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  • Amélioration de la version de base

    13 September 2013

    Jolie sélection multiple
    Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
    Il suffit pour cela d’activer le plugin Chosen (Configuration générale du site > Gestion des plugins), puis de configurer le plugin (Les squelettes > Chosen) en activant l’utilisation de Chosen dans le site public et en spécifiant les éléments de formulaires à améliorer, par exemple select[multiple] pour les listes à sélection multiple (...)

  • Menus personnalisés

    14 November 2010, by

    MediaSPIP utilise le plugin Menus pour gérer plusieurs menus configurables pour la navigation.
    Cela permet de laisser aux administrateurs de canaux la possibilité de configurer finement ces menus.
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    Par défaut trois menus sont créés automatiquement à l’initialisation du site : Le menu principal; Identifiant : barrenav; Ce menu s’insère en général en haut de la page après le bloc d’entête, son identifiant le rend compatible avec les squelettes basés sur Zpip; Éléments (...)

  • Configuration spécifique pour PHP5

    4 February 2011, by

    PHP5 est obligatoire, vous pouvez l’installer en suivant ce tutoriel spécifique.
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    Il est nécessaire d’installer certains modules PHP spécifiques, via le gestionnaire de paquet de votre distribution ou manuellement : php5-mysql pour la connectivité avec la (...)

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

    23 July 2014, by 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 July 2014, by 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 July 2014, by 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

    The post AppRTC : Google’s WebRTC test app and its parameters first appeared on ginger’s thoughts.

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