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coreaudio: refactor output callback

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AudioLayer contains generic procedure to get AudioSamples and
resample them. We can remove duplicated code in CoreLayer

Change-Id: I3c756a1ff71e96a277b3b86552aa8912c05abb90
Tuleap: #324
This commit is contained in:
Alexandre Lision
2016-02-02 14:19:12 -05:00
parent 4c27b23493
commit 50763c6e64
1 changed files with 8 additions and 70 deletions
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78
src/media/audio/coreaudio/corelayer.cpp
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@ -285,80 +285,18 @@ void CoreLayer::write(AudioUnitRenderActionFlags* ioActionFlags,
UInt32 inNumberFrames,
AudioBufferList* ioData)
{
// Checks for resampling
AudioFormat mainBufferAudioFormat = Manager::instance().getRingBufferPool().getInternalAudioFormat();
bool resample = audioFormat_.sample_rate != mainBufferAudioFormat.sample_rate;
auto& ringBuff = getToRing(audioFormat_, inNumberFrames);
auto& playBuff = getToPlay(audioFormat_, inNumberFrames);
unsigned urgentFramesToGet = urgentRingBuffer_.availableForGet(RingBufferPool::DEFAULT_ID);
if (urgentFramesToGet > 0) {
RING_WARN("Getting urgent frames.");
size_t totSample = std::min(inNumberFrames, urgentFramesToGet);
playbackBuff_.setFormat(audioFormat_);
playbackBuff_.resize(totSample);
urgentRingBuffer_.get(playbackBuff_, RingBufferPool::DEFAULT_ID);
playbackBuff_.applyGain(isPlaybackMuted_ ? 0.0 : playbackGain_);
auto& toPlay = ringBuff.frames() > 0 ? ringBuff : playBuff;
if (toPlay.frames() == 0) {
for (int i = 0; i < audioFormat_.nb_channels; ++i)
playbackBuff_.channelToFloat((Float32*)ioData->mBuffers[i].mData, i); // Write
Manager::instance().getRingBufferPool().discard(totSample, RingBufferPool::DEFAULT_ID);
memset((Float32*)ioData->mBuffers[i].mData, 0, ioData->mBuffers[i].mDataByteSize);
}
unsigned normalFramesToGet = Manager::instance().getRingBufferPool().availableForGet(RingBufferPool::DEFAULT_ID);
if (normalFramesToGet > 0) {
double resampleFactor = 1.0;
unsigned readableSamples = inNumberFrames;
if (resample) {
resampleFactor = static_cast<double>(audioFormat_.sample_rate) / mainBufferAudioFormat.sample_rate;
readableSamples = std::ceil(inNumberFrames / resampleFactor);
}
readableSamples = std::min(readableSamples, normalFramesToGet);
size_t nResampled = (double) readableSamples * resampleFactor;
playbackBuff_.setFormat(mainBufferAudioFormat);
playbackBuff_.resize(readableSamples);
Manager::instance().getRingBufferPool().getData(
playbackBuff_, RingBufferPool::DEFAULT_ID);
playbackBuff_.applyGain(isPlaybackMuted_ ? 0.0 : playbackGain_);
if (resample) {
AudioBuffer resampledOutput(readableSamples, audioFormat_);
resampler_->resample(playbackBuff_, resampledOutput);
for (int i = 0; i < audioFormat_.nb_channels; ++i)
resampledOutput.channelToFloat((Float32*)ioData->mBuffers[i].mData, i);
} else {
for (int i = 0; i < audioFormat_.nb_channels; ++i)
playbackBuff_.channelToFloat((Float32*)ioData->mBuffers[i].mData, i);
}
}
if (normalFramesToGet <= 0) {
AudioLoop* tone = Manager::instance().getTelephoneTone();
AudioLoop* file_tone = Manager::instance().getTelephoneFile();
playbackBuff_.setFormat(audioFormat_);
playbackBuff_.resize(inNumberFrames);
if (tone) {
tone->getNext(playbackBuff_, playbackGain_);
}
else if (file_tone) {
file_tone->getNext(playbackBuff_, playbackGain_);
}
else {
playbackBuff_.reset();
}
for (int i = 0; i < audioFormat_.nb_channels; ++i) {
playbackBuff_.channelToFloat((Float32*)ioData->mBuffers[i].mData, i);
}
else {
for (int i = 0; i < audioFormat_.nb_channels; ++i)
toPlay.channelToFloat((Float32*)ioData->mBuffers[i].mData, i);
}
}