使用live555 建立rtsp 服务

╰+哭是因爲堅強的太久メ 2024-02-23 02:04 41阅读 0赞

1、 live555  
live555除了可以建立客户端，当然也可以做一个服务端了，快速做一个服务端就把live555的lib文件编译好，放进来进行。

以h264 为例子，实际上，无论是读取文件也好，还是web摄像头编码也好，需要的其实就是sps，pps信息，就是这么简单。

1.1 读取文件  
如果是读取文件，读的过程中要把sps，pps记录下来，如果是加密流，分为两种情况，一是解密，二是直接传输。

1.2 摄像头编码发送  
摄像头采用编码h264或者h265后可以发送到连接的客户端，建立一个CameraFrameSource从FramedSource继承，声明如下：

class CameraFrameSource : public FramedSource {
       public:
          static CameraFrameSource* createNew(UsageEnvironment&, CameraH264Encoder*);
          CameraFrameSource(UsageEnvironment& env, CameraH264Encoder*);
          ~CameraFrameSource() = default;
    
       private:
          static void deliverFrameStub(void* clientData) {
             ((CameraFrameSource*)clientData)->deliverFrame();
          };
          void doGetNextFrame() override;
          void deliverFrame();
          void doStopGettingFrames() override;
          void onFrame();
    
       private:
          CameraH264Encoder   *fEncoder;
          EventTriggerId       fEventTriggerId;
       };

主要转发的代码如下

void CameraFrameSource::deliverFrame()
       {
          if (!isCurrentlyAwaitingData()) return; // we're not ready for the buff yet
    
          static uint8_t* newFrameDataStart;
          static unsigned newFrameSize = 0;
    
          /* get the buff frame from the Encoding thread.. */
          if (fEncoder->fetch_packet(&newFrameDataStart, &newFrameSize)) {
             if (newFrameDataStart != nullptr) {
                /* This should never happen, but check anyway.. */
                if (newFrameSize > fMaxSize) {
                   fFrameSize = fMaxSize;
                   fNumTruncatedBytes = newFrameSize - fMaxSize;
                }
                else {
                   fFrameSize = newFrameSize;
                }
    
                gettimeofday(&fPresentationTime, nullptr);
                memcpy(fTo, newFrameDataStart, fFrameSize);
    
                //delete newFrameDataStart;
                //newFrameSize = 0;
    
                fEncoder->release_packet();
             }
             else {
                fFrameSize = 0;
                fTo = nullptr;
                handleClosure(this);
             }
          }
          else {
             fFrameSize = 0;
          }
    
          if (fFrameSize > 0)
             FramedSource::afterGetting(this);
       };

主main函数里面实现rtspserver的run，把刚才的framesource加入到里面。如果是h265，同样的流程。

CameraRTSPServer::CameraRTSPServer(CameraH264Encoder * enc, int port, int httpPort)
          : fH264Encoder(enc), fPortNum(port), fHttpTunnelingPort(httpPort), fQuit(0), fBitRate(5120) //in kbs
       {
          fQuit = 0;
       };
    
       CameraRTSPServer::~CameraRTSPServer()
       {
       };
    
       void CameraRTSPServer::run()
       {
          TaskScheduler    *scheduler;
          UsageEnvironment *env;
          char stream_name[] = "your name";
    
          scheduler = BasicTaskScheduler::createNew();
          env = BasicUsageEnvironment::createNew(*scheduler);
    
          UserAuthenticationDatabase* authDB = nullptr;
    
          // if (m_Enable_Pass){
          // 	authDB = new UserAuthenticationDatabase;
          // 	authDB->addUserRecord(UserN, PassW);
          // }
    
          OutPacketBuffer::increaseMaxSizeTo(5242880);  // 1M
          RTSPServer* rtspServer = RTSPServer::createNew(*env, fPortNum, authDB);
    
          if (rtspServer == nullptr)
          {
             *env << "LIVE555: Failed to create RTSP server: %s\n", env->getResultMsg();
          }
          else {
             if (fHttpTunnelingPort)
             {
                rtspServer->setUpTunnelingOverHTTP(fHttpTunnelingPort);
             }
    
             char const* descriptionString = "Combined Multiple Cameras Streaming Session";
    
             CameraFrameSource* source = CameraFrameSource::createNew(*env, fH264Encoder);
             StreamReplicator* inputDevice = StreamReplicator::createNew(*env, source, false);
    
             ServerMediaSession* sms = ServerMediaSession::createNew(*env, stream_name, stream_name, descriptionString); 
             CameraMediaSubsession* sub = CameraMediaSubsession::createNew(*env, inputDevice);
    
             sub->set_PPS_NAL(fH264Encoder->PPS_NAL(), fH264Encoder->PPS_NAL_size());
             sub->set_SPS_NAL(fH264Encoder->SPS_NAL(), fH264Encoder->SPS_NAL_size());
             if (fH264Encoder->h264_bit_rate() > 102400) {
                sub->set_bit_rate(fH264Encoder->h264_bit_rate());
             }
             else {
                sub->set_bit_rate(static_cast<uint64_t>(fH264Encoder->pict_width() * fH264Encoder->pict_height() * fH264Encoder->frame_rate() / 20));
             }
             
             sms->addSubsession(sub);
             rtspServer->addServerMediaSession(sms);
    
             char* url = rtspServer->rtspURL(sms);
             *env << "Play this stream using the URL \"" << url << "\"\n";
             delete[] url;
    
             //signal(SIGNIT,sighandler);
             env->taskScheduler().doEventLoop(&fQuit); // does not return
    
             Medium::close(rtspServer);
             Medium::close(inputDevice);
          }
    
          env->reclaim();
          delete scheduler;
       };
    }

### 2、推流到其他服务器 ###

2.1.OPTIONS会话
        OPTIONS rtsp://x.x.x.x.:445/live/1001 RTSP/1.0
        User-Agent: youragentname
        CSeq: 1
    2.2 ANNOUNCE
        ANNOUNCE rtsp://x.x.x.x:445/live/1001 RTSP/1.0
        User-Agent: youranentname
        CSeq: 2
        Content-Type: application/sdp
        Content-Length: number
    
        v=0
        o=- 1464329709034587 1 IN IP4 127.0.0.1
        s= h264 Transport Stream, streamed by qianbo
        m= video 0 RTP/AVP 96
        c=IN IP4 0.0.0.0
        a=control:control:streamid=0
        m=audio 0 RTP/AVP 97
    	a=rtpmap:97 MPEG4-GENERIC/44100/1
    	a=fmtp:97 profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3; 				   					config=1208
    	a=control:streamid=1
    2.3 SETUP tcp
        SETUP rtsp://x.x.x.x:445/live/trackid1 RTSP/1.0
        Transport: RTP/AVP/TCP;unicast;interleaved=0-1;mode=record
        User-Agent: youragentname
        CSeq: 3
    2.4 RECORD
        RECORD rtsp://x.x.x.x:445/live RTSP/1.0
        Session: 0x11111
        User-Agent: youragentname
        CSeq: 4
    2.5 TEARDOWN
        TEARDOWN rtsp://x.x.x.x:445/live RTSP/1.0
        Session: 0x11111
        User-Agent: youragentname
        CSeq: 4

这个方式并不是使用rtsp server 来接收下级的流，而是使用live555 来push到上级server，主代码如下所示

#include "liveMedia.hh"
    #include "BasicUsageEnvironment.hh"
    
    UsageEnvironment* env;
    
    // To make the second and subsequent client for each stream reuse the same
    // True：The client that is started later always starts playing from the position that the first client has already played
    // False：Every client plays video files from the beginning
    Boolean reuseFirstSource = False;
    
    // This function prints relevant information
    static void announceStream(RTSPServer* rtspServer, ServerMediaSession* sms, char const* streamName);
    
    int main(int argc, char** argv) {
    
        // 1. Create a task scheduler and initialize the use environment
        TaskScheduler* scheduler = BasicTaskScheduler::createNew();
    
        // 2. Create an interactive environment
        env = BasicUsageEnvironment::createNew(*scheduler);
        // The following is the code for permission control. After setting, the client without permission cannot connect
        UserAuthenticationDatabase* authDB = NULL;
    
        authDB = new UserAuthenticationDatabase;
        authDB->addUserRecord("admin", "Suntek123"); // replace these with real strings  
    
    
        // 3. Create an RTSP server and start monitoring the connection of the module client. 
        // Note that if the port here is not the default port 554, you must specify the port number when accessing the URL:
        RTSPServer* rtspServer = RTSPServer::createNew(*env, 554, authDB);
        if (rtspServer == NULL) {
            *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
            exit(1);
        }
    
        char const* descriptionString = "Session streamed by \"h264LiveMediaServer\"";
    
        // A H.264 video elementary stream:
        {
            char const* streamName = NULL; // Stream name, media name
    
        // 4. Create a media session 
            // When the client orders, the stream name streamName must be input to tell the RTSP server which stream is ordering.
            // Media session manages session-related information such as session description, session duration, stream name, etc.
            // When running h264LiveMediaServer through the IDE, live555 pushes the video or audio in the project working directory. The working directory is the directory at the same level as *.vcxproj,
            //第二个参数:媒体名、三:媒体信息、四:媒体描述
            ServerMediaSession* sms = ServerMediaSession::createNew(*env, streamName, streamName, descriptionString);
    
        //5. Add 264 sub-session. The file name here is the name of the file that is actually opened.
            sms->addSubsession(H264LiveVideoServerMediaSubsession::createNew(*env, reuseFirstSource));
        //6. Add session for rtspserver, but streamName = NULL
            rtspServer->addServerMediaSession(sms);
            announceStream(rtspServer, sms, streamName);
        }
    
        // Trying to create an HTTP server for the RTSP-over-HTTP channel.
        if (rtspServer->setUpTunnelingOverHTTP(80) || rtspServer->setUpTunnelingOverHTTP(8000) || rtspServer->setUpTunnelingOverHTTP(8080)) {
            *env << "\n(We use port " << rtspServer->httpServerPortNum() << " for optional RTSP-over-HTTP tunneling.)\n";
        }
        else {
            *env << "\n(RTSP-over-HTTP tunneling is not available.)\n";
        }
    
       // Entering the event loop, the read event of the socket and the delayed sending of the media file are all completed in this loop.
        env->taskScheduler().doEventLoop(); // does not return
    
        return 0; // only to prevent compiler warning
    }
    
    static void announceStream(RTSPServer* rtspServer, ServerMediaSession* sms, char const* streamName) {
        char* url = rtspServer->rtspURL(sms);
        UsageEnvironment& env = rtspServer->envir();
        env << "\nstreamName: " << streamName << "\n";
        env << "Play using the URL \"" << url << "\"\n";
        delete[] url;
    }