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  • MediaSPIP version 0.1 Beta

    16 avril 2011, par

    MediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
    Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
    Pour avoir une installation fonctionnelle, 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 (...)

  • 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 (...)

  • Amélioration de la version de base

    13 septembre 2013

    Jolie sélection multiple
    Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
    Il suffit pour cela d’activer le plugin Chosen (Configuration générale du site > Gestion des plugins), puis de configurer le plugin (Les squelettes > Chosen) en activant l’utilisation de Chosen dans le site public et en spécifiant les éléments de formulaires à améliorer, par exemple select[multiple] pour les listes à sélection multiple (...)

Sur d’autres sites (13847)

  • Create MPEG-DASH Initialization segment

    5 janvier 2016, par Mahout

    I am looking to convert between HLS and MPEG Dash. I do not access to the original fully concatenated video file, only the individual HLS segments.

    In doing this transformation to MPEG Dash I need to supply an initialziation segment for the Dash manifest .mpd file.

    My questions are :

    1. What is the structure of a Dash video initialization segment ?
    2. How can I generate/create one without the need for the original full file ?

    Perhaps a solution would involve getting MP4Box to convert the ’.ts’ HLS segments to Dash ’.m4s’ segments which are self initializing, but I am unsure how to go about this this ?

    Any ideas are much appreciated.

    Many thanks.

    UPDATE :
    Snippet to stream using original hls segments. Video plays all the way through but is just black.

     <representation width="426" height="238" framerate="25" bandwidth="400000">
       <segmentlist timescale="25000" duration="112500">
              <segmenturl media="video_0_400000/hls/segment_0.ts"></segmenturl>
              <segmenturl media="video_0_400000/hls/segment_1.ts"></segmenturl>
             <segmenturl media="video_0_400000/hls/segment_2.ts"></segmenturl>
       </segmentlist>
      </representation>
  • How to re-encode an audio to match another one, to avoid re-encoding the whole audio

    21 mars 2024, par Bernard Wiesner

    I have an audio editor in the browser using ffmpeg (WebAssembly), and I want to insert new audio into the existing audio without having to re-encode everything. Re-encoding everything takes a long time, especially in the browser, so I would like to only re-encode the inserted file, match it to the original one and concatenate them using the copy command.

    &#xA;

    On ffmpeg concatenate docs it says :

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    &#xA;

    All files must have the same streams (same codecs, same time base, etc.)

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    But it is not clear what is meant by time base. So far I have observed I need to match :

    &#xA;

      &#xA;
    • codec
    • &#xA;

    • bit rate
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    • sample rate
    • &#xA;

    • channels (mono, stereo)
    • &#xA;

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    Is there anything else I need to match so that the resulting audio is not corrupt/broken when concatenating ?

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    I have observed with mp3 for example it has VBR, CBR, and ABR. If the original audio has a bit rate of 128 kb/s, I am assuming it is a CBR, so I match it with :

    &#xA;

    ffmpeg -i original.mp3&#xA;# > Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 128 kb/s&#xA;&#xA;ffmpeg -i input.mp3 -b:a 128k -ar 44100 -ac 2 re_encoded.mp3&#xA;&#xA;# then merge&#xA;# concat_list.txt contains the original audio and the re_encoded.mp3&#xA;&#xA;ffmpeg -f concat -i concat_list.txt -safe 0 -c copy merged.mp3&#xA;

    &#xA;

    And that works fine for CBR such as 8, 16, 24, 32, 40, 48, 64, 80, 96, 112, 128, 160, 192, 224, 256, or 320 (docs), as far as I have tested.

    &#xA;

    The issue is when the original.mp3 has a VBR (variable bit rate) or ABR, such as 150 kb/s.

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    If I try to match it like below :

    &#xA;

    ffmpeg -i input.mp3 -b:a 150k -ar 44100 -ac 2 re_encoded.mp3&#xA;ffmpeg -i re_encoded.mp3&#xA;# Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 160 kb/s&#xA;

    &#xA;

    The resulting bitrate is rounded to the nearest CBR which is 160.

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    I can solve this with mp3 by using -abr 1 :

    &#xA;

    ffmpeg -i input.mp3 -abr 1 -b:a 150k -ar 44100 -ac 2 re_encoded.mp3&#xA;ffmpeg -i re_encoded.mp3&#xA;# Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 150 kb/s&#xA;

    &#xA;

    Now the bitrate matches the original audio, however I am not sure this is correct since I am modifying the new audio to an ABR and concatenating it with a VBR ? I am not even sure how to check with ffmpeg if the audio is VBR, CBR or ABR, or if that even matters when concatenating.

    &#xA;

    Another issue also happens with aac files. When I try to match the original audio bitrate I can't.

    &#xA;

    ffmpeg -i input.mp3 -b:a 128k -ar 44100 -ac 2 re_encoded.aac&#xA;ffmpeg -i re_encoded.aac&#xA;# Stream #0:0: Audio: aac (LC), 44100 Hz, stereo, fltp, 135 kb/s&#xA;

    &#xA;

    The resulting bitrate always seems to be variable (135 in this case), and hence I can't match it to the original one.

    &#xA;

    So my question is, what conditions need to be met when concatenating audios with different streams, and how can I achieve re-encoding only one audio to match the other one. Or if there is some package that can do this, it would be of great help.

    &#xA;

  • MPEG4 quality saved in MATLAB

    17 août 2017, par guyts

    I need to get an mpeg4 file to use in another application, from an original mpeg4 video I loaded into matlab and edited (frame by frame).
    To do so, I tried using VideoWriter, setting the quality to 100% :

    newVid = VideoWriter(outputfilename, 'MPEG-4');
    newVid.FrameRate = fps;
    newVid.Quality = 100;

    However, the result I’m getting is very poor and if the original unedited video size was 50MB, the one I get post-edit in matlab is around 20MB, and I don’t know how to keep the quality and size as they were.
    I also tried saving as .avi and converting to mpeg4 with ffmpeg, but it gave even poorer results.
    Any ideas ?