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• 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 ?

  


• 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.

• what protocol must use for broadcast live video ?

  19 août 2021, par user3807476

  A server is in the middle
And we want you to send live video to it

  


  And on the other hand, watch it through the HTTPS like https://server/live.pm4

  


  

  


  What protocols can be used for this purpose ?

  


  I used to do this experimentally with nodejs dgram and ffmpeg on the raw UDP and it worked fine !
but stability and security is an issue that must be observed !