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tg2sip/libtgvoip/VoIPController.h

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//
// 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.
//
#ifndef __VOIPCONTROLLER_H
#define __VOIPCONTROLLER_H
#ifndef _WIN32
#include <arpa/inet.h>
#include <netinet/in.h>
#endif
#ifdef __APPLE__
#include <TargetConditionals.h>
#include "os/darwin/AudioUnitIO.h"
#endif
#include <stdint.h>
#include <vector>
#include <string>
#include <unordered_map>
#include <map>
#include <memory>
#include "video/VideoSource.h"
#include "video/VideoRenderer.h"
#include <atomic>
#if defined(TGVOIP_USE_SOFTWARE_AUDIO)
#include <pjsua2.hpp>
#include "audio/SoftwareAudioInput.h"
#include "audio/SoftwareAudioOutput.h"
#endif
#include "audio/AudioInput.h"
#include "BlockingQueue.h"
#include "audio/AudioOutput.h"
#include "audio/AudioIO.h"
#include "JitterBuffer.h"
#include "OpusDecoder.h"
#include "OpusEncoder.h"
#include "EchoCanceller.h"
#include "CongestionControl.h"
#include "NetworkSocket.h"
#include "Buffers.h"
#include "PacketReassembler.h"
#include "MessageThread.h"
#include "utils.h"
#define LIBTGVOIP_VERSION "2.4.3"
#ifdef _WIN32
#undef GetCurrentTime
#undef ERROR_TIMEOUT
#endif
#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 Endpoint{
friend class VoIPController;
friend class VoIPGroupController;
public:
enum Type{
UDP_P2P_INET=1,
UDP_P2P_LAN,
UDP_RELAY,
TCP_RELAY
};
Endpoint(int64_t id, uint16_t port, IPv4Address& address, IPv6Address& v6address, Type type, unsigned char* peerTag);
Endpoint();
~Endpoint();
const NetworkAddress& GetAddress() const;
NetworkAddress& GetAddress();
bool IsIPv6Only() const;
int64_t id;
uint16_t port;
IPv4Address address;
IPv6Address v6address;
Type type;
unsigned char peerTag[16];
private:
double lastPingTime;
uint32_t lastPingSeq;
HistoricBuffer<double, 6> rtts;
double averageRTT;
NetworkSocket* socket;
int udpPongCount;
};
class AudioDevice{
public:
std::string id;
std::string displayName;
};
class AudioOutputDevice : public AudioDevice{
};
class AudioInputDevice : public AudioDevice{
};
class AudioInputTester{
public:
AudioInputTester(const std::string deviceID);
~AudioInputTester();
TGVOIP_DISALLOW_COPY_AND_ASSIGN(AudioInputTester);
float GetAndResetLevel();
bool Failed(){
return io && io->Failed();
}
private:
void Update(int16_t* samples, size_t count);
audio::AudioIO* io=NULL;
audio::AudioInput* input=NULL;
int16_t maxSample=0;
std::string deviceID;
};
class VoIPController{
friend class VoIPGroupController;
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;
uint64_t bytesRecvdWifi;
uint64_t bytesSentMobile;
uint64_t bytesRecvdMobile;
};
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(char* key, bool isOutgoing);
/**
*
* @param cfg
*/
void SetConfig(const Config& cfg);
void DebugCtl(int request, int param);
/**
*
* @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(unsigned char* 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 int32_t GetConnectionMaxLayer(){
return 92;
};
/**
* 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);
#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 unsigned char*);
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
#if defined(TGVOIP_USE_SOFTWARE_AUDIO)
pj::AudioMedia * AudioMediaInput() {return softwareMediaInput;};
pj::AudioMedia * AudioMediaOutput() {return softwareMediaOutput;};
#endif
private:
struct Stream;
struct UnacknowledgedExtraData;
protected:
struct RecentOutgoingPacket{
uint32_t seq;
uint16_t id; // for group calls only
double sendTime;
double ackTime;
};
struct PendingOutgoingPacket{
PendingOutgoingPacket(uint32_t seq, unsigned char type, size_t len, Buffer&& data, int64_t endpoint){
this->seq=seq;
this->type=type;
this->len=len;
this->data=std::move(data);
this->endpoint=endpoint;
}
PendingOutgoingPacket(PendingOutgoingPacket&& other){
seq=other.seq;
type=other.type;
len=other.len;
data=std::move(other.data);
endpoint=other.endpoint;
}
PendingOutgoingPacket& operator=(PendingOutgoingPacket&& other){
if(this!=&other){
seq=other.seq;
type=other.type;
len=other.len;
data=std::move(other.data);
endpoint=other.endpoint;
}
return *this;
}
TGVOIP_DISALLOW_COPY_AND_ASSIGN(PendingOutgoingPacket);
uint32_t seq;
unsigned char type;
size_t len;
Buffer data;
int64_t endpoint;
};
struct QueuedPacket{
#if defined(_MSC_VER) && _MSC_VER <= 1800 // VS2013 doesn't support auto-generating move constructors
//TGVOIP_DISALLOW_COPY_AND_ASSIGN(QueuedPacket);
QueuedPacket(QueuedPacket&& other){
data=std::move(other.data);
type=other.type;
seqs=other.seqs;
firstSentTime=other.firstSentTime;
lastSentTime=other.lastSentTime;
retryInterval=other.retryInterval;
timeout=other.timeout;
}
QueuedPacket(){
}
#endif
Buffer data;
unsigned char type;
HistoricBuffer<uint32_t, 16> seqs;
double firstSentTime;
double lastSentTime;
double retryInterval;
double timeout;
};
virtual void ProcessIncomingPacket(NetworkPacket& packet, Endpoint& srcEndpoint);
virtual void ProcessExtraData(Buffer& data);
virtual void WritePacketHeader(uint32_t seq, BufferOutputStream* s, unsigned char type, uint32_t length);
virtual void SendPacket(unsigned char* data, size_t len, Endpoint& ep, PendingOutgoingPacket& srcPacket);
virtual void SendInit();
virtual void SendUdpPing(Endpoint& endpoint);
virtual void SendRelayPings();
virtual void OnAudioOutputReady();
virtual void SendExtra(Buffer& data, unsigned char type);
void SendStreamFlags(Stream& stream);
void SendStreamCSD(Stream& stream);
void InitializeTimers();
void ResetEndpointPingStats();
void SendVideoFrame(const Buffer& frame, uint32_t flags);
void ProcessIncomingVideoFrame(Buffer frame, uint32_t pts, bool keyframe);
std::shared_ptr<Stream> GetStreamByType(int type, bool outgoing);
Endpoint* GetEndpointForPacket(const PendingOutgoingPacket& pkt);
bool SendOrEnqueuePacket(PendingOutgoingPacket pkt, bool enqueue=true);
static std::string NetworkTypeToString(int type);
CellularCarrierInfo GetCarrierInfo();
private:
struct Stream{
int32_t userID;
unsigned char id;
unsigned char type;
uint32_t codec;
bool enabled;
bool extraECEnabled;
uint16_t frameDuration;
std::shared_ptr<JitterBuffer> jitterBuffer;
std::shared_ptr<OpusDecoder> decoder;
std::shared_ptr<PacketReassembler> packetReassembler;
std::shared_ptr<CallbackWrapper> callbackWrapper;
std::vector<Buffer> codecSpecificData;
bool csdIsValid=false;
unsigned int width=0;
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;
};
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);
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;
#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;
/*** 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;
#if defined(TGVOIP_USE_SOFTWARE_AUDIO)
tgvoip::audio::SoftwareAudioInput* softwareMediaInput;
tgvoip::audio::SoftwareAudioOutput* softwareMediaOutput;
#endif
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
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