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Autres articles (51)

  • Encodage et transformation en formats lisibles sur Internet

    10 avril 2011

    MediaSPIP transforme et ré-encode les documents mis en ligne afin de les rendre lisibles sur Internet et automatiquement utilisables sans intervention du créateur de contenu.
    Les vidéos sont automatiquement encodées dans les formats supportés par HTML5 : MP4, Ogv et WebM. La version "MP4" est également utilisée pour le lecteur flash de secours nécessaire aux anciens navigateurs.
    Les documents audios sont également ré-encodés dans les deux formats utilisables par HTML5 :MP3 et Ogg. La version "MP3" (...)

  • 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.

  • 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 (...)

Sur d’autres sites (4577)

  • ffmpeg segment doesn't show file size update in real time

    9 juillet, par Lucas Cardoso

    I'm trying to run ffmpeg mp3 stream with segmentation for each hour. Everything is working perfectly, except for one thing : when i run the command, the file size doesn't grow in real-time as i need, it only grows in packages of 256k.

    



    Is there a way to turn a "real-time mode" ?

    



    I'm using ubuntu 18.04 with ffmpeg 3.4.6

    



    This is the code i'm trying to run on linux terminal :
    
ffmpeg -i http://radiocentova.conectastm.com:8363/stream -y -acodec libmp3lame -b:a 16k -ac 1 -ar 11025 -vn -strftime 1 -f segment -segment_time 3600 -flush_packets 1 @test_%Y%m%d%H%M%S+00.mp3

    



    
Recording with segment :
Recording with segment
Recording without segment :
Recording without segment

    


  • Randomly silencing part of input audio in real time

    3 septembre 2021, par Nobody

    My machine is running Ubuntu 20 LTS. I want to manipulate the input live audio in real-time. I have achieved pitch shifting using sox. The command being -

    


     sox -t pulseaudio default -t pulseaudio null pitch +1000


    


    and then routing the audio from "Monitor of Nullsink" .

    


    What I actually want to do is, silence randomized parts of the input audio, with a range. What I mean is, randomly mute 1-2s of the input audio.

    


    The final goal of this project will be to write a script that manipulates my voice and makes it seems like my network is bad.

    


    There is no restriction in method of achieving. That is we may use any language, make an extension, directly manipulate the input audio with sox, ffmpeg etc. Anything goes.

    


  • Add buffering to real time input stream with c++ ffmpeg

    7 octobre 2018, par Joel

    I am writing a c++ program in which I am handling a real time udp video stream with the ffmpeg library.

    The video input fps is 25 frames per second in avarage. The gap between two frames can be 10 , 20, 40 milliseconds, but sometimes it can be around 80 milliseconds.
    In those times, when the gap is around 80 milliseconds, the video seems choppy or stuck,

    When I open the same stream with the ffplay player(using the ffplay.exe), using a simple
    "ffplay.exe udp ://ip:port" command, the video has a little delay (around 50 millisends) with respect to the origin stream, but it runs much more smoothly.

    Can I set a buffering time or any minimal delay so that the delay between the frames will be much more steady, in my c++ program ?
    If so, how can I do this ?

    Thanks,
    Joel

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