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Merge pull request #15637 from kencooke/audio-init-racecond-crashfix

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BUGZ-158: Crash when switching audio devices on Mac
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Shannon Romano 2019-05-25 16:26:56 -07:00 committed by GitHub
commit 12996b2d7d
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GPG key ID: 4AEE18F83AFDEB23
2 changed files with 47 additions and 41 deletions
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87
libraries/audio-client/src/AudioClient.cpp
View file

@ -1869,6 +1869,7 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo outputDeviceI
if (_audioOutput) {
_audioOutputIODevice.close();
_audioOutput->stop();
_audioOutputInitialized = false;
//must be deleted in next eventloop cycle when its called from notify()
_audioOutput->deleteLater();
@ -1939,52 +1940,50 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo outputDeviceI
int requestedSize = _sessionOutputBufferSizeFrames * frameSize * AudioConstants::SAMPLE_SIZE;
_audioOutput->setBufferSize(requestedSize);
// initialize mix buffers on the _audioOutput thread to avoid races
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;
// 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
int networkPeriod = _localToOutputResampler ? _localToOutputResampler->getMaxOutput(AudioConstants::NETWORK_FRAME_SAMPLES_STEREO) : 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();
int bufferSamples = bufferSize / AudioConstants::SAMPLE_SIZE;
int bufferFrames = bufferSamples / (float)frameSize;
qCDebug(audioclient) << "frame (samples):" << frameSize;
qCDebug(audioclient) << "buffer (frames):" << bufferFrames;
qCDebug(audioclient) << "buffer (samples):" << bufferSamples;
qCDebug(audioclient) << "buffer (bytes):" << bufferSize;
qCDebug(audioclient) << "requested (bytes):" << requestedSize;
qCDebug(audioclient) << "period (samples):" << _outputPeriod;
qCDebug(audioclient) << "local buffer (samples):" << localPeriod;
disconnect(_audioOutput, &QAudioOutput::stateChanged, 0, 0);
// unlock to avoid a deadlock with the device callback (which always succeeds this initialization)
localAudioLock.unlock();
}
});
connect(_audioOutput, &QAudioOutput::notify, this, &AudioClient::outputNotify);
// start the output device
_audioOutputIODevice.start();
_audioOutput->start(&_audioOutputIODevice);
// initialize mix buffers
// restrict device callback to _outputPeriod samples
_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
int networkPeriod = _localToOutputResampler ? _localToOutputResampler->getMaxOutput(AudioConstants::NETWORK_FRAME_SAMPLES_STEREO) : 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);
_audioOutputInitialized = true;
int bufferSize = _audioOutput->bufferSize();
int bufferSamples = bufferSize / AudioConstants::SAMPLE_SIZE;
int bufferFrames = bufferSamples / (float)frameSize;
qCDebug(audioclient) << "frame (samples):" << frameSize;
qCDebug(audioclient) << "buffer (frames):" << bufferFrames;
qCDebug(audioclient) << "buffer (samples):" << bufferSamples;
qCDebug(audioclient) << "buffer (bytes):" << bufferSize;
qCDebug(audioclient) << "requested (bytes):" << requestedSize;
qCDebug(audioclient) << "period (samples):" << _outputPeriod;
qCDebug(audioclient) << "local buffer (samples):" << localPeriod;
// unlock to avoid a deadlock with the device callback (which always succeeds this initialization)
localAudioLock.unlock();
// setup a loopback audio output device
_loopbackAudioOutput = new QAudioOutput(outputDeviceInfo, _outputFormat, this);
@ -2082,6 +2081,12 @@ float AudioClient::gainForSource(float distance, float volume) {
qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) {
// lock-free wait for initialization to avoid races
if (!_audio->_audioOutputInitialized.load(std::memory_order_acquire)) {
memset(data, 0, maxSize);
return maxSize;
}
// samples requested from OUTPUT_CHANNEL_COUNT
int deviceChannelCount = _audio->_outputFormat.channelCount();
int samplesRequested = (int)(maxSize / AudioConstants::SAMPLE_SIZE) * OUTPUT_CHANNEL_COUNT / deviceChannelCount;

1
libraries/audio-client/src/AudioClient.h
View file

@ -340,6 +340,7 @@ private:
QIODevice* _inputDevice;
int _numInputCallbackBytes;
QAudioOutput* _audioOutput;
std::atomic<bool> _audioOutputInitialized { false };
QAudioFormat _desiredOutputFormat;
QAudioFormat _outputFormat;
int _outputFrameSize;