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Autres articles (40)

  • Contribute to a better visual interface

    13 avril 2011

    MediaSPIP is based on a system of themes and templates. Templates define the placement of information on the page, and can be adapted to a wide range of uses. Themes define the overall graphic appearance of the site.
    Anyone can submit a new graphic theme or template and make it available to the MediaSPIP community.

  • Contribute to translation

    13 avril 2011

    You can help us to improve the language used in the software interface to make MediaSPIP more accessible and user-friendly. You can also translate the interface into any language that allows it to spread to new linguistic communities.
    To do this, we use the translation interface of SPIP where the all the language modules of MediaSPIP are available. Just subscribe to the mailing list and request further informantion on translation.
    MediaSPIP is currently available in French and English (...)

  • Le plugin : Gestion de la mutualisation

    2 mars 2010, par

    Le plugin de Gestion de mutualisation permet de gérer les différents canaux de mediaspip depuis un site maître. Il a pour but de fournir une solution pure SPIP afin de remplacer cette ancienne solution.
    Installation basique
    On installe les fichiers de SPIP sur le serveur.
    On ajoute ensuite le plugin "mutualisation" à la racine du site comme décrit ici.
    On customise le fichier mes_options.php central comme on le souhaite. Voilà pour l’exemple celui de la plateforme mediaspip.net :
    < ?php (...)

Sur d’autres sites (3536)

  • x264vfw - is the H.264/MPEG-4 AVC codec advisable to use ?

    26 avril 2012, par Dhaval Kariya

    I have generated a graph using the GraphEdit utility of Directshow.

    My Graph is as follow :

    osprey card -> x264vfw - H.264/MPEG-4 AVC codec -> GDCL Multiplexer -> File writer (.mp4    
    file)

    I used x264vfw - H.264/MPEG-4 AVC codec, downloaded from
    http://sourceforge.net/projects/x264vfw/.

    I want to know if it is advisable to use this codec to encode the live video streams ?
    And what about the support and licensing for it ?

  • Revision 2b59e14a00 : Parse out arm isa targets from dumpmachine The current parsing logic of the dum

    15 août 2012, par Mike Frysinger

    Changed Paths : Modify /build/make/configure.sh Parse out arm isa targets from dumpmachine The current parsing logic of the dumpmachine tuple lacks any arm cases which means tgt_isa never gets set, so for all arm targets, we get detected as generic-gnu. Add some basic arm checks here so the (...)

  • FFMPEG update from 5.0 to 6.0 out_0_0 buffer queued [closed]

    16 mai 2023, par Kevitto

    I've been using ffmpeg 5.0 for some time, encoding an audio stream to an rtp server, but since I updated to ffmpeg 6.0 I get this :

    &#xA;

    [out_0_0 @ 0x55ac187b60] 100 buffers queued in out_0_0, something may be wrong.&#xA;

    &#xA;

    Below is the ffmpeg call :

    &#xA;

    ffmpeg -re -f alsa -i default:CARD:card1 -ac 2 -af aresample=async=1 -acodec libopus -b:a 48000 -f rtp "rtp://127.0.0.1:5002"&#xA;

    &#xA;

    And here is the full startup log :

    &#xA;

    ffmpeg version 549430e Copyright (c) 2000-2023 the FFmpeg developers&#xA;  built with gcc 10 (Debian 10.2.1-6)&#xA;  configuration: --extra-cflags=-I/usr/local/include --extra-ldflags=-L/usr/local/lib --extra-libs=&#x27;-lpthread -lm -latomic&#x27; --arch=arm64 --enable-gmp --enable-gpl --enable-libopus --enable-nonfree --enable-version3 --target-os=linux --enable-pthreads --enable-openssl --enable-hardcoded-tables&#xA;  libavutil      58.  2.100 / 58.  2.100&#xA;  libavcodec     60.  3.100 / 60.  3.100&#xA;  libavformat    60.  3.100 / 60.  3.100&#xA;  libavdevice    60.  1.100 / 60.  1.100&#xA;  libavfilter     9.  3.100 /  9.  3.100&#xA;  libswscale      7.  1.100 /  7.  1.100&#xA;  libswresample   4. 10.100 /  4. 10.100&#xA;  libpostproc    57.  1.100 / 57.  1.100&#xA;Guessed Channel Layout for Input Stream #0.0 : stereo&#xA;Input #0, alsa, from &#x27;default:CARD=pisound&#x27;:&#xA;  Duration: N/A, start: 1684250059.973334, bitrate: 1536 kb/s&#xA;  Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s&#xA;Stream mapping:&#xA;  Stream #0:0 -> #0:0 (pcm_s16le (native) -> opus (libopus))&#xA;Press [q] to stop, [?] for help&#xA;Output #0, rtp, to &#x27;rtp://127.0.0.1:5002&#x27;:&#xA;  Metadata:&#xA;    encoder         : Lavf60.3.100&#xA;  Stream #0:0: Audio: opus, 48000 Hz, stereo, s16, 48 kb/s&#xA;    Metadata:&#xA;      encoder         : Lavc60.3.100 libopus&#xA;SDP:&#xA;v=0&#xA;o=- 0 0 IN IP4 127.0.0.1&#xA;s=No Name&#xA;c=IN IP4 127.0.0.1&#xA;t=0 0&#xA;a=tool:libavformat 60.3.100&#xA;m=audio 5002 RTP/AVP 97&#xA;b=AS:48&#xA;a=rtpmap:97 opus/48000/2&#xA;a=fmtp:97 sprop-stereo=1&#xA;&#xA;size=       0kB time=-577014:32:22.77 bitrate=  -0.0kbits/s speed=N/A    &#xA;size=       0kB time=-577014:32:22.77 bitrate=  -0.0kbits/s speed=N/A    &#xA;[out_0_0 @ 0x559f530c60] 100 buffers queued in out_0_0, something may be wrong.&#xA;size=       0kB time=-577014:32:22.77 bitrate=  -0.0kbits/s speed=N/A    &#xA;size=       0kB time=-577014:32:22.77 bitrate=  -0.0kbits/s speed=N/A    &#xA;size=       0kB time=-00:00:00.00 bitrate=  -0.0kbits/s speed=N/A    &#xA;size=       0kB time=-00:00:00.00 bitrate=  -0.0kbits/s speed=N/A    &#xA;[alsa @ 0x559f4db460] ALSA buffer xrun.&#xA;size=       6kB time=00:00:02.81 bitrate=  16.7kbits/s speed=0.93x    &#xA;size=       6kB time=00:00:02.81 bitrate=  16.7kbits/s speed=0.797x    &#xA;size=       6kB time=00:00:02.83 bitrate=  16.6kbits/s speed=0.703x    &#xA;size=       6kB time=00:00:02.83 bitrate=  16.6kbits/s speed=0.624x    &#xA;size=       6kB time=00:00:02.83 bitrate=  16.6kbits/s speed=0.562x    &#xA;size=       6kB time=00:00:02.83 bitrate=  16.6kbits/s speed=0.511x    &#xA;[alsa @ 0x559f4db460] ALSA buffer xrun.&#xA;size=      10kB time=00:00:05.67 bitrate=  14.8kbits/s speed=0.939x    &#xA;size=      10kB time=00:00:05.67 bitrate=  14.8kbits/s speed=0.866x    &#xA;size=      10kB time=00:00:05.67 bitrate=  14.8kbits/s speed=0.805x    &#xA;size=      10kB time=00:00:05.67 bitrate=  14.8kbits/s speed=0.751x    &#xA;size=      10kB time=00:00:05.69 bitrate=  14.8kbits/s speed=0.707x    &#xA;[alsa @ 0x559f4db460] ALSA buffer xrun.&#xA;size=      13kB time=00:00:05.95 bitrate=  17.8kbits/s speed=0.696x    &#xA;size=      16kB time=00:00:08.51 bitrate=  15.2kbits/s speed=0.939x    &#xA;size=      16kB time=00:00:08.51 bitrate=  15.2kbits/s speed=0.89x    &#xA;size=      16kB time=00:00:08.53 bitrate=  15.2kbits/s speed=0.847x    &#xA;size=      16kB time=00:00:08.53 bitrate=  15.2kbits/s speed=0.806x    &#xA;size=      16kB time=00:00:08.53 bitrate=  15.2kbits/s speed=0.77x    &#xA;[alsa @ 0x559f4db460] ALSA buffer xrun.&#xA;size=      21kB time=00:00:11.37 bitrate=  14.8kbits/s speed=0.981x  &#xA;

    &#xA;

    I tried changing the output to -f null /dev/null to see if the rtp was the issue, but I get the same thing. I made sure the user running it was a member to the "audio" group and arecord -l and aplay -l both show the card with the right name and information. I even tried to use its hw code instead of the default name, and same issue.

    &#xA;