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  • 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 (...)

  • Multilang : améliorer l’interface pour les blocs multilingues

    18 février 2011, par

    Multilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
    Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela.

Sur d’autres sites (3640)

  • Révision 22931 : la fonction recuperer_url_cache sort un tableau plus complet que recuperer_lapage

    16 mars 2016, par erational@erational.org

  • ffmpeg cannot open a simple microsoft wav file exported with Audacity

    18 février 2014, par sebpiq

    I have exported a sound file to microsoft wav using Audacity.
    I am trying to open this file with ffmpeg :

    ffmpeg -i steps-stereo-16b-44khz.wav /tmp/test.ogg

    and here's the ouput I get :

    fmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers
    
  built on Jun 12 2013 13:46:11 with Apple clang version 4.1 (tags/Apple/clang-421.11.66) (based on LLVM 3.1svn)
    
  configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-libtheora --enable-libschroedinger --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libspeex --enable-libass --enable-libbluray --enable-gnutls --enable-libfreetype --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --arch=x86_64 --enable-yasm --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid
    
  libavutil      52. 18.100 / 52. 18.100
    
  libavcodec     54. 92.100 / 54. 92.100
    
  libavformat    54. 63.104 / 54. 63.104
    
  libavdevice    54.  3.103 / 54.  3.103
    
  libavfilter     3. 42.103 /  3. 42.103
    
  libswscale      2.  2.100 /  2.  2.100
    
  libswresample   0. 17.102 /  0. 17.102
    
  libpostproc    52.  2.100 / 52.  2.100
    
[dca @ 0x7fd30c013600] Not a valid DCA frame
    

    
... SNIP ...
    

    
[dca @ 0x7fd5bc013600] Invalid bit allocation index
    
[dca @ 0x7fd5bc013600] error decoding block
    
    Last message repeated 3 times
    
[dca @ 0x7fd5bc013600] Didn't get subframe DSYNC
    
[dca @ 0x7fd5bc013600] error decoding block
    
[wav @ 0x7fd5bc013000] max_analyze_duration 5000000 reached at 5009070 microseconds
    
[wav @ 0x7fd5bc013000] decoding for stream 0 failed
    
[wav @ 0x7fd5bc013000] Could not find codec parameters for stream 0 (Audio: dts ([1][0][0][0] / 0x0001), 192000 Hz, 2 channels, fltp, 0 kb/s): no decodable DTS frames
    
Consider increasing the value for the 'analyzeduration' and 'probesize' options
    
steps-stereo-16b-44khz.wav: could not find codec parameters

    If I export the same file to .ogg or .aiff, no problem, the following works fine :

    ffmpeg -i steps-stereo-16b-44khz.aiff /tmp/test.ogg

    Any idea what could be wrong ?

    A link to my wav file so you can try to reproduce.

    NB my final goal is to slice the audio file. I know I can export file directly to .ogg with audacity. This is just a test case.

    EDIT

    Getting file info with another program like sox, works well :

    sox --info steps-stereo-16b-44khz.wav
    

    
Input File     : 'steps-stereo-16b-44khz.wav'
    
Channels       : 2
    
Sample Rate    : 44100
    
Precision      : 16-bit
    
Duration       : 00:00:02.10 = 92608 samples = 157.497 CDDA sectors
    
File Size      : 370k
    
Bit Rate       : 1.41M
    
Sample Encoding: 16-bit Signed Integer PCM

  • ffmpef cannot open a simple microsoft wav file exported with Audacity

    23 juillet 2013, par sebpiq

    I have exported a sound file to microsoft wav using Audacity.
    I am trying to open this file with ffmpeg :

    ffmpeg -i steps-stereo-16b-44khz.wav /tmp/test.ogg

    and here's the ouput I get :

    fmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers
    
  built on Jun 12 2013 13:46:11 with Apple clang version 4.1 (tags/Apple/clang-421.11.66) (based on LLVM 3.1svn)
    
  configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-libtheora --enable-libschroedinger --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libspeex --enable-libass --enable-libbluray --enable-gnutls --enable-libfreetype --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --arch=x86_64 --enable-yasm --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid
    
  libavutil      52. 18.100 / 52. 18.100
    
  libavcodec     54. 92.100 / 54. 92.100
    
  libavformat    54. 63.104 / 54. 63.104
    
  libavdevice    54.  3.103 / 54.  3.103
    
  libavfilter     3. 42.103 /  3. 42.103
    
  libswscale      2.  2.100 /  2.  2.100
    
  libswresample   0. 17.102 /  0. 17.102
    
  libpostproc    52.  2.100 / 52.  2.100
    
[dca @ 0x7fd30c013600] Not a valid DCA frame
    

    
... SNIP ...
    

    
[dca @ 0x7fd5bc013600] Invalid bit allocation index
    
[dca @ 0x7fd5bc013600] error decoding block
    
    Last message repeated 3 times
    
[dca @ 0x7fd5bc013600] Didn't get subframe DSYNC
    
[dca @ 0x7fd5bc013600] error decoding block
    
[wav @ 0x7fd5bc013000] max_analyze_duration 5000000 reached at 5009070 microseconds
    
[wav @ 0x7fd5bc013000] decoding for stream 0 failed
    
[wav @ 0x7fd5bc013000] Could not find codec parameters for stream 0 (Audio: dts ([1][0][0][0] / 0x0001), 192000 Hz, 2 channels, fltp, 0 kb/s): no decodable DTS frames
    
Consider increasing the value for the 'analyzeduration' and 'probesize' options
    
steps-stereo-16b-44khz.wav: could not find codec parameters

    If I export the same file to .ogg or .aiff, no problem, the following works fine :

    ffmpeg -i steps-stereo-16b-44khz.aiff /tmp/test.ogg

    Any idea what could be wrong ?

    A link to my wav file so you can try to reproduce.

    NB my final goal is to slice the audio file. I know I can export file directly to .ogg with audacity. This is just a test case.

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