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  • MediaSPIP version 0.1 Beta

    16 avril 2011, par

    MediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
    Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
    Pour avoir une installation fonctionnelle, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
    Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...)

  • MediaSPIP 0.1 Beta version

    25 avril 2011, par

    MediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
    The zip file provided here only contains the sources of MediaSPIP in its standalone version.
    To get a working installation, you must manually install all-software dependencies on the server.
    If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)

  • Amélioration de la version de base

    13 septembre 2013

    Jolie sélection multiple
    Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
    Il suffit pour cela d’activer le plugin Chosen (Configuration générale du site > Gestion des plugins), puis de configurer le plugin (Les squelettes > Chosen) en activant l’utilisation de Chosen dans le site public et en spécifiant les éléments de formulaires à améliorer, par exemple select[multiple] pour les listes à sélection multiple (...)

Sur d’autres sites (12666)

  • Install ffmpeg on elastic beanstalk using ebextensions config

    18 mai 2017, par user3581244

    I’m attempting to install an up to date version of ffmpeg on an elastic beanstalk instance on amazon servers. I’ve created my config file and added these container_commands :

       container_commands:
           01-ffmpeg:
               command: wget -O/usr/local/bin/ffmpeg http://ffmpeg.gusari.org/static/64bit/ffmpeg.static.64bit.2014-03-05.tar.gz
               leader_only: false
           02-ffmpeg:
               command: tar -xzf /usr/local/bin/ffmpeg
               leader_only: false
           03-ffmpeg:
               command: ln -s /usr/local/bin/ffmpeg /usr/bin/ffmpeg
               leader_only: false

    Command 01 and 03 seems to work perfectly but 02 doesn’t seem to work so ffmpeg doesn’t unzip. Any ideas what the issue might be ?

    Thanks,
    Helen

  • FFmpeg RTSP stream to remote MediaMTX server disconnects after a few seconds [closed]

    13 juin, par Rorschy

    I'm new to RTSP and MediaMTX, and I'm trying to live stream my screen using FFmpeg and MediaMTX for a specific use case.

    


    Everything works perfectly when both FFmpeg and MediaMTX run on the same machine.
However, when I move MediaMTX to a remote server, the stream becomes unstable — I can't maintain a connection or view the stream reliably.

    


    Here is the FFmpeg command I'm using from the client machine :

    


    ffmpeg -f gdigrab -framerate 10 -offset_x 0 -offset_y 0 -video_size 1920x1080 -i desktop -f lavfi -i anullsrc -vcodec libx264 -tune zerolatency -g 30 -sc_threshold 0 -preset ultrafast -tune zerolatency -f rtsp rtsp:///live/stream


    


    And here’s the relevant MediaMTX log output on the remote server :

    


    2025/06/12 14:28:44 INF [RTSP] [conn :35798] opened
2025/06/12 14:28:44 INF [RTSP] [session 2e487869] created by :35798
2025/06/12 14:28:44 INF [RTSP] [session 2e487869] is publishing to path 'live/stream', 2 tracks (H264, MPEG-4 Audio)
2025/06/12 14:28:45 INF [WebRTC] [session 8a909818] created by :47296
2025/06/12 14:28:45 WAR [WebRTC] [session 8a909818] skipping track 2 (MPEG-4 Audio)
2025/06/12 14:28:47 INF [WebRTC] [session dd0d3af7] created by :46306
2025/06/12 14:28:47 WAR [WebRTC] [session dd0d3af7] skipping track 2 (MPEG-4 Audio)
2025/06/12 14:28:49 INF [WebRTC] [session 5f853024] created by :46320
2025/06/12 14:28:49 WAR [WebRTC] [session 5f853024] skipping track 2 (MPEG-4 Audio)
2025/06/12 14:28:51 INF [WebRTC] [session 3edba9a8] created by :46342
2025/06/12 14:28:51 WAR [WebRTC] [session 3edba9a8] skipping track 2 (MPEG-4 Audio)
2025/06/12 14:28:53 INF [WebRTC] [session 4be5bd9b] created by :46352
2025/06/12 14:28:53 WAR [WebRTC] [session 4be5bd9b] skipping track 2 (MPEG-4 Audio)
2025/06/12 14:28:54 INF [RTSP] [conn :35798] closed: terminated
2025/06/12 14:28:54 INF [RTSP] [session 2e487869] destroyed: session timed out
2025/06/12 14:28:54 INF [WebRTC] [session 8a909818] closed: terminated
2025/06/12 14:28:54 INF [WebRTC] [session 3edba9a8] closed: terminated
2025/06/12 14:28:54 INF [WebRTC] [session 5f853024] closed: terminated


    


    My questions :

    


      

    1. What could be causing the RTSP stream to disconnect when streaming to a remote MediaMTX server ?
    2. 


    3. Are there any recommended network settings or MediaMTX configuration tweaks to ensure a stable stream over the internet ?
    4. 


    


    Any help or guidance would be greatly appreciated. Thanks !

    


  • Changing video frame rate and keeping duration intact

    6 mai 2018, par Idan

    I am trying to segment a video file, transcode each of the segments and then stitch them all back to one. The process is working but I have noticed that the final file has a different duration than the original one. Probably because I am transcoding to a fixed frame rate which is different from the source frame rate.

    To segment the video I use :

    ffmpeg -y -i video_only.mp4 -c copy -flags -global_header -segment_time 50 -break_non_keyframes 0 -reset_timestamps 0 \
     -segment_list file_segs.list -segment_format mp4 -segment_list_type ffconcat -write_empty_segments 0 \
     -f segment file_seg-%d.mp4

    Then, each segment is transcoded with this command :

    ffmpeg -y -i $f -vcodec libx264 -vsync 1 -crf 18 -vf "fps=30" -pix_fmt yuv420p -preset:v fast -profile:v main -level:v 4.1 transcoded/$f

    And finally, concat :

    ffmpeg -y -f concat -i file_segs.list -c:v copy -c:a copy -movflags +faststart video_final.mp4

    I’ve tried using "-r 30" instead on the filter (-vf), the same result.
    As this is a part of a transcoding system I prefer not to check for the input file fps before transcoding as it can be anything - variable fps too (which is not so easy to detect)

    Is there any way I can keep the duration intact while changing the frame rate in this type of transcoding flow ?