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Autres articles (38)

  • MediaSPIP v0.2

    21 juin 2013, par

    MediaSPIP 0.2 est la première version de MediaSPIP stable.
    Sa date de sortie officielle est le 21 juin 2013 et est annoncée ici.
    Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
    Comme pour la version précédente, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
    Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...)

  • Mise à disposition des fichiers

    14 avril 2011, par

    Par défaut, lors de son initialisation, MediaSPIP ne permet pas aux visiteurs de télécharger les fichiers qu’ils soient originaux ou le résultat de leur transformation ou encodage. Il permet uniquement de les visualiser.
    Cependant, il est possible et facile d’autoriser les visiteurs à avoir accès à ces documents et ce sous différentes formes.
    Tout cela se passe dans la page de configuration du squelette. Il vous faut aller dans l’espace d’administration du canal, et choisir dans la navigation (...)

  • 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 (5809)

  • Ffmpeg mixing audio only for n number of videos using offset and altered duraion

    26 août 2022, par Turgut

    I want to mix audio stream of n amount of video files with ffmpeg, with certain parameters such as :

    


      

    • Duration : I want to specify how long each audio is going to play for each output. For example if I specify ffmpeg -i -d:5 first.mp4 -i second.mp4 it should play the audio from first.mp4 for 5 seconds (I don't know if -d:5 is a real tag it's just an example I've made ).
    • 


    • Starting point : I want to specify when a given file is going start on the output, I probably need to use -itsoffset 5 but I don't know if it's the right one in terms of interacting with other commands, in on itself it works fine. For example : ffmpeg -i -isoffset 5 first.mp -i second.mp4 causes the second.mp4 to start immediately, and first.mp4 to start after 5 seconds.
    • 


    • Segmentation : This is the tricky one, I want to specify at which point the input's audio should start. It's like the -ss flag but the problem is it's not working together with -itsoffset. For example when I say ffmpeg -i -ss 5 first.mp4 -i second.mp4 both files should start immediately on the output, but first.mp4 should start on it's 5 seconds. So the 5th second of first.mp4 is heard at the 1st second of the output.
    • 


    


    This is what I'm trying to achieve, my problem is that I don't know how to implement 'duration' and -ss is not working together with -itsoffset.

    


    At the end I should have something similar to this :

    


    ffmpeg -y -d 5 -itsoffset 3.5 -i first.mp4 -d 10 -ss 10 -itsoffset 5.3 -i 3 -vn -copyts -async 1 -filter_complex amix=inputs=2 out.mp


    


    Which should result in an audio that sounds like this : The first 3.5 seconds are empty, no audio is heard. Then first.mp4 is heard from it's beginning for 5 seconds. When the outputs timestamp reaches 5.3, the 10th seconds of second.mp4 is heard (while first.mp4 is still playing, it's supposed to play until 8.5, so I should hear both files at the same time.) for 10 seconds.

    


    I can't find an example of this and some sources are out-of date.

    


  • Real time livestreaming - RPI FFmpeg and H5 Player

    29 avril 2022, par Victor

    I work at a telehealth company and we are using connected medical devices in order to provide the doctor with real time information from these equipements, the equipements are used by a trained health Professional.

    


    Those devices work with video and audio. Right now, we are using them with peerjs (so peer to peer connection) but we are trying to move away from that and have a RPI with his only job to stream data (so streaming audio and video).

    


    Because the equipements are supposed to be used with instructions from a doctor we need the doctor to receive the data in real time.

    


    But we also need the trained health professional to see what he is doing (so we need a local feed from the equipement)

    


    How do we capture audio and video

    


    We are using ffmpeg with a go client that is in charge of managing the ffmpeg clients and stream them to a SRS server.
This works but we are having a 2-3 sec delay when streaming the data. (rtmp from ffmpeg and flv on the front end)

    


    ffmpeg settings :

    


    ("ffmpeg", "-f", "v4l2", `-i`, "*/video0", "-f", "flv", "-vcodec", "libx264", "-x264opts", "keyint=15", "-preset", "ultrafast", "-tune", "zerolatency", "-fflags", "nobuffer", "-b:a", "160k", "-threads", "0", "-g", "0", "rtmp://srs-url")


    


    My questions

    


      

    • Is there a way for this set up to achieve low latency (<1 sec) (for the nurse and for the doctor) ?
    • &#xA;

    • Is the way I want to achieve this good ? Is there a batter way ?
    • &#xA;

    &#xA;

    Flow schema

    &#xA;

    Data exchange and use case flow :

    &#xA;

    Data exchange and use case flow

    &#xA;

    &#xA;

    Note : The nurse and doctor use HTTP-FLV to play the live stream, for low latency.

    &#xA;

    &#xA;

  • Real time livestreaming - RPI

    24 avril 2022, par Victor

    I work at a telehealth company and we are using connected medical devices in order to provide the doctor with real time information from these equipements, the equipements are used by a trained health Professional.

    &#xA;

    Those devices work with video and audio. Right now, we are using them with peerjs (so peer to peer connection) but we are trying to move away from that and have a RPI with his only job to stream data (so streaming audio and video).

    &#xA;

    Because the equipements are supposed to be used with instructions from a doctor we need the doctor to receive the data in real time.

    &#xA;

    But we also need the trained health professional to see what he is doing (so we need a local feed from the equipement)

    &#xA;

    How do we capture audio and video

    &#xA;

    We are using ffmpeg with a go client that is in charge of managing the ffmpeg clients and stream them to a SRS server.&#xA;This works but we are having a 2-3 sec delay when streaming the data. (rtmp from ffmpeg and flv on the front end)

    &#xA;

    ffmpeg settings :

    &#xA;

    ("ffmpeg", "-f", "v4l2", `-i`, "*/video0", "-f", "flv", "-vcodec", "libx264", "-x264opts", "keyint=15", "-preset", "ultrafast", "-tune", "zerolatency", "-fflags", "nobuffer", "-b:a", "160k", "-threads", "0", "-g", "0", "rtmp://srs-url")&#xA;

    &#xA;

    My questions

    &#xA;

      &#xA;
    • Is there a way for this set up to achieve low latency (<1 sec) (for the nurse and for the doctor) ?
    • &#xA;

    • Is the way I want to achieve this good ? Is there a batter way ?
    • &#xA;

    &#xA;

    Flow schema

    &#xA;

    Data exchange and use case flow

    &#xA;