Merge remote-tracking branch 'upstream/public' into tdesktop

.gitignore

@@ -19,3 +19,5 @@ bld/
 
 # Visual Studio 2015 cache/options directory
 .vs/
+
+xcuserdata/

EchoCanceller.cpp

@@ -80,6 +80,8 @@ EchoCanceller::EchoCanceller(bool enableAEC, bool enableNS, bool enableAGC){
 		running=true;
 
 		start_thread(bufferFarendThread, EchoCanceller::StartBufferFarendThread, this);
+	}else{
+		aec=NULL;
 	}
 
 	if(enableNS){
@@ -92,6 +94,8 @@ EchoCanceller::EchoCanceller(bool enableAEC, bool enableNS, bool enableAGC){
 		WebRtcNs_Init((NsHandle*)ns, 48000);
 		WebRtcNs_set_policy((NsHandle*)ns, 1);
 #endif*/
+	}else{
+		ns=NULL;
 	}
 
 	if(enableAGC){
@@ -103,6 +107,8 @@ EchoCanceller::EchoCanceller(bool enableAEC, bool enableNS, bool enableAGC){
 		WebRtcAgc_Init(agc, 0, 255, kAgcModeAdaptiveDigital, 48000);
 		WebRtcAgc_set_config(agc, agcConfig);
 		agcMicLevel=0;
+	}else{
+		agc=NULL;
 	}
 #endif
 }
@@ -354,6 +360,17 @@ void EchoCanceller::ProcessInput(unsigned char* data, unsigned char* out, size_t
 	memcpy(samplesOut, bufIn->ibuf_const()->bands(0)[0], 960*2);
 }
 
+void EchoCanceller::SetAECStrength(int strength){
+	if(aec){
+#ifndef TGVOIP_USE_DESKTOP_DSP
+		AecmConfig cfg;
+		cfg.cngMode=AecmFalse;
+		cfg.echoMode=(int16_t) strength;
+		WebRtcAecm_set_config(aec, cfg);
+#endif
+	}
+}
+
 AudioEffect::~AudioEffect(){
 
 }

EchoCanceller.h

@@ -23,6 +23,7 @@ class EchoCanceller{
 	void SpeakerOutCallback(unsigned char* data, size_t len);
 	void Enable(bool enabled);
 	void ProcessInput(unsigned char* data, unsigned char* out, size_t len);
+	void SetAECStrength(int strength);
 
 private:
 	bool enableAEC;

VoIPController.cpp

@@ -232,6 +232,7 @@ VoIPController::VoIPController() : activeNetItfName(""),
 	udpSocket=NetworkSocket::Create(PROTO_UDP);
 	realUdpSocket=udpSocket;
 	udpConnectivityState=UDP_UNKNOWN;
+	echoCancellationStrength=1;
 
 	outputAGC=NULL;
 	outputAGCEnabled=false;
@@ -1147,6 +1148,7 @@ simpleAudioBlock random_id:long random_bytes:string raw_data:string = DecryptedA
 					audioOutput=tgvoip::audio::AudioOutput::Create(currentAudioOutput);
 					audioOutput->Configure(48000, 16, 1);
 					echoCanceller=new EchoCanceller(config.enableAEC, config.enableNS, config.enableAGC);
+					echoCanceller->SetAECStrength(echoCancellationStrength);
 					encoder=new OpusEncoder(audioInput);
 					encoder->SetCallback(AudioInputCallback, this);
 					encoder->SetOutputFrameDuration(outgoingAudioStream->frameDuration);
@@ -2472,6 +2474,12 @@ void VoIPController::SetAudioOutputGainControlEnabled(bool enabled){
 		outputAGC->SetPassThrough(!enabled);
 }
 
+void VoIPController::SetEchoCancellationStrength(int strength){
+	echoCancellationStrength=strength;
+	if(echoCanceller)
+		echoCanceller->SetAECStrength(strength);
+}
+
 Endpoint::Endpoint(int64_t id, uint16_t port, IPv4Address& _address, IPv6Address& _v6address, char type, unsigned char peerTag[16]) : address(_address), v6address(_v6address){
 	this->id=id;
 	this->port=port;

VoIPController.h

@@ -29,7 +29,7 @@
 #include "CongestionControl.h"
 #include "NetworkSocket.h"
 
-#define LIBTGVOIP_VERSION "1.0.1"
+#define LIBTGVOIP_VERSION "1.0.3"
 
 #define STATE_WAIT_INIT 1
 #define STATE_WAIT_INIT_ACK 2
@@ -341,6 +341,7 @@ class VoIPController
 		 * @param enabled I usually pick argument names to be self-explanatory
 		 */
 		void SetAudioOutputGainControlEnabled(bool enabled);
+		void SetEchoCancellationStrength(int strength);
 
 private:
 	struct PendingOutgoingPacket{
@@ -464,6 +465,7 @@ class VoIPController
 	int udpConnectivityState;
 	double lastUdpPingTime;
 	int udpPingCount;
+	int echoCancellationStrength;
 
 	int proxyProtocol;
 	std::string proxyAddress;

client/android/tg_voip_jni.cpp

@@ -303,6 +303,10 @@ extern "C" JNIEXPORT void Java_org_telegram_messenger_voip_VoIPController_native
 	((VoIPController*)(intptr_t)inst)->SetAudioOutputGainControlEnabled(enabled);
 }
 
+extern "C" JNIEXPORT void Java_org_telegram_messenger_voip_VoIPController_nativeSetEchoCancellationStrength(JNIEnv* env, jclass cls, jlong inst, jint strength){
+	((VoIPController*)(intptr_t)inst)->SetEchoCancellationStrength(strength);
+}
+
 extern "C" JNIEXPORT jint Java_org_telegram_messenger_voip_Resampler_convert44to48(JNIEnv* env, jclass cls, jobject from, jobject to){
 	return tgvoip::audio::Resampler::Convert44To48((int16_t *) env->GetDirectBufferAddress(from), (int16_t *) env->GetDirectBufferAddress(to), (size_t) (env->GetDirectBufferCapacity(from)/2), (size_t) (env->GetDirectBufferCapacity(to)/2));
 }

os/linux/AudioInputALSA.cpp

@@ -71,6 +71,7 @@ void AudioInputALSA::Stop(){
 
 void* AudioInputALSA::StartThread(void* arg){
 	((AudioInputALSA*)arg)->RunThread();
+	return NULL;
 }
 
 void AudioInputALSA::RunThread(){

os/linux/AudioOutputALSA.cpp

@@ -74,6 +74,7 @@ bool AudioOutputALSA::IsPlaying(){
 
 void* AudioOutputALSA::StartThread(void* arg){
 	((AudioOutputALSA*)arg)->RunThread();
+	return NULL;
 }
 
 void AudioOutputALSA::RunThread(){

os/windows/CXWrapper.cpp

@@ -41,7 +41,6 @@ VoIPControllerWrapper::VoIPControllerWrapper(){
 	controller->implData=(void*)this;
 	controller->SetStateCallback(VoIPControllerWrapper::OnStateChanged);
 	controller->SetSignalBarsCountCallback(VoIPControllerWrapper::OnSignalBarsChanged);
-	stateCallback=nullptr;
 }
 
 VoIPControllerWrapper::~VoIPControllerWrapper(){
@@ -86,10 +85,6 @@ void VoIPControllerWrapper::SetNetworkType(NetworkType type){
 	controller->SetNetworkType((int)type);
 }
 
-void VoIPControllerWrapper::SetStateCallback(IStateCallback^ callback){
-	stateCallback=callback;
-}
-
 void VoIPControllerWrapper::SetMicMute(bool mute){
 	controller->SetMicMute(mute);
 }
@@ -142,13 +137,11 @@ void VoIPControllerWrapper::OnSignalBarsChanged(VoIPController* c, int count){
 }
 
 void VoIPControllerWrapper::OnStateChangedInternal(int state){
-	if(stateCallback)
-		stateCallback->OnCallStateChanged((CallState)state);
+	CallStateChanged(this, (CallState)state);
 }
 
 void VoIPControllerWrapper::OnSignalBarsChangedInternal(int count){
-	if(stateCallback)
-		stateCallback->OnSignalBarsChanged(count);
+	SignalBarsChanged(this, count);
 }
 
 void VoIPControllerWrapper::SetConfig(double initTimeout, double recvTimeout, DataSavingMode dataSavingMode, bool enableAEC, bool enableNS, bool enableAGC, Platform::String^ logFilePath, Platform::String^ statsDumpFilePath){

os/windows/CXWrapper.h

@@ -58,10 +58,11 @@ namespace libtgvoip{
 		SOCKS5
 	};
 
-	public interface class IStateCallback{
-		void OnCallStateChanged(CallState newState);
-		void OnSignalBarsChanged(int count);
-	};
+	ref class VoIPControllerWrapper;
+	public delegate void CallStateChangedEventHandler(VoIPControllerWrapper^ sender, CallState newState);
+
+	ref class VoIPControllerWrapper;
+	public delegate void SignalBarsChangedEventHandler(VoIPControllerWrapper^ sender, int newCount);
 
     public ref class VoIPControllerWrapper sealed{
     public:
@@ -71,7 +72,6 @@ namespace libtgvoip{
 		void Connect();
 		void SetPublicEndpoints(const Platform::Array<Endpoint^>^ endpoints, bool allowP2P);
 		void SetNetworkType(NetworkType type);
-		void SetStateCallback(IStateCallback^ callback);
 		void SetMicMute(bool mute);
 		void SetEncryptionKey(const Platform::Array<uint8>^ key, bool isOutgoing);
 		void SetConfig(double initTimeout, double recvTimeout, DataSavingMode dataSavingMode, bool enableAEC, bool enableNS, bool enableAGC, Platform::String^ logFilePath, Platform::String^ statsDumpFilePath);
@@ -85,13 +85,16 @@ namespace libtgvoip{
 		static void UpdateServerConfig(Platform::String^ json);
 		static void SwitchSpeaker(bool external);
 		//static Platform::String^ TestAesIge();
+
+		event CallStateChangedEventHandler^ CallStateChanged;
+		event SignalBarsChangedEventHandler^ SignalBarsChanged;
+
 	private:
 		static void OnStateChanged(tgvoip::VoIPController* c, int state);
 		static void OnSignalBarsChanged(tgvoip::VoIPController* c, int count);
 		void OnStateChangedInternal(int state);
 		void OnSignalBarsChangedInternal(int count);
 		tgvoip::VoIPController* controller;
-		IStateCallback^ stateCallback;
     };
 
 	ref class MicrosoftCryptoImpl{