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  • Submit bugs and patches

    13 avril 2011

    Unfortunately a software is never perfect.
    If you think you have found a bug, report it using our ticket system. Please to help us to fix it by providing the following information : the browser you are using, including the exact version as precise an explanation as possible of the problem if possible, the steps taken resulting in the problem a link to the site / page in question
    If you think you have solved the bug, fill in a ticket and attach to it a corrective patch.
    You may also (...)

  • 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 (...)

  • De l’upload à la vidéo finale [version standalone]

    31 janvier 2010, par

    Le chemin d’un document audio ou vidéo dans SPIPMotion est divisé en trois étapes distinctes.
    Upload et récupération d’informations de la vidéo source
    Dans un premier temps, il est nécessaire de créer un article SPIP et de lui joindre le document vidéo "source".
    Au moment où ce document est joint à l’article, deux actions supplémentaires au comportement normal sont exécutées : La récupération des informations techniques des flux audio et video du fichier ; La génération d’une vignette : extraction d’une (...)

Sur d’autres sites (5112)

  • ffmpeg from .mp3 to .al appears to slow and deepen the audio

    6 juillet 2020, par Darpeh

    I have an mp3 input file that is 22050Hz and 48kb/s. I need to convert it to a .al file in order to stream it to a camera on my network for output.

    


    ffmpeg -i file.mp3 file.al 


    


    creates a file that does play but the bitrate of the output is 176kb/s and it clearly slower/deeper sounding. I have tried -b:a 48k and -filter:a without success but I could be doing it wrong.

    


    Documentation states that this is the format I need : PCM ADPCM G.711A G.711.Mu G.726 G.729 MPEG2 AMR AAC

    


  • FFmpeg : how to access to RTP layer for FEC handling

    3 juin 2016, par Akon

    I need to receive to RTP streams of H.264 video (actually, an RTP payload is not significant for the question) : one stream is initial packets, and other is FEC packets (streams come to different ports). Then process them to recover loss packets resulting one output stream to be fed to a decoder.

    Also, arrived packets in both streams may be reordered due to network transmission, so some RTP-jitter preprocessing is required.

    I have this task completed with GS1treamer, but how can I do such processing with FFmpeg ?

    Thanks.

  • ffmpeg from .mp3 to .al appears to slow and deepen the audio. I've narrowed it down to bitrate - how to get -b:a to stick ?

    6 juillet 2020, par Darpeh

    I have an mp3 input file that is 22050Hz and 48kb/s. I need to convert it to a .al file in order to stream it to a camera on my network for output.

    


    ffmpeg -i file.mp3 file.al 


    


    creates a file that does play but the bitrate of the output is 176kb/s and it clearly slower/deeper sounding. I have tried -b:a 48k and -filter:a without success but I could be doing it wrong.