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    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 ;

  • 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

  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
    The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
    For older browsers the Flowplayer flash fallback is used.
    MediaSPIP allows for media playback on major mobile platforms with the above (...)

Sur d’autres sites (15639)

  • Merge commit ’528daa399018af74d52192eb1861d2b59d256111’

    28 mai 2013, par Michael Niedermayer
    Merge commit ’528daa399018af74d52192eb1861d2b59d256111’

    * commit ’528daa399018af74d52192eb1861d2b59d256111’ :
    
  wavpack : switch to planar output
    

    Merged-by : Michael Niedermayer <michaelni@gmx.at>
    • [DH] libavcodec/wavpack.c

  • af_asyncts : fix offset calculation

    8 mai 2013, par Anton Khirnov
    af_asyncts : fix offset calculation

    delta is in samples, not bytes. Also the sample format is not guaranteed
    
to be planar.
    

    CC:libav-stable@libav.org
    • [DBH] libavfilter/af_asyncts.c

  • Audio decoding using ffms2(ffmegsource)

    1er mai 2013, par praks411

    I'm using ffms2(ffmpegsource) a wrapper around libav to get video and audio frame from a file.
    Video decoding is working fine. However I'm facing some issues with audio decoding.
    FFMS2 provide a simple function FFMS_GetAudio(FFMS_AudioSource *A, void *Buf, int64_t Start, int64_t Count, FFMS_ErrorInfo *ErrorInfo); api to get the decoded buffer. The decoded data is return in buffer provided by user.

    For single channel the data is interpretation is straight forward with data byte starting from first location of user buffer. However when it comes to two channel there are two possibilities the decoded data could be planar or interleaved depending upon sample format return by FFMS_GetAudioProperties. In my case the sample format is always planar which means that decoded data will in two sperate data plane data[0] and data[1]. And this is what is explained by libav/ffmpeg and also by portaudio which consider planar data to be in two separate data plane.

    However FFMS_GetAudio just take single buffer from user. So can I assume for planar data
    data[0] = buf, data[1] = buf + offset, where offset is half the length of buffer return by FFMS_GetAudio.

    FFMS does not provide any good document for this interpretation. It would be great help if some can provide more information on this.

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