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webrtc视频内容传输

程序员文章站 2022-07-01 15:34:30
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1、main.js

核心代码:

  const videoTracks = stream.getVideoTracks();
  const audioTracks = stream.getAudioTracks();
  if (videoTracks.length > 0) {
    console.log(`Using video device: ${videoTracks[0].label}`);
  }
  if (audioTracks.length > 0) {
    console.log(`Using audio device: ${audioTracks[0].label}`);
  }
  const servers = null;
  pc1 = new RTCPeerConnection(servers);
  console.log('Created local peer connection object pc1');
  pc1.onicecandidate = e => onIceCandidate(pc1, e);
  pc2 = new RTCPeerConnection(servers);
  console.log('Created remote peer connection object pc2');
  pc2.onicecandidate = e => onIceCandidate(pc2, e);
  pc1.oniceconnectionstatechange = e => onIceStateChange(pc1, e);
  pc2.oniceconnectionstatechange = e => onIceStateChange(pc2, e);
  pc2.ontrack = gotRemoteStream;

  stream.getTracks().forEach(track => pc1.addTrack(track, stream));
  console.log('Added local stream to pc1');

  console.log('pc1 createOffer start');
  pc1.createOffer(onCreateOfferSuccess, onCreateSessionDescriptionError, offerOptions);
/*
*  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
*
*  Use of this source code is governed by a BSD-style license
*  that can be found in the LICENSE file in the root of the source
*  tree.
*/

'use strict';

const leftVideo = document.getElementById('leftVideo');
const rightVideo = document.getElementById('rightVideo');

let stream;

let pc1;
let pc2;
const offerOptions = {
  offerToReceiveAudio: 1,
  offerToReceiveVideo: 1
};

let startTime;

function maybeCreateStream() {
  if (stream) {
    return;
  }
  if (leftVideo.captureStream) {
    stream = leftVideo.captureStream();
    console.log('Captured stream from leftVideo with captureStream',
      stream);
    call();
  } else if (leftVideo.mozCaptureStream) {
    stream = leftVideo.mozCaptureStream();
    console.log('Captured stream from leftVideo with mozCaptureStream()',
      stream);
    call();
  } else {
    console.log('captureStream() not supported');
  }
}

// Video tag capture must be set up after video tracks are enumerated.
leftVideo.oncanplay = maybeCreateStream;
if (leftVideo.readyState >= 3) { // HAVE_FUTURE_DATA
  // Video is already ready to play, call maybeCreateStream in case oncanplay
  // fired before we registered the event handler.
  maybeCreateStream();
}

leftVideo.play();

rightVideo.onloadedmetadata = () => {
  console.log(`Remote video videoWidth: ${this.videoWidth}px,  videoHeight: ${this.videoHeight}px`);
};

rightVideo.onresize = () => {
  console.log(`Remote video size changed to ${rightVideo.videoWidth}x${rightVideo.videoHeight}`);
  // We'll use the first onresize callback as an indication that
  // video has started playing out.
  if (startTime) {
    const elapsedTime = window.performance.now() - startTime;
    console.log('Setup time: ' + elapsedTime.toFixed(3) + 'ms');
    startTime = null;
  }
};

function call() {
  console.log('Starting call');
  startTime = window.performance.now();
  const videoTracks = stream.getVideoTracks();
  const audioTracks = stream.getAudioTracks();
  if (videoTracks.length > 0) {
    console.log(`Using video device: ${videoTracks[0].label}`);
  }
  if (audioTracks.length > 0) {
    console.log(`Using audio device: ${audioTracks[0].label}`);
  }
  const servers = null;
  pc1 = new RTCPeerConnection(servers);
  console.log('Created local peer connection object pc1');
  pc1.onicecandidate = e => onIceCandidate(pc1, e);
  pc2 = new RTCPeerConnection(servers);
  console.log('Created remote peer connection object pc2');
  pc2.onicecandidate = e => onIceCandidate(pc2, e);
  pc1.oniceconnectionstatechange = e => onIceStateChange(pc1, e);
  pc2.oniceconnectionstatechange = e => onIceStateChange(pc2, e);
  pc2.ontrack = gotRemoteStream;

  stream.getTracks().forEach(track => pc1.addTrack(track, stream));
  console.log('Added local stream to pc1');

  console.log('pc1 createOffer start');
  pc1.createOffer(onCreateOfferSuccess, onCreateSessionDescriptionError, offerOptions);
}

function onCreateSessionDescriptionError(error) {
  console.log(`Failed to create session description: ${error.toString()}`);
}

function onCreateOfferSuccess(desc) {
  console.log(`Offer from pc1
${desc.sdp}`);
  console.log('pc1 setLocalDescription start');
  pc1.setLocalDescription(desc, () => onSetLocalSuccess(pc1), onSetSessionDescriptionError);
  console.log('pc2 setRemoteDescription start');
  pc2.setRemoteDescription(desc, () => onSetRemoteSuccess(pc2), onSetSessionDescriptionError);
  console.log('pc2 createAnswer start');
  // Since the 'remote' side has no media stream we need
  // to pass in the right constraints in order for it to
  // accept the incoming offer of audio and video.
  pc2.createAnswer(onCreateAnswerSuccess, onCreateSessionDescriptionError);
}

function onSetLocalSuccess(pc) {
  console.log(`${getName(pc)} setLocalDescription complete`);
}

function onSetRemoteSuccess(pc) {
  console.log(`${getName(pc)} setRemoteDescription complete`);
}

function onSetSessionDescriptionError(error) {
  console.log(`Failed to set session description: ${error.toString()}`);
}

function gotRemoteStream(event) {
  if (rightVideo.srcObject !== event.streams[0]) {
    rightVideo.srcObject = event.streams[0];
    console.log('pc2 received remote stream', event);
  }
}

function onCreateAnswerSuccess(desc) {
  console.log(`Answer from pc2:
${desc.sdp}`);
  console.log('pc2 setLocalDescription start');
  pc2.setLocalDescription(desc, () => onSetLocalSuccess(pc2), onSetSessionDescriptionError);
  console.log('pc1 setRemoteDescription start');
  pc1.setRemoteDescription(desc, () => onSetRemoteSuccess(pc1), onSetSessionDescriptionError);
}

function onIceCandidate(pc, event) {
  getOtherPc(pc).addIceCandidate(event.candidate)
    .then(
      () => onAddIceCandidateSuccess(pc),
      err => onAddIceCandidateError(pc, err)
    );
  console.log(`${getName(pc)} ICE candidate: 
${event.candidate ?
    event.candidate.candidate : '(null)'}`);
}

function onAddIceCandidateSuccess(pc) {
  console.log(`${getName(pc)} addIceCandidate success`);
}

function onAddIceCandidateError(pc, error) {
  console.log(`${getName(pc)} failed to add ICE Candidate: ${error.toString()}`);
}

function onIceStateChange(pc, event) {
  if (pc) {
    console.log(`${getName(pc)} ICE state: ${pc.iceConnectionState}`);
    console.log('ICE state change event: ', event);
  }
}

function getName(pc) {
  return (pc === pc1) ? 'pc1' : 'pc2';
}

function getOtherPc(pc) {
  return (pc === pc1) ? pc2 : pc1;
}

2、index.html

<!DOCTYPE html>
<!--
 *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree.
-->
<html>
<head>

    <meta charset="utf-8">
    <meta name="description" content="WebRTC code samples">
    <meta name="viewport" content="width=device-width, user-scalable=yes, initial-scale=1, maximum-scale=1">
    <meta itemprop="description" content="Client-side WebRTC code samples">
    <meta itemprop="image" content="../../../images/webrtc-icon-192x192.png">
    <meta itemprop="name" content="WebRTC code samples">
    <meta name="mobile-web-app-capable" content="yes">
    <meta id="theme-color" name="theme-color" content="#ffffff">

    <base target="_blank">

    <title>Video to peer connection</title>

    <link rel="icon" sizes="192x192" href="../../../images/webrtc-icon-192x192.png">
    <link href="//fonts.googleapis.com/css?family=Roboto:300,400,500,700" rel="stylesheet" type="text/css">
    <link rel="stylesheet" href="main.css"/>
    <link rel="stylesheet" href="css/main.css"/>

</head>

<body>

<div id="container">

    <h1><a href="//webrtc.github.io/samples/" title="WebRTC samples homepage">WebRTC samples</a> <span>Stream from a video to a peer connection</span>
    </h1>

    <video id="leftVideo" playsinline controls muted>
        <!-- NB CORS: https://bugzilla.mozilla.org/show_bug.cgi?id=1177793 -->
        <source src="chrome.webm" type="video/webm"/>
        <source src="chrome.mp4" type="video/mp4"/>
        <p>This browser does not support the video element.</p>
    </video>

    <video id="rightVideo" playsinline autoplay controls></video>

    <p>This demo requires Firefox 47, Chrome 53 with <strong>Experimental Web Platform features</strong> enabled from
        <tt>chrome://flags</tt>.</p>

    <p>A stream is captured from the video on the left using the <code>captureStream()</code> method, and streamed via a
        peer connection to the video element on the right.</p>

    <p>View the browser console to see logging.</p>

    <p>Several variables are in global scope, so you can inspect them from the console: <code>localPeerConnection</code>,
        <code>remotePeerConnection</code> and <code>stream</code>.</p>

    <p>For more information about RTCPeerConnection, see <a href="http://www.html5rocks.com/en/tutorials/webrtc/basics/"
                                                            title="HTML5 Rocks article about WebRTC by Sam Dutton">Getting
        Started With WebRTC</a>.</p>

    <a href="https://github.com/webrtc/samples/tree/gh-pages/src/content/capture/video-pc"
       title="View source for this page on GitHub" id="viewSource">View source on GitHub</a>

</div>

<script src="https://webrtc.github.io/adapter/adapter-latest.js"></script>
<script src="main6.js" async></script>

<script src="ga.js"></script>
</body>
</html>

 

相关标签: webrtc