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  • Personnaliser en ajoutant son logo, sa bannière ou son image de fond

    5 septembre 2013, par

    Certains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;

  • La file d’attente de SPIPmotion

    28 novembre 2010, par

    Une file d’attente stockée dans la base de donnée
    Lors de son installation, SPIPmotion crée une nouvelle table dans la base de donnée intitulée spip_spipmotion_attentes.
    Cette nouvelle table est constituée des champs suivants : id_spipmotion_attente, l’identifiant numérique unique de la tâche à traiter ; id_document, l’identifiant numérique du document original à encoder ; id_objet l’identifiant unique de l’objet auquel le document encodé devra être attaché automatiquement ; objet, le type d’objet auquel (...)

  • Ecrire une actualité

    21 juin 2013, par

    Présentez les changements dans votre MédiaSPIP ou les actualités de vos projets sur votre MédiaSPIP grâce à la rubrique actualités.
    Dans le thème par défaut spipeo de MédiaSPIP, les actualités sont affichées en bas de la page principale sous les éditoriaux.
    Vous pouvez personnaliser le formulaire de création d’une actualité.
    Formulaire de création d’une actualité Dans le cas d’un document de type actualité, les champs proposés par défaut sont : Date de publication ( personnaliser la date de publication ) (...)

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  • Live AAC and H264 data into live stream

    10 mai 2024, par tzuleger

    I have a remote camera that captures H264 encoded video data and AAC encoded audio data, places the data into a custom ring buffer, which then is sent to a Node.js socket server, where the packet of information is detected as audio or video and then handled accordingly. That data should turn into a live stream, the protocol doesn't matter, but the delay has to be around 4 seconds and can be played on iOS and Android devices.

    


    After reading hundreds of pages of documentation, questions, or solutions on the internet, I can't seem to find anything about handling two separate streams of AAC and H264 data to create a live stream.

    


    Despite attempting many different ways of achieving this goal, even having a working implementation of HLS, I want to revisit ALL options of live streaming, and I am hoping someone out there can give me advice or guidance to specific documentation on how to achieve this goal.

    


    To be specific, this is our goal :

    


      

    • Stream AAC and H264 data from remote cellular camera to a server which will do some work on that data to live stream to one user (possibly more users in the future) on a mobile iOS or Android device

      • 


    • Delay of the live stream should be a maximum of 4 seconds, if the user has bad signal, then a longer delay is okay, as we obviously cannot do anything about that.

      • 


    • We should not have to re-encode our data. We've explored WebRTC, but that requires OPUS audio packets and thus requires us to re-encode the data, which would be expensive for our server to run.

      • 


    


    Any and all help, ranging from re-visiting an old approach we took to exploring new ones, is appreciated.

    


    I can provide code snippets as well for our current implementation of LLHLS if it helps, but I figured this post is already long enough.

    


    I've tried FFmpeg with named pipes, I expected it to just work, but FFmpeg kept blocking on the first named pipe input. I thought of just writing the data out to two files and then using FFmpeg, but it's continuous data and I don't have enough knowledge on FFmpeg on how I could use that type of implementation to create one live stream.

    


    I've tried implementing our own RTSP server on the camera using Gstreamer (our camera had its RTSP server stripped out, wasn't my call) but the camera's flash storage cannot handle having GStreamer on it, so that wasn't an option.

    


    My latest attempt was using a derivation of hls-parser to create an HLS manifest and mux.js to create MP4 containers for .m4s fragmented mp4 files and do an HLS live stream. This was my most successful attempt, where we successfully had a live stream going, but the delay was up to 16 seconds, as one would expect with HLS live streaming. We could drop the target duration down to 2 seconds and get about 6-8 seconds delay, but this could be unreliable, as these cameras could have no signal making it relatively expensive to send so many IDR frames with such low bandwidth.

    


    With the delay being the only factor left, I attempted to upgrade the implementation to support Apple's Low Latency HLS. It seems to work, as the right partial segments are getting requested and everything that makes LLHLS is working as intended, but the delay isn't going down when played on iOS' native AVPlayer, as a matter of fact, it looks like it worsened.

    


    I would also like to disclaim, my knowledge on media streaming is fairly limited. I've learned most of what I speak of in this post over the past 3 months by reading RFCs, documentation, and stackoverflow/reddit questions and answers. If anything appears to be confusing, it might be just my lack of understanding of it.

    


  • flac_parser.c : fix case when final frame is a false positive

    28 juin 2013, par Michael Chinen
    flac_parser.c : fix case when final frame is a false positive

    Should fix https://ffmpeg.org/trac/ffmpeg/ticket/2552
    
Only did minimal testing on a few files and fate.
    

    Signed-off-by : Michael Niedermayer <michaelni@gmx.at>
    • [DH] libavcodec/flac_parser.c

  • tiny_ssim : add per-frame metrics and final ssim db number.

    17 février 2014, par Ronald S. Bultje
    tiny_ssim : add per-frame metrics and final ssim db number.

    Signed-off-by : Michael Niedermayer <michaelni@gmx.at>
    • [DH] tests/tiny_ssim.c

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