mirror of
https://github.com/danog/libtgvoip.git
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365 lines
11 KiB
C++
365 lines
11 KiB
C++
//
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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 "EchoCanceller.h"
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#include "audio/AudioOutput.h"
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#include "audio/AudioInput.h"
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#include "logging.h"
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#include <string.h>
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#include <stdio.h>
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#ifndef TGVOIP_NO_DSP
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#ifndef TGVOIP_USE_DESKTOP_DSP
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#include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
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#include "webrtc/modules/audio_processing/ns/noise_suppression_x.h"
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#else
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#include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
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//#include "webrtc/modules/audio_processing/ns/noise_suppression.h"
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#include "webrtc/modules/audio_processing/ns/noise_suppression_x.h"
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#endif
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#include "webrtc/modules/audio_processing/splitting_filter.h"
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#include "webrtc/common_audio/channel_buffer.h"
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#include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"
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#endif
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#define AEC_FRAME_SIZE 160
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#define OFFSET_STEP AEC_FRAME_SIZE*2
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//#define CLAMP(x, min, max) (x<max ? (x>min ? x : min) : max)
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#define CLAMP(x, min, max) x
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using namespace tgvoip;
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#ifdef TGVOIP_USE_DESKTOP_DSP
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namespace webrtc{
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void WebRtcAec_enable_delay_agnostic(AecCore* self, int enable);
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}
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#endif
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EchoCanceller::EchoCanceller(bool enableAEC, bool enableNS, bool enableAGC){
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this->enableAEC=enableAEC;
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this->enableAGC=enableAGC;
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this->enableNS=enableNS;
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#ifndef TGVOIP_NO_DSP
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splittingFilter=new webrtc::SplittingFilter(1, 3, 960);
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splittingFilterFarend=new webrtc::SplittingFilter(1, 3, 960);
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splittingFilterIn=new webrtc::IFChannelBuffer(960, 1, 1);
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splittingFilterFarendIn=new webrtc::IFChannelBuffer(960, 1, 1);
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splittingFilterOut=new webrtc::IFChannelBuffer(960, 1, 3);
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splittingFilterFarendOut=new webrtc::IFChannelBuffer(960, 1, 3);
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if(enableAEC){
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init_mutex(aecMutex);
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#ifndef TGVOIP_USE_DESKTOP_DSP
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aec=WebRtcAecm_Create();
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WebRtcAecm_Init(aec, 16000);
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AecmConfig cfg;
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cfg.cngMode=AecmFalse;
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cfg.echoMode=1;
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WebRtcAecm_set_config(aec, cfg);
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#else
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aec=webrtc::WebRtcAec_Create();
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webrtc::WebRtcAec_Init(aec, 48000, 48000);
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//webrtc::WebRtcAec_enable_delay_agnostic(webrtc::WebRtcAec_aec_core(aec), 1);
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webrtc::AecConfig config;
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config.metricsMode=webrtc::kAecFalse;
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config.nlpMode=webrtc::kAecNlpAggressive;
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config.skewMode=webrtc::kAecFalse;
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config.delay_logging=webrtc::kAecFalse;
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webrtc::WebRtcAec_set_config(aec, config);
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#endif
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farendQueue=new BlockingQueue<int16_t*>(11);
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farendBufferPool=new BufferPool(960*2, 10);
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running=true;
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start_thread(bufferFarendThread, EchoCanceller::StartBufferFarendThread, this);
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}
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if(enableNS){
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//#ifndef TGVOIP_USE_DESKTOP_DSP
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ns=WebRtcNsx_Create();
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WebRtcNsx_Init((NsxHandle*)ns, 48000);
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WebRtcNsx_set_policy((NsxHandle*)ns, 2);
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/*#else
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ns=WebRtcNs_Create();
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WebRtcNs_Init((NsHandle*)ns, 48000);
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WebRtcNs_set_policy((NsHandle*)ns, 1);
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#endif*/
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}
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if(enableAGC){
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agc=WebRtcAgc_Create();
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WebRtcAgcConfig agcConfig;
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agcConfig.compressionGaindB = 9;
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agcConfig.limiterEnable = 1;
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#ifndef TGVOIP_USE_DESKTOP_DSP
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agcConfig.targetLevelDbfs = 3;
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#else
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agcConfig.targetLevelDbfs = 9;
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#endif
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WebRtcAgc_Init(agc, 0, 255, kAgcModeAdaptiveAnalog, 48000);
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WebRtcAgc_set_config(agc, agcConfig);
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agcMicLevel=128;
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}
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/*state=webrtc::WebRtcAec_Create();
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webrtc::WebRtcAec_Init(state, 16000, 16000);
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webrtc::WebRtcAec_enable_delay_agnostic(webrtc::WebRtcAec_aec_core(state), 1);*/
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#endif
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}
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EchoCanceller::~EchoCanceller(){
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if(enableAEC){
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running=false;
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farendQueue->Put(NULL);
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join_thread(bufferFarendThread);
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delete farendQueue;
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delete farendBufferPool;
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#ifndef TGVOIP_USE_DESKTOP_DSP
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WebRtcAecm_Free(aec);
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#else
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webrtc::WebRtcAec_Free(aec);
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#endif
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}
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if(enableNS){
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//#ifndef TGVOIP_USE_DESKTOP_DSP
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WebRtcNsx_Free((NsxHandle*)ns);
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/*#else
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WebRtcNs_Free((NsHandle*)ns);
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#endif*/
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}
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if(enableAGC){
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WebRtcAgc_Free(agc);
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}
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//webrtc::WebRtcAec_Free(state);
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delete (webrtc::SplittingFilter*)splittingFilter;
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delete (webrtc::SplittingFilter*)splittingFilterFarend;
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delete (webrtc::IFChannelBuffer*)splittingFilterIn;
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delete (webrtc::IFChannelBuffer*)splittingFilterOut;
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delete (webrtc::IFChannelBuffer*)splittingFilterFarendIn;
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delete (webrtc::IFChannelBuffer*)splittingFilterFarendOut;
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if (this->enableAEC) {
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free_mutex(aecMutex);
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}
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}
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void EchoCanceller::Start(){
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}
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void EchoCanceller::Stop(){
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}
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void EchoCanceller::SpeakerOutCallback(unsigned char* data, size_t len){
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if(len!=960*2 || !enableAEC)
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return;
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/*size_t offset=0;
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while(offset<len){
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WebRtcAecm_BufferFarend(state, (int16_t*)(data+offset), AEC_FRAME_SIZE);
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offset+=OFFSET_STEP;
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}*/
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int16_t* buf=(int16_t*)farendBufferPool->Get();
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if(buf){
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memcpy(buf, data, 960*2);
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farendQueue->Put(buf);
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}
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}
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void *EchoCanceller::StartBufferFarendThread(void *arg){
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((EchoCanceller*)arg)->RunBufferFarendThread();
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return NULL;
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}
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void EchoCanceller::RunBufferFarendThread(){
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while(running){
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int16_t* samplesIn=farendQueue->GetBlocking();
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if(samplesIn){
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webrtc::IFChannelBuffer* bufIn=(webrtc::IFChannelBuffer*) splittingFilterFarendIn;
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webrtc::IFChannelBuffer* bufOut=(webrtc::IFChannelBuffer*) splittingFilterFarendOut;
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memcpy(bufIn->ibuf()->bands(0)[0], samplesIn, 960*2);
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farendBufferPool->Reuse((unsigned char *) samplesIn);
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((webrtc::SplittingFilter*)splittingFilterFarend)->Analysis(bufIn, bufOut);
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lock_mutex(aecMutex);
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#ifndef TGVOIP_USE_DESKTOP_DSP
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WebRtcAecm_BufferFarend(aec, bufOut->ibuf_const()->bands(0)[0], 160);
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WebRtcAecm_BufferFarend(aec, bufOut->ibuf_const()->bands(0)[0]+160, 160);
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#else
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webrtc::WebRtcAec_BufferFarend(aec, bufOut->fbuf_const()->bands(0)[0], 160);
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webrtc::WebRtcAec_BufferFarend(aec, bufOut->fbuf_const()->bands(0)[0]+160, 160);
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#endif
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unlock_mutex(aecMutex);
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didBufferFarend=true;
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}
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}
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}
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void EchoCanceller::Enable(bool enabled){
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//isOn=enabled;
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}
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void EchoCanceller::ProcessInput(unsigned char* data, unsigned char* out, size_t len){
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int i;
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if(!enableAEC && !enableAGC && !enableNS){
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memcpy(out, data, len);
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return;
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}
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int16_t* samplesIn=(int16_t*)data;
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int16_t* samplesOut=(int16_t*)out;
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webrtc::IFChannelBuffer* bufIn=(webrtc::IFChannelBuffer*) splittingFilterIn;
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webrtc::IFChannelBuffer* bufOut=(webrtc::IFChannelBuffer*) splittingFilterOut;
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memcpy(bufIn->ibuf()->bands(0)[0], samplesIn, 960*2);
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((webrtc::SplittingFilter*)splittingFilter)->Analysis(bufIn, bufOut);
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if(enableAGC){
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int16_t _agcOut[3][320];
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int16_t* agcIn[3];
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int16_t* agcOut[3];
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for(i=0;i<3;i++){
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agcIn[i]=(int16_t*)bufOut->ibuf_const()->bands(0)[i];
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agcOut[i]=_agcOut[i];
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}
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uint8_t saturation;
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WebRtcAgc_AddMic(agc, agcIn, 3, 160);
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WebRtcAgc_Process(agc, (const int16_t *const *) agcIn, 3, 160, agcOut, agcMicLevel, &agcMicLevel, 0, &saturation);
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for(i=0;i<3;i++){
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agcOut[i]+=160;
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agcIn[i]+=160;
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}
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WebRtcAgc_AddMic(agc, agcIn, 3, 160);
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WebRtcAgc_Process(agc, (const int16_t *const *) agcIn, 3, 160, agcOut, agcMicLevel, &agcMicLevel, 0, &saturation);
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//LOGV("AGC mic level %d", agcMicLevel);
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memcpy(bufOut->ibuf()->bands(0)[0], _agcOut[0], 320*2);
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memcpy(bufOut->ibuf()->bands(0)[1], _agcOut[1], 320*2);
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memcpy(bufOut->ibuf()->bands(0)[2], _agcOut[2], 320*2);
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}
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#ifndef TGVOIP_USE_DESKTOP_DSP
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if(enableAEC && enableNS){
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int16_t _nsOut[3][320];
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int16_t* nsIn[3];
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int16_t* nsOut[3];
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for(i=0;i<3;i++){
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nsIn[i]=(int16_t*)bufOut->ibuf_const()->bands(0)[i];
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nsOut[i]=_nsOut[i];
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}
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WebRtcNsx_Process((NsxHandle*)ns, (const short *const *) nsIn, 3, nsOut);
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for(i=0;i<3;i++){
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nsOut[i]+=160;
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nsIn[i]+=160;
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}
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WebRtcNsx_Process((NsxHandle*)ns, (const short *const *) nsIn, 3, nsOut);
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memcpy(bufOut->ibuf()->bands(0)[1], _nsOut[1], 320*2*2);
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lock_mutex(aecMutex);
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WebRtcAecm_Process(aec, bufOut->ibuf()->bands(0)[0], _nsOut[0], samplesOut, AEC_FRAME_SIZE, (int16_t) tgvoip::audio::AudioOutput::GetEstimatedDelay());
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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()));
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unlock_mutex(aecMutex);
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memcpy(bufOut->ibuf()->bands(0)[0], samplesOut, 320*2);
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}else if(enableAEC){
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lock_mutex(aecMutex);
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WebRtcAecm_Process(aec, bufOut->ibuf()->bands(0)[0], NULL, samplesOut, AEC_FRAME_SIZE, (int16_t) tgvoip::audio::AudioOutput::GetEstimatedDelay());
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WebRtcAecm_Process(aec, bufOut->ibuf()->bands(0)[0]+160, NULL, samplesOut+160, AEC_FRAME_SIZE, (int16_t) (tgvoip::audio::AudioOutput::GetEstimatedDelay()+audio::AudioInput::GetEstimatedDelay()));
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unlock_mutex(aecMutex);
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memcpy(bufOut->ibuf()->bands(0)[0], samplesOut, 320*2);
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}else if(enableNS){
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int16_t _nsOut[3][320];
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int16_t* nsIn[3];
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int16_t* nsOut[3];
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for(i=0;i<3;i++){
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nsIn[i]=(int16_t*)bufOut->ibuf_const()->bands(0)[i];
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nsOut[i]=_nsOut[i];
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}
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WebRtcNsx_Process((NsxHandle*)ns, (const short *const *) nsIn, 3, nsOut);
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for(i=0;i<3;i++){
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nsOut[i]+=160;
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nsIn[i]+=160;
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}
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WebRtcNsx_Process((NsxHandle*)ns, (const short *const *) nsIn, 3, nsOut);
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memcpy(bufOut->ibuf()->bands(0)[0], _nsOut[0], 320*2);
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memcpy(bufOut->ibuf()->bands(0)[1], _nsOut[1], 320*2);
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memcpy(bufOut->ibuf()->bands(0)[2], _nsOut[2], 320*2);
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}
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#else
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/*if(enableNS){
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float _nsOut[3][320];
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const float* nsIn[3];
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float* nsOut[3];
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for(i=0;i<3;i++){
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nsIn[i]=bufOut->fbuf_const()->bands(0)[i];
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nsOut[i]=_nsOut[i];
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}
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WebRtcNs_Process((NsHandle*)ns, nsIn, 3, nsOut);
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for(i=0;i<3;i++){
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nsOut[i]+=160;
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nsIn[i]+=160;
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}
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WebRtcNs_Process((NsHandle*)ns, nsIn, 3, nsOut);
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memcpy(bufOut->fbuf()->bands(0)[0], _nsOut[0], 320*4);
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memcpy(bufOut->fbuf()->bands(0)[1], _nsOut[1], 320*4);
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memcpy(bufOut->fbuf()->bands(0)[2], _nsOut[2], 320*4);
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}*/
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if(enableNS){
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int16_t _nsOut[3][320];
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int16_t* nsIn[3];
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int16_t* nsOut[3];
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for(i=0;i<3;i++){
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nsIn[i]=(int16_t*)bufOut->ibuf_const()->bands(0)[i];
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nsOut[i]=_nsOut[i];
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}
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WebRtcNsx_Process((NsxHandle*)ns, (const short *const *)nsIn, 3, nsOut);
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for(i=0;i<3;i++){
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nsOut[i]+=160;
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nsIn[i]+=160;
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}
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WebRtcNsx_Process((NsxHandle*)ns, (const short *const *)nsIn, 3, nsOut);
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memcpy(bufOut->ibuf()->bands(0)[0], _nsOut[0], 320*2);
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memcpy(bufOut->ibuf()->bands(0)[1], _nsOut[1], 320*2);
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memcpy(bufOut->ibuf()->bands(0)[2], _nsOut[2], 320*2);
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}
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if(enableAEC){
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const float* aecIn[3];
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float* aecOut[3];
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float _aecOut[3][320];
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for(i=0;i<3;i++){
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aecIn[i]=bufOut->fbuf_const()->bands(0)[i];
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aecOut[i]=_aecOut[i];
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}
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webrtc::WebRtcAec_Process(aec, aecIn, 3, aecOut, AEC_FRAME_SIZE, audio::AudioOutput::GetEstimatedDelay()+audio::AudioInput::GetEstimatedDelay(), 0);
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for(i=0;i<3;i++){
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aecOut[i]+=160;
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aecIn[i]+=160;
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}
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webrtc::WebRtcAec_Process(aec, aecIn, 3, aecOut, AEC_FRAME_SIZE, audio::AudioOutput::GetEstimatedDelay()+audio::AudioInput::GetEstimatedDelay(), 0);
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memcpy(bufOut->fbuf()->bands(0)[0], _aecOut[0], 320*4);
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memcpy(bufOut->fbuf()->bands(0)[1], _aecOut[1], 320*4);
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memcpy(bufOut->fbuf()->bands(0)[2], _aecOut[2], 320*4);
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}
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#endif
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((webrtc::SplittingFilter*)splittingFilter)->Synthesis(bufOut, bufIn);
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memcpy(samplesOut, bufIn->ibuf_const()->bands(0)[0], 960*2);
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}
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