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  • Personnaliser en ajoutant son logo, sa bannière ou son image de fond

    5 septembre 2013, par

    Certains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;

  • Ecrire une actualité

    21 juin 2013, par

    Présentez les changements dans votre MédiaSPIP ou les actualités de vos projets sur votre MédiaSPIP grâce à la rubrique actualités.
    Dans le thème par défaut spipeo de MédiaSPIP, les actualités sont affichées en bas de la page principale sous les éditoriaux.
    Vous pouvez personnaliser le formulaire de création d’une actualité.
    Formulaire de création d’une actualité Dans le cas d’un document de type actualité, les champs proposés par défaut sont : Date de publication ( personnaliser la date de publication ) (...)

  • Le profil des utilisateurs

    12 avril 2011, par

    Chaque utilisateur dispose d’une page de profil lui permettant de modifier ses informations personnelle. Dans le menu de haut de page par défaut, un élément de menu est automatiquement créé à l’initialisation de MediaSPIP, visible uniquement si le visiteur est identifié sur le site.
    L’utilisateur a accès à la modification de profil depuis sa page auteur, un lien dans la navigation "Modifier votre profil" est (...)

Sur d’autres sites (6820)

  • SWF to Mov using FFMPEG

    3 juillet 2013, par RNJ

    I am trying to use ffmpeg to convert an swf file to a mov file. When this happens I only get the audio. If I do

    ffplay -stats mine.swf

    then I just get the audio and no video. I dont think the swf codec in installed. I find this confusing because it says that it is present in the installation according to the documentation.

    Does anyone know how I can do this ? Can I download another codec ?

    Thanks

  • 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.

  • ffmpeg for a android (using tutorial : "ffmpeg and Android.mk")

    7 avril 2016, par Matthias

    I am trying to compile ffmpeg for a android. I have found several posts on this theme but non of these seems to work. If tried to build ffmpeg like it is posted on [1]. Did anybody successfully compile ffmpeg using theses tutorial ?
    I am not sure how to realize step 4 to 5.

    STEP4 : Configuring ...

    STEP5 : cd to your NDK root dir, type make TARGET_ARCH=arm APP=ffmpeg-org

    It seems to me that building an application like it is explained in the tutorial in step 5 need some previous steps. Unfortunately I have no app in the folder to make. I am using the current android ndk release 3 and checked out the actual ffmpeg releases from [3] and [4]. I am thankful for every advice.

    [1] http://slworkthings.wordpress.com/
    [2] http://gitorious.org/ olvaffe/ffmpeg/ffmpeg-android
    [3] http://ffmpeg.org/download.html