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  • Use, discuss, criticize

    13 avril 2011, par

    Talk to people directly involved in MediaSPIP’s development, or to people around you who could use MediaSPIP to share, enhance or develop their creative projects.
    The bigger the community, the more MediaSPIP’s potential will be explored and the faster the software will evolve.
    A discussion list is available for all exchanges between users.

  • Formulaire personnalisable

    21 juin 2013, par

    Cette page présente les champs disponibles dans le formulaire de publication d’un média et il indique les différents champs qu’on peut ajouter. Formulaire de création d’un Media
    Dans le cas d’un document de type média, les champs proposés par défaut sont : Texte Activer/Désactiver le forum ( on peut désactiver l’invite au commentaire pour chaque article ) Licence Ajout/suppression d’auteurs Tags
    On peut modifier ce formulaire dans la partie :
    Administration > Configuration des masques de formulaire. (...)

  • Les tâches Cron régulières de la ferme

    1er décembre 2010, par

    La gestion de la ferme passe par l’exécution à intervalle régulier de plusieurs tâches répétitives dites Cron.
    Le super Cron (gestion_mutu_super_cron)
    Cette tâche, planifiée chaque minute, a pour simple effet d’appeler le Cron de l’ensemble des instances de la mutualisation régulièrement. Couplée avec un Cron système sur le site central de la mutualisation, cela permet de simplement générer des visites régulières sur les différents sites et éviter que les tâches des sites peu visités soient trop (...)

Sur d’autres sites (9436)

  • Evolution #3560 : Supprimer l’option expérimentale Google Closure Compiler

    3 mai 2017, par cedric -

    oui !

  • AppRTC : Google’s WebRTC test app and its parameters

    23 juillet 2014, par silvia

    If you’ve been interested in WebRTC and haven’t lived under a rock, you will know about Google’s open source testing application for WebRTC : AppRTC.

    When you go to the site, a new video conferencing room is automatically created for you and you can share the provided URL with somebody else and thus connect (make sure you’re using Google Chrome, Opera or Mozilla Firefox).

    We’ve been using this application forever to check whether any issues with our own WebRTC applications are due to network connectivity issues, firewall issues, or browser bugs, in which case AppRTC breaks down, too. Otherwise we’re pretty sure to have to dig deeper into our own code.

    Now, AppRTC creates a pretty poor quality video conference, because the browsers use a 640×480 resolution by default. However, there are many query parameters that can be added to the AppRTC URL through which the connection can be manipulated.

    Here are my favourite parameters :

    • hd=true : turns on high definition, ie. minWidth=1280,minHeight=720
    • stereo=true : turns on stereo audio
    • debug=loopback : connect to yourself (great to check your own firewalls)
    • tt=60 : by default, the channel is closed after 30min – this gives you 60 (max 1440)

    For example, here’s how a stereo, HD loopback test would look like : https://apprtc.appspot.com/?r=82313387&hd=true&stereo=true&debug=loopback .

    This is not the limit of the available parameter, though. Here are some others that you may find interesting for some more in-depth geekery :

    • ss=[stunserver] : in case you want to test a different STUN server to the default Google ones
    • ts=[turnserver] : in case you want to test a different TURN server to the default Google ones
    • tp=[password] : password for the TURN server
    • audio=true&video=false : audio-only call
    • audio=false : video-only call
    • audio=googEchoCancellation=false,googAutoGainControl=true : disable echo cancellation and enable gain control
    • audio=googNoiseReduction=true : enable noise reduction (more Google-specific parameters)
    • asc=ISAC/16000 : preferred audio send codec is ISAC at 16kHz (use on Android)
    • arc=opus/48000 : preferred audio receive codec is opus at 48kHz
    • dtls=false : disable datagram transport layer security
    • dscp=true : enable DSCP
    • ipv6=true : enable IPv6

    AppRTC’s source code is available here. And here is the file with the parameters (in case you want to check if they have changed).

    Have fun playing with the main and always up-to-date WebRTC application : AppRTC.

    UPDATE 12 May 2014

    AppRTC now also supports the following bitrate controls :

    • arbr=[bitrate] : set audio receive bitrate
    • asbr=[bitrate] : set audio send bitrate
    • vsbr=[bitrate] : set video receive bitrate
    • vrbr=[bitrate] : set video send bitrate

    Example usage : https://apprtc.appspot.com/?r=&asbr=128&vsbr=4096&hd=true

  • AppRTC : Google’s WebRTC test app and its parameters

    23 juillet 2014, par silvia

    If you’ve been interested in WebRTC and haven’t lived under a rock, you will know about Google’s open source testing application for WebRTC : AppRTC.

    When you go to the site, a new video conferencing room is automatically created for you and you can share the provided URL with somebody else and thus connect (make sure you’re using Google Chrome, Opera or Mozilla Firefox).

    We’ve been using this application forever to check whether any issues with our own WebRTC applications are due to network connectivity issues, firewall issues, or browser bugs, in which case AppRTC breaks down, too. Otherwise we’re pretty sure to have to dig deeper into our own code.

    Now, AppRTC creates a pretty poor quality video conference, because the browsers use a 640×480 resolution by default. However, there are many query parameters that can be added to the AppRTC URL through which the connection can be manipulated.

    Here are my favourite parameters :

    • hd=true : turns on high definition, ie. minWidth=1280,minHeight=720
    • stereo=true : turns on stereo audio
    • debug=loopback : connect to yourself (great to check your own firewalls)
    • tt=60 : by default, the channel is closed after 30min – this gives you 60 (max 1440)

    For example, here’s how a stereo, HD loopback test would look like : https://apprtc.appspot.com/?r=82313387&hd=true&stereo=true&debug=loopback .

    This is not the limit of the available parameter, though. Here are some others that you may find interesting for some more in-depth geekery :

    • ss=[stunserver] : in case you want to test a different STUN server to the default Google ones
    • ts=[turnserver] : in case you want to test a different TURN server to the default Google ones
    • tp=[password] : password for the TURN server
    • audio=true&video=false : audio-only call
    • audio=false : video-only call
    • audio=googEchoCancellation=false,googAutoGainControl=true : disable echo cancellation and enable gain control
    • audio=googNoiseReduction=true : enable noise reduction (more Google-specific parameters)
    • asc=ISAC/16000 : preferred audio send codec is ISAC at 16kHz (use on Android)
    • arc=opus/48000 : preferred audio receive codec is opus at 48kHz
    • dtls=false : disable datagram transport layer security
    • dscp=true : enable DSCP
    • ipv6=true : enable IPv6

    AppRTC’s source code is available here. And here is the file with the parameters (in case you want to check if they have changed).

    Have fun playing with the main and always up-to-date WebRTC application : AppRTC.

    UPDATE 12 May 2014

    AppRTC now also supports the following bitrate controls :

    • arbr=[bitrate] : set audio receive bitrate
    • asbr=[bitrate] : set audio send bitrate
    • vsbr=[bitrate] : set video receive bitrate
    • vrbr=[bitrate] : set video send bitrate

    Example usage : https://apprtc.appspot.com/?r=&asbr=128&vsbr=4096&hd=true

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