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  • Qu’est ce qu’un masque de formulaire

    13 juin 2013, par

    Un masque de formulaire consiste en la personnalisation du formulaire de mise en ligne des médias, rubriques, actualités, éditoriaux et liens vers des sites.
    Chaque formulaire de publication d’objet peut donc être personnalisé.
    Pour accéder à la personnalisation des champs de formulaires, il est nécessaire d’aller dans l’administration de votre MediaSPIP puis de sélectionner "Configuration des masques de formulaires".
    Sélectionnez ensuite le formulaire à modifier en cliquant sur sont type d’objet. (...)

  • Keeping control of your media in your hands

    13 avril 2011, par

    The vocabulary used on this site and around MediaSPIP in general, aims to avoid reference to Web 2.0 and the companies that profit from media-sharing.
    While using MediaSPIP, you are invited to avoid using words like "Brand", "Cloud" and "Market".
    MediaSPIP is designed to facilitate the sharing of creative media online, while allowing authors to retain complete control of their work.
    MediaSPIP aims to be accessible to as many people as possible and development is based on expanding the (...)

  • 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 (...)

Sur d’autres sites (3908)

  • 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.

  • 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
  • ImgFlip Gif Generator

    30 juillet 2013, par user1960364

    Reference : http://imgflip.com/gifgenerator

    So, I don't need the web UI and all the cusomizations. But I've been trying to figure out how to slice and convert a mp4 or mov to an animated gif automagically. I've looked at several ways of doing it with imagemagick/ffmpeg but none of them rendered results as good as imgflip. I always end up with some 24bit color-looking gif instead or some impractically large file.

    Any suggestions/ideas on how to do it as efficiently as imgflip ?

    Thanks :)

    — Edit —
    I found this on reddit :

    #!/bin/bash

    # Extracts each frame of the video as a single gif
    ffmpeg -i $1 out%04d.gif

    # Combines all the frames into one very nicely animated gif.
    convert -delay 4 out*.gif anim.gif

    # Optimizes the gif using imagemagick
    convert -layers Optimize anim.gif optimized_output.gif

    # Cleans up the leftovers
    rm out*
    rm anim.gif

    Using that, I created (and yes, I'm on windows) :

    ffmpeg -i v.mp4 -r 10 -q:v 1 tmp/out%04d.jpg
    convert -delay 10 tmp/out*.jpg jpg-d10.gif
    echo y | del tmp

    And got decent quality, even this creates an 8MB file for a small 6sec 480x480 clip at 10fps. And because it's a video and not vector or series of images with very few colors, extracting jpg images does better compression than -layers Optimize. However, for some reason, -qscale/-q:v on the frame extraction does not effect the final file size, just the quality.