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  • Les tâches Cron régulières de la ferme

    1er décembre 2010, par

    La gestion de la ferme passe par l’exécution à intervalle régulier de plusieurs tâches répétitives dites Cron.
    Le super Cron (gestion_mutu_super_cron)
    Cette tâche, planifiée chaque minute, a pour simple effet d’appeler le Cron de l’ensemble des instances de la mutualisation régulièrement. Couplée avec un Cron système sur le site central de la mutualisation, cela permet de simplement générer des visites régulières sur les différents sites et éviter que les tâches des sites peu visités soient trop (...)

  • Keeping control of your media in your hands

    13 avril 2011, par

    The vocabulary used on this site and around MediaSPIP in general, aims to avoid reference to Web 2.0 and the companies that profit from media-sharing.
    While using MediaSPIP, you are invited to avoid using words like "Brand", "Cloud" and "Market".
    MediaSPIP is designed to facilitate the sharing of creative media online, while allowing authors to retain complete control of their work.
    MediaSPIP aims to be accessible to as many people as possible and development is based on expanding the (...)

  • Le profil des utilisateurs

    12 avril 2011, par

    Chaque utilisateur dispose d’une page de profil lui permettant de modifier ses informations personnelle. Dans le menu de haut de page par défaut, un élément de menu est automatiquement créé à l’initialisation de MediaSPIP, visible uniquement si le visiteur est identifié sur le site.
    L’utilisateur a accès à la modification de profil depuis sa page auteur, un lien dans la navigation "Modifier votre profil" est (...)

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  • how to automatically oversample and downsample when using filters ?

    4 juin 2021, par Yue Wang

    I am researching avfilters. Say I have an audio with sample rate s, and bit depth d,

    


    What I want to do is to write a graph that

    


      

    • upsample s by 4x, and set precision to 64bit float
    • 


    • apply some biquad filters with 64bit precision
    • 


    • downsample by 4x back to s, and set bit depth back to d.
    • 


    


    The reason to oversample is to get better filtering result by antialiasing, and the reason to downsample is to stream using the original source format.

    


    I don't know if there's way that I can do it automatically in the graph.

    


    ashowinfo could print out the sample rate, but seems there's no way to use the value later in the pipeline.

    


    asoftclip has a oversample factor. but it's not available in other filters.

    


  • bad audio mic recording quality with ffmpeg compared to sox

    1er juillet 2021, par user2355330

    I am contacting you as after 3 days of searching I am stuck on a really simple point.

    


    I want to record the sound of my mic on MacOS using ffmpeg.

    


    I managed to do it using the following command :

    


    ffmpeg -f avfoundation -audio_device_index 2 -i "none:-" -c:a pcm_s32l alexspeaking.wav -y -loglevel debug


    


    The issue is that each time I am speaking, there are cracks and pop in the sound...

    


    I tried to use sox and it gave me a perfect and crystal clear sound and I have no idea why... Below is the output of the sox command :

    


    sox -t coreaudio "G935 Gaming Headset" toto.wav -V6
sox:      SoX v
time:     Nov 15 2020 01:06:02
uname:    Darwin MacBook-Pro.local 20.5.0 Darwin Kernel Version 20.5.0: Sat May  8 05:10:33 PDT 2021; root:xnu-7195.121.3~9/RELEASE_X86_64 x86_64
compiler: gcc Apple LLVM 12.0.0 (clang-1200.0.32.27)
arch:     1288 48 88 L
sox INFO coreaudio: Found Audio Device "DELL U2721DE"
sox INFO coreaudio: Found Audio Device "G935 Gaming "
sox DBUG coreaudio: audio device did not accept 2 channels. Use 1 channels instead.
sox DBUG coreaudio: audio device did not accept 44100 sample rate. Use 48000 instead.
Input File     : 'G935 Gaming Headset' (coreaudio)
Channels       : 1
Sample Rate    : 48000
Precision      : 32-bit
Sample Encoding: 32-bit Signed Integer PCM
Endian Type    : little
Reverse Nibbles: no
Reverse Bits   : no
sox INFO sox: Overwriting `toto.wav'
sox DBUG wav: Writing Wave file: Microsoft PCM format, 1 channel, 48000 samp/sec
sox DBUG wav:         192000 byte/sec, 4 block align, 32 bits/samp
Output File    : 'toto.wav'
Channels       : 1
Sample Rate    : 48000
Precision      : 32-bit
Sample Encoding: 32-bit Signed Integer PCM
Endian Type    : little
Reverse Nibbles: no
Reverse Bits   : no
Comment        : 'Processed by SoX'
sox DBUG effects: sox_add_effect: extending effects table, new size = 8
sox INFO sox: effects chain: input        48000Hz  1 channels (multi) 32 bits unknown length
sox INFO sox: effects chain: output       48000Hz  1 channels (multi) 32 bits unknown length
sox DBUG sox: start-up time = 0.051332
In:0.00% 00:00:07.13 [00:00:00.00] Out:340k  [      |      ]        Clip:0    ^C
sox DBUG input: output buffer still held 2048 samples; dropped.
Aborted.
sox DBUG wav: Finished writing Wave file, 1359872 data bytes 339968 samples


    


    I am pretty sure the issue is linked to the way the encoding is done and the params I used with ffmpeg but I don't seem to be able to grasp which one I must use.

    


    Any ideas if there are ffmpeg experts here ?

    


  • libavcodec/libx265 : add user data unregistered SEI encoding

    12 juillet 2021, par Brad Hards
    libavcodec/libx265 : add user data unregistered SEI encoding
    

    MISB ST 0604 and ST 2101 require user data unregistered SEI messages
    (precision timestamps and sensor identifiers) to be included. That
    currently isn't supported for libx265. This patch adds support
    for user data unregistered SEI messages in accordance with
    ISO/IEC 23008-2:2020 Section D.2.7

    The design is based on nvenc, with support finished up at
    57de80673cb

    Signed-off-by : Derek Buitenhuis <derek.buitenhuis@gmail.com>

    • [DH] libavcodec/libx265.c