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use network period multiple for local audio buffer

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Zach Pomerantz 2017-05-08 20:05:14 -04:00
parent 6da925a626
commit b6f652e0ea
1 changed files with 10 additions and 1 deletions
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11
libraries/audio-client/src/AudioClient.cpp
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@ -1525,14 +1525,23 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo& outputDevice
connect(_audioOutput, &QAudioOutput::stateChanged, [&, frameSize, requestedSize](QAudio::State state) {
if (state == QAudio::ActiveState) {
// restrict device callback to _outputPeriod samples
_outputPeriod = (_audioOutput->periodSize() / AudioConstants::SAMPLE_SIZE) * 2;
_outputPeriod = _audioOutput->periodSize() / AudioConstants::SAMPLE_SIZE;
// device callback may exceed reported period, so double it to avoid stutter
_outputPeriod *= 2;
_outputMixBuffer = new float[_outputPeriod];
_outputScratchBuffer = new int16_t[_outputPeriod];
// size local output mix buffer based on resampled network frame size
_networkPeriod = _localToOutputResampler->getMaxOutput(AudioConstants::NETWORK_FRAME_SAMPLES_STEREO);
_localOutputMixBuffer = new float[_networkPeriod];
// local period should be at least twice the output period,
// in case two device reads happen before more data can be read (worst case)
int localPeriod = _outputPeriod * 2;
// round up to an exact multiple of _networkPeriod
localPeriod = ((localPeriod + _networkPeriod - 1) / _networkPeriod) * _networkPeriod;
// this ensures lowest latency without stutter from underrun
_localInjectorsStream.resizeForFrameSize(localPeriod);
int bufferSize = _audioOutput->bufferSize();