mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-01 17:11:31 +08:00
208 lines
7.6 KiB
C++
208 lines
7.6 KiB
C++
/*
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* MIT License
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*
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* Copyright (c) 2016 xiongziliang <771730766@qq.com>
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*
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* This file is part of ZLMediaKit(https://github.com/xiongziliang/ZLMediaKit).
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in all
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* copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
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* SOFTWARE.
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*/
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#ifndef SESSION_RTSPSESSION_H_
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#define SESSION_RTSPSESSION_H_
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#include <set>
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#include <vector>
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#include <unordered_map>
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#include "Util/util.h"
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#include "Util/logger.h"
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#include "Common/config.h"
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#include "Network/TcpSession.h"
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#include "Player/PlayerBase.h"
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#include "Rtsp.h"
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#include "RtpBroadCaster.h"
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#include "RtspMediaSource.h"
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#include "RtspSplitter.h"
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#include "RtpReceiver.h"
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#include "RtspToRtmpMediaSource.h"
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using namespace std;
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using namespace toolkit;
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namespace mediakit {
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class RtspSession;
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class BufferRtp : public Buffer{
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public:
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typedef std::shared_ptr<BufferRtp> Ptr;
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BufferRtp(const RtpPacket::Ptr & pkt,uint32_t offset = 0 ):_rtp(pkt),_offset(offset){}
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virtual ~BufferRtp(){}
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char *data() const override {
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return (char *)_rtp->payload + _offset;
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}
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uint32_t size() const override {
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return _rtp->length - _offset;
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}
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private:
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RtpPacket::Ptr _rtp;
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uint32_t _offset;
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};
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class RtspSession: public TcpSession, public RtspSplitter, public RtpReceiver , public MediaSourceEvent{
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public:
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typedef std::shared_ptr<RtspSession> Ptr;
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typedef std::function<void(const string &realm)> onGetRealm;
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//encrypted为true是则表明是md5加密的密码，否则是明文密码
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//在请求明文密码时如果提供md5密码者则会导致认证失败
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typedef std::function<void(bool encrypted,const string &pwd_or_md5)> onAuth;
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RtspSession(const std::shared_ptr<ThreadPool> &pTh, const Socket::Ptr &pSock);
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virtual ~RtspSession();
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void onRecv(const Buffer::Ptr &pBuf) override;
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void onError(const SockException &err) override;
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void onManager() override;
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protected:
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//RtspSplitter override
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/**
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* 收到完整的rtsp包回调，包括sdp等content数据
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* @param parser rtsp包
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*/
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void onWholeRtspPacket(Parser &parser) override;
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/**
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* 收到rtp包回调
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* @param data
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* @param len
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*/
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void onRtpPacket(const char *data,uint64_t len) override;
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/**
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* 从rtsp头中获取Content长度
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* @param parser
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* @return
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*/
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int64_t getContentLength(Parser &parser) override;
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//RtpReceiver override
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void onRtpSorted(const RtpPacket::Ptr &rtppt, int trackidx) override;
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//MediaSourceEvent override
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bool close() override ;
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//TcpSession override
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int send(const Buffer::Ptr &pkt) override;
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private:
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bool handleReq_Options(const Parser &parser); //处理options方法
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bool handleReq_Describe(const Parser &parser); //处理describe方法
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bool handleReq_ANNOUNCE(const Parser &parser); //处理options方法
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bool handleReq_RECORD(const Parser &parser); //处理options方法
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bool handleReq_Setup(const Parser &parser); //处理setup方法
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bool handleReq_Play(const Parser &parser); //处理play方法
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bool handleReq_Pause(const Parser &parser); //处理pause方法
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bool handleReq_Teardown(const Parser &parser); //处理teardown方法
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bool handleReq_Get(const Parser &parser); //处理Get方法
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bool handleReq_Post(const Parser &parser); //处理Post方法
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bool handleReq_SET_PARAMETER(const Parser &parser); //处理SET_PARAMETER方法
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void inline send_StreamNotFound(); //rtsp资源未找到
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void inline send_UnsupportedTransport(); //不支持的传输模式
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void inline send_SessionNotFound(); //会话id错误
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void inline send_NotAcceptable(); //rtsp同时播放数限制
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inline bool findStream(); //根据rtsp url查找 MediaSource实例
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inline void findStream(const function<void(bool)> &cb); //根据rtsp url查找 MediaSource实例
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inline string printSSRC(uint32_t ui32Ssrc);
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inline int getTrackIndexByTrackType(TrackType type);
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inline int getTrackIndexByControlSuffix(const string &controlSuffix);
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inline int getTrackIndexByInterleaved(int interleaved);
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inline void onRcvPeerUdpData(int iTrackIdx, const Buffer::Ptr &pBuf, const struct sockaddr &addr);
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inline void startListenPeerUdpData(int iTrackIdx);
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//认证相关
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static void onAuthSuccess(const weak_ptr<RtspSession> &weakSelf);
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static void onAuthFailed(const weak_ptr<RtspSession> &weakSelf,const string &realm);
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static void onAuthUser(const weak_ptr<RtspSession> &weakSelf,const string &realm,const string &authorization);
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static void onAuthBasic(const weak_ptr<RtspSession> &weakSelf,const string &realm,const string &strBase64);
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static void onAuthDigest(const weak_ptr<RtspSession> &weakSelf,const string &realm,const string &strMd5);
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void doDelay(int delaySec,const std::function<void()> &fun);
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void cancelDelyaTask();
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inline void sendRtpPacket(const RtpPacket::Ptr &pkt);
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bool sendRtspResponse(const string &res_code,const std::initializer_list<string> &header, const string &sdp = "" , const char *protocol = "RTSP/1.0");
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bool sendRtspResponse(const string &res_code,const StrCaseMap &header = StrCaseMap(), const string &sdp = "",const char *protocol = "RTSP/1.0");
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private:
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Ticker _ticker;
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int _iCseq = 0;
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string _strUrl;
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string _strSdp;
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string _strSession;
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bool _bFirstPlay = true;
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MediaInfo _mediaInfo;
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std::weak_ptr<RtspMediaSource> _pMediaSrc;
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RingBuffer<RtpPacket::Ptr>::RingReader::Ptr _pRtpReader;
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PlayerBase::eRtpType _rtpType = PlayerBase::RTP_Invalid;
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vector<SdpTrack::Ptr> _aTrackInfo;
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//RTP over udp
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bool _bGotAllPeerUdp = false;
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bool _abGotPeerUdp[2] = { false, false }; //获取客户端udp端口计数
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Socket::Ptr _apRtpSock[2]; //RTP端口,trackid idx 为数组下标
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Socket::Ptr _apRtcpSock[2];//RTCP端口,trackid idx 为数组下标
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std::shared_ptr<struct sockaddr> _apPeerRtpPortAddr[2]; //播放器接收RTP的地址,trackid idx 为数组下标
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//RTP over udp_multicast
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RtpBroadCaster::Ptr _pBrdcaster;
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//登录认证
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string _strNonce;
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//消耗的总流量
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uint64_t _ui64TotalBytes = 0;
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//RTSP over HTTP
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//quicktime 请求rtsp会产生两次tcp连接，
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//一次发送 get 一次发送post，需要通过x-sessioncookie关联起来
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string _http_x_sessioncookie;
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function<void(const Buffer::Ptr &pBuf)> _onRecv;
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std::function<void()> _delayTask;
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uint32_t _iTaskTimeLine = 0;
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atomic<bool> _enableSendRtp;
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//rtsp推流相关
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RtspToRtmpMediaSource::Ptr _pushSrc;
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#ifdef RTSP_SEND_RTCP
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RtcpCounter _aRtcpCnt[2]; //rtcp统计,trackid idx 为数组下标
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Ticker _aRtcpTicker[2]; //rtcp发送时间,trackid idx 为数组下标
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inline void sendRTCP();
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#endif
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};
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/**
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* 支持ssl加密的rtsp服务器，可用于诸如亚马逊echo show这样的设备访问
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*/
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typedef TcpSessionWithSSL<RtspSession> RtspSessionWithSSL;
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} /* namespace mediakit */
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#endif /* SESSION_RTSPSESSION_H_ */
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