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mirror of https://github.com/danog/libtgvoip.git synced 2024-11-27 04:34:42 +01:00
libtgvoip/OpusEncoder.cpp
Grishka 3216b76349 v0.4.1
Better jitter buffer with packet rescaling
Tried to fix some issues on Linux (telegramdesktop/tdesktop#3413)
Fixes for Windows Phone
2017-05-21 17:50:23 +03:00

161 lines
4.4 KiB
C++

//
// libtgvoip is free and unencumbered public domain software.
// For more information, see http://unlicense.org or the UNLICENSE file
// you should have received with this source code distribution.
//
#include "OpusEncoder.h"
#include <assert.h>
#include "logging.h"
#include "VoIPServerConfig.h"
tgvoip::OpusEncoder::OpusEncoder(MediaStreamItf *source):queue(11), bufferPool(960*2, 10){
this->source=source;
source->SetCallback(tgvoip::OpusEncoder::Callback, this);
enc=opus_encoder_create(48000, 1, OPUS_APPLICATION_VOIP, NULL);
opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(10));
opus_encoder_ctl(enc, OPUS_SET_PACKET_LOSS_PERC(15));
opus_encoder_ctl(enc, OPUS_SET_INBAND_FEC(1));
opus_encoder_ctl(enc, OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE));
opus_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_FULLBAND));
requestedBitrate=32000;
currentBitrate=0;
running=false;
echoCanceller=NULL;
complexity=10;
frameDuration=20;
mediumCorrectionBitrate=ServerConfig::GetSharedInstance()->GetInt("audio_medium_fec_bitrate", 10000);
strongCorrectionBitrate=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);
}
tgvoip::OpusEncoder::~OpusEncoder(){
opus_encoder_destroy(enc);
}
void tgvoip::OpusEncoder::Start(){
if(running)
return;
running=true;
start_thread(thread, StartThread, this);
set_thread_priority(thread, get_thread_max_priority());
set_thread_name(thread, "opus_encoder");
}
void tgvoip::OpusEncoder::Stop(){
if(!running)
return;
running=false;
queue.Put(NULL);
join_thread(thread);
}
void tgvoip::OpusEncoder::SetBitrate(uint32_t bitrate){
requestedBitrate=bitrate;
}
void tgvoip::OpusEncoder::Encode(unsigned char *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);
}
int32_t r=opus_encode(enc, (int16_t*)data, len/2, buffer, 4096);
if(r<=0){
LOGE("Error encoding: %d", r);
}else if(r==1){
LOGW("DTX");
}else if(running){
//LOGV("Packet size = %d", r);
InvokeCallback(buffer, (size_t)r);
}
}
size_t tgvoip::OpusEncoder::Callback(unsigned char *data, size_t len, void* param){
OpusEncoder* e=(OpusEncoder*)param;
unsigned char* buf=e->bufferPool.Get();
if(buf){
assert(len==960*2);
memcpy(buf, data, 960*2);
e->queue.Put(buf);
}else{
LOGW("opus_encoder: no buffer slots left");
if(e->complexity>1){
e->complexity--;
opus_encoder_ctl(e->enc, OPUS_SET_COMPLEXITY(e->complexity));
}
}
return 0;
}
uint32_t tgvoip::OpusEncoder::GetBitrate(){
return requestedBitrate;
}
void tgvoip::OpusEncoder::SetEchoCanceller(EchoCanceller* aec){
echoCanceller=aec;
}
void* tgvoip::OpusEncoder::StartThread(void* arg){
((OpusEncoder*)arg)->RunThread();
return NULL;
}
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;
if(packetsPerFrame>1)
frame=(unsigned char *) malloc(960*2*packetsPerFrame);
else
frame=NULL;
while(running){
unsigned char* packet=(unsigned char*)queue.GetBlocking();
if(packet){
if(echoCanceller)
echoCanceller->ProcessInput(packet, buf, 960*2);
else
memcpy(buf, packet, 960*2);
if(packetsPerFrame==1){
Encode(buf, 960*2);
}else{
memcpy(frame+(960*2*bufferedCount), buf, 960*2);
bufferedCount++;
if(bufferedCount==packetsPerFrame){
Encode(frame, 960*2*packetsPerFrame);
bufferedCount=0;
}
}
bufferPool.Reuse(packet);
}
}
if(frame)
free(frame);
}
void tgvoip::OpusEncoder::SetOutputFrameDuration(uint32_t duration){
frameDuration=duration;
}
void tgvoip::OpusEncoder::SetPacketLoss(int percent){
packetLossPercent=percent;
double multiplier=1;
if(currentBitrate<=strongCorrectionBitrate)
multiplier=strongCorrectionMultiplier;
else if(currentBitrate<=mediumCorrectionBitrate)
multiplier=mediumCorrectionMultiplier;
opus_encoder_ctl(enc, OPUS_SET_PACKET_LOSS_PERC((int)(percent*multiplier)));
opus_encoder_ctl(enc, OPUS_SET_BANDWIDTH(percent>17 ? OPUS_AUTO : OPUS_BANDWIDTH_FULLBAND));
}
int tgvoip::OpusEncoder::GetPacketLoss(){
return packetLossPercent;
}