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changed Mac downstream ringbuffer to 100 frames

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wangyix 2014-07-30 11:13:15 -07:00
parent 093e78ac52
commit 2b77a72aba
1 changed files with 12 additions and 8 deletions
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20
interface/src/Audio.cpp
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@ -75,11 +75,7 @@ Audio::Audio(QObject* parent) :
// slower than real time (or at least the desired sample rate). If you increase the size of the ring buffer, then it
// this delay will slowly add up and the longer someone runs, they more delayed their audio will be.
_inputRingBuffer(0),
#ifdef _WIN32
_receivedAudioStream(NETWORK_BUFFER_LENGTH_SAMPLES_STEREO, 100, true, 20, true),
#else
_receivedAudioStream(NETWORK_BUFFER_LENGTH_SAMPLES_STEREO, 10, true, 20, true), // DO NOT CHANGE THIS UNLESS YOU SOLVE THE AUDIO DEVICE DRIFT PROBLEM!!!
#endif
_receivedAudioStream(NETWORK_BUFFER_LENGTH_SAMPLES_STEREO, 100, true, 0, true),
_isStereoInput(false),
_averagedLatency(0.0),
_lastInputLoudness(0),
@ -896,6 +892,12 @@ void Audio::processReceivedAudio(const QByteArray& audioByteArray) {
// parse audio data
_receivedAudioStream.parseData(audioByteArray);
// This call has been moved to handleAudioInput. handleAudioInput is called at a much more regular interval
// than processReceivedAudio since handleAudioInput does not experience network-related jitter.
// This way, we reduce the jitter of the frames being pushed to the audio output, allowing us to use a reduced
// buffer size for it, which reduces latency.
//pushAudioToOutput();
}
@ -1659,11 +1661,13 @@ bool Audio::switchOutputToAudioDevice(const QAudioDeviceInfo& outputDeviceInfo)
if (adjustedFormatForAudioDevice(outputDeviceInfo, _desiredOutputFormat, _outputFormat)) {
qDebug() << "The format to be used for audio output is" << _outputFormat;
const int AUDIO_OUTPUT_BUFFER_SIZE_FRAMES = 10;
// setup our general output device for audio-mixer audio
_audioOutput = new QAudioOutput(outputDeviceInfo, _outputFormat, this);
_audioOutput->setBufferSize(/*_receivedAudioStream.getFrameCapacity()*/ 10 * _outputFormat.bytesForDuration(BUFFER_SEND_INTERVAL_USECS));
qDebug() << "Ring Buffer capacity in frames: " << _receivedAudioStream.getFrameCapacity();
_audioOutput->setBufferSize(AUDIO_OUTPUT_BUFFER_SIZE_FRAMES * _outputFormat.bytesForDuration(BUFFER_SEND_INTERVAL_USECS));
qDebug() << "Ring Buffer capacity in frames: " << AUDIO_OUTPUT_BUFFER_SIZE_FRAMES;
_outputDevice = _audioOutput->start();
// setup a loopback audio output device