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Updated WebRTC APM

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I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
This commit is contained in:
Grishka
2018-11-21 21:22:31 +03:00
parent cc0cf35725
commit 5caaaafa42
644 changed files with 90797 additions and 9287 deletions
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31
OpusEncoder.cpp
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@@ -30,8 +30,8 @@ tgvoip::OpusEncoder::OpusEncoder(MediaStreamItf *source, bool needSecondary):que
complexity=10;
frameDuration=20;
levelMeter=NULL;
mediumCorrectionBitrate=ServerConfig::GetSharedInstance()->GetInt("audio_medium_fec_bitrate", 10000);
strongCorrectionBitrate=ServerConfig::GetSharedInstance()->GetInt("audio_strong_fec_bitrate", 8000);
mediumCorrectionBitrate=static_cast<uint32_t>(ServerConfig::GetSharedInstance()->GetInt("audio_medium_fec_bitrate", 10000));
strongCorrectionBitrate=static_cast<uint32_t>(ServerConfig::GetSharedInstance()->GetInt("audio_strong_fec_bitrate", 8000));
mediumCorrectionMultiplier=ServerConfig::GetSharedInstance()->GetDouble("audio_medium_fec_multiplier", 1.5);
strongCorrectionMultiplier=ServerConfig::GetSharedInstance()->GetDouble("audio_strong_fec_multiplier", 2.0);
secondaryEncoderEnabled=false;
@@ -80,15 +80,15 @@ void tgvoip::OpusEncoder::SetBitrate(uint32_t bitrate){
requestedBitrate=bitrate;
}
void tgvoip::OpusEncoder::Encode(unsigned char *data, size_t len){
void tgvoip::OpusEncoder::Encode(int16_t* data, size_t len){
if(requestedBitrate!=currentBitrate){
opus_encoder_ctl(enc, OPUS_SET_BITRATE(requestedBitrate));
currentBitrate=requestedBitrate;
LOGV("opus_encoder: setting bitrate to %u", currentBitrate);
}
if(levelMeter)
levelMeter->Update(reinterpret_cast<int16_t *>(data), len/2);
int32_t r=opus_encode(enc, (int16_t*)data, len/2, buffer, 4096);
levelMeter->Update(data, len);
int32_t r=opus_encode(enc, data, static_cast<int>(len), buffer, 4096);
if(r<=0){
LOGE("Error encoding: %d", r);
}else if(r==1){
@@ -98,7 +98,7 @@ void tgvoip::OpusEncoder::Encode(unsigned char *data, size_t len){
int32_t secondaryLen=0;
unsigned char secondaryBuffer[128];
if(secondaryEncoderEnabled && secondaryEncoder){
secondaryLen=opus_encode(secondaryEncoder, (int16_t*)data, len/2, secondaryBuffer, sizeof(secondaryBuffer));
secondaryLen=opus_encode(secondaryEncoder, data, static_cast<int>(len), secondaryBuffer, sizeof(secondaryBuffer));
//LOGV("secondaryLen %d", secondaryLen);
}
InvokeCallback(buffer, (size_t)r, secondaryBuffer, (size_t)secondaryLen);
@@ -132,33 +132,30 @@ void tgvoip::OpusEncoder::SetEchoCanceller(EchoCanceller* aec){
}
void tgvoip::OpusEncoder::RunThread(){
unsigned char buf[960*2];
uint32_t bufferedCount=0;
uint32_t packetsPerFrame=frameDuration/20;
LOGV("starting encoder, packets per frame=%d", packetsPerFrame);
unsigned char* frame;
int16_t* frame;
if(packetsPerFrame>1)
frame=(unsigned char *) malloc(960*2*packetsPerFrame);
frame=(int16_t*) malloc(960*2*packetsPerFrame);
else
frame=NULL;
while(running){
unsigned char* packet=(unsigned char*)queue.GetBlocking();
int16_t* packet=(int16_t*)queue.GetBlocking();
if(packet){
if(echoCanceller)
echoCanceller->ProcessInput(packet, buf, 960*2);
else
memcpy(buf, packet, 960*2);
echoCanceller->ProcessInput(packet, 960);
if(packetsPerFrame==1){
Encode(buf, 960*2);
Encode(packet, 960);
}else{
memcpy(frame+(960*2*bufferedCount), buf, 960*2);
memcpy(frame+(960*bufferedCount), packet, 960*2);
bufferedCount++;
if(bufferedCount==packetsPerFrame){
Encode(frame, 960*2*packetsPerFrame);
Encode(frame, 960*packetsPerFrame);
bufferedCount=0;
}
}
bufferPool.Reuse(packet);
bufferPool.Reuse(reinterpret_cast<unsigned char *>(packet));
}
}
if(frame)