mirror of
https://github.com/danog/libtgvoip.git
synced 2024-12-02 09:37:52 +01:00
Merge branch 'public' of https://github.com/grishka/libtgvoip into public
This commit is contained in:
Daniil Gentili
commit
c5fa50730f
@ -97,12 +97,12 @@ EchoCanceller::EchoCanceller(bool enableAEC, bool enableNS, bool enableAGC){
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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.compressionGaindB = 20;
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agcConfig.limiterEnable = 1;
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agcConfig.targetLevelDbfs = 3;
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WebRtcAgc_Init(agc, 0, 255, kAgcModeAdaptiveAnalog, 48000);
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agcConfig.targetLevelDbfs = 9;
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WebRtcAgc_Init(agc, 0, 255, kAgcModeAdaptiveDigital, 48000);
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WebRtcAgc_set_config(agc, agcConfig);
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agcMicLevel=128;
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agcMicLevel=0;
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}
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#endif
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}
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@ -354,3 +354,75 @@ void EchoCanceller::ProcessInput(unsigned char* data, unsigned char* out, size_t
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memcpy(samplesOut, bufIn->ibuf_const()->bands(0)[0], 960*2);
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}
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AudioEffect::~AudioEffect(){
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}
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void AudioEffect::SetPassThrough(bool passThrough){
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this->passThrough=passThrough;
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}
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AutomaticGainControl::AutomaticGainControl(){
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splittingFilter=new webrtc::SplittingFilter(1, 3, 960);
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splittingFilterIn=new webrtc::IFChannelBuffer(960, 1, 1);
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splittingFilterOut=new webrtc::IFChannelBuffer(960, 1, 3);
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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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agcConfig.targetLevelDbfs = 3;
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WebRtcAgc_Init(agc, 0, 255, kAgcModeAdaptiveDigital, 48000);
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WebRtcAgc_set_config(agc, agcConfig);
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agcMicLevel=0;
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}
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AutomaticGainControl::~AutomaticGainControl(){
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delete (webrtc::SplittingFilter*)splittingFilter;
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delete (webrtc::IFChannelBuffer*)splittingFilterIn;
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delete (webrtc::IFChannelBuffer*)splittingFilterOut;
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WebRtcAgc_Free(agc);
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}
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void AutomaticGainControl::Process(int16_t *inOut, size_t numSamples){
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if(passThrough)
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return;
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if(numSamples!=960){
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LOGW("AutomaticGainControl only works on 960-sample buffers (got %u samples)", numSamples);
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return;
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}
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//LOGV("processing frame through AGC");
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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], inOut, 960*2);
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((webrtc::SplittingFilter*)splittingFilter)->Analysis(bufIn, bufOut);
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int i;
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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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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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((webrtc::SplittingFilter*)splittingFilter)->Synthesis(bufOut, bufIn);
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memcpy(inOut, bufIn->ibuf_const()->bands(0)[0], 960*2);
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}
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@ -10,6 +10,7 @@
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#include "threading.h"
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#include "BufferPool.h"
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#include "BlockingQueue.h"
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#include "MediaStreamItf.h"
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namespace tgvoip{
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class EchoCanceller{
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@ -48,6 +49,29 @@ private:
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int32_t agcMicLevel;
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#endif
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};
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}
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class AudioEffect{
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public:
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virtual ~AudioEffect()=0;
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virtual void Process(int16_t* inOut, size_t numSamples)=0;
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virtual void SetPassThrough(bool passThrough);
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protected:
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bool passThrough;
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};
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class AutomaticGainControl : public AudioEffect{
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public:
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AutomaticGainControl();
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virtual ~AutomaticGainControl();
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virtual void Process(int16_t* inOut, size_t numSamples);
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private:
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void* agc;
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void* splittingFilter;
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void* splittingFilterIn;
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void* splittingFilterOut;
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int32_t agcMicLevel;
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};
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};
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#endif //LIBTGVOIP_ECHOCANCELLER_H
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@ -8,6 +8,7 @@
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#include "audio/Resampler.h"
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#include "logging.h"
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#include <assert.h>
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#include <algorithm>
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#define PACKET_SIZE (960*2)
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@ -222,6 +223,9 @@ void tgvoip::OpusDecoder::RunThread(){
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unsigned char *buf=bufferPool->Get();
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if(buf){
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if(size>0){
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for(std::vector<AudioEffect*>::iterator effect=postProcEffects.begin();effect!=postProcEffects.end();++effect){
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(*effect)->Process(reinterpret_cast<int16_t*>(processedBuffer+(PACKET_SIZE*i)), 960);
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}
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memcpy(buf, processedBuffer+(PACKET_SIZE*i), PACKET_SIZE);
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}else{
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LOGE("Error decoding, result=%d", size);
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@ -255,3 +259,13 @@ void tgvoip::OpusDecoder::ResetQueue(){
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void tgvoip::OpusDecoder::SetJitterBuffer(JitterBuffer* jitterBuffer){
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this->jitterBuffer=jitterBuffer;
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}
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void tgvoip::OpusDecoder::AddAudioEffect(AudioEffect *effect){
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postProcEffects.push_back(effect);
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}
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void tgvoip::OpusDecoder::RemoveAudioEffect(AudioEffect *effect){
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std::vector<AudioEffect*>::iterator i=std::find(postProcEffects.begin(), postProcEffects.end(), effect);
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if(i!=postProcEffects.end())
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postProcEffects.erase(i);
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}
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@ -16,6 +16,7 @@
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#include "EchoCanceller.h"
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#include "JitterBuffer.h"
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#include <stdio.h>
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#include <vector>
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namespace tgvoip{
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class OpusDecoder {
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@ -31,6 +32,8 @@ public:
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void SetFrameDuration(uint32_t duration);
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void ResetQueue();
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void SetJitterBuffer(JitterBuffer* jitterBuffer);
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void AddAudioEffect(AudioEffect* effect);
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void RemoveAudioEffect(AudioEffect* effect);
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private:
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static size_t Callback(unsigned char* data, size_t len, void* param);
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@ -50,6 +53,7 @@ private:
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uint32_t frameDuration;
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EchoCanceller* echoCanceller;
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JitterBuffer* jitterBuffer;
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std::vector<AudioEffect*> postProcEffects;
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};
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}
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@ -234,6 +234,9 @@ VoIPController::VoIPController() : activeNetItfName(""),
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realUdpSocket=udpSocket;
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udpConnectivityState=UDP_UNKNOWN;
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outputAGC=NULL;
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outputAGCEnabled=false;
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maxAudioBitrate=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_max_bitrate", 20000);
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maxAudioBitrateGPRS=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_max_bitrate_gprs", 8000);
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maxAudioBitrateEDGE=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_max_bitrate_edge", 16000);
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@ -264,6 +267,8 @@ VoIPController::VoIPController() : activeNetItfName(""),
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stm->enabled=1;
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stm->frameDuration=60;
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outgoingStreams.push_back(stm);
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memset(signalBarsHistory, 0, sizeof(signalBarsHistory));
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}
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VoIPController::~VoIPController(){
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@ -361,6 +366,8 @@ VoIPController::~VoIPController(){
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if(resolvedProxyAddress)
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delete resolvedProxyAddress;
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delete selectCanceller;
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if(outputAGC)
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delete outputAGC;
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LOGD("Left VoIPController::~VoIPController");
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}
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@ -1182,10 +1189,13 @@ simpleAudioBlock random_id:long random_bytes:string raw_data:string = DecryptedA
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UpdateAudioBitrate();
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jitterBuffer=new JitterBuffer(NULL, incomingAudioStream->frameDuration);
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outputAGC=new AutomaticGainControl();
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outputAGC->SetPassThrough(!outputAGCEnabled);
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decoder=new OpusDecoder(audioOutput);
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decoder->SetEchoCanceller(echoCanceller);
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decoder->SetJitterBuffer(jitterBuffer);
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decoder->SetFrameDuration(incomingAudioStream->frameDuration);
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decoder->AddAudioEffect(outputAGC);
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decoder->Start();
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if(incomingAudioStream->frameDuration>50)
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jitterBuffer->SetMinPacketCount((uint32_t) ServerConfig::GetSharedInstance()->GetInt("jitter_initial_delay_60", 3));
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@ -1243,7 +1253,7 @@ simpleAudioBlock random_id:long random_bytes:string raw_data:string = DecryptedA
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audioOutput->Start();
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audioOutStarted=true;
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}
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if(jitterBuffer)
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if(jitterBuffer && in.Remaining()>=sdlen)
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jitterBuffer->HandleInput((unsigned char*) (buffer+in.GetOffset()), sdlen, pts);
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if(i<count-1)
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in.Seek(in.GetOffset()+sdlen);
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@ -1376,7 +1386,7 @@ void VoIPController::RunTickThread(){
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#else
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Sleep(100);
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#endif
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int prevSignalBarCount=signalBarCount;
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int prevSignalBarCount=GetSignalBarsCount();
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signalBarCount=4;
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tickCount++;
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if(connectionInitTime==0)
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@ -1559,12 +1569,12 @@ void VoIPController::RunTickThread(){
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double avgDelay=jitterBuffer->GetAverageDelay();
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double avgLateCount[3];
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jitterBuffer->GetAverageLateCount(avgLateCount);
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if(avgDelay>=5)
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/*if(avgDelay>=5)
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signalBarCount=1;
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else if(avgDelay>=4)
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signalBarCount=MIN(signalBarCount, 2);
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else if(avgDelay>=3)
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signalBarCount=MIN(signalBarCount, 3);
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signalBarCount=MIN(signalBarCount, 3);*/
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if(avgLateCount[2]>=0.2)
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signalBarCount=1;
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@ -1711,10 +1721,12 @@ void VoIPController::RunTickThread(){
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setEstablishedAt=0;
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}
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if(signalBarCount!=prevSignalBarCount){
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LOGD("SIGNAL BAR COUNT CHANGED: %d", signalBarCount);
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signalBarsHistory[tickCount%sizeof(signalBarsHistory)]=(unsigned char)signalBarCount;
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int _signalBarCount=GetSignalBarsCount();
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if(_signalBarCount!=prevSignalBarCount){
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LOGD("SIGNAL BAR COUNT CHANGED: %d", _signalBarCount);
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if(signalBarCountCallback)
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signalBarCountCallback(this, signalBarCount);
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signalBarCountCallback(this, _signalBarCount);
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}
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||||
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@ -2460,13 +2472,23 @@ void VoIPController::SendUdpPing(Endpoint *endpoint){
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}
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int VoIPController::GetSignalBarsCount(){
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return signalBarCount;
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unsigned char avg=0;
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for(int i=0;i<sizeof(signalBarsHistory);i++)
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avg+=signalBarsHistory[i];
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return avg >> 2;
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}
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void VoIPController::SetSignalBarsCountCallback(void (*f)(VoIPController *, int)){
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signalBarCountCallback=f;
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}
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void VoIPController::SetAudioOutputGainControlEnabled(bool enabled){
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LOGD("New output AGC state: %d", enabled);
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outputAGCEnabled=enabled;
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if(outputAGC)
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outputAGC->SetPassThrough(!enabled);
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}
|
||||
|
||||
Endpoint::Endpoint(int64_t id, uint16_t port, IPv4Address& _address, IPv6Address& _v6address, char type, unsigned char peerTag[16]) : address(_address), v6address(_v6address){
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this->id=id;
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this->port=port;
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||||
|
||||
@ -318,7 +318,7 @@ public:
|
||||
std::string GetCurrentAudioOutputID();
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||||
/**
|
||||
* 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.
|
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* 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;
|
||||
|
||||
@ -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));
|
||||
}
|
||||
|
||||
@ -47,12 +47,8 @@
|
||||
692AB9201E675F7000706ACC /* AudioUnit.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 692AB91D1E675F7000706ACC /* AudioUnit.framework */; };
|
||||
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|
||||
695B20621EBD39FF00E31757 /* DarwinSpecific.h in Headers */ = {isa = PBXBuildFile; fileRef = 695B20601EBD39FF00E31757 /* DarwinSpecific.h */; };
|
||||
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||||
698848411F4B39F700076DF0 /* AudioInputAudioUnit.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 6988483B1F4B39F700076DF0 /* AudioInputAudioUnit.cpp */; };
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||||
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||||
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||||
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||||
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@ -189,8 +185,12 @@
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||||
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@ -1264,9 +1282,9 @@
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69F8423F1E67540700C110F7 /* Debug */ = {
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@ -1303,9 +1321,9 @@
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69F842401E67540700C110F7 /* Release Hockeyapp */ = {
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buildSettings = {
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CLANG_CXX_LANGUAGE_STANDARD = "c++0x";
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@ -1342,7 +1360,7 @@
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||||
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||||
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||||
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||||
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@ -1434,7 +1452,7 @@
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||||
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D04D01CB1E678C230086DDC0 /* Hockeyapp */ = {
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||||
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||||
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||||
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||||
@ -1477,9 +1495,9 @@
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||||
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||||
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||||
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||||
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||||
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||||
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||||
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||||
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||||
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||||
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||||
@ -1516,7 +1534,7 @@
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||||
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||||
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||||
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||||
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||||
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||||
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||||
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||||
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||||
@ -1524,24 +1542,24 @@
|
||||
69F842301E67540700C110F7 /* Build configuration list for PBXProject "libtgvoip_osx" */ = {
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||||
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||||
buildConfigurations = (
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||||
69F8423C1E67540700C110F7 /* Debug */,
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||||
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||||
D04D01C31E678C0D0086DDC0 /* Debug AppStore */,
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||||
69F8423D1E67540700C110F7 /* Release */,
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||||
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||||
69F8423D1E67540700C110F7 /* Release Hockeyapp */,
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||||
D04D01CB1E678C230086DDC0 /* Release AppStore */,
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||||
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|
||||
defaultConfigurationIsVisible = 0;
|
||||
defaultConfigurationName = Release;
|
||||
defaultConfigurationName = "Release Hockeyapp";
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||||
};
|
||||
69F8423E1E67540700C110F7 /* Build configuration list for PBXNativeTarget "libtgvoip" */ = {
|
||||
isa = XCConfigurationList;
|
||||
buildConfigurations = (
|
||||
69F8423F1E67540700C110F7 /* Debug */,
|
||||
69F8423F1E67540700C110F7 /* Debug Hockeyapp */,
|
||||
D04D01C41E678C0D0086DDC0 /* Debug AppStore */,
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||||
69F842401E67540700C110F7 /* Release */,
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||||
D04D01CC1E678C230086DDC0 /* Hockeyapp */,
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||||
69F842401E67540700C110F7 /* Release Hockeyapp */,
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||||
D04D01CC1E678C230086DDC0 /* Release AppStore */,
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||||
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||||
defaultConfigurationIsVisible = 0;
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||||
defaultConfigurationName = Release;
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||||
defaultConfigurationName = "Release Hockeyapp";
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||||
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||||
/* End XCConfigurationList section */
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||||
};
|
||||
|
||||
@ -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
|
||||
|
||||
@ -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);
|
||||
|
||||
@ -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;
|
||||
|
||||
@ -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;
|
||||
};
|
||||
|
||||
Loading…
Reference in New Issue
Block a user