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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 ) (...)

  • Les vidéos

    21 avril 2011, par

    Comme les documents de type "audio", Mediaspip affiche dans la mesure du possible les vidéos grâce à la balise html5 .
    Un des inconvénients de cette balise est qu’elle n’est pas reconnue correctement par certains navigateurs (Internet Explorer pour ne pas le nommer) et que chaque navigateur ne gère en natif que certains formats de vidéos.
    Son avantage principal quant à lui est de bénéficier de la prise en charge native de vidéos dans les navigateur et donc de se passer de l’utilisation de Flash et (...)

Sur d’autres sites (10034)

  • aaccoder : Implement Perceptual Noise Substitution for AAC

    15 avril 2015, par Rostislav Pehlivanov
    aaccoder : Implement Perceptual Noise Substitution for AAC
    

    This commit implements the perceptual noise substitution AAC extension. This is a proof of concept
    implementation, and as such, is not enabled by default. This is the fourth revision of this patch,
    made after some problems were noted out. Any changes made since the previous revisions have been indicated.

    In order to extend the encoder to use an additional codebook, the array holding each codebook has been
    modified with two additional entries - 13 for the NOISE_BT codebook and 12 which has a placeholder function.
    The cost system was modified to skip the 12th entry using an array to map the input and outputs it has. It
    also does not accept using the 13th codebook for any band which is not marked as containing noise, thereby
    restricting its ability to arbitrarily choose it for bands. The use of arrays allows the system to be easily
    extended to allow for intensity stereo encoding, which uses additional codebooks.

    The 12th entry in the codebook function array points to a function which stops the execution of the program
    by calling an assert with an always ’false’ argument. It was pointed out in an email discussion with
    Claudio Freire that having a ’NULL’ entry can result in unexpected behaviour and could be used as
    a security hole. There is no danger of this function being called during encoding due to the codebook maps introduced.

    Another change from version 1 of the patch is the addition of an argument to the encoder, ’-aac_pns’ to
    enable and disable the PNS. This currently defaults to disable the PNS, as it is experimental.
    The switch will be removed in the future, when the algorithm to select noise bands has been improved.
    The current algorithm simply compares the energy to the threshold (multiplied by a constant) to determine
    noise, however the FFPsyBand structure contains other useful figures to determine which bands carry noise more accurately.

    Some of the sample files provided triggered an assertion when the parameter to tune the threshold was set to
    a value of ’2.2’. Claudio Freire reported the problem’s source could be in the range of the scalefactor
    indices for noise and advised to measure the minimal index and clip anything above the maximum allowed
    value. This has been implemented and all the files which used to trigger the asserion now encode without error.

    The third revision of the problem also removes unneded variabes and comparisons. All of them were
    redundant and were of little use for when the PNS implementation would be extended.

    The fourth revision moved the clipping of the noise scalefactors outside the second loop of the two-loop
    algorithm in order to prevent their redundant calculations. Also, freq_mult has been changed to a float
    variable due to the fact that rounding errors can prove to be a problem at low frequencies.
    Considerations were taken whether the entire expression could be evaluated inside the expression
    , but in the end it was decided that it would be for the best if just the type of the variable were
    to change. Claudio Freire reported the two problems. There is no change of functionality
    (except for low sampling frequencies) so the spectral demonstrations at the end of this commit’s message were not updated.

    Finally, the way energy values are converted to scalefactor indices has changed since the first commit,
    as per the suggestion of Claudio Freire. This may still have some drawbacks, but unlike the first commit
    it works without having redundant offsets and outputs what the decoder expects to have, in terms of the
    ranges of the scalefactor indices.

    Some spectral comparisons : https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/Original.png (original),
    https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS_NO.png (encoded without PNS),
    https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS1.2.png (encoded with PNS, const = 1.2),
    https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/Difference1.png (spectral difference).
    The constant is the value which multiplies the threshold when it gets compared to the energy, larger
    values means more noise will be substituded by PNS values. Example when const = 2.2 :
    https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS_2.2.png

    Reviewed-by : Claudio Freire <klaussfreire@gmail.com>
    Signed-off-by : Michael Niedermayer <michaelni@gmx.at>

    • [DH] libavcodec/aaccoder.c
    • [DH] libavcodec/aacenc.c
    • [DH] libavcodec/aacenc.h
  • aac : Correctly map multichannel ADTS AAC with non-zero channel_config + PCE

    3 juin 2015, par nu774
    aac : Correctly map multichannel ADTS AAC with non-zero channel_config + PCE
    

    The decoder assigns channels using default channel configuration
    for 5.1ch when it parses an ADTS frame header using consecutive
    channel ids.

    When a PCE comes, it reassigns channels using PCE configuration
    using directly the ids provided. They can be arbitrary.

    Always use consecutive channel ids to avoid decoding glitches due
    spurious reconfigurations due the channel ids mismatch between the
    two otherwise-identical channel maps.

    Signed-off-by : Luca Barbato <lu_zero@gentoo.org>

    • [DH] libavcodec/aacdec.c
  • Streams mixed when using -filter_complex amerge in FFmpeg

    19 juin 2016, par BrainStone

    I am currently having issues with ffmpeg and one of its filters.

    I am trying to merge 2 audio streams of a video into one. for this purpose I tried this command :

    ffmpeg -i /home/maniaplanet/Videos/ManiaPlanet\ 2014-08-21\ 20-09-13-082.avi.output.mkv -filter_complex "[0:1][0:2] amerge=inputs=2"-c:v copy -c:a libvo_aacenc -b:a 256k /var/www/files/output.mp4

    But I get this output :

    ffmpeg version 1.0.10 Copyright (c) 2000-2014 the FFmpeg developers
     built on Jul 25 2014 07:50:40 with gcc 4.7 (Debian 4.7.2-5)
     configuration: --prefix=/usr --extra-cflags='-g -O2 -fstack-protector --param=ssp-buffer-size=4 -Wformat -Werror=format-security ' --extra-ldflags='-Wl,-z,relro' --cc='ccache cc' --enable-shared --enable-libmp3lame --enable-gpl --enable-nonfree --enable-libvorbis --enable-pthreads --enable-libfaac --enable-libxvid --enable-postproc --enable-x11grab --enable-libgsm --enable-libtheora --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-libspeex --enable-nonfree --disable-stripping --enable-libvpx --enable-libschroedinger --disable-encoder=libschroedinger --enable-version3 --enable-libopenjpeg --enable-librtmp --enable-avfilter --enable-libfreetype --enable-libvo-aacenc --disable-decoder=amrnb --enable-libvo-amrwbenc --enable-libaacplus --libdir=/usr/lib/x86_64-linux-gnu --disable-vda --enable-libbluray --enable-libcdio --enable-gnutls --enable-frei0r --enable-openssl --enable-libass --enable-libopus --enable-fontconfig --enable-libfdk-aac --enable-libdc1394 --disable-altivec --dis  libavutil      51. 73.101 / 51. 73.101
     libavcodec     54. 59.100 / 54. 59.100
     libavformat    54. 29.104 / 54. 29.104
     libavdevice    54.  2.101 / 54.  2.101
     libavfilter     3. 17.100 /  3. 17.100
     libswscale      2.  1.101 /  2.  1.101
     libswresample   0. 15.100 /  0. 15.100
     libpostproc    52.  0.100 / 52.  0.100
    Input #0, matroska,webm, from '/home/maniaplanet/Videos/ManiaPlanet 2014-08-21 20-09-13-082.avi.output.mkv':
     Metadata:
       ISRC            : Video:RGB24 Audio0:Headset (2- Plantronics .Audio 655 DSP) Audio1:Headset (2- Plantronics .Audio 655 DSP)
       ENCODER         : Lavf55.37.100
     Duration: 01:49:48.47, start: 0.000000, bitrate: 3867 kb/s
       Stream #0:0: Video: h264 (High), yuv420p, 1280x1024, SAR 1:1 DAR 5:4, 30 fps, 30 tbr, 1k tbn, 60 tbc (default)
       Stream #0:1: Audio: mp3, 48000 Hz, stereo, s16, 320 kb/s (default)
       Stream #0:2: Audio: mp3, 48000 Hz, stereo, s16, 320 kb/s (default)
    File '/var/www/files/output.mp4' already exists. Overwrite ? [y/N] y
    Input channel layouts overlap: output layout will be determined by the number of distinct input channels
    [libvo_aacenc @ 0x7ae800] Unable to set encoding parameters
    Output #0, mp4, to '/var/www/files/output.mp4':
     Metadata:
       ISRC            : Video:RGB24 Audio0:Headset (2- Plantronics .Audio 655 DSP) Audio1:Headset (2- Plantronics .Audio 655 DSP)
       ENCODER         : Lavf55.37.100
       Stream #0:0: Audio: aac, 48000 Hz, 4.0, s16, 256 kb/s
       Stream #0:1: Video: h264, yuv420p, 1280x1024 [SAR 1:1 DAR 5:4], q=2-31, 30 fps, 90k tbn, 1k tbc (default)
    Stream mapping:
     Stream #0:1 (mp3) -> amerge:in0
     Stream #0:2 (mp3) -> amerge:in1
     amerge -> Stream #0:0 (libvo_aacenc)
     Stream #0:0 -> #0:1 (copy)
    Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height

    I think the important part is at the bottom :

    Stream mapping:
     Stream #0:1 (mp3) -> amerge:in0
     Stream #0:2 (mp3) -> amerge:in1
     amerge -> Stream #0:0 (libvo_aacenc)
     Stream #0:0 -> #0:1 (copy)

    It maps the video stream as the second stream and the audio gets first. How do i resolve this ? -map did not help. (Maybe I just used it wrong)