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• Live555 : X264 Stream Live source based on "testOnDemandRTSPServer"

  12 janvier 2017, par user2660369

  I am trying to create a rtsp Server that streams the OpenGL output of my program. I had a look at How to write a Live555 FramedSource to allow me to stream H.264 live, but I need the stream to be unicast. So I had a look at testOnDemandRTSPServer. Using the same Code fails. To my understanding I need to provide memory in which I store my h264 frames so the OnDemandServer can read them on Demand.

  H264VideoStreamServerMediaSubsession.cpp

  H264VideoStreamServerMediaSubsession*
  
  H264VideoStreamServerMediaSubsession::createNew(UsageEnvironment& env,
  
                            Boolean reuseFirstSource) {
  
    return new H264VideoStreamServerMediaSubsession(env, reuseFirstSource);
  
  }
  
  
  
  H264VideoStreamServerMediaSubsession::H264VideoStreamServerMediaSubsession(UsageEnvironment& env, Boolean reuseFirstSource)
  
    : OnDemandServerMediaSubsession(env, reuseFirstSource), fAuxSDPLine(NULL), fDoneFlag(0), fDummyRTPSink(NULL) {
  
  }
  
  
  
  H264VideoStreamServerMediaSubsession::~H264VideoStreamServerMediaSubsession() {
  
    delete[] fAuxSDPLine;
  
  }
  
  
  
  static void afterPlayingDummy(void* clientData) {
  
    H264VideoStreamServerMediaSubsession* subsess = (H264VideoStreamServerMediaSubsession*)clientData;
  
    subsess->afterPlayingDummy1();
  
  }
  
  
  
  void H264VideoStreamServerMediaSubsession::afterPlayingDummy1() {
  
    // Unschedule any pending 'checking' task:
  
    envir().taskScheduler().unscheduleDelayedTask(nextTask());
  
    // Signal the event loop that we're done:
  
    setDoneFlag();
  
  }
  
  
  
  static void checkForAuxSDPLine(void* clientData) {
  
    H264VideoStreamServerMediaSubsession* subsess = (H264VideoStreamServerMediaSubsession*)clientData;
  
    subsess->checkForAuxSDPLine1();
  
  }
  
  
  
  void H264VideoStreamServerMediaSubsession::checkForAuxSDPLine1() {
  
    char const* dasl;
  
  
  
    if (fAuxSDPLine != NULL) {
  
      // Signal the event loop that we're done:
  
      setDoneFlag();
  
    } else if (fDummyRTPSink != NULL && (dasl = fDummyRTPSink->auxSDPLine()) != NULL) {
  
      fAuxSDPLine = strDup(dasl);
  
      fDummyRTPSink = NULL;
  
  
  
      // Signal the event loop that we're done:
  
      setDoneFlag();
  
    } else {
  
      // try again after a brief delay:
  
      int uSecsToDelay = 100000; // 100 ms
  
      nextTask() = envir().taskScheduler().scheduleDelayedTask(uSecsToDelay,
  
                    (TaskFunc*)checkForAuxSDPLine, this);
  
    }
  
  }
  
  
  
  char const* H264VideoStreamServerMediaSubsession::getAuxSDPLine(RTPSink* rtpSink, FramedSource* inputSource) {
  
    if (fAuxSDPLine != NULL) return fAuxSDPLine; // it's already been set up (for a previous client)
  
  
  
    if (fDummyRTPSink == NULL) { // we're not already setting it up for another, concurrent stream
  
      // Note: For H264 video files, the 'config' information ("profile-level-id" and "sprop-parameter-sets") isn't known
  
      // until we start reading the file.  This means that "rtpSink"s "auxSDPLine()" will be NULL initially,
  
      // and we need to start reading data from our file until this changes.
  
      fDummyRTPSink = rtpSink;
  
  
  
      // Start reading the file:
  
      fDummyRTPSink->startPlaying(*inputSource, afterPlayingDummy, this);
  
  
  
      // Check whether the sink's 'auxSDPLine()' is ready:
  
      checkForAuxSDPLine(this);
  
    }
  
  
  
    envir().taskScheduler().doEventLoop(&fDoneFlag);
  
  
  
    return fAuxSDPLine;
  
  }
  
  
  
  FramedSource* H264VideoStreamServerMediaSubsession::createNewStreamSource(unsigned /*clientSessionId*/, unsigned& estBitrate) {
  
    estBitrate = 500; // kb
  
    megamol::remotecontrol::View3D_MRC *parent = (megamol::remotecontrol::View3D_MRC*)this->parent;
  
    return H264VideoStreamFramer::createNew(envir(), parent->h264FramedSource);
  
  }
  
  
  
  RTPSink* H264VideoStreamServerMediaSubsession::createNewRTPSink(Groupsock* rtpGroupsock, unsigned char rtpPayloadTypeIfDynamic, FramedSource* /*inputSource*/) {
  
    return H264VideoRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic);
  
  }

  FramedSource.cpp

  H264FramedSource* H264FramedSource::createNew(UsageEnvironment& env,
  
                                            unsigned preferredFrameSize,
  
                                            unsigned playTimePerFrame)
  
  {
  
      return new H264FramedSource(env, preferredFrameSize, playTimePerFrame);
  
  }
  
  
  
  H264FramedSource::H264FramedSource(UsageEnvironment& env,
  
                                 unsigned preferredFrameSize,
  
                                 unsigned playTimePerFrame)
  
      : FramedSource(env),
  
      fPreferredFrameSize(fMaxSize),
  
      fPlayTimePerFrame(playTimePerFrame),
  
      fLastPlayTime(0),
  
      fCurIndex(0)
  
  {
  
  
  
      x264_param_default_preset(&param, "veryfast", "zerolatency");
  
      param.i_threads = 1;
  
      param.i_width = 1024;
  
      param.i_height = 768;
  
      param.i_fps_num = 30;
  
      param.i_fps_den = 1;
  
      // Intra refres:
  
      param.i_keyint_max = 60;
  
      param.b_intra_refresh = 1;
  
      //Rate control:
  
      param.rc.i_rc_method = X264_RC_CRF;
  
      param.rc.f_rf_constant = 25;
  
      param.rc.f_rf_constant_max = 35;
  
      param.i_sps_id = 7;
  
      //For streaming:
  
      param.b_repeat_headers = 1;
  
      param.b_annexb = 1;
  
      x264_param_apply_profile(&param, "baseline");
  
  
  
      param.i_log_level = X264_LOG_ERROR;
  
  
  
      encoder = x264_encoder_open(&param);
  
      pic_in.i_type            = X264_TYPE_AUTO;
  
      pic_in.i_qpplus1         = 0;
  
      pic_in.img.i_csp         = X264_CSP_I420;
  
      pic_in.img.i_plane       = 3;
  
  
  
  
  
      x264_picture_alloc(&pic_in, X264_CSP_I420, 1024, 768);
  
  
  
      convertCtx = sws_getContext(1024, 768, PIX_FMT_RGBA, 1024, 768, PIX_FMT_YUV420P, SWS_FAST_BILINEAR, NULL, NULL, NULL);
  
      eventTriggerId = envir().taskScheduler().createEventTrigger(deliverFrame0);
  
  }
  
  
  
  H264FramedSource::~H264FramedSource()
  
  {
  
      envir().taskScheduler().deleteEventTrigger(eventTriggerId);
  
      eventTriggerId = 0;
  
  }
  
  
  
  void H264FramedSource::AddToBuffer(uint8_t* buf, int surfaceSizeInBytes)
  
  {
  
      uint8_t* surfaceData = (new uint8_t[surfaceSizeInBytes]);
  
  
  
      memcpy(surfaceData, buf, surfaceSizeInBytes);
  
  
  
      int srcstride = 1024*4;
  
      sws_scale(convertCtx, &surfaceData, &srcstride,0, 768, pic_in.img.plane, pic_in.img.i_stride);
  
      x264_nal_t* nals = NULL;
  
      int i_nals = 0;
  
      int frame_size = -1;
  
  
  
  
  
      frame_size = x264_encoder_encode(encoder, &nals, &i_nals, &pic_in, &pic_out);
  
  
  
      static bool finished = false;
  
  
  
      if (frame_size >= 0)
  
      {
  
      static bool alreadydone = false;
  
      if(!alreadydone)
  
      {
  
  
  
          x264_encoder_headers(encoder, &nals, &i_nals);
  
          alreadydone = true;
  
      }
  
      for(int i = 0; i < i_nals; ++i)
  
      {
  
          m_queue.push(nals[i]);
  
      }
  
      }
  
      delete [] surfaceData;
  
      surfaceData = nullptr;
  
  
  
      envir().taskScheduler().triggerEvent(eventTriggerId, this);
  
  }
  
  
  
  void H264FramedSource::doGetNextFrame()
  
  {
  
      deliverFrame();
  
  }
  
  
  
  void H264FramedSource::deliverFrame0(void* clientData)
  
  {
  
      ((H264FramedSource*)clientData)->deliverFrame();
  
  }
  
  
  
  void H264FramedSource::deliverFrame()
  
  {
  
      x264_nal_t nalToDeliver;
  
  
  
      if (fPlayTimePerFrame > 0 && fPreferredFrameSize > 0) {
  
      if (fPresentationTime.tv_sec == 0 && fPresentationTime.tv_usec == 0) {
  
          // This is the first frame, so use the current time:
  
          gettimeofday(&fPresentationTime, NULL);
  
      } else {
  
          // Increment by the play time of the previous data:
  
          unsigned uSeconds   = fPresentationTime.tv_usec + fLastPlayTime;
  
          fPresentationTime.tv_sec += uSeconds/1000000;
  
          fPresentationTime.tv_usec = uSeconds%1000000;
  
      }
  
  
  
      // Remember the play time of this data:
  
      fLastPlayTime = (fPlayTimePerFrame*fFrameSize)/fPreferredFrameSize;
  
      fDurationInMicroseconds = fLastPlayTime;
  
      } else {
  
      // We don't know a specific play time duration for this data,
  
      // so just record the current time as being the 'presentation time':
  
      gettimeofday(&fPresentationTime, NULL);
  
      }
  
  
  
      if(!m_queue.empty())
  
      {
  
      m_queue.wait_and_pop(nalToDeliver);
  
  
  
      uint8_t* newFrameDataStart = (uint8_t*)0xD15EA5E;
  
  
  
      newFrameDataStart = (uint8_t*)(nalToDeliver.p_payload);
  
      unsigned newFrameSize = nalToDeliver.i_payload;
  
  
  
      // Deliver the data here:
  
      if (newFrameSize > fMaxSize) {
  
          fFrameSize = fMaxSize;
  
          fNumTruncatedBytes = newFrameSize - fMaxSize;
  
      }
  
      else {
  
          fFrameSize = newFrameSize;
  
      }
  
  
  
      memcpy(fTo, nalToDeliver.p_payload, nalToDeliver.i_payload);
  
  
  
      FramedSource::afterGetting(this);
  
      }
  
  }

  Relevant part of the RTSP-Server Therad

    RTSPServer* rtspServer = RTSPServer::createNew(*(parent->env), 8554, NULL);
  
    if (rtspServer == NULL) {
  
      *(parent->env) << "Failed to create RTSP server: " << (parent->env)->getResultMsg() << "\n";
  
      exit(1);
  
    }
  
    char const* streamName = "Stream";
  
    parent->h264FramedSource = H264FramedSource::createNew(*(parent->env), 0, 0);
  
    H264VideoStreamServerMediaSubsession *h264VideoStreamServerMediaSubsession = H264VideoStreamServerMediaSubsession::createNew(*(parent->env), true);
  
    h264VideoStreamServerMediaSubsession->parent = parent;
  
    sms->addSubsession(h264VideoStreamServerMediaSubsession);
  
    rtspServer->addServerMediaSession(sms);
  
  
  
    parent->env->taskScheduler().doEventLoop(); // does not return

  Once a connection exists the render loop calls

  h264FramedSource->AddToBuffer(videoData, 1024*768*4);

• PHP FFmpeg metadata usage

  28 novembre 2013, par user2976292

   shell_exec("d:/wamp/ffmpg/ffmpeg  -metadata title="Movie Title" -metadata year="2010"  -i $vaw  -b:v 64k  -bufsize 64k $mp3");

  i need add meta data to new output file with utf8 chars in installed windows localhost.

  How can i add metadata txt have you any examples for metadata.txt ?

• On ALAC’s Open Sourcing

  1er novembre 2011, par Multimedia Mike — Codec Technology

  Apple open sourced their lossless audio codec last week. Pretty awesome ! I have a theory that, given enough time, absolutely every codec will be open source in one way or another.

  I know I shouldn’t bother reading internet conversation around any news related to multimedia technology. And if I do read it, I shouldn’t waste any effort getting annoyed about them. But here are some general corrections :

  • ALAC is not in the same league as — nor is it a suitable replacement for — MP3/AAC/Vorbis or any other commonly used perceptual audio codec. It’s not a matter of better or worse ; they’re just different families of codecs designed for different purposes.
  • Apple open sourced ALAC, not AAC– easy mistake, though there’s nothing to ‘open source’ about AAC (though people can, and will, argue about its absolute ‘open-ness’).
  • There’s not much technical room to argue between ALAC and FLAC, the leading open source lossless audio compressor. Both perform similarly in terms of codec speeds (screamingly fast) and compression efficiency (results vary slightly depending on source material).
  • Perhaps the most frustrating facet is the blithe ignorance about ALAC’s current open source status. While this event simply added an official “open source” status to the codec, ALAC has effectively been open source for a very long time. According to my notes, the ALAC decoding algorithm was reverse engineered in 2005 and added into FFmpeg in March of the same year. Then in 2008, Google — through their Summer of Code program — sponsored an open source ALAC encoder.

  From the multimedia-savvy who are versed in these concepts, the conversation revolves around which would win in a fight, ALAC or FLAC ? And who between Apple and FFmpeg/Libav has a faster ALAC decoder ? The faster and more efficient ALAC encoder ? I contend that these issues don’t really matter. If you have any experience working with lossless audio encoders, you know that they tend to be ridiculously fast to both encode and decode and that many different lossless codecs compress at roughly the same ratios.

  As for which encoder is the fastest : use whatever encoder is handiest and most familiar, either iTunes or FFmpeg/Libav.

  As for whether to use FLAC or ALAC — if you’ve already been using one or the other for years, keep on using it. Support isn’t going to vanish. If you’re deciding which to use for a new project, again, perhaps choose based on software you’re already familiar with. Also, consider hardware support– ALAC enjoys iPod support, FLAC is probably better supported in a variety of non-iPod devices, though that may change going forward due to this open sourcing event.

  For my part, I’m just ecstatic that the question of moral superiority based on open source status has been removed from the equation.

  Code-wise, I’m interested in studying the official ALAC code to see if it has any corner-case modes that the existing open source decoders don’t yet account for. The source makes mention of multichannel (i.e., greater than stereo) configurations, but I don’t know if that’s in FFmpeg/Libav.