mirror of
https://github.com/danog/libtgvoip.git
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883 lines
25 KiB
C++
Executable File
883 lines
25 KiB
C++
Executable File
//
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// libtgvoip is free and unencumbered public domain software.
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// For more information, see http://unlicense.org or the UNLICENSE file
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// you should have received with this source code distribution.
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//
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#ifndef __VOIPCONTROLLER_H
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#define __VOIPCONTROLLER_H
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#ifndef _WIN32
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#include <arpa/inet.h>
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#include <netinet/in.h>
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#endif
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#ifdef __APPLE__
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#include <TargetConditionals.h>
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#include "os/darwin/AudioUnitIO.h"
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#endif
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#include <stdint.h>
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#include <vector>
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#include <string>
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#include <unordered_map>
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#include <map>
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#include <memory>
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#include "video/VideoSource.h"
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#include "video/VideoRenderer.h"
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#include <atomic>
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#include "video/ScreamCongestionController.h"
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#include "audio/AudioInput.h"
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#include "BlockingQueue.h"
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#include "audio/AudioOutput.h"
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#include "audio/AudioIO.h"
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#include "JitterBuffer.h"
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#include "OpusDecoder.h"
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#include "OpusEncoder.h"
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#include "EchoCanceller.h"
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#include "CongestionControl.h"
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#include "NetworkSocket.h"
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#include "Buffers.h"
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#include "PacketReassembler.h"
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#include "MessageThread.h"
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#include "utils.h"
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#define LIBTGVOIP_VERSION "2.4.4"
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#ifdef _WIN32
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#undef GetCurrentTime
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#undef ERROR_TIMEOUT
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#endif
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#define TGVOIP_PEER_CAP_GROUP_CALLS 1
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#define TGVOIP_PEER_CAP_VIDEO_CAPTURE 2
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#define TGVOIP_PEER_CAP_VIDEO_DISPLAY 4
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namespace tgvoip{
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enum{
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PROXY_NONE=0,
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PROXY_SOCKS5,
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//PROXY_HTTP
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};
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enum{
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STATE_WAIT_INIT=1,
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STATE_WAIT_INIT_ACK,
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STATE_ESTABLISHED,
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STATE_FAILED,
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STATE_RECONNECTING
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};
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enum{
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ERROR_UNKNOWN=0,
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ERROR_INCOMPATIBLE,
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ERROR_TIMEOUT,
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ERROR_AUDIO_IO,
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ERROR_PROXY
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};
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enum{
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NET_TYPE_UNKNOWN=0,
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NET_TYPE_GPRS,
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NET_TYPE_EDGE,
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NET_TYPE_3G,
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NET_TYPE_HSPA,
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NET_TYPE_LTE,
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NET_TYPE_WIFI,
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NET_TYPE_ETHERNET,
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NET_TYPE_OTHER_HIGH_SPEED,
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NET_TYPE_OTHER_LOW_SPEED,
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NET_TYPE_DIALUP,
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NET_TYPE_OTHER_MOBILE
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};
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enum{
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DATA_SAVING_NEVER=0,
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DATA_SAVING_MOBILE,
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DATA_SAVING_ALWAYS
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};
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struct CryptoFunctions{
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void (*rand_bytes)(uint8_t* buffer, size_t length);
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void (*sha1)(uint8_t* msg, size_t length, uint8_t* output);
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void (*sha256)(uint8_t* msg, size_t length, uint8_t* output);
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void (*aes_ige_encrypt)(uint8_t* in, uint8_t* out, size_t length, uint8_t* key, uint8_t* iv);
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void (*aes_ige_decrypt)(uint8_t* in, uint8_t* out, size_t length, uint8_t* key, uint8_t* iv);
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void (*aes_ctr_encrypt)(uint8_t* inout, size_t length, uint8_t* key, uint8_t* iv, uint8_t* ecount, uint32_t* num);
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void (*aes_cbc_encrypt)(uint8_t* in, uint8_t* out, size_t length, uint8_t* key, uint8_t* iv);
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void (*aes_cbc_decrypt)(uint8_t* in, uint8_t* out, size_t length, uint8_t* key, uint8_t* iv);
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};
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struct CellularCarrierInfo{
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std::string name;
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std::string mcc;
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std::string mnc;
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std::string countryCode;
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};
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class Endpoint{
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friend class VoIPController;
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friend class VoIPGroupController;
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public:
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enum Type{
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UDP_P2P_INET=1,
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UDP_P2P_LAN,
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UDP_RELAY,
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TCP_RELAY
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};
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Endpoint(int64_t id, uint16_t port, const IPv4Address& address, const IPv6Address& v6address, Type type, unsigned char* peerTag);
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Endpoint();
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~Endpoint();
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const NetworkAddress& GetAddress() const;
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NetworkAddress& GetAddress();
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bool IsIPv6Only() const;
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int64_t id;
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uint16_t port;
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IPv4Address address;
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IPv6Address v6address;
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Type type;
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unsigned char peerTag[16];
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private:
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double lastPingTime;
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uint32_t lastPingSeq;
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HistoricBuffer<double, 6> rtts;
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double averageRTT;
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NetworkSocket* socket;
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int udpPongCount;
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};
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class AudioDevice{
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public:
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std::string id;
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std::string displayName;
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};
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class AudioOutputDevice : public AudioDevice{
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};
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class AudioInputDevice : public AudioDevice{
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};
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class AudioInputTester{
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public:
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AudioInputTester(const std::string deviceID);
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~AudioInputTester();
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TGVOIP_DISALLOW_COPY_AND_ASSIGN(AudioInputTester);
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float GetAndResetLevel();
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bool Failed(){
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return io && io->Failed();
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}
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private:
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void Update(int16_t* samples, size_t count);
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audio::AudioIO* io=NULL;
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audio::AudioInput* input=NULL;
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int16_t maxSample=0;
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std::string deviceID;
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};
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class VoIPController{
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friend class VoIPGroupController;
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public:
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TGVOIP_DISALLOW_COPY_AND_ASSIGN(VoIPController);
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struct Config{
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Config(double initTimeout=30.0, double recvTimeout=20.0, int dataSaving=DATA_SAVING_NEVER, bool enableAEC=false, bool enableNS=false, bool enableAGC=false, bool enableCallUpgrade=false){
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this->initTimeout=initTimeout;
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this->recvTimeout=recvTimeout;
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this->dataSaving=dataSaving;
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this->enableAEC=enableAEC;
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this->enableNS=enableNS;
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this->enableAGC=enableAGC;
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this->enableCallUpgrade=enableCallUpgrade;
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}
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double initTimeout;
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double recvTimeout;
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int dataSaving;
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#ifndef _WIN32
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std::string logFilePath="";
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std::string statsDumpFilePath="";
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#else
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std::wstring logFilePath=L"";
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std::wstring statsDumpFilePath=L"";
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#endif
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bool enableAEC;
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bool enableNS;
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bool enableAGC;
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bool enableCallUpgrade;
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bool logPacketStats=false;
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bool enableVolumeControl=false;
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bool enableVideoSend=false;
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bool enableVideoReceive=false;
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};
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struct TrafficStats{
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uint64_t bytesSentWifi;
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uint64_t bytesRecvdWifi;
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uint64_t bytesSentMobile;
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uint64_t bytesRecvdMobile;
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};
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VoIPController();
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virtual ~VoIPController();
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/**
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* Set the initial endpoints (relays)
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* @param endpoints Endpoints converted from phone.PhoneConnection TL objects
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* @param allowP2p Whether p2p connectivity is allowed
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* @param connectionMaxLayer The max_layer field from the phoneCallProtocol object returned by Telegram server.
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* DO NOT HARDCODE THIS VALUE, it's extremely important for backwards compatibility.
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*/
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void SetRemoteEndpoints(std::vector<Endpoint> endpoints, bool allowP2p, int32_t connectionMaxLayer);
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/**
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* Initialize and start all the internal threads
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*/
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void Start();
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/**
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* Stop any internal threads. Don't call any other methods after this.
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*/
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void Stop();
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/**
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* Initiate connection
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*/
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void Connect();
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Endpoint& GetRemoteEndpoint();
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/**
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* Get the debug info string to be displayed in client UI
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*/
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virtual std::string GetDebugString();
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/**
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* Notify the library of network type change
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* @param type The new network type
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*/
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virtual void SetNetworkType(int type);
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/**
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* Get the average round-trip time for network packets
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* @return
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*/
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double GetAverageRTT();
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static double GetCurrentTime();
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/**
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* Use this field to store any of your context data associated with this call
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*/
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void* implData;
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/**
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*
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* @param mute
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*/
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virtual void SetMicMute(bool mute);
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/**
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*
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* @param key
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* @param isOutgoing
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*/
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void SetEncryptionKey(char* key, bool isOutgoing);
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/**
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*
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* @param cfg
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*/
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void SetConfig(const Config& cfg);
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void DebugCtl(int request, int param);
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/**
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*
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* @param stats
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*/
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void GetStats(TrafficStats* stats);
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/**
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*
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* @return
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*/
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int64_t GetPreferredRelayID();
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/**
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*
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* @return
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*/
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int GetLastError();
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/**
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*
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*/
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static CryptoFunctions crypto;
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/**
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*
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* @return
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*/
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static const char* GetVersion();
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/**
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*
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* @return
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*/
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std::string GetDebugLog();
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/**
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*
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* @return
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*/
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static std::vector<AudioInputDevice> EnumerateAudioInputs();
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/**
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*
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* @return
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*/
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static std::vector<AudioOutputDevice> EnumerateAudioOutputs();
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/**
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*
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* @param id
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*/
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void SetCurrentAudioInput(std::string id);
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/**
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*
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* @param id
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*/
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void SetCurrentAudioOutput(std::string id);
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/**
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*
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* @return
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*/
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std::string GetCurrentAudioInputID();
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/**
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*
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* @return
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*/
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std::string GetCurrentAudioOutputID();
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/**
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* Set the proxy server to route the data through. Call this before connecting.
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* @param protocol PROXY_NONE or PROXY_SOCKS5
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* @param address IP address or domain name of the server
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* @param port Port of the server
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* @param username Username; empty string for anonymous
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* @param password Password; empty string if none
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*/
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void SetProxy(int protocol, std::string address, uint16_t port, std::string username, std::string password);
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/**
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* Get the number of signal bars to display in the client UI.
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* @return the number of signal bars, from 1 to 4
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*/
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int GetSignalBarsCount();
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/**
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* Enable or disable AGC (automatic gain control) on audio output. Should only be enabled on phones when the earpiece speaker is being used.
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* 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.
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* @param enabled I usually pick argument names to be self-explanatory
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*/
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void SetAudioOutputGainControlEnabled(bool enabled);
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/**
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* Get the additional capabilities of the peer client app
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* @return corresponding TGVOIP_PEER_CAP_* flags OR'ed together
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*/
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uint32_t GetPeerCapabilities();
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/**
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* Send the peer the key for the group call to prepare this private call to an upgrade to a E2E group call.
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* The peer must have the TGVOIP_PEER_CAP_GROUP_CALLS capability. After the peer acknowledges the key, Callbacks::groupCallKeySent will be called.
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* @param key newly-generated group call key, must be exactly 265 bytes long
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*/
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void SendGroupCallKey(unsigned char* key);
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/**
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* In an incoming call, request the peer to generate a new encryption key, send it to you and upgrade this call to a E2E group call.
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*/
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void RequestCallUpgrade();
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void SetEchoCancellationStrength(int strength);
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int GetConnectionState();
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bool NeedRate();
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/**
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* Get the maximum connection layer supported by this libtgvoip version.
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* Pass this as <code>max_layer</code> in the phone.phoneConnection TL object when requesting and accepting calls.
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*/
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static int32_t GetConnectionMaxLayer(){
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return 92;
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};
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/**
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* Get the persistable state of the library, like proxy capabilities, to save somewhere on the disk. Call this at the end of the call.
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* Using this will speed up the connection establishment in some cases.
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*/
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std::vector<uint8_t> GetPersistentState();
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/**
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* Load the persistable state. Call this before starting the call.
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*/
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void SetPersistentState(std::vector<uint8_t> state);
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#if defined(TGVOIP_USE_CALLBACK_AUDIO_IO)
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void SetAudioDataCallbacks(std::function<void(int16_t*, size_t)> input, std::function<void(int16_t*, size_t)> output, std::function<void(int16_t*, size_t)> preprocessed);
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#endif
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void SetVideoCodecSpecificData(const std::vector<Buffer>& data);
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struct Callbacks{
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void (*connectionStateChanged)(VoIPController*, int);
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void (*signalBarCountChanged)(VoIPController*, int);
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void (*groupCallKeySent)(VoIPController*);
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void (*groupCallKeyReceived)(VoIPController*, const unsigned char*);
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void (*upgradeToGroupCallRequested)(VoIPController*);
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};
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void SetCallbacks(Callbacks callbacks);
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float GetOutputLevel(){
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return 0.0f;
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};
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int GetVideoResolutionForCurrentBitrate();
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void SetVideoSource(video::VideoSource* source);
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void SetVideoRenderer(video::VideoRenderer* renderer);
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void SetInputVolume(float level);
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void SetOutputVolume(float level);
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#if defined(__APPLE__) && defined(TARGET_OS_OSX)
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void SetAudioOutputDuckingEnabled(bool enabled);
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#endif
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private:
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struct Stream;
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struct UnacknowledgedExtraData;
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protected:
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struct RecentOutgoingPacket{
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uint32_t seq;
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uint16_t id; // for group calls only
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double sendTime;
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double ackTime;
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uint8_t type;
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uint32_t size;
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};
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struct PendingOutgoingPacket{
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PendingOutgoingPacket(uint32_t seq, unsigned char type, size_t len, Buffer&& data, int64_t endpoint){
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this->seq=seq;
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this->type=type;
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this->len=len;
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this->data=std::move(data);
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this->endpoint=endpoint;
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}
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PendingOutgoingPacket(PendingOutgoingPacket&& other){
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seq=other.seq;
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type=other.type;
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len=other.len;
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data=std::move(other.data);
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endpoint=other.endpoint;
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}
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PendingOutgoingPacket& operator=(PendingOutgoingPacket&& other){
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if(this!=&other){
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seq=other.seq;
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type=other.type;
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len=other.len;
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data=std::move(other.data);
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endpoint=other.endpoint;
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}
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return *this;
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}
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TGVOIP_DISALLOW_COPY_AND_ASSIGN(PendingOutgoingPacket);
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uint32_t seq;
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unsigned char type;
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size_t len;
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Buffer data;
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int64_t endpoint;
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};
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struct QueuedPacket{
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#if defined(_MSC_VER) && _MSC_VER <= 1800 // VS2013 doesn't support auto-generating move constructors
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//TGVOIP_DISALLOW_COPY_AND_ASSIGN(QueuedPacket);
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QueuedPacket(QueuedPacket&& other){
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data=std::move(other.data);
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type=other.type;
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seqs=other.seqs;
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firstSentTime=other.firstSentTime;
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lastSentTime=other.lastSentTime;
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retryInterval=other.retryInterval;
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timeout=other.timeout;
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}
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QueuedPacket(){
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}
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#endif
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Buffer data;
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unsigned char type;
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HistoricBuffer<uint32_t, 16> seqs;
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double firstSentTime;
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double lastSentTime;
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double retryInterval;
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double timeout;
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};
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virtual void ProcessIncomingPacket(NetworkPacket& packet, Endpoint& srcEndpoint);
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virtual void ProcessExtraData(Buffer& data);
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virtual void WritePacketHeader(uint32_t seq, BufferOutputStream* s, unsigned char type, uint32_t length);
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virtual void SendPacket(unsigned char* data, size_t len, Endpoint& ep, PendingOutgoingPacket& srcPacket);
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virtual void SendInit();
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virtual void SendUdpPing(Endpoint& endpoint);
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virtual void SendRelayPings();
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virtual void OnAudioOutputReady();
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virtual void SendExtra(Buffer& data, unsigned char type);
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void SendStreamFlags(Stream& stream);
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void SendStreamCSD(Stream& stream);
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void InitializeTimers();
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void ResetEndpointPingStats();
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void SendVideoFrame(const Buffer& frame, uint32_t flags);
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void ProcessIncomingVideoFrame(Buffer frame, uint32_t pts, bool keyframe);
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std::shared_ptr<Stream> GetStreamByType(int type, bool outgoing);
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Endpoint* GetEndpointForPacket(const PendingOutgoingPacket& pkt);
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bool SendOrEnqueuePacket(PendingOutgoingPacket pkt, bool enqueue=true);
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static std::string NetworkTypeToString(int type);
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CellularCarrierInfo GetCarrierInfo();
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private:
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struct Stream{
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int32_t userID;
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unsigned char id;
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unsigned char type;
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uint32_t codec;
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bool enabled;
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bool extraECEnabled;
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uint16_t frameDuration;
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std::shared_ptr<JitterBuffer> jitterBuffer;
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std::shared_ptr<OpusDecoder> decoder;
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std::shared_ptr<PacketReassembler> packetReassembler;
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std::shared_ptr<CallbackWrapper> callbackWrapper;
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std::vector<Buffer> codecSpecificData;
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bool csdIsValid=false;
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int resolution;
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unsigned int width=0;
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unsigned int height=0;
|
|
uint16_t rotation=0;
|
|
};
|
|
struct UnacknowledgedExtraData{
|
|
#if defined(_MSC_VER) && _MSC_VER <= 1800 // VS2013 doesn't support auto-generating move constructors
|
|
UnacknowledgedExtraData(UnacknowledgedExtraData&& other){
|
|
type=other.type;
|
|
data=std::move(other.data);
|
|
firstContainingSeq=other.firstContainingSeq;
|
|
}
|
|
UnacknowledgedExtraData(unsigned char _type, Buffer&& _data, uint32_t _firstContainingSeq){
|
|
type=_type;
|
|
data=_data;
|
|
firstContainingSeq=_firstContainingSeq;
|
|
}
|
|
#endif
|
|
unsigned char type;
|
|
Buffer data;
|
|
uint32_t firstContainingSeq;
|
|
};
|
|
enum{
|
|
UDP_UNKNOWN=0,
|
|
UDP_PING_PENDING,
|
|
UDP_PING_SENT,
|
|
UDP_AVAILABLE,
|
|
UDP_NOT_AVAILABLE,
|
|
UDP_BAD
|
|
};
|
|
struct DebugLoggedPacket{
|
|
int32_t seq;
|
|
double timestamp;
|
|
int32_t length;
|
|
};
|
|
struct SentVideoFrame{
|
|
uint32_t num;
|
|
uint32_t fragmentCount;
|
|
std::vector<uint32_t> unacknowledgedPackets;
|
|
uint32_t fragmentsInQueue;
|
|
};
|
|
struct PendingVideoFrameFragment{
|
|
uint32_t pts;
|
|
Buffer data;
|
|
};
|
|
|
|
void RunRecvThread();
|
|
void RunSendThread();
|
|
void HandleAudioInput(unsigned char* data, size_t len, unsigned char* secondaryData, size_t secondaryLen);
|
|
void UpdateAudioBitrateLimit();
|
|
void SetState(int state);
|
|
void UpdateAudioOutputState();
|
|
void InitUDPProxy();
|
|
void UpdateDataSavingState();
|
|
void KDF(unsigned char* msgKey, size_t x, unsigned char* aesKey, unsigned char* aesIv);
|
|
void KDF2(unsigned char* msgKey, size_t x, unsigned char* aesKey, unsigned char* aesIv);
|
|
static void AudioInputCallback(unsigned char* data, size_t length, unsigned char* secondaryData, size_t secondaryLength, void* param);
|
|
void SendPublicEndpointsRequest();
|
|
void SendPublicEndpointsRequest(const Endpoint& relay);
|
|
Endpoint& GetEndpointByType(int type);
|
|
void SendPacketReliably(unsigned char type, unsigned char* data, size_t len, double retryInterval, double timeout);
|
|
uint32_t GenerateOutSeq();
|
|
void ActuallySendPacket(NetworkPacket& pkt, Endpoint& ep);
|
|
void InitializeAudio();
|
|
void StartAudio();
|
|
void ProcessAcknowledgedOutgoingExtra(UnacknowledgedExtraData& extra);
|
|
void AddIPv6Relays();
|
|
void AddTCPRelays();
|
|
void SendUdpPings();
|
|
void EvaluateUdpPingResults();
|
|
void UpdateRTT();
|
|
void UpdateCongestion();
|
|
void UpdateAudioBitrate();
|
|
void UpdateSignalBars();
|
|
void UpdateQueuedPackets();
|
|
void SendNopPacket();
|
|
void TickJitterBufferAngCongestionControl();
|
|
void ResetUdpAvailability();
|
|
std::string GetPacketTypeString(unsigned char type);
|
|
void SetupOutgoingVideoStream();
|
|
bool WasOutgoingPacketAcknowledged(uint32_t seq);
|
|
RecentOutgoingPacket* GetRecentOutgoingPacket(uint32_t seq);
|
|
|
|
int state;
|
|
std::map<int64_t, Endpoint> endpoints;
|
|
int64_t currentEndpoint=0;
|
|
int64_t preferredRelay=0;
|
|
int64_t peerPreferredRelay=0;
|
|
bool runReceiver;
|
|
std::atomic<uint32_t> seq;
|
|
uint32_t lastRemoteSeq;
|
|
uint32_t lastRemoteAckSeq;
|
|
uint32_t lastSentSeq;
|
|
std::vector<RecentOutgoingPacket> recentOutgoingPackets;
|
|
double recvPacketTimes[32];
|
|
HistoricBuffer<uint32_t, 10, double> sendLossCountHistory;
|
|
uint32_t audioTimestampIn;
|
|
uint32_t audioTimestampOut;
|
|
tgvoip::audio::AudioIO* audioIO=NULL;
|
|
tgvoip::audio::AudioInput* audioInput=NULL;
|
|
tgvoip::audio::AudioOutput* audioOutput=NULL;
|
|
OpusEncoder* encoder;
|
|
std::vector<PendingOutgoingPacket> sendQueue;
|
|
EchoCanceller* echoCanceller;
|
|
Mutex sendBufferMutex;
|
|
Mutex endpointsMutex;
|
|
Mutex socketSelectMutex;
|
|
bool stopping;
|
|
bool audioOutStarted;
|
|
Thread* recvThread;
|
|
Thread* sendThread;
|
|
uint32_t packetsReceived;
|
|
uint32_t recvLossCount;
|
|
uint32_t prevSendLossCount;
|
|
uint32_t firstSentPing;
|
|
HistoricBuffer<double, 32> rttHistory;
|
|
bool waitingForAcks;
|
|
int networkType;
|
|
int dontSendPackets;
|
|
int lastError;
|
|
bool micMuted;
|
|
uint32_t maxBitrate;
|
|
std::vector<std::shared_ptr<Stream>> outgoingStreams;
|
|
std::vector<std::shared_ptr<Stream>> incomingStreams;
|
|
unsigned char encryptionKey[256];
|
|
unsigned char keyFingerprint[8];
|
|
unsigned char callID[16];
|
|
double stateChangeTime;
|
|
bool waitingForRelayPeerInfo;
|
|
bool allowP2p;
|
|
bool dataSavingMode;
|
|
bool dataSavingRequestedByPeer;
|
|
std::string activeNetItfName;
|
|
double publicEndpointsReqTime;
|
|
std::vector<QueuedPacket> queuedPackets;
|
|
Mutex audioIOMutex;
|
|
Mutex queuedPacketsMutex;
|
|
double connectionInitTime;
|
|
double lastRecvPacketTime;
|
|
Config config;
|
|
int32_t peerVersion;
|
|
CongestionControl* conctl;
|
|
TrafficStats stats;
|
|
bool receivedInit;
|
|
bool receivedInitAck;
|
|
bool isOutgoing;
|
|
NetworkSocket* udpSocket;
|
|
NetworkSocket* realUdpSocket;
|
|
FILE* statsDump;
|
|
std::string currentAudioInput;
|
|
std::string currentAudioOutput;
|
|
bool useTCP;
|
|
bool useUDP;
|
|
bool didAddTcpRelays;
|
|
SocketSelectCanceller* selectCanceller;
|
|
HistoricBuffer<unsigned char, 4, int> signalBarsHistory;
|
|
bool audioStarted=false;
|
|
|
|
int udpConnectivityState;
|
|
double lastUdpPingTime;
|
|
int udpPingCount;
|
|
int echoCancellationStrength;
|
|
|
|
int proxyProtocol;
|
|
std::string proxyAddress;
|
|
uint16_t proxyPort;
|
|
std::string proxyUsername;
|
|
std::string proxyPassword;
|
|
IPv4Address* resolvedProxyAddress;
|
|
|
|
uint32_t peerCapabilities;
|
|
Callbacks callbacks;
|
|
bool didReceiveGroupCallKey;
|
|
bool didReceiveGroupCallKeyAck;
|
|
bool didSendGroupCallKey;
|
|
bool didSendUpgradeRequest;
|
|
bool didInvokeUpgradeCallback;
|
|
|
|
int32_t connectionMaxLayer;
|
|
bool useMTProto2;
|
|
bool setCurrentEndpointToTCP;
|
|
|
|
std::vector<UnacknowledgedExtraData> currentExtras;
|
|
std::unordered_map<uint8_t, uint64_t> lastReceivedExtrasByType;
|
|
bool useIPv6;
|
|
bool peerIPv6Available;
|
|
IPv6Address myIPv6;
|
|
bool shittyInternetMode;
|
|
int extraEcLevel=0;
|
|
std::vector<Buffer> ecAudioPackets;
|
|
bool didAddIPv6Relays;
|
|
bool didSendIPv6Endpoint;
|
|
int publicEndpointsReqCount=0;
|
|
MessageThread messageThread;
|
|
bool wasEstablished=false;
|
|
bool receivedFirstStreamPacket=false;
|
|
std::atomic<unsigned int> unsentStreamPackets;
|
|
HistoricBuffer<unsigned int, 5> unsentStreamPacketsHistory;
|
|
bool needReInitUdpProxy=true;
|
|
bool needRate=false;
|
|
std::vector<DebugLoggedPacket> debugLoggedPackets;
|
|
|
|
uint32_t initTimeoutID=MessageThread::INVALID_ID;
|
|
uint32_t noStreamsNopID=MessageThread::INVALID_ID;
|
|
uint32_t udpPingTimeoutID=MessageThread::INVALID_ID;
|
|
|
|
effects::Volume outputVolume;
|
|
effects::Volume inputVolume;
|
|
|
|
std::vector<uint32_t> peerVideoDecoders;
|
|
int peerMaxVideoResolution=0;
|
|
|
|
#if defined(TGVOIP_USE_CALLBACK_AUDIO_IO)
|
|
std::function<void(int16_t*, size_t)> audioInputDataCallback;
|
|
std::function<void(int16_t*, size_t)> audioOutputDataCallback;
|
|
std::function<void(int16_t*, size_t)> audioPreprocDataCallback;
|
|
::OpusDecoder* preprocDecoder=nullptr;
|
|
int16_t preprocBuffer[4096];
|
|
#endif
|
|
#if defined(__APPLE__) && defined(TARGET_OS_OSX)
|
|
bool macAudioDuckingEnabled=true;
|
|
#endif
|
|
|
|
video::VideoSource* videoSource=NULL;
|
|
video::VideoRenderer* videoRenderer=NULL;
|
|
double firstVideoFrameTime=0.0;
|
|
uint32_t videoFrameCount=0;
|
|
uint32_t lastReceivedVideoFrameNumber=UINT32_MAX;
|
|
std::vector<SentVideoFrame> sentVideoFrames;
|
|
Mutex sentVideoFramesMutex;
|
|
bool videoKeyframeRequested=false;
|
|
video::ScreamCongestionController videoCongestionControl;
|
|
std::vector<PendingVideoFrameFragment> videoPacingQueue;
|
|
uint32_t sendVideoPacketID=MessageThread::INVALID_ID;
|
|
uint32_t videoPacketLossCount=0;
|
|
uint32_t currentVideoBitrate=0;
|
|
double lastVideoResolutionChangeTime=0.0;
|
|
|
|
/*** debug report problems ***/
|
|
bool wasReconnecting=false;
|
|
bool wasExtraEC=false;
|
|
bool wasEncoderLaggy=false;
|
|
bool wasNetworkHandover=false;
|
|
|
|
/*** persistable state values ***/
|
|
bool proxySupportsUDP=true;
|
|
bool proxySupportsTCP=true;
|
|
std::string lastTestedProxyServer="";
|
|
|
|
/*** server config values ***/
|
|
uint32_t maxAudioBitrate;
|
|
uint32_t maxAudioBitrateEDGE;
|
|
uint32_t maxAudioBitrateGPRS;
|
|
uint32_t maxAudioBitrateSaving;
|
|
uint32_t initAudioBitrate;
|
|
uint32_t initAudioBitrateEDGE;
|
|
uint32_t initAudioBitrateGPRS;
|
|
uint32_t initAudioBitrateSaving;
|
|
uint32_t minAudioBitrate;
|
|
uint32_t audioBitrateStepIncr;
|
|
uint32_t audioBitrateStepDecr;
|
|
double relaySwitchThreshold;
|
|
double p2pToRelaySwitchThreshold;
|
|
double relayToP2pSwitchThreshold;
|
|
double reconnectingTimeout;
|
|
uint32_t needRateFlags;
|
|
double rateMaxAcceptableRTT;
|
|
double rateMaxAcceptableSendLoss;
|
|
double packetLossToEnableExtraEC;
|
|
uint32_t maxUnsentStreamPackets;
|
|
|
|
public:
|
|
#ifdef __APPLE__
|
|
static double machTimebase;
|
|
static uint64_t machTimestart;
|
|
#endif
|
|
#ifdef _WIN32
|
|
static int64_t win32TimeScale;
|
|
static bool didInitWin32TimeScale;
|
|
#endif
|
|
};
|
|
|
|
class VoIPGroupController : public VoIPController{
|
|
public:
|
|
VoIPGroupController(int32_t timeDifference);
|
|
virtual ~VoIPGroupController();
|
|
void SetGroupCallInfo(unsigned char* encryptionKey, unsigned char* reflectorGroupTag, unsigned char* reflectorSelfTag, unsigned char* reflectorSelfSecret, unsigned char* reflectorSelfTagHash, int32_t selfUserID, IPv4Address reflectorAddress, IPv6Address reflectorAddressV6, uint16_t reflectorPort);
|
|
void AddGroupCallParticipant(int32_t userID, unsigned char* memberTagHash, unsigned char* serializedStreams, size_t streamsLength);
|
|
void RemoveGroupCallParticipant(int32_t userID);
|
|
float GetParticipantAudioLevel(int32_t userID);
|
|
virtual void SetMicMute(bool mute);
|
|
void SetParticipantVolume(int32_t userID, float volume);
|
|
void SetParticipantStreams(int32_t userID, unsigned char* serializedStreams, size_t length);
|
|
static size_t GetInitialStreams(unsigned char* buf, size_t size);
|
|
|
|
struct Callbacks : public VoIPController::Callbacks{
|
|
void (*updateStreams)(VoIPGroupController*, unsigned char*, size_t);
|
|
void (*participantAudioStateChanged)(VoIPGroupController*, int32_t, bool);
|
|
|
|
};
|
|
void SetCallbacks(Callbacks callbacks);
|
|
virtual std::string GetDebugString();
|
|
virtual void SetNetworkType(int type);
|
|
protected:
|
|
virtual void ProcessIncomingPacket(NetworkPacket& packet, Endpoint& srcEndpoint);
|
|
virtual void SendInit();
|
|
virtual void SendUdpPing(Endpoint& endpoint);
|
|
virtual void SendRelayPings();
|
|
virtual void SendPacket(unsigned char* data, size_t len, Endpoint& ep, PendingOutgoingPacket& srcPacket);
|
|
virtual void WritePacketHeader(uint32_t seq, BufferOutputStream* s, unsigned char type, uint32_t length);
|
|
virtual void OnAudioOutputReady();
|
|
private:
|
|
int32_t GetCurrentUnixtime();
|
|
std::vector<std::shared_ptr<Stream>> DeserializeStreams(BufferInputStream& in);
|
|
void SendRecentPacketsRequest();
|
|
void SendSpecialReflectorRequest(unsigned char* data, size_t len);
|
|
void SerializeAndUpdateOutgoingStreams();
|
|
struct GroupCallParticipant{
|
|
int32_t userID;
|
|
unsigned char memberTagHash[32];
|
|
std::vector<std::shared_ptr<Stream>> streams;
|
|
AudioLevelMeter* levelMeter;
|
|
};
|
|
std::vector<GroupCallParticipant> participants;
|
|
unsigned char reflectorSelfTag[16];
|
|
unsigned char reflectorSelfSecret[16];
|
|
unsigned char reflectorSelfTagHash[32];
|
|
int32_t userSelfID;
|
|
Endpoint groupReflector;
|
|
AudioMixer* audioMixer;
|
|
AudioLevelMeter selfLevelMeter;
|
|
Callbacks groupCallbacks;
|
|
struct PacketIdMapping{
|
|
uint32_t seq;
|
|
uint16_t id;
|
|
double ackTime;
|
|
};
|
|
std::vector<PacketIdMapping> recentSentPackets;
|
|
Mutex sentPacketsMutex;
|
|
Mutex participantsMutex;
|
|
int32_t timeDifference;
|
|
};
|
|
|
|
};
|
|
|
|
#endif
|