ZLMediaKit/src/Rtsp/RtspSession.h
2018-09-25 09:55:41 +08:00

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/*
* MIT License
*
* Copyright (c) 2016 xiongziliang <771730766@qq.com>
*
* This file is part of ZLMediaKit(https://github.com/xiongziliang/ZLMediaKit).
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all
* copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
* SOFTWARE.
*/
#ifndef SESSION_RTSPSESSION_H_
#define SESSION_RTSPSESSION_H_
#include <set>
#include <vector>
#include <unordered_map>
#include "Common/config.h"
#include "Rtsp.h"
#include "RtpBroadCaster.h"
#include "RtspMediaSource.h"
#include "Player/PlayerBase.h"
#include "Util/util.h"
#include "Util/logger.h"
#include "Network/TcpSession.h"
#include "Http/HttpRequestSplitter.h"
using namespace std;
using namespace ZL::Util;
using namespace ZL::Rtsp;
using namespace ZL::Player;
using namespace ZL::Network;
using namespace ZL::Http;
namespace ZL {
namespace Rtsp {
class RtspSession;
class BufferRtp : public Buffer{
public:
typedef std::shared_ptr<BufferRtp> Ptr;
BufferRtp(const RtpPacket::Ptr & pkt,uint32_t offset = 0 ):_rtp(pkt),_offset(offset){}
virtual ~BufferRtp(){}
char *data() override {
return (char *)_rtp->payload + _offset;
}
uint32_t size() const override {
return _rtp->length - _offset;
}
private:
RtpPacket::Ptr _rtp;
uint32_t _offset;
};
class RtspSession: public TcpSession, public HttpRequestSplitter {
public:
typedef std::shared_ptr<RtspSession> Ptr;
typedef std::function<void(const string &realm)> onGetRealm;
//encrypted为true是则表明是md5加密的密码否则是明文密码
//在请求明文密码时如果提供md5密码者则会导致认证失败
typedef std::function<void(bool encrypted,const string &pwd_or_md5)> onAuth;
RtspSession(const std::shared_ptr<ThreadPool> &pTh, const Socket::Ptr &pSock);
virtual ~RtspSession();
void onRecv(const Buffer::Ptr &pBuf) override;
void onError(const SockException &err) override;
void onManager() override;
protected:
//HttpRequestSplitter override
int64_t onRecvHeader(const char *data,uint64_t len) override ;
private:
void inputRtspOrRtcp(const char *data,uint64_t len);
int send(const string &strBuf) override {
m_ui64TotalBytes += strBuf.size();
return m_pSender->send(strBuf);
}
int send(string &&strBuf) override {
m_ui64TotalBytes += strBuf.size();
return m_pSender->send(std::move(strBuf));
}
int send(const char *pcBuf, int iSize) override {
m_ui64TotalBytes += iSize;
return m_pSender->send(pcBuf, iSize);
}
int send(const Buffer::Ptr &pkt) override{
m_ui64TotalBytes += pkt->size();
return m_pSender->send(pkt,SOCKET_DEFAULE_FLAGS | FLAG_MORE);
}
void shutdown() override ;
void shutdown_l(bool close);
bool handleReq_Options(); //处理options方法
bool handleReq_Describe(); //处理describe方法
bool handleReq_Setup(); //处理setup方法
bool handleReq_Play(); //处理play方法
bool handleReq_Pause(); //处理pause方法
bool handleReq_Teardown(); //处理teardown方法
bool handleReq_Get(); //处理Get方法
bool handleReq_Post(); //处理Post方法
bool handleReq_SET_PARAMETER(); //处理SET_PARAMETER方法
void inline send_StreamNotFound(); //rtsp资源未找到
void inline send_UnsupportedTransport(); //不支持的传输模式
void inline send_SessionNotFound(); //会话id错误
void inline send_NotAcceptable(); //rtsp同时播放数限制
inline bool findStream(); //根据rtsp url查找 MediaSource实例
inline void initSender(const std::shared_ptr<RtspSession> &pSession); //处理rtsp over httpquicktime使用的
inline void sendRtpPacket(const RtpPacket::Ptr &pkt);
inline string printSSRC(uint32_t ui32Ssrc) {
char tmp[9] = { 0 };
ui32Ssrc = htonl(ui32Ssrc);
uint8_t *pSsrc = (uint8_t *) &ui32Ssrc;
for (int i = 0; i < 4; i++) {
sprintf(tmp + 2 * i, "%02X", pSsrc[i]);
}
return tmp;
}
inline int getTrackIndexByTrackType(TrackType type) {
for (unsigned int i = 0; i < m_uiTrackCnt; i++) {
if (type == m_aTrackInfo[i].type) {
return i;
}
}
return -1;
}
inline int getTrackIndexByControlSuffix(const string &controlSuffix) {
for (unsigned int i = 0; i < m_uiTrackCnt; i++) {
if (controlSuffix == m_aTrackInfo[i].controlSuffix) {
return i;
}
}
return -1;
}
inline void onRcvPeerUdpData(int iTrackIdx, const Buffer::Ptr &pBuf, const struct sockaddr &addr);
inline void startListenPeerUdpData();
//认证相关
static void onAuthSuccess(const weak_ptr<RtspSession> &weakSelf);
static void onAuthFailed(const weak_ptr<RtspSession> &weakSelf,const string &realm);
static void onAuthUser(const weak_ptr<RtspSession> &weakSelf,const string &realm,const string &authorization);
static void onAuthBasic(const weak_ptr<RtspSession> &weakSelf,const string &realm,const string &strBase64);
static void onAuthDigest(const weak_ptr<RtspSession> &weakSelf,const string &realm,const string &strMd5);
private:
char *m_pcBuf = nullptr;
Ticker m_ticker;
Parser m_parser; //rtsp解析类
string m_strUrl;
string m_strSdp;
string m_strSession;
bool m_bFirstPlay = true;
MediaInfo m_mediaInfo;
std::weak_ptr<RtspMediaSource> m_pMediaSrc;
//RTP缓冲
weak_ptr<RingBuffer<RtpPacket::Ptr> > m_pWeakRing;
RingBuffer<RtpPacket::Ptr>::RingReader::Ptr m_pRtpReader;
PlayerBase::eRtpType m_rtpType = PlayerBase::RTP_UDP;
bool m_bSetUped = false;
int m_iCseq = 0;
unsigned int m_uiTrackCnt = 0; //媒体track个数
RtspTrack m_aTrackInfo[2]; //媒体track信息,trackid idx 为数组下标
bool m_bGotAllPeerUdp = false;
#ifdef RTSP_SEND_RTCP
RtcpCounter m_aRtcpCnt[2]; //rtcp统计,trackid idx 为数组下标
Ticker m_aRtcpTicker[2]; //rtcp发送时间,trackid idx 为数组下标
inline void sendRTCP();
#endif
//RTP over UDP
bool m_abGotPeerUdp[2] = { false, false }; //获取客户端udp端口计数
weak_ptr<Socket> m_apUdpSock[2]; //发送RTP的UDP端口,trackid idx 为数组下标
std::shared_ptr<struct sockaddr> m_apPeerUdpAddr[2]; //播放器接收RTP的地址,trackid idx 为数组下标
bool m_bListenPeerUdpData = false;
RtpBroadCaster::Ptr m_pBrdcaster;
//登录认证
string m_strNonce;
//RTSP over HTTP
function<void(void)> m_onDestory;
bool m_bBase64need = false; //是否需要base64解码
Socket::Ptr m_pSender; //回复rtsp时走的tcp通道供quicktime用
//quicktime 请求rtsp会产生两次tcp连接
//一次发送 get 一次发送post需要通过sessioncookie关联起来
string m_strSessionCookie;
//消耗的总流量
uint64_t m_ui64TotalBytes = 0;
static recursive_mutex g_mtxGetter; //对quicktime上锁保护
static recursive_mutex g_mtxPostter; //对quicktime上锁保护
static unordered_map<string, weak_ptr<RtspSession> > g_mapGetter;
static unordered_map<void *, std::shared_ptr<RtspSession> > g_mapPostter;
};
} /* namespace Session */
} /* namespace ZL */
#endif /* SESSION_RTSPSESSION_H_ */