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  • Supporting all media types

    13 avril 2011, par

    Unlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)

  • Ajouter notes et légendes aux images

    7 février 2011, par

    Pour pouvoir ajouter notes et légendes aux images, la première étape est d’installer le plugin "Légendes".
    Une fois le plugin activé, vous pouvez le configurer dans l’espace de configuration afin de modifier les droits de création / modification et de suppression des notes. Par défaut seuls les administrateurs du site peuvent ajouter des notes aux images.
    Modification lors de l’ajout d’un média
    Lors de l’ajout d’un média de type "image" un nouveau bouton apparait au dessus de la prévisualisation (...)

  • Selection of projects using MediaSPIP

    2 mai 2011, par

    The examples below are representative elements of MediaSPIP specific uses for specific projects.
    MediaSPIP farm @ Infini
    The non profit organizationInfini develops hospitality activities, internet access point, training, realizing innovative projects in the field of information and communication technologies and Communication, and hosting of websites. It plays a unique and prominent role in the Brest (France) area, at the national level, among the half-dozen such association. Its members (...)

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  • How to decouple between ffmpeg and rtp server ?

    2 février 2023, par dror

    I have a ffmpeg-based worker that handles video-generation jobs at very high throughput.
Long videos need to be streamed while being generated.

    


    For that purpose, I have introduced a WebRTC server named Janus-Gateway with its streaming plugin, and set the application's output to an rtp:// endpoint at that server (ffmpeg can stream a single stream using the ​RTP protocol).

    


    In order to avoid buffering problems on the other hand, the streaming is done through the ffmpeg's -re option, which means that the stream will be streamed in real-time, i.e. it slows it down to simulate live streaming.

    


    [ffmpeg-based app] (#1)--> [rtp://janus:port @ webrtc server] (#2)--> [webrtc subscribers]


    


    How can I continue processing video jobs at high throughput while streaming the results at real-time speed ? I need somehow to decouple ffmpeg output (stage #1) so that consumers at stage #2 get streams at natural playback speed.

    


  • avcodec/cfhd, cfhddata : Simplify check for escape

    3 septembre 2022, par Andreas Rheinhardt
    avcodec/cfhd, cfhddata : Simplify check for escape

    cfhd.c checked for level being equal to a certain codebook-
    
dependent constant and to run being two. The first check is
    
actually redundant, as all codebooks contain only one (real)
    
entry with run == 2 (as is usual with VLCs, this one real entry
    
has several corresponding entries in the table). But given
    
that no entry has a run of zero (except incomplete entries
    
which just signal that one needs to do another round of parsing),
    
one can actually use that as sentinel. This patch does so.
    

    Reviewed-by : Paul B Mahol <onemda@gmail.com>
    
Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
    • [DH] libavcodec/cfhd.c
    • [DH] libavcodec/cfhddata.c

  • TV audio extracted using ffmeg does not work in iOS (but it works in the simulator)

    26 janvier 2014, par Genar

    I can process a TV signal (well I have a .ts video which comes from a TV channel), using ffmpeg but the audio cannot be understood in an iPhone/iPad. The most strange is that the audio (and video) works properly in the simulator (and also in an Android real device, but this is another point), but in a real iPhone/iPad device the video is OK but the audio sounds like a metallic box and nothing can be understood.

    I have created the ffmpeg libraries for iOS (I have tried the version 2.0.2 and also the version 2.1.3) using the information provided in the following link :

    Installing ffmpeg ios libraries armv7, armv7s, i386 and universal on Mac with 10.8

    The aforementioned link explains how to create the ffmpeg include folders and the universal libraries which I have included into my project (the libraries created are : libavcodec.a, libavdevice.a, libavfilter.a, libavformat.a, libavresample.a, libavutil.a, libswresample.a and libswscale.a).

    The sampling frequency used is 48000.

    The audio got from ffmpeg is then stored into a buffer and then "inserted" using OpenAL ; but, from the same TV content (the same .ts video file) the audio data which is generated from ffmpeg in an iPhone/iPad is totally different from the audio data generated from ffmpeg in the simulator (which can reproduce both the audio and video perfectly).

    Thanks in advance,

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