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  • Les autorisations surchargées par les plugins

    27 avril 2010, par

    Mediaspip core
    autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs

  • 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 (...)

  • Librairies et logiciels spécifiques aux médias

    10 décembre 2010, par

    Pour un fonctionnement correct et optimal, plusieurs choses sont à prendre en considération.
    Il est important, après avoir installé apache2, mysql et php5, d’installer d’autres logiciels nécessaires dont les installations sont décrites dans les liens afférants. Un ensemble de librairies multimedias (x264, libtheora, libvpx) utilisées pour l’encodage et le décodage des vidéos et sons afin de supporter le plus grand nombre de fichiers possibles. Cf. : ce tutoriel ; FFMpeg avec le maximum de décodeurs et (...)

Sur d’autres sites (7617)

  • ffmpeg, download video stream from url

    1er décembre 2014, par samiullah

    I am developing application to save any online webinar(video or audio) given the url using ffmpeg library. Url usually contain other contents like text, images as well. So i have problem, how to separately get video stream from url using ffmpeg(or some other better free library). Url may be for any site, not only for youtube, but as an example, link may be like
    http://wildkingdom.magnify.net/video/Dangerous-Moments-in-the-Wild-K ;recent or
    http://www.youtube.com/watch?v=wnrJJYm7qIw
    etc

  • Record linux screen (audio/video) and stream to an API

    5 juin 2024, par Joel

    I'm running a browser on a Linux server, and I'm trying to figure out the best way to capture the audio/video of the screen, and stream that with as little latency as possible to an API on another server.

    


    Requirements :

    


      

    • Stream the audio from server A to server B so that server B can pipe that forward to an online transcription service.
    • 


    • Stream the audio and video from server A to server B so that server B can store the contents to some kind of blob storage. If the stream is killed for some reason before it's ended, the partial contents should be saved and still be playable.
    • 


    


    I've successfully captured the audio/video to disk using ffmpeg (see below), but I'm really unsure how I should change that to stream the contents to an API instead.

    


    The API is currently ASP.NET Core 8, but it could be node/python/whatever really.

    


    ffmpeg -y -video_size 1920x1080 -framerate 30 -f x11grab -i :0 -f pulse -i default -c:v libx264 -pix_fmt yuv420p -c:a aac -strict experimental /app/recordings/output.mp4


    


  • Merge audio (m4s) segments into one

    20 avril 2022, par akinuri

    I recently started learning Laravel, and currently watching an online course. Online courses are fine, but I like to have local copies, so I'm trying to download/merge segmented audio from Laracasts : Laravel 8 From Scratch series.

    


    I've written some scripts (in Python) that does the following :

    


      

    1. Download the master.json
    2. 


    3. Read master.json and download audio segments
    4. 


    5. Merge the segments into a single file (the file is not playable yet)
    6. 


    7. Process the audio file via ffmpeg (now it's playable, but has issues)
    8. 


    


    I think there's a problem with the step 3 and/or 4.

    


    In step/script 3, I create a new file, and add the contents of the segments to the file in binary.

    


    Then (step/script 4), run a ffmpeg command in python : ffmpeg -i merged-file.mp4 -c copy processed-file.mp4

    


    However, the final file doesn't work/play as expected. There's a delay in the beginning, and some parts seem to be cut off/skipped.

    


    There are three possibilities :

    


      

    • Segment files are problematic (not likely ?)
    • 


    • I'm doing the merging wrong
    • 


    • I'm doing the ffmpeg processing wrong
    • 


    


    Can someone guide me here ?

    



    


    The issues/colored parts in the ffmpeg output are :

    


    ...
[mov,mp4,m4a,3gp,3g2,mj2 @ 000001cfbc0de780] could not find corresponding track id 2
[mov,mp4,m4a,3gp,3g2,mj2 @ 000001cfbc0de780] could not find corresponding trex (id 2)
...
[aac @ 000001cfbc0f0380] Number of bands (31) exceeds limit (6).
...
[mp4 @ 000001cfbc20ecc0] track 0: codec frame size is not set
...
[mp4 @ 000001cfbc20ecc0] Non-monotonous DTS in output stream 0:0; previous: 318318, current: 286286; changing to 318319. This may result in incorrect timestamps in the output file.
...


    


    Everything required for a test case is located in GitHub (akinuri/dump/m4s-segments/). Screenshot of the contents :

    


    repo folder contents

    



    


    Note : there are two types/formats of audio in the master.json : mp42 and dash. dash works as expected, and seem to be used in limited videos/courses. On the other hand, mp42 appears more. So I need a way to make mp42 work.