[Live-devel] Adding Audio Stream Session

[Ashfaque]

Hi Ross, 

Been using Live555 for Video streaming on multiple platforms, used on Linux, iOS and now ported to WinCE7. Video streaming works like a charm.
Now I want to add the audio stream in the media session. As a test I want stream a mp3 file with video stream (H.264 video data generated by a encoder source). 
I am assuming to use MP3AudioFileServerMediaSubsession as audio subsession and add it to ServerMediaSession.
Below is my code:
  const Port rtpPort(rtpPortNum);
  const Port rtcpPort(rtcpPortNum);

  Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl);
  rtpGroupsock.multicastSendOnly(); // we're a SSM source
  Groupsock rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl);
  rtcpGroupsock.multicastSendOnly(); // we're a SSM source

  // Create a 'H264 Video RTP' sink from the RTP 'groupsock':
  OutPacketBuffer::maxSize = 500000;
  videoSink = H264VideoRTPSink::createNew(*env, &rtpGroupsock, 96);

  // Create (and start) a 'RTCP instance' for this RTP sink:
  const unsigned estimatedSessionBandwidth = 500; // 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 */,
                True /* we're a SSM source */);
  // Note: This starts RTCP running automatically

  RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554);
  if (rtspServer == NULL) {
    *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
    exit(1);
  }
  ServerMediaSession* sms
    = ServerMediaSession::createNew(*env, "testStream", inputFileName,
           "Session streamed by \"testH264VideoStreamer\"",
                       True /*SSM*/);

   ServerMediaSubsession* VideoSession = NULL;
  VideoSession = PassiveServerMediaSubsession::createNew(*videoSink, rtcp);
  sms->addSubsession(VideoSession);

  //rtspServer->addServerMediaSession(sms);

  ServerMediaSubsession* AudioSession = NULL;
 
  AudioSession = MP3AudioFileServerMediaSubsession
               ::createNew(*env, inputaudioFileName, reuseFirstSource,
                   true, NULL);
  sms->addSubsession(AudioSession);

  rtspServer->addServerMediaSession(sms);

  char* url = rtspServer->rtspURL(sms);

Please let me know whether this is the correct method of adding audio stream with video. I don’t have a audio codec source, hence want to test with MP3 file only.

Thanks in advance.
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