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• Video Conferencing in HTML5 : WebRTC via Socket.io

  
  5 février 2013, par silvia

  Six months ago I experimented with Web sockets for WebRTC and the early implementations of PeerConnection in Chrome. Last week I gave a presentation about WebRTC at Linux.conf.au, so it was time to update that codebase.

  I decided to use socket.io for the signalling following the idea of Luc, which made the server code even smaller and reduced it to a mere reflector :

   var app = require(’http’).createServer().listen(1337) ;
   var io = require(’socket.io’).listen(app) ;
   io.sockets.on(’connection’, function(socket) 
  
           socket.on(’message’, function(message) 
  
           socket.broadcast.emit(’message’, message) ;
  
       ) ;
  
   ) ;
  

  Then I turned to the client code. I was surprised to see the massive changes that PeerConnection has gone through. Check out my slide deck to see the different components that are now necessary to create a PeerConnection.

  I was particularly surprised to see the SDP object now fully exposed to JavaScript and thus the ability to manipulate it directly rather than through some API. This allows Web developers to manipulate the type of session that they are asking the browsers to set up. I can imaging e.g. if they have support for a video codec in JavaScript that the browser does not provide built-in, they can add that codec to the set of choices to be offered to the peer. While it is flexible, I am concerned if this might create more problems than it solves. I guess we’ll have to wait and see.

  I was also surprised by the need to use ICE, even though in my experiment I got away with an empty list of ICE servers – the ICE messages just got exchanged through the socket.io server. I am not sure whether this is a bug, but I was very happy about it because it meant I could run the whole demo on a completely separate network from the Internet.

  The most exciting news since my talk is that Mozilla and Google have managed to get a PeerConnection working between Firefox and Chrome – this is the first cross-browser video conference call without a plugin ! The code differences are minor.

  Since the specification of the WebRTC API and of the MediaStream API are now official Working Drafts at the W3C, I expect other browsers will follow. I am also looking forward to the possibilities of :

  • multi-peer video conferencing like the efforts around webrtc.io,
  • the media stream recording API,
  • and the peer-to-peer data API.

  The best places to learn about the latest possibilities of WebRTC are webrtc.org and the W3C WebRTC WG. code.google.com has open source code that continues to be updated to the latest released and interoperable features in browsers.

  The video of my talk is in the process of being published. There is a MP4 version on the Linux Australia mirror server, but I expect it will be published properly soon. I will update the blog post when that happens.

• Error while extracting subtitle from mkv or m2ts to srt with FFmpeg

  14 février 2013, par user2071701

  I need to extract subtitles from different video files to .srt format (to use it in html5 video).
  I tried a lot of variant i found with google. But every time i get this error :

  Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height

  I think, this error means that ffmpeg can decode source subtitle but cant encode it to .srt format. All codecs a enabled (i compiled the later ffmpeg version from git a few times with a different configuration).

  Here is the output :

      # /usr/local/bin/ffmpeg -i /var/video/sources/Balbesy1.m2ts -an -vn  -copyinkf -scodec srt -f srt -y  sub.srt
  
  ffmpeg version N-49947-g9f16cb9 Copyright (c) 2000-2013 the FFmpeg developers
  
    built on Feb 14 2013 14:26:10 with gcc 4.4.5 (Debian 4.4.5-8)
  
    configuration: --enable-encoder=dvdsub --enable-decoder=dvdsub --enable-decoder=pgssub --enable-encoder=srt --enable-decoder=srt --enable-encoder=srt --enable-decoder=srt
  
    libavutil      52. 17.101 / 52. 17.101
  
    libavcodec     54. 91.103 / 54. 91.103
  
    libavformat    54. 63.100 / 54. 63.100
  
    libavdevice    54.  3.103 / 54.  3.103
  
    libavfilter     3. 37.101 /  3. 37.101
  
    libswscale      2.  2.100 /  2.  2.100
  
    libswresample   0. 17.102 /  0. 17.102
  
  [mpegts @ 0x2719040] Stream #5: not enough frames to estimate rate; consider increasing probesize
  
  [mpegts @ 0x2719040] Could not find codec parameters for stream 5 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size
  
  Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
  [NULL @ 0x271fec0] start time is not set in estimate_timings_from_pts
  
  Input #0, mpegts, from '/var/video/sources/Balbesy1.m2ts':
  
    Duration: 01:18:11.89, start: 599.958300, bitrate: 37378 kb/s
  
    Program 1 
  
      Stream #0:0[0x1011]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc
  
      Stream #0:1[0x1100]: Audio: dts (DTS-HD MA) ([134][0][0][0] / 0x0086), 48000 Hz, 5.1(side), fltp, 768 kb/s
  
      Stream #0:2[0x1101]: Audio: dts (DTS-HD MA) ([134][0][0][0] / 0x0086), 48000 Hz, 5.1(side), fltp, 768 kb/s
  
      Stream #0:3[0x1102]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s
  
      Stream #0:4[0x1103]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s
  
      Stream #0:5[0x1200]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)
  
  Output #0, srt, to 'sub.srt':
  
      Stream #0:0: Subtitle: srt
  
  Stream mapping:
  
    Stream #0:5 -> #0:0 (pgssub -> srt)
  
  Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height

  sorry for my english

• FFMPEG Errors : "max_analyze_duration" "buffer underflow" "packet too large" What to do ?

  8 février 2013, par Leon5x

  I have an error while just copying a video with ffmpeg.
  I use the command :

  ffmpeg -i leon.mpg -vcodec copy -acodec copy leon2.mpg

  The errors :

  [mpeg @ 00000000020ebd20] max_analyze_duration 5000000 reached at 5004000 microseconds
  
  [mpeg @ 00000000042f4020] buffer underflow i=0 bufi=11286 size=14824
  
  [mpeg @ 00000000042f4020] packet too large, ignoring buffer limits to mux it

  What do I have to set that the buffer errors don't occur anymore ?

  Here is a picture of what ffmpeg gives out first. After that the red errors repeat really many times. There you also can see what the film's codec and so on is.
  See this picture :

  I use the ffmpeg Version git-5ce023b (2013-01-15) - Win64 - static build from Zeranoe.
  I tried some things but nothing happened. I searched a while but found no solution.