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  • 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 (...)

  • Des sites réalisés avec MediaSPIP

    2 mai 2011, par

    Cette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
    Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page.

  • Les autorisations surchargées par les plugins

    27 avril 2010, par

    Mediaspip core
    autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs

Sur d’autres sites (6402)

  • How to split bulk video fast by using ffmpeg ?

    9 avril 2014, par user3513568

    I have a lot of videos, so I want to split them automatically. And they will be divided into 2 parts :

    • Part 1 : 15 minutes
    • Part 2 : the rest

    Searched a lot, but did not find. Please, help.

  • AAC's converted from CAFs for HTTP Live Streaming

    13 mars 2014, par user2901994

    I have several AAC files that were converted from CAF files, for use in HTTP Live Streaming. The stream works, however there is a small gap between each AAC file. It is my understanding that this gap is caused by the "Priming" and "Remainder" frames that are attached to AAC files when they are transcoded from CAFs.

    My question is, is there any way to remove this gap ? Or use FFMpeg to wrap the files, (possibly in m4a ?) so that audio players (VLC, JWPlayer) will understand to skip the gap ?

  • using pocketsphinx_continuous with a .wav file

    3 avril 2013, par user2242131

    I am attempting to write an application that will allow a user to speak a small set of commands from a remote system and have them executed on my server. Using pocketsphinx to parse the spoken text. When run locally with the microphone, pocketsphinx_continuous works perfectly no matter how I slur the words. But when importing the audio file and using ffmpeg to downsample the audio to a single channel, 16 bit PCM file, it will parse the first word without difficulty. Then it will skip everything else and treat it as . I am confident that the problem is in the file format and not in the pocketsphinx configuration.

    Using command line
    ffmpeg -y -i Sound\AddSheet.wav -ac 1 -f s16le -acodec pcm_s16le -ar 16k AddTmp.wav
    in a batch file.

    The bottom of the output I get is :

    INFO: fsg_search.c(1407): Start node ADD.0:5:47
    
INFO: fsg_search.c(1407): Start node <sil>.0:2:49
    
INFO: fsg_search.c(1446): End node <sil>.126:128:305 (-486)
    
INFO: fsg_search.c(1662): lattice start node <s>.0 end node <sil>.126
    
INFO: ps_lattice.c(1352): Normalizer P(O) = alpha(<sil>:126:305) = -175371
    
INFO: ps_lattice.c(1390): Joint P(O,S) = -176076 P(S|O) = -705
    
000000000: ADD USER
    
</sil></sil></s></sil></sil>

    Which is not the audio in the file. The words spoken in the file are "ADD SPREADSHEET", which works perfectly from the same microphone without the intervening .wav file.

    I have tried increasing the audio volume and decreasing the background noise using sox :

    sox -v 3.0 Sound\%1 Sound\%1-loud.wav ffmpeg -i Sound\%1-loud.wav -vn -ss 00:00:00 -t 00:00:01 -y Sound\%1-noiseaud.wav 
    
sox Sound\%1-noiseaud.wav -n noiseprof Sound\%1-noise.prof 
    
sox Sound\%1 Sound\%1-clean.wav noisered sound\noise.prof 0.21 
    
ffmpeg -y -i Sound\%1-clean.wav -ac 1 -f s16le -acodec pcm_s16le -ar 16k AddTmp.wav

    with no noticeable effect on the final results.

    If you look at the output you will notice that fsg_search.c has found ADD as the start node, then silence for the remainder. Please help on this.

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