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Autres articles (62)

  • 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

  • La sauvegarde automatique de canaux SPIP

    1er avril 2010, par

    Dans le cadre de la mise en place d’une plateforme ouverte, il est important pour les hébergeurs de pouvoir disposer de sauvegardes assez régulières pour parer à tout problème éventuel.
    Pour réaliser cette tâche on se base sur deux plugins SPIP : Saveauto qui permet une sauvegarde régulière de la base de donnée sous la forme d’un dump mysql (utilisable dans phpmyadmin) mes_fichiers_2 qui permet de réaliser une archive au format zip des données importantes du site (les documents, les éléments (...)

  • Gestion de la ferme

    2 mars 2010, par

    La ferme est gérée dans son ensemble par des "super admins".
    Certains réglages peuvent être fais afin de réguler les besoins des différents canaux.
    Dans un premier temps il utilise le plugin "Gestion de mutualisation"

Sur d’autres sites (7898)

  • 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.

  • h264 : convert loop filter strength dsp function to yasm.

    28 juillet 2012, par Ronald S. Bultje

    h264 : convert loop filter strength dsp function to yasm.

  • How to convert mp3 to flac [migrated]

    4 août 2012, par Cupidvogel

    Can someone tell me how to convert a mp3 audio file to flac format using ffmpeg ? I Googled extensively, but almost all resources point to converting flac to mp3, something like ffmpeg -i "input.flac" -ab 320k -map_meta_data 0:0 "output.mp3".

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