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  • 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 (...)

  • 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 (...)

  • 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 (12821)

  • Stream microphone from client browser to remote server and pass audio in real time to ffmpeg to combine with a second video source

    4 mai 2021, par fakeguybrushthreepwood

    As a beginner at working with these kinds of real-time streaming services, I've spent hours trying to work out how this is possible, but can't seem to work out I'd precisely go about it.

    


    I'm prototyping a personal basic web app that does the following :

    


      

    1. In a web browser, the web application has a button that says 'Stream Microphone' - when pressed it streams the audio from the user's microphone (the user obviously has to consent to give permission to send their microphone audio) through to the server which I was presuming would be running node.js (no specific reason at this point, just thought this is how I'd go about doing it).

      


    2. 


    3. The server receives the audio close enough to real-time somehow (not sure how I'd do this).

      


    4. 


    5. I can then run ffmpeg on the command line and take the real-time audio coming in real-time and add it as the sound to a video file (let's just say I'm going to play testmovie.mp4) that I want to play.

      


    6. 


    


    I've looked at various solutions - such as maybe using WebRTC, RTP/RTSP, Piping audio into ffmpeg, Gstreamer, Kurento, Flashphoner and/or Wowza - but somehow they look overly complicated and usually seem to focus on video along with audio. I just need to work with audio.

    


  • ffmpeg - generate moov atom

    9 juillet 2021, par mmx

    I am currently recording video with blackmagic's prorecorder. I transcode video on-the-fly to mp4 video container with ffmpeg. Duration is unknown as I'm transcoding .ts that prorecorder is outputting to named pipe.

    


    My goal is to try to play this file with browser, while stream is still being recorded - Playback is great, but problem is that when I open file, duration is defined to current recording time.

    


    So, question is - I would like to generate a "fake" moov atom for duration of fe. 8 hours with ffmpeg, and then start recording mp4 file as I'm already doing. How could I do this ?

    


    Documentation that I checked and I think it's relevant - https://ffmpeg.org/ffmpeg-formats.html#MOV_002fMP4_002fISMV
I also saw this on stack : Is it possible to fake a (mp4) moov atom ?

    


  • FFMPEG segmented radio stream shows incorrect duration after a concat

    27 juillet 2021, par darbid

    On MacOS using Automator I am recording a radio stream. I thought the best way to do this so as to avoid an internet drop out was to segment it as follows.

    


    /opt/homebrew/Cellar/ffmpeg/4.4_2/bin/ffmpeg -i http://s2.viastreaming.net:7100 -t 02:10:00 -c copy -f segment -segment_time 1800 -segment_list /Volumes/BackUp/Radio/segmentlist.txt -segment_list_entry_prefix "file /Volumes/BackUp/Radio/" /Volumes/BackUp/Radio/Segments_%d.m4a


    


    I then bring it all together with the following command.

    


    /opt/homebrew/Cellar/ffmpeg/4.4_2/bin/ffmpeg -f concat -safe 0 -i /Volumes/BackUp/Radio/segmentlist.txt -c copy /Volumes/BackUp/Radio/capture.m4a


    


    Looking at the get info of the capture.m4a file it shows a duration of 2 hours 10 minutes, however, when I play it in for example iTunes it shows exactly 5:10. It also shows 5:10 for example in Emby (a local media server solution).

    


    Can someone please point me in the right direction on fixing my ffmpeg commands ?