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  • Les autorisations surchargées par les plugins

    27 avril 2010, par

    Mediaspip core
    autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs

  • List of compatible distributions

    26 avril 2011, par

    The table below is the list of Linux distributions compatible with the automated installation script of MediaSPIP. Distribution nameVersion nameVersion number Debian Squeeze 6.x.x Debian Weezy 7.x.x Debian Jessie 8.x.x Ubuntu The Precise Pangolin 12.04 LTS Ubuntu The Trusty Tahr 14.04
    If you want to help us improve this list, you can provide us access to a machine whose distribution is not mentioned above or send the necessary fixes to add (...)

  • Le profil des utilisateurs

    12 avril 2011, par

    Chaque utilisateur dispose d’une page de profil lui permettant de modifier ses informations personnelle. Dans le menu de haut de page par défaut, un élément de menu est automatiquement créé à l’initialisation de MediaSPIP, visible uniquement si le visiteur est identifié sur le site.
    L’utilisateur a accès à la modification de profil depuis sa page auteur, un lien dans la navigation "Modifier votre profil" est (...)

Sur d’autres sites (5316)

  • FFmpeg : getting correct output bitrate

    13 novembre 2015, par user2809516

    I’m trying to do something which seems relatively simple, creating a ffmpeg command which will take any mp3 file and change the bitrate to 48k on the output mp3. However, I can’t seem to do so without forcing the sampling rate down to 16k, even though the tool says the output bitrate is 48.

    Sample commands :

    ffmpeg -i input.mp3 -b:a 48k output.mp3

    ffmpeg says gives this output :

    size=     193kB time=00:00:32.88 bitrate=  48.1kbits/s

    However, when I open the file in VLC, it says the bitrate is 64k.

    I can get proper output only by changing the sample rate to 16k :

    ffmpeg -i input.mp3 -b:a 48k -ar 16000 output.mp3

    When I play this in VLC it shows the proper bitrate of 48k.

    Is there any way to get ffmpeg to output a 48k file with a sample rate of 44.1k, and if so, how is this done ?

  • Correct ffmpeg scale command syntax

    10 novembre 2015, par user3132858

    I am new to ffmpeg and need some help with the correct syntax for a scale command I found here.

    Below is my code :

    $width=300;
    $height=200;
    $aspect=$width/$height;


    $command = "/usr/local/bin/ffmpeg -y -i 1.mp4 -vf scale=min(1\,gt(iw\,".$width.")+gt(ih\,".$height.")) * (gte(a\,".$aspect.")*".$width." + \
    lt(a\,".$aspect.")*((".$height."*iw)/ih)) + not(min(1\,gt(iw\,".$width.")+gt(ih\,".$height.")))*iw : \
    min(1\,gt(iw\,".$width.")+gt(ih\,".$height.")) * (lte(a\,".$aspect.")*".$height." + \
    gt(a\,".$aspect.")*((".$width."*ih)/iw)) + not(min(1\,gt(iw\,".$width.")+gt(ih\,".$height.")))*ih  -b 1200k -acodec aac -strict -2 2.mp4 ";

    Video is successfully converted however, the scaling does not go trough.

    The code is copied word for word, I only make two changes :

    • removed the " before scale word and from the end
    • replaced the $FW with $width, $FH with $height and $FA with $aspect.

    Any suggestion at what might be wrong ?

  • Concatenate TS files with correct timestamps

    23 mai 2016, par Pete9119

    I’m trying to merge multiple ts chunk files to one single file, without any loss of quality or reencoding. The files are taken from a live stream, however I’m trying to merge them in a diffrent order and not the order they were streamed.

    Example of files :

    0000000033.ts
    0000000034.ts
    0000000039.ts
    0000000044.ts

    I tried :

    cat 0000000033.ts 0000000034.ts 0000000039.ts 0000000044.ts >combined.ts

    and

    ffmpeg -i "concat:0000000033.ts|concat:0000000034.ts|concat:0000000039.ts|concat:0000000044.ts" -c copy -bsf:a aac_adtstoasc output.mp4

    This kinda works, however I instead of beeing 4 seconds long it’s around 15. It plays this way :

    [first 2 clips]
    [5 secs pause]
    [39.ts]
    [5 secs pause]
    [44.ts]
    [done]

    This happens to both the cat and ffmpeg combined version. So it seems the ts chunks contain timestamps from the stream that are beeing used.

    How can I fix that to make it one continous clip ?

    The chunks here are more of an example, the chunks will be dynamically selected.