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  • MediaSPIP 0.1 Beta version

    25 avril 2011, par

    MediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
    The zip file provided here only contains the sources of MediaSPIP in its standalone version.
    To get a working installation, you must manually install all-software dependencies on the server.
    If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)

  • Multilang : améliorer l’interface pour les blocs multilingues

    18 février 2011, par

    Multilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
    Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela.

  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
    The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
    For older browsers the Flowplayer flash fallback is used.
    MediaSPIP allows for media playback on major mobile platforms with the above (...)

Sur d’autres sites (9543)

  • I want to record live stream and display it on my site, how to do it ? (ffmeg)

    18 octobre 2011, par user893856

    I want to record live stream and display it on my site, how to do it ? (ffmeg)
    I know FFMPEG might know it, but I didnt find a way to record from the display. If it can record and stream toward a site, how could it display it realtime ?

  • Error in converting audio file format from ogg to wav [on hold]

    9 juin 2014, par Sumit Bisht

    I am trying to convert an ogg format file that was created using webrtc (html5 usermedia content generated on firefox) and transferred and decoded on the server into a wav file through ffmpeg but am getting this error on cmmand line while trying to convert :

    $ ffmpeg -i 2014-6-5_16-17-54.ogg res1.wav
    
ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg developers
    
  built on May  1 2014 13:12:12 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-4)
    
  configuration: --enable-gpl --enable-version3 --enable-shared --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid
    
  libavutil      52. 38.100 / 52. 38.100
    
  libavcodec     55. 18.102 / 55. 18.102
    
  libavformat    55. 12.100 / 55. 12.100
    
  libavdevice    55.  3.100 / 55.  3.100
    
  libavfilter     3. 79.101 /  3. 79.101
    
  libswscale      2.  3.100 /  2.  3.100
    
  libswresample   0. 17.102 /  0. 17.102
    
  libpostproc    52.  3.100 / 52.  3.100
    
Guessed Channel Layout for  Input Stream #0.0 : mono
    
Input #0, ogg, from '2014-6-5_16-17-54.ogg':
    
  Duration: 00:00:01.84, start: 0.000000, bitrate: 18 kb/s
    
    Stream #0:0: Audio: opus, 48000 Hz, mono
    
    Metadata:
    
      ENCODER         : Mozilla29.0.1
    
[graph 0 input from stream 0:0 @ 0x18dca20] Invalid sample format (null)
    
Error opening filters!

    Although, I am able to play the file on server and using the same command, am able to convert .ogg files generated somewhere else. What might be I missing ?

    Edit :
    Here’s the source code that is used to write to the file :

    1) During startup - use the methods of getUserMedia API.

    navigator.getUserMedia({
    
        audio: true,
    
        video: false
    
    }, function(stream) {
    
        audioStream = RecordRTC(stream, {
    
            bufferSize: 16384
    
        });
    
        audioStream.startRecording();

    2) During stopping of the recording - extracting the recorded information.

    function(audioDataURL) {
    
     var audioFile = {};
    
     audioFile = {
    
       contents: audioDataURL
    
     **strong text**};

    On server end, the following code is creating a file from this data :

    dataURL = dataURL.split(',').pop(); // dataURL is the audioDataURL as defined above
    
fileBuffer = new Buffer(dataURL, 'base64');
    
fs.writeFileSync(filePath, fileBuffer);

  • Révision 17716 : implémenter les attributs ARIA sur les zones live (inclusion et formulaires ajax...

    13 avril 2011, par cedric -

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