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Autres articles (90)

  • Encoding and processing into web-friendly formats

    13 avril 2011, par

    MediaSPIP automatically converts uploaded files to internet-compatible formats.
    Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
    Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
    Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
    All uploaded files are stored online in their original format, so you can (...)

  • Websites made ​​with MediaSPIP

    2 mai 2011, par

    This page lists some websites based on MediaSPIP.

  • Publier sur MédiaSpip

    13 juin 2013

    Puis-je poster des contenus à partir d’une tablette Ipad ?
    Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir

Sur d’autres sites (6508)

  • avcodec/libx265 : add support for setting chroma sample location

    29 août 2021, par Jan Ekström
    avcodec/libx265 : add support for setting chroma sample location

    Unlike libx264, libx265 does not handle the chroma format check
    
on its own side, so in order to not write out values which are
    
supposed to be ignored according to the specification, we limit
    
the writing out of chroma sample location to 4:2:0 only.
    • [DH] libavcodec/libx265.c

  • hevc : Fix 4K sample video

    14 juin 2014, par Kieran
    hevc : Fix 4K sample video

    Reviewed-by : smarter
    
Signed-off-by : Michael Niedermayer <michaelni@gmx.at>
    • [DH] libavcodec/hevc_ps.c

  • ffmpeg disconnects with different audio sample rate

    11 juillet 2017, par HRG

    Am using ffmpeg transcoder tool to convert a streaming audio input to another format.The raw audio data comes from a input device in wav format (.wmv) which can be accessed via server port (ex.8080) . The audio format is in wav with 16bits/sample and sample rate of 6000 hz.
    When I use ffmpeg to read this audio input with above specs and convert to another format it works fine.

    Ex.
    ffmpeg -f s16le -ar 6000 -ac 1 -i http://local:8080/sampleaudio.wmv -f webm out.webm

    But if i use fmmpeg to read the audio input at sample rate of 4000hz (other than the actual rate) ,it disconnects from the server always .

    Ex.
    ffmpeg -f s16le -ar 4000 -ac 1 -i http://local:8080/sampleaudio.wmv -f webm out.webm

    The error in the ffmpeg console is "Stream ends prematurely at 194604, should be 4800044"

    I know the input rate at which ffmpeg is reading and the actual audio output is different .But am curious to know

    why ffmpeg is disconnecting from the server ,what is happening behind the scene.

    Also how to enable logs for network connections(http,sockets..)

    Thanks

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