/* * Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved. * * This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit). * * Use of this source code is governed by MIT-like license that can be found in the * LICENSE file in the root of the source tree. All contributing project authors * may be found in the AUTHORS file in the root of the source tree. */ #ifndef SRC_RTSPPLAYER_RTSPPLAYER_H_TXT_ #define SRC_RTSPPLAYER_RTSPPLAYER_H_TXT_ #include #include #include "Util/TimeTicker.h" #include "Poller/Timer.h" #include "Network/Socket.h" #include "Player/PlayerBase.h" #include "Network/TcpClient.h" #include "RtspSplitter.h" #include "RtpReceiver.h" #include "Rtcp/RtcpContext.h" namespace mediakit { // 实现了rtsp播放器协议部分的功能，及数据接收功能 [AUTO-TRANSLATED:c1ed5c0f] // Implemented the rtsp player protocol part functionality, and data receiving functionality class RtspPlayer : public PlayerBase, public toolkit::TcpClient, public RtspSplitter, public RtpReceiver { public: using Ptr = std::shared_ptr; RtspPlayer(const toolkit::EventPoller::Ptr &poller); ~RtspPlayer() override; void play(const std::string &strUrl) override; void pause(bool pause) override; void speed(float speed) override; void teardown() override; float getPacketLossRate(TrackType type) const override; protected: // 派生类回调函数 [AUTO-TRANSLATED:61e20903] // Derived class callback function virtual bool onCheckSDP(const std::string &sdp) = 0; virtual void onRecvRTP(RtpPacket::Ptr rtp, const SdpTrack::Ptr &track) = 0; uint32_t getProgressMilliSecond() const; void seekToMilliSecond(uint32_t ms); /** * 收到完整的rtsp包回调，包括sdp等content数据 * @param parser rtsp包 * Callback for receiving a complete rtsp packet, including sdp and other content data * @param parser rtsp packet * [AUTO-TRANSLATED:4d3c2056] */ void onWholeRtspPacket(Parser &parser) override ; /** * 收到rtp包回调 * @param data * @param len * Callback for receiving rtp packet * @param data * @param len * [AUTO-TRANSLATED:c8f7c9bb] */ void onRtpPacket(const char *data,size_t len) override ; /** * rtp数据包排序后输出 * @param rtp rtp数据包 * @param track_idx track索引 * Output rtp data packets after sorting * @param rtp rtp data packet * @param track_idx track index * [AUTO-TRANSLATED:8f9ca364] */ void onRtpSorted(RtpPacket::Ptr rtp, int track_idx) override; /** * 解析出rtp但还未排序 * @param rtp rtp数据包 * @param track_index track索引 * Parse out rtp but not yet sorted * @param rtp rtp data packet * @param track_index track index * [AUTO-TRANSLATED:c1636911] */ void onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_index) override; /** * 收到RTCP包回调 * @param track_idx track索引 * @param track sdp相关信息 * @param data rtcp内容 * @param len rtcp内容长度 * Callback for receiving RTCP packet * @param track_idx track index * @param track sdp related information * @param data rtcp content * @param len rtcp content length * [AUTO-TRANSLATED:1a2cfa4f] */ virtual void onRtcpPacket(int track_idx, SdpTrack::Ptr &track, uint8_t *data, size_t len); /////////////TcpClient override///////////// void onConnect(const toolkit::SockException &err) override; void onRecv(const toolkit::Buffer::Ptr &buf) override; void onError(const toolkit::SockException &ex) override; private: void onPlayResult_l(const toolkit::SockException &ex , bool handshake_done); int getTrackIndexByInterleaved(int interleaved) const; int getTrackIndexByTrackType(TrackType track_type) const; void handleResSETUP(const Parser &parser, unsigned int track_idx); void handleResDESCRIBE(const Parser &parser); bool handleAuthenticationFailure(const std::string &wwwAuthenticateParamsStr); void handleResPAUSE(const Parser &parser, int type); bool handleResponse(const std::string &cmd, const Parser &parser); void sendOptions(); void sendSetup(unsigned int track_idx); void sendPause(int type , uint32_t ms); void sendDescribe(); void sendTeardown(); void sendKeepAlive(); void sendRtspRequest(const std::string &cmd, const std::string &url ,const StrCaseMap &header = StrCaseMap()); void sendRtspRequest(const std::string &cmd, const std::string &url ,const std::initializer_list &header); void createUdpSockIfNecessary(int track_idx); private: // 是否为性能测试模式 [AUTO-TRANSLATED:1fde8234] // Whether it is performance test mode bool _benchmark_mode = false; // 轮流发送rtcp与GET_PARAMETER保活 [AUTO-TRANSLATED:5b6f9c37] // Send rtcp and GET_PARAMETER keep-alive in turn bool _send_rtcp[2] = {true, true}; // 心跳类型 [AUTO-TRANSLATED:c22abb05] // Heartbeat type uint32_t _beat_type = 0; // 心跳保护间隔 [AUTO-TRANSLATED:de16d9c9] // Heartbeat protection interval uint32_t _beat_interval_ms = 0; std::string _play_url; // rtsp开始倍速 [AUTO-TRANSLATED:9ab84508] // Rtsp start speed float _speed= 0.0f; std::vector _sdp_track; std::function _on_response; // RTP端口,trackid idx 为数组下标 [AUTO-TRANSLATED:77c186bb] // RTP port, trackid idx is the array subscript toolkit::Socket::Ptr _rtp_sock[2]; // RTCP端口,trackid idx 为数组下标 [AUTO-TRANSLATED:446a7861] // RTCP port, trackid idx is the array subscript toolkit::Socket::Ptr _rtcp_sock[2]; // rtsp鉴权相关 [AUTO-TRANSLATED:947dc6a3] // Rtsp authentication related std::string _md5_nonce; std::string _realm; //rtsp info std::string _session_id; uint32_t _cseq_send = 1; std::string _content_base; std::string _control_url; Rtsp::eRtpType _rtp_type = Rtsp::RTP_TCP; // 当前rtp时间戳 [AUTO-TRANSLATED:410f2691] // Current rtp timestamp uint32_t _stamp[2] = {0, 0}; // 超时功能实现 [AUTO-TRANSLATED:1d603b3a] // Timeout function implementation toolkit::Ticker _rtp_recv_ticker; std::shared_ptr _play_check_timer; std::shared_ptr _rtp_check_timer; // 服务器支持的命令 [AUTO-TRANSLATED:f7f589bf] // Server supported commands std::set _supported_cmd; ////////// rtcp //////////////// // rtcp发送时间,trackid idx 为数组下标 [AUTO-TRANSLATED:bf3248b1] // Rtcp send time, trackid idx is the array subscript toolkit::Ticker _rtcp_send_ticker[2]; // 统计rtp并发送rtcp [AUTO-TRANSLATED:0ac2b665] // Statistics rtp and send rtcp std::vector _rtcp_context; }; } /* namespace mediakit */ #endif /* SRC_RTSPPLAYER_RTSPPLAYER_H_TXT_ */