forks
/
tg2sip
1
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity
tg2sip/EchoCanceller.cpp

461 lines
14 KiB
C++
Executable File
Raw Blame History

//
// 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 "EchoCanceller.h"
#include "audio/AudioOutput.h"
#include "audio/AudioInput.h"
#include "logging.h"
#include <string.h>
#include <stdio.h>
#ifndef TGVOIP_NO_DSP
#ifndef TGVOIP_USE_DESKTOP_DSP
#include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
#include "webrtc/modules/audio_processing/ns/noise_suppression_x.h"
#else
#include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
//#include "webrtc/modules/audio_processing/ns/noise_suppression.h"
#include "webrtc/modules/audio_processing/ns/noise_suppression_x.h"
#endif
#include "webrtc/modules/audio_processing/splitting_filter.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"
#endif
#define AEC_FRAME_SIZE 160
#define OFFSET_STEP AEC_FRAME_SIZE*2
//#define CLAMP(x, min, max) (x<max ? (x>min ? x : min) : max)
#define CLAMP(x, min, max) x
using namespace tgvoip;
#ifdef TGVOIP_USE_DESKTOP_DSP
namespace webrtc{
void WebRtcAec_enable_delay_agnostic(AecCore* self, int enable);
}
#endif
EchoCanceller::EchoCanceller(bool enableAEC, bool enableNS, bool enableAGC){
#ifndef TGVOIP_NO_DSP
this->enableAEC=enableAEC;
this->enableAGC=enableAGC;
this->enableNS=enableNS;
isOn=true;
splittingFilter=new webrtc::SplittingFilter(1, 3, 960);
splittingFilterFarend=new webrtc::SplittingFilter(1, 3, 960);
splittingFilterIn=new webrtc::IFChannelBuffer(960, 1, 1);
splittingFilterFarendIn=new webrtc::IFChannelBuffer(960, 1, 1);
splittingFilterOut=new webrtc::IFChannelBuffer(960, 1, 3);
splittingFilterFarendOut=new webrtc::IFChannelBuffer(960, 1, 3);
if(enableAEC){
#ifndef TGVOIP_USE_DESKTOP_DSP
aec=WebRtcAecm_Create();
WebRtcAecm_Init(aec, 16000);
AecmConfig cfg;
cfg.cngMode=AecmFalse;
cfg.echoMode=0;
WebRtcAecm_set_config(aec, cfg);
#else
aec=webrtc::WebRtcAec_Create();
webrtc::WebRtcAec_Init(aec, 48000, 48000);
webrtc::WebRtcAec_enable_delay_agnostic(webrtc::WebRtcAec_aec_core(aec), 1);
webrtc::AecConfig config;
config.metricsMode=webrtc::kAecFalse;
config.nlpMode=webrtc::kAecNlpAggressive;
config.skewMode=webrtc::kAecFalse;
config.delay_logging=webrtc::kAecFalse;
webrtc::WebRtcAec_set_config(aec, config);
#endif
farendQueue=new BlockingQueue<int16_t*>(11);
farendBufferPool=new BufferPool(960*2, 10);
running=true;
bufferFarendThread=new Thread(new MethodPointer<EchoCanceller>(&EchoCanceller::RunBufferFarendThread, this), NULL);
bufferFarendThread->Start();
}else{
aec=NULL;
}
if(enableNS){
//#ifndef TGVOIP_USE_DESKTOP_DSP
ns=WebRtcNsx_Create();
WebRtcNsx_Init((NsxHandle*)ns, 48000);
WebRtcNsx_set_policy((NsxHandle*)ns, 0);
/*#else
ns=WebRtcNs_Create();
WebRtcNs_Init((NsHandle*)ns, 48000);
WebRtcNs_set_policy((NsHandle*)ns, 1);
#endif*/
}else{
ns=NULL;
}
if(enableAGC){
agc=WebRtcAgc_Create();
WebRtcAgcConfig agcConfig;
agcConfig.compressionGaindB = 20;
agcConfig.limiterEnable = 1;
agcConfig.targetLevelDbfs = 9;
WebRtcAgc_Init(agc, 0, 255, kAgcModeAdaptiveDigital, 48000);
WebRtcAgc_set_config(agc, agcConfig);
agcMicLevel=0;
}else{
agc=NULL;
}
#else
this->enableAEC=this->enableAGC=enableAGC=this->enableNS=enableNS=false;
isOn=true;
#endif
}
EchoCanceller::~EchoCanceller(){
#ifndef TGVOIP_NO_DSP
if(enableAEC){
running=false;
farendQueue->Put(NULL);
bufferFarendThread->Join();
delete bufferFarendThread;
delete farendQueue;
delete farendBufferPool;
#ifndef TGVOIP_USE_DESKTOP_DSP
WebRtcAecm_Free(aec);
#else
webrtc::WebRtcAec_Free(aec);
#endif
}
if(enableNS){
//#ifndef TGVOIP_USE_DESKTOP_DSP
WebRtcNsx_Free((NsxHandle*)ns);
/*#else
WebRtcNs_Free((NsHandle*)ns);
#endif*/
}
if(enableAGC){
WebRtcAgc_Free(agc);
}
//webrtc::WebRtcAec_Free(state);
delete (webrtc::SplittingFilter*)splittingFilter;
delete (webrtc::SplittingFilter*)splittingFilterFarend;
delete (webrtc::IFChannelBuffer*)splittingFilterIn;
delete (webrtc::IFChannelBuffer*)splittingFilterOut;
delete (webrtc::IFChannelBuffer*)splittingFilterFarendIn;
delete (webrtc::IFChannelBuffer*)splittingFilterFarendOut;
#endif
}
void EchoCanceller::Start(){
}
void EchoCanceller::Stop(){
}
void EchoCanceller::SpeakerOutCallback(unsigned char* data, size_t len){
if(len!=960*2 || !enableAEC || !isOn)
return;
/*size_t offset=0;
while(offset<len){
WebRtcAecm_BufferFarend(state, (int16_t*)(data+offset), AEC_FRAME_SIZE);
offset+=OFFSET_STEP;
}*/
#ifndef TGVOIP_NO_DSP
int16_t* buf=(int16_t*)farendBufferPool->Get();
if(buf){
memcpy(buf, data, 960*2);
farendQueue->Put(buf);
}
#endif
}
#ifndef TGVOIP_NO_DSP
void EchoCanceller::RunBufferFarendThread(void* arg){
while(running){
int16_t* samplesIn=farendQueue->GetBlocking();
if(samplesIn){
webrtc::IFChannelBuffer* bufIn=(webrtc::IFChannelBuffer*) splittingFilterFarendIn;
webrtc::IFChannelBuffer* bufOut=(webrtc::IFChannelBuffer*) splittingFilterFarendOut;
memcpy(bufIn->ibuf()->bands(0)[0], samplesIn, 960*2);
farendBufferPool->Reuse((unsigned char *) samplesIn);
((webrtc::SplittingFilter*)splittingFilterFarend)->Analysis(bufIn, bufOut);
aecMutex.Lock();
//outstandingFarendFrames++;
//LOGV("BufferFarend: %d frames", outstandingFarendFrames);
#ifndef TGVOIP_USE_DESKTOP_DSP
WebRtcAecm_BufferFarend(aec, bufOut->ibuf_const()->bands(0)[0], 160);
WebRtcAecm_BufferFarend(aec, bufOut->ibuf_const()->bands(0)[0]+160, 160);
#else
webrtc::WebRtcAec_BufferFarend(aec, bufOut->fbuf_const()->bands(0)[0], 160);
webrtc::WebRtcAec_BufferFarend(aec, bufOut->fbuf_const()->bands(0)[0]+160, 160);
#endif
aecMutex.Unlock();
didBufferFarend=true;
}
}
}
#endif
void EchoCanceller::Enable(bool enabled){
isOn=enabled;
}
void EchoCanceller::ProcessInput(unsigned char* data, unsigned char* out, size_t len){
int i;
if(!isOn || (!enableAEC && !enableAGC && !enableNS)){
memcpy(out, data, len);
return;
}
#ifndef TGVOIP_NO_DSP
int16_t* samplesIn=(int16_t*)data;
int16_t* samplesOut=(int16_t*)out;
webrtc::IFChannelBuffer* bufIn=(webrtc::IFChannelBuffer*) splittingFilterIn;
webrtc::IFChannelBuffer* bufOut=(webrtc::IFChannelBuffer*) splittingFilterOut;
memcpy(bufIn->ibuf()->bands(0)[0], samplesIn, 960*2);
((webrtc::SplittingFilter*)splittingFilter)->Analysis(bufIn, bufOut);
#ifndef TGVOIP_USE_DESKTOP_DSP
if(enableAEC && enableNS){
int16_t _nsOut[3][320];
int16_t* nsIn[3];
int16_t* nsOut[3];
for(i=0;i<3;i++){
nsIn[i]=(int16_t*)bufOut->ibuf_const()->bands(0)[i];
nsOut[i]=_nsOut[i];
}
WebRtcNsx_Process((NsxHandle*)ns, (const short *const *) nsIn, 3, nsOut);
for(i=0;i<3;i++){
nsOut[i]+=160;
nsIn[i]+=160;
}
WebRtcNsx_Process((NsxHandle*)ns, (const short *const *) nsIn, 3, nsOut);
memcpy(bufOut->ibuf()->bands(0)[1], _nsOut[1], 320*2*2);
aecMutex.Lock();
WebRtcAecm_Process(aec, bufOut->ibuf()->bands(0)[0], _nsOut[0], samplesOut, AEC_FRAME_SIZE, (int16_t) tgvoip::audio::AudioOutput::GetEstimatedDelay());
WebRtcAecm_Process(aec, bufOut->ibuf()->bands(0)[0]+160, _nsOut[0]+160, samplesOut+160, AEC_FRAME_SIZE, (int16_t) (tgvoip::audio::AudioOutput::GetEstimatedDelay()+audio::AudioInput::GetEstimatedDelay()));
aecMutex.Unlock();
memcpy(bufOut->ibuf()->bands(0)[0], samplesOut, 320*2);
}else if(enableAEC){
aecMutex.Lock();
WebRtcAecm_Process(aec, bufOut->ibuf()->bands(0)[0], NULL, samplesOut, AEC_FRAME_SIZE, (int16_t) tgvoip::audio::AudioOutput::GetEstimatedDelay());
WebRtcAecm_Process(aec, bufOut->ibuf()->bands(0)[0]+160, NULL, samplesOut+160, AEC_FRAME_SIZE, (int16_t) (tgvoip::audio::AudioOutput::GetEstimatedDelay()+audio::AudioInput::GetEstimatedDelay()));
aecMutex.Unlock();
memcpy(bufOut->ibuf()->bands(0)[0], samplesOut, 320*2);
}else if(enableNS){
int16_t _nsOut[3][320];
int16_t* nsIn[3];
int16_t* nsOut[3];
for(i=0;i<3;i++){
nsIn[i]=(int16_t*)bufOut->ibuf_const()->bands(0)[i];
nsOut[i]=_nsOut[i];
}
WebRtcNsx_Process((NsxHandle*)ns, (const short *const *) nsIn, 3, nsOut);
for(i=0;i<3;i++){
nsOut[i]+=160;
nsIn[i]+=160;
}
WebRtcNsx_Process((NsxHandle*)ns, (const short *const *) nsIn, 3, nsOut);
memcpy(bufOut->ibuf()->bands(0)[0], _nsOut[0], 320*2);
memcpy(bufOut->ibuf()->bands(0)[1], _nsOut[1], 320*2);
memcpy(bufOut->ibuf()->bands(0)[2], _nsOut[2], 320*2);
}
#else
/*if(enableNS){
float _nsOut[3][320];
const float* nsIn[3];
float* nsOut[3];
for(i=0;i<3;i++){
nsIn[i]=bufOut->fbuf_const()->bands(0)[i];
nsOut[i]=_nsOut[i];
}
WebRtcNs_Process((NsHandle*)ns, nsIn, 3, nsOut);
for(i=0;i<3;i++){
nsOut[i]+=160;
nsIn[i]+=160;
}
WebRtcNs_Process((NsHandle*)ns, nsIn, 3, nsOut);
memcpy(bufOut->fbuf()->bands(0)[0], _nsOut[0], 320*4);
memcpy(bufOut->fbuf()->bands(0)[1], _nsOut[1], 320*4);
memcpy(bufOut->fbuf()->bands(0)[2], _nsOut[2], 320*4);
}*/
if(enableNS){
int16_t _nsOut[3][320];
int16_t* nsIn[3];
int16_t* nsOut[3];
for(i=0;i<3;i++){
nsIn[i]=(int16_t*)bufOut->ibuf_const()->bands(0)[i];
nsOut[i]=_nsOut[i];
}
WebRtcNsx_Process((NsxHandle*)ns, (const short *const *)nsIn, 3, nsOut);
for(i=0;i<3;i++){
nsOut[i]+=160;
nsIn[i]+=160;
}
WebRtcNsx_Process((NsxHandle*)ns, (const short *const *)nsIn, 3, nsOut);
memcpy(bufOut->ibuf()->bands(0)[0], _nsOut[0], 320*2);
memcpy(bufOut->ibuf()->bands(0)[1], _nsOut[1], 320*2);
memcpy(bufOut->ibuf()->bands(0)[2], _nsOut[2], 320*2);
}
if(enableAEC){
const float* aecIn[3];
float* aecOut[3];
float _aecOut[3][320];
for(i=0;i<3;i++){
aecIn[i]=bufOut->fbuf_const()->bands(0)[i];
aecOut[i]=_aecOut[i];
}
webrtc::WebRtcAec_Process(aec, aecIn, 3, aecOut, AEC_FRAME_SIZE, audio::AudioOutput::GetEstimatedDelay()+audio::AudioInput::GetEstimatedDelay(), 0);
for(i=0;i<3;i++){
aecOut[i]+=160;
aecIn[i]+=160;
}
webrtc::WebRtcAec_Process(aec, aecIn, 3, aecOut, AEC_FRAME_SIZE, audio::AudioOutput::GetEstimatedDelay()+audio::AudioInput::GetEstimatedDelay(), 0);
//outstandingFarendFrames--;
//LOGV("Process: %d frames", outstandingFarendFrames);
memcpy(bufOut->fbuf()->bands(0)[0], _aecOut[0], 320*4);
memcpy(bufOut->fbuf()->bands(0)[1], _aecOut[1], 320*4);
memcpy(bufOut->fbuf()->bands(0)[2], _aecOut[2], 320*4);
}
#endif
if(enableAGC){
int16_t _agcOut[3][320];
int16_t* agcIn[3];
int16_t* agcOut[3];
for(i=0;i<3;i++){
agcIn[i]=(int16_t*)bufOut->ibuf_const()->bands(0)[i];
agcOut[i]=_agcOut[i];
}
uint8_t saturation;
WebRtcAgc_AddMic(agc, agcIn, 3, 160);
WebRtcAgc_Process(agc, (const int16_t *const *) agcIn, 3, 160, agcOut, agcMicLevel, &agcMicLevel, 0, &saturation);
for(i=0;i<3;i++){
agcOut[i]+=160;
agcIn[i]+=160;
}
WebRtcAgc_AddMic(agc, agcIn, 3, 160);
WebRtcAgc_Process(agc, (const int16_t *const *) agcIn, 3, 160, agcOut, agcMicLevel, &agcMicLevel, 0, &saturation);
//LOGV("AGC mic level %d", agcMicLevel);
memcpy(bufOut->ibuf()->bands(0)[0], _agcOut[0], 320*2);
memcpy(bufOut->ibuf()->bands(0)[1], _agcOut[1], 320*2);
memcpy(bufOut->ibuf()->bands(0)[2], _agcOut[2], 320*2);
}
((webrtc::SplittingFilter*)splittingFilter)->Synthesis(bufOut, bufIn);
memcpy(samplesOut, bufIn->ibuf_const()->bands(0)[0], 960*2);
#endif
}
void EchoCanceller::SetAECStrength(int strength){
#ifndef TGVOIP_NO_DSP
if(aec){
#ifndef TGVOIP_USE_DESKTOP_DSP
AecmConfig cfg;
cfg.cngMode=AecmFalse;
cfg.echoMode=(int16_t) strength;
WebRtcAecm_set_config(aec, cfg);
#endif
}
#endif
}
AudioEffect::~AudioEffect(){
}
void AudioEffect::SetPassThrough(bool passThrough){
this->passThrough=passThrough;
}
AutomaticGainControl::AutomaticGainControl(){
#ifndef TGVOIP_NO_DSP
splittingFilter=new webrtc::SplittingFilter(1, 3, 960);
splittingFilterIn=new webrtc::IFChannelBuffer(960, 1, 1);
splittingFilterOut=new webrtc::IFChannelBuffer(960, 1, 3);
agc=WebRtcAgc_Create();
WebRtcAgcConfig agcConfig;
agcConfig.compressionGaindB = 9;
agcConfig.limiterEnable = 1;
agcConfig.targetLevelDbfs = 3;
WebRtcAgc_Init(agc, 0, 255, kAgcModeAdaptiveDigital, 48000);
WebRtcAgc_set_config(agc, agcConfig);
agcMicLevel=0;
#endif
}
AutomaticGainControl::~AutomaticGainControl(){
#ifndef TGVOIP_NO_DSP
delete (webrtc::SplittingFilter*)splittingFilter;
delete (webrtc::IFChannelBuffer*)splittingFilterIn;
delete (webrtc::IFChannelBuffer*)splittingFilterOut;
WebRtcAgc_Free(agc);
#endif
}
void AutomaticGainControl::Process(int16_t *inOut, size_t numSamples){
#ifndef TGVOIP_NO_DSP
if(passThrough)
return;
if(numSamples!=960){
LOGW("AutomaticGainControl only works on 960-sample buffers (got %u samples)", (unsigned int)numSamples);
return;
}
//LOGV("processing frame through AGC");
webrtc::IFChannelBuffer* bufIn=(webrtc::IFChannelBuffer*) splittingFilterIn;
webrtc::IFChannelBuffer* bufOut=(webrtc::IFChannelBuffer*) splittingFilterOut;
memcpy(bufIn->ibuf()->bands(0)[0], inOut, 960*2);
((webrtc::SplittingFilter*)splittingFilter)->Analysis(bufIn, bufOut);
int i;
int16_t _agcOut[3][320];
int16_t* agcIn[3];
int16_t* agcOut[3];
for(i=0;i<3;i++){
agcIn[i]=(int16_t*)bufOut->ibuf_const()->bands(0)[i];
agcOut[i]=_agcOut[i];
}
uint8_t saturation;
WebRtcAgc_AddMic(agc, agcIn, 3, 160);
WebRtcAgc_Process(agc, (const int16_t *const *) agcIn, 3, 160, agcOut, agcMicLevel, &agcMicLevel, 0, &saturation);
for(i=0;i<3;i++){
agcOut[i]+=160;
agcIn[i]+=160;
}
WebRtcAgc_AddMic(agc, agcIn, 3, 160);
WebRtcAgc_Process(agc, (const int16_t *const *) agcIn, 3, 160, agcOut, agcMicLevel, &agcMicLevel, 0, &saturation);
memcpy(bufOut->ibuf()->bands(0)[0], _agcOut[0], 320*2);
memcpy(bufOut->ibuf()->bands(0)[1], _agcOut[1], 320*2);
memcpy(bufOut->ibuf()->bands(0)[2], _agcOut[2], 320*2);
((webrtc::SplittingFilter*)splittingFilter)->Synthesis(bufOut, bufIn);
memcpy(inOut, bufIn->ibuf_const()->bands(0)[0], 960*2);
#endif
}
Reference in New Issue View Git Blame Copy Permalink