Recherche avancée

Médias (91)

Autres articles (42)

  • La sauvegarde automatique de canaux SPIP

    1er avril 2010, par

    Dans le cadre de la mise en place d’une plateforme ouverte, il est important pour les hébergeurs de pouvoir disposer de sauvegardes assez régulières pour parer à tout problème éventuel.
    Pour réaliser cette tâche on se base sur deux plugins SPIP : Saveauto qui permet une sauvegarde régulière de la base de donnée sous la forme d’un dump mysql (utilisable dans phpmyadmin) mes_fichiers_2 qui permet de réaliser une archive au format zip des données importantes du site (les documents, les éléments (...)

  • Contribute to documentation

    13 avril 2011

    Documentation is vital to the development of improved technical capabilities.
    MediaSPIP welcomes documentation by users as well as developers - including : critique of existing features and functions articles contributed by developers, administrators, content producers and editors screenshots to illustrate the above translations of existing documentation into other languages
    To contribute, register to the project users’ mailing (...)

  • Use, discuss, criticize

    13 avril 2011, par

    Talk to people directly involved in MediaSPIP’s development, or to people around you who could use MediaSPIP to share, enhance or develop their creative projects.
    The bigger the community, the more MediaSPIP’s potential will be explored and the faster the software will evolve.
    A discussion list is available for all exchanges between users.

Sur d’autres sites (2167)

  • 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>

    • [DH] libavcodec/arm/fft_init_arm.c
    • [DH] libavcodec/arm/fft_vfp.S
  • checkasm : Add vc1dsp inverse transform tests

    31 mars 2022, par Ben Avison
    checkasm : Add vc1dsp inverse transform tests
    

    This test deliberately doesn't exercise the full range of inputs described in
    the committee draft VC-1 standard. It says :

    input coefficients in frequency domain, D, satisfy -2048 <= D < 2047
    intermediate coefficients, E, satisfy -4096 <= E < 4095
    fully inverse-transformed coefficients, R, satisfy -512 <= R < 511

    For one thing, the inequalities look odd. Did they mean them to go the
    other way round ? That would make more sense because the equations generally
    both add and subtract coefficients multiplied by constants, including powers
    of 2. Requiring the most-negative values to be valid extends the number of
    bits to represent the intermediate values just for the sake of that one case !

    For another thing, the extreme values don't look to occur in real streams -
    both in my experience and supported by the following comment in the AArch32
    decoder :

    tNhalf is half of the value of tN (as described in vc1_inv_trans_8x8_c).
    This is done because sometimes files have input that causes tN + tM to
    overflow. To avoid this overflow, we compute tNhalf, then compute
    tNhalf + tM (which doesn't overflow), and then we use vhadd to compute
    (tNhalf + (tNhalf + tM)) >> 1 which does not overflow because it is
    one instruction.

    My AArch64 decoder goes further than this. It calculates tNhalf and tM
    then does an SRA (essentially a fused halve and add) to compute
    (tN + tM) >> 1 without ever having to hold (tNhalf + tM) in a 16-bit element
    without overflowing. It only encounters difficulties if either tNhalf or
    tM overflow in isolation.

    I haven't had sight of the final standard, so it's possible that these
    issues were dealt with during finalisation, which could explain the lack
    of usage of extreme inputs in real streams. Or a preponderance of decoders
    that only support 16-bit intermediate values in their inverse transforms
    might have caused encoders to steer clear of such cases.

    I have effectively followed this approach in the test, and limited the
    scale of the coefficients sufficient that both the existing AArch32 decoder
    and my new AArch64 decoder both pass.

    Signed-off-by : Ben Avison <bavison@riscosopen.org>
    Signed-off-by : Martin Storsjö <martin@martin.st>

    • [DH] tests/checkasm/vc1dsp.c
  • lavf doxy : add some general lavf information.

    11 décembre 2011, par Anton Khirnov

    lavf doxy : add some general lavf information.