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  • Keeping control of your media in your hands

    13 avril 2011, par

    The vocabulary used on this site and around MediaSPIP in general, aims to avoid reference to Web 2.0 and the companies that profit from media-sharing.
    While using MediaSPIP, you are invited to avoid using words like "Brand", "Cloud" and "Market".
    MediaSPIP is designed to facilitate the sharing of creative media online, while allowing authors to retain complete control of their work.
    MediaSPIP aims to be accessible to as many people as possible and development is based on expanding the (...)

  • 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 (...)

Sur d’autres sites (8255)

  • Gstreamer RTSP Server over UDP artifacts on video when there is a packet loss

    28 août 2023, par Usama

    I want to have a RTSP server which should work on top of UDP , but when I tried to use Gstreamer RTSP server its giving video artifact/glitches when there is a packet loss.

    


    I have tried to follow this example for RTSP re-transmission.

    


    gst_rtsp_media_factory_set_profiles (factory, GST_RTSP_PROFILE_AVPF);

    


    When the profile is set like above , ffmpeg / mplayer / VLC is not able to play and says that it is an unsupported protocol. I don't know which player supports this protocol, only the gstreamer player is able to play the video.

    


    When the gst_rtsp_media_factory_set_profiles line is removed then all the players are working but there is a lot of artifact on the video if there is packet loss.

    


    everything is working fine on local env but on cloud there are artifact because of packet loss

    


    I also tried making retransission_time to 10 sec but it did not work.

    


    gst_rtsp_media_factory_set_retransmission_time (factory, 10000 * GST_MSECOND);

    


    Tried enabling RTCP as required for NACK (did not work i can see RTCP messages in wireshark )

    


    gst_rtsp_media_factory_set_enable_rtcp(factory , 1);

    


    Tried setting retransmission true (Did not worked )

    


    gst_rtsp_media_factory_set_do_retransmission(factory,1);

    


    I still get the artifacts / glitches on the video some time video just stuck on the client side

    


    I can confirm that its due to packet loss because when i switch the RTSP to use TCP instead of UDP everything works fine.

    


  • Mixed Reality WebRTC without Signalling Server

    25 mai 2021, par SilverLife

    I am trying to find a way, which allows me to use Mixed Reality WebRTC (link to git-repo) without a signalling server.
In detail, I want to create a sdp-file from my ffmpeg Video sender and use this sdp-description in my unity-Project to bypass the signaling process and receive the ffmpeg video stream.
Is there a way of doing so with Mixed Reality WebRTC ? I was already searching for the line of code, where the sdp-file is created within MR WebRTC but I didn´t find it.

    


    I am relatively new to this topic and I am not sure if this works at all but since ffmpeg is not directly compatible with webrtc I was thinking that this might be the most promising approach.

    


  • How to install youtube-to-gif in the Linux server ?

    8 janvier 2016, par Chinu

    Hi I am using Linux cloud server. I have successfully installed ffmpeg and nodeJS by server support team. But I am unable to install youtube-to-gif

    My nodeJS version is - 2.5.1
    ffmpeg version is - 0.8.17-4:0.8.17-0ubuntu0.12.04.1

    I have tried to installed youtube-to-gif in following command

    $ sudo npm install youtube-to-gif -g

    or

    $ npm install youtube-to-gif -g

    This two command not working. I saw youtube-to-gif installed in the node_modules directory but when i am typing $ youtube-to-gif -h I got this error message

    -bash: youtube-to-gif: command not found

    enter image description here

    YouTube to GIF command

    youtube-to-gif -u https://www.youtube.com/watch?v=NqxSgp385N0 -b 30 -d 5

    SERVER

    enter image description here

    Local

    enter image description here

    Document - https://www.npmjs.com/package/youtube-to-gif

    I am not sure this is perfectly installed or not. But youtube-to-gif command not working. Can you please help me.

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