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  • La file d’attente de SPIPmotion

    28 novembre 2010, par

    Une file d’attente stockée dans la base de donnée
    Lors de son installation, SPIPmotion crée une nouvelle table dans la base de donnée intitulée spip_spipmotion_attentes.
    Cette nouvelle table est constituée des champs suivants : id_spipmotion_attente, l’identifiant numérique unique de la tâche à traiter ; id_document, l’identifiant numérique du document original à encoder ; id_objet l’identifiant unique de l’objet auquel le document encodé devra être attaché automatiquement ; objet, le type d’objet auquel (...)

  • 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 (5195)

  • Could not find com.arthenica:ffmpeg-kit-full:6.0-2

    6 mai, par gabocalero

    I started receiving this message when I build my project

    


    > Could not find com.arthenica:ffmpeg-kit-full:6.0-2.
  Searched in the following locations:
    - https://dl.google.com/dl/android/maven2/com/arthenica/ffmpeg-kit-full/6.0-2/ffmpeg-kit-full-6.0-2.pom
    - https://repo.maven.apache.org/maven2/com/arthenica/ffmpeg-kit-full/6.0-2/ffmpeg-kit-full-6.0-2.pom
    - https://jcenter.bintray.com/com/arthenica/ffmpeg-kit-full/6.0-2/ffmpeg-kit-full-6.0-2.pom
    - https://jitpack.io/com/arthenica/ffmpeg-kit-full/6.0-2/ffmpeg-kit-full-6.0-2.pom
    - https://pkgs.dev.azure.com/MicrosoftDeviceSDK/DuoSDK-Public/_packaging/Duo-SDK-Feed/maven/v1/com/arthenica/ffmpeg-kit-full/6.0-2/ffmpeg-kit-full-6.0-2.pom
    - https://oss.sonatype.org/content/repositories/snapshots/com/arthenica/ffmpeg-kit-full/6.0-2/ffmpeg-kit-full-6.0-2.pom
  Required by:
      project :presentation > project :domain


    


    These are the project's repositories

    


    allprojects {     
   repositories {
      google()
      mavenCentral()
      jcenter()
      maven { url 'https://jitpack.io' }
      maven { url "https://oss.sonatype.org/content/repositories/snapshots" }     
   } 
}


    


    And this is the dependency I'm adding

    


        implementation(libs.arthenica.ffmpeg.full)


    


    This is my libs.version.toml

    


    ffmpeg = "6.0-2"
arthenica-ffmpeg-full = {  group = "com.arthenica", name = "ffmpeg-kit-full", version.ref = "ffmpeg" }


    


    AFAICS in the project's github page, it will not be maintained anymore
ffmpeg-kit is archived on April 21, 2025

    


    In the short term, do you know any other repository that still servers this dependency ?

    


    On the other hand, do you know any other project to replace Arthenica ffmpeg-kit-full dependency ?

    


    Thank you very much

    


  • FFMPEG send RTP audio at 8k bytes/sec

    10 mai, par Muzza

    I'm trying to use FFMPEG to mimick a device that transmits G711U audio over UDP/RTP at 8k bytes per second.
The device im mimicking sends rtp packets every 20ms with 160byte payload.

    


    I've had limited success using the following command

    


    ffmpeg -f dshow -i audio="Microphone (Realtek(R) Audio)" -ac 1 -ar 8000 -ab 8 -acodec pcm_mulaw -f rtp rtp://127.0.0.1:12345?pkt_size=160


    


    This sends G711U encoded audio, in 160byte chunks, but streams at 64kB/s, not the 8kB/s that my device is expected, so the device errors out ?

    


    Any idea's would be massively appreciated !

    


    Thank you

    


    Log from FFMPEG

    


    >ffmpeg -f dshow -i audio="Microphone (Realtek(R) Audio)" -ac 1 -ar 8000 -ab 8 -acodec pcm_mulaw -f rtp rtp://127.0.0.1:12345?pkt_size=160
ffmpeg version 2025-04-23-git-25b0a8e295-essentials_build-www.gyan.dev Copyright (c) 2000-2025 the FFmpeg developers
  built with gcc 14.2.0 (Rev3, Built by MSYS2 project)
  configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-sdl2 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-mediafoundation --enable-libass --enable-libfreetype --enable-libfribidi --enable-libharfbuzz --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-dxva2 --enable-d3d11va --enable-d3d12va --enable-ffnvcodec --enable-libvpl --enable-nvdec --enable-nvenc --enable-vaapi --enable-libgme --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvo-amrwbenc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-librubberband
  libavutil      60.  2.100 / 60.  2.100
  libavcodec     62.  0.101 / 62.  0.101
  libavformat    62.  0.100 / 62.  0.100
  libavdevice    62.  0.100 / 62.  0.100
  libavfilter    11.  0.100 / 11.  0.100
  libswscale      9.  0.100 /  9.  0.100
  libswresample   6.  0.100 /  6.  0.100
  libpostproc    59.  1.100 / 59.  1.100
[aist#0:0/pcm_s16le @ 00000198256b73c0] Guessed Channel Layout: stereo
Input #0, dshow, from 'audio=Microphone (Realtek(R) Audio)':
  Duration: N/A, start: 135470.702000, bitrate: 1411 kb/s
  Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s, Start-Time 135470.702s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_mulaw (native))
Press [q] to stop, [?] for help
[pcm_mulaw @ 00000198256cf240] Bitrate 8 is extremely low, maybe you mean 8k
Output #0, rtp, to 'rtp://127.0.0.1:12345?pkt_size=160':
  Metadata:
    encoder         : Lavf62.0.100
  Stream #0:0: Audio: pcm_mulaw, 8000 Hz, mono, s16 (8 bit), 64 kb/s
    Metadata:
      encoder         : Lavc62.0.101 pcm_mulaw
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 62.0.100
m=audio 12345 RTP/AVP 0
b=AS:64

[out#0/rtp @ 00000198256cdd00] video:0KiB audio:973KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: 8.467470%
size=    1055KiB time=00:02:04.51 bitrate=  69.4kbits/s speed=   1x
Exiting normally, received signal 2.


    


    Wireshark :
Wireshark Log

    


    Shows packets being sent every 0.20ms

    


  • Revision 28933 : on bouge

    31 mai 2009, par ben.spip@… — Log

    on bouge