bufferList, UInt32 frameCount)
{
if(!IsOpen() || nullptr == bufferList || 0 == frameCount)
return 0;
OSStatus result = AudioConverterFillComplexBuffer(mConverter, myAudioConverterComplexInputDataProc, mConverterState.get(), &frameCount, bufferList, nullptr);
if(noErr != result)
return 0;
return frameCount;
}
static OSStatus convertProc(void *inRefCon, AudioUnitRenderActionFlags *inActionFlags,
const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber,
UInt32 inNumFrames, AudioBufferList *ioData)
{
AudioStreamPacketDescription* outPacketDescription = NULL;
mpg123_coreaudio_t* ca = (mpg123_coreaudio_t*)inRefCon;
OSStatus err= noErr;
err = AudioConverterFillComplexBuffer(ca->converter, playProc, inRefCon, &inNumFrames, ioData, outPacketDescription);
return err;
}
void CocoaCaConverter::loadData( BufferList *ioData )
{
std::shared_ptr<AudioBufferList> nativeBufferList = createCaBufferList( ioData );
UInt32 aSampleCount = ioData->mBuffers[0].mSampleCount;
AudioStreamPacketDescription * outputPacketDescriptions = new AudioStreamPacketDescription[aSampleCount];
OSStatus err = AudioConverterFillComplexBuffer( mConverter, CocoaCaConverter::dataInputCallback, (void *)this, &aSampleCount, nativeBufferList.get(), outputPacketDescriptions );
delete [] outputPacketDescriptions;
if( err ) {
//throw
}
fillBufferListFromCaBufferList( ioData, nativeBufferList.get(), (uint32_t)aSampleCount );
}
// _______________________________________________________________________________________
//
void CAAudioFile::WritePacketsFromCallback(
AudioConverterComplexInputDataProc inInputDataProc,
void * inInputDataProcUserData)
{
while (true) {
// keep writing until we exhaust the input (temporary stop), or produce no output (EOF)
UInt32 numEncodedPackets = mIOBufferSizePackets;
mIOBufferList.mBuffers[0].mDataByteSize = mIOBufferSizeBytes;
#if CAAUDIOFILE_PROFILE
mInConverter = true;
#endif
StartTiming(this, fill);
OSStatus err = AudioConverterFillComplexBuffer(mConverter, inInputDataProc, inInputDataProcUserData,
&numEncodedPackets, &mIOBufferList, mPacketDescs);
ElapsedTime(this, fill, mTicksInConverter);
#if CAAUDIOFILE_PROFILE
mInConverter = false;
#endif
XThrowIf(err != 0 && err != kNoMoreInputRightNow, err, "convert audio packets (write)");
if (numEncodedPackets == 0)
break;
Byte *buf = (Byte *)mIOBufferList.mBuffers[0].mData;
#if VERBOSE_IO
printf("CAAudioFile::WritePacketsFromCallback: wrote %ld packets, %ld bytes\n", numEncodedPackets, mIOBufferList.mBuffers[0].mDataByteSize);
if (mPacketDescs) {
for (UInt32 i = 0; i < numEncodedPackets; ++i) {
printf(" write packet %qd : offset %qd, length %ld\n", mPacketMark + i, mPacketDescs[i].mStartOffset, mPacketDescs[i].mDataByteSize);
#if VERBOSE_IO >= 2
hexdump(buf + mPacketDescs[i].mStartOffset, mPacketDescs[i].mDataByteSize);
#endif
}
}
#endif
StartTiming(this, write);
XThrowIfError(AudioFileWritePackets(mAudioFile, mUseCache, mIOBufferList.mBuffers[0].mDataByteSize, mPacketDescs, mPacketMark, &numEncodedPackets, buf), "write audio file");
ElapsedTime(this, write, mTicksInIO);
mPacketMark += numEncodedPackets;
//mNumberPackets += numEncodedPackets;
if (mFileDataFormat.mFramesPerPacket > 0)
mFrameMark += numEncodedPackets * mFileDataFormat.mFramesPerPacket;
else {
for (UInt32 i = 0; i < numEncodedPackets; ++i)
mFrameMark += mPacketDescs[i].mVariableFramesInPacket;
}
if (err == kNoMoreInputRightNow)
break;
}
}
void
AACEncode::pushBuffer(const uint8_t* const data, size_t size, IMetadata& metadata)
{
const size_t sampleCount = size / m_bytesPerSample;
const size_t aac_packet_count = sampleCount / kSamplesPerFrame;
const size_t required_bytes = aac_packet_count * m_outputPacketMaxSize;
if(m_outputBuffer.total() < (required_bytes)) {
m_outputBuffer.resize(required_bytes);
}
uint8_t* p = m_outputBuffer();
uint8_t* p_out = (uint8_t*)data;
for ( size_t i = 0 ; i < aac_packet_count ; ++i ) {
UInt32 num_packets = 1;
AudioBufferList l;
l.mNumberBuffers=1;
l.mBuffers[0].mDataByteSize = m_outputPacketMaxSize * num_packets;
l.mBuffers[0].mData = p;
std::unique_ptr<UserData> ud(new UserData());
ud->size = static_cast<int>(kSamplesPerFrame * m_bytesPerSample);
ud->data = const_cast<uint8_t*>(p_out);
ud->packetSize = static_cast<int>(m_bytesPerSample);
AudioStreamPacketDescription output_packet_desc[num_packets];
m_converterMutex.lock();
AudioConverterFillComplexBuffer(m_audioConverter, AACEncode::ioProc, ud.get(), &num_packets, &l, output_packet_desc);
m_converterMutex.unlock();
p += output_packet_desc[0].mDataByteSize;
p_out += kSamplesPerFrame * m_bytesPerSample;
}
const size_t totalBytes = p - m_outputBuffer();
auto output = m_output.lock();
if(output && totalBytes) {
if(!m_sentConfig) {
output->pushBuffer((const uint8_t*)m_asc, sizeof(m_asc), metadata);
m_sentConfig = true;
}
output->pushBuffer(m_outputBuffer(), totalBytes, metadata);
}
}
void AudioFile::read(Float32 *data, UInt64 *cursor, UInt32 *numFrames)
{
AudioFramePacketTranslation t;
UInt32 size = sizeof(AudioFramePacketTranslation);
t.mFrame = *cursor;
AudioFileGetProperty(mAudioFileID, kAudioFilePropertyFrameToPacket, &size, &t);
*mCursor = t.mPacket;
AudioFramePacketTranslation t2;
t2.mFrame = *numFrames;
AudioFileGetProperty(mAudioFileID, kAudioFilePropertyFrameToPacket, &size, &t2);
UInt32 numPacketsToRead = t2.mPacket ? t2.mPacket : 1;
AudioBytePacketTranslation t3;
t3.mPacket = numPacketsToRead;
size = sizeof(AudioBytePacketTranslation);
AudioFileGetProperty(mAudioFileID, kAudioFilePropertyPacketToByte, &size, &t3);
if (mConverterBuffer) free(mConverterBuffer);
mConverterBuffer = (char*)malloc(t3.mByte);
mNumPacketsToRead = numPacketsToRead;
UInt32 outNumBytes;
checkError(AudioFileReadPackets(mAudioFileID, false, &outNumBytes, mPacketDescs, *mCursor, &numPacketsToRead, mConverterBuffer), "AudioFileReadPackets");
mConvertByteSize = outNumBytes;
UInt32 numFramesToConvert = t.mFrameOffsetInPacket + *numFrames;
bool interleaved = true;
interleaved = !(mClientFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved);
AudioBufferList* tmpbuf = AudioSourceNode::createAudioBufferList(2, interleaved, numFramesToConvert, sizeof(Float32));
checkError(AudioConverterFillComplexBuffer(mAudioConverterRef, encoderProc, this, &numFramesToConvert, tmpbuf, NULL),
"AudioConverterFillComplexBuffer");
if (interleaved) {
Float32* sample = (Float32*)tmpbuf->mBuffers[0].mData;
memcpy(data, &sample[t.mFrameOffsetInPacket], numFramesToConvert * sizeof(Float32) * mClientFormat.mChannelsPerFrame);
}
AudioSourceNode::deleteAudioBufferList(tmpbuf);
if (numFramesToConvert == 0) {
AudioConverterReset(mAudioConverterRef);
}
*numFrames = numFramesToConvert;
}
OSStatus MT32Synth::Render(AudioUnitRenderActionFlags &ioActionFlags, const AudioTimeStamp &inTimeStamp, UInt32 inNumberFrames) {
if(!synth) {
return noErr;
}
AUOutputElement* outputBus = GetOutput(0);
outputBus->PrepareBuffer(inNumberFrames);
AudioBufferList& outputBufList = outputBus->GetBufferList();
AUBufferList::ZeroBuffer(outputBufList);
UInt32 ioOutputDataPackets = inNumberFrames * destFormat.mFramesPerPacket;
AudioConverterFillComplexBuffer(audioConverterRef, EncoderDataProc, (void*) this, &ioOutputDataPackets, &outputBufList, NULL);
return noErr;
}
void AudioFile::read(AudioBufferList* buf, UInt64* cursor, UInt32* numFrames)
{
AudioFramePacketTranslation t;
UInt32 size = sizeof(AudioFramePacketTranslation);
t.mFrame = *cursor;
AudioFileGetProperty(mAudioFileID, kAudioFilePropertyFrameToPacket, &size, &t);
*mCursor = t.mPacket;
UInt32 numFramesToRead = *numFrames;//t.mFrameOffsetInPacket + *numFrames;
AudioBufferList* tmpbuf = AudioSourceNode::createAudioBufferList(2, false, numFramesToRead, sizeof(Float32));
checkError(AudioConverterFillComplexBuffer(mAudioConverterRef, encoderProc, this, &numFramesToRead, tmpbuf, NULL),
"AudioConverterFillComplexBuffer");
memcpy(buf->mBuffers[0].mData, tmpbuf->mBuffers[0].mData, *numFrames * sizeof(Float32));
memcpy(buf->mBuffers[1].mData, tmpbuf->mBuffers[1].mData, *numFrames * sizeof(Float32));
AudioSourceNode::deleteAudioBufferList(tmpbuf);
if (numFramesToRead == 0) {
AudioConverterReset(mAudioConverterRef);
}
*numFrames = numFramesToRead;
}
OSStatus AudioConverterConvertComplexBuffer(AudioConverterRef inAudioConverter, UInt32 inNumberPCMFrames, const AudioBufferList *inInputData, AudioBufferList *outOutputData)
{
OSStatus status;
UInt32 dataPacketSize;
int64_t totalOutBytes = 0;
AudioConverterComplexInputDataProc proc = [](AudioConverterRef audioConverter, UInt32* numberDataPackets, AudioBufferList* data, AudioStreamPacketDescription** dataPacketDescription, void* userData) -> OSStatus
{
const AudioBufferList *inInputData = (AudioBufferList*) inInputData;
if (dataPacketDescription)
*dataPacketDescription = nullptr;
return unimpErr;
};
for (UInt32 i = 0; i < outOutputData->mNumberBuffers; i++)
totalOutBytes += outOutputData->mBuffers[i].mDataByteSize;
dataPacketSize = totalOutBytes / inAudioConverter->frameSize();
status = AudioConverterFillComplexBuffer(inAudioConverter, proc, (void*) inInputData, &dataPacketSize, outOutputData, nullptr);
return status;
}
}
hb_buffer_t *obuf;
AudioStreamPacketDescription odesc = { 0 };
AudioBufferList obuflist =
{
.mNumberBuffers = 1,
.mBuffers = { { .mNumberChannels = pv->nchannels } },
};
obuf = hb_buffer_init(pv->omaxpacket);
obuflist.mBuffers[0].mDataByteSize = obuf->size;
obuflist.mBuffers[0].mData = obuf->data;
OSStatus err = AudioConverterFillComplexBuffer(pv->converter,
inInputDataProc, pv,
&npackets, &obuflist, &odesc);
if (err != noErr && err != 1)
{
hb_log("encCoreAudio: unexpected error in AudioConverterFillComplexBuffer()");
}
// only drop the output buffer if it's actually empty
if (!npackets || odesc.mDataByteSize <= 0)
{
hb_log("encCoreAudio: 0 packets returned");
return NULL;
}
obuf->size = odesc.mDataByteSize;
obuf->s.start = 90000LL * pv->samples / pv->osamplerate;
MediaResult
AppleATDecoder::DecodeSample(MediaRawData* aSample)
{
MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
// Array containing the queued decoded audio frames, about to be output.
nsTArray<AudioDataValue> outputData;
UInt32 channels = mOutputFormat.mChannelsPerFrame;
// Pick a multiple of the frame size close to a power of two
// for efficient allocation.
const uint32_t MAX_AUDIO_FRAMES = 128;
const uint32_t maxDecodedSamples = MAX_AUDIO_FRAMES * channels;
// Descriptions for _decompressed_ audio packets. ignored.
auto packets = MakeUnique<AudioStreamPacketDescription[]>(MAX_AUDIO_FRAMES);
// This API insists on having packets spoon-fed to it from a callback.
// This structure exists only to pass our state.
PassthroughUserData userData =
{ channels, (UInt32)aSample->Size(), aSample->Data() };
// Decompressed audio buffer
AlignedAudioBuffer decoded(maxDecodedSamples);
if (!decoded) {
return NS_ERROR_OUT_OF_MEMORY;
}
do {
AudioBufferList decBuffer;
decBuffer.mNumberBuffers = 1;
decBuffer.mBuffers[0].mNumberChannels = channels;
decBuffer.mBuffers[0].mDataByteSize =
maxDecodedSamples * sizeof(AudioDataValue);
decBuffer.mBuffers[0].mData = decoded.get();
// in: the max number of packets we can handle from the decoder.
// out: the number of packets the decoder is actually returning.
UInt32 numFrames = MAX_AUDIO_FRAMES;
OSStatus rv = AudioConverterFillComplexBuffer(mConverter,
_PassthroughInputDataCallback,
&userData,
&numFrames /* in/out */,
&decBuffer,
packets.get());
if (rv && rv != kNoMoreDataErr) {
LOG("Error decoding audio sample: %d\n", rv);
return MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
RESULT_DETAIL("Error decoding audio sample: %d @ %lld",
rv, aSample->mTime));
}
if (numFrames) {
outputData.AppendElements(decoded.get(), numFrames * channels);
}
if (rv == kNoMoreDataErr) {
break;
}
} while (true);
if (outputData.IsEmpty()) {
return NS_OK;
}
size_t numFrames = outputData.Length() / channels;
int rate = mOutputFormat.mSampleRate;
media::TimeUnit duration = FramesToTimeUnit(numFrames, rate);
if (!duration.IsValid()) {
NS_WARNING("Invalid count of accumulated audio samples");
return MediaResult(
NS_ERROR_DOM_MEDIA_OVERFLOW_ERR,
RESULT_DETAIL(
"Invalid count of accumulated audio samples: num:%llu rate:%d",
uint64_t(numFrames), rate));
}
#ifdef LOG_SAMPLE_DECODE
LOG("pushed audio at time %lfs; duration %lfs\n",
(double)aSample->mTime / USECS_PER_S,
duration.ToSeconds());
#endif
AudioSampleBuffer data(outputData.Elements(), outputData.Length());
if (!data.Data()) {
return NS_ERROR_OUT_OF_MEMORY;
}
if (mChannelLayout && !mAudioConverter) {
AudioConfig in(*mChannelLayout.get(), rate);
AudioConfig out(channels, rate);
if (!in.IsValid() || !out.IsValid()) {
return MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
RESULT_DETAIL("Invalid audio config"));
}
mAudioConverter = MakeUnique<AudioConverter>(in, out);
}
if (mAudioConverter) {
MOZ_ASSERT(mAudioConverter->CanWorkInPlace());
data = mAudioConverter->Process(Move(data));
}
RefPtr<AudioData> audio = new AudioData(aSample->mOffset,
aSample->mTime,
duration.ToMicroseconds(),
numFrames,
data.Forget(),
channels,
rate);
mCallback->Output(audio);
return NS_OK;
}
void
AppleATDecoder::SampleCallback(uint32_t aNumBytes,
uint32_t aNumPackets,
const void* aData,
AudioStreamPacketDescription* aPackets)
{
// Pick a multiple of the frame size close to a power of two
// for efficient allocation.
const uint32_t MAX_AUDIO_FRAMES = 128;
const uint32_t decodedSize = MAX_AUDIO_FRAMES * mConfig.channel_count *
sizeof(AudioDataValue);
// Descriptions for _decompressed_ audio packets. ignored.
nsAutoArrayPtr<AudioStreamPacketDescription>
packets(new AudioStreamPacketDescription[MAX_AUDIO_FRAMES]);
// This API insists on having packets spoon-fed to it from a callback.
// This structure exists only to pass our state and the result of the
// parser on to the callback above.
PassthroughUserData userData =
{ this, aNumPackets, aNumBytes, aData, aPackets, false };
do {
// Decompressed audio buffer
nsAutoArrayPtr<uint8_t> decoded(new uint8_t[decodedSize]);
AudioBufferList decBuffer;
decBuffer.mNumberBuffers = 1;
decBuffer.mBuffers[0].mNumberChannels = mOutputFormat.mChannelsPerFrame;
decBuffer.mBuffers[0].mDataByteSize = decodedSize;
decBuffer.mBuffers[0].mData = decoded.get();
// in: the max number of packets we can handle from the decoder.
// out: the number of packets the decoder is actually returning.
UInt32 numFrames = MAX_AUDIO_FRAMES;
OSStatus rv = AudioConverterFillComplexBuffer(mConverter,
_PassthroughInputDataCallback,
&userData,
&numFrames /* in/out */,
&decBuffer,
packets.get());
if (rv && rv != kNeedMoreData) {
LOG("Error decoding audio stream: %#x\n", rv);
mCallback->Error();
break;
}
LOG("%d frames decoded", numFrames);
// If we decoded zero frames then AudioConverterFillComplexBuffer is out
// of data to provide. We drained its internal buffer completely on the
// last pass.
if (numFrames == 0 && rv == kNeedMoreData) {
LOG("FillComplexBuffer out of data exactly\n");
mCallback->InputExhausted();
break;
}
const int rate = mOutputFormat.mSampleRate;
const int channels = mOutputFormat.mChannelsPerFrame;
int64_t time = mCurrentAudioTimestamp;
int64_t duration = FramesToUsecs(numFrames, rate).value();
LOG("pushed audio at time %lfs; duration %lfs\n",
(double)time / USECS_PER_S, (double)duration / USECS_PER_S);
AudioData* audio = new AudioData(mSamplePosition,
time, duration, numFrames,
reinterpret_cast<AudioDataValue*>(decoded.forget()),
channels, rate);
mCallback->Output(audio);
mHaveOutput = true;
if (rv == kNeedMoreData) {
// No error; we just need more data.
LOG("FillComplexBuffer out of data\n");
mCallback->InputExhausted();
break;
}
} while (true);
}
OSStatus readRenderProc(void *inRefCon,
AudioUnitRenderActionFlags *inActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumFrames,
AudioBufferList *ioData)
{
CAData *d=(CAData*)inRefCon;
OSStatus err = noErr;
err = AudioUnitRender(d->caInAudioUnit, inActionFlags, inTimeStamp, inBusNumber,
inNumFrames, d->fAudioBuffer);
if(err != noErr)
{
ms_error("AudioUnitRender %d size = %d", err, d->fAudioBuffer->mBuffers[0].mDataByteSize);
return err;
}
UInt32 AvailableOutputBytes = inNumFrames * sizeof (float) * d->caInASBD.mChannelsPerFrame;
UInt32 propertySize = sizeof (AvailableOutputBytes);
err = AudioConverterGetProperty (d->caInConverter,
kAudioConverterPropertyCalculateOutputBufferSize,
&propertySize,
&AvailableOutputBytes);
if(err != noErr)
{
ms_error("AudioConverterGetProperty kAudioConverterPropertyCalculateOutputBufferSize %d", err);
return err;
}
if (AvailableOutputBytes>d->fMSBuffer->mBuffers[0].mDataByteSize)
{
DestroyAudioBufferList(d->fMSBuffer);
d->fMSBuffer = AllocateAudioBufferList(d->stereo ? 2 : 1,
AvailableOutputBytes);
}
UInt32 ActualOutputFrames = AvailableOutputBytes / ((d->bits / 8) * 1) / d->caInASBD.mChannelsPerFrame;
err = AudioConverterFillComplexBuffer (d->caInConverter,
(AudioConverterComplexInputDataProc)(readACInputProc),
inRefCon,
&ActualOutputFrames,
d->fMSBuffer,
NULL);
if(err != noErr)
{
ms_error("readRenderProc:AudioConverterFillComplexBuffer %d", err);
return err;
}
mblk_t *rm=NULL;
rm=allocb(ActualOutputFrames*2,0);
memcpy(rm->b_wptr, d->fMSBuffer->mBuffers[0].mData, ActualOutputFrames*2);
rm->b_wptr+=ActualOutputFrames*2;
if (gain_volume_in != 1.0f)
{
int16_t *ptr=(int16_t *)rm->b_rptr;
for (;ptr<(int16_t *)rm->b_wptr;ptr++)
{
*ptr=(int16_t)(((float)(*ptr))*gain_volume_in);
}
}
ms_mutex_lock(&d->mutex);
putq(&d->rq,rm);
ms_mutex_unlock(&d->mutex);
rm=NULL;
return err;
}
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;
}
std::shared_ptr<Buffer>
AudioMixer::resample(const uint8_t* const buffer,
size_t size,
AudioBufferMetadata& metadata)
{
const auto inFrequncyInHz = metadata.getData<kAudioMetadataFrequencyInHz>();
const auto inBitsPerChannel = metadata.getData<kAudioMetadataBitsPerChannel>();
const auto inChannelCount = metadata.getData<kAudioMetadataChannelCount>();
const auto inFlags = metadata.getData<kAudioMetadataFlags>();
const auto inBytesPerFrame = metadata.getData<kAudioMetadataBytesPerFrame>();
const auto inNumberFrames = metadata.getData<kAudioMetadataNumberFrames>();
const auto inUsesOSStruct = metadata.getData<kAudioMetadataUsesOSStruct>();
if(m_outFrequencyInHz == inFrequncyInHz &&
m_outBitsPerChannel == inBitsPerChannel &&
m_outChannelCount == inChannelCount
&& !(inFlags & kAudioFormatFlagIsNonInterleaved)
&& !(inFlags & kAudioFormatFlagIsFloat))
{
// No resampling necessary
return std::make_shared<Buffer>();
}
uint64_t hash = uint64_t(inBytesPerFrame&0xFF) << 56 | uint64_t(inFlags&0xFF) << 48 | uint64_t(inChannelCount&0xFF) << 40
| uint64_t(inBitsPerChannel&0xFF) << 32 | inFrequncyInHz;
auto it = m_converters.find(hash) ;
ConverterInst converter = {0};
if(it == m_converters.end()) {
AudioStreamBasicDescription in = {0};
AudioStreamBasicDescription out = {0};
in.mFormatID = kAudioFormatLinearPCM;
in.mFormatFlags = inFlags;
in.mChannelsPerFrame = inChannelCount;
in.mSampleRate = inFrequncyInHz;
in.mBitsPerChannel = inBitsPerChannel;
in.mBytesPerFrame = inBytesPerFrame;
in.mFramesPerPacket = 1;
in.mBytesPerPacket = in.mBytesPerFrame * in.mFramesPerPacket;
out.mFormatID = kAudioFormatLinearPCM;
out.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
out.mChannelsPerFrame = m_outChannelCount;
out.mSampleRate = m_outFrequencyInHz;
out.mBitsPerChannel = m_outBitsPerChannel;
out.mBytesPerFrame = (out.mBitsPerChannel * out.mChannelsPerFrame) / 8;
out.mFramesPerPacket = 1;
out.mBytesPerPacket = out.mBytesPerFrame * out.mFramesPerPacket;
converter.asbdIn = in;
converter.asbdOut = out;
OSStatus ret = AudioConverterNew(&in, &out, &converter.converter);
AudioConverterSetProperty(converter.converter,
kAudioConverterSampleRateConverterComplexity,
sizeof(s_samplingRateConverterComplexity),
&s_samplingRateConverterComplexity);
AudioConverterSetProperty(converter.converter,
kAudioConverterSampleRateConverterQuality,
sizeof(s_samplingRateConverterQuality),
&s_samplingRateConverterQuality);
auto prime = kConverterPrimeMethod_None;
AudioConverterSetProperty(converter.converter,
kAudioConverterPrimeMethod,
sizeof(prime),
&prime);
m_converters[hash] = converter;
if(ret != noErr) {
DLog("ret = %d (%x)", (int)ret, (unsigned)ret);
}
} else {
converter = it->second;
}
auto & in = converter.asbdIn;
auto & out = converter.asbdOut;
const double inSampleCount = inNumberFrames;
const double ratio = static_cast<double>(inFrequncyInHz) / static_cast<double>(m_outFrequencyInHz);
const double outBufferSampleCount = std::round(double(inSampleCount) / ratio);
const size_t outBufferSize = out.mBytesPerPacket * outBufferSampleCount;
const auto outBuffer = std::make_shared<Buffer>(outBufferSize);
std::unique_ptr<UserData> ud(new UserData());
ud->size = static_cast<int>(size);
ud->data = const_cast<uint8_t*>(buffer);
ud->p = inUsesOSStruct ? 0 : ud->data;
ud->packetSize = in.mBytesPerPacket;
ud->numberPackets = inSampleCount;
ud->numChannels = inChannelCount;
ud->isInterleaved = !(inFlags & kAudioFormatFlagIsNonInterleaved);
ud->usesOSStruct = inUsesOSStruct;
AudioBufferList outBufferList;
outBufferList.mNumberBuffers = 1;
outBufferList.mBuffers[0].mDataByteSize = static_cast<UInt32>(outBufferSize);
outBufferList.mBuffers[0].mNumberChannels = m_outChannelCount;
outBufferList.mBuffers[0].mData = (*outBuffer)();
UInt32 sampleCount = outBufferSampleCount;
OSStatus ret = AudioConverterFillComplexBuffer(converter.converter, /* AudioConverterRef inAudioConverter */
AudioMixer::ioProc, /* AudioConverterComplexInputDataProc inInputDataProc */
ud.get(), /* void *inInputDataProcUserData */
&sampleCount, /* UInt32 *ioOutputDataPacketSize */
&outBufferList, /* AudioBufferList *outOutputData */
NULL /* AudioStreamPacketDescription *outPacketDescription */
);
if(ret != noErr) {
DLog("ret = %d (%x)", (int)ret, (unsigned)ret);
}
outBuffer->setSize(outBufferList.mBuffers[0].mDataByteSize);
return outBuffer;
}
/* Encode */
output->size = 0;
#if defined(PJMEDIA_ILBC_CODEC_USE_COREAUDIO)&& PJMEDIA_ILBC_CODEC_USE_COREAUDIO
npackets = nsamples / ilbc_codec->enc_samples_per_frame;
theABL.mNumberBuffers = 1;
theABL.mBuffers[0].mNumberChannels = 1;
theABL.mBuffers[0].mDataByteSize = output_buf_len;
theABL.mBuffers[0].mData = output->buf;
ilbc_codec->enc_total_packets = npackets;
ilbc_codec->enc_buffer = (char *)input->buf;
ilbc_codec->enc_buffer_offset = 0;
err = AudioConverterFillComplexBuffer(ilbc_codec->enc, encodeDataProc,
ilbc_codec, &npackets,
&theABL, NULL);
if (err == noErr) {
output->size = npackets * ilbc_codec->enc_frame_size;
}
#else
while (nsamples >= ilbc_codec->enc_samples_per_frame) {
unsigned i;
/* Convert to float */
for (i=0; i<ilbc_codec->enc_samples_per_frame; ++i) {
ilbc_codec->enc_block[i] = (float) (*pcm_in++);
}
iLBC_encode((unsigned char *)output->buf + output->size,
ilbc_codec->enc_block,
static OSStatus gviHardwareCaptureIOProc(AudioDeviceID inDevice,
const AudioTimeStamp * inNow,
const AudioBufferList * inInputData,
const AudioTimeStamp * inInputTime,
AudioBufferList * outOutputData,
const AudioTimeStamp * inOutputTime,
void * inClientData)
{
GVIDevice * device = (GVIDevice *)inClientData;
GVIHardwareData * data = (GVIHardwareData *)device->m_data;
OSStatus result;
UInt32 size;
GVICaptureConverterData converterData;
AudioBufferList bufferList;
GVICapturedFrame * frame;
// get a lock on the device
if(!gviLockDevice(data))
return (OSStatus)1;
// make sure we are capturing
if(data->m_capturing)
{
// setup the buffer list
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0].mNumberChannels = 1;
bufferList.mBuffers[0].mDataByteSize = GVIBytesPerFrame;
bufferList.mBuffers[0].mData = data->m_captureBuffer;
// setup the converter data struct
converterData.m_device = device;
converterData.m_capturedAudio = (AudioBufferList *)inInputData;
converterData.m_used = GVFalse;
// loop while it is converting data
do
{
// request one frame
size = GVISamplesPerFrame;
// convert the captured data into our format
result = AudioConverterFillComplexBuffer(data->m_captureConverter, gviAudioConverterCaptureProc, &converterData, &size, &bufferList, NULL);
// was there enough to fill a buffer
if(result == noErr)
{
// get a frame
frame = gviPopFirstFrame(&data->m_captureAvailableFrames);
// if there aren't any available frames, repurpose the oldest captured frame
if(!frame)
{
frame = gviPopFirstFrame(&data->m_capturedFrames);
assert(frame);
if(!frame)
break;
}
// setup the frame
memcpy(frame->m_frame, data->m_captureBuffer, GVIBytesPerFrame);
frame->m_frameStamp = data->m_captureClock;
// increment the capture clock
data->m_captureClock++;
// add this frame to the end of the capture list
gviPushLastFrame(&data->m_capturedFrames, frame);
}
}
while(result == noErr);
}
// release the device lock
gviUnlockDevice(data);
return noErr;
}
void Convert(MyAudioConverterSettings *mySettings)
{
// create audioConverter object
AudioConverterRef audioConverter;
CheckResult (AudioConverterNew(&mySettings->inputFormat, &mySettings->outputFormat, &audioConverter),
"AudioConveterNew failed");
// allocate packet descriptions if the input file is VBR
UInt32 packetsPerBuffer = 0;
UInt32 outputBufferSize = 32 * 1024; // 32 KB is a good starting point
UInt32 sizePerPacket = mySettings->inputFormat.mBytesPerPacket;
if (sizePerPacket == 0)
{
UInt32 size = sizeof(sizePerPacket);
CheckResult(AudioConverterGetProperty(audioConverter, kAudioConverterPropertyMaximumOutputPacketSize, &size, &sizePerPacket),
"Couldn't get kAudioConverterPropertyMaximumOutputPacketSize");
// make sure the buffer is large enough to hold at least one packet
if (sizePerPacket > outputBufferSize)
outputBufferSize = sizePerPacket;
packetsPerBuffer = outputBufferSize / sizePerPacket;
mySettings->inputFilePacketDescriptions = (AudioStreamPacketDescription*)malloc(sizeof(AudioStreamPacketDescription) * packetsPerBuffer);
}
else
{
packetsPerBuffer = outputBufferSize / sizePerPacket;
}
// allocate destination buffer
UInt8 *outputBuffer = (UInt8 *)malloc(sizeof(UInt8) * outputBufferSize); // CHRIS: not sizeof(UInt8*). check book text!
UInt32 outputFilePacketPosition = 0; //in bytes
while(1)
{
// wrap the destination buffer in an AudioBufferList
AudioBufferList convertedData;
convertedData.mNumberBuffers = 1;
convertedData.mBuffers[0].mNumberChannels = mySettings->inputFormat.mChannelsPerFrame;
convertedData.mBuffers[0].mDataByteSize = outputBufferSize;
convertedData.mBuffers[0].mData = outputBuffer;
// now call the audioConverter to transcode the data. This function will call
// the callback function as many times as required to fulfill the request.
UInt32 ioOutputDataPackets = packetsPerBuffer;
OSStatus error = AudioConverterFillComplexBuffer(audioConverter,
MyAudioConverterCallback,
mySettings,
&ioOutputDataPackets,
&convertedData,
(mySettings->inputFilePacketDescriptions ? mySettings->inputFilePacketDescriptions : nil));
if (error || !ioOutputDataPackets)
{
// fprintf(stderr, "err: %ld, packets: %ld\n", err, ioOutputDataPackets);
break; // this is our termination condition
}
// write the converted data to the output file
// KEVIN: QUESTION: 3rd arg seems like it should be a byte count, not packets. why does this work?
CheckResult (AudioFileWritePackets(mySettings->outputFile,
FALSE,
ioOutputDataPackets,
NULL,
outputFilePacketPosition / mySettings->outputFormat.mBytesPerPacket,
&ioOutputDataPackets,
convertedData.mBuffers[0].mData),
"Couldn't write packets to file");
// advance the output file write location
outputFilePacketPosition += (ioOutputDataPackets * mySettings->outputFormat.mBytesPerPacket);
}
AudioConverterDispose(audioConverter);
}
int adin_mic_read(void *buffer, int nsamples) {
OSStatus status;
#ifdef DEBUG
jlog("Stat: adin_darwin: read: %d samples required\n", nsamples);
#endif
if (!CoreAudioHasInputDevice)
return -1;
if (!CoreAudioRecordStarted) {
status = AudioOutputUnitStart(InputUnit);
CoreAudioRecordStarted = TRUE;
}
UInt32 capacity = BUF_SAMPLES * OutputSamplesPerPacket;
UInt32 npackets = nsamples * OutputSamplesPerPacket;
UInt32 numDataPacketsNeeded;
Sample* inputDataBuf = (Sample*)(BufListConverted->mBuffers[0].mData);
numDataPacketsNeeded = npackets < capacity ? npackets : capacity;
#ifdef DEBUG
jlog("Stat: adin_darwin: numDataPacketsNeeded=%d\n", numDataPacketsNeeded);
#endif
status = AudioConverterFillComplexBuffer(Converter,
ConvInputProc,
NULL, // user data
&numDataPacketsNeeded,
BufListConverted,
NULL // packet description
);
if (status != noErr) {
jlog("Error: adin_darwin: AudioConverterFillComplexBuffer: failed\n");
return -1;
}
#ifdef DEBUG
jlog("Stat: adin_darwin: %d bytes filled (BufListConverted)\n",
BufListConverted->mBuffers[0].mDataByteSize);
#endif
int providedSamples = numDataPacketsNeeded / OutputSamplesPerPacket;
pthread_mutex_lock(&MutexInput);
#ifdef DEBUG
jlog("Stat: adin_darwin: provided samples: %d\n", providedSamples);
#endif
Sample* dst_data = (Sample*)buffer;
int i;
int count = 0;
for (i = 0; i < providedSamples; i++) {
dst_data[i] = inputDataBuf[i];
if (dst_data[i] == 0) count++;
}
//jlog("Stat: adin_darwin: %d zero samples\n", count);
pthread_mutex_unlock(&MutexInput);
#ifdef DEBUG
jlog("Stat: adindarwin: EXIT: %d samples provided\n", providedSamples);
#endif
return providedSamples;
}
OSStatus FeedSound(void *ref,AudioUnitRenderActionFlags *flags,const AudioTimeStamp *time,UInt32 bus,UInt32 count,AudioBufferList *blist){
coreaudio *audio;
audio=(coreaudio*)ref;
return AudioConverterFillComplexBuffer(audio->conv,Feed,ref,&count,blist,0);
}
// _______________________________________________________________________________________
//
void CAAudioFile::Read(UInt32 &ioNumPackets, AudioBufferList *ioData)
// May read fewer packets than requested if:
// buffer is not big enough
// file does not contain that many more packets
// Note that eofErr is not fatal, just results in 0 packets returned
// ioData's buffer sizes may be shortened
{
XThrowIf(mClientMaxPacketSize == 0, kExtAudioFileError_MaxPacketSizeUnknown, "client maximum packet size is 0");
if (mIOBufferList.mBuffers[0].mData == NULL) {
#if DEBUG
printf("warning: CAAudioFile::AllocateBuffers called from ReadPackets\n");
#endif
AllocateBuffers();
}
UInt32 bufferSizeBytes = ioData->mBuffers[0].mDataByteSize;
UInt32 maxNumPackets = bufferSizeBytes / mClientMaxPacketSize;
// older versions of AudioConverterFillComplexBuffer don't do this, so do our own sanity check
UInt32 nPackets = std::min(ioNumPackets, maxNumPackets);
mMaxPacketsToRead = ~0UL;
if (mClientDataFormat.mFramesPerPacket == 1) { // PCM or equivalent
while (mFramesToSkipFollowingSeek > 0) {
UInt32 skipFrames = std::min(mFramesToSkipFollowingSeek, maxNumPackets);
UInt32 framesPerPacket;
if ((framesPerPacket=mFileDataFormat.mFramesPerPacket) > 0)
mMaxPacketsToRead = (skipFrames + framesPerPacket - 1) / framesPerPacket;
if (mConverter == NULL) {
XThrowIfError(ReadInputProc(NULL, &skipFrames, ioData, NULL, this), "read audio file");
} else {
#if CAAUDIOFILE_PROFILE
mInConverter = true;
#endif
StartTiming(this, fill);
XThrowIfError(AudioConverterFillComplexBuffer(mConverter, ReadInputProc, this, &skipFrames, ioData, NULL), "convert audio packets (pcm read)");
ElapsedTime(this, fill, mTicksInConverter);
#if CAAUDIOFILE_PROFILE
mInConverter = false;
#endif
}
if (skipFrames == 0) { // hit EOF
ioNumPackets = 0;
return;
}
mFrameMark += skipFrames;
#if VERBOSE_IO
printf("CAAudioFile::ReadPackets: skipped %ld frames\n", skipFrames);
#endif
mFramesToSkipFollowingSeek -= skipFrames;
// restore mDataByteSize
for (int i = ioData->mNumberBuffers; --i >= 0 ; )
ioData->mBuffers[i].mDataByteSize = bufferSizeBytes;
}
}
if (mFileDataFormat.mFramesPerPacket > 0)
// don't read more packets than we are being asked to produce
mMaxPacketsToRead = nPackets / mFileDataFormat.mFramesPerPacket + 1;
if (mConverter == NULL) {
XThrowIfError(ReadInputProc(NULL, &nPackets, ioData, NULL, this), "read audio file");
} else {
#if CAAUDIOFILE_PROFILE
mInConverter = true;
#endif
StartTiming(this, fill);
XThrowIfError(AudioConverterFillComplexBuffer(mConverter, ReadInputProc, this, &nPackets, ioData, NULL), "convert audio packets (read)");
ElapsedTime(this, fill, mTicksInConverter);
#if CAAUDIOFILE_PROFILE
mInConverter = false;
#endif
}
if (mClientDataFormat.mFramesPerPacket == 1)
mFrameMark += nPackets;
ioNumPackets = nPackets;
}
/* This is called by audio file stream parser when it finds packets of audio */
void Audio_Stream::streamDataCallback(void *inClientData, UInt32 inNumberBytes, UInt32 inNumberPackets, const void *inInputData, AudioStreamPacketDescription *inPacketDescriptions)
{
AS_TRACE("%s: inNumberBytes %u, inNumberPackets %u\n", __FUNCTION__, inNumberBytes, inNumberPackets);
Audio_Stream *THIS = static_cast<Audio_Stream*>(inClientData);
if (!THIS->m_audioStreamParserRunning) {
AS_TRACE("%s: stray callback detected!\n", __PRETTY_FUNCTION__);
return;
}
for (int i = 0; i < inNumberPackets; i++) {
/* Allocate the packet */
UInt32 size = inPacketDescriptions[i].mDataByteSize;
queued_packet_t *packet = (queued_packet_t *)malloc(sizeof(queued_packet_t) + size);
if (THIS->m_bitrateBufferIndex < kAudioStreamBitrateBufferSize) {
// Only keep sampling for one buffer cycle; this is to keep the counters (for instance) duration
// stable.
THIS->m_bitrateBuffer[THIS->m_bitrateBufferIndex++] = 8 * inPacketDescriptions[i].mDataByteSize / THIS->m_packetDuration;
}
/* Prepare the packet */
packet->next = NULL;
packet->desc = inPacketDescriptions[i];
packet->desc.mStartOffset = 0;
memcpy(packet->data, (const char *)inInputData + inPacketDescriptions[i].mStartOffset,
size);
if (THIS->m_queuedHead == NULL) {
THIS->m_queuedHead = THIS->m_queuedTail = packet;
} else {
THIS->m_queuedTail->next = packet;
THIS->m_queuedTail = packet;
}
}
int count = 0;
queued_packet_t *cur = THIS->m_queuedHead;
while (cur) {
cur = cur->next;
count++;
}
Stream_Configuration *config = Stream_Configuration::configuration();
if (count > config->decodeQueueSize) {
AudioBufferList outputBufferList;
outputBufferList.mNumberBuffers = 1;
outputBufferList.mBuffers[0].mNumberChannels = THIS->m_dstFormat.mChannelsPerFrame;
outputBufferList.mBuffers[0].mDataByteSize = THIS->m_outputBufferSize;
outputBufferList.mBuffers[0].mData = THIS->m_outputBuffer;
AudioStreamPacketDescription description;
description.mStartOffset = 0;
description.mDataByteSize = THIS->m_outputBufferSize;
description.mVariableFramesInPacket = 0;
UInt32 ioOutputDataPackets = THIS->m_outputBufferSize / THIS->m_dstFormat.mBytesPerPacket;
AS_TRACE("calling AudioConverterFillComplexBuffer\n");
OSStatus err = AudioConverterFillComplexBuffer(THIS->m_audioConverter,
&encoderDataCallback,
THIS,
&ioOutputDataPackets,
&outputBufferList,
NULL);
if (err == noErr) {
AS_TRACE("%i output bytes available for the audio queue\n", (unsigned int)ioOutputDataPackets);
if (THIS->m_watchdogTimer) {
AS_TRACE("The stream started to play, canceling the watchdog\n");
CFRunLoopTimerInvalidate(THIS->m_watchdogTimer);
CFRelease(THIS->m_watchdogTimer), THIS->m_watchdogTimer = 0;
}
THIS->setState(PLAYING);
THIS->audioQueue()->handleAudioPackets(outputBufferList.mBuffers[0].mDataByteSize,
outputBufferList.mNumberBuffers,
outputBufferList.mBuffers[0].mData,
&description);
if (THIS->m_delegate) {
THIS->m_delegate->samplesAvailable(outputBufferList, description);
}
for(std::list<queued_packet_t*>::iterator iter = THIS->m_processedPackets.begin();
iter != THIS->m_processedPackets.end(); iter++) {
queued_packet_t *cur = *iter;
free(cur);
}
THIS->m_processedPackets.clear();
} else {
AS_TRACE("AudioConverterFillComplexBuffer failed, error %i\n", err);
}
} else {
AS_TRACE("Less than %i packets queued, returning...\n", config->decodeQueueSize);
}
}
nsresult
AppleATDecoder::DecodeSample(mp4_demuxer::MP4Sample* aSample)
{
// Array containing the queued decoded audio frames, about to be output.
nsTArray<AudioDataValue> outputData;
UInt32 channels = mOutputFormat.mChannelsPerFrame;
// Pick a multiple of the frame size close to a power of two
// for efficient allocation.
const uint32_t MAX_AUDIO_FRAMES = 128;
const uint32_t maxDecodedSamples = MAX_AUDIO_FRAMES * channels;
// Descriptions for _decompressed_ audio packets. ignored.
nsAutoArrayPtr<AudioStreamPacketDescription>
packets(new AudioStreamPacketDescription[MAX_AUDIO_FRAMES]);
// This API insists on having packets spoon-fed to it from a callback.
// This structure exists only to pass our state.
PassthroughUserData userData =
{ channels, (UInt32)aSample->size, aSample->data };
// Decompressed audio buffer
nsAutoArrayPtr<AudioDataValue> decoded(new AudioDataValue[maxDecodedSamples]);
do {
AudioBufferList decBuffer;
decBuffer.mNumberBuffers = 1;
decBuffer.mBuffers[0].mNumberChannels = channels;
decBuffer.mBuffers[0].mDataByteSize =
maxDecodedSamples * sizeof(AudioDataValue);
decBuffer.mBuffers[0].mData = decoded.get();
// in: the max number of packets we can handle from the decoder.
// out: the number of packets the decoder is actually returning.
UInt32 numFrames = MAX_AUDIO_FRAMES;
OSStatus rv = AudioConverterFillComplexBuffer(mConverter,
_PassthroughInputDataCallback,
&userData,
&numFrames /* in/out */,
&decBuffer,
packets.get());
if (rv && rv != kNoMoreDataErr) {
LOG("Error decoding audio stream: %d\n", rv);
return NS_ERROR_FAILURE;
}
if (numFrames) {
outputData.AppendElements(decoded.get(), numFrames * channels);
}
if (rv == kNoMoreDataErr) {
break;
}
} while (true);
if (outputData.IsEmpty()) {
return NS_OK;
}
size_t numFrames = outputData.Length() / channels;
int rate = mOutputFormat.mSampleRate;
CheckedInt<Microseconds> duration = FramesToUsecs(numFrames, rate);
if (!duration.isValid()) {
NS_WARNING("Invalid count of accumulated audio samples");
return NS_ERROR_FAILURE;
}
#ifdef LOG_SAMPLE_DECODE
LOG("pushed audio at time %lfs; duration %lfs\n",
(double)aSample->composition_timestamp / USECS_PER_S,
(double)duration.value() / USECS_PER_S);
#endif
nsAutoArrayPtr<AudioDataValue> data(new AudioDataValue[outputData.Length()]);
PodCopy(data.get(), &outputData[0], outputData.Length());
nsRefPtr<AudioData> audio = new AudioData(aSample->byte_offset,
aSample->composition_timestamp,
duration.value(),
numFrames,
data.forget(),
channels,
rate);
mCallback->Output(audio);
return NS_OK;
}
/*
* This callback is called when |AudioFileStreamParseBytes| has enough data to
* extract one or more MP3 packets.
*/
void
AppleMP3Reader::AudioSampleCallback(UInt32 aNumBytes,
UInt32 aNumPackets,
const void *aData,
AudioStreamPacketDescription *aPackets)
{
LOGD("got %u bytes, %u packets\n", aNumBytes, aNumPackets);
// 1 frame per packet * num channels * 32-bit float
uint32_t decodedSize = MAX_AUDIO_FRAMES * mAudioChannels *
sizeof(AudioDataValue);
// descriptions for _decompressed_ audio packets. ignored.
nsAutoArrayPtr<AudioStreamPacketDescription>
packets(new AudioStreamPacketDescription[MAX_AUDIO_FRAMES]);
// This API insists on having MP3 packets spoon-fed to it from a callback.
// This structure exists only to pass our state and the result of the parser
// on to the callback above.
PassthroughUserData userData = { this, aNumPackets, aNumBytes, aData, aPackets, false };
do {
// Decompressed audio buffer
nsAutoArrayPtr<uint8_t> decoded(new uint8_t[decodedSize]);
AudioBufferList decBuffer;
decBuffer.mNumberBuffers = 1;
decBuffer.mBuffers[0].mNumberChannels = mAudioChannels;
decBuffer.mBuffers[0].mDataByteSize = decodedSize;
decBuffer.mBuffers[0].mData = decoded.get();
// in: the max number of packets we can handle from the decoder.
// out: the number of packets the decoder is actually returning.
UInt32 numFrames = MAX_AUDIO_FRAMES;
OSStatus rv = AudioConverterFillComplexBuffer(mAudioConverter,
PassthroughInputDataCallback,
&userData,
&numFrames /* in/out */,
&decBuffer,
packets.get());
if (rv && rv != kNeedMoreData) {
LOGE("Error decoding audio stream: %x\n", rv);
break;
}
// If we decoded zero frames then AudiOConverterFillComplexBuffer is out
// of data to provide. We drained its internal buffer completely on the
// last pass.
if (numFrames == 0 && rv == kNeedMoreData) {
LOGD("FillComplexBuffer out of data exactly\n");
break;
}
int64_t time = FramesToUsecs(mCurrentAudioFrame, mAudioSampleRate).value();
int64_t duration = FramesToUsecs(numFrames, mAudioSampleRate).value();
LOGD("pushed audio at time %lfs; duration %lfs\n",
(double)time / USECS_PER_S, (double)duration / USECS_PER_S);
AudioData *audio = new AudioData(mDecoder->GetResource()->Tell(),
time, duration, numFrames,
reinterpret_cast<AudioDataValue *>(decoded.forget()),
mAudioChannels, mAudioSampleRate);
mAudioQueue.Push(audio);
mCurrentAudioFrame += numFrames;
if (rv == kNeedMoreData) {
// No error; we just need more data.
LOGD("FillComplexBuffer out of data\n");
break;
}
} while (true);
}
OSStatus FCoreAudioSoundSource::CoreAudioRenderCallback( void *InRefCon, AudioUnitRenderActionFlags *IOActionFlags,
const AudioTimeStamp *InTimeStamp, UInt32 InBusNumber,
UInt32 InNumberFrames, AudioBufferList *IOData )
{
OSStatus Status = noErr;
FCoreAudioSoundSource *Source = ( FCoreAudioSoundSource *)InRefCon;
uint32 DataByteSize = InNumberFrames * sizeof( Float32 );
uint32 PacketsRequested = InNumberFrames;
uint32 PacketsObtained = 0;
// AudioBufferList itself holds only one buffer, while AudioConverterFillComplexBuffer expects a couple of them
struct
{
AudioBufferList BufferList;
AudioBuffer AdditionalBuffers[5];
} LocalBuffers;
AudioBufferList *LocalBufferList = &LocalBuffers.BufferList;
LocalBufferList->mNumberBuffers = IOData->mNumberBuffers;
if( Source->Buffer && Source->Playing )
{
while( PacketsObtained < PacketsRequested )
{
int32 BufferFilledBytes = PacketsObtained * sizeof( Float32 );
for( uint32 Index = 0; Index < LocalBufferList->mNumberBuffers; Index++ )
{
LocalBufferList->mBuffers[Index].mDataByteSize = DataByteSize - BufferFilledBytes;
LocalBufferList->mBuffers[Index].mData = ( uint8 *)IOData->mBuffers[Index].mData + BufferFilledBytes;
}
uint32 PacketCount = PacketsRequested - PacketsObtained;
Status = AudioConverterFillComplexBuffer( Source->CoreAudioConverter, &CoreAudioConvertCallback, InRefCon, &PacketCount, LocalBufferList, NULL );
PacketsObtained += PacketCount;
if( PacketCount == 0 || Status != noErr )
{
AudioConverterReset( Source->CoreAudioConverter );
break;
}
}
if( PacketsObtained == 0 )
{
*IOActionFlags |= kAudioUnitRenderAction_OutputIsSilence;
}
}
else
{
*IOActionFlags |= kAudioUnitRenderAction_OutputIsSilence;
}
if( PacketsObtained < PacketsRequested )
{
// Fill the rest of buffers provided with zeroes
int32 BufferFilledBytes = PacketsObtained * sizeof( Float32 );
for( uint32 Index = 0; Index < IOData->mNumberBuffers; ++Index )
{
FMemory::Memzero( ( uint8 *)IOData->mBuffers[Index].mData + BufferFilledBytes, DataByteSize - BufferFilledBytes );
}
}
return Status;
}