tg2sip/EchoCanceller.cpp

241 lines
5.9 KiB
C++
Executable File

//
// 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.
//
#ifndef TGVOIP_NO_DSP
#include "webrtc_dsp/modules/audio_processing/include/audio_processing.h"
#include "webrtc_dsp/api/audio/audio_frame.h"
#endif
#include "EchoCanceller.h"
#include "audio/AudioOutput.h"
#include "audio/AudioInput.h"
#include "logging.h"
#include "VoIPServerConfig.h"
#include <string.h>
#include <stdio.h>
#include <math.h>
using namespace tgvoip;
EchoCanceller::EchoCanceller(bool enableAEC, bool enableNS, bool enableAGC){
#ifndef TGVOIP_NO_DSP
this->enableAEC=enableAEC;
this->enableAGC=enableAGC;
this->enableNS=enableNS;
isOn=true;
webrtc::Config extraConfig;
#ifdef TGVOIP_USE_DESKTOP_DSP
extraConfig.Set(new webrtc::DelayAgnostic(true));
#endif
apm=webrtc::AudioProcessingBuilder().Create(extraConfig);
webrtc::AudioProcessing::Config config;
config.echo_canceller.enabled = enableAEC;
#ifndef TGVOIP_USE_DESKTOP_DSP
config.echo_canceller.mobile_mode = true;
#else
config.echo_canceller.mobile_mode = false;
#endif
config.high_pass_filter.enabled = enableAEC;
config.gain_controller2.enabled = enableAGC;
apm->ApplyConfig(config);
webrtc::NoiseSuppression::Level nsLevel;
#ifdef __APPLE__
switch(ServerConfig::GetSharedInstance()->GetInt("webrtc_ns_level_vpio", 0)){
#else
switch(ServerConfig::GetSharedInstance()->GetInt("webrtc_ns_level", 2)){
#endif
case 0:
nsLevel=webrtc::NoiseSuppression::Level::kLow;
break;
case 1:
nsLevel=webrtc::NoiseSuppression::Level::kModerate;
break;
case 3:
nsLevel=webrtc::NoiseSuppression::Level::kVeryHigh;
break;
case 2:
default:
nsLevel=webrtc::NoiseSuppression::Level::kHigh;
break;
}
apm->noise_suppression()->set_level(nsLevel);
apm->noise_suppression()->Enable(enableNS);
if(enableAGC){
apm->gain_control()->set_mode(webrtc::GainControl::Mode::kAdaptiveDigital);
apm->gain_control()->set_target_level_dbfs(ServerConfig::GetSharedInstance()->GetInt("webrtc_agc_target_level", 9));
apm->gain_control()->enable_limiter(ServerConfig::GetSharedInstance()->GetBoolean("webrtc_agc_enable_limiter", true));
apm->gain_control()->set_compression_gain_db(ServerConfig::GetSharedInstance()->GetInt("webrtc_agc_compression_gain", 20));
}
apm->voice_detection()->set_likelihood(webrtc::VoiceDetection::Likelihood::kVeryLowLikelihood);
audioFrame=new webrtc::AudioFrame();
audioFrame->samples_per_channel_=480;
audioFrame->sample_rate_hz_=48000;
audioFrame->num_channels_=1;
farendQueue=new BlockingQueue<int16_t*>(11);
farendBufferPool=new BufferPool(960*2, 10);
running=true;
bufferFarendThread=new Thread(std::bind(&EchoCanceller::RunBufferFarendThread, this));
bufferFarendThread->Start();
#else
this->enableAEC=this->enableAGC=enableAGC=this->enableNS=enableNS=false;
isOn=true;
#endif
}
EchoCanceller::~EchoCanceller(){
#ifndef TGVOIP_NO_DSP
delete apm;
delete audioFrame;
#endif
}
void EchoCanceller::Start(){
}
void EchoCanceller::Stop(){
}
void EchoCanceller::SpeakerOutCallback(unsigned char* data, size_t len){
if(len!=960*2 || !enableAEC || !isOn)
return;
#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(){
webrtc::AudioFrame frame;
frame.num_channels_=1;
frame.sample_rate_hz_=48000;
frame.samples_per_channel_=480;
while(running){
int16_t* samplesIn=farendQueue->GetBlocking();
if(samplesIn){
memcpy(frame.mutable_data(), samplesIn, 480*2);
apm->ProcessReverseStream(&frame);
memcpy(frame.mutable_data(), samplesIn+480, 480*2);
apm->ProcessReverseStream(&frame);
didBufferFarend=true;
farendBufferPool->Reuse(reinterpret_cast<unsigned char*>(samplesIn));
}
}
}
#endif
void EchoCanceller::Enable(bool enabled){
isOn=enabled;
}
void EchoCanceller::ProcessInput(int16_t* inOut, size_t numSamples, bool& hasVoice){
#ifndef TGVOIP_NO_DSP
if(!isOn || (!enableAEC && !enableAGC && !enableNS)){
return;
}
int delay=audio::AudioInput::GetEstimatedDelay()+audio::AudioOutput::GetEstimatedDelay();
assert(numSamples==960);
memcpy(audioFrame->mutable_data(), inOut, 480*2);
if(enableAEC)
apm->set_stream_delay_ms(delay);
apm->ProcessStream(audioFrame);
if(enableVAD)
hasVoice=apm->voice_detection()->stream_has_voice();
memcpy(inOut, audioFrame->data(), 480*2);
memcpy(audioFrame->mutable_data(), inOut+480, 480*2);
if(enableAEC)
apm->set_stream_delay_ms(delay);
apm->ProcessStream(audioFrame);
if(enableVAD){
hasVoice=hasVoice || apm->voice_detection()->stream_has_voice();
}
memcpy(inOut+480, audioFrame->data(), 480*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
}
void EchoCanceller::SetVoiceDetectionEnabled(bool enabled){
enableVAD=enabled;
#ifndef TGVOIP_NO_DSP
apm->voice_detection()->Enable(enabled);
#endif
}
using namespace tgvoip::effects;
AudioEffect::~AudioEffect(){
}
void AudioEffect::SetPassThrough(bool passThrough){
this->passThrough=passThrough;
}
Volume::Volume(){
}
Volume::~Volume(){
}
void Volume::Process(int16_t* inOut, size_t numSamples){
if(level==1.0f || passThrough){
return;
}
for(size_t i=0;i<numSamples;i++){
float sample=(float)inOut[i]*multiplier;
if(sample>32767.0f)
inOut[i]=INT16_MAX;
else if(sample<-32768.0f)
inOut[i]=INT16_MIN;
else
inOut[i]=(int16_t)sample;
}
}
void Volume::SetLevel(float level){
this->level=level;
float db;
if(level<1.0f)
db=-50.0f*(1.0f-level);
else if(level>1.0f && level<=2.0f)
db=10.0f*(level-1.0f);
else
db=0.0f;
multiplier=expf(db/20.0f * logf(10.0f));
}
float Volume::GetLevel(){
return level;
}