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  • Des sites réalisés avec MediaSPIP

    2 mai 2011, par

    Cette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
    Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page.

  • Support audio et vidéo HTML5

    10 avril 2011

    MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
    Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
    Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
    Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)

  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
    The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
    For older browsers the Flowplayer flash fallback is used.
    MediaSPIP allows for media playback on major mobile platforms with the above (...)

Sur d’autres sites (4684)

  • ffmpeg custom buffer sink filter

    25 novembre 2018, par NadavRub

    Environment

    • Ubuntu 18.04
    • C++
    • ffmpeg 3.4 (git master)
    • ffmpeg is used as a shared lib (InProc) via the C++ API

    Use-case

    • Per this link I am trying to use the ’avfilter_graph_*’ APIs to create ffmpeg graph
    • I would like the graph output to be sent out to my custom code ( part of the hosting application )

    Considered implementations

    • [A] Implement a custom sink filter ( part of libavfilter ) to implement my custom logic
    • [B] Implement a custom sink filter to grab the output samples and send them out to my application ( something similar to DShow SampleGrabber )

    Problem at hand

    With either of the above mentioned approaches the ffmpeg code has to be modified, and this impose an overhead in supporting future ffmpeg releases

    I wonder if there is any straight forward approach for an external ( hosting ) application to grab the graph output w/ minimal copying of the payload.

    Is there any way to use a custom AVIOContext to achieve that ? can I construct a graph connected to an output AVIOContext ? can I create a custom filter implemented in a module external to libavfilter and associate it w/ the graph using ’AVFilterContext’ ?

  • ffmpeg output file smaller than input file

    3 mai 2020, par Debug255

    I am using ffmpeg to rotate videos 90 or 180 degrees in a Python script. It works great. But, I am curious as to why the output file would be a smaller amount of bytes than the input file.

    



    Here are the commands I use :

    



    180 degrees :

    



    ffmpeg -i ./input.mp4 -preset veryslow -vf "transpose=2,transpose=2,format=yuv420p" -metadata:s:v rotate=0 -codec:v libx264 -codec:a copy ./output.mp4

    



    90 degrees :

    



    ffmpeg -i ./input.mp4 -vf "transpose=2" ./output.mp4

    



    For example, a GoPro Hero 3 MP4 file was originally 2.0 GB. The resulting output file was 480.9 MB. Another GoPro file was 2.0 and its resulting file was 671.5 MB. Is this maybe because the GoPro files were 2.0 but contains empty space, sort of like how some NTFS filesystems make a minimal 4k file, even when there is less bytes in it ?

    



    If this isn't the GoPro Hero 3, how do I rotate the files 90 or 180 degrees but ensure the output file size is the same ? Or, is data loss expected ? Does the data loss have to do with the format ?

    



    Note that the quality of the video doesn't appear to be damaged, which is good. So, I am interested in learning more about why this is happening, then I can read the section of ffmpeg documentation that is relevant to this.

    



    Thank you !

    


  • How to stream the video from one PC to another with an acceptable quality and synchronization ?

    15 juin 2021, par ErickSkrauch

    I have the following task : to organize the broadcast of several gamers on the director's computer, which will switch the image to, to put it simply, the one who currently has more interesting gameplay.

    


    The obvious solution would be to raise an RTMP server and broadcast to it. We tried that. The image quality clearly correlates with the bitrate of the broadcast, but the streams aren't synchronized and there is no way to synchronize them. As far as I know, it's just not built into the RTMP protocol.

    


    We also tried streaming via UDP, SRT and RTSP protocols. We got minimal delay but a very blurry image and artifacts from lost packets. It feels like all these formats are trying to achieve constant FPS and sacrifice the quality.

    


    What we need :

    


      

    • A quality image.
    • 


    • Broken frames can be discarded (it's okay to have not constant FPS).
    • 


    • Latency isn't important.
    • 


    • The streams should be synchronized within a second or two.
    • 


    


    There is an assumption that broadcasting on UDP should be a solution, but some kind of intermediate buffer is needed to provide the necessary broadcasting conditions. But I don't know how to do that. I assume that we need an intermediate ffmpeg instance, which will read the incoming stream, buffer it and publish the result to some local port, from which the picture will be already taken by the director's OBS.

    


    Is there any solution to achieve our goals ?