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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

  • Publier sur MédiaSpip

    13 juin 2013

    Puis-je poster des contenus à partir d’une tablette Ipad ?
    Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir

  • Encoding and processing into web-friendly formats

    13 avril 2011, par

    MediaSPIP automatically converts uploaded files to internet-compatible formats.
    Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
    Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
    Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
    All uploaded files are stored online in their original format, so you can (...)

Sur d’autres sites (9674)

  • Concatenation of files using ffmpeg does not work as I expected. Why ?

    19 décembre 2018, par Peter Olson

    I execute the following command line

    ffmpeg.exe
       -i C:\Beema\video-source\DO_U_BEEMA176x144short.avi
       -i C:\Beema\video-source\DO_U_BEEMA176x144short.avi
       -i C:\Beema\temp\9016730-51056331-stitcheds.avi  
       -i C:\Beema\video-source\GOTTA_BEEMA176x144short.avi
       -y -ac 1 -r 24 -b 25K
       C:\Beema\video-out\9a062fb6-d448-48fe-b006-a85d51adf8a1.mpg

    The output file in video-out ends up having a single copy of DO_U_BEEMA. I do not understand why ffmpeg is not concatenating.

    Any help is dramatically appreciated,

  • How the ffmpeg astats crest factor value of an audio track is calculated

    29 août 2017, par FranGar

    I’m scripting a ffmpeg chain process for my work. The aim is normalizing/compressing lot of audio files (mp3’s).
    It’s done in Python and the critical part is the line :

    ffmpeg -y -i "Input.mp3" -codec:a libmp3lame -b:a 96k -af acompressor=threshold=-15dB:ratio=5:attack=0.01:release=1000:knee=2,dynaudnorm=g=3:m=2:p=0.95 "Output.mp3"

    The python script it’s complete and working BUT the nature of the audios (voice recordings) are very different so I can’t use the same params for all of them.

    I make some experimenting with the values of the ffmpeg filter astats and i discovered that the crest factor (Standard ratio of peak to RMS level ) gave a good reference to programatically get the better params.

    In fact I saw that a recording with a nice dynamic range sound and smooth in shape, get crest values around 9-15 (the compress/normlz params will be somehow conservative). But audios with crest around 22-30 need more aggressive processing.
    (All empirically)

    Somebody can clarify how the crest values are really calculated ? Which are the peaks taken to account ? (Why the flat factor is always 0 ?)
    Or if somebody knows how to get a value representing the sound ’smoothness’ will be nice also.

    Thanks for the ideas.

  • How the ffmpeg astats crest factor of an audio track is calculated

    29 août 2017, par FranGar

    I’m scripting a ffmpeg chain process for my work. The aim is normalizing/compressing lot of audio files (mp3’s).
    It’s done in Python and the critical part is the line :

    ffmpeg -y -i "Input.mp3" -codec:a libmp3lame -b:a 96k -af acompressor=threshold=-15dB:ratio=5:attack=0.01:release=1000:knee=2,dynaudnorm=g=3:m=2:p=0.95 "Output.mp3"

    The python script it’s complete and working BUT the nature of the audios (voice recordings) are very different so I can’t use the same params for all of them.

    I make some experimenting with the values of the ffmpeg filter astats and i discovered that the crest factor (Standard ratio of peak to RMS level ) gave a good reference to programatically get the better params.

    In fact I saw that a recording with a nice dynamic range sound and smooth in shape, get crest values around 9-15 (the compress/normlz params will be somehow conservative). But audios with crest around 22-30 need more aggressive processing.
    (All empirically)

    Somebody can clarify how the crest values are really calculated ? Which are the peaks taken to account ? (Why the flat factor is always 0 ?)
    Or if somebody knows how to get a value representing the sound ’smoothness’ will be nice also.

    Thanks for the ideas.