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  • Support audio et vidéo HTML5

    10 avril 2011

    MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
    Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
    Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
    Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)

  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
    The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
    For older browsers the Flowplayer flash fallback is used.
    MediaSPIP allows for media playback on major mobile platforms with the above (...)

  • De l’upload à la vidéo finale [version standalone]

    31 janvier 2010, par

    Le chemin d’un document audio ou vidéo dans SPIPMotion est divisé en trois étapes distinctes.
    Upload et récupération d’informations de la vidéo source
    Dans un premier temps, il est nécessaire de créer un article SPIP et de lui joindre le document vidéo "source".
    Au moment où ce document est joint à l’article, deux actions supplémentaires au comportement normal sont exécutées : La récupération des informations techniques des flux audio et video du fichier ; La génération d’une vignette : extraction d’une (...)

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  • 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.

  • Revision 36889 : Le début d’une page d’info concernant la configuration de FFMPEG sur le ...

    3 avril 2010, par kent1@… — Log

    Le début d’une page d’info concernant la configuration de FFMPEG sur le serveur.
    On vire le PHP du squelette du formulaire de configuration
    On prépare le passage aux pressets

  • Revision 36891 : On liste les codecs gérés...

    3 avril 2010, par kent1@… — Log

    On liste les codecs gérés…

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