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• Revision 896b6a9b57 : Merge snapshot 2 development history Create a merge between the experimental de

  12 juillet 2012, par John Koleszar

  Merge snapshot 2 development history Create a merge between the experimental development history and the previously published snapshot. Change-Id : If320df72a6bbefec53833626d08dbc9678be2d2d

• Solving The XVD Puzzle

  15 avril 2012, par Multimedia Mike — General, multimedi archaeology, silicon valley, vg2, xvd, zygo

  I downloaded a multimedia file a long time ago (at least, I strongly suspected it was a multimedia file which is why I downloaded it). It went by the name of ‘lamborghini_850kbps.vg2′. I have had it in my collection for at least 7 years. I couldn’t remember where I found it. I downloaded it before it occurred to me to take notes about this sort of stuff.

  I found myself staring at the file again today and Googled the filename. This led me to a few Japanese sites which also contained working URLs for a few more .vg2 samples. Some other clues led me to a Russian language forum where someone had linked to a site that had Win32 codec modules that could process the files. The site was defunct but the Internet Archive Wayback Machine kept a copy for me, as well as copies of several more .vg2 samples from a defunct Japanese site previously involved with this codec.

  Sometimes this internet technology works really well. But I digress.

  Anyway, through all this, I finally found a clue : XVD. and wouldn’t you know, there is already a basic page on the MultimediaWiki describing the technology. In fact, while VG2 is a custom container, the MultimediaWiki states that the video component has a FourCC of VGMV, and there is already a file named VGMV.avi in the root V-codecs/ samples directory, something I vow to correct (that’s a big pet peeve of mine– putting samples in the root V-codecs/ or A-codecs/ directories).

  XVD… XVD… XVD… why does that sound so familiar ? Oh, of course ; there is a company named XVD and they have an office in the Bay Area which I have passed on numerous occasions, like this morning :

  
  <

  Someone originally connected with the multimedia technology in question operates a website which contains an unofficial history of the XVD tech. At first, I was wondering if the technology was completely defunct (and should therefore be open sourced). But if XVD’s solutions page (dated 2010) is to be believed, the technology is still in service, and purported to be better than H.264 and VC-1 : “The current generation of XVD video compression technology provides better video quality at any given data rate than standards-based codecs (H.264 or VC-1) with four times lower encoding complexity (when compared with H.264 Main Profile).”

  If they say so. For my part, I’m just happy that I have finally figured out what this lamborghini_850kbps.vg2 is so that I can properly catalog it on the samples site, which I have now done, along with other samples and various codecs modules.

  This episode reminds me that there’s a branch office of Zygo Corporation close to my home (though the headquarters are far, far away). The companies you see in Silicon Valley. Anyway, long-time open source multimedia hackers will no doubt recognize Zygo from the ZyGo FourCC & video codec transported in QuickTime files that was almost decode-able using an H.263 decoder.

  
  
  

  I may never learn what Zygo’s core competency actually is, but I will always remember their multimedia tech every time I run past their office.

• streaming H.264 over RTP with libavformat

  16 avril 2012, par Jacob Peddicord

  I've been trying over the past week to implement H.264 streaming over RTP, using x264 as an encoder and libavformat to pack and send the stream. Problem is, as far as I can tell it's not working correctly.

  Right now I'm just encoding random data (x264_picture_alloc) and extracting NAL frames from libx264. This is fairly simple :

  x264_picture_t pic_out;
  
  x264_nal_t* nals;
  
  int num_nals;
  
  int frame_size = x264_encoder_encode(this->encoder, &nals, &num_nals, this->pic_in, &pic_out);
  
  
  
  if (frame_size <= 0)
  
  {
  
      return frame_size;
  
  }
  
  
  
  // push NALs into the queue
  
  for (int i = 0; i < num_nals; i++)
  
  {
  
      // create a NAL storage unit
  
      NAL nal;
  
      nal.size = nals[i].i_payload;
  
      nal.payload = new uint8_t[nal.size];
  
      memcpy(nal.payload, nals[i].p_payload, nal.size);
  
  
  
      // push the storage into the NAL queue
  
      {
  
          // lock and push the NAL to the queue
  
          boost::mutex::scoped_lock lock(this->nal_lock);
  
          this->nal_queue.push(nal);
  
      }
  
  }

  nal_queue is used for safely passing frames over to a Streamer class which will then send the frames out. Right now it's not threaded, as I'm just testing to try to get this to work. Before encoding individual frames, I've made sure to initialize the encoder.

  But I don't believe x264 is the issue, as I can see frame data in the NALs it returns back.
  Streaming the data is accomplished with libavformat, which is first initialized in a Streamer class :

  Streamer::Streamer(Encoder* encoder, string rtp_address, int rtp_port, int width, int height, int fps, int bitrate)
  
  {
  
      this->encoder = encoder;
  
  
  
      // initalize the AV context
  
      this->ctx = avformat_alloc_context();
  
      if (!this->ctx)
  
      {
  
          throw runtime_error("Couldn't initalize AVFormat output context");
  
      }
  
  
  
      // get the output format
  
      this->fmt = av_guess_format("rtp", NULL, NULL);
  
      if (!this->fmt)
  
      {
  
          throw runtime_error("Unsuitable output format");
  
      }
  
      this->ctx->oformat = this->fmt;
  
  
  
      // try to open the RTP stream
  
      snprintf(this->ctx->filename, sizeof(this->ctx->filename), "rtp://%s:%d", rtp_address.c_str(), rtp_port);
  
      if (url_fopen(&(this->ctx->pb), this->ctx->filename, URL_WRONLY) < 0)
  
      {
  
          throw runtime_error("Couldn't open RTP output stream");
  
      }
  
  
  
      // add an H.264 stream
  
      this->stream = av_new_stream(this->ctx, 1);
  
      if (!this->stream)
  
      {
  
          throw runtime_error("Couldn't allocate H.264 stream");
  
      }
  
  
  
      // initalize codec
  
      AVCodecContext* c = this->stream->codec;
  
      c->codec_id = CODEC_ID_H264;
  
      c->codec_type = AVMEDIA_TYPE_VIDEO;
  
      c->bit_rate = bitrate;
  
      c->width = width;
  
      c->height = height;
  
      c->time_base.den = fps;
  
      c->time_base.num = 1;
  
  
  
      // write the header
  
      av_write_header(this->ctx);
  
  }

  This is where things seem to go wrong. av_write_header above seems to do absolutely nothing ; I've used wireshark to verify this. For reference, I use Streamer streamer(&enc, "10.89.6.3", 49990, 800, 600, 30, 40000); to initialize the Streamer instance, with enc being a reference to an Encoder object used to handle x264 previously.

  Now when I want to stream out a NAL, I use this :

  // grab a NAL
  
  NAL nal = this->encoder->nal_pop();
  
  cout << "NAL popped with size " << nal.size << endl;
  
  
  
  // initalize a packet
  
  AVPacket p;
  
  av_init_packet(&p);
  
  p.data = nal.payload;
  
  p.size = nal.size;
  
  p.stream_index = this->stream->index;
  
  
  
  // send it out
  
  av_write_frame(this->ctx, &p);

  At this point, I can see RTP data appearing over the network, and it looks like the frames I've been sending, even including a little copyright blob from x264. But, no player I've used has been able to make any sense of the data. VLC quits wanting an SDP description, which apparently isn't required.

  I then tried to play it through gst-launch :

  gst-launch udpsrc port=49990 ! rtph264depay ! decodebin ! xvimagesink

  This will sit waiting for UDP data, but when it is received, I get :

  ERROR : element /GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0 : No RTP
  format was negotiated. Additional debug info :
  gstbasertpdepayload.c(372) : gst_base_rtp_depayload_chain () :
  /GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0 : Input buffers
  need to have RTP caps set on them. This is usually achieved by setting
  the 'caps' property of the upstream source element (often udpsrc or
  appsrc), or by putting a capsfilter element before the depayloader and
  setting the 'caps' property on that. Also see
  http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README

  As I'm not using GStreamer to stream itself, I'm not quite sure what it means with RTP caps. But, it makes me wonder if I'm not sending enough information over RTP to describe the stream. I'm pretty new to video and I feel like there's some key thing I'm missing here. Any hints ?