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  • Récupération d’informations sur le site maître à l’installation d’une instance

    26 novembre 2010, par

    Utilité
    Sur le site principal, une instance de mutualisation est définie par plusieurs choses : Les données dans la table spip_mutus ; Son logo ; Son auteur principal (id_admin dans la table spip_mutus correspondant à un id_auteur de la table spip_auteurs)qui sera le seul à pouvoir créer définitivement l’instance de mutualisation ;
    Il peut donc être tout à fait judicieux de vouloir récupérer certaines de ces informations afin de compléter l’installation d’une instance pour, par exemple : récupérer le (...)

  • Publier sur MédiaSpip

    13 juin 2013

    Puis-je poster des contenus à partir d’une tablette Ipad ?
    Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir

  • Encoding and processing into web-friendly formats

    13 avril 2011, par

    MediaSPIP automatically converts uploaded files to internet-compatible formats.
    Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
    Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
    Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
    All uploaded files are stored online in their original format, so you can (...)

Sur d’autres sites (4688)

  • converting mp3 to ts with ffmpeg [closed]

    29 janvier, par tamirg

    i have an mp3 of about 50 kb.
if i use ffmpeg to convert to ts :

    


    ffmpeg -i sample.mp3 sample.ts


    


    i end up getting a file of almost 200kb.

    


    while if i use some online software which converts mp3 to ts (https://clideo.com/editor/?convertTo=TS for example), i get a file with similar size to the original mp3.

    


    why does the ffmpeg conversion creates such a large file ? is it possible to reduce that ?

    


  • FFmpeg -> JSMpeg Websocket Closes Repeatedly

    13 mars 2018, par Kyle Martin

    I’m trying to create a fairly simple streaming server/site. Here’s the current flow :

    • OBS streams to an RTMP URL
    • Nginx accepts the RTMP stream and uses exec-push to have FFmpeg pick up the stream and transcode it
    • FFmpeg transcodes the stream and outputs it to a JSMpeg application, which displays the stream on a webpage.

    When I have my exec_push statement as follows, everything seems to work perfectly, except the browser says Possible garbage data. Skipping. on every frame it receives :

    exec_push /usr/bin/ffmpeg -re -i rtmp://127.0.0.1:1935/$app/$name -f mpeg1video  http://localhost:8080/supersecret;

    This behavior is understandable, because JSMpeg must receive MPEG-TS data, not MPEG1 data. It sees the MPEG1 frames and thinks they’re garbage.

    So through some online research, I found this :

    exec_push /usr/bin/ffmpeg -re -i rtmp://127.0.0.1:1935/$app/$name -c:v copy -c:a copy -f mpegts http://localhost:8080/supersecret;

    Supposedly, this is supposed to transcode my RTMP stream into an MPEG-TS format, which should be compatible with JSMpeg.

    However, with the second version of the command, my FFmpeg -> JSMpeg stream keeps connecting and disconnecting, connecting and disconnecting, and so on. This behavior is observed in terminal :

    Stream Connected: ::1:40208
    close
    Stream Connected: ::1:40212
    close
    Stream Connected: ::1:40216
    close
    Stream Connected: ::1:40220
    close
    Stream Connected: ::1:40224
    close
    ...

    What would cause this ? I am pretty certain the issue is in my exec_push command. OBS is perfectly content, which tells me that the stream is making it to the server, and if I do a push, I can do a test push to Ustream just fine, which tells me that Nginx is at least processing the stream with some reasonable degree of success.


    Disclaimer : I have no idea what I’m talking about. Everything I know about FFmpeg and JSMpeg/Node is from snippets of code that I found online.

  • ffmpeg start/stop rtmp streams on the fly

    14 août 2020, par almosnow

    I have a media stream that I transmit as rtmp to several endpoints, for this I use ffmpeg and the tee pseudo-format.

    


    Sometimes one of the sites may come online (or go offline) and I would like to start/stop streaming to it accordingly.

    


    Restarting ffmpeg is not an optimal choice as all the streams that are currently being transmitted would get cut.

    


    Is there a way to add/remove targets to tee on the fly ? Or a way to achieve a similar thing with ffmpeg and tools from a linux environment ? Perhaps there's something that already does this with rtmp streams.