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Move WebRTC far-end FIFO into AudioClient.h

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This commit is contained in:
Ken Cooke 2019-07-26 12:04:00 -07:00 committed by Seth Alves
parent 8ec4505b95
commit 5c86c22076
2 changed files with 13 additions and 13 deletions
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19
libraries/audio-client/src/AudioClient.cpp
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@ -1092,9 +1092,6 @@ void AudioClient::setReverbOptions(const AudioEffectOptions* options) {
#if defined(WEBRTC_ENABLED)
static const int WEBRTC_FRAMES_MAX = webrtc::AudioProcessing::kChunkSizeMs * webrtc::AudioProcessing::kMaxNativeSampleRateHz / 1000;
static const int WEBRTC_CHANNELS_MAX = 2;
static void deinterleaveToFloat(const int16_t* src, float* const* dst, int numFrames, int numChannels) {
for (int i = 0; i < numFrames; i++) {
for (int ch = 0; ch < numChannels; ch++) {
@ -1145,10 +1142,6 @@ void AudioClient::configureWebrtc() {
// rebuffer into 10ms chunks
void AudioClient::processWebrtcFarEnd(const int16_t* samples, int numFrames, int numChannels, int sampleRate) {
// TODO: move to AudioClient.h
static int16_t _fifo[WEBRTC_CHANNELS_MAX * WEBRTC_FRAMES_MAX];
static int _numFifo = 0; // numFrames saved in fifo
const webrtc::StreamConfig streamConfig = webrtc::StreamConfig(sampleRate, numChannels);
const int numChunk = (int)streamConfig.num_frames();
@ -1164,26 +1157,26 @@ void AudioClient::processWebrtcFarEnd(const int16_t* samples, int numFrames, int
while (numFrames > 0) {
// number of frames to fill
int numFill = std::min(numFrames, numChunk - _numFifo);
int numFill = std::min(numFrames, numChunk - _numFifoFarEnd);
// refill fifo
memcpy(&_fifo[_numFifo], samples, numFill * numChannels * sizeof(int16_t));
memcpy(&_fifoFarEnd[_numFifoFarEnd], samples, numFill * numChannels * sizeof(int16_t));
samples += numFill * numChannels;
numFrames -= numFill;
_numFifo += numFill;
_numFifoFarEnd += numFill;
if (_numFifo == numChunk) {
if (_numFifoFarEnd == numChunk) {
// convert audio format
float buffer[WEBRTC_CHANNELS_MAX][WEBRTC_FRAMES_MAX];
float* const buffers[WEBRTC_CHANNELS_MAX] = { buffer[0], buffer[1] };
deinterleaveToFloat(_fifo, buffers, numChunk, numChannels);
deinterleaveToFloat(_fifoFarEnd, buffers, numChunk, numChannels);
// process one chunk
if (_apm->kNoError != _apm->ProcessReverseStream(buffers, streamConfig, streamConfig, buffers)) {
qCWarning(audioclient) << "WebRTC ProcessReverseStream() returned an ERROR.";
}
_numFifo = 0;
_numFifoFarEnd = 0;
}
}
}

7
libraries/audio-client/src/AudioClient.h
View file

@ -423,7 +423,14 @@ private:
void handleLocalEchoAndReverb(QByteArray& inputByteArray);
#if defined(WEBRTC_ENABLED)
static const int WEBRTC_FRAMES_MAX = webrtc::AudioProcessing::kChunkSizeMs * webrtc::AudioProcessing::kMaxNativeSampleRateHz / 1000;
static const int WEBRTC_CHANNELS_MAX = 2;
webrtc::AudioProcessing* _apm { nullptr };
int16_t _fifoFarEnd[WEBRTC_CHANNELS_MAX * WEBRTC_FRAMES_MAX] {};
int _numFifoFarEnd = 0; // numFrames saved in fifo
void configureWebrtc();
void processWebrtcFarEnd(const int16_t* samples, int numFrames, int numChannels, int sampleRate);
void processWebrtcNearEnd(int16_t* samples, int numFrames, int numChannels, int sampleRate);