mirror of
https://github.com/NekoX-Dev/NekoX.git
synced 2024-12-16 04:19:26 +01:00
291 lines
8.0 KiB
C++
Executable File
291 lines
8.0 KiB
C++
Executable File
//
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// libtgvoip is free and unencumbered public domain software.
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// For more information, see http://unlicense.org or the UNLICENSE file
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// you should have received with this source code distribution.
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//
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#include "OpusDecoder.h"
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#include "audio/Resampler.h"
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#include "logging.h"
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#include <assert.h>
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#include <math.h>
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#include <algorithm>
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#ifdef HAVE_CONFIG_H
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#include <opus/opus.h>
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#else
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#include "opus.h"
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#endif
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#include "VoIPController.h"
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#define PACKET_SIZE (960*2)
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using namespace tgvoip;
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tgvoip::OpusDecoder::OpusDecoder(const std::shared_ptr<MediaStreamItf>& dst, bool isAsync, bool needEC){
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dst->SetCallback(OpusDecoder::Callback, this);
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Initialize(isAsync, needEC);
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}
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tgvoip::OpusDecoder::OpusDecoder(const std::unique_ptr<MediaStreamItf>& dst, bool isAsync, bool needEC){
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dst->SetCallback(OpusDecoder::Callback, this);
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Initialize(isAsync, needEC);
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}
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tgvoip::OpusDecoder::OpusDecoder(MediaStreamItf* dst, bool isAsync, bool needEC){
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dst->SetCallback(OpusDecoder::Callback, this);
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Initialize(isAsync, needEC);
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}
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void tgvoip::OpusDecoder::Initialize(bool isAsync, bool needEC){
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async=isAsync;
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if(async){
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decodedQueue=new BlockingQueue<unsigned char*>(33);
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bufferPool=new BufferPool(PACKET_SIZE, 32);
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semaphore=new Semaphore(32, 0);
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}else{
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decodedQueue=NULL;
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bufferPool=NULL;
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semaphore=NULL;
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}
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dec=opus_decoder_create(48000, 1, NULL);
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if(needEC)
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ecDec=opus_decoder_create(48000, 1, NULL);
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else
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ecDec=NULL;
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buffer=(unsigned char *) malloc(8192);
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lastDecoded=NULL;
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outputBufferSize=0;
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echoCanceller=NULL;
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frameDuration=20;
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consecutiveLostPackets=0;
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enableDTX=false;
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silentPacketCount=0;
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levelMeter=NULL;
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nextLen=0;
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running=false;
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remainingDataLen=0;
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processedBuffer=NULL;
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prevWasEC=false;
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prevLastSample=0;
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}
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tgvoip::OpusDecoder::~OpusDecoder(){
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opus_decoder_destroy(dec);
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if(ecDec)
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opus_decoder_destroy(ecDec);
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free(buffer);
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if(bufferPool)
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delete bufferPool;
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if(decodedQueue)
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delete decodedQueue;
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if(semaphore)
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delete semaphore;
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}
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void tgvoip::OpusDecoder::SetEchoCanceller(EchoCanceller* canceller){
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echoCanceller=canceller;
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}
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size_t tgvoip::OpusDecoder::Callback(unsigned char *data, size_t len, void *param){
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return ((OpusDecoder*)param)->HandleCallback(data, len);
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}
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size_t tgvoip::OpusDecoder::HandleCallback(unsigned char *data, size_t len){
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if(async){
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if(!running){
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memset(data, 0, len);
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return 0;
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}
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if(outputBufferSize==0){
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outputBufferSize=len;
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int packetsNeeded;
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if(len>PACKET_SIZE)
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packetsNeeded=len/PACKET_SIZE;
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else
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packetsNeeded=1;
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packetsNeeded*=2;
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semaphore->Release(packetsNeeded);
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}
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assert(outputBufferSize==len && "output buffer size is supposed to be the same throughout callbacks");
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if(len==PACKET_SIZE){
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lastDecoded=(unsigned char *) decodedQueue->GetBlocking();
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if(!lastDecoded)
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return 0;
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memcpy(data, lastDecoded, PACKET_SIZE);
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bufferPool->Reuse(lastDecoded);
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semaphore->Release();
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if(silentPacketCount>0){
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silentPacketCount--;
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if(levelMeter)
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levelMeter->Update(reinterpret_cast<int16_t *>(data), 0);
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return 0;
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}
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if(echoCanceller){
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echoCanceller->SpeakerOutCallback(data, PACKET_SIZE);
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}
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}else{
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LOGE("Opus decoder buffer length != 960 samples");
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abort();
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}
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}else{
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if(remainingDataLen==0 && silentPacketCount==0){
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int duration=DecodeNextFrame();
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remainingDataLen=(size_t) (duration/20*960*2);
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}
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if(silentPacketCount>0 || remainingDataLen==0 || !processedBuffer){
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if(silentPacketCount>0)
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silentPacketCount--;
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memset(data, 0, 960*2);
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if(levelMeter)
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levelMeter->Update(reinterpret_cast<int16_t *>(data), 0);
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return 0;
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}
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memcpy(data, processedBuffer, 960*2);
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remainingDataLen-=960*2;
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if(remainingDataLen>0){
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memmove(processedBuffer, processedBuffer+960*2, remainingDataLen);
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}
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}
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if(levelMeter)
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levelMeter->Update(reinterpret_cast<int16_t *>(data), len/2);
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return len;
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}
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void tgvoip::OpusDecoder::Start(){
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if(!async)
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return;
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running=true;
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thread=new Thread(std::bind(&tgvoip::OpusDecoder::RunThread, this));
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thread->SetName("opus_decoder");
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thread->SetMaxPriority();
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thread->Start();
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}
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void tgvoip::OpusDecoder::Stop(){
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if(!running || !async)
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return;
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running=false;
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semaphore->Release();
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thread->Join();
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delete thread;
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}
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void tgvoip::OpusDecoder::RunThread(){
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int i;
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LOGI("decoder: packets per frame %d", packetsPerFrame);
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while(running){
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int playbackDuration=DecodeNextFrame();
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for(i=0;i<playbackDuration/20;i++){
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semaphore->Acquire();
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if(!running){
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LOGI("==== decoder exiting ====");
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return;
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}
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unsigned char *buf=bufferPool->Get();
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if(buf){
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if(remainingDataLen>0){
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for(effects::AudioEffect*& effect:postProcEffects){
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effect->Process(reinterpret_cast<int16_t*>(processedBuffer+(PACKET_SIZE*i)), 960);
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}
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memcpy(buf, processedBuffer+(PACKET_SIZE*i), PACKET_SIZE);
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}else{
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//LOGE("Error decoding, result=%d", size);
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memset(buf, 0, PACKET_SIZE);
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}
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decodedQueue->Put(buf);
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}else{
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LOGW("decoder: no buffers left!");
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}
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}
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}
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}
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int tgvoip::OpusDecoder::DecodeNextFrame(){
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int playbackDuration=0;
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bool isEC=false;
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size_t len=jitterBuffer->HandleOutput(buffer, 8192, 0, true, playbackDuration, isEC);
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bool fec=false;
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if(!len){
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fec=true;
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len=jitterBuffer->HandleOutput(buffer, 8192, 0, false, playbackDuration, isEC);
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//if(len)
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// LOGV("Trying FEC...");
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}
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int size;
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if(len){
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size=opus_decode(isEC ? ecDec : dec, buffer, len, (opus_int16 *) decodeBuffer, packetsPerFrame*960, fec ? 1 : 0);
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consecutiveLostPackets=0;
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if(prevWasEC!=isEC && size){
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// It turns out the waveforms generated by the PLC feature are also great to help smooth out the
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// otherwise audible transition between the frames from different decoders. Those are basically an extrapolation
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// of the previous successfully decoded data -- which is exactly what we need here.
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size=opus_decode(prevWasEC ? ecDec : dec, NULL, 0, (opus_int16*)nextBuffer, packetsPerFrame*960, 0);
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if(size){
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int16_t* plcSamples=reinterpret_cast<int16_t*>(nextBuffer);
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int16_t* samples=reinterpret_cast<int16_t*>(decodeBuffer);
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constexpr float coeffs[]={0.999802, 0.995062, 0.984031, 0.966778, 0.943413, 0.914084, 0.878975, 0.838309, 0.792344,
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0.741368, 0.685706, 0.625708, 0.561754, 0.494249, 0.423619, 0.350311, 0.274788, 0.197527, 0.119018, 0.039757};
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for(int i=0;i<20;i++){
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samples[i]=(int16_t)round((plcSamples[i]*coeffs[i]+(float)samples[i]*(1.0-coeffs[i])));
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}
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}
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}
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prevWasEC=isEC;
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prevLastSample=decodeBuffer[size-1];
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}else{ // do packet loss concealment
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consecutiveLostPackets++;
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if(consecutiveLostPackets>2 && enableDTX){
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silentPacketCount+=packetsPerFrame;
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size=packetsPerFrame*960;
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}else{
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size=opus_decode(prevWasEC ? ecDec : dec, NULL, 0, (opus_int16 *) decodeBuffer, packetsPerFrame*960, 0);
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//LOGV("PLC");
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}
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}
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if(size<0)
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LOGW("decoder: opus_decode error %d", size);
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remainingDataLen=size;
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if(playbackDuration==80){
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processedBuffer=buffer;
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audio::Resampler::Rescale60To80((int16_t*) decodeBuffer, (int16_t*) processedBuffer);
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}else if(playbackDuration==40){
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processedBuffer=buffer;
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audio::Resampler::Rescale60To40((int16_t*) decodeBuffer, (int16_t*) processedBuffer);
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}else{
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processedBuffer=decodeBuffer;
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}
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return playbackDuration;
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}
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void tgvoip::OpusDecoder::SetFrameDuration(uint32_t duration){
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frameDuration=duration;
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packetsPerFrame=frameDuration/20;
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}
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void tgvoip::OpusDecoder::SetJitterBuffer(std::shared_ptr<JitterBuffer> jitterBuffer){
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this->jitterBuffer=jitterBuffer;
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}
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void tgvoip::OpusDecoder::SetDTX(bool enable){
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enableDTX=enable;
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}
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void tgvoip::OpusDecoder::SetLevelMeter(AudioLevelMeter *levelMeter){
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this->levelMeter=levelMeter;
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}
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void tgvoip::OpusDecoder::AddAudioEffect(effects::AudioEffect *effect){
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postProcEffects.push_back(effect);
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}
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void tgvoip::OpusDecoder::RemoveAudioEffect(effects::AudioEffect *effect){
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std::vector<effects::AudioEffect*>::iterator i=std::find(postProcEffects.begin(), postProcEffects.end(), effect);
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if(i!=postProcEffects.end())
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postProcEffects.erase(i);
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}
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