socket_pair.h

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/*
 *  Copyright (C) 2004-2026 Savoir-faire Linux Inc.
 *  Copyright (C) 2012 VLC authors and VideoLAN
 *
 *  This program is free software: you can redistribute it and/or modify
 *  it under the terms of the GNU General Public License as published by
 *  the Free Software Foundation, either version 3 of the License, or
 *  (at your option) any later version.
 *
 *  This program is distributed in the hope that it will be useful,
 *  but WITHOUT ANY WARRANTY; without even the implied warranty of
 *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
 *  GNU General Public License for more details.
 *
 *  You should have received a copy of the GNU General Public License
 *  along with this program. If not, see <https://www.gnu.org/licenses/>.
 */
#pragma once
 
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
 
#include "media_io_handle.h"
 
#ifndef _WIN32
#include <sys/socket.h>
#include <netdb.h>
#include <poll.h>
#else
#include <winsock2.h>
#include <ws2tcpip.h>
using socklen_t = int;
#endif
 
#include <dhtnet/ip_utils.h>
#include <dhtnet/ice_socket.h>
 
#include <cstdint>
#include <mutex>
#include <memory>
#include <atomic>
#include <list>
#include <vector>
#include <condition_variable>
#include <functional>
 
namespace jami {
 
class SRTPProtoContext;
 
typedef struct
{
#ifdef WORDS_BIGENDIAN
    uint32_t version : 2; /* protocol version */
    uint32_t p : 1;       /* padding flag */
    uint32_t rc : 5;      /* reception report count must be 201 for report */
 
#else
    uint32_t rc : 5;      /* reception report count must be 201 for report */
    uint32_t p : 1;       /* padding flag */
    uint32_t version : 2; /* protocol version */
#endif
    uint32_t pt : 8;               /* payload type */
    uint32_t len : 16;             /* length of RTCP packet */
    uint32_t ssrc;                 /* synchronization source identifier of packet send */
    uint32_t id;                   /* synchronization source identifier of first source */
    uint32_t fraction_lost : 8;    /* 8 bits of fraction, 24 bits of total packets lost */
    uint32_t cum_lost_packet : 24; /* cumulative number packet lost */
    uint32_t ext_high;             /* Extended highest sequence number received */
    uint32_t jitter;               /* jitter */
    uint32_t lsr;                  /* last SR timestamp */
    uint32_t dlsr;                 /* Delay since last SR timestamp */
} rtcpRRHeader;
 
typedef struct
{
#ifdef WORDS_BIGENDIAN
    uint32_t version : 2; /* protocol version */
    uint32_t p : 1;       /* padding flag */
    uint32_t rc : 5;      /* reception report count must be 201 for report */
 
#else
    uint32_t rc : 5;      /* reception report count must be 201 for report */
    uint32_t p : 1;       /* padding flag */
    uint32_t version : 2; /* protocol version */
#endif
    uint32_t pt : 8;       /* payload type */
    uint32_t len : 16;     /* length of RTCP packet */
    uint32_t ssrc;         /* synchronization source identifier of packet send */
    uint32_t timestampMSB; /* timestamp MSB */
    uint32_t timestampLSB; /* timestamp LSB */
    uint32_t timestampRTP; /* RTP timestamp */
    uint32_t spc;          /* Sender's packet count */
    uint32_t soc;          /* Sender's octet count */
} rtcpSRHeader;
 
typedef struct
{
#ifdef WORDS_BIGENDIAN
    uint32_t version : 2; /* protocol version */
    uint32_t p : 1;       /* padding flag always 0 */
    uint32_t fmt : 5;     /* Feedback message type always 15 */
 
#else
    uint32_t fmt : 5;     /* Feedback message type always 15 */
    uint32_t p : 1;       /* padding flag always 0 */
    uint32_t version : 2; /* protocol version */
#endif
    uint32_t pt : 8;         /* payload type */
    uint32_t len : 16;       /* length of RTCP packet */
    uint32_t ssrc;           /* synchronization source identifier of packet sender */
    uint32_t ssrc_source;    /* synchronization source identifier of first source alway 0*/
    uint32_t uid;            /* Unique identifier Always ‘R’ ‘E’ ‘M’ ‘B’ (4 ASCII characters). */
    uint32_t n_ssrc : 8;     /* Number of SSRCs in this message. */
    uint32_t br_exp : 6;     /* BR Exp */
    uint32_t br_mantis : 18; /* BR Mantissa */
    uint32_t f_ssrc;         /* SSRC feedback */
} rtcpREMBHeader;
 
typedef struct
{
    uint64_t last_send_ts;
    std::chrono::steady_clock::time_point last_receive_ts;
    uint64_t send_ts;
    std::chrono::steady_clock::time_point receive_ts;
} TS_Frame;
 
class SocketPair
{
public:
    SocketPair(const char* uri, int localPort);
    SocketPair(std::unique_ptr<dhtnet::IceSocket> rtp_sock, std::unique_ptr<dhtnet::IceSocket> rtcp_sock);
    ~SocketPair();
 
    void interrupt();
 
    // Set the read blocking mode.
    // By default, the read operation will block until data is available
    // on the socket. This method allows to switch to unblocking mode
    // if to stop the receiver thread to exit and there is no more data
    // to read (if the peer mutes/stops the media/RTP stream).
    void setReadBlockingMode(bool blocking);
 
    MediaIOHandle* createIOContext(const uint16_t mtu);
 
    void openSockets(const char* uri, int localPort);
    void closeSockets();
 
    /*
       Supported suites are:
 
       AES_CM_128_HMAC_SHA1_80
       SRTP_AES128_CM_HMAC_SHA1_80
       AES_CM_128_HMAC_SHA1_32
       SRTP_AES128_CM_HMAC_SHA1_3
 
       Example (unsecure) usage:
       createSRTP("AES_CM_128_HMAC_SHA1_80",
                  "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmn",
                  "AES_CM_128_HMAC_SHA1_80",
                  "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmn");
 
       Will throw an std::runtime_error on failure, should be handled at a higher level
    */
    void createSRTP(const char* out_suite, const char* out_params, const char* in_suite, const char* in_params);
 
    void stopSendOp(bool state = true);
    std::list<rtcpRRHeader> getRtcpRR();
    std::list<rtcpREMBHeader> getRtcpREMB();
 
    bool waitForRTCP(std::chrono::seconds interval);
    double getLastLatency();
 
    void setPacketLossCallback(std::function<void(void)> cb) { packetLossCallback_ = std::move(cb); }
    void setRtpDelayCallback(std::function<void(int, int)> cb);
 
    int writeData(uint8_t* buf, int buf_size);
 
    uint16_t lastSeqValOut();
 
private:
    NON_COPYABLE(SocketPair);
    using clock = std::chrono::steady_clock;
    using time_point = clock::time_point;
 
    int readCallback(uint8_t* buf, int buf_size);
    int writeCallback(uint8_t* buf, int buf_size);
 
    int waitForData();
    int readRtpData(void* buf, int buf_size);
    int readRtcpData(void* buf, int buf_size);
    void saveRtcpRRPacket(uint8_t* buf, size_t len);
    void saveRtcpREMBPacket(uint8_t* buf, size_t len);
 
    std::mutex dataBuffMutex_;
    std::condition_variable cv_;
    std::list<std::vector<uint8_t>> rtpDataBuff_;
    std::list<std::vector<uint8_t>> rtcpDataBuff_;
 
    std::unique_ptr<dhtnet::IceSocket> rtp_sock_;
    std::unique_ptr<dhtnet::IceSocket> rtcp_sock_;
 
    int rtpHandle_ {-1};
    int rtcpHandle_ {-1};
    dhtnet::IpAddr rtpDestAddr_;
    dhtnet::IpAddr rtcpDestAddr_;
    std::atomic_bool interrupted_ {false};
    // Read operations are blocking. This will allow unblocking the
    // receiver thread if the peer stops/mutes the media (RTP)
    std::atomic_bool readBlockingMode_ {false};
    std::atomic_bool noWrite_ {false};
    std::unique_ptr<SRTPProtoContext> srtpContext_;
    std::function<void(void)> packetLossCallback_;
    std::function<void(int, int)> rtpDelayCallback_;
    bool getOneWayDelayGradient(float sendTS, bool marker, int32_t* gradient, int32_t* deltaR);
    bool parse_RTP_ext(uint8_t* buf, float* abs);
 
    std::list<rtcpRRHeader> listRtcpRRHeader_;
    std::list<rtcpREMBHeader> listRtcpREMBHeader_;
    std::mutex rtcpInfo_mutex_;
    std::condition_variable cvRtcpPacketReadyToRead_;
    static constexpr unsigned MAX_LIST_SIZE {10};
 
    mutable std::atomic_bool rtcpPacketLoss_ {false};
    double lastSRTS_ {};
    uint32_t lastDLSR_ {};
 
    std::list<double> histoLatency_;
 
    time_point lastRR_time;
    uint16_t lastSeqNumIn_ {0};
    float lastSendTS_ {0.0f};
    time_point lastReceiveTS_ {};
    time_point arrival_TS {};
 
    TS_Frame svgTS = {};
};
 
} // namespace jami