playcommon.cpp

流媒体传输协议的实现代码,非常有用.可以支持rtsp mms等流媒体传输协议

      if (sscanf(argv[2], "%u", &socketInputBufferSize) != 1) {
	usage();
      }
      ++argv; --argc;
      break;
    }

    // Note: The following option is deprecated, and may someday be removed:
    case 'l': { // try to compensate for packet loss by repeating frames
      packetLossCompensate = True;
      break;
    }

    case 'y': { // synchronize audio and video streams
      syncStreams = True;
      break;
    }

    case 'H': { // generate hint tracks (as well as the regular data tracks)
      generateHintTracks = True;
      break;
    }

    case 'Q': { // output QOS measurements
      qosMeasurementIntervalMS = 1000; // default: 1 second

      if (argc > 3 && argv[2][0] != '-') {
	// The next argument is the measurement interval,
	// in multiples of 100 ms
	if (sscanf(argv[2], "%u", &qosMeasurementIntervalMS) != 1) {
	  usage();
	}
	qosMeasurementIntervalMS *= 100;
	++argv; --argc;
      }
      break;
    }

    case 'R': { // inject received data into a RTSP server
      destRTSPURL = argv[2];
      ++argv; --argc;
      break;
    }

    default: {
      usage();
      break;
    }
    }

    ++argv; --argc;
  }
  if (argc != 2) usage();
  if (outputQuickTimeFile && outputAVIFile) {
    *env << "The -i and -q (or -4) flags cannot both be used!\n";
    usage();
  }
  Boolean outputCompositeFile = outputQuickTimeFile || outputAVIFile;
  if (!createReceivers && outputCompositeFile) {
    *env << "The -r and -q (or -4 or -i) flags cannot both be used!\n";
    usage();
  }
  if (destRTSPURL != NULL && (!createReceivers || outputCompositeFile)) {
    *env << "The -R flag cannot be used with -r, -q, or -i!\n";
    usage();
  }
  if (outputCompositeFile && !movieWidthOptionSet) {
    *env << "Warning: The -q, -4 or -i option was used, but not -w.  Assuming a video width of "
	 << movieWidth << " pixels\n";
  }
  if (outputCompositeFile && !movieHeightOptionSet) {
    *env << "Warning: The -q, -4 or -i option was used, but not -h.  Assuming a video height of "
	 << movieHeight << " pixels\n";
  }
  if (outputCompositeFile && !movieFPSOptionSet) {
    *env << "Warning: The -q, -4 or -i option was used, but not -f.  Assuming a video frame rate of "
	 << movieFPS << " frames-per-second\n";
  }
  if (audioOnly && videoOnly) {
    *env << "The -a and -v flags cannot both be used!\n";
    usage();
  }
  if (sendOptionsRequestOnly && !sendOptionsRequest) {
    *env << "The -o and -O flags cannot both be used!\n";
    usage();
  }
  if (tunnelOverHTTPPortNum > 0) {
    if (streamUsingTCP) {
      *env << "The -t and -T flags cannot both be used!\n";
      usage();
    } else {
      streamUsingTCP = True;
    }
  }
  if (!createReceivers && notifyOnPacketArrival) {
    *env << "Warning: Because we're not receiving stream data, the -n flag has no effect\n";
  }
  if (endTimeSlop < 0) {
    // This parameter wasn't set, so use a default value.
    // If we're measuring QOS stats, then don't add any slop, to avoid
    // having 'empty' measurement intervals at the end.
    endTimeSlop = qosMeasurementIntervalMS > 0 ? 0.0 : 5.0;
  }

  char* url = argv[1];

  // Create our client object:
  ourClient = createClient(*env, verbosityLevel, progName);
  if (ourClient == NULL) {
    *env << "Failed to create " << clientProtocolName
		<< " client: " << env->getResultMsg() << "\n";
    shutdown();
  }

  if (sendOptionsRequest) {
    // Begin by sending an "OPTIONS" command:
    char* optionsResponse = getOptionsResponse(ourClient, url);
    if (sendOptionsRequestOnly) {
      if (optionsResponse == NULL) {
	*env << clientProtocolName << " \"OPTIONS\" request failed: "
	     << env->getResultMsg() << "\n";
      } else {
	*env << clientProtocolName << " \"OPTIONS\" request returned: "
	     << optionsResponse << "\n";
      }
      shutdown();
    }
    delete[] optionsResponse;
  }

  // Open the URL, to get a SDP description:
  char* sdpDescription
    = getSDPDescriptionFromURL(ourClient, url, username, password,
			       proxyServerName, proxyServerPortNum,
			       desiredPortNum);
  if (sdpDescription == NULL) {
    *env << "Failed to get a SDP description from URL \"" << url
		<< "\": " << env->getResultMsg() << "\n";
    shutdown();
  }

  *env << "Opened URL \"" << url
	  << "\", returning a SDP description:\n" << sdpDescription << "\n";

  // Create a media session object from this SDP description:
  session = MediaSession::createNew(*env, sdpDescription);
  delete[] sdpDescription;
  if (session == NULL) {
    *env << "Failed to create a MediaSession object from the SDP description: " << env->getResultMsg() << "\n";
    shutdown();
  } else if (!session->hasSubsessions()) {
    *env << "This session has no media subsessions (i.e., \"m=\" lines)\n";
    shutdown();
  }

  // Then, setup the "RTPSource"s for the session:
  MediaSubsessionIterator iter(*session);
  MediaSubsession *subsession;
  Boolean madeProgress = False;
  char const* singleMediumToTest = singleMedium;
  while ((subsession = iter.next()) != NULL) {
    // If we've asked to receive only a single medium, then check this now:
    if (singleMediumToTest != NULL) {
      if (strcmp(subsession->mediumName(), singleMediumToTest) != 0) {
		  *env << "Ignoring \"" << subsession->mediumName()
			  << "/" << subsession->codecName()
			  << "\" subsession, because we've asked to receive a single " << singleMedium
			  << " session only\n";
	continue;
      } else {
	// Receive this subsession only
	singleMediumToTest = "xxxxx";
	    // this hack ensures that we get only 1 subsession of this type
      }
    }

    if (desiredPortNum != 0) {
      subsession->setClientPortNum(desiredPortNum);
      desiredPortNum += 2;
    }

    if (createReceivers) {
      if (!subsession->initiate(simpleRTPoffsetArg)) {
		*env << "Unable to create receiver for \"" << subsession->mediumName()
			<< "/" << subsession->codecName()
			<< "\" subsession: " << env->getResultMsg() << "\n";
      } else {
		*env << "Created receiver for \"" << subsession->mediumName()
			<< "/" << subsession->codecName()
			<< "\" subsession (client ports " << subsession->clientPortNum()
			<< "-" << subsession->clientPortNum()+1 << ")\n";
		madeProgress = True;

		if (subsession->rtpSource() != NULL) {
		  // Because we're saving the incoming data, rather than playing
		  // it in real time, allow an especially large time threshold
		  // (1 second) for reordering misordered incoming packets:
		  unsigned const thresh = 1000000; // 1 second 
		  subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);

		  if (socketInputBufferSize > 0) {
		    // Set the RTP source's input buffer size as specified:
		    int socketNum
		      = subsession->rtpSource()->RTPgs()->socketNum();
		    unsigned curBufferSize
		      = getReceiveBufferSize(*env, socketNum);
		    unsigned newBufferSize
		      = setReceiveBufferTo(*env, socketNum, socketInputBufferSize);
		    *env << "Changed socket receive buffer size for the \""
			 << subsession->mediumName()
			 << "/" << subsession->codecName()
			 << "\" subsession from "
			 << curBufferSize << " to "
			 << newBufferSize << " bytes\n";
		  }
		}
      }
    } else {
      if (subsession->clientPortNum() == 0) {
	*env << "No client port was specified for the \""
	     << subsession->mediumName()
	     << "/" << subsession->codecName()
	     << "\" subsession.  (Try adding the \"-p <portNum>\" option.)\n";
      } else {	
		madeProgress = True;
      }
    }
  }
  if (!madeProgress) shutdown();

  // Perform additional 'setup' on each subsession, before playing them:
  setupStreams();

  // Create output files:
  if (createReceivers) {
    if (outputQuickTimeFile) {
      // Create a "QuickTimeFileSink", to write to 'stdout':
      qtOut = QuickTimeFileSink::createNew(*env, *session, "stdout",
					   fileSinkBufferSize,
					   movieWidth, movieHeight,
					   movieFPS,
					   packetLossCompensate,
					   syncStreams,
					   generateHintTracks,
					   generateMP4Format);
      if (qtOut == NULL) {
	*env << "Failed to create QuickTime file sink for stdout: " << env->getResultMsg();
	shutdown();
      }

      qtOut->startPlaying(sessionAfterPlaying, NULL);
    } else if (outputAVIFile) {
      // Create an "AVIFileSink", to write to 'stdout':
      aviOut = AVIFileSink::createNew(*env, *session, "stdout",
				      fileSinkBufferSize,
				      movieWidth, movieHeight,
				      movieFPS,
				      packetLossCompensate);
      if (aviOut == NULL) {
	*env << "Failed to create AVI file sink for stdout: " << env->getResultMsg();
	shutdown();
      }

      aviOut->startPlaying(sessionAfterPlaying, NULL);
#ifdef SUPPORT_REAL_RTSP
    } else if (session->isRealNetworksRDT) {
      // For RealNetworks' sessions, we create a single output file,
      // named "output.rm".
      char outFileName[1000];
      if (singleMedium == NULL) {
	snprintf(outFileName, sizeof outFileName, "%soutput.rm", fileNamePrefix);
      } else {
	// output to 'stdout' as normal, even though we actually output all media
	sprintf(outFileName, "stdout");
      }
      FileSink* fileSink = FileSink::createNew(*env, outFileName,
					       fileSinkBufferSize, oneFilePerFrame);

      // The output file needs to begin with a special 'RMFF' header,
      // in order for it to be usable.  Write this header first:
      unsigned headerSize;
      unsigned char* headerData = RealGenerateRMFFHeader(session, headerSize);
      struct timeval timeNow;
      gettimeofday(&timeNow, NULL);
      fileSink->addData(headerData, headerSize, timeNow);
      delete[] headerData;

      // Start playing the output file from the first subsession.
      // (Hack: Because all subsessions' data is actually multiplexed on the
      // single RTSP TCP connection, playing from one subsession is sufficient.)
      iter.reset();
      madeProgress = False;
      while ((subsession = iter.next()) != NULL) {
	if (subsession->readSource() == NULL) continue; // was not initiated

	  fileSink->startPlaying(*(subsession->readSource()),
					 subsessionAfterPlaying,
					 subsession);
	  madeProgress = True;
	  break; // play from one subsession only
      }
      if (!madeProgress) shutdown();
#endif
    } else if (destRTSPURL != NULL) {
      // Announce the session into a (separate) RTSP server,
      // and create one or more "RTPTranslator"s to tie the source
      // and destination together:
      if (setupDestinationRTSPServer()) {
		  *env << "Set up destination RTSP session for \"" << destRTSPURL << "\"\n";
      } else {
		  *env << "Failed to set up destination RTSP session for \"" << destRTSPURL
			  << "\": " << env->getResultMsg() << "\n";
		shutdown();
      }
    } else {
      // Create and start "FileSink"s for each subsession:
      madeProgress = False;
      iter.reset();
      while ((subsession = iter.next()) != NULL) {
	if (subsession->readSource() == NULL) continue; // was not initiated
	
	// Create an output file for each desired stream:
	char outFileName[1000];
	if (singleMedium == NULL) {
	  // Output file name is
	  //     "<filename-prefix><medium_name>-<codec_name>-<counter>"
	  static unsigned streamCounter = 0;
	  snprintf(outFileName, sizeof outFileName, "%s%s-%s-%d",
		   fileNamePrefix, subsession->mediumName(),
		   subsession->codecName(), ++streamCounter);
	} else {
	  sprintf(outFileName, "stdout");
	}
	FileSink* fileSink;
	if (strcmp(subsession->mediumName(), "audio") == 0 &&
	    (strcmp(subsession->codecName(), "AMR") == 0 ||
	     strcmp(subsession->codecName(), "AMR-WB") == 0)) {
	  // For AMR audio streams, we use a special sink that inserts AMR frame hdrs:
	  fileSink = AMRAudioFileSink::createNew(*env, outFileName,
						 fileSinkBufferSize, oneFilePerFrame);
	} else {
	  // Normal case:
	  fileSink = FileSink::createNew(*env, outFileName,
					 fileSinkBufferSize, oneFilePerFrame);
	}
	subsession->sink = fileSink;
	if (subsession->sink == NULL) {
	  *env << "Failed to create FileSink for \"" << outFileName
		  << "\": " << env->getResultMsg() << "\n";
	} else {
	  if (singleMedium == NULL) {
	    *env << "Created output file: \"" << outFileName << "\"\n";
	  } else {
	    *env << "Outputting data from the \"" << subsession->mediumName()
			<< "/" << subsession->codecName()
			<< "\" subsession to 'stdout'\n";
	  }

	  if (strcmp(subsession->mediumName(), "video") == 0 &&
	      strcmp(subsession->codecName(), "MP4V-ES") == 0 &&
	      subsession->fmtp_config() != NULL) {
	    // For MPEG-4 video RTP streams, the 'config' information
	    // from the SDP description contains useful VOL etc. headers.
	    // Insert this data at the front of the output file:
	    unsigned configLen;
	    unsigned char* configData
	      = parseGeneralConfigStr(subsession->fmtp_config(), configLen);
	    struct timeval timeNow;
	    gettimeofday(&timeNow, NULL);
	    fileSink->addData(configData, configLen, timeNow);
	    delete[] configData;
	  }