Recherche avancée

Médias (91)

Autres articles (20)

  • D’autres logiciels intéressants

    12 avril 2011, par

    On ne revendique pas d’être les seuls à faire ce que l’on fait ... et on ne revendique surtout pas d’être les meilleurs non plus ... Ce que l’on fait, on essaie juste de le faire bien, et de mieux en mieux...
    La liste suivante correspond à des logiciels qui tendent peu ou prou à faire comme MediaSPIP ou que MediaSPIP tente peu ou prou à faire pareil, peu importe ...
    On ne les connais pas, on ne les a pas essayé, mais vous pouvez peut être y jeter un coup d’oeil.
    Videopress
    Site Internet : (...)

  • 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 (...)

  • Emballe médias : à quoi cela sert ?

    4 février 2011, par

    Ce plugin vise à gérer des sites de mise en ligne de documents de tous types.
    Il crée des "médias", à savoir : un "média" est un article au sens SPIP créé automatiquement lors du téléversement d’un document qu’il soit audio, vidéo, image ou textuel ; un seul document ne peut être lié à un article dit "média" ;

Sur d’autres sites (4423)

  • 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.

    


  • 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


    


  • How to use the result of guardian-ffmpeg-Android ?

    4 avril 2013, par user1914692

    I am trying to build ffmpeg on Android. There are many tutorials. Some are very old.
    So I want to try one that can use newer version of ffmpeg and Android NDK.
    After long time searching, I find one, guardianproject / android-ffmpeg

    The project was updated several months ago.
    NDK r8 is used. ffmpeg is put from online, so a latest version.

    After I follow all the instruction, I am confused which result I should use, and how to use it.
    The README mentions testing, like :

    # embedding metadata into a matroska video /data/local/ffmpeg -y -i test.mp4 \
       -attach attach.txt -metadata:s:2 mimetype=text/plain \
       -acodec copy -vcodec copy testattach.mkv

    First, I fail to find the path : /data/local
    Second, this is a command. How will I use it in Android ?

    Totall confused.
    Any light ?