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• mpeg2 ts android ffmpeg openmax

  10 octobre 2014, par WLGfx

  The setup is as follows :

  1. Multicast server 1000Mbs, UDP, Mpeg2-TS Part 1 (H.222) streaming live TV-channels.
  2. Quad core 1.5Ghz Android 4.2.2 GLES 2.0 renderer.
  3. FFMpeg library.
  4. Eclipse Kepler, Android SDK/NDK, etc. Running on Windows 8.1.
  5. Output screen 1920 x 1080, I am using a texture of 2048 x 1024 and getting between 35 and 45 Frames per second.

  The app :

  1. Renderer thread runs continuously and updates a single texture by uploading segments to the gpu when media images are ready.
  2. Media handler thread, downloads and processes media from server/or local storage.
  3. Video thread(s), one for buffering the UDP packets and another for decoding the packets into frames.

  I am connecting ffmpeg to the UDP stream just fine and the packets are being buffered and seemingly decoded fine. The packet buffers are plenty, no under/over-flows. The problem I am facing is it appears to be chopping up frames (ie only playing back 1 out of every so many frames). I understand that I need to distinguish I/P/B frames, but at the moment, hands up, I ain’t got a clue. I’ve even tried a hack to detect I frames to no avail. Plus, I am only rendering the frames to less than a quarter of the screen. So I’m not using full screen decoding.

  The decoded frames are also stored in separate buffers to cut out page tearing. The number of buffers I’ve changed too, from 1 to 10 with no luck.

  From what I’ve found about OpenMax IL, is it only handles MPeg2-TS Part 3 (H.264 and AAC), but you can use your own decoder. I understand that you can add your own decode component to it. Would it be worth me trying this route or should I continue on with ffmpeg ?

  The frame decoder (only the renderer will convert and scale the frames when ready)
  /*
  * This function will run through the packets and keep decoding
  * until a frame is ready first, or out of packets
  */

  while (packetsUsed[decCurrent])
  
  {
  
      hack_for_i_frame:
  
      i = avcodec_decode_video2(pCodecCtx, pFrame, &frameFinished, &packets[decCurrent]);
  
      packetsUsed[decCurrent] = 0; // finished with this one
  
      i = packets[decCurrent].flags & 0x0001;
  
      decCurrent++;
  
      if (decCurrent >= MAXPACKETS) decCurrent = 0;
  
      if (frameFinished)
  
      {
  
          ready_pFrame = pFrame;
  
          frameReady = true;  // notify renderer
  
          frameCounter++;
  
          if (frameCounter>=MAXFRAMES) frameCounter = 0;
  
          pFrame = pFrames[frameCounter];
  
          return 0;
  
      }
  
      else if (i)
  
          goto hack_for_i_frame;
  
  }
  
  
  
  return 0;

  The packet reader (spawned as a pthread)
  void *mainPacketReader(void *voidptr)
  
  int res ;

  while ( threadState == TS_RUNNING )
  
  {
  
      if (packetsUsed[prCurrent])
  
      {
  
          LOGE("Packet buffer overflow, dropping packet...");
  
          av_read_frame( pFormatCtx, &packet );
  
      }
  
      else if ( av_read_frame( pFormatCtx, &packets[prCurrent] ) >= 0 )
  
      {
  
          if ( packets[prCurrent].stream_index == videoStream )
  
          {
  
              packetsUsed[prCurrent] = 1; // flag as used
  
              prCurrent++;
  
              if ( prCurrent >= MAXPACKETS )
  
              {
  
                  prCurrent = 0;
  
              }
  
          }
  
  
  
          // here check if the packet is audio and add to audio buffer
  
      }
  
  }
  
  return NULL;

  And the renderer just simply does this
  // texture has already been bound before calling this function

  if ( frameReady == false ) return;
  
  
  
  AVFrame *temp;  // set to frame 'not' currently being decoded
  
  temp = ready_pFrame;
  
  
  
  sws_scale(sws_ctx,(uint8_t const* const *)temp->data,
  
          temp->linesize, 0, pCodecCtx->height,
  
          pFrameRGB->data, pFrameRGB->linesize);
  
  
  
  glTexSubImage2D(GL_TEXTURE_2D, 0,
  
          XPOS, YPOS, WID, HGT,
  
          GL_RGBA, GL_UNSIGNED_BYTE, buffer);
  
  
  
  frameReady = false;

  In the past, libvlc had audio syncing problems too, so that is my decision for going with ffmpeg and doing all the donkey work from scratch.

  If anybody has any pointers of how to stop the choppiness of the video playback (works great in VLC player) or possibly another route to go down, it would be seriously appreciated.

  EDIT I removed the hack for the I-frame (completely useless). Move the sws_scale function from the renderer to the packet decoder. And I left the udp packet reader thread alone.

  In the meantime I’ve also changed the packet reader thread and the packet decoder threads priority to real-time. Since doing that I don’t get shed loads of dropped packets.

• Evolution #3229 (Nouveau) : Utiliser une méthode standard dès que l’on a besoin d’un module JS

  14 juin 2014, par RastaPopoulos ♥

  Actuellement, on a tendance :
  - soit à charger du JS sur toutes les pages
  - soit à le charger au moment où on en a besoin
  - soit, mieux mais fastidieux, à gérer soi-même à la main un mécanisme de chargement unique d’un module (cf GIS dernièrement)

  Il semblerait pourtant qu’il existe depuis plusieurs années enfin des méthodes pour appeler un ou plusieurs modules uniquement lorsque nécessaire, et en pouvant les appeler plusieurs fois sans risque. Bref, l’équivalent des inclusions PHP.

  Pour jQuery de base je ne sais pas, mais ce serait au moins utile pour tel ou tel module de jQueryUI, pour le JS de datation de SPIP, (inc-dateur), pour Leaflet, etc.

  Doit-on vraiment réinventer la roue à chaque fois qu’on veut n’inclure un JS qu’une seule fois ?

  http://requirejs.org/docs/whyamd.html
  http://developer.telerik.com/featured/jquery-using-only-what-you-need/

  Qu’en pensent les experts en cache et en performance ? :)

• Evolution #3394 (Fermé) : Fonction {trunchtml}

  27 juillet 2015, par cedric -

  C’est une autre fonction qui est candidate, intégrée pour le moment dans spip-bonux
  http://zone.spip.org/trac/spip-zone/changeset/87957/_plugins_/spip-bonux-3