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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.

  • Les tâches Cron régulières de la ferme

    1er décembre 2010, par

    La gestion de la ferme passe par l’exécution à intervalle régulier de plusieurs tâches répétitives dites Cron.
    Le super Cron (gestion_mutu_super_cron)
    Cette tâche, planifiée chaque minute, a pour simple effet d’appeler le Cron de l’ensemble des instances de la mutualisation régulièrement. Couplée avec un Cron système sur le site central de la mutualisation, cela permet de simplement générer des visites régulières sur les différents sites et éviter que les tâches des sites peu visités soient trop (...)

Sur d’autres sites (12683)

  • lavu/tx : require output argument to match input for inplace transforms

    26 février 2021, par Lynne
    lavu/tx : require output argument to match input for inplace transforms

    This simplifies some assembly code by a lot, by either saving a branch
    
or saving an entire duplicated function.
    • [DH] libavutil/tx.h
    • [DH] libavutil/tx_template.c

  • 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.

    


    On ffmpeg concatenate docs it says :

    


    


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

    


    


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

    


      

    • codec

      • 


    • bit rate

      • 


    • sample rate

      • 


    • channels (mono, stereo)

      • 


    


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

    


    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 :

    


    ffmpeg -i original.mp3
# > Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 128 kb/s

ffmpeg -i input.mp3 -b:a 128k -ar 44100 -ac 2 re_encoded.mp3

# then merge
# concat_list.txt contains the original audio and the re_encoded.mp3

ffmpeg -f concat -i concat_list.txt -safe 0 -c copy merged.mp3


    


    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.

    


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

    


    If I try to match it like below :

    


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


    


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

    


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

    


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


    


    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.

    


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

    


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


    


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

    


    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.

    


  • ffv1 : Drop unnecessary casts and const qualifiers to match function signatures

    18 décembre 2014, par Diego Biurrun
    ffv1 : Drop unnecessary casts and const qualifiers to match function signatures

    libavcodec/ffv1dec.c:898:36 : warning : cast discards ‘const’ qualifier from pointer target type
    • [DBH] libavcodec/ffv1dec.c

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