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Autres articles (11)

  • Support de tous types de médias

    10 avril 2011

    Contrairement à beaucoup de logiciels et autres plate-formes modernes de partage de documents, MediaSPIP a l’ambition de gérer un maximum de formats de documents différents qu’ils soient de type : images (png, gif, jpg, bmp et autres...) ; audio (MP3, Ogg, Wav et autres...) ; vidéo (Avi, MP4, Ogv, mpg, mov, wmv et autres...) ; contenu textuel, code ou autres (open office, microsoft office (tableur, présentation), web (html, css), LaTeX, Google Earth) (...)

  • 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 (...)

  • Support audio et vidéo HTML5

    10 avril 2011

    MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
    Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
    Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
    Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)

Sur d’autres sites (2803)

  • FFmpeg : AAC muxing into two different files in the same loop : The 2nd file is corrupted [on hold]

    1er septembre 2013, par user2677612

    We are using FFmpeg libraries git-ee94362 libavformat v55.2.100.
    Our purpose is to mux synthetic video and audio-file audio to 2 different files (in the same loop)
    with the the same format and parameters.

    With MP3 input, both M3U8 outputs are OK :

    Output file #0 : Total number of frames is : video = 229, audio = 347.

    Output file #1 : Total number of frames is : video = 229, audio = 347.

    But with AAC input, the 1st M3U8 output is OK, whereas the 2nd one - is not :

    Output file #0 : Total number of frames is : video = 403, audio = 690.

    Output file #1 : Total number of frames is : video = 3, audio = 690.

    It is exactly the same code in the main loop. We do not use the "aac_adtstoasc" filter in case of M3U output.
    By the way, the problem exist both with and without the filter,
    since in case of MP4 output, the 1st output file is OK, whereas the 2nd one - is not, - as well.

    Moreover :

    1) I have implemented the main loop as Windows threads, separated for every output file.

    2) I have separated input files, as well (i.e. use two identical input files).
    The problem remains !

    It seems that video PTS begins to be corrupted (wrong-calculated) for the 2nd (3rd, etc.) output file.

    If somebody is known about such a problem ? If yes, what can be a solution/workaround ?
    Thank you.

  • ffmpeg stops pushing stream of h264 frames to the media server after 180s (3 minutes)

    19 décembre 2019, par blvvet

    My purpose is to let tons of h264 frames play on the browser.

    What I try to do is pushing these frames to the node-media-server(I also tried srs, still breaks down at 180s..So I think node-media-server should not be the problem) by ffmpeg and play it on the browser by flv.js. It works fine for the first 180 seconds and It always breaks down at 180s.

    The node process keeps alive after 180s and the node-media-server receives nothing after 180s. So I think the problem may caused on ffmpeg. Since the number is always 180 seconds, I think maybe I did something wrong about the parameters of ffmpeg.

    Thanks to everyone who gives a glance at my question
    Here’s the code from the nodejs side :

    const spawn = require('child_process').spawn
    const GET_FRAME = require('XXX') // just a cpp addon to get frame buffer encoded of h264
    let args = [
       '-i', 'pipe:0',
       '-use_wallclock_as_timestamps', '1',
       '-max_delay', '5',
       '-r', '25',
       '-c', 'copy',
       '-preset:v', 'ultrafast',
       '-tune:v', 'zerolatency',
       '-f', 'flv',
       '-g', '5',
       'rtmp://127.0.0.1/live/test'
    ]
    let $ffmpeg = spawn('ffmpeg', args)
    setInterval(() => {
       let $FrameH264 = GET_FRAME()
       console.log($FrameH264) // buffers keep printing after 180s
       $ffmpeg.stdin.write($FrameH264)
    }, 1000 / 30)
  • avformat/mpegtsenc : fix flushing of audio packets

    26 août 2019, par Marton Balint
    avformat/mpegtsenc : fix flushing of audio packets
    

    7d097a0fc57f0fa8385962a539c657c2f40b5ed0 had the same purpose as
    3700f655c55e2001b57215210b957b169d66b50f but the former is much simpler, so
    let's remove the latter.

    Unfortunately both checks were wrong, because in order to make sure DTS > PCR
    we have to give us some headroom, so instead of using a dts_difference <
    max_delay check let's use a dts_difference < max_delay/2 check.

    Fixes DTS < PCR errors with this command line :

    ./ffmpeg -loglevel verbose -y -f lavfi -i \
    "testsrc=s=64x64:d=20,split=2[out0][tmp1] ;[tmp1]vflip[out1] ;sine=d=20,asetnsamples=1000[out2]" \
    -flags +bitexact -fflags +bitexact -sws_flags +accurate_rnd+bitexact \
    -codec:v libx264 -codec:a mp2 -b:a 32k -pix_fmt yuv420p \
    -map '0:v:0' \
    -map '0:v:1' \
    -map '0:a:0' \
    -muxrate 800000 \
    -program st=0:st=2 -program st=1:st=2 -program st=2 -program st=0 -f mpegts out1.ts

    Signed-off-by : Marton Balint <cus@passwd.hu>

    • [DH] libavformat/mpegtsenc.c