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  • 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 (...)

  • ANNEXE : Les plugins utilisés spécifiquement pour la ferme

    5 mars 2010, par

    Le site central/maître de la ferme a besoin d’utiliser plusieurs plugins supplémentaires vis à vis des canaux pour son bon fonctionnement. le plugin Gestion de la mutualisation ; le plugin inscription3 pour gérer les inscriptions et les demandes de création d’instance de mutualisation dès l’inscription des utilisateurs ; le plugin verifier qui fournit une API de vérification des champs (utilisé par inscription3) ; le plugin champs extras v2 nécessité par inscription3 (...)

Sur d’autres sites (7965)

  • 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.

  • 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 ?

  • 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.

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