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Refactor the audio output callback, to expose the final output (far-end) needed for AEC

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Ken Cooke 2019-07-18 11:43:46 -07:00 committed by Seth Alves
parent 32d5ab7b1f
commit 80aeeaa3e0
1 changed files with 39 additions and 36 deletions
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75
libraries/audio-client/src/AudioClient.cpp
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@ -2114,15 +2114,16 @@ qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) {
return maxSize;
}
// samples requested from OUTPUT_CHANNEL_COUNT
// max samples requested from OUTPUT_CHANNEL_COUNT
int deviceChannelCount = _audio->_outputFormat.channelCount();
int samplesRequested = (int)(maxSize / AudioConstants::SAMPLE_SIZE) * OUTPUT_CHANNEL_COUNT / deviceChannelCount;
int maxSamplesRequested = (int)(maxSize / AudioConstants::SAMPLE_SIZE) * OUTPUT_CHANNEL_COUNT / deviceChannelCount;
// restrict samplesRequested to the size of our mix/scratch buffers
samplesRequested = std::min(samplesRequested, _audio->_outputPeriod);
maxSamplesRequested = std::min(maxSamplesRequested, _audio->_outputPeriod);
int16_t* scratchBuffer = _audio->_outputScratchBuffer;
float* mixBuffer = _audio->_outputMixBuffer;
int samplesRequested = maxSamplesRequested;
int networkSamplesPopped;
if ((networkSamplesPopped = _receivedAudioStream.popSamples(samplesRequested, false)) > 0) {
qCDebug(audiostream, "Read %d samples from buffer (%d available, %d requested)", networkSamplesPopped, _receivedAudioStream.getSamplesAvailable(), samplesRequested);
@ -2167,40 +2168,42 @@ qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) {
});
int samplesPopped = std::max(networkSamplesPopped, injectorSamplesPopped);
int framesPopped = samplesPopped / AudioConstants::STEREO;
int bytesWritten;
if (samplesPopped > 0) {
// apply output gain
float newGain = _audio->_outputGain.load(std::memory_order_acquire);
float oldGain = _audio->_lastOutputGain;
_audio->_lastOutputGain = newGain;
applyGainSmoothing<OUTPUT_CHANNEL_COUNT>(mixBuffer, framesPopped, oldGain, newGain);
if (deviceChannelCount == OUTPUT_CHANNEL_COUNT) {
// limit the audio
_audio->_audioLimiter.render(mixBuffer, (int16_t*)data, framesPopped);
} else {
_audio->_audioLimiter.render(mixBuffer, scratchBuffer, framesPopped);
// upmix or downmix to deviceChannelCount
if (deviceChannelCount > OUTPUT_CHANNEL_COUNT) {
int extraChannels = deviceChannelCount - OUTPUT_CHANNEL_COUNT;
channelUpmix(scratchBuffer, (int16_t*)data, samplesPopped, extraChannels);
} else {
channelDownmix(scratchBuffer, (int16_t*)data, samplesPopped);
}
}
bytesWritten = framesPopped * AudioConstants::SAMPLE_SIZE * deviceChannelCount;
assert(bytesWritten <= maxSize);
} else {
// nothing on network, don't grab anything from injectors, and just return 0s
memset(data, 0, maxSize);
bytesWritten = maxSize;
if (samplesPopped == 0) {
// nothing on network, don't grab anything from injectors, and fill with silence
samplesPopped = maxSamplesRequested;
memset(mixBuffer, 0, samplesPopped * sizeof(float));
}
int framesPopped = samplesPopped / OUTPUT_CHANNEL_COUNT;
// apply output gain
float newGain = _audio->_outputGain.load(std::memory_order_acquire);
float oldGain = _audio->_lastOutputGain;
_audio->_lastOutputGain = newGain;
applyGainSmoothing<OUTPUT_CHANNEL_COUNT>(mixBuffer, framesPopped, oldGain, newGain);
// limit the audio
_audio->_audioLimiter.render(mixBuffer, scratchBuffer, framesPopped);
// TODO:
// At this point, scratchBuffer contains the final (mixed, limited) output audio.
// format = interleaved int16_t
// samples = samplesPopped
// channels = OUTPUT_CHANNEL_COUNT
// sampleRate = _outputFormat.sampleRate()
// This can be used as the far-end signal for AEC.
// if required, upmix or downmix to deviceChannelCount
if (deviceChannelCount == OUTPUT_CHANNEL_COUNT) {
memcpy(data, scratchBuffer, samplesPopped * AudioConstants::SAMPLE_SIZE);
} else if (deviceChannelCount > OUTPUT_CHANNEL_COUNT) {
int extraChannels = deviceChannelCount - OUTPUT_CHANNEL_COUNT;
channelUpmix(scratchBuffer, (int16_t*)data, samplesPopped, extraChannels);
} else {
channelDownmix(scratchBuffer, (int16_t*)data, samplesPopped);
}
int bytesWritten = framesPopped * AudioConstants::SAMPLE_SIZE * deviceChannelCount;
assert(bytesWritten <= maxSize);
// send output buffer for recording
if (_audio->_isRecording) {