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  • Des sites réalisés avec MediaSPIP

    2 mai 2011, par

    Cette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
    Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page.

  • 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 (...)

  • De l’upload à la vidéo finale [version standalone]

    31 janvier 2010, par

    Le chemin d’un document audio ou vidéo dans SPIPMotion est divisé en trois étapes distinctes.
    Upload et récupération d’informations de la vidéo source
    Dans un premier temps, il est nécessaire de créer un article SPIP et de lui joindre le document vidéo "source".
    Au moment où ce document est joint à l’article, deux actions supplémentaires au comportement normal sont exécutées : La récupération des informations techniques des flux audio et video du fichier ; La génération d’une vignette : extraction d’une (...)

Sur d’autres sites (13082)

  • 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
  • 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 : Martin Storsjö <martin@martin.st>

    • [DBH] libavcodec/arm/mdct_vfp.S