EchoCanceller.cpp

//
// 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
}