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

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This commit is contained in:
John Preston 2017-11-23 09:39:42 +04:00
commit bd2a34476c
15 changed files with 195 additions and 29 deletions
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80
EchoCanceller.cpp
View File

@ -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);
}

26
EchoCanceller.h
View File

@ -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

14
OpusDecoder.cpp
View File

@ -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);
}

4
OpusDecoder.h
View File

@ -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;
};
}

38
VoIPController.cpp
View File

@ -233,6 +233,9 @@ 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);
@ -263,6 +266,8 @@ VoIPController::VoIPController() : activeNetItfName(""),
stm->enabled=1;
stm->frameDuration=60;
outgoingStreams.push_back(stm);
memset(signalBarsHistory, 0, sizeof(signalBarsHistory));
}
VoIPController::~VoIPController(){
@ -359,6 +364,8 @@ VoIPController::~VoIPController(){
if(resolvedProxyAddress)
delete resolvedProxyAddress;
delete selectCanceller;
if(outputAGC)
delete outputAGC;
LOGD("Left VoIPController::~VoIPController");
}
@ -1165,10 +1172,13 @@ 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));
@ -1226,7 +1236,7 @@ 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);
@ -1359,7 +1369,7 @@ void VoIPController::RunTickThread(){
#else
Sleep(100);
#endif
int prevSignalBarCount=signalBarCount;
int prevSignalBarCount=GetSignalBarsCount();
signalBarCount=4;
tickCount++;
if(connectionInitTime==0)
@ -1542,12 +1552,12 @@ 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;
@ -1694,10 +1704,12 @@ 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);
}
@ -2443,13 +2455,23 @@ 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;

15
VoIPController.h
View File

@ -29,7 +29,7 @@
#include "CongestionControl.h"
#include "NetworkSocket.h"
#define LIBTGVOIP_VERSION "1.0"
#define LIBTGVOIP_VERSION "1.0.1"
#define STATE_WAIT_INIT 1
#define STATE_WAIT_INIT_ACK 2
@ -317,7 +317,7 @@ 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
@ -334,6 +334,13 @@ 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{
@ -451,6 +458,7 @@ private:
double setEstablishedAt;
SocketSelectCanceller* selectCanceller;
NetworkSocket* openingTcpSocket;
unsigned char signalBarsHistory[4];
BufferPool outgoingPacketsBufferPool;
int udpConnectivityState;
@ -466,6 +474,9 @@ private:
int signalBarCount;
void (*signalBarCountCallback)(VoIPController*, int);
AutomaticGainControl* outputAGC;
bool outputAGCEnabled;
/*** server config values ***/
uint32_t maxAudioBitrate;

4
audio/AudioOutput.cpp
View File

@ -37,9 +37,9 @@ int32_t AudioOutput::estimatedDelay=60;
AudioOutput *AudioOutput::Create(std::string deviceID){
#if defined(__ANDROID__)
if(systemVersion<21)
//if(systemVersion<21)
return new AudioOutputAndroid();
return new AudioOutputOpenSLES();
//return new AudioOutputOpenSLES();
#elif defined(__APPLE__)
#if TARGET_OS_OSX
if(kCFCoreFoundationVersionNumber<kCFCoreFoundationVersionNumber10_7)

4
client/android/tg_voip_jni.cpp
View File

@ -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));
}

7
logging.h
View File

@ -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

11
os/darwin/AudioOutputAudioUnit.cpp
View File

@ -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);

12
os/darwin/AudioUnitIO.cpp
View File

@ -14,6 +14,12 @@
#define CHECK_AU_ERROR(res, msg) if(res!=noErr){ LOGE(msg": OSStatus=%d", (int)res); failed=true; return; }
#define BUFFER_SIZE 960 // 20 ms
#if TARGET_OS_OSX
#define INPUT_BUFFER_SIZE 20480
#else
#define INPUT_BUFFER_SIZE 10240
#endif
#define kOutputBus 0
#define kInputBus 1
@ -30,8 +36,8 @@ AudioUnitIO::AudioUnitIO(){
outputEnabled=false;
failed=false;
started=false;
inBufferList.mBuffers[0].mData=malloc(10240);
inBufferList.mBuffers[0].mDataByteSize=10240;
inBufferList.mBuffers[0].mData=malloc(INPUT_BUFFER_SIZE);
inBufferList.mBuffers[0].mDataByteSize=INPUT_BUFFER_SIZE;
inBufferList.mNumberBuffers=1;
OSStatus status;
@ -159,7 +165,7 @@ void AudioUnitIO::BufferCallback(AudioUnitRenderActionFlags *ioActionFlags, cons
memset(ioData->mBuffers[0].mData, 0, ioData->mBuffers[0].mDataByteSize);
}
}else if(bus==kInputBus){
inBufferList.mBuffers[0].mDataByteSize=10240;
inBufferList.mBuffers[0].mDataByteSize=INPUT_BUFFER_SIZE;
AudioUnitRender(unit, ioActionFlags, inTimeStamp, bus, numFrames, &inBufferList);
if(input && inputEnabled){
input->HandleBufferCallback(&inBufferList);

2
os/linux/AudioInputALSA.cpp
View File

@ -15,7 +15,7 @@
using namespace tgvoip::audio;
#define BUFFER_SIZE 960
#define CHECK_ERROR(res, msg) if(res<0){LOGE(msg ": %s", _snd_strerror(res));}
#define CHECK_ERROR(res, msg) if(res<0){LOGE(msg ": %s", _snd_strerror(res)); failed=true; return;}
#define CHECK_DL_ERROR(res, msg) if(!res){LOGE(msg ": %s", dlerror()); failed=true; return;}
#define LOAD_FUNCTION(lib, name, ref) {ref=(typeof(ref))dlsym(lib, name); CHECK_DL_ERROR(ref, "Error getting entry point for " name);}

2
os/linux/AudioOutputALSA.cpp
View File

@ -12,7 +12,7 @@
#include "../../VoIPController.h"
#define BUFFER_SIZE 960
#define CHECK_ERROR(res, msg) if(res<0){LOGE(msg ": %s", _snd_strerror(res));}
#define CHECK_ERROR(res, msg) if(res<0){LOGE(msg ": %s", _snd_strerror(res)); failed=true; return;}
#define CHECK_DL_ERROR(res, msg) if(!res){LOGE(msg ": %s", dlerror()); failed=true; return;}
#define LOAD_FUNCTION(lib, name, ref) {ref=(typeof(ref))dlsym(lib, name); CHECK_DL_ERROR(ref, "Error getting entry point for " name);}

4
os/windows/CXWrapper.cpp
View File

@ -180,6 +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;

1
os/windows/CXWrapper.h
View File

@ -81,6 +81,7 @@ 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();