diff --git a/EchoCanceller.cpp b/EchoCanceller.cpp
index 2106532..23ccc94 100644
--- a/EchoCanceller.cpp
+++ b/EchoCanceller.cpp
@@ -97,12 +97,12 @@ EchoCanceller::EchoCanceller(bool enableAEC, bool enableNS, bool enableAGC){
 	if(enableAGC){
 		agc=WebRtcAgc_Create();
 		WebRtcAgcConfig agcConfig;
-		agcConfig.compressionGaindB = 9;
+		agcConfig.compressionGaindB = 20;
 		agcConfig.limiterEnable = 1;
-		agcConfig.targetLevelDbfs = 3;
-		WebRtcAgc_Init(agc, 0, 255, kAgcModeAdaptiveAnalog, 48000);
+		agcConfig.targetLevelDbfs = 9;
+		WebRtcAgc_Init(agc, 0, 255, kAgcModeAdaptiveDigital, 48000);
 		WebRtcAgc_set_config(agc, agcConfig);
-		agcMicLevel=128;
+		agcMicLevel=0;
 	}
 #endif
 }
@@ -354,3 +354,75 @@ void EchoCanceller::ProcessInput(unsigned char* data, unsigned char* out, size_t
 	memcpy(samplesOut, bufIn->ibuf_const()->bands(0)[0], 960*2);
 }
 
+AudioEffect::~AudioEffect(){
+
+}
+
+void AudioEffect::SetPassThrough(bool passThrough){
+	this->passThrough=passThrough;
+}
+
+AutomaticGainControl::AutomaticGainControl(){
+	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;
+}
+
+AutomaticGainControl::~AutomaticGainControl(){
+	delete (webrtc::SplittingFilter*)splittingFilter;
+	delete (webrtc::IFChannelBuffer*)splittingFilterIn;
+	delete (webrtc::IFChannelBuffer*)splittingFilterOut;
+	WebRtcAgc_Free(agc);
+}
+
+void AutomaticGainControl::Process(int16_t *inOut, size_t numSamples){
+	if(passThrough)
+		return;
+	if(numSamples!=960){
+		LOGW("AutomaticGainControl only works on 960-sample buffers (got %u samples)", 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);
+}
+
diff --git a/EchoCanceller.h b/EchoCanceller.h
index 982bd4a..a915ec1 100644
--- a/EchoCanceller.h
+++ b/EchoCanceller.h
@@ -10,6 +10,7 @@
 #include "threading.h"
 #include "BufferPool.h"
 #include "BlockingQueue.h"
+#include "MediaStreamItf.h"
 
 namespace tgvoip{
 class EchoCanceller{
@@ -48,6 +49,29 @@ private:
 	int32_t agcMicLevel;
 #endif
 };
-}
+
+	class AudioEffect{
+	public:
+		virtual ~AudioEffect()=0;
+		virtual void Process(int16_t* inOut, size_t numSamples)=0;
+		virtual void SetPassThrough(bool passThrough);
+	protected:
+		bool passThrough;
+	};
+
+	class AutomaticGainControl : public AudioEffect{
+	public:
+		AutomaticGainControl();
+		virtual ~AutomaticGainControl();
+		virtual void Process(int16_t* inOut, size_t numSamples);
+
+	private:
+		void* agc;
+		void* splittingFilter;
+		void* splittingFilterIn;
+		void* splittingFilterOut;
+		int32_t agcMicLevel;
+	};
+};
 
 #endif //LIBTGVOIP_ECHOCANCELLER_H
diff --git a/OpusDecoder.cpp b/OpusDecoder.cpp
index 6f6626a..77fcb8d 100644
--- a/OpusDecoder.cpp
+++ b/OpusDecoder.cpp
@@ -8,6 +8,7 @@
 #include "audio/Resampler.h"
 #include "logging.h"
 #include <assert.h>
+#include <algorithm>
 
 #define PACKET_SIZE (960*2)
 
@@ -222,6 +223,9 @@ void tgvoip::OpusDecoder::RunThread(){
 			unsigned char *buf=bufferPool->Get();
 			if(buf){
 				if(size>0){
+					for(std::vector<AudioEffect*>::iterator effect=postProcEffects.begin();effect!=postProcEffects.end();++effect){
+						(*effect)->Process(reinterpret_cast<int16_t*>(processedBuffer+(PACKET_SIZE*i)), 960);
+					}
 					memcpy(buf, processedBuffer+(PACKET_SIZE*i), PACKET_SIZE);
 				}else{
 					LOGE("Error decoding, result=%d", size);
@@ -255,3 +259,13 @@ void tgvoip::OpusDecoder::ResetQueue(){
 void tgvoip::OpusDecoder::SetJitterBuffer(JitterBuffer* jitterBuffer){
 	this->jitterBuffer=jitterBuffer;
 }
+
+void tgvoip::OpusDecoder::AddAudioEffect(AudioEffect *effect){
+	postProcEffects.push_back(effect);
+}
+
+void tgvoip::OpusDecoder::RemoveAudioEffect(AudioEffect *effect){
+	std::vector<AudioEffect*>::iterator i=std::find(postProcEffects.begin(), postProcEffects.end(), effect);
+	if(i!=postProcEffects.end())
+		postProcEffects.erase(i);
+}
diff --git a/OpusDecoder.h b/OpusDecoder.h
index 51bbbac..1a88ce5 100644
--- a/OpusDecoder.h
+++ b/OpusDecoder.h
@@ -16,6 +16,7 @@
 #include "EchoCanceller.h"
 #include "JitterBuffer.h"
 #include <stdio.h>
+#include <vector>
 
 namespace tgvoip{
 class OpusDecoder {
@@ -31,6 +32,8 @@ public:
 	void SetFrameDuration(uint32_t duration);
 	void ResetQueue();
 	void SetJitterBuffer(JitterBuffer* jitterBuffer);
+	void AddAudioEffect(AudioEffect* effect);
+	void RemoveAudioEffect(AudioEffect* effect);
 
 private:
 	static size_t Callback(unsigned char* data, size_t len, void* param);
@@ -50,6 +53,7 @@ private:
 	uint32_t frameDuration;
 	EchoCanceller* echoCanceller;
 	JitterBuffer* jitterBuffer;
+	std::vector<AudioEffect*> postProcEffects;
 };
 }
 
diff --git a/VoIPController.cpp b/VoIPController.cpp
index 7f4e87c..9623c00 100644
--- a/VoIPController.cpp
+++ b/VoIPController.cpp
@@ -234,6 +234,9 @@ VoIPController::VoIPController() : activeNetItfName(""),
 	realUdpSocket=udpSocket;
 	udpConnectivityState=UDP_UNKNOWN;
 
+	outputAGC=NULL;
+	outputAGCEnabled=false;
+
 	maxAudioBitrate=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_max_bitrate", 20000);
 	maxAudioBitrateGPRS=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_max_bitrate_gprs", 8000);
 	maxAudioBitrateEDGE=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_max_bitrate_edge", 16000);
@@ -264,6 +267,8 @@ VoIPController::VoIPController() : activeNetItfName(""),
 	stm->enabled=1;
 	stm->frameDuration=60;
 	outgoingStreams.push_back(stm);
+										
+	memset(signalBarsHistory, 0, sizeof(signalBarsHistory));
 }
 
 VoIPController::~VoIPController(){
@@ -361,6 +366,8 @@ VoIPController::~VoIPController(){
 	if(resolvedProxyAddress)
 		delete resolvedProxyAddress;
 	delete selectCanceller;
+	if(outputAGC)
+		delete outputAGC;
 	LOGD("Left VoIPController::~VoIPController");
 }
 
@@ -1182,10 +1189,13 @@ simpleAudioBlock random_id:long random_bytes:string raw_data:string = DecryptedA
 					UpdateAudioBitrate();
 
 					jitterBuffer=new JitterBuffer(NULL, incomingAudioStream->frameDuration);
+					outputAGC=new AutomaticGainControl();
+					outputAGC->SetPassThrough(!outputAGCEnabled);
 					decoder=new OpusDecoder(audioOutput);
 					decoder->SetEchoCanceller(echoCanceller);
 					decoder->SetJitterBuffer(jitterBuffer);
 					decoder->SetFrameDuration(incomingAudioStream->frameDuration);
+					decoder->AddAudioEffect(outputAGC);
 					decoder->Start();
 					if(incomingAudioStream->frameDuration>50)
 						jitterBuffer->SetMinPacketCount((uint32_t) ServerConfig::GetSharedInstance()->GetInt("jitter_initial_delay_60", 3));
@@ -1243,7 +1253,7 @@ simpleAudioBlock random_id:long random_bytes:string raw_data:string = DecryptedA
 					audioOutput->Start();
 					audioOutStarted=true;
 				}
-				if(jitterBuffer)
+				if(jitterBuffer && in.Remaining()>=sdlen)
 					jitterBuffer->HandleInput((unsigned char*) (buffer+in.GetOffset()), sdlen, pts);
 				if(i<count-1)
 					in.Seek(in.GetOffset()+sdlen);
@@ -1376,7 +1386,7 @@ void VoIPController::RunTickThread(){
 #else
 		Sleep(100);
 #endif
-		int prevSignalBarCount=signalBarCount;
+		int prevSignalBarCount=GetSignalBarsCount();
 		signalBarCount=4;
 		tickCount++;
 		if(connectionInitTime==0)
@@ -1559,12 +1569,12 @@ void VoIPController::RunTickThread(){
 			double avgDelay=jitterBuffer->GetAverageDelay();
 			double avgLateCount[3];
 			jitterBuffer->GetAverageLateCount(avgLateCount);
-			if(avgDelay>=5)
+			/*if(avgDelay>=5)
 				signalBarCount=1;
 			else if(avgDelay>=4)
 				signalBarCount=MIN(signalBarCount, 2);
 			else if(avgDelay>=3)
-				signalBarCount=MIN(signalBarCount, 3);
+				signalBarCount=MIN(signalBarCount, 3);*/
 
 			if(avgLateCount[2]>=0.2)
 				signalBarCount=1;
@@ -1711,10 +1721,12 @@ void VoIPController::RunTickThread(){
 			setEstablishedAt=0;
 		}
 
-		if(signalBarCount!=prevSignalBarCount){
-			LOGD("SIGNAL BAR COUNT CHANGED: %d", signalBarCount);
+		signalBarsHistory[tickCount%sizeof(signalBarsHistory)]=(unsigned char)signalBarCount;
+		int _signalBarCount=GetSignalBarsCount();
+		if(_signalBarCount!=prevSignalBarCount){
+			LOGD("SIGNAL BAR COUNT CHANGED: %d", _signalBarCount);
 			if(signalBarCountCallback)
-				signalBarCountCallback(this, signalBarCount);
+				signalBarCountCallback(this, _signalBarCount);
 		}
 
 
@@ -2460,13 +2472,23 @@ void VoIPController::SendUdpPing(Endpoint *endpoint){
 }
 
 int VoIPController::GetSignalBarsCount(){
-	return signalBarCount;
+	unsigned char avg=0;
+	for(int i=0;i<sizeof(signalBarsHistory);i++)
+		avg+=signalBarsHistory[i];
+	return avg >> 2;
 }
 
 void VoIPController::SetSignalBarsCountCallback(void (*f)(VoIPController *, int)){
 	signalBarCountCallback=f;
 }
 
+void VoIPController::SetAudioOutputGainControlEnabled(bool enabled){
+	LOGD("New output AGC state: %d", enabled);
+	outputAGCEnabled=enabled;
+	if(outputAGC)
+		outputAGC->SetPassThrough(!enabled);
+}
+
 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;
diff --git a/VoIPController.h b/VoIPController.h
index 2b1f4db..88f3a64 100644
--- a/VoIPController.h
+++ b/VoIPController.h
@@ -318,7 +318,7 @@ public:
 	std::string GetCurrentAudioOutputID();
 	/**
 	 * Set the proxy server to route the data through. Call this before connecting.
-	 * @param protocol PROXY_NONE, PROXY_SOCKS4, or PROXY_SOCKS5
+	 * @param protocol PROXY_NONE or PROXY_SOCKS5
 	 * @param address IP address or domain name of the server
 	 * @param port Port of the server
 	 * @param username Username; empty string for anonymous
@@ -335,6 +335,13 @@ public:
 		 * @param f
 		 */
 		void SetSignalBarsCountCallback(void (*f)(VoIPController*, int));
+		/**
+		 * Enable or disable AGC (automatic gain control) on audio output. Should only be enabled on phones when the earpiece speaker is being used.
+		 * The audio output will be louder with this on.
+		 * AGC with speakerphone or other kinds of loud speakers has detrimental effects on some echo cancellation implementations.
+		 * @param enabled I usually pick argument names to be self-explanatory
+		 */
+		void SetAudioOutputGainControlEnabled(bool enabled);
 
 private:
 	struct PendingOutgoingPacket{
@@ -452,6 +459,7 @@ private:
 	double setEstablishedAt;
 	SocketSelectCanceller* selectCanceller;
 	NetworkSocket* openingTcpSocket;
+	unsigned char signalBarsHistory[4];
 
 	BufferPool outgoingPacketsBufferPool;
 	int udpConnectivityState;
@@ -467,6 +475,9 @@ private:
 
 		int signalBarCount;
 		void (*signalBarCountCallback)(VoIPController*, int);
+
+		AutomaticGainControl* outputAGC;
+		bool outputAGCEnabled;
 	
 	/*** server config values ***/
 	uint32_t maxAudioBitrate;
diff --git a/client/android/tg_voip_jni.cpp b/client/android/tg_voip_jni.cpp
index dcb18df..f88b338 100644
--- a/client/android/tg_voip_jni.cpp
+++ b/client/android/tg_voip_jni.cpp
@@ -299,6 +299,10 @@ extern "C" JNIEXPORT jstring Java_org_telegram_messenger_voip_VoIPController_nat
 	return env->NewStringUTF(log.c_str());
 }
 
+extern "C" JNIEXPORT void Java_org_telegram_messenger_voip_VoIPController_nativeSetAudioOutputGainControlEnabled(JNIEnv* env, jclass clasz, jlong inst, jboolean enabled){
+	((VoIPController*)(intptr_t)inst)->SetAudioOutputGainControlEnabled(enabled);
+}
+
 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));
 }
diff --git a/libtgvoip_osx.xcodeproj/project.pbxproj b/libtgvoip_osx.xcodeproj/project.pbxproj
index c4d966f..d5195f1 100644
--- a/libtgvoip_osx.xcodeproj/project.pbxproj
+++ b/libtgvoip_osx.xcodeproj/project.pbxproj
@@ -47,12 +47,8 @@
 		692AB9201E675F7000706ACC /* AudioUnit.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 692AB91D1E675F7000706ACC /* AudioUnit.framework */; };
 		692AB9211E675F7000706ACC /* CoreAudio.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 692AB91E1E675F7000706ACC /* CoreAudio.framework */; };
 		695B20621EBD39FF00E31757 /* DarwinSpecific.h in Headers */ = {isa = PBXBuildFile; fileRef = 695B20601EBD39FF00E31757 /* DarwinSpecific.h */; };
-		695B20631EBD39FF00E31757 /* DarwinSpecific.mm in Sources */ = {isa = PBXBuildFile; fileRef = 695B20611EBD39FF00E31757 /* DarwinSpecific.mm */; };
-		698848411F4B39F700076DF0 /* AudioInputAudioUnit.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 6988483B1F4B39F700076DF0 /* AudioInputAudioUnit.cpp */; };
 		698848421F4B39F700076DF0 /* AudioInputAudioUnit.h in Headers */ = {isa = PBXBuildFile; fileRef = 6988483C1F4B39F700076DF0 /* AudioInputAudioUnit.h */; };
-		698848431F4B39F700076DF0 /* AudioOutputAudioUnit.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 6988483D1F4B39F700076DF0 /* AudioOutputAudioUnit.cpp */; };
 		698848441F4B39F700076DF0 /* AudioOutputAudioUnit.h in Headers */ = {isa = PBXBuildFile; fileRef = 6988483E1F4B39F700076DF0 /* AudioOutputAudioUnit.h */; };
-		698848451F4B39F700076DF0 /* AudioUnitIO.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 6988483F1F4B39F700076DF0 /* AudioUnitIO.cpp */; };
 		698848461F4B39F700076DF0 /* AudioUnitIO.h in Headers */ = {isa = PBXBuildFile; fileRef = 698848401F4B39F700076DF0 /* AudioUnitIO.h */; };
 		69A6DEB91E96149300000E69 /* array_view.h in Headers */ = {isa = PBXBuildFile; fileRef = 69A6DE231E96149300000E69 /* array_view.h */; };
 		69A6DEBA1E96149300000E69 /* atomicops.h in Headers */ = {isa = PBXBuildFile; fileRef = 69A6DE241E96149300000E69 /* atomicops.h */; };
@@ -189,8 +185,12 @@
 		69A6DF441E9614B700000E69 /* AudioInputAudioUnitOSX.h in Headers */ = {isa = PBXBuildFile; fileRef = 69A6DF401E9614B700000E69 /* AudioInputAudioUnitOSX.h */; };
 		69A6DF451E9614B700000E69 /* AudioOutputAudioUnitOSX.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 69A6DF411E9614B700000E69 /* AudioOutputAudioUnitOSX.cpp */; };
 		69A6DF461E9614B700000E69 /* AudioOutputAudioUnitOSX.h in Headers */ = {isa = PBXBuildFile; fileRef = 69A6DF421E9614B700000E69 /* AudioOutputAudioUnitOSX.h */; };
-		69AC14901F4B41CF00AC3173 /* Resampler.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 69AC148E1F4B41CF00AC3173 /* Resampler.cpp */; };
 		69AC14911F4B41CF00AC3173 /* Resampler.h in Headers */ = {isa = PBXBuildFile; fileRef = 69AC148F1F4B41CF00AC3173 /* Resampler.h */; };
+		C2A87DD81F4B6A33002D3F73 /* Resampler.cpp in Sources */ = {isa = PBXBuildFile; fileRef = C2A87DD71F4B6A33002D3F73 /* Resampler.cpp */; };
+		C2A87DDA1F4B6A57002D3F73 /* DarwinSpecific.mm in Sources */ = {isa = PBXBuildFile; fileRef = C2A87DD91F4B6A57002D3F73 /* DarwinSpecific.mm */; };
+		C2A87DDF1F4B6A61002D3F73 /* AudioInputAudioUnit.cpp in Sources */ = {isa = PBXBuildFile; fileRef = C2A87DDB1F4B6A61002D3F73 /* AudioInputAudioUnit.cpp */; };
+		C2A87DE01F4B6A61002D3F73 /* AudioOutputAudioUnit.cpp in Sources */ = {isa = PBXBuildFile; fileRef = C2A87DDD1F4B6A61002D3F73 /* AudioOutputAudioUnit.cpp */; };
+		C2A87DE41F4B6AD3002D3F73 /* AudioUnitIO.cpp in Sources */ = {isa = PBXBuildFile; fileRef = C2A87DE31F4B6AD3002D3F73 /* AudioUnitIO.cpp */; };
 /* End PBXBuildFile section */
 
 /* Begin PBXContainerItemProxy section */
@@ -425,6 +425,15 @@
 		69AC148E1F4B41CF00AC3173 /* Resampler.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = Resampler.cpp; path = "../../../../Telegram-iOS/submodules/libtgvoip/audio/Resampler.cpp"; sourceTree = "<group>"; };
 		69AC148F1F4B41CF00AC3173 /* Resampler.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = Resampler.h; path = "../../../../Telegram-iOS/submodules/libtgvoip/audio/Resampler.h"; sourceTree = "<group>"; };
 		69F842361E67540700C110F7 /* libtgvoip.framework */ = {isa = PBXFileReference; explicitFileType = wrapper.framework; includeInIndex = 0; path = libtgvoip.framework; sourceTree = BUILT_PRODUCTS_DIR; };
+		C2A87DD71F4B6A33002D3F73 /* Resampler.cpp */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; name = Resampler.cpp; path = audio/Resampler.cpp; sourceTree = "<group>"; };
+		C2A87DD91F4B6A57002D3F73 /* DarwinSpecific.mm */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.objcpp; name = DarwinSpecific.mm; path = os/darwin/DarwinSpecific.mm; sourceTree = "<group>"; };
+		C2A87DDB1F4B6A61002D3F73 /* AudioInputAudioUnit.cpp */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; name = AudioInputAudioUnit.cpp; path = os/darwin/AudioInputAudioUnit.cpp; sourceTree = "<group>"; };
+		C2A87DDC1F4B6A61002D3F73 /* AudioInputAudioUnitOSX.cpp */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; name = AudioInputAudioUnitOSX.cpp; path = os/darwin/AudioInputAudioUnitOSX.cpp; sourceTree = "<group>"; };
+		C2A87DDD1F4B6A61002D3F73 /* AudioOutputAudioUnit.cpp */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; name = AudioOutputAudioUnit.cpp; path = os/darwin/AudioOutputAudioUnit.cpp; sourceTree = "<group>"; };
+		C2A87DDE1F4B6A61002D3F73 /* AudioOutputAudioUnitOSX.cpp */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; name = AudioOutputAudioUnitOSX.cpp; path = os/darwin/AudioOutputAudioUnitOSX.cpp; sourceTree = "<group>"; };
+		C2A87DE11F4B6A89002D3F73 /* AudioInput.cpp */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; name = AudioInput.cpp; path = audio/AudioInput.cpp; sourceTree = "<group>"; };
+		C2A87DE21F4B6A89002D3F73 /* AudioOutput.cpp */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; name = AudioOutput.cpp; path = audio/AudioOutput.cpp; sourceTree = "<group>"; };
+		C2A87DE31F4B6AD3002D3F73 /* AudioUnitIO.cpp */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; name = AudioUnitIO.cpp; path = os/darwin/AudioUnitIO.cpp; sourceTree = "<group>"; };
 /* End PBXFileReference section */
 
 /* Begin PBXFrameworksBuildPhase section */
@@ -840,6 +849,15 @@
 		69F8422C1E67540700C110F7 = {
 			isa = PBXGroup;
 			children = (
+				C2A87DE31F4B6AD3002D3F73 /* AudioUnitIO.cpp */,
+				C2A87DE11F4B6A89002D3F73 /* AudioInput.cpp */,
+				C2A87DE21F4B6A89002D3F73 /* AudioOutput.cpp */,
+				C2A87DDB1F4B6A61002D3F73 /* AudioInputAudioUnit.cpp */,
+				C2A87DDC1F4B6A61002D3F73 /* AudioInputAudioUnitOSX.cpp */,
+				C2A87DDD1F4B6A61002D3F73 /* AudioOutputAudioUnit.cpp */,
+				C2A87DDE1F4B6A61002D3F73 /* AudioOutputAudioUnitOSX.cpp */,
+				C2A87DD91F4B6A57002D3F73 /* DarwinSpecific.mm */,
+				C2A87DD71F4B6A33002D3F73 /* Resampler.cpp */,
 				692AB8861E6759BF00706ACC /* libtgvoip */,
 				69F842371E67540700C110F7 /* Products */,
 				692AB9061E675E8700706ACC /* Frameworks */,
@@ -1072,6 +1090,7 @@
 				69A6DEF41E96149300000E69 /* resample_fractional.c in Sources */,
 				6915307B1E6B5BAB004F643F /* logging.cpp in Sources */,
 				69A6DEE11E96149300000E69 /* get_scaling_square.c in Sources */,
+				C2A87DE41F4B6AD3002D3F73 /* AudioUnitIO.cpp in Sources */,
 				690725C21EBBD5F2005D860B /* NetworkSocket.cpp in Sources */,
 				69A6DEC71E96149300000E69 /* channel_buffer.cc in Sources */,
 				69A6DF191E96149300000E69 /* apm_data_dumper.cc in Sources */,
@@ -1086,12 +1105,10 @@
 				69A6DEF71E96149300000E69 /* spl_sqrt.c in Sources */,
 				69A6DEED1E96149300000E69 /* real_fft.c in Sources */,
 				692AB9021E6759DD00706ACC /* VoIPController.cpp in Sources */,
-				69A6DF431E9614B700000E69 /* AudioInputAudioUnitOSX.cpp in Sources */,
 				69A6DF0D1E96149300000E69 /* aecm_core.cc in Sources */,
 				69A6DF101E96149300000E69 /* aecm_core_neon.cc in Sources */,
 				69A6DED71E96149300000E69 /* division_operations.c in Sources */,
 				69A6DEDB1E96149300000E69 /* energy.c in Sources */,
-				695B20631EBD39FF00E31757 /* DarwinSpecific.mm in Sources */,
 				69A6DEC61E96149300000E69 /* audio_util.cc in Sources */,
 				69A6DF141E96149300000E69 /* analog_agc.c in Sources */,
 				69A6DEF81E96149300000E69 /* spl_sqrt_floor.c in Sources */,
@@ -1109,24 +1126,28 @@
 				692AB8EB1E6759DD00706ACC /* OpusDecoder.cpp in Sources */,
 				69A6DED81E96149300000E69 /* dot_product_with_scale.c in Sources */,
 				69A6DF331E96149300000E69 /* ooura_fft.cc in Sources */,
-				698848451F4B39F700076DF0 /* AudioUnitIO.cpp in Sources */,
 				69A6DEF11E96149300000E69 /* resample_by_2.c in Sources */,
 				69A6DEEC1E96149300000E69 /* randomization_functions.c in Sources */,
 				69A6DEEE1E96149300000E69 /* refl_coef_to_lpc.c in Sources */,
+				C2A87DDF1F4B6A61002D3F73 /* AudioInputAudioUnit.cpp in Sources */,
+				69A6DF431E9614B700000E69 /* AudioInputAudioUnitOSX.cpp in Sources */,
+				C2A87DE01F4B6A61002D3F73 /* AudioOutputAudioUnit.cpp in Sources */,
+				69A6DF451E9614B700000E69 /* AudioOutputAudioUnitOSX.cpp in Sources */,
 				69A6DEFC1E96149300000E69 /* vector_scaling_operations.c in Sources */,
 				692AB8E61E6759DD00706ACC /* JitterBuffer.cpp in Sources */,
+				692AB8CB1E6759DD00706ACC /* AudioInput.cpp in Sources */,
+				692AB8CD1E6759DD00706ACC /* AudioOutput.cpp in Sources */,
+				C2A87DDA1F4B6A57002D3F73 /* DarwinSpecific.mm in Sources */,
+				C2A87DD81F4B6A33002D3F73 /* Resampler.cpp in Sources */,
 				69A6DEFA1E96149300000E69 /* splitting_filter_impl.c in Sources */,
-				69AC14901F4B41CF00AC3173 /* Resampler.cpp in Sources */,
 				69A6DEE01E96149300000E69 /* get_hanning_window.c in Sources */,
 				69A6DF161E96149300000E69 /* digital_agc.c in Sources */,
 				69A6DF061E96149300000E69 /* aec_core_neon.cc in Sources */,
 				69A6DF201E96149300000E69 /* ns_core.c in Sources */,
-				698848431F4B39F700076DF0 /* AudioOutputAudioUnit.cpp in Sources */,
 				69A6DF091E96149300000E69 /* aec_resampler.cc in Sources */,
 				692AB8D11E6759DD00706ACC /* BufferInputStream.cpp in Sources */,
 				692AB8E91E6759DD00706ACC /* MediaStreamItf.cpp in Sources */,
 				69A6DF2C1E96149300000E69 /* block_mean_calculator.cc in Sources */,
-				69A6DF451E9614B700000E69 /* AudioOutputAudioUnitOSX.cpp in Sources */,
 				69A6DEBC1E96149300000E69 /* checks.cc in Sources */,
 				692AB8DA1E6759DD00706ACC /* EchoCanceller.cpp in Sources */,
 				69A6DF281E96149300000E69 /* splitting_filter.cc in Sources */,
@@ -1143,18 +1164,15 @@
 				69A6DF1E1E96149300000E69 /* noise_suppression_x.c in Sources */,
 				692AB8D51E6759DD00706ACC /* BufferPool.cpp in Sources */,
 				69A6DED01E96149300000E69 /* complex_bit_reverse.c in Sources */,
-				692AB8CB1E6759DD00706ACC /* AudioInput.cpp in Sources */,
 				69A6DEDF1E96149300000E69 /* filter_ma_fast_q12.c in Sources */,
 				69A6DEFF1E96149300000E69 /* wav_file.cc in Sources */,
 				69A6DF351E96149300000E69 /* ooura_fft_neon.cc in Sources */,
 				69A6DECE1E96149300000E69 /* auto_corr_to_refl_coef.c in Sources */,
 				69A6DEFD1E96149300000E69 /* sparse_fir_filter.cc in Sources */,
 				69A6DED91E96149300000E69 /* downsample_fast.c in Sources */,
-				692AB8CD1E6759DD00706ACC /* AudioOutput.cpp in Sources */,
 				69A6DF251E96149300000E69 /* nsx_core_neon.c in Sources */,
 				69A6DF081E96149300000E69 /* aec_core_sse2.cc in Sources */,
 				69A6DEEA1E96149300000E69 /* min_max_operations.c in Sources */,
-				698848411F4B39F700076DF0 /* AudioInputAudioUnit.cpp in Sources */,
 				69A6DF361E96149300000E69 /* ooura_fft_sse2.cc in Sources */,
 				69A6DED51E96149300000E69 /* cross_correlation.c in Sources */,
 				69A6DF3D1E96149300000E69 /* cpu_features.cc in Sources */,
@@ -1170,7 +1188,7 @@
 /* End PBXSourcesBuildPhase section */
 
 /* Begin XCBuildConfiguration section */
-		69F8423C1E67540700C110F7 /* Debug */ = {
+		69F8423C1E67540700C110F7 /* Debug Hockeyapp */ = {
 			isa = XCBuildConfiguration;
 			buildSettings = {
 				ALWAYS_SEARCH_USER_PATHS = NO;
@@ -1219,9 +1237,9 @@
 				VERSIONING_SYSTEM = "apple-generic";
 				VERSION_INFO_PREFIX = "";
 			};
-			name = Debug;
+			name = "Debug Hockeyapp";
 		};
-		69F8423D1E67540700C110F7 /* Release */ = {
+		69F8423D1E67540700C110F7 /* Release Hockeyapp */ = {
 			isa = XCBuildConfiguration;
 			buildSettings = {
 				ALWAYS_SEARCH_USER_PATHS = NO;
@@ -1264,9 +1282,9 @@
 				VERSIONING_SYSTEM = "apple-generic";
 				VERSION_INFO_PREFIX = "";
 			};
-			name = Release;
+			name = "Release Hockeyapp";
 		};
-		69F8423F1E67540700C110F7 /* Debug */ = {
+		69F8423F1E67540700C110F7 /* Debug Hockeyapp */ = {
 			isa = XCBuildConfiguration;
 			buildSettings = {
 				CLANG_CXX_LANGUAGE_STANDARD = "c++0x";
@@ -1303,9 +1321,9 @@
 				SDKROOT = macosx;
 				SKIP_INSTALL = YES;
 			};
-			name = Debug;
+			name = "Debug Hockeyapp";
 		};
-		69F842401E67540700C110F7 /* Release */ = {
+		69F842401E67540700C110F7 /* Release Hockeyapp */ = {
 			isa = XCBuildConfiguration;
 			buildSettings = {
 				CLANG_CXX_LANGUAGE_STANDARD = "c++0x";
@@ -1342,7 +1360,7 @@
 				SDKROOT = macosx;
 				SKIP_INSTALL = YES;
 			};
-			name = Release;
+			name = "Release Hockeyapp";
 		};
 		D04D01C31E678C0D0086DDC0 /* Debug AppStore */ = {
 			isa = XCBuildConfiguration;
@@ -1434,7 +1452,7 @@
 			};
 			name = "Debug AppStore";
 		};
-		D04D01CB1E678C230086DDC0 /* Hockeyapp */ = {
+		D04D01CB1E678C230086DDC0 /* Release AppStore */ = {
 			isa = XCBuildConfiguration;
 			buildSettings = {
 				ALWAYS_SEARCH_USER_PATHS = NO;
@@ -1477,9 +1495,9 @@
 				VERSIONING_SYSTEM = "apple-generic";
 				VERSION_INFO_PREFIX = "";
 			};
-			name = Hockeyapp;
+			name = "Release AppStore";
 		};
-		D04D01CC1E678C230086DDC0 /* Hockeyapp */ = {
+		D04D01CC1E678C230086DDC0 /* Release AppStore */ = {
 			isa = XCBuildConfiguration;
 			buildSettings = {
 				CLANG_CXX_LANGUAGE_STANDARD = "c++0x";
@@ -1516,7 +1534,7 @@
 				SDKROOT = macosx;
 				SKIP_INSTALL = YES;
 			};
-			name = Hockeyapp;
+			name = "Release AppStore";
 		};
 /* End XCBuildConfiguration section */
 
@@ -1524,24 +1542,24 @@
 		69F842301E67540700C110F7 /* Build configuration list for PBXProject "libtgvoip_osx" */ = {
 			isa = XCConfigurationList;
 			buildConfigurations = (
-				69F8423C1E67540700C110F7 /* Debug */,
+				69F8423C1E67540700C110F7 /* Debug Hockeyapp */,
 				D04D01C31E678C0D0086DDC0 /* Debug AppStore */,
-				69F8423D1E67540700C110F7 /* Release */,
-				D04D01CB1E678C230086DDC0 /* Hockeyapp */,
+				69F8423D1E67540700C110F7 /* Release Hockeyapp */,
+				D04D01CB1E678C230086DDC0 /* Release AppStore */,
 			);
 			defaultConfigurationIsVisible = 0;
-			defaultConfigurationName = Release;
+			defaultConfigurationName = "Release Hockeyapp";
 		};
 		69F8423E1E67540700C110F7 /* Build configuration list for PBXNativeTarget "libtgvoip" */ = {
 			isa = XCConfigurationList;
 			buildConfigurations = (
-				69F8423F1E67540700C110F7 /* Debug */,
+				69F8423F1E67540700C110F7 /* Debug Hockeyapp */,
 				D04D01C41E678C0D0086DDC0 /* Debug AppStore */,
-				69F842401E67540700C110F7 /* Release */,
-				D04D01CC1E678C230086DDC0 /* Hockeyapp */,
+				69F842401E67540700C110F7 /* Release Hockeyapp */,
+				D04D01CC1E678C230086DDC0 /* Release AppStore */,
 			);
 			defaultConfigurationIsVisible = 0;
-			defaultConfigurationName = Release;
+			defaultConfigurationName = "Release Hockeyapp";
 		};
 /* End XCConfigurationList section */
 	};
diff --git a/logging.h b/logging.h
index cbe811b..3168bc4 100644
--- a/logging.h
+++ b/logging.h
@@ -44,10 +44,6 @@ void tgvoip_log_file_write_header();
 #include <windows.h>
 #include <stdio.h>
 
-#if !defined(snprintf) && defined(_WIN32) && defined(__cplusplus_winrt)
-#define snprintf _snprintf
-#endif
-
 #define _TGVOIP_W32_LOG_PRINT(verb, msg, ...){ char __log_buf[1024]; snprintf(__log_buf, 1024, "%c/tgvoip: " msg "\n", verb, ##__VA_ARGS__); OutputDebugStringA(__log_buf); tgvoip_log_file_printf((char)verb, msg, __VA_ARGS__);}
 
 #define LOGV(msg, ...) _TGVOIP_W32_LOG_PRINT('V', msg, ##__VA_ARGS__)
@@ -70,6 +66,9 @@ void tgvoip_log_file_write_header();
 
 #endif
 
+#if !defined(snprintf) && defined(_WIN32) && defined(__cplusplus_winrt)
+#define snprintf _snprintf
+#endif
 
 #ifdef TGVOIP_LOG_VERBOSITY
 #if TGVOIP_LOG_VERBOSITY<5
diff --git a/os/darwin/AudioOutputAudioUnit.cpp b/os/darwin/AudioOutputAudioUnit.cpp
index 7114d4d..791dfb2 100644
--- a/os/darwin/AudioOutputAudioUnit.cpp
+++ b/os/darwin/AudioOutputAudioUnit.cpp
@@ -75,12 +75,12 @@ void AudioOutputAudioUnit::HandleBufferCallback(AudioBufferList *ioData){
 			memset(buf.mData, 0, buf.mDataByteSize);
 			return;
 		}
-		while(remainingDataSize<buf.mDataByteSize){
-			assert(remainingDataSize+BUFFER_SIZE*2<10240);
+#if TARGET_OS_OSX
+		while(remainingDataSize<buf.mDataByteSize/2){
+			assert(remainingDataSize+BUFFER_SIZE*2<sizeof(remainingData));
 			InvokeCallback(remainingData+remainingDataSize, BUFFER_SIZE*2);
 			remainingDataSize+=BUFFER_SIZE*2;
 		}
-#if TARGET_OS_OSX
 		float* dst=reinterpret_cast<float*>(buf.mData);
 		int16_t* src=reinterpret_cast<int16_t*>(remainingData);
 		for(k=0;k<buf.mDataByteSize/4;k++){
@@ -89,6 +89,11 @@ void AudioOutputAudioUnit::HandleBufferCallback(AudioBufferList *ioData){
 		remainingDataSize-=buf.mDataByteSize/2;
 		memmove(remainingData, remainingData+buf.mDataByteSize/2, remainingDataSize);
 #else
+		while(remainingDataSize<buf.mDataByteSize){
+			assert(remainingDataSize+BUFFER_SIZE*2<sizeof(remainingData));
+			InvokeCallback(remainingData+remainingDataSize, BUFFER_SIZE*2);
+			remainingDataSize+=BUFFER_SIZE*2;
+		}
 		memcpy(buf.mData, remainingData, buf.mDataByteSize);
 		remainingDataSize-=buf.mDataByteSize;
 		memmove(remainingData, remainingData+buf.mDataByteSize, remainingDataSize);
diff --git a/os/windows/CXWrapper.cpp b/os/windows/CXWrapper.cpp
index 5f71f3b..bbc1279 100755
--- a/os/windows/CXWrapper.cpp
+++ b/os/windows/CXWrapper.cpp
@@ -40,6 +40,7 @@ VoIPControllerWrapper::VoIPControllerWrapper(){
 	controller=new VoIPController();
 	controller->implData=(void*)this;
 	controller->SetStateCallback(VoIPControllerWrapper::OnStateChanged);
+	controller->SetSignalBarsCountCallback(VoIPControllerWrapper::OnSignalBarsChanged);
 	stateCallback=nullptr;
 }
 
@@ -136,11 +137,20 @@ void VoIPControllerWrapper::OnStateChanged(VoIPController* c, int state){
 	reinterpret_cast<VoIPControllerWrapper^>(c->implData)->OnStateChangedInternal(state);
 }
 
+void VoIPControllerWrapper::OnSignalBarsChanged(VoIPController* c, int count){
+	reinterpret_cast<VoIPControllerWrapper^>(c->implData)->OnSignalBarsChangedInternal(count);
+}
+
 void VoIPControllerWrapper::OnStateChangedInternal(int state){
 	if(stateCallback)
 		stateCallback->OnCallStateChanged((CallState)state);
 }
 
+void VoIPControllerWrapper::OnSignalBarsChangedInternal(int count){
+	if(stateCallback)
+		stateCallback->OnSignalBarsChanged(count);
+}
+
 void VoIPControllerWrapper::SetConfig(double initTimeout, double recvTimeout, DataSavingMode dataSavingMode, bool enableAEC, bool enableNS, bool enableAGC, Platform::String^ logFilePath, Platform::String^ statsDumpFilePath){
 	voip_config_t config{0};
 	config.init_timeout=initTimeout;
@@ -170,6 +180,10 @@ void VoIPControllerWrapper::SetProxy(ProxyProtocol protocol, Platform::String^ a
 	controller->SetProxy((int)protocol, _address, port, _username, _password);
 }
 
+void VoIPControllerWrapper::SetAudioOutputGainControlEnabled(bool enabled){
+	controller->SetAudioOutputGainControlEnabled(enabled);
+}
+
 void VoIPControllerWrapper::UpdateServerConfig(Platform::String^ json){
 	JsonObject^ jconfig=JsonValue::Parse(json)->GetObject();
 	std::map<std::string, std::string> config;
diff --git a/os/windows/CXWrapper.h b/os/windows/CXWrapper.h
index d00d5fb..38e0f0b 100755
--- a/os/windows/CXWrapper.h
+++ b/os/windows/CXWrapper.h
@@ -60,6 +60,7 @@ namespace libtgvoip{
 
 	public interface class IStateCallback{
 		void OnCallStateChanged(CallState newState);
+		void OnSignalBarsChanged(int count);
 	};
 
     public ref class VoIPControllerWrapper sealed{
@@ -80,12 +81,15 @@ namespace libtgvoip{
 		Error GetLastError();
 		static Platform::String^ GetVersion();
 		int64 GetPreferredRelayID();
+		void SetAudioOutputGainControlEnabled(bool enabled);
 		static void UpdateServerConfig(Platform::String^ json);
 		static void SwitchSpeaker(bool external);
 		//static Platform::String^ TestAesIge();
 	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;
     };