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add audio limiting to device callback

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Zach Pomerantz 2017-01-08 23:24:23 -05:00
parent 24d53ea13c
commit a7ecf41a42
2 changed files with 42 additions and 7 deletions
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44
libraries/audio-client/src/AudioClient.cpp
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@ -1386,6 +1386,12 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo& outputDevice
_loopbackOutputDevice = NULL;
delete _loopbackAudioOutput;
_loopbackAudioOutput = NULL;
delete[] _limitMixBuffer;
_limitMixBuffer = NULL;
delete[] _limitScratchBuffer;
_limitScratchBuffer = NULL;
}
if (_networkToOutputResampler) {
@ -1436,6 +1442,12 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo& outputDevice
_audioOutput->start(&_audioOutputIODevice);
lock.unlock();
int periodSampleSize = _audioOutput->periodSize() / AudioConstants::SAMPLE_SIZE;
// device callback is not restricted to periodSampleSize, so double the mix/scratch buffer sizes
_limitPeriod = periodSampleSize * 2;
_limitMixBuffer = new float[_limitPeriod];
_limitScratchBuffer = new int16_t[_limitPeriod];
qCDebug(audioclient) << "Output Buffer capacity in frames: " << _audioOutput->bufferSize() / AudioConstants::SAMPLE_SIZE / (float)deviceFrameSize <<
"requested bytes:" << requestedSize << "actual bytes:" << _audioOutput->bufferSize() <<
"os default:" << osDefaultBufferSize << "period size:" << _audioOutput->periodSize();
@ -1545,6 +1557,8 @@ qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) {
// samples requested from OUTPUT_CHANNEL_COUNT
int deviceChannelCount = _audio->_outputFormat.channelCount();
int samplesRequested = (int)(maxSize / AudioConstants::SAMPLE_SIZE) * OUTPUT_CHANNEL_COUNT / deviceChannelCount;
// restrict samplesRequested to the size of our mix/scratch buffers
samplesRequested = std::min(samplesRequested, _audio->_limitPeriod);
int samplesPopped;
int bytesWritten;
@ -1553,14 +1567,30 @@ qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) {
qCDebug(audiostream, "Read %d samples from buffer (%d available)", samplesPopped, _receivedAudioStream.getSamplesAvailable());
AudioRingBuffer::ConstIterator lastPopOutput = _receivedAudioStream.getLastPopOutput();
// if required, upmix or downmix to deviceChannelCount
if (deviceChannelCount == OUTPUT_CHANNEL_COUNT) {
lastPopOutput.readSamples((int16_t*)data, samplesPopped);
} else if (deviceChannelCount > OUTPUT_CHANNEL_COUNT) {
lastPopOutput.readSamplesWithUpmix((int16_t*)data, samplesPopped, deviceChannelCount - OUTPUT_CHANNEL_COUNT);
} else {
lastPopOutput.readSamplesWithDownmix((int16_t*)data, samplesPopped);
int16_t* scratchBuffer = _audio->_limitScratchBuffer;
lastPopOutput.readSamples(scratchBuffer, samplesPopped);
float* mixBuffer = _audio->_limitMixBuffer;
for (int i = 0; i < samplesPopped; i++) {
mixBuffer[i] = (float)scratchBuffer[i] * (1/32768.0f);
}
int framesPopped = samplesPopped / AudioConstants::STEREO;
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 = (samplesPopped * AudioConstants::SAMPLE_SIZE) * deviceChannelCount / OUTPUT_CHANNEL_COUNT;
} else {
// nothing on network, don't grab anything from injectors, and just return 0s

5
libraries/audio-client/src/AudioClient.h
View file

@ -300,6 +300,11 @@ private:
// for local hrtf-ing
float _mixBuffer[AudioConstants::NETWORK_FRAME_SAMPLES_STEREO];
int16_t _scratchBuffer[AudioConstants::NETWORK_FRAME_SAMPLES_AMBISONIC];
// for limiting
int _limitPeriod { 0 };
float* _limitMixBuffer { NULL };
int16_t* _limitScratchBuffer { NULL };
AudioLimiter _audioLimiter;
// Adds Reverb