mirror of
https://github.com/danog/libtgvoip.git
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847 lines
25 KiB
C++
Executable File
847 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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#pragma once
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//#define LOG_PACKETS 1
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#include <array>
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#include "controller/Constants.h"
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#include "controller/protocol/VersionInfo.h"
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#include "controller/protocol/protocol/Index.h"
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#if defined HAVE_CONFIG_H || defined TGVOIP_USE_INSTALLED_OPUS
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#include <opus/opus.h>
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#else
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#include <opus/opus.h>
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#endif
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#ifdef __ANDROID__
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#include "controller/net/NetworkSocket.h"
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#include "os/android/AudioInputAndroid.h"
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#include "os/android/JNIUtilities.h"
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extern jclass jniUtilitiesClass;
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#endif
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#if defined(TGVOIP_USE_CALLBACK_AUDIO_IO)
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#include "audio/AudioIOCallback.h"
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#endif
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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 "os/darwin/AudioUnitIO.h"
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#include <TargetConditionals.h>
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#endif
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#include "audio/AudioIO.h"
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#include "audio/AudioInput.h"
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#include "audio/AudioOutput.h"
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#include "audio/Device.h"
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#include "controller/audio/EchoCanceller.h"
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#include "controller/audio/OpusDecoder.h"
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#include "controller/audio/OpusEncoder.h"
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#include "controller/net/CongestionControl.h"
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#include "controller/net/Endpoint.h"
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#include "controller/net/JitterBuffer.h"
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#include "controller/net/PacketReassembler.h"
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#include "controller/protocol/Stream.h"
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#include "controller/protocol/packets/PacketManager.h"
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#include "controller/protocol/packets/PacketStructs.h"
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#include "controller/protocol/protocol/Extra.h"
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#include "tools/BlockingQueue.h"
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#include "tools/Buffers.h"
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#include "tools/MessageThread.h"
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#include "tools/utils.h"
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#include "video/ScreamCongestionController.h"
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#include "video/VideoRenderer.h"
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#include "video/VideoSource.h"
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#include <atomic>
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#include <deque>
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#include <fstream>
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#include <iomanip>
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#include <map>
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#include <memory>
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#include <stdint.h>
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#include <string>
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#include <unordered_map>
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#include <vector>
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#define LIBTGVOIP_VERSION "2.5"
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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 ENFORCE_MSG_THREAD assert(messageThread.IsCurrent())
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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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{
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enum
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{
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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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{
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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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{
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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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{
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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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{
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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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{
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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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{
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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 PacketSender;
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class VoIPController
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{
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friend class VoIPGroupController;
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friend class PacketSender;
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friend class AudioPacketSender;
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friend class video::VideoPacketSender;
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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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{
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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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{
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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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{
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uint64_t bytesSentWifi = 0;
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uint64_t bytesRecvdWifi = 0;
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uint64_t bytesSentMobile = 0;
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uint64_t bytesRecvdMobile = 0;
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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(std::vector<uint8_t>, 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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/**
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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(uint8_t *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 const int32_t GetConnectionMaxLayer()
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{
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return 110;
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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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{
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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 uint8_t *);
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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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{
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return 0.0f;
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};
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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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protected:
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virtual void ProcessIncomingPacket(NetworkPacket &packet, Endpoint &srcEndpoint);
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virtual void ProcessIncomingPacket(Packet &packet, Endpoint &srcEndpoint);
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virtual void ProcessExtraData(const Wrapped<Extra> &data, Endpoint &srcEndpoint);
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//virtual uint8_t WritePacketHeader(PendingOutgoingPacket &pkt, BufferOutputStream &s, PacketSender *source);
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virtual void SendUdpPing(Endpoint &endpoint);
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virtual void SendRelayPings();
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PendingOutgoingPacket PreparePacket(unsigned char *data, size_t len, Endpoint &ep, CongestionControlPacket &&pkt);
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void SendPacket(OutgoingPacket &&pkt, double retryInterval = 0.5, double timeout = 5.0, uint8_t tries = 0);
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void SendOrEnqueuePacket(PendingOutgoingPacket &pkt, bool enqueue = true);
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void SendPacketReliably(PendingOutgoingPacket &pkt, double retryInterval, double timeout, uint8_t tries = 0xFF);
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void SendNopPacket(int64_t endpointId = 0, double retryInterval = 0.5, double timeout = 5.0, uint8_t tries = 0);
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virtual void SendInit();
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virtual void SendDataSavingMode();
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virtual void SendExtra(Wrapped<Extra> &&extra, int64_t endpointId = 0);
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virtual void SendExtra(std::shared_ptr<Extra> &&_d, int64_t endpointId = 0);
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virtual void SendExtra(std::shared_ptr<Extra> &_d, int64_t endpointId = 0);
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template <class T>
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void SendStreamFlags(const T &stream)
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{
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ENFORCE_MSG_THREAD;
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auto flags = std::make_shared<ExtraStreamFlags>();
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flags->streamId = stream.id;
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if (stream.enabled)
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flags->flags |= ExtraStreamFlags::Flags::Enabled;
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if constexpr (std::is_same_v<OutgoingAudioStream, T>)
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if (stream.extraECEnabled)
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flags->flags |= ExtraStreamFlags::Flags::ExtraEC;
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if (stream.paused)
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flags->flags |= ExtraStreamFlags::Flags::Paused;
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LOGV("My stream state: id %u flags %u", (unsigned int)stream.id, (unsigned int)flags->flags);
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SendExtra(Wrapped<Extra>(std::move(flags)));
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};
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virtual void OnAudioOutputReady();
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void InitializeTimers();
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void ResetEndpointPingStats();
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void ProcessIncomingVideoFrame(Buffer frame, uint32_t pts, bool keyframe, uint16_t rotation);
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Endpoint *GetEndpointForPacket(const PendingOutgoingPacket &pkt);
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Endpoint *GetEndpointForPacket(const OutgoingPacket &pkt);
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Endpoint *GetEndpointById(const int64_t id);
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CellularCarrierInfo GetCarrierInfo();
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template <class T>
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std::shared_ptr<T> GetStreamByTypeShared()
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{
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if constexpr (T::OUTGOING)
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{
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for (auto &ss : outgoingStreams)
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{
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if (ss->type == T::TYPE)
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return dynamic_pointer_cast<T>(ss);
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}
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}
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else
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{
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for (auto &ss : incomingStreams)
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{
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if (ss->type == T::TYPE)
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return dynamic_pointer_cast<T>(ss);
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}
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}
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return nullptr;
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}
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template <class T>
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T *GetStreamByType()
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{
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if constexpr (T::OUTGOING)
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{
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for (auto &ss : outgoingStreams)
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{
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if (ss->type == T::TYPE)
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return dynamic_cast<T *>(ss.get());
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}
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}
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else
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{
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for (auto &ss : incomingStreams)
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{
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if (ss->type == T::TYPE)
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return dynamic_cast<T *>(ss.get());
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}
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}
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return nullptr;
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}
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template <class T>
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T *GetStreamByID(uint8_t id)
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{
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if constexpr (T::OUTGOING)
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{
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if (id < outgoingStreams.size())
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{
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return dynamic_cast<T *>(outgoingStreams[id].get());
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}
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}
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else
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{
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if (id < incomingStreams.size())
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{
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return dynamic_cast<T *>(incomingStreams[id].get());
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}
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}
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return nullptr;
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}
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private:
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struct RawPendingOutgoingPacket
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{
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TGVOIP_MOVE_ONLY(RawPendingOutgoingPacket);
|
|
NetworkPacket packet;
|
|
std::shared_ptr<NetworkSocket> socket;
|
|
};
|
|
enum
|
|
{
|
|
UDP_UNKNOWN = 0,
|
|
UDP_PING_PENDING,
|
|
UDP_PING_SENT,
|
|
UDP_AVAILABLE,
|
|
UDP_NOT_AVAILABLE,
|
|
UDP_BAD
|
|
};
|
|
|
|
void RunRecvThread();
|
|
void RunSendThread();
|
|
void UpdateAudioBitrateLimit();
|
|
void SetState(int state);
|
|
void UpdateAudioOutputState();
|
|
void InitUDPProxy();
|
|
void UpdateDataSavingState();
|
|
|
|
size_t decryptPacket(unsigned char *buffer, BufferInputStream &in);
|
|
void encryptPacket(unsigned char *data, size_t len, BufferOutputStream &out);
|
|
|
|
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);
|
|
|
|
void SendPublicEndpointsRequest();
|
|
void SendPublicEndpointsRequest(const Endpoint &relay);
|
|
Endpoint &GetEndpointByType(const Endpoint::Type type);
|
|
|
|
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 UpdateReliablePackets();
|
|
void TickJitterBufferAndCongestionControl();
|
|
void ResetUdpAvailability();
|
|
inline static std::string NetworkTypeToString(int type)
|
|
{
|
|
switch (type)
|
|
{
|
|
case NET_TYPE_WIFI:
|
|
return "wifi";
|
|
case NET_TYPE_GPRS:
|
|
return "gprs";
|
|
case NET_TYPE_EDGE:
|
|
return "edge";
|
|
case NET_TYPE_3G:
|
|
return "3g";
|
|
case NET_TYPE_HSPA:
|
|
return "hspa";
|
|
case NET_TYPE_LTE:
|
|
return "lte";
|
|
case NET_TYPE_ETHERNET:
|
|
return "ethernet";
|
|
case NET_TYPE_OTHER_HIGH_SPEED:
|
|
return "other_high_speed";
|
|
case NET_TYPE_OTHER_LOW_SPEED:
|
|
return "other_low_speed";
|
|
case NET_TYPE_DIALUP:
|
|
return "dialup";
|
|
case NET_TYPE_OTHER_MOBILE:
|
|
return "other_mobile";
|
|
default:
|
|
return "unknown";
|
|
}
|
|
}
|
|
public:
|
|
inline static std::string GetPacketTypeString(unsigned char type)
|
|
{
|
|
switch (type)
|
|
{
|
|
case PKT_INIT:
|
|
return "init";
|
|
case PKT_INIT_ACK:
|
|
return "init_ack";
|
|
case PKT_STREAM_STATE:
|
|
return "stream_state";
|
|
case PKT_STREAM_DATA:
|
|
return "stream_data";
|
|
case PKT_PING:
|
|
return "ping";
|
|
case PKT_PONG:
|
|
return "pong";
|
|
case PKT_LAN_ENDPOINT:
|
|
return "lan_endpoint";
|
|
case PKT_NETWORK_CHANGED:
|
|
return "network_changed";
|
|
case PKT_NOP:
|
|
return "nop";
|
|
case PKT_STREAM_EC:
|
|
return "stream_ec";
|
|
}
|
|
return string("unknown(") + std::to_string(type) + ')';
|
|
}
|
|
private:
|
|
|
|
bool parseRelayPacket(const BufferInputStream &in, Endpoint &srcEndpoint);
|
|
|
|
void HandleReliablePackets(const PacketManager &pm);
|
|
|
|
void SetupOutgoingVideoStream();
|
|
void NetworkPacketReceived(std::shared_ptr<NetworkPacket> packet);
|
|
void TrySendOutgoingPackets();
|
|
|
|
int state = STATE_WAIT_INIT;
|
|
std::map<int64_t, Endpoint> endpoints;
|
|
int64_t currentEndpoint = 0;
|
|
int64_t preferredRelay = 0;
|
|
int64_t peerPreferredRelay = 0;
|
|
std::atomic<bool> runReceiver = ATOMIC_VAR_INIT(false);
|
|
|
|
HistoricBuffer<uint32_t, 10, double> sendLossCountHistory;
|
|
HistoricBuffer<uint32_t, 10, double> packetCountHistory;
|
|
|
|
std::shared_ptr<OpusEncoder> encoder;
|
|
std::unique_ptr<tgvoip::audio::AudioIO> audioIO;
|
|
|
|
// Obtained from audioIO
|
|
std::shared_ptr<tgvoip::audio::AudioInput> audioInput;
|
|
std::shared_ptr<tgvoip::audio::AudioOutput> audioOutput;
|
|
|
|
// Shared between encoder and decoder
|
|
std::shared_ptr<EchoCanceller> echoCanceller;
|
|
|
|
std::unique_ptr<Thread> recvThread;
|
|
std::unique_ptr<Thread> sendThread;
|
|
|
|
std::vector<PendingOutgoingPacket> sendQueue;
|
|
std::atomic<bool> stopping = ATOMIC_VAR_INIT(false);
|
|
bool audioOutStarted = false;
|
|
uint32_t packetsReceived = 0;
|
|
uint32_t recvLossCount = 0;
|
|
uint32_t prevSendLossCount = 0;
|
|
uint32_t prevSeq = 1;
|
|
uint32_t firstSentPing;
|
|
HistoricBuffer<double, 32> rttHistory;
|
|
bool waitingForAcks = false;
|
|
int networkType = NET_TYPE_UNKNOWN;
|
|
int dontSendPackets = 0;
|
|
int lastError;
|
|
bool micMuted = false;
|
|
uint32_t maxBitrate;
|
|
|
|
//
|
|
std::vector<std::shared_ptr<OutgoingStream>> outgoingStreams;
|
|
std::vector<std::shared_ptr<IncomingStream>> incomingStreams;
|
|
|
|
PacketManager &getBestPacketManager();
|
|
|
|
unsigned char encryptionKey[256];
|
|
unsigned char keyFingerprint[8];
|
|
unsigned char callID[16];
|
|
double stateChangeTime;
|
|
bool waitingForRelayPeerInfo = false;
|
|
bool allowP2p = true;
|
|
bool dataSavingMode = false;
|
|
bool dataSavingRequestedByPeer = false;
|
|
std::string activeNetItfName;
|
|
double publicEndpointsReqTime = 0;
|
|
std::vector<ReliableOutgoingPacket> reliablePackets;
|
|
double connectionInitTime = 0;
|
|
double lastRecvPacketTime = 0;
|
|
Config config;
|
|
CongestionControl conctl;
|
|
TrafficStats stats;
|
|
bool receivedInit = false;
|
|
bool receivedInitAck = false;
|
|
bool isOutgoing;
|
|
|
|
// Might point to the same or different objects
|
|
std::shared_ptr<NetworkSocket> udpSocket;
|
|
std::shared_ptr<NetworkSocket> realUdpSocket;
|
|
|
|
std::ofstream statsDump;
|
|
std::string currentAudioInput;
|
|
std::string currentAudioOutput;
|
|
bool useTCP = false;
|
|
bool useUDP = true;
|
|
bool didAddTcpRelays = false;
|
|
std::unique_ptr<SocketSelectCanceller> selectCanceller;
|
|
HistoricBuffer<unsigned char, 4, int> signalBarsHistory;
|
|
bool audioStarted = false;
|
|
|
|
int udpConnectivityState = UDP_UNKNOWN;
|
|
double lastUdpPingTime = 0;
|
|
int udpPingCount = 0;
|
|
int echoCancellationStrength = 1;
|
|
|
|
int proxyProtocol = PROXY_NONE;
|
|
std::string proxyAddress;
|
|
uint16_t proxyPort = 0;
|
|
std::string proxyUsername;
|
|
std::string proxyPassword;
|
|
NetworkAddress resolvedProxyAddress = NetworkAddress::Empty();
|
|
|
|
uint32_t peerCapabilities = 0;
|
|
Callbacks callbacks{0};
|
|
bool didReceiveGroupCallKey = false;
|
|
bool didReceiveGroupCallKeyAck = false;
|
|
bool didSendGroupCallKey = false;
|
|
bool didSendUpgradeRequest = false;
|
|
bool didInvokeUpgradeCallback = false;
|
|
|
|
bool useMTProto2 = false;
|
|
bool setCurrentEndpointToTCP = false;
|
|
|
|
std::vector<UnacknowledgedExtraData> currentExtras;
|
|
std::unordered_map<uint8_t, uint64_t> lastReceivedExtrasByType;
|
|
bool useIPv6 = false;
|
|
bool peerIPv6Available = false;
|
|
NetworkAddress myIPv6{NetworkAddress::Empty()};
|
|
bool didAddIPv6Relays = false;
|
|
bool didSendIPv6Endpoint = false;
|
|
int publicEndpointsReqCount = 0;
|
|
bool wasEstablished = false;
|
|
bool receivedFirstStreamPacket = false;
|
|
std::atomic<unsigned int> unsentStreamPackets = ATOMIC_VAR_INIT(0);
|
|
HistoricBuffer<unsigned int, 5> unsentStreamPacketsHistory;
|
|
bool needReInitUdpProxy = true;
|
|
bool needRate = false;
|
|
BlockingQueue<RawPendingOutgoingPacket> rawSendQueue;
|
|
|
|
uint32_t initTimeoutID = MessageThread::INVALID_ID;
|
|
uint32_t udpPingTimeoutID = MessageThread::INVALID_ID;
|
|
|
|
// Using a shared_ptr is redundant here, but it allows more flexibility in the OpusEncoder API
|
|
std::shared_ptr<effects::Volume> outputVolume = std::make_shared<effects::Volume>();
|
|
std::shared_ptr<effects::Volume> inputVolume = std::make_shared<effects::Volume>();
|
|
|
|
std::vector<uint32_t> peerVideoDecoders;
|
|
|
|
MessageThread messageThread;
|
|
|
|
// Locked whenever the endpoints vector is modified (but not endpoints themselves) and whenever iterated outside of messageThread.
|
|
// After the call is started, only messageThread is allowed to modify the endpoints vector.
|
|
Mutex endpointsMutex;
|
|
// Locked while audio i/o is being initialized and deinitialized so as to allow it to fully initialize before deinitialization begins.
|
|
Mutex audioIOMutex;
|
|
|
|
#if defined(TGVOIP_USE_CALLBACK_AUDIO_IO)
|
|
std::function<void(int16_t *, size_t)> audioInputDataCallback;
|
|
std::function<void(int16_t *, size_t)> audioOutputDataCallback;
|
|
#endif
|
|
#if defined(__APPLE__) && defined(TARGET_OS_OSX)
|
|
bool macAudioDuckingEnabled = true;
|
|
#endif
|
|
|
|
video::VideoRenderer *videoRenderer = nullptr;
|
|
uint32_t lastReceivedVideoFrameNumber = UINT32_MAX;
|
|
|
|
uint32_t sendLosses = 0;
|
|
uint32_t unacknowledgedIncomingPacketCount = 0;
|
|
|
|
VersionInfo ver;
|
|
|
|
/*** 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;
|
|
uint32_t unackNopThreshold;
|
|
|
|
public:
|
|
#ifdef __APPLE__
|
|
static double machTimebase;
|
|
static uint64_t machTimestart = 0;
|
|
#endif
|
|
#ifdef _WIN32
|
|
static int64_t win32TimeScale;
|
|
static bool didInitWin32TimeScale;
|
|
#endif
|
|
};
|
|
|
|
}; // namespace tgvoip
|