testProgs/playCommon.cpp

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/**********
This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 2.1 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)

This library is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
more details.

You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
**********/
// Copyright (c) 1996-2008, Live Networks, Inc.  All rights reserved
// A common framework, used for the "openRTSP" and "playSIP" applications
// Implementation

#include "playCommon.hh"
#include "BasicUsageEnvironment.hh"
#include "GroupsockHelper.hh"
#ifdef SUPPORT_REAL_RTSP
#include "../RealRTSP/include/RealRTSP.hh"
#endif

#if defined(__WIN32__) || defined(_WIN32)
#define snprintf _snprintf
#else
#include <signal.h>
#define USE_SIGNALS 1
#endif

// Forward function definitions:
void setupStreams();
void startPlayingStreams();
void tearDownStreams();
void closeMediaSinks();
void subsessionAfterPlaying(void* clientData);
void subsessionByeHandler(void* clientData);
void sessionAfterPlaying(void* clientData = NULL);
void sessionTimerHandler(void* clientData);
void shutdown(int exitCode = 1);
void signalHandlerShutdown(int sig);
void checkForPacketArrival(void* clientData);
void checkInterPacketGaps(void* clientData);
void beginQOSMeasurement();

char const* progName;
UsageEnvironment* env;
Medium* ourClient = NULL;
MediaSession* session = NULL;
TaskToken sessionTimerTask = NULL;
TaskToken arrivalCheckTimerTask = NULL;
TaskToken interPacketGapCheckTimerTask = NULL;
TaskToken qosMeasurementTimerTask = NULL;
Boolean createReceivers = True;
Boolean outputQuickTimeFile = False;
Boolean generateMP4Format = False;
QuickTimeFileSink* qtOut = NULL;
Boolean outputAVIFile = False;
AVIFileSink* aviOut = NULL;
Boolean audioOnly = False;
Boolean videoOnly = False;
char const* singleMedium = NULL;
int verbosityLevel = 1; // by default, print verbose output
double duration = 0;
double durationSlop = -1.0; // extra seconds to play at the end
double initialSeekTime = 0.0f;
double scale = 1.0f;
unsigned interPacketGapMaxTime = 0;
unsigned totNumPacketsReceived = ~0; // used if checking inter-packet gaps
Boolean playContinuously = False;
int simpleRTPoffsetArg = -1;
Boolean sendOptionsRequest = True;
Boolean sendOptionsRequestOnly = False;
Boolean oneFilePerFrame = False;
Boolean notifyOnPacketArrival = False;
Boolean streamUsingTCP = False;
portNumBits tunnelOverHTTPPortNum = 0;
char* username = NULL;
char* password = NULL;
char* proxyServerName = NULL;
unsigned short proxyServerPortNum = 0;
unsigned char desiredAudioRTPPayloadFormat = 0;
char* mimeSubtype = NULL;
unsigned short movieWidth = 240; // default
Boolean movieWidthOptionSet = False;
unsigned short movieHeight = 180; // default
Boolean movieHeightOptionSet = False;
unsigned movieFPS = 15; // default
Boolean movieFPSOptionSet = False;
char* fileNamePrefix = "";
unsigned fileSinkBufferSize = 20000;
unsigned socketInputBufferSize = 0;
Boolean packetLossCompensate = False;
Boolean syncStreams = False;
Boolean generateHintTracks = False;
unsigned qosMeasurementIntervalMS = 0; // 0 means: Don't output QOS data
unsigned statusCode = 0;

struct timeval startTime;

void usage() {
  *env << "Usage: " << progName
       << " [-p <startPortNum>] [-r|-q|-4|-i] [-a|-v] [-V] [-d <duration>] [-D <max-inter-packet-gap-time> [-c] [-S <offset>] [-n] [-O]"
           << (controlConnectionUsesTCP ? " [-t|-T <http-port>]" : "")
       << " [-u <username> <password>"
           << (allowProxyServers ? " [<proxy-server> [<proxy-server-port>]]" : "")
       << "]" << (supportCodecSelection ? " [-A <audio-codec-rtp-payload-format-code>|-M <mime-subtype-name>]" : "")
       << " [-s <initial-seek-time>] [-z <scale>]"
       << " [-w <width> -h <height>] [-f <frames-per-second>] [-y] [-H] [-Q [<measurement-interval>]] [-F <filename-prefix>] [-b <file-sink-buffer-size>] [-B <input-socket-buffer-size>] [-I <input-interface-ip-address>] [-m] <url> (or " << progName << " -o [-V] <url>)\n";
  //##### Add "-R <dest-rtsp-url>" #####
  shutdown();
}

int main(int argc, char** argv) {
  // Begin by setting up our usage environment:
  TaskScheduler* scheduler = BasicTaskScheduler::createNew();
  env = BasicUsageEnvironment::createNew(*scheduler);

  progName = argv[0];

  gettimeofday(&startTime, NULL);

#ifdef USE_SIGNALS
  // Allow ourselves to be shut down gracefully by a SIGHUP or a SIGUSR1:
  signal(SIGHUP, signalHandlerShutdown);
  signal(SIGUSR1, signalHandlerShutdown);
#endif

  unsigned short desiredPortNum = 0;

  // unfortunately we can't use getopt() here, as Windoze doesn't have it
  while (argc > 2) {
    char* const opt = argv[1];
    if (opt[0] != '-') usage();
    switch (opt[1]) {

    case 'p': { // specify start port number
      int portArg;
      if (sscanf(argv[2], "%d", &portArg) != 1) {
        usage();
      }
      if (portArg <= 0 || portArg >= 65536 || portArg&1) {
        *env << "bad port number: " << portArg
                << " (must be even, and in the range (0,65536))\n";
        usage();
      }
      desiredPortNum = (unsigned short)portArg;
      ++argv; --argc;
      break;
    }

    case 'r': { // do not receive data (instead, just 'play' the stream(s))
      createReceivers = False;
      break;
    }

    case 'q': { // output a QuickTime file (to stdout)
      outputQuickTimeFile = True;
      break;
    }

    case '4': { // output a 'mp4'-format file (to stdout)
      outputQuickTimeFile = True;
      generateMP4Format = True;
      break;
    }

    case 'i': { // output an AVI file (to stdout)
      outputAVIFile = True;
      break;
    }

    case 'I': { // specify input interface... 
      NetAddressList addresses(argv[2]);
      if (addresses.numAddresses() == 0) {
        *env << "Failed to find network address for \"" << argv[2] << "\"";
        break;
      }
      ReceivingInterfaceAddr = *(unsigned*)(addresses.firstAddress()->data());
      ++argv; --argc;
      break;
    }

    case 'a': { // receive/record an audio stream only
      audioOnly = True;
      singleMedium = "audio";
      break;
    }

    case 'v': { // receive/record a video stream only
      videoOnly = True;
      singleMedium = "video";
      break;
    }

    case 'V': { // disable verbose output
      verbosityLevel = 0;
      break;
    }

    case 'd': { // specify duration, or how much to delay after end time
      float arg;
      if (sscanf(argv[2], "%g", &arg) != 1) {
        usage();
      }
      if (argv[2][0] == '-') { // not "arg<0", in case argv[2] was "-0"
        // a 'negative' argument was specified; use this for "durationSlop":
        duration = 0; // use whatever's in the SDP
        durationSlop = -arg;
      } else {
        duration = arg;
        durationSlop = 0;
      }
      ++argv; --argc;
      break;
    }

    case 'D': { // specify maximum number of seconds to wait for packets:
      if (sscanf(argv[2], "%u", &interPacketGapMaxTime) != 1) {
        usage();
      }
      ++argv; --argc;
      break;
    }

    case 'c': { // play continuously
      playContinuously = True;
      break;
    }

    case 'S': { // specify an offset to use with "SimpleRTPSource"s
      if (sscanf(argv[2], "%d", &simpleRTPoffsetArg) != 1) {
        usage();
      }
      if (simpleRTPoffsetArg < 0) {
        *env << "offset argument to \"-S\" must be >= 0\n";
        usage();
      }
      ++argv; --argc;
      break;
    }

    case 'O': { // Don't send an "OPTIONS" request before "DESCRIBE"
      sendOptionsRequest = False;
      break;
    }

    case 'o': { // Send only the "OPTIONS" request to the server
      sendOptionsRequestOnly = True;
      break;
    }

    case 'm': { // output multiple files - one for each frame
      oneFilePerFrame = True;
      break;
    }

    case 'n': { // notify the user when the first data packet arrives
      notifyOnPacketArrival = True;
      break;
    }

    case 't': {
      // stream RTP and RTCP over the TCP 'control' connection
      if (controlConnectionUsesTCP) {
        streamUsingTCP = True;
      } else {
        usage();
      }
      break;
    }

    case 'T': {
      // stream RTP and RTCP over a HTTP connection
      if (controlConnectionUsesTCP) {
        if (argc > 3 && argv[2][0] != '-') {
          // The next argument is the HTTP server port number:
          if (sscanf(argv[2], "%hu", &tunnelOverHTTPPortNum) == 1
              && tunnelOverHTTPPortNum > 0) {
            ++argv; --argc;
            break;
          }
        }
      }

      // If we get here, the option was specified incorrectly:
      usage();
      break;
    }

    case 'u': { // specify a username and password
      username = argv[2];
      password = argv[3];
      argv+=2; argc-=2;
      if (allowProxyServers && argc > 3 && argv[2][0] != '-') {
        // The next argument is the name of a proxy server:
        proxyServerName = argv[2];
        ++argv; --argc;

        if (argc > 3 && argv[2][0] != '-') {
          // The next argument is the proxy server port number:
          if (sscanf(argv[2], "%hu", &proxyServerPortNum) != 1) {
            usage();
          }
          ++argv; --argc;
        }
      }
      break;
    }

    case 'A': { // specify a desired audio RTP payload format
      unsigned formatArg;
      if (sscanf(argv[2], "%u", &formatArg) != 1
          || formatArg >= 96) {
        usage();
      }
      desiredAudioRTPPayloadFormat = (unsigned char)formatArg;
      ++argv; --argc;
      break;
    }

    case 'M': { // specify a MIME subtype for a dynamic RTP payload type
      mimeSubtype = argv[2];
      if (desiredAudioRTPPayloadFormat==0) desiredAudioRTPPayloadFormat =96;
      ++argv; --argc;
      break;
    }

    case 'w': { // specify a width (pixels) for an output QuickTime or AVI movie
      if (sscanf(argv[2], "%hu", &movieWidth) != 1) {
        usage();
      }
      movieWidthOptionSet = True;
      ++argv; --argc;
      break;
    }

    case 'h': { // specify a height (pixels) for an output QuickTime or AVI movie
      if (sscanf(argv[2], "%hu", &movieHeight) != 1) {
        usage();
      }
      movieHeightOptionSet = True;
      ++argv; --argc;
      break;
    }

    case 'f': { // specify a frame rate (per second) for an output QT or AVI movie
      if (sscanf(argv[2], "%u", &movieFPS) != 1) {
        usage();
      }
      movieFPSOptionSet = True;
      ++argv; --argc;
      break;
    }

    case 'F': { // specify a prefix for the audio and video output files
      fileNamePrefix = argv[2];
      ++argv; --argc;
      break;
    }

    case 'b': { // specify the size of buffers for "FileSink"s
      if (sscanf(argv[2], "%u", &fileSinkBufferSize) != 1) {
        usage();
      }
      ++argv; --argc;
      break;
    }

    case 'B': { // specify the size of input socket buffers
      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 's': { // specify initial seek time (trick play)
      float arg;
      if (sscanf(argv[2], "%g", &arg) != 1 || arg < 0) {
        usage();
      }
      initialSeekTime = arg;
      ++argv; --argc;
      break;
    }

    case 'z': { // scale (trick play)
      float arg;
      if (sscanf(argv[2], "%g", &arg) != 1 || arg == 0.0f) {
        usage();
      }
      scale = arg;
      ++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 (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 (durationSlop < 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.
    durationSlop = 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, username, password);
    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 {
      // 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 if (strcmp(subsession->mediumName(), "video") == 0 &&
            (strcmp(subsession->codecName(), "H264") == 0)) {
      // For H.264 video stream, we use a special sink that insert start_codes:
          fileSink = H264VideoFileSink::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;
          }

          subsession->sink->startPlaying(*(subsession->readSource()),
                                         subsessionAfterPlaying,
                                         subsession);
          
          // Also set a handler to be called if a RTCP "BYE" arrives
          // for this subsession:
          if (subsession->rtcpInstance() != NULL) {
            subsession->rtcpInstance()->setByeHandler(subsessionByeHandler,
                                                      subsession);
          }

          madeProgress = True;
        }
      }
      if (!madeProgress) shutdown();
    }
  }
    
  // Finally, start playing each subsession, to start the data flow:

  startPlayingStreams();

  env->taskScheduler().doEventLoop(); // does not return

  return 0; // only to prevent compiler warning
}


void setupStreams() {
  MediaSubsessionIterator iter(*session);
  MediaSubsession *subsession;
  Boolean madeProgress = False;

  while ((subsession = iter.next()) != NULL) {
    if (subsession->clientPortNum() == 0) continue; // port # was not set

    if (!clientSetupSubsession(ourClient, subsession, streamUsingTCP)) {
      *env << "Failed to setup \"" << subsession->mediumName()
                << "/" << subsession->codecName()
                << "\" subsession: " << env->getResultMsg() << "\n";
    } else {
      *env << "Setup \"" << subsession->mediumName()
                << "/" << subsession->codecName()
                << "\" subsession (client ports " << subsession->clientPortNum()
                << "-" << subsession->clientPortNum()+1 << ")\n";
      madeProgress = True;
    }
  }
  if (!madeProgress) shutdown();
}

void startPlayingStreams() {
  if (duration == 0) {
    if (scale > 0) duration = session->playEndTime() - initialSeekTime; // use SDP end time
    else if (scale < 0) duration = initialSeekTime;
  }
  if (duration < 0) duration = 0.0;

  if (!clientStartPlayingSession(ourClient, session)) {
    *env << "Failed to start playing session: " << env->getResultMsg() << "\n";
    shutdown();
  } else {
    *env << "Started playing session\n";
  }

  if (qosMeasurementIntervalMS > 0) {
    // Begin periodic QOS measurements:
    beginQOSMeasurement();
  }

  // Figure out how long to delay (if at all) before shutting down, or
  // repeating the playing
  Boolean timerIsBeingUsed = False;
  double secondsToDelay = duration;
  if (duration > 0) {
    double const maxDelayTime
      = (double)( ((unsigned)0x7FFFFFFF)/1000000.0 );
    if (