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Autres articles (109)

  • Websites made ​​with MediaSPIP

    2 mai 2011, par

    This page lists some websites based on MediaSPIP.

  • 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.

  • 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 (12944)

  • FFmpeg RTP_Mpegts over RTP protocol

    7 mars 2020, par Nicolò

    I’m tryin to implement a client/server application based on FFmpeg. Unfortunately RTP_MPEGTS isn’t documented in the official FFmpeg Documentation - Formats.
    Anyway i found inspiration from this old thread.

    Server Side

    (1) Capture mic audio as input. (2)Encode it as pcm 8khz mono and (3) send it locally as RTP_MPEGTS format over rtp protocol.

    ffmpeg -f avfoundation -i none:2  -ar 8000 -acodec pcm_u8 -ac 1 -f rtp_mpegts rtp://127.0.0.1:41954
    • This works, but on initiation it alerts "[mpegts @ 0x7fda13024600] frame size not set"

    Client Side (on the same machine)

    (1) Receive rtp audio stream input (2) write it in a file or playback.

    ffmpeg -i rtp://127.0.0.1:41954 -vcodec copy -y "output.wav"
    • I’m using -vcodec copy because i’ve already verified it in another rtp stream in which -acodec copy didn’t work.
    • This stuck and while closing with Ctrl+C shortcut it prints :

      Input #0, rtp, from 'rtp://127.0.0.1:41954':
      Duration: N/A, start: 8.956122, bitrate: N/A
      Program 1
      Metadata:
       service_name    : Service01
       service_provider: FFmpeg
      Stream #0:0: Data: bin_data ([6][0][0][0] / 0x0006)
      Output #0, wav, to 'output.wav':
      Output file #0 does not contain any stream

    1. I don’t understand if the client didn’t receive any stream, or it cannot write rtp packets into "output.wav" file. (Client or server problem ?)
    2. In the old thread is explained a workaround. On server could run 2 ffmpeg instance :
      One produces "tmp.ts" file due to mpegts, and the other takes "tmp.ts" as input and streams it over rtp. Is it possibile ?

    3. Is there any better way to do implement this client/server with the lowest latency possible ?


    Thanks for any help provided.

  • Page.startScreencast Chrome DevTools Protocol low FPS issue

    31 mars 2022, par Murat Colyaran

    With puppeteer I'm using Chrome DevTools Protocol Page.startScreencast to taking screenshots of webpages and converting those screenshots to a video with FFMPEG

    


    This is a simplified version of the tool I created :

    


    const browser = await puppeteer.launch({
     headless: true,
});
const page = await browser.newPage();
page.setViewport({
    width: 678,
    height: 1080,
});

await page.goto(`https://youtu.be/7G6FfKXvZX8`,
     {
        waitUntil: 'networkidle0',
     }
);

const client = await this.page.target().createCDPSession();
client.on('Page.screencastFrame', async (frameObject) => {

   await writeImageFilename(frameObject.data);
   await client.send('Page.screencastFrameAck', {
      sessionId: frameObject.sessionId,
   });
}

client.send('Page.startScreencast', {
   format: 'png',
   quality: 100,
   maxWidth: 678,
   maxHeight: 1080,
   everyNthFrame: 1,
});
await page.waitForTimeout(6000);
client.send('Page.stopScreencast');

// Helper func
async writeImageFilename(data) {
   const filename = path.join(
      '/tmp',
      Date.now().toString() + '.png'
    );
    fs.writeFileSync(filename, data, 'base64');
    return filename;
}


    


    (I didn't include the ffmpeg part to keep it simple.)

    


    The script above basicly goes to youtube and takes 60 screeenshots in 6 seconds. This way I can create a FFMPEG 10 fps video.

    


    But that's the problem. Capturing 60 frames in 6 seconds is a very slow performance.

    


    Is there any way to increase this number ?

    


  • avformat : implement SChannel SSP TLS protocol

    4 octobre 2015, par Hendrik Leppkes
    avformat : implement SChannel SSP TLS protocol
    

    This implementation does not support TLS listen sockets and loading
    CA/Certs from files.

    The Windows API does not support loading PEM certs, and would either
    require a manual loader or instead be limited to loading Windows PFX
    certificates

    TLS listen sockets would have to be implemented quite separately, as many
    of the APIs are different for server-mode (as opposed to client mode).

    • [DH] configure
    • [DH] libavformat/Makefile
    • [DH] libavformat/allformats.c
    • [DH] libavformat/tls.h
    • [DH] libavformat/tls_schannel.c