// // 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 #include #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) (xmin ? 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){ this->enableAEC=enableAEC; this->enableAGC=enableAGC; this->enableNS=enableNS; #ifndef TGVOIP_NO_DSP 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){ init_mutex(aecMutex); #ifndef TGVOIP_USE_DESKTOP_DSP aec=WebRtcAecm_Create(); WebRtcAecm_Init(aec, 16000); AecmConfig cfg; cfg.cngMode=AecmFalse; cfg.echoMode=1; 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(11); farendBufferPool=new BufferPool(960*2, 10); running=true; start_thread(bufferFarendThread, EchoCanceller::StartBufferFarendThread, this); } if(enableNS){ //#ifndef TGVOIP_USE_DESKTOP_DSP ns=WebRtcNsx_Create(); WebRtcNsx_Init((NsxHandle*)ns, 48000); WebRtcNsx_set_policy((NsxHandle*)ns, 2); /*#else ns=WebRtcNs_Create(); WebRtcNs_Init((NsHandle*)ns, 48000); WebRtcNs_set_policy((NsHandle*)ns, 1); #endif*/ } if(enableAGC){ agc=WebRtcAgc_Create(); WebRtcAgcConfig agcConfig; agcConfig.compressionGaindB = 9; agcConfig.limiterEnable = 1; #ifndef TGVOIP_USE_DESKTOP_DSP agcConfig.targetLevelDbfs = 3; #else agcConfig.targetLevelDbfs = 9; #endif WebRtcAgc_Init(agc, 0, 255, kAgcModeAdaptiveAnalog, 48000); WebRtcAgc_set_config(agc, agcConfig); agcMicLevel=128; } /*state=webrtc::WebRtcAec_Create(); webrtc::WebRtcAec_Init(state, 16000, 16000); webrtc::WebRtcAec_enable_delay_agnostic(webrtc::WebRtcAec_aec_core(state), 1);*/ #endif } EchoCanceller::~EchoCanceller(){ if(enableAEC){ running=false; farendQueue->Put(NULL); join_thread(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; if (this->enableAEC) { free_mutex(aecMutex); } } void EchoCanceller::Start(){ } void EchoCanceller::Stop(){ } void EchoCanceller::SpeakerOutCallback(unsigned char* data, size_t len){ if(len!=960*2 || !enableAEC) return; /*size_t offset=0; while(offsetGet(); if(buf){ memcpy(buf, data, 960*2); farendQueue->Put(buf); } } void *EchoCanceller::StartBufferFarendThread(void *arg){ ((EchoCanceller*)arg)->RunBufferFarendThread(); return NULL; } void EchoCanceller::RunBufferFarendThread(){ while(running){ int16_t* samplesIn=(int16_t *) 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); lock_mutex(aecMutex); #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 unlock_mutex(aecMutex); didBufferFarend=true; } } } void EchoCanceller::Enable(bool enabled){ //isOn=enabled; } void EchoCanceller::ProcessInput(unsigned char* data, unsigned char* out, size_t len){ int i; if(!enableAEC && !enableAGC && !enableNS){ memcpy(out, data, len); return; } 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); 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); } #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); lock_mutex(aecMutex); 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())); unlock_mutex(aecMutex); memcpy(bufOut->ibuf()->bands(0)[0], samplesOut, 320*2); }else if(enableAEC){ lock_mutex(aecMutex); 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())); unlock_mutex(aecMutex); 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); 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 ((webrtc::SplittingFilter*)splittingFilter)->Synthesis(bufOut, bufIn); memcpy(samplesOut, bufIn->ibuf_const()->bands(0)[0], 960*2); }