/usr/include/libtgvoip/VoIPController.h is in libtgvoip-dev 1.0.3-3.
This file is owned by root:root, with mode 0o644.
The actual contents of the file can be viewed below.
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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>
#endif
#include <stdint.h>
#include <vector>
#include <string>
#include <map>
#include "audio/AudioInput.h"
#include "BlockingQueue.h"
#include "BufferOutputStream.h"
#include "audio/AudioOutput.h"
#include "JitterBuffer.h"
#include "OpusDecoder.h"
#include "OpusEncoder.h"
#include "EchoCanceller.h"
#include "CongestionControl.h"
#include "NetworkSocket.h"
#define LIBTGVOIP_VERSION "1.0.3"
#define STATE_WAIT_INIT 1
#define STATE_WAIT_INIT_ACK 2
#define STATE_ESTABLISHED 3
#define STATE_FAILED 4
#define STATE_RECONNECTING 5
#define TGVOIP_ERROR_UNKNOWN 0
#define TGVOIP_ERROR_INCOMPATIBLE 1
#define TGVOIP_ERROR_TIMEOUT 2
#define TGVOIP_ERROR_AUDIO_IO 3
#define NET_TYPE_UNKNOWN 0
#define NET_TYPE_GPRS 1
#define NET_TYPE_EDGE 2
#define NET_TYPE_3G 3
#define NET_TYPE_HSPA 4
#define NET_TYPE_LTE 5
#define NET_TYPE_WIFI 6
#define NET_TYPE_ETHERNET 7
#define NET_TYPE_OTHER_HIGH_SPEED 8
#define NET_TYPE_OTHER_LOW_SPEED 9
#define NET_TYPE_DIALUP 10
#define NET_TYPE_OTHER_MOBILE 11
#define EP_TYPE_UDP_P2P_INET 1
#define EP_TYPE_UDP_P2P_LAN 2
#define EP_TYPE_UDP_RELAY 3
#define EP_TYPE_TCP_RELAY 4
#define DATA_SAVING_NEVER 0
#define DATA_SAVING_MOBILE 1
#define DATA_SAVING_ALWAYS 2
#ifdef _WIN32
#undef GetCurrentTime
#endif
struct voip_stream_t{
int32_t userID;
unsigned char id;
unsigned char type;
unsigned char codec;
bool enabled;
uint16_t frameDuration;
};
typedef struct voip_stream_t voip_stream_t;
struct voip_queued_packet_t{
unsigned char type;
unsigned char* data;
size_t length;
uint32_t seqs[16];
double firstSentTime;
double lastSentTime;
double retryInterval;
double timeout;
};
typedef struct voip_queued_packet_t voip_queued_packet_t;
struct voip_config_t{
double init_timeout;
double recv_timeout;
int data_saving;
char logFilePath[256];
char statsDumpFilePath[256];
bool enableAEC;
bool enableNS;
bool enableAGC;
};
typedef struct voip_config_t voip_config_t;
#if defined(__APPLE__) && TARGET_OS_IPHONE
// temporary fix for nasty linking errors
struct voip_legacy_endpoint_t{
const char* address;
const char* address6;
uint16_t port;
int64_t id;
unsigned char peerTag[16];
};
typedef struct voip_legacy_endpoint_t voip_legacy_endpoint_t;
#endif
struct voip_stats_t{
uint64_t bytesSentWifi;
uint64_t bytesRecvdWifi;
uint64_t bytesSentMobile;
uint64_t bytesRecvdMobile;
};
typedef struct voip_stats_t voip_stats_t;
struct voip_crypto_functions_t{
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);
};
typedef struct voip_crypto_functions_t voip_crypto_functions_t;
#define SEQ_MAX 0xFFFFFFFF
inline bool seqgt(uint32_t s1, uint32_t s2){
return ((s1>s2) && (s1-s2<=SEQ_MAX/2)) || ((s1<s2) && (s2-s1>SEQ_MAX/2));
}
namespace tgvoip{
enum{
PROXY_NONE=0,
PROXY_SOCKS5,
//PROXY_HTTP
};
class Endpoint{
friend class VoIPController;
public:
Endpoint(int64_t id, uint16_t port, IPv4Address& address, IPv6Address& v6address, char type, unsigned char* peerTag);
Endpoint();
int64_t id;
uint16_t port;
IPv4Address address;
IPv6Address v6address;
char type;
unsigned char peerTag[16];
private:
double lastPingTime;
uint32_t lastPingSeq;
double rtts[6];
double averageRTT;
NetworkSocket* socket;
};
class AudioDevice{
public:
std::string id;
std::string displayName;
};
class AudioOutputDevice : public AudioDevice{
};
class AudioInputDevice : public AudioDevice{
};
class VoIPController
{
public:
VoIPController();
~VoIPController();
/**
* Set the initial endpoints (relays)
* @param endpoints Endpoints converted from phone.PhoneConnection TL objects
* @param allowP2p Whether p2p connectivity is allowed
*/
void SetRemoteEndpoints(std::vector<Endpoint> endpoints, bool allowP2p);
/**
* Initialize and start all the internal threads
*/
void Start();
/**
* Initiate connection
*/
void Connect();
Endpoint& GetRemoteEndpoint();
/**
* Get the debug info string to be displayed in client UI
* @param buffer The buffer to put the string into
* @param len The length of the buffer
*/
void GetDebugString(char* buffer, size_t len);
/**
* Notify the library of network type change
* @param type The new network type
*/
void SetNetworkType(int type);
/**
* Get the average round-trip time for network packets
* @return
*/
double GetAverageRTT();
/**
* Set the function to be called whenever the connection state changes
* @param f
*/
void SetStateCallback(void (*f)(VoIPController*, int));
static double GetCurrentTime();
/**
* Use this field to store any of your context data associated with this call
*/
void* implData;
/**
*
* @param mute
*/
void SetMicMute(bool mute);
/**
*
* @param key
* @param isOutgoing
*/
void SetEncryptionKey(char* key, bool isOutgoing);
/**
*
* @param cfg
*/
void SetConfig(voip_config_t* cfg);
float GetOutputLevel();
void DebugCtl(int request, int param);
/**
*
* @param stats
*/
void GetStats(voip_stats_t* stats);
/**
*
* @return
*/
int64_t GetPreferredRelayID();
/**
*
* @return
*/
int GetLastError();
/**
*
*/
static voip_crypto_functions_t crypto;
/**
*
* @return
*/
static const char* GetVersion();
#ifdef TGVOIP_USE_AUDIO_SESSION
void SetAcquireAudioSession(void (^)(void (^)()));
void ReleaseAudioSession(void (^completion)());
#endif
/**
*
* @return
*/
std::string GetDebugLog();
/**
*
* @param buffer
*/
void GetDebugLog(char* buffer);
size_t GetDebugLogLength();
/**
*
* @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();
/**
* Set the callback to be called when the signal bar count changes.
* @param f
*/
void SetSignalBarsCountCallback(void (*f)(VoIPController*, int));
/**
* Enable or disable AGC (automatic gain control) on audio output. Should only be enabled on phones when the earpiece speaker is being used.
* The audio output will be louder with this on.
* 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);
void SetEchoCancellationStrength(int strength);
private:
struct PendingOutgoingPacket{
uint32_t seq;
unsigned char type;
size_t len;
unsigned char* data;
Endpoint* endpoint;
};
enum{
UDP_UNKNOWN=0,
UDP_PING_SENT,
UDP_AVAILABIE,
UDP_NOT_AVAILABLE
};
static void* StartRecvThread(void* arg);
static void* StartSendThread(void* arg);
static void* StartTickThread(void* arg);
void RunRecvThread();
void RunSendThread();
void RunTickThread();
void SendPacket(unsigned char* data, size_t len, Endpoint* ep);
void HandleAudioInput(unsigned char* data, size_t len);
void UpdateAudioBitrate();
void SetState(int state);
void UpdateAudioOutputState();
void SendInit();
void InitUDPProxy();
void UpdateDataSavingState();
void KDF(unsigned char* msgKey, size_t x, unsigned char* aesKey, unsigned char* aesIv);
void WritePacketHeader(uint32_t seq, BufferOutputStream* s, unsigned char type, uint32_t length);
static size_t AudioInputCallback(unsigned char* data, size_t length, void* param);
void SendPublicEndpointsRequest();
void SendPublicEndpointsRequest(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 LogDebugInfo();
void SendUdpPing(Endpoint* endpoint);
int state;
std::vector<Endpoint*> endpoints;
Endpoint* currentEndpoint;
Endpoint* preferredRelay;
Endpoint* peerPreferredRelay;
bool runReceiver;
uint32_t seq;
uint32_t lastRemoteSeq;
uint32_t lastRemoteAckSeq;
uint32_t lastSentSeq;
double remoteAcks[32];
double sentPacketTimes[32];
double recvPacketTimes[32];
uint32_t sendLossCountHistory[32];
uint32_t audioTimestampIn;
uint32_t audioTimestampOut;
tgvoip::audio::AudioInput* audioInput;
tgvoip::audio::AudioOutput* audioOutput;
JitterBuffer* jitterBuffer;
OpusDecoder* decoder;
OpusEncoder* encoder;
BlockingQueue<PendingOutgoingPacket>* sendQueue;
EchoCanceller* echoCanceller;
tgvoip_mutex_t sendBufferMutex;
tgvoip_mutex_t endpointsMutex;
bool stopping;
bool audioOutStarted;
tgvoip_thread_t recvThread;
tgvoip_thread_t sendThread;
tgvoip_thread_t tickThread;
uint32_t packetsRecieved;
uint32_t recvLossCount;
uint32_t prevSendLossCount;
uint32_t firstSentPing;
double rttHistory[32];
bool waitingForAcks;
int networkType;
int dontSendPackets;
int lastError;
bool micMuted;
uint32_t maxBitrate;
void (*stateCallback)(VoIPController*, int);
std::vector<voip_stream_t*> outgoingStreams;
std::vector<voip_stream_t*> 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<voip_queued_packet_t*> queuedPackets;
tgvoip_mutex_t queuedPacketsMutex;
double connectionInitTime;
double lastRecvPacketTime;
voip_config_t config;
int32_t peerVersion;
CongestionControl* conctl;
voip_stats_t stats;
bool receivedInit;
bool receivedInitAck;
std::vector<std::string> debugLogs;
bool isOutgoing;
NetworkSocket* udpSocket;
NetworkSocket* realUdpSocket;
FILE* statsDump;
std::string currentAudioInput;
std::string currentAudioOutput;
bool useTCP;
bool useUDP;
bool didAddTcpRelays;
double setEstablishedAt;
SocketSelectCanceller* selectCanceller;
NetworkSocket* openingTcpSocket;
unsigned char signalBarsHistory[4];
BufferPool outgoingPacketsBufferPool;
int udpConnectivityState;
double lastUdpPingTime;
int udpPingCount;
int echoCancellationStrength;
int proxyProtocol;
std::string proxyAddress;
uint16_t proxyPort;
std::string proxyUsername;
std::string proxyPassword;
IPv4Address* resolvedProxyAddress;
int signalBarCount;
void (*signalBarCountCallback)(VoIPController*, int);
AutomaticGainControl* outputAGC;
bool outputAGCEnabled;
/*** 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;
#ifdef TGVOIP_USE_AUDIO_SESSION
void (^acquireAudioSession)(void (^)());
bool needNotifyAcquiredAudioSession;
#endif
public:
#ifdef __APPLE__
static double machTimebase;
static uint64_t machTimestart;
#if TARGET_OS_IPHONE
// temporary fix for nasty linking errors
void SetRemoteEndpoints(voip_legacy_endpoint_t* buffer, size_t count, bool allowP2P);
#endif
#endif
#ifdef _WIN32
static int64_t win32TimeScale;
static bool didInitWin32TimeScale;
#endif
};
}
#endif
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