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libtgvoip/VoIPController.h

847 lines
25 KiB
C++
Executable File

//
// libtgvoip is free and unencumbered public domain software.
// For more information, see http://unlicense.org or the UNLICENSE file
// you should have received with this source code distribution.
//
#pragma once
//#define LOG_PACKETS 1
#include <array>
#include "controller/Constants.h"
#include "controller/protocol/VersionInfo.h"
#include "controller/protocol/protocol/Index.h"
#if defined HAVE_CONFIG_H || defined TGVOIP_USE_INSTALLED_OPUS
#include <opus/opus.h>
#else
#include <opus/opus.h>
#endif
#ifdef __ANDROID__
#include "controller/net/NetworkSocket.h"
#include "os/android/AudioInputAndroid.h"
#include "os/android/JNIUtilities.h"
extern jclass jniUtilitiesClass;
#endif
#if defined(TGVOIP_USE_CALLBACK_AUDIO_IO)
#include "audio/AudioIOCallback.h"
#endif
#ifndef _WIN32
#include <arpa/inet.h>
#include <netinet/in.h>
#endif
#ifdef __APPLE__
#include "os/darwin/AudioUnitIO.h"
#include <TargetConditionals.h>
#endif
#include "audio/AudioIO.h"
#include "audio/AudioInput.h"
#include "audio/AudioOutput.h"
#include "audio/Device.h"
#include "controller/audio/EchoCanceller.h"
#include "controller/audio/OpusDecoder.h"
#include "controller/audio/OpusEncoder.h"
#include "controller/net/CongestionControl.h"
#include "controller/net/Endpoint.h"
#include "controller/net/JitterBuffer.h"
#include "controller/net/PacketReassembler.h"
#include "controller/protocol/Stream.h"
#include "controller/protocol/packets/PacketManager.h"
#include "controller/protocol/packets/PacketStructs.h"
#include "controller/protocol/protocol/Extra.h"
#include "tools/BlockingQueue.h"
#include "tools/Buffers.h"
#include "tools/MessageThread.h"
#include "tools/utils.h"
#include "video/ScreamCongestionController.h"
#include "video/VideoRenderer.h"
#include "video/VideoSource.h"
#include <atomic>
#include <deque>
#include <fstream>
#include <iomanip>
#include <map>
#include <memory>
#include <stdint.h>
#include <string>
#include <unordered_map>
#include <vector>
#define LIBTGVOIP_VERSION "2.5"
#ifdef _WIN32
#undef GetCurrentTime
#undef ERROR_TIMEOUT
#endif
#define ENFORCE_MSG_THREAD assert(messageThread.IsCurrent())
#define TGVOIP_PEER_CAP_GROUP_CALLS 1
#define TGVOIP_PEER_CAP_VIDEO_CAPTURE 2
#define TGVOIP_PEER_CAP_VIDEO_DISPLAY 4
namespace tgvoip
{
enum
{
PROXY_NONE = 0,
PROXY_SOCKS5,
//PROXY_HTTP
};
enum
{
STATE_WAIT_INIT = 1,
STATE_WAIT_INIT_ACK,
STATE_ESTABLISHED,
STATE_FAILED,
STATE_RECONNECTING
};
enum
{
ERROR_UNKNOWN = 0,
ERROR_INCOMPATIBLE,
ERROR_TIMEOUT,
ERROR_AUDIO_IO,
ERROR_PROXY
};
enum
{
NET_TYPE_UNKNOWN = 0,
NET_TYPE_GPRS,
NET_TYPE_EDGE,
NET_TYPE_3G,
NET_TYPE_HSPA,
NET_TYPE_LTE,
NET_TYPE_WIFI,
NET_TYPE_ETHERNET,
NET_TYPE_OTHER_HIGH_SPEED,
NET_TYPE_OTHER_LOW_SPEED,
NET_TYPE_DIALUP,
NET_TYPE_OTHER_MOBILE
};
enum
{
DATA_SAVING_NEVER = 0,
DATA_SAVING_MOBILE,
DATA_SAVING_ALWAYS
};
struct CryptoFunctions
{
void (*rand_bytes)(uint8_t *buffer, size_t length);
void (*sha1)(uint8_t *msg, size_t length, uint8_t *output);
void (*sha256)(uint8_t *msg, size_t length, uint8_t *output);
void (*aes_ige_encrypt)(uint8_t *in, uint8_t *out, size_t length, uint8_t *key, uint8_t *iv);
void (*aes_ige_decrypt)(uint8_t *in, uint8_t *out, size_t length, uint8_t *key, uint8_t *iv);
void (*aes_ctr_encrypt)(uint8_t *inout, size_t length, uint8_t *key, uint8_t *iv, uint8_t *ecount, uint32_t *num);
void (*aes_cbc_encrypt)(uint8_t *in, uint8_t *out, size_t length, uint8_t *key, uint8_t *iv);
void (*aes_cbc_decrypt)(uint8_t *in, uint8_t *out, size_t length, uint8_t *key, uint8_t *iv);
};
struct CellularCarrierInfo
{
std::string name;
std::string mcc;
std::string mnc;
std::string countryCode;
};
class PacketSender;
class VoIPController
{
friend class VoIPGroupController;
friend class PacketSender;
friend class AudioPacketSender;
friend class video::VideoPacketSender;
public:
TGVOIP_DISALLOW_COPY_AND_ASSIGN(VoIPController);
struct Config
{
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)
{
this->initTimeout = initTimeout;
this->recvTimeout = recvTimeout;
this->dataSaving = dataSaving;
this->enableAEC = enableAEC;
this->enableNS = enableNS;
this->enableAGC = enableAGC;
this->enableCallUpgrade = enableCallUpgrade;
}
double initTimeout;
double recvTimeout;
int dataSaving;
#ifndef _WIN32
std::string logFilePath = "";
std::string statsDumpFilePath = "";
#else
std::wstring logFilePath = L"";
std::wstring statsDumpFilePath = L"";
#endif
bool enableAEC;
bool enableNS;
bool enableAGC;
bool enableCallUpgrade;
bool logPacketStats = false;
bool enableVolumeControl = false;
bool enableVideoSend = false;
bool enableVideoReceive = false;
};
struct TrafficStats
{
uint64_t bytesSentWifi = 0;
uint64_t bytesRecvdWifi = 0;
uint64_t bytesSentMobile = 0;
uint64_t bytesRecvdMobile = 0;
};
VoIPController();
virtual ~VoIPController();
/**
* Set the initial endpoints (relays)
* @param endpoints Endpoints converted from phone.PhoneConnection TL objects
* @param allowP2p Whether p2p connectivity is allowed
* @param connectionMaxLayer The max_layer field from the phoneCallProtocol object returned by Telegram server.
* DO NOT HARDCODE THIS VALUE, it's extremely important for backwards compatibility.
*/
void SetRemoteEndpoints(std::vector<Endpoint> endpoints, bool allowP2p, int32_t connectionMaxLayer);
/**
* Initialize and start all the internal threads
*/
void Start();
/**
* Stop any internal threads. Don't call any other methods after this.
*/
void Stop();
/**
* Initiate connection
*/
void Connect();
Endpoint &GetRemoteEndpoint();
/**
* Get the debug info string to be displayed in client UI
*/
virtual std::string GetDebugString();
/**
* Notify the library of network type change
* @param type The new network type
*/
virtual void SetNetworkType(int type);
/**
* Get the average round-trip time for network packets
* @return
*/
double GetAverageRTT();
static double GetCurrentTime();
/**
* Use this field to store any of your context data associated with this call
*/
void *implData;
/**
*
* @param mute
*/
virtual void SetMicMute(bool mute);
/**
*
* @param key
* @param isOutgoing
*/
void SetEncryptionKey(std::vector<uint8_t>, bool isOutgoing);
/**
*
* @param cfg
*/
void SetConfig(const Config &cfg);
/**
*
* @param stats
*/
void GetStats(TrafficStats *stats);
/**
*
* @return
*/
int64_t GetPreferredRelayID();
/**
*
* @return
*/
int GetLastError();
/**
*
*/
static CryptoFunctions crypto;
/**
*
* @return
*/
static const char *GetVersion();
/**
*
* @return
*/
std::string GetDebugLog();
/**
*
* @return
*/
static std::vector<AudioInputDevice> EnumerateAudioInputs();
/**
*
* @return
*/
static std::vector<AudioOutputDevice> EnumerateAudioOutputs();
/**
*
* @param id
*/
void SetCurrentAudioInput(std::string id);
/**
*
* @param id
*/
void SetCurrentAudioOutput(std::string id);
/**
*
* @return
*/
std::string GetCurrentAudioInputID();
/**
*
* @return
*/
std::string GetCurrentAudioOutputID();
/**
* Set the proxy server to route the data through. Call this before connecting.
* @param protocol PROXY_NONE or PROXY_SOCKS5
* @param address IP address or domain name of the server
* @param port Port of the server
* @param username Username; empty string for anonymous
* @param password Password; empty string if none
*/
void SetProxy(int protocol, std::string address, uint16_t port, std::string username, std::string password);
/**
* Get the number of signal bars to display in the client UI.
* @return the number of signal bars, from 1 to 4
*/
int GetSignalBarsCount();
/**
* Enable or disable AGC (automatic gain control) on audio output. Should only be enabled on phones when the earpiece speaker is being used.
* The audio output will be louder with this on.
* AGC with speakerphone or other kinds of loud speakers has detrimental effects on some echo cancellation implementations.
* @param enabled I usually pick argument names to be self-explanatory
*/
void SetAudioOutputGainControlEnabled(bool enabled);
/**
* Get the additional capabilities of the peer client app
* @return corresponding TGVOIP_PEER_CAP_* flags OR'ed together
*/
uint32_t GetPeerCapabilities();
/**
* Send the peer the key for the group call to prepare this private call to an upgrade to a E2E group call.
* The peer must have the TGVOIP_PEER_CAP_GROUP_CALLS capability. After the peer acknowledges the key, Callbacks::groupCallKeySent will be called.
* @param key newly-generated group call key, must be exactly 265 bytes long
*/
void SendGroupCallKey(uint8_t *key);
/**
* 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.
*/
void RequestCallUpgrade();
void SetEchoCancellationStrength(int strength);
int GetConnectionState();
bool NeedRate();
/**
* Get the maximum connection layer supported by this libtgvoip version.
* Pass this as <code>max_layer</code> in the phone.phoneConnection TL object when requesting and accepting calls.
*/
static const int32_t GetConnectionMaxLayer()
{
return 110;
};
/**
* Get the persistable state of the library, like proxy capabilities, to save somewhere on the disk. Call this at the end of the call.
* Using this will speed up the connection establishment in some cases.
*/
std::vector<uint8_t> GetPersistentState();
/**
* Load the persistable state. Call this before starting the call.
*/
void SetPersistentState(std::vector<uint8_t> state);
#if defined(TGVOIP_USE_CALLBACK_AUDIO_IO)
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);
#endif
void SetVideoCodecSpecificData(const std::vector<Buffer> &data);
struct Callbacks
{
void (*connectionStateChanged)(VoIPController *, int);
void (*signalBarCountChanged)(VoIPController *, int);
void (*groupCallKeySent)(VoIPController *);
void (*groupCallKeyReceived)(VoIPController *, const uint8_t *);
void (*upgradeToGroupCallRequested)(VoIPController *);
};
void SetCallbacks(Callbacks callbacks);
float GetOutputLevel()
{
return 0.0f;
};
void SetVideoSource(video::VideoSource *source);
void SetVideoRenderer(video::VideoRenderer *renderer);
void SetInputVolume(float level);
void SetOutputVolume(float level);
#if defined(__APPLE__) && defined(TARGET_OS_OSX)
void SetAudioOutputDuckingEnabled(bool enabled);
#endif
protected:
virtual void ProcessIncomingPacket(NetworkPacket &packet, Endpoint &srcEndpoint);
virtual void ProcessIncomingPacket(Packet &packet, Endpoint &srcEndpoint);
virtual void ProcessExtraData(const Wrapped<Extra> &data, Endpoint &srcEndpoint);
//virtual uint8_t WritePacketHeader(PendingOutgoingPacket &pkt, BufferOutputStream &s, PacketSender *source);
virtual void SendUdpPing(Endpoint &endpoint);
virtual void SendRelayPings();
PendingOutgoingPacket PreparePacket(unsigned char *data, size_t len, Endpoint &ep, CongestionControlPacket &&pkt);
void SendPacket(OutgoingPacket &&pkt, double retryInterval = 0.5, double timeout = 5.0, uint8_t tries = 0);
void SendOrEnqueuePacket(PendingOutgoingPacket &pkt, bool enqueue = true);
void SendPacketReliably(PendingOutgoingPacket &pkt, double retryInterval, double timeout, uint8_t tries = 0xFF);
void SendNopPacket(int64_t endpointId = 0, double retryInterval = 0.5, double timeout = 5.0, uint8_t tries = 0);
virtual void SendInit();
virtual void SendDataSavingMode();
virtual void SendExtra(Wrapped<Extra> &&extra, int64_t endpointId = 0);
virtual void SendExtra(std::shared_ptr<Extra> &&_d, int64_t endpointId = 0);
virtual void SendExtra(std::shared_ptr<Extra> &_d, int64_t endpointId = 0);
template <class T>
void SendStreamFlags(const T &stream)
{
ENFORCE_MSG_THREAD;
auto flags = std::make_shared<ExtraStreamFlags>();
flags->streamId = stream.id;
if (stream.enabled)
flags->flags |= ExtraStreamFlags::Flags::Enabled;
if constexpr (std::is_same_v<OutgoingAudioStream, T>)
if (stream.extraECEnabled)
flags->flags |= ExtraStreamFlags::Flags::ExtraEC;
if (stream.paused)
flags->flags |= ExtraStreamFlags::Flags::Paused;
LOGV("My stream state: id %u flags %u", (unsigned int)stream.id, (unsigned int)flags->flags);
SendExtra(Wrapped<Extra>(std::move(flags)));
};
virtual void OnAudioOutputReady();
void InitializeTimers();
void ResetEndpointPingStats();
void ProcessIncomingVideoFrame(Buffer frame, uint32_t pts, bool keyframe, uint16_t rotation);
Endpoint *GetEndpointForPacket(const PendingOutgoingPacket &pkt);
Endpoint *GetEndpointForPacket(const OutgoingPacket &pkt);
Endpoint *GetEndpointById(const int64_t id);
CellularCarrierInfo GetCarrierInfo();
template <class T>
std::shared_ptr<T> GetStreamByTypeShared()
{
if constexpr (T::OUTGOING)
{
for (auto &ss : outgoingStreams)
{
if (ss->type == T::TYPE)
return dynamic_pointer_cast<T>(ss);
}
}
else
{
for (auto &ss : incomingStreams)
{
if (ss->type == T::TYPE)
return dynamic_pointer_cast<T>(ss);
}
}
return nullptr;
}
template <class T>
T *GetStreamByType()
{
if constexpr (T::OUTGOING)
{
for (auto &ss : outgoingStreams)
{
if (ss->type == T::TYPE)
return dynamic_cast<T *>(ss.get());
}
}
else
{
for (auto &ss : incomingStreams)
{
if (ss->type == T::TYPE)
return dynamic_cast<T *>(ss.get());
}
}
return nullptr;
}
template <class T>
T *GetStreamByID(uint8_t id)
{
if constexpr (T::OUTGOING)
{
if (id < outgoingStreams.size())
{
return dynamic_cast<T *>(outgoingStreams[id].get());
}
}
else
{
if (id < incomingStreams.size())
{
return dynamic_cast<T *>(incomingStreams[id].get());
}
}
return nullptr;
}
private:
struct RawPendingOutgoingPacket
{
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