更改 WebRTC 流中的播放延迟

Question

I'm trying to cast a live MediaStream (Eventually from the camera) from peerA to peerB and I want peerB to receive the live stream in real time and then replay it with an added delay. Unfortunately in isn't possible to simply pause the stream and resume with play since it jump forward to the live moment.

So I have figured out that I can use MediaRecorder + SourceBuffer rewatch the live stream. Record the stream and append the buffers to MSE (SourceBuffer) and play it 5 seconds later. This works grate on the local device (stream). But when I try to use Media Recorder on the receivers MediaStream (from pc.onaddstream ) is looks like it gets some data and it's able to append the buffer to the sourceBuffer. however it dose not replay. sometime i get just one frame.

const [pc1, pc2] = localPeerConnectionLoop()
const canvasStream = canvas.captureStream(200)

videoA.srcObject = canvasStream
videoA.play()

// Note: using two MediaRecorder at the same time seem problematic
// But this one works
// stream2mediaSorce(canvasStream, videoB)
// setTimeout(videoB.play.bind(videoB), 5000)

pc1.addTransceiver(canvasStream.getTracks()[0], {
  streams: [ canvasStream ]
})

pc2.onaddstream = (evt) => {
  videoC.srcObject = evt.stream
  videoC.play()

  // Note: using two MediaRecorder at the same time seem problematic
  // THIS DOSE NOT WORK
  stream2mediaSorce(evt.stream, videoD)
  setTimeout(() => videoD.play(), 2000)
}

/**
 * Turn a MediaStream into a SourceBuffer
 * 
 * @param  {MediaStream}      stream   Live Stream to record
 * @param  {HTMLVideoElement} videoElm Video element to play the recorded video in
 * @return {undefined}
 */
function stream2mediaSorce (stream, videoElm) {
  const RECORDER_MIME_TYPE = 'video/webm;codecs=vp9'
  const recorder = new MediaRecorder(stream, { mimeType : RECORDER_MIME_TYPE })

  const mediaSource = new MediaSource()
  videoElm.src = URL.createObjectURL(mediaSource)
  mediaSource.onsourceopen = (e) => {
    sourceBuffer = mediaSource.addSourceBuffer(RECORDER_MIME_TYPE);

    const fr = new FileReader()
    fr.onerror = console.log
    fr.onload = ({ target }) => {
      console.log(target.result)
      sourceBuffer.appendBuffer(target.result)
    }
    recorder.ondataavailable = ({ data }) => {
      console.log(data)
      fr.readAsArrayBuffer(data)
    }
    setInterval(recorder.requestData.bind(recorder), 1000)
  }

  console.log('Recorder created')
  recorder.start() 
}

Do you know why it won't play the video?

I have created a fiddle with all the necessary code to try it out, the javascript tab is the same code as above, (the html is mostly irrelevant and dose not need to be changed)

Some try to reduce the latency, but I actually want to increase it to ~10 seconds to rewatch something you did wrong in a golf swing or something, and if possible avoid MediaRecorder altogether

EDIT: I found something called "playout-delay" in some RTC extension

that allows the sender to control the minimum and maximum latency from capture to render time

• https://webrtc.org/experiments/rtp-hdrext/playout-delay/

How can i use it? Will it be of any help to me?

Answer 1

Update, there is new feature that will enable this, called playoutDelayHint .

We want to provide means for javascript applications to set their preferences on how fast they want to render audio or video data. As fast as possible might be beneficial for applications which concentrates on real time experience. For others additional data buffering may provide smother experience in case of network issues.

Refs:
https://discourse.wicg.io/t/hint-attribute-in-webrtc-to-influence-underlying-audio-video-buffering/4038

https://bugs.chromium.org/p/webrtc/issues/detail?id=10287

Demo: https://jsfiddle.net/rvekxns5/ doe i was only able to set max 10s in my browser but it's more up to the UA vendor to do it's best it can with the resources available

 import('https://jimmy.warting.se/packages/dummycontent/canvas-clock.js') .then(({AnalogClock}) => { const {canvas} = new AnalogClock(100) document.querySelector('canvas').replaceWith(canvas) const [pc1, pc2] = localPeerConnectionLoop() const canvasStream = canvas.captureStream(200) videoA.srcObject = canvasStream videoA.play() pc1.addTransceiver(canvasStream.getTracks()[0], { streams: [ canvasStream ] }) pc2.onaddstream = (evt) => { videoC.srcObject = evt.stream videoC.play() } $dur.onchange = () => { pc2.getReceivers()[0].playoutDelayHint = $dur.valueAsNumber } })

 <!-- all the irrelevant part, that you don't need to know anything about --> <h3 style="border-bottom: 1px solid">Original canvas</h3> <canvas id="canvas" width="100" height="100"></canvas> <script> function localPeerConnectionLoop(cfg = {sdpSemantics: 'unified-plan'}) { const setD = (d, a, b) => Promise.all([a.setLocalDescription(d), b.setRemoteDescription(d)]); return [0, 1].map(() => new RTCPeerConnection(cfg)).map((pc, i, pcs) => Object.assign(pc, { onicecandidate: e => e.candidate && pcs[i ^ 1].addIceCandidate(e.candidate), onnegotiationneeded: async e => { try { await setD(await pc.createOffer(), pc, pcs[i ^ 1]); await setD(await pcs[i ^ 1].createAnswer(), pcs[i ^ 1], pc); } catch (e) { console.log(e); } } })); } </script> <h3 style="border-bottom: 1px solid">Local peer (PC1)</h3> <video id="videoA" muted width="100" height="100"></video> <h3 style="border-bottom: 1px solid">Remote peer (PC2)</h3> <video id="videoC" muted width="100" height="100"></video> <label> Change playoutDelayHint <input type="number" value="1" id="$dur"> </label>