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  • Configurer la prise en compte des langues

    15 novembre 2010, par

    Accéder à la configuration et ajouter des langues prises en compte
    Afin de configurer la prise en compte de nouvelles langues, il est nécessaire de se rendre dans la partie "Administrer" du site.
    De là, dans le menu de navigation, vous pouvez accéder à une partie "Gestion des langues" permettant d’activer la prise en compte de nouvelles langues.
    Chaque nouvelle langue ajoutée reste désactivable tant qu’aucun objet n’est créé dans cette langue. Dans ce cas, elle devient grisée dans la configuration et (...)

  • Le profil des utilisateurs

    12 avril 2011, par

    Chaque utilisateur dispose d’une page de profil lui permettant de modifier ses informations personnelle. Dans le menu de haut de page par défaut, un élément de menu est automatiquement créé à l’initialisation de MediaSPIP, visible uniquement si le visiteur est identifié sur le site.
    L’utilisateur a accès à la modification de profil depuis sa page auteur, un lien dans la navigation "Modifier votre profil" est (...)

  • 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 (...)

Sur d’autres sites (8726)

  • Volume Detect on half second window sizes [duplicate]

    14 juin 2021, par user82395214

    How can I get the max volume of individual slices of a video using FFMpeg ?

    


    Using this command, you can obtain the max volume for the whole clip :

    


    ffmpeg -i video.avi -af "volumedetect" -vn -sn -dn -f null /dev/null

    


    However, I want the max volume for every half second window in the audio file. How can I achieve this with ffmpeg ?

    


  • Google Speech API doesn't give correct result when audio is sent in file

    4 août 2012, par Cupidvogel

    I chanced upon the article at Google Speech API which suggested a mechanism for extracting text from audio file through Perl. Now I have recorded a audio file, which you will find at http://vocaroo.com/i/s0lPN5d3YQJj. It is a simple piece of audio, reading I love you. When I go to the Google speech API in Chrome, and speak those words, I get the right result. When I try the code at the above mentioned link with the audio file I pointed out, it returns strange results, like logan. How can I make it more accurate ? This is just a sample audio, what I am generally doing is extracting the audio from a video file through FFMpeg using something like ffmpeg -i input.avi -vn -ar 44100 -ac 2 -ab 192 -f mp3 output.mp3, followed by ffmpeg -i input.mp3 output.flac.

  • How to best decide what VM to use on google cloud ? Any best practices ? [closed]

    2 juillet 2024, par Prabhjot Kaur

    I have a script that reads google sheet for urls and then records those url videos, then merges it with my "test" video. both videos are about 3 minutes long. I am using e2-standard-8 Instance with ubuntu on it. Then running my script in node using puppeteer for recording and ffmpeg for merging videos. It takes 5 minutes for every video.

    


    My question is that should I run concurrent processed and use a stronger VM that will complete it in lesser time, or should i use a slow one ? It doesnt have to run 24/7, because I only have to generate certain amount of videos every week.

    


    Please provide the guidance that I need. Thanks in advance.

    


    I tried creating instance with more CPUs with free credits and ran out with them fairly quickly. I wonder if there is some other service i could use that will make the process faster ?

    


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