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Correctly handle time-varying buffer size due to loopback resampling

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This commit is contained in:
Ken Cooke 2019-06-01 14:27:32 -07:00
parent 451cf60e54
commit a56942e494
1 changed files with 25 additions and 19 deletions
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44
libraries/audio-client/src/AudioClient.cpp
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@ -657,11 +657,11 @@ bool adjustedFormatForAudioDevice(const QAudioDeviceInfo& audioDevice,
return false; // a supported format could not be found
}
bool sampleChannelConversion(const int16_t* sourceSamples, int16_t* destinationSamples, unsigned int numSourceSamples,
bool sampleChannelConversion(const int16_t* sourceSamples, int16_t* destinationSamples, int numSourceSamples,
const int sourceChannelCount, const int destinationChannelCount) {
if (sourceChannelCount == 2 && destinationChannelCount == 1) {
// loop through the stereo input audio samples and average every two samples
for (uint i = 0; i < numSourceSamples; i += 2) {
for (int i = 0; i < numSourceSamples; i += 2) {
destinationSamples[i / 2] = (sourceSamples[i] / 2) + (sourceSamples[i + 1] / 2);
}
@ -669,7 +669,7 @@ bool sampleChannelConversion(const int16_t* sourceSamples, int16_t* destinationS
} else if (sourceChannelCount == 1 && destinationChannelCount == 2) {
// loop through the mono input audio and repeat each sample twice
for (uint i = 0; i < numSourceSamples; ++i) {
for (int i = 0; i < numSourceSamples; ++i) {
destinationSamples[i * 2] = destinationSamples[(i * 2) + 1] = sourceSamples[i];
}
@ -679,10 +679,13 @@ bool sampleChannelConversion(const int16_t* sourceSamples, int16_t* destinationS
return false;
}
void possibleResampling(AudioSRC* resampler,
const int16_t* sourceSamples, int16_t* destinationSamples,
unsigned int numSourceSamples, unsigned int numDestinationSamples,
const int sourceChannelCount, const int destinationChannelCount) {
int possibleResampling(AudioSRC* resampler,
const int16_t* sourceSamples, int16_t* destinationSamples,
int numSourceSamples, int maxDestinationSamples,
const int sourceChannelCount, const int destinationChannelCount) {
int numSourceFrames = numSourceSamples / sourceChannelCount;
int numDestinationFrames = 0;
if (numSourceSamples > 0) {
if (!resampler) {
@ -691,10 +694,8 @@ void possibleResampling(AudioSRC* resampler,
// no conversion, we can copy the samples directly across
memcpy(destinationSamples, sourceSamples, numSourceSamples * AudioConstants::SAMPLE_SIZE);
}
numDestinationFrames = numSourceFrames;
} else {
int numSourceFrames = numSourceSamples / sourceChannelCount;
if (sourceChannelCount != destinationChannelCount) {
int16_t* channelConversionSamples = new int16_t[numSourceFrames * destinationChannelCount];
@ -702,14 +703,17 @@ void possibleResampling(AudioSRC* resampler,
sampleChannelConversion(sourceSamples, channelConversionSamples, numSourceSamples,
sourceChannelCount, destinationChannelCount);
resampler->render(channelConversionSamples, destinationSamples, numSourceFrames);
numDestinationFrames = resampler->render(channelConversionSamples, destinationSamples, numSourceFrames);
delete[] channelConversionSamples;
} else {
resampler->render(sourceSamples, destinationSamples, numSourceFrames);
numDestinationFrames = resampler->render(sourceSamples, destinationSamples, numSourceFrames);
}
}
}
int numDestinationSamples = numDestinationFrames * destinationChannelCount;
return numDestinationSamples;
}
void AudioClient::start() {
@ -1104,18 +1108,20 @@ void AudioClient::handleLocalEchoAndReverb(QByteArray& inputByteArray) {
int numInputSamples = inputByteArray.size() / AudioConstants::SAMPLE_SIZE;
int numInputFrames = numInputSamples / _inputFormat.channelCount();
int numLoopbackFrames = (numInputFrames * _outputFormat.sampleRate() + _inputFormat.sampleRate() - 1) / _inputFormat.sampleRate();
int numLoopbackSamples = numLoopbackFrames * OUTPUT_CHANNEL_COUNT;
int maxLoopbackFrames = _loopbackResampler ? _loopbackResampler->getMaxOutput(numInputFrames) : numInputFrames;
int maxLoopbackSamples = maxLoopbackFrames * OUTPUT_CHANNEL_COUNT;
loopBackByteArray.resize(numLoopbackSamples * AudioConstants::SAMPLE_SIZE);
loopBackByteArray.resize(maxLoopbackSamples * AudioConstants::SAMPLE_SIZE);
int16_t* inputSamples = reinterpret_cast<int16_t*>(inputByteArray.data());
int16_t* loopbackSamples = reinterpret_cast<int16_t*>(loopBackByteArray.data());
possibleResampling(_loopbackResampler,
inputSamples, loopbackSamples,
numInputSamples, numLoopbackSamples,
_inputFormat.channelCount(), OUTPUT_CHANNEL_COUNT);
int numLoopbackSamples = possibleResampling(_loopbackResampler,
inputSamples, loopbackSamples,
numInputSamples, maxLoopbackSamples,
_inputFormat.channelCount(), OUTPUT_CHANNEL_COUNT);
loopBackByteArray.resize(numLoopbackSamples * AudioConstants::SAMPLE_SIZE);
// apply stereo reverb at the source, to the loopback audio
if (!_shouldEchoLocally && hasReverb) {