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  • Modifier la date de publication

    21 juin 2013, par

    Comment changer la date de publication d’un média ?
    Il faut au préalable rajouter un champ "Date de publication" dans le masque de formulaire adéquat :
    Administrer > Configuration des masques de formulaires > Sélectionner "Un média"
    Dans la rubrique "Champs à ajouter, cocher "Date de publication "
    Cliquer en bas de la page sur Enregistrer

  • 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

  • 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 (...)

Sur d’autres sites (5699)

  • avcodec/huffman/ff_huff_gen_len_table : support skiping stat=0 entries

    9 juin 2014, par Michael Niedermayer
    avcodec/huffman/ff_huff_gen_len_table : support skiping stat=0 entries
    

    This is probably not the simplest solution but as this is needed for a bugfix,
    simplification is left for later.

    Signed-off-by : Michael Niedermayer <michaelni@gmx.at>

    • [DH] libavcodec/huffman.c
    • [DH] libavcodec/huffman.h
    • [DH] libavcodec/huffyuvenc.c
    • [DH] libavcodec/utvideoenc.c
  • armv6 : Accelerate ff_imdct_half for general case (mdct_bits != 6)

    11 juillet 2014, par Ben Avison
    armv6 : Accelerate ff_imdct_half for general case (mdct_bits != 6)
    

    The previous implementation targeted DTS Coherent Acoustics, which only
    requires mdct_bits == 6. This relatively small size lent itself to
    unrolling the loops a small number of times, and encoding offsets
    calculated at assembly time within the load/store instructions of each
    iteration.

    In the more general case (codecs such as AAC and AC3) much larger arrays
    are used - mdct_bits == [8, 9, 11]. The old method does not scale for
    these cases, so more integer registers are used with non-unrolled versions
    of the loops (and with some stack spillage). The postrotation filter loop
    is still unrolled by a factor of 2 to permit the double-buffering of some
    VFP registers to facilitate overlap of neighbouring iterations.

    I benchmarked the result by measuring the number of gperftools samples
    that hit anywhere in the AAC decoder (starting from aac_decode_frame())
    or specifically in ff_imdct_half_c / ff_imdct_half_vfp, for the same
    example AAC stream :

    Before After
    Mean StdDev Mean StdDev Confidence Change
    aac_decode_frame 2368.1 35.8 2117.2 35.3 100.0% +11.8%
    ff_imdct_half_* 457.5 22.4 251.2 16.2 100.0% +82.1%

    Signed-off-by : Michael Niedermayer <michaelni@gmx.at>

    • [DH] libavcodec/arm/mdct_vfp.S
  • armv6 : Accelerate ff_fft_calc for general case (nbits != 4)

    16 juillet 2014, par Ben Avison
    armv6 : Accelerate ff_fft_calc for general case (nbits != 4)
    

    The previous implementation targeted DTS Coherent Acoustics, which only
    requires nbits == 4 (fft16()). This case was (and still is) linked directly
    rather than being indirected through ff_fft_calc_vfp(), but now the full
    range from radix-4 up to radix-65536 is available. This benefits other codecs
    such as AAC and AC3.

    The implementaion is based upon the C version, with each routine larger than
    radix-16 calling a hierarchy of smaller FFT functions, then performing a
    post-processing pass. This pass benefits a lot from loop unrolling to
    counter the long pipelines in the VFP. A relaxed calling standard also
    reduces the overhead of the call hierarchy, and avoiding the excessive
    inlining performed by GCC probably helps with I-cache utilisation too.

    I benchmarked the result by measuring the number of gperftools samples that
    hit anywhere in the AAC decoder (starting from aac_decode_frame()) or
    specifically in the FFT routines (fft4() to fft512() and pass()) for the
    same sample AAC stream :

    Before After
    Mean StdDev Mean StdDev Confidence Change
    Audio decode 2245.5 53.1 1599.6 43.8 100.0% +40.4%
    FFT routines 940.6 22.0 348.1 20.8 100.0% +170.2%

    Signed-off-by : Martin Storsjö <martin@martin.st>

    • [DBH] libavcodec/arm/fft_init_arm.c
    • [DBH] libavcodec/arm/fft_vfp.S