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  • Pas question de marché, de cloud etc...

    10 avril 2011

    Le vocabulaire utilisé sur ce site essaie d’éviter toute référence à la mode qui fleurit allègrement
    sur le web 2.0 et dans les entreprises qui en vivent.
    Vous êtes donc invité à bannir l’utilisation des termes "Brand", "Cloud", "Marché" etc...
    Notre motivation est avant tout de créer un outil simple, accessible à pour tout le monde, favorisant
    le partage de créations sur Internet et permettant aux auteurs de garder une autonomie optimale.
    Aucun "contrat Gold ou Premium" n’est donc prévu, aucun (...)

  • Dépôt de média et thèmes par FTP

    31 mai 2013, par

    L’outil MédiaSPIP traite aussi les média transférés par la voie FTP. Si vous préférez déposer par cette voie, récupérez les identifiants d’accès vers votre site MédiaSPIP et utilisez votre client FTP favori.
    Vous trouverez dès le départ les dossiers suivants dans votre espace FTP : config/ : dossier de configuration du site IMG/ : dossier des média déjà traités et en ligne sur le site local/ : répertoire cache du site web themes/ : les thèmes ou les feuilles de style personnalisées tmp/ : dossier de travail (...)

  • Activation de l’inscription des visiteurs

    12 avril 2011, par

    Il est également possible d’activer l’inscription des visiteurs ce qui permettra à tout un chacun d’ouvrir soit même un compte sur le canal en question dans le cadre de projets ouverts par exemple.
    Pour ce faire, il suffit d’aller dans l’espace de configuration du site en choisissant le sous menus "Gestion des utilisateurs". Le premier formulaire visible correspond à cette fonctionnalité.
    Par défaut, MediaSPIP a créé lors de son initialisation un élément de menu dans le menu du haut de la page menant (...)

Sur d’autres sites (8874)

  • Convert video with ffmpeg to ogg theora [closed]

    19 juillet 2012, par Marcus

    I'm looking at this example http://blog.archive.org/2008/11/25/fast-and-reliable-way-to-encode-theora-ogg-videos-using-ffmpeg-libtheora-and-liboggz/ but it seems like ffmpeg now has support for converting videos to ogg theora out of the box. Does anyone have a detailed example of this ?

  • Objective-c. How to convert wav file to flac file ?

    23 juillet 2012, par user941313

    I have ffmpeg library and libFLAC, what library is better to use in xcode project and how to use ffmpeg for convert wav file to FLAC file ?

  • How to configure ffmpeg on ubuntu to convert *.3gp to pcm *.wav ? [migrated]

    31 juillet 2012, par Monica Sol

    I'm using linux Ubuntu ver 10.04.
    I need to convert file *.3gp to PCM *.wav. I'm using for that ffmpeg program.

    When it's installed from repository by using aptitude install ffmpeg it's installing some basic version of it and I cannot convert what I need.

    I've read some stuff on the Internet and I've made what there was written.
    I've installed the latest yasm ver.1.1.0 and the newest x264 - 0.125.2208. After that I got ffmpeg using git from http://ffmpeg.org/download.html (git clone git ://source.ffmpeg.org/ffmpeg.git ffmpeg).

    I`ve tried to configure ffmpeg by myself using :

    ./configure --enable-gpl --enable-version3 --enable-postproc 
    
--enable-nonfree --enable-swscale --enable-pthreads --enable-libmp3lame 
    
--enable-libx264 --enable-libopencore-amrnb --enable-libopencore-amrwb

    than : time make && make install.

    Till this time everything was ok. After conversion (ffmpeg -i audiotest.3gp -f s16le -ar 8000 -acodec pcm_s16le audio.wav) I wanted to check information about this PCM *.wav file (ffmpeg -i audio.wav) and I`ve got this error :

    ~# ffmpeg -i audio.wav
    

    
ffmpeg version N-42619-g6b7849e Copyright (c) 2000-2012 the FFmpeg developers
    
  built on Jul 21 2012 00:50:52 with gcc 4.4.3
    
  configuration: --enable-gpl --enable-version3 --enable-postproc --enable-nonfree --enable-swscale --enable-pthreads --enable-libmp3lame --enable-libx264 --enable-libopencore-amrnb --enable-libopencore-amrwb
    

    
  libavutil      51. 65.100 / 51. 65.100
    
  libavcodec     54. 41.100 / 54. 41.100
    
  libavformat    54. 17.100 / 54. 17.100
    
  libavdevice    54.  1.100 / 54.  1.100
    
  libavfilter     3.  2.100 /  3.  2.100
    
  libswscale      2.  1.100 /  2.  1.100
    
  libswresample   0. 15.100 /  0. 15.100
    
  libpostproc    52.  0.100 / 52.  0.100
    
[aac @ 0x943d4e0] Format aac detected only with low score of 1, misdetection possible!
    
[aac @ 0x9443740] channel element 0.0 is not allocated
    
    Last message repeated 2 times
    
[aac @ 0x9443740] More than one AAC RDB per ADTS frame is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
    
[aac @ 0x9443740] Input buffer exhausted before END element found
    
[aac @ 0x9443740] Number of bands (16) exceeds limit (4).
    
[aac @ 0x9443740] Number of bands (7) exceeds limit (2).
    
[aac @ 0x9443740] Input buffer exhausted before END element found
    
[aac @ 0x9443740] SSR not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
    
[aac @ 0x9443740] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list.
    
[aac @ 0x9443740] channel element 2.0 is not allocated
    
[aac @ 0x9443740] Error decoding AAC frame header.
    
[aac @ 0x9443740] Reserved bit set.
    
[aac @ 0x9443740] Invalid Predictor Reset Group.
    
[aac @ 0x9443740] channel element 0.0 is not allocated
    
[aac @ 0x9443740] Number of bands (31) exceeds limit (1).
    
[aac @ 0x9443740] Error decoding AAC frame header.
    
[aac @ 0x9443740] Input buffer exhausted before END element found
    
[aac @ 0x9443740] Number of bands (16) exceeds limit (2).
    
[aac @ 0x9443740] channel element 0.7 is not allocated
    
[aac @ 0x9443740] Invalid Predictor Reset Group.
    
[aac @ 0x9443740] Error decoding AAC frame header.
    
[aac @ 0x9443740] Number of scalefactor bands in group (62) exceeds limit (41).
    
[aac @ 0x9443740] Invalid Predictor Reset Group.
    
[aac @ 0x9443740] Input buffer exhausted before END element found
    
[aac @ 0x9443740] Invalid Predictor Reset Group.
    
[aac @ 0x9443740] channel element 0.2 is not allocated
    
[aac @ 0x9443740] Input buffer exhausted before END element found
    
[aac @ 0x9443740] Invalid Predictor Reset Group.
    
[aac @ 0x9443740] Reserved bit set.
    
[aac @ 0x9443740] Input buffer exhausted before END element found
    
[aac @ 0x9443740] channel element 0.15 is not allocated
    
[aac @ 0x9443740] Pulse data corrupt or invalid.
    
[aac @ 0x9443740] Number of scalefactor bands in group (48) exceeds limit (41).
    
[aac @ 0x9443740] channel element 2.0 is not allocated
    
[aac @ 0x9443740] Invalid Predictor Reset Group.
    
[aac @ 0x9443740] Reserved bit set.
    
[aac @ 0x9443740] Number of bands (16) exceeds limit (4).
    
[aac @ 0x9443740] Error decoding AAC frame header.
    
[aac @ 0x9443740] Input buffer exhausted before END element found
    
[aac @ 0x9443740] Error decoding AAC frame header.
    
[aac @ 0x9443740] Reserved bit set.
    
    Last message repeated 1 times
    
[aac @ 0x9443740] Error decoding AAC frame header.
    
[aac @ 0x9443740] channel element 2.0 is not allocated
    
[aac @ 0x9443740] Number of bands (31) exceeds limit (4).
    
[aac @ 0x9443740] Pulse data corrupt or invalid.
    
[aac @ 0x9443740] Reserved bit set.
    
[aac @ 0x9443740] Error decoding AAC frame header.
    
[aac @ 0x9443740] Reserved bit set.
    
[aac @ 0x9443740] SSR not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
    
[aac @ 0x9443740] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list.
    
[aac @ 0x9443740] Input buffer exhausted before END element found
    
[aac @ 0x9443740] channel element 0.0 is not allocated
    
[aac @ 0x9443740] Error decoding AAC frame header.
    
[aac @ 0x9443740] Reserved bit set.
    
[aac @ 0x9443740] Invalid Predictor Reset Group.
    
[aac @ 0x9443740] channel element 0.3 is not allocated
    
[aac @ 0x9443740] Pulse data corrupt or invalid.
    
[aac @ 0x9443740] Invalid Predictor Reset Group.
    
[aac @ 0x9443740] Input buffer exhausted before END element found
    
[aac @ 0x9443740] Number of bands (35) exceeds limit (16).
    
[aac @ 0x9443740] Number of scalefactor bands in group (63) exceeds limit (41).
    
[aac @ 0x9443740] Input buffer exhausted before END element found
    
[aac @ 0x9443740] channel element 0.0 is not allocated
    
[aac @ 0x9443740] Input buffer exhausted before END element found
    
[aac @ 0x9443740] Invalid Predictor Reset Group.
    
[aac @ 0x9443740] Reserved bit set.
    
[aac @ 0x9443740] Invalid Predictor Reset Group.
    
[aac @ 0x9443740] Reserved bit set.
    
[aac @ 0x9443740] Invalid Predictor Reset Group.
    
[aac @ 0x9443740] channel element 0.0 is not allocated
    
[aac @ 0x9443740] Number of bands (38) exceeds limit (10).
    
[aac @ 0x9443740] Error decoding AAC frame header.
    
[aac @ 0x9443740] Invalid Predictor Reset Group.
    
[aac @ 0x9443740] channel element 0.2 is not allocated
    
[aac @ 0x9443740] channel element 0.7 is not allocated
    
[aac @ 0x9443740] Reserved bit set.
    
    Last message repeated 2 times
    
[aac @ 0x9443740] channel element 0.2 is not allocated
    
[aac @ 0x9443740] Error decoding AAC frame header.
    
[aac @ 0x9443740] Reserved bit set.
    
    Last message repeated 1 times
    
[aac @ 0x9443740] Invalid Predictor Reset Group.
    
[aac @ 0x9443740] Input buffer exhausted before END element found
    
[aac @ 0x9443740] decode_band_types: Input buffer exhausted before END element found
    
[aac @ 0x9443740] Error decoding AAC frame header.
    
[aac @ 0x9443740] Reserved bit set.
    
[aac @ 0x9443740] Error decoding AAC frame header.
    
    Last message repeated 1 times
    
[aac @ 0x9443740] Reserved bit set.
    
    Last message repeated 1 times
    
[aac @ 0x9443740] Number of bands (4) exceeds limit (1).
    
[aac @ 0x9443740] Invalid Predictor Reset Group.
    
[aac @ 0x9443740] Reserved bit set.
    
[aac @ 0x9443740] Error decoding AAC frame header.
    
[aac @ 0x9443740] Number of bands (31) exceeds limit (8).
    
[aac @ 0x9443740] Invalid Predictor Reset Group.
    
[aac @ 0x9443740] Number of bands (31) exceeds limit (2).
    
[aac @ 0x9443740] Number of bands (28) exceeds limit (1).
    
[aac @ 0x9443740] channel element 0.0 is not allocated
    
[aac @ 0x9443740] Input buffer exhausted before END element found
    
[aac @ 0x9443740] Number of bands (16) exceeds limit (2).
    
[aac @ 0x9443740] Error decoding AAC frame header.
    
[aac @ 0x943d4e0] decoding for stream 0 failed
    
[aac @ 0x943d4e0] Could not find codec parameters for stream 0 (Audio: aac, 4.0, s16, 383 kb/s): unspecified sample rate
    
Consider increasing the value for the 'analyzeduration' and 'probesize' options
    
[aac @ 0x943d4e0] Estimating duration from bitrate, this may be inaccurate
    
audio.wav: could not find codec parameters

    Can anyone help me with this ? What I'm doing wrong ? I'm linux newbie, but I really need to get this thing works.

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