[Live-devel] Question regarding streaming application

[kriti singhal]

Hello sir,

i made an streamer using your libraries which streams the live video from
camera
int initLm555Settings(void)
{
scheduler = BasicTaskScheduler::createNew();
  env = BasicUsageEnvironment::createNew(*scheduler);
destinationAddressStr

#ifdef USE_SSM

= "232.255.42.42";

#else

= StreamingIp;
#endif
 const unsigned short rtpPortNum = 18888;
const unsigned short rtcpPortNum = rtpPortNum+1;
  const unsigned char ttl = 7;
struct in_addr destinationAddress;

 destinationAddress.s_addr = our_inet_addr(destinationAddressStr);

const Port rtpPort(rtpPortNum);
  const Port rtcpPort(rtcpPortNum);


  Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl);
Groupsock rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl);

#ifdef USE_SSM

 rtpGroupsock.multicastSendOnly();

 rtcpGroupsock.multicastSendOnly();

#endif
g_ExitEventLoop = 0;
ideoSink =
    SimpleRTPSink::createNew(*env, &rtpGroupsock, 33, 90000, "video",
"MP2T",
                 1, True, False /*no 'M' bit*/);
 setSendBufferTo(*env, rtpGroupsock.socketNum(), 1024 * 1024);

  // Create (and start) a 'RTCP instance' for this RTP sink:
  const unsigned estimatedSessionBandwidth = 5000; // in kbps; for RTCP b/w
share
  const unsigned maxCNAMElen = 100;
  unsigned char CNAME[maxCNAMElen+1];
  gethostname((char*)CNAME, maxCNAMElen);
  CNAME[maxCNAMElen] = '\0'; // just in case

  RTCPInstance* rtcp =
    RTCPInstance::createNew(*env, &rtcpGroupsock,
                estimatedSessionBandwidth, CNAME,
                videoSink, NULL /* we're a server */, isSSM);

 UserAuthenticationDatabase* authDB = NULL;
      portNumBits rtspServerPortNum = 554;
      unsigned reclamationTestSeconds=65U;
rtspServer = RTSPServer::createNew(*env,rtspServerPortNum, authDB,
reclamationTestSeconds);
    if (rtspServer == NULL)
    {
    *env << "Failed to create RTSP server: " <<env->getResultMsg()<<"\n";

        rtspServerPortNum = 8554;
        rtspServer = RTSPServer::createNew(*env,rtspServerPortNum);
        if (rtspServer == NULL)
        {
                     return 0;
        }
        else
        {

            *env << "Created RTSP server.."<<"\n.";
        }
    }
    else
    {
        *env << "Created RTSP server.."<<"\n.";

    Boolean const inputStreamIsRawUDP = False;
char const* descriptionString={"Session streamed by \"testOnDemandRT\""};
sms= ServerMediaSession::createNew(*env, streamName,
streamName,descriptionString);
sms->addSubsession(MPEG2TransportUDPServerMediaSubsession::createNew(*env,destinationAddressStr,rtpPortNum1,inputStreamIsRawUDP));
    rtspServer->addServerMediaSession(sms);

    char* url = rtspServer->rtspURL(sms);
*env << "Play this stream using the URL \"" << url << "\"\n";
    delete[] url;
if (rtspServer->setUpTunnelingOverHTTP(sport) ||
rtspServer->setUpTunnelingOverHTTP(sport) ||
rtspServer->setUpTunnelingOverHTTP(sport))
        {
cout<<"\n\n\n(We use port "<<rtspServer->httpServerPortNum()<<" for
optional RTSP-over-HTTP tunneling.)\n";

        }
        else
        {

            cout<<"\n\n\n(RTSP-over-HTTP tunneling is not available.)";
        }
  play();
  env->taskScheduler().doEventLoop(&g_ExitEventLoop);
 if(rtspServer)
    Medium::close(rtspServer);
   if(rtcp)
  Medium::close(rtcp);

   if(videoSink)
  Medium::close(videoSink);
   if(fileSource)
  Medium::close(fileSource);
 rtpGroupsock.removeAllDestinations();
  rtcpGroupsock.removeAllDestinations();

  env->reclaim();

  delete scheduler;

    return 0; // only to prevent compiler warning
}
void afterPlaying(void* /*clientData*/) {
  *env << "...done reading from file\n";

  videoSink->stopPlaying();
  // Note that this also closes the input file that this source read from.
  Medium::close(videoSource);


  // Start playing once again:
  play();
}

//================================================================
// play(): Play the input source.
//=================================================================
void play() {
  // Open the input file as a 'byte-stream file source':

  fi_params.nFICardFrameSize = TRANSPORT_PACKETS_PER_NETWORK_PACKET *
TRANSPORT_PACKET_SIZE;
  fi_params.pfnGetRTPPayload = GetRTPPayload;
  fi_params.socketNum = videoSink->groupsockBeingUsed().socketNum();

  DeviceParameters temp;

    fileSource = DeviceSourceFICard::createNew(*env, fi_params, temp);
  if (fileSource == NULL) {
    *env << "Unable to open Foresight card as a byte-stream file source\n";
    exit(1);
  }
  FramedSource* videoES = fileSource;

  // Create a framer for the Video Elementary Stream:

  videoSource = MPEG1or2VideoStreamDiscreteFramer::createNew(*env,
videoES);//original


  // Finally, start playing:
  *env << "Beginning to read from file...\n";
  videoSink->startPlaying(*videoSource, afterPlaying, videoSink);

//    env->taskScheduler().scheduleDelayedTask(uSecsToDelay,
(TaskFunc*)periodicbrMeasurement1, videoSink);
}


void StartRTPProcess(void)
{

    g_hRtpComThread = CreateThread((LPSECURITY_ATTRIBUTES) NULL, 0,
        (LPTHREAD_START_ROUTINE)initLm555Settings, 0, 0,
&g_dwRtpComThreadID);

    if(g_hRtpComThread) SetThreadPriority(g_hRtpComThread,
THREAD_PRIORITY_LOWEST/*THREAD_PRIORITY_NORMAL*/);

}
int StopRTProcess(void)
{

    try{

          if( videoSource )
    videoSource->stopGettingFrames();


 *env <<"in  StopRTProcess\n";
  Sleep(500);

 Medium::close(rtspServer);
  g_ExitEventLoop = 1;
     g_ExitEventLoop = 0;
  g_hRtpComThread = 0;
  g_dwRtpComThreadID = 0;

  return 0;
}

The streaming done by above streamer is catched by proxy server to which i
give the url given by streamer at the line "char* url".
this streaming is then seen by the client by using the proxy server
URL,when the client say to stop stream the streamer calls its method
StopRTProcess(void) but i got stuck in the line "
Medium::close(rtspServer);",can you please tell why?
I know i have modified your code but still need some of your help


Thanks
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