mirror of
https://github.com/danog/libtgvoip.git
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524 lines
13 KiB
C++
524 lines
13 KiB
C++
//
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// libtgvoip is free and unencumbered public domain software.
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// For more information, see http://unlicense.org or the UNLICENSE file
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// you should have received with this source code distribution.
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//
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#ifndef __VOIPCONTROLLER_H
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#define __VOIPCONTROLLER_H
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#ifndef _WIN32
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#include <arpa/inet.h>
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#include <netinet/in.h>
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#endif
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#ifdef __APPLE__
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#include <TargetConditionals.h>
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#endif
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#include <stdint.h>
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#include <vector>
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#include <string>
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#include <map>
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#include "audio/AudioInput.h"
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#include "BlockingQueue.h"
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#include "BufferOutputStream.h"
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#include "audio/AudioOutput.h"
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#include "JitterBuffer.h"
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#include "OpusDecoder.h"
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#include "OpusEncoder.h"
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#include "EchoCanceller.h"
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#include "CongestionControl.h"
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#include "NetworkSocket.h"
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#define LIBTGVOIP_VERSION "1.0.3"
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#define STATE_CREATED 0
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#define STATE_WAIT_INIT 1
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#define STATE_WAIT_INIT_ACK 2
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#define STATE_ESTABLISHED 3
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#define STATE_FAILED 4
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#define STATE_RECONNECTING 5
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#define TGVOIP_ERROR_UNKNOWN 0
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#define TGVOIP_ERROR_INCOMPATIBLE 1
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#define TGVOIP_ERROR_TIMEOUT 2
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#define TGVOIP_ERROR_AUDIO_IO 3
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#define NET_TYPE_UNKNOWN 0
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#define NET_TYPE_GPRS 1
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#define NET_TYPE_EDGE 2
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#define NET_TYPE_3G 3
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#define NET_TYPE_HSPA 4
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#define NET_TYPE_LTE 5
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#define NET_TYPE_WIFI 6
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#define NET_TYPE_ETHERNET 7
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#define NET_TYPE_OTHER_HIGH_SPEED 8
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#define NET_TYPE_OTHER_LOW_SPEED 9
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#define NET_TYPE_DIALUP 10
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#define NET_TYPE_OTHER_MOBILE 11
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#define EP_TYPE_UDP_P2P_INET 1
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#define EP_TYPE_UDP_P2P_LAN 2
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#define EP_TYPE_UDP_RELAY 3
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#define EP_TYPE_TCP_RELAY 4
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#define DATA_SAVING_NEVER 0
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#define DATA_SAVING_MOBILE 1
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#define DATA_SAVING_ALWAYS 2
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#ifdef _WIN32
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#undef GetCurrentTime
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#endif
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struct voip_stream_t{
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int32_t userID;
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unsigned char id;
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unsigned char type;
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unsigned char codec;
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bool enabled;
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uint16_t frameDuration;
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};
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typedef struct voip_stream_t voip_stream_t;
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struct voip_queued_packet_t{
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unsigned char type;
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unsigned char* data;
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size_t length;
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uint32_t seqs[16];
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double firstSentTime;
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double lastSentTime;
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double retryInterval;
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double timeout;
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};
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typedef struct voip_queued_packet_t voip_queued_packet_t;
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struct voip_config_t{
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double init_timeout;
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double recv_timeout;
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int data_saving;
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char logFilePath[256];
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char statsDumpFilePath[256];
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bool enableAEC;
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bool enableNS;
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bool enableAGC;
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};
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typedef struct voip_config_t voip_config_t;
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#if defined(__APPLE__) && TARGET_OS_IPHONE
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// temporary fix for nasty linking errors
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struct voip_legacy_endpoint_t{
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const char* address;
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const char* address6;
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uint16_t port;
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int64_t id;
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unsigned char peerTag[16];
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};
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typedef struct voip_legacy_endpoint_t voip_legacy_endpoint_t;
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#endif
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struct voip_stats_t{
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uint64_t bytesSentWifi;
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uint64_t bytesRecvdWifi;
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uint64_t bytesSentMobile;
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uint64_t bytesRecvdMobile;
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};
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typedef struct voip_stats_t voip_stats_t;
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struct voip_crypto_functions_t{
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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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};
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typedef struct voip_crypto_functions_t voip_crypto_functions_t;
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#define SEQ_MAX 0xFFFFFFFF
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inline bool seqgt(uint32_t s1, uint32_t s2){
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return ((s1>s2) && (s1-s2<=SEQ_MAX/2)) || ((s1<s2) && (s2-s1>SEQ_MAX/2));
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}
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namespace tgvoip{
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enum{
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PROXY_NONE=0,
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PROXY_SOCKS5,
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//PROXY_HTTP
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};
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class Endpoint{
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friend class VoIPController;
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public:
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Endpoint(int64_t id, uint16_t port, IPv4Address& address, IPv6Address& v6address, char type, unsigned char* peerTag);
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Endpoint();
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int64_t id;
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uint16_t port;
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IPv4Address address;
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IPv6Address v6address;
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char type;
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unsigned char peerTag[16];
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private:
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double lastPingTime;
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uint32_t lastPingSeq;
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double rtts[6];
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double averageRTT;
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NetworkSocket* socket;
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};
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class AudioDevice{
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public:
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std::string id;
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std::string displayName;
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};
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class AudioOutputDevice : public AudioDevice{
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};
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class AudioInputDevice : public AudioDevice{
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};
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class VoIPController
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{
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public:
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VoIPController();
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~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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*/
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void SetRemoteEndpoints(std::vector<Endpoint> endpoints, bool allowP2p);
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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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* 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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* @param buffer The buffer to put the string into
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* @param len The length of the buffer
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*/
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void GetDebugString(char* buffer, size_t len);
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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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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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/**
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* Set the function to be called whenever the connection state changes
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* @param f
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*/
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void SetStateCallback(void (*f)(VoIPController*, int));
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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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void SetMicMute(bool mute);
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/**
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*
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* @param key
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* @param isOutgoing
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*/
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void SetEncryptionKey(char* key, bool isOutgoing);
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/**
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*
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* @param cfg
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*/
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void SetConfig(voip_config_t* cfg);
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float GetOutputLevel();
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void DebugCtl(int request, int param);
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/**
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*
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* @param stats
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*/
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void GetStats(voip_stats_t* 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 voip_crypto_functions_t 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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#ifdef TGVOIP_USE_AUDIO_SESSION
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void SetAcquireAudioSession(void (^)(void (^)()));
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void ReleaseAudioSession(void (^completion)());
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#endif
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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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* @param buffer
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*/
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void GetDebugLog(char* buffer);
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size_t GetDebugLogLength();
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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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* Set the callback to be called when the signal bar count changes.
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* @param f
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*/
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void SetSignalBarsCountCallback(void (*f)(VoIPController*, int));
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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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void SetEchoCancellationStrength(int strength);
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private:
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struct PendingOutgoingPacket{
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uint32_t seq;
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unsigned char type;
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size_t len;
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unsigned char* data;
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Endpoint* endpoint;
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};
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enum{
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UDP_UNKNOWN=0,
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UDP_PING_SENT,
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UDP_AVAILABLE,
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UDP_NOT_AVAILABLE
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};
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static void* StartRecvThread(void* arg);
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static void* StartSendThread(void* arg);
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static void* StartTickThread(void* arg);
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void RunRecvThread();
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void RunSendThread();
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void RunTickThread();
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void SendPacket(unsigned char* data, size_t len, Endpoint* ep);
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void HandleAudioInput(unsigned char* data, size_t len);
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void UpdateAudioBitrate();
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void SetState(int state);
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void UpdateAudioOutputState();
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void SendInit();
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void InitUDPProxy();
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void UpdateDataSavingState();
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void KDF(unsigned char* msgKey, size_t x, unsigned char* aesKey, unsigned char* aesIv);
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void WritePacketHeader(uint32_t seq, BufferOutputStream* s, unsigned char type, uint32_t length);
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static size_t AudioInputCallback(unsigned char* data, size_t length, void* param);
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void SendPublicEndpointsRequest();
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void SendPublicEndpointsRequest(Endpoint& relay);
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Endpoint* GetEndpointByType(int type);
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void SendPacketReliably(unsigned char type, unsigned char* data, size_t len, double retryInterval, double timeout);
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uint32_t GenerateOutSeq();
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void LogDebugInfo();
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void SendUdpPing(Endpoint* endpoint);
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int state;
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std::vector<Endpoint*> endpoints;
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Endpoint* currentEndpoint;
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Endpoint* preferredRelay;
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Endpoint* peerPreferredRelay;
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bool runReceiver;
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uint32_t seq;
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uint32_t lastRemoteSeq;
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uint32_t lastRemoteAckSeq;
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uint32_t lastSentSeq;
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double remoteAcks[32];
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double sentPacketTimes[32];
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double recvPacketTimes[32];
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uint32_t sendLossCountHistory[32];
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uint32_t audioTimestampIn;
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uint32_t audioTimestampOut;
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tgvoip::audio::AudioInput* audioInput;
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tgvoip::audio::AudioOutput* audioOutput;
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JitterBuffer* jitterBuffer;
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OpusDecoder* decoder;
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OpusEncoder* encoder;
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BlockingQueue<PendingOutgoingPacket>* sendQueue;
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EchoCanceller* echoCanceller;
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tgvoip_mutex_t sendBufferMutex;
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tgvoip_mutex_t endpointsMutex;
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bool stopping;
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bool audioOutStarted;
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tgvoip_thread_t recvThread;
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tgvoip_thread_t sendThread;
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tgvoip_thread_t tickThread;
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uint32_t packetsRecieved;
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uint32_t recvLossCount;
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uint32_t prevSendLossCount;
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uint32_t firstSentPing;
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double rttHistory[32];
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bool waitingForAcks;
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int networkType;
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int dontSendPackets;
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int lastError;
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bool micMuted;
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uint32_t maxBitrate;
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void (*stateCallback)(VoIPController*, int);
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std::vector<voip_stream_t*> outgoingStreams;
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std::vector<voip_stream_t*> incomingStreams;
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unsigned char encryptionKey[256];
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unsigned char keyFingerprint[8];
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unsigned char callID[16];
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double stateChangeTime;
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bool waitingForRelayPeerInfo;
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bool allowP2p;
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bool dataSavingMode;
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bool dataSavingRequestedByPeer;
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std::string activeNetItfName;
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double publicEndpointsReqTime;
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std::vector<voip_queued_packet_t*> queuedPackets;
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tgvoip_mutex_t queuedPacketsMutex;
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double connectionInitTime;
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double lastRecvPacketTime;
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voip_config_t config;
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int32_t peerVersion;
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CongestionControl* conctl;
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voip_stats_t stats;
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bool receivedInit;
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bool receivedInitAck;
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std::vector<std::string> debugLogs;
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bool isOutgoing;
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NetworkSocket* udpSocket;
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NetworkSocket* realUdpSocket;
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FILE* statsDump;
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std::string currentAudioInput;
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std::string currentAudioOutput;
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bool useTCP;
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bool useUDP;
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bool didAddTcpRelays;
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double setEstablishedAt;
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SocketSelectCanceller* selectCanceller;
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NetworkSocket* openingTcpSocket;
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unsigned char signalBarsHistory[4];
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BufferPool outgoingPacketsBufferPool;
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int udpConnectivityState;
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double lastUdpPingTime;
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int udpPingCount;
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int echoCancellationStrength;
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int proxyProtocol;
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std::string proxyAddress;
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uint16_t proxyPort;
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std::string proxyUsername;
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std::string proxyPassword;
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IPv4Address* resolvedProxyAddress;
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int signalBarCount;
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void (*signalBarCountCallback)(VoIPController*, int);
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AutomaticGainControl* outputAGC;
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bool outputAGCEnabled;
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/*** server config values ***/
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uint32_t maxAudioBitrate;
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uint32_t maxAudioBitrateEDGE;
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uint32_t maxAudioBitrateGPRS;
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uint32_t maxAudioBitrateSaving;
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uint32_t initAudioBitrate;
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uint32_t initAudioBitrateEDGE;
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uint32_t initAudioBitrateGPRS;
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uint32_t initAudioBitrateSaving;
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uint32_t minAudioBitrate;
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uint32_t audioBitrateStepIncr;
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uint32_t audioBitrateStepDecr;
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double relaySwitchThreshold;
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double p2pToRelaySwitchThreshold;
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double relayToP2pSwitchThreshold;
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double reconnectingTimeout;
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#ifdef TGVOIP_USE_AUDIO_SESSION
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void (^acquireAudioSession)(void (^)());
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bool needNotifyAcquiredAudioSession;
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#endif
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public:
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#ifdef __APPLE__
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static double machTimebase;
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static uint64_t machTimestart;
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#if TARGET_OS_IPHONE
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// temporary fix for nasty linking errors
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void SetRemoteEndpoints(voip_legacy_endpoint_t* buffer, size_t count, bool allowP2P);
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#endif
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#endif
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#ifdef _WIN32
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static int64_t win32TimeScale;
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static bool didInitWin32TimeScale;
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#endif
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};
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}
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#endif
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