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  • Participer à sa traduction

    10 avril 2011

    Vous pouvez nous aider à améliorer les locutions utilisées dans le logiciel ou à traduire celui-ci dans n’importe qu’elle nouvelle langue permettant sa diffusion à de nouvelles communautés linguistiques.
    Pour ce faire, on utilise l’interface de traduction de SPIP où l’ensemble des modules de langue de MediaSPIP sont à disposition. ll vous suffit de vous inscrire sur la liste de discussion des traducteurs pour demander plus d’informations.
    Actuellement MediaSPIP n’est disponible qu’en français et (...)

  • 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

Sur d’autres sites (10257)

  • Revision f82c61b886 : Merge "make DC_PRED for i4x4 to use real pixels only" into experimental

    22 avril 2013, par Paul Wilkins

    Merge "make DC_PRED for i4x4 to use real pixels only" into experimental

  • Revision 6ce718eb18 : Merge "End of orientation zero group experiment" into experimental

    22 avril 2013, par Deb Mukherjee

    Changed Paths : Modify /vp9/decoder/vp9_decodframe.c Modify /vp9/encoder/vp9_encodemb.c Modify /vp9/encoder/vp9_rdopt.c Merge "End of orientation zero group experiment" into experimental

  • 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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