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  • Des sites réalisés avec MediaSPIP

    2 mai 2011, par

    Cette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
    Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page.

  • Keeping control of your media in your hands

    13 avril 2011, par

    The vocabulary used on this site and around MediaSPIP in general, aims to avoid reference to Web 2.0 and the companies that profit from media-sharing.
    While using MediaSPIP, you are invited to avoid using words like "Brand", "Cloud" and "Market".
    MediaSPIP is designed to facilitate the sharing of creative media online, while allowing authors to retain complete control of their work.
    MediaSPIP aims to be accessible to as many people as possible and development is based on expanding the (...)

  • MediaSPIP v0.2

    21 juin 2013, par

    MediaSPIP 0.2 est la première version de MediaSPIP stable.
    Sa date de sortie officielle est le 21 juin 2013 et est annoncée ici.
    Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
    Comme pour la version précédente, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
    Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...)

Sur d’autres sites (7214)

  • 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);
  • How to scale and position watermark to scale ?

    26 avril 2013, par Cobra_Fast

    I'm scaling a video and applying a watermark like so :

    ffmpeg -ss 0:0:0.000 -i video.mp4 -y -an -t 0:0:10.000
    -vf \"[in]scale=400:316[middle]\" -b:v 2000k -r 20
    -vf 'movie=watermark.png,pad=400:316:0:0:0x00000000 [watermark];[middle] [watermark]overlay=0:0[out]'
    out.flv

    However, the applied watermark seems to be scaled to the original video size rather than the smaller scaled video size.

    This command line worked on ffmpeg version 0.8.6.git and now behaves differently after an upgrade to version N-52381-g2288c77.

    How do I get it to work again ?


    Update 2013-04-26 :
    I now have tried to use the overlay filter's X and Y parameters instead of padding without success.
  • How to make FFmpeg download only the required segments [closed]

    25 juillet, par daniil_

    I'm developing a tool, that create timelapse. For this purpose I use ffmpeg 7.1.1 verion.
I have a playlist file called index.m3u8 (inside it there are URLs to TS segments—typically around 5,000 of them). Here’s a small excerpt :

    


    #EXTM3U
#EXT-X-TARGETDURATION:12
#EXT-X-VERSION:6
#EXT-X-MEDIA-SEQUENCE:0
#EXT-X-PLAYLIST-TYPE:EVENT
#EXT-X-START:TIME-OFFSET=8,PRECISE=YES
#EXT-X-PROGRAM-DATE-TIME:2025-07-22T07:59:51.173Z
#EXTINF:10.080,
https://my-storage.io/receiver/archives/3a1b162d-392d-d45a-089a-dc93fb842a35/files/2025-07-22/07/59/1753171191173_aa_00000_00000_10080.ts?exp=1753543086&signature=MEYCIQCx_hDnFVwk7WJyQG5QujV5ZGhLoBqkD5uoQFebha1knQIhAMJm1KA5DLEuqD4_5zS5QlzTGIuj5q9TCUcML7MEPMl7
#EXT-X-PROGRAM-DATE-TIME:2025-07-22T08:00:01.253Z
#EXTINF:10.080,
https://my-storage.io/receiver/archives/3a1b162d-392d-d45a-089a-dc93fb842a35/files/2025-07-22/08/00/1753171201253_aa_00000_00000_10080.ts?exp=1753543086&signature=MEYCIQCx_hDnFVwk7WJyQG5QujV5ZGhLoBqkD5uoQFebha1knQIhAMJm1KA5DLEuqD4_5zS5QlzTGIuj5q9TCUcML7MEPMl7
#EXT-X-PROGRAM-DATE-TIME:2025-07-22T08:00:11.333Z
#EXTINF:10.021,
https://my-storage.io/receiver/archives/3a1b162d-392d-d45a-089a-dc93fb842a35/files/2025-07-22/08/00/1753171211333_aa_00000_00000_10021.ts?exp=1753543086&signature=MEYCIQCx_hDnFVwk7WJyQG5QujV5ZGhLoBqkD5uoQFebha1knQIhAMJm1KA5DLEuqD4_5zS5QlzTGIuj5q9TCUcML7MEPMl7
#EXT-X-PROGRAM-DATE-TIME:2025-07-22T08:00:21.354Z
#EXTINF:10.079,
https://my-storage.io/receiver/archives/3a1b162d-392d-d45a-089a-dc93fb842a35/files/2025-07-22/08/00/1753171221354_aa_00000_00000_10079.ts?exp=1753543086&signature=MEYCIQCx_hDnFVwk7WJyQG5QujV5ZGhLoBqkD5uoQFebha1knQIhAMJm1KA5DLEuqD4_5zS5QlzTGIuj5q9TCUcML7MEPMl7
#EXT-X-PROGRAM-DATE-TIME:2025-07-22T08:00:31.433Z
...


    


    I’m trying to extract still images from the stream at specific moments.

    


    When I run a command like :

    


    ffmpeg \
  -analyzeduration 5000000 \
  -probesize 5000000 \
  -err_detect ignore_err \
  -protocol_whitelist file,http,https,tcp,tls \
  -allowed_extensions ALL \
  -f hls \
  -y \
  -ss 11316.719 \
  -i /Users/daniil/Desktop/test/exports/.../index.m3u8 \
  -frames:v 1 \
  -q:v 2 \
  /Users/daniil/Desktop/test/exports/.../frames/frame_00100.png


    


    ffmpeg downloads segments 1 and 2 (presumably to probe the format), and then it downloads two more TS files around the target timestamp.

    


    But as soon as I push the -ss value beyond a certain point—in my case -ss 8487.54—it starts downloading every TS segment from the very start of the playlist up to the one I need. That can easily be 1,000 TS files or more. In other words, when -ss is between 0 and 8374.372, everything works fine, but beyond that it exhibits this strange behavior.