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Autres articles (104)

  • MediaSPIP 0.1 Beta version

    25 avril 2011, par

    MediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
    The zip file provided here only contains the sources of MediaSPIP in its standalone version.
    To get a working installation, you must manually install all-software dependencies on the server.
    If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)

  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
    The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
    For older browsers the Flowplayer flash fallback is used.
    MediaSPIP allows for media playback on major mobile platforms with the above (...)

  • Support audio et vidéo HTML5

    10 avril 2011

    MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
    Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
    Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
    Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)

Sur d’autres sites (13862)

  • Anomalie #2985 (Nouveau) : Les docs attachés à un auteur ne sont pas visibles pour les rédacs dans...

    26 avril 2013, par Suske -

    Voir http://forum.spip.net/fr_251736.html

    Je confirme.

    Par contre ils s’afficheraient dans le public, ce que je veux bien croire mais pas testé.

    Une question d’autorisation j’imagine.

  • how to upload a video to google driver use paperclip or carriwave

    14 janvier 2016, par bách trần nguyên

    i want to upload video to google driver.
    code models
    video model

    class Video < ActiveRecord::Base
     has_attached_file :video,
      :storage => :google_drive,
      :google_drive_credentials => {:client_id => AppConfig.gg_drive.client_id,
                                 :client_secret => AppConfig.gg_drive.client_secret,
                                 :refresh_token => AppConfig.gg_drive.refresh_token,
                                 :scope => AppConfig.gg_drive.scope,
                                 :access_token => Token.cache_access_token_google_drive
                                 },
     :styles => {
       :medium => {
         :geometry => "640x480",
         :format => 'mp4'
       },
       :thumb => { :geometry => "160x120", :format => 'jpeg', :time => 10}
     },# hello 123
     :processors => [:transcoder],
     :google_drive_options => {
       :path => proc { |style| "#{style}_#{id}_#{image.original_filename}" },
       :public_folder_id => '0B0VNyOkzIwUZZFFGeVhycFk0dnc'
     }
    end

    in Gemfile

    gem 'google-api-client'
    gem 'paperclip'
    gem 'paperclip-googledrive'
    gem 'paperclip-av-transcoder'
    gem "paperclip-ffmpeg"

    in controller

    def create
       if params[:videos]
         params[:videos].each { |video| Video.create(video: video) }
       end
    end

    when i run , this display error

    [AV] Running command : if command -v avprobe 2>/dev/null ; then echo "true" ; else echo "false" ; fi
    [AV] Running command : if command -v ffmpeg 2>/dev/null ; then echo "true" ; else echo "false" ; fi
    Av::UnableToDetect in AlbumsController#create
    Unable to detect any supported library

    pls. how to fix this errors

  • 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

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