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• Given 2 input videos of different lengths at 60 FPS with multiple common sections in between, what is the best way to sync them at the common section ?

  27 janvier 2024, par PirateApp

  
input vs output

  


  

  • In my actual case, there are 4 videos (basically gaming videos recorded from 4 different machines)

  


  • Each video may start or end at different duration (manually controlled by people starting and stopping recording and therefore their lengths vary slightly)

  


  • Each video has one or more common sections (basically a cutscene playing at the same time)

  


  • The distance between the common sections is the same in every video

  


  • How do I sync the videos at the common section ?

  


  • Basically I want to create a split screen video such that the common part plays at the same time on all the sub parts of the video

  


  • Keep in mind that the common sections wont be EXACTLY equal (because of graphics differences between machines but highly similar like dx9 vs dx11 or gamma)
What have I tried ?

  


  • Naive approach

  


  • Start video 1 at 0, start video 2 at 0.017, render the whole thing and check if it is in sync

  


  • Start video 1 at 0, start video 2 at 0.033, render the whole thing and check if it is in sync

  


  • Basically loop through the length of the video at each frame interval, render the full video and check if we get a synced version

  


  


  This sounds like a terrible idea honestly

  


    for (t1 in length of video a) {
    for (t2 in length of video b) {
       render a video with a and b starting at t1, t2 respectively
       increment t2 by 0.017 for b
    }
    increment t1 by 0.017 for a
  }


  


  Final output with 4 videos should look like this all synced at common sections


  


  Any better ideas ?

  


• varying RTP stream result from custom SIP implementation

  1er février, par Nik Hendricks

  I am in the process of creating my own SIP implementation in Node.js. As well as a b2bua as a learning project.

  


  Finding people wise in the ways of SIP has proved to be difficult elsewhere but here I have had good results

  


  this is the GitHub of my library so far node.js-sip

  


  this is the GitHub of my PBX so far FlowPBX

  


  Currently, everything is working as I expect. Although I really have some questions on possible errors in my implementation.

  


  My main issue is with RTP streams. Currently I am utilizing ffmpeg.

  


  my function goes as follows

  


  start_stream(call_id, sdp){
        console.log('Starting Stream')
        let port = sdp.match(/m=audio (\d+) RTP/)[1];
        let ip = sdp.match(/c=IN IP4 (\d+\.\d+\.\d+\.\d+)/)[1];
        let codec_ids = sdp.match(/m=audio \d+ RTP\/AVP (.+)/)[1].split(' ');
        let ffmpeg_codec_map = {
            'opus': 'libopus',
            'PCMU': 'pcm_mulaw',
            'PCMA': 'pcm_alaw',
            'telephone-event': 'pcm_mulaw',
            'speex': 'speex',
            'G722': 'g722',
            'G729': 'g729',
            'GSM': 'gsm',
            'AMR': 'amr',
            'AMR-WB': 'amr_wb',
            'iLBC': 'ilbc',
            'iSAC': 'isac',
        }

        let codecs = [];
        sdp.split('\n').forEach(line => {
            if(line.includes('a=rtpmap')){
                let codec = line.match(/a=rtpmap:(\d+) (.+)/)[2];
                let c_id = line.match(/a=rtpmap:(\d+) (.+)/)[1];
                codecs.push({                    
                    name: codec.split('/')[0],
                    rate: codec.split('/')[1],
                    channels: codec.split('/')[2] !== undefined ? codec.split('/')[2] : 1,
                    id: c_id
                })
            }
        })

        console.log('codecs')
        console.log(codecs)

        let selected_codec = codecs[0]
        if(selected_codec.name == 'telephone-event'){
            selected_codec = codecs[1]
            console.log(selected_codec)
        }

        //see if opus is available
        codecs.forEach(codec => {
            if(codec.name == 'opus'){
                selected_codec = codec;
            }
        })

        if(selected_codec.name != 'opus'){
            //check if g729 is available
            codecs.forEach(codec => {
                if(codec.name == 'G729'){
                    selected_codec = codec;
                }
            })
        }

        console.log('selected_codec')
        console.log(selected_codec)

        let spawn = require('child_process').spawn;
        let ffmpegArgs = [
            '-re',
            '-i', 'song.mp3',
            '-acodec', ffmpeg_codec_map[selected_codec.name],
            '-ar', selected_codec.rate,
            '-ac', selected_codec.channels,
            '-payload_type', selected_codec.id,
            '-f', 'rtp', `rtp://${ip}:${port}`
        ];

        let ffmpeg = spawn('ffmpeg', ffmpegArgs);

        ffmpeg.stdout.on('data', (data) => {
            console.log(`stdout: ${data}`);
        });
        ffmpeg.stderr.on('data', (data) => {
            console.error(`stderr: ${data}`);
        });




}


  


  When using zoiper to test it works great. I have seen the mobile version negotiate speex
and the desktop version negotiate opus mostly for the codec.

  


  today I tried to register a grandstream phone to my pbx and the rtp stream is blank audio.
opus is available and I have tried to prefer that in my stream but still even when selecting that I cannot get audio to the grandstream phone. This is the same case for a yealink phone. I can only get zoiper to work so far.

  


  what could be causing this behavior ? there is a clear path of communication between everything just like the zoiper client's I have used.

  


  Additionally in my sip implementation,
how important is the concept of a dialog ? currently, I just match messages by Call-ID

  


  and then choose what to send based on the method or response. is there any other underlying dialog functionality that I may need to implement ?

  


  It would just be awesome to get someone who really knows what they are talking about eyes on some of my code to direct this large codebase in the right direction but I realize that a big ask lol.

  


• Revision 36889 : Le début d’une page d’info concernant la configuration de FFMPEG sur le ...

  3 avril 2010, par kent1@… — Log

  Le début d’une page d’info concernant la configuration de FFMPEG sur le serveur.
  On vire le PHP du squelette du formulaire de configuration
  On prépare le passage aux pressets