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Bypass WebRTC processing when not enabled.

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Note: WebRTC processing with all features disabled still holds locks and performs resampling in the callback.
This commit is contained in:
Ken Cooke 2019-07-26 12:45:21 -07:00 committed by Seth Alves
parent e6e8491098
commit 8c6cee4e4d
1 changed files with 8 additions and 6 deletions
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14
libraries/audio-client/src/AudioClient.cpp
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@ -1135,8 +1135,6 @@ void AudioClient::configureWebrtc() {
config.level_estimation.enabled = false; config.level_estimation.enabled = false;
_apm->ApplyConfig(config); _apm->ApplyConfig(config);
qCDebug(audioclient) << "WebRTC enabled for acoustic echo cancellation.";
} }
// rebuffer into 10ms chunks // rebuffer into 10ms chunks
@ -1207,7 +1205,7 @@ void AudioClient::processWebrtcNearEnd(int16_t* samples, int numFrames, int numC
// process one chunk // process one chunk
int error = _apm->ProcessStream(buffers, streamConfig, streamConfig, buffers); int error = _apm->ProcessStream(buffers, streamConfig, streamConfig, buffers);
if (error =! _apm->kNoError) { if (error != _apm->kNoError) {
qCWarning(audioclient) << "WebRTC ProcessStream() returned ERROR:" << error; qCWarning(audioclient) << "WebRTC ProcessStream() returned ERROR:" << error;
} else { } else {
// modify samples in-place // modify samples in-place
@ -1396,8 +1394,10 @@ void AudioClient::handleMicAudioInput() {
_inputRingBuffer.readSamples(inputAudioSamples.get(), inputSamplesRequired); _inputRingBuffer.readSamples(inputAudioSamples.get(), inputSamplesRequired);
#if defined(WEBRTC_ENABLED) #if defined(WEBRTC_ENABLED)
processWebrtcNearEnd(inputAudioSamples.get(), inputSamplesRequired / _inputFormat.channelCount(), if (_isAECEnabled) {
_inputFormat.channelCount(), _inputFormat.sampleRate()); processWebrtcNearEnd(inputAudioSamples.get(), inputSamplesRequired / _inputFormat.channelCount(),
_inputFormat.channelCount(), _inputFormat.sampleRate());
}
#endif #endif
// detect loudness and clipping on the raw input // detect loudness and clipping on the raw input
@ -2326,7 +2326,9 @@ qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) {
_audio->_audioLimiter.render(mixBuffer, scratchBuffer, framesPopped); _audio->_audioLimiter.render(mixBuffer, scratchBuffer, framesPopped);
#if defined(WEBRTC_ENABLED) #if defined(WEBRTC_ENABLED)
_audio->processWebrtcFarEnd(scratchBuffer, framesPopped, OUTPUT_CHANNEL_COUNT, _audio->_outputFormat.sampleRate()); if (_audio->_isAECEnabled) {
_audio->processWebrtcFarEnd(scratchBuffer, framesPopped, OUTPUT_CHANNEL_COUNT, _audio->_outputFormat.sampleRate());
}
#endif #endif
// if required, upmix or downmix to deviceChannelCount // if required, upmix or downmix to deviceChannelCount