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<!DOCTYPE html>
<!--
* Copyright (c) 2014 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 Javascript code samples">
<meta name="viewport" content="width=device-width, user-scalable=no, initial-scale=1, maximum-scale=1">
<meta itemprop="description" content="Client-side WebRTC code samples">
<meta itemprop="image" content="src/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>WebRTC samples</title>
<link rel="icon" sizes="192x192" href="src/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="src/css/main.css" />
<style>
h2 {
font-size: 1.5em;
font-weight: 500;
}
h3 {
border-top: none;
}
section {
border-bottom: 1px solid #eee;
margin: 0 0 1.5em 0;
padding: 0 0 1.5em 0;
}
section:last-child {
border-bottom: none;
margin: 0;
padding: 0;
}
</style>
</head>
<body>
<div id="container">
<h1>WebRTC samples</h1>
<section>
<p>This is a repository for the WebRTC Javascript code samples. The source for these samples is available at <a href="//github.com/webrtc/samples" title="View GitHub repository for these files">github.com/webrtc/samples</a>.</p>
<p>Some of the samples use new browser features. They may only work in <a href="//www.google.com/chrome/browser/canary.html" title="Download Chrome Canary">Chrome Canary</a> and/or <a href="http://www.mozilla.org/firefox/beta/" title="Download Firefox Beta">Firefox Beta</a>, and may require flags to be set.</p>
<p>Most of the samples use <a href="//github.com/webrtc/adapter">adapter.js</a>, a shim to insulate apps from spec changes and prefix differences. (In fact, the standards and protocols used for WebRTC implementations are highly stable, and there are only a few prefixed names. For full interop information, see <a href="//www.webrtc.org/web-apis/interop">webrtc.org/web-apis/interop</a>.)</p>
<p>Please note that all samples that use <code>getUserMedia()</code> must be run from a server. Calling <code>getUserMedia()</code> from a file:// URL will result in a PERMISSION_DENIED NavigatorUserMediaError.</p>
<p><a href="http://www.webrtc.org/testing" title="Command-line flags for WebRTC testing">webrtc.org/testing</a> lists command line flags useful for development and testing with Chrome.</p>
<p>For more information about WebRTC, we maintain a list of <a href="//docs.google.com/document/d/1idl_NYQhllFEFqkGQOLv8KBK8M3EVzyvxnKkHl4SuM8/edit">WebRTC Resources</a>. If you've never worked with WebRTC, we recommend you start with the 2013 Google I/O <a href="//www.youtube.com/watch?v=p2HzZkd2A40">WebRTC presentation</a>.</p>
<p>Patches and issues welcome! See <a href="https://github.com/webrtc/samples/blob/master/CONTRIBUTING.md">CONTRIBUTING.md</a> for instructions. The <a href="https://bit.ly/webrtcdevguide">Developer's Guide</a> for this repo has more information about code style, structure and validation.</p>
</section>
<section>
<h2 id="the-demos">The demos</h2>
<h3 id="getusermedia">getUserMedia</h3>
<p><a href="src/content/getusermedia/gum">Basic getUserMedia demo</a></p>
<p><a href="src/content/getusermedia/canvas">Use getUserMedia with canvas</a></p>
<p><a href="src/content/getusermedia/filter">Use getUserMedia with canvas and CSS filters</a></p>
<p><a href="src/content/getusermedia/resolution">Choose camera resolution</a></p>
<p><a href="src/content/getusermedia/source">Choose camera and microphone</a></p>
<p><a href="src/content/getusermedia/audio">Audio-only getUserMedia() output to local audio element</a></p>
<p><a href="src/content/getusermedia/volume">Audio-only getUserMedia() displaying volume</a></p>
<p><a href="src/content/getusermedia/face">Face tracking, using getUserMedia and canvas</a></p>
<h3 id="peerconnection">RTCPeerConnection</h3>
<p><a href="src/content/peerconnection/pc1">Basic peer connection demo</a></p>
<p><a href="src/content/peerconnection/audio">Audio-only peer connection demo</a></p>
<p><a href="src/content/peerconnection/multiple">Multiple peer connections at once</a></p>
<p><a href="src/content/peerconnection/multiple-relay">Forward the output of one PC into another</a></p>
<p><a href="src/content/peerconnection/munge-sdp">Munge SDP parameters</a></p>
<p><a href="src/content/peerconnection/pr-answer">Use pranswer when setting up a peer connection</a></p>
<p><a href="src/content/peerconnection/constraints">Constraints and stats</a></p>
<p><a href="src/content/peerconnection/create-offer">Display createOffer output for various scenarios</a></p>
<p><a href="src/content/peerconnection/dtmf">Use RTCDTMFSender</a></p>
<p><a href="src/content/peerconnection/states">Display peer connection states</a></p>
<p><a href="src/content/peerconnection/trickle-ice">ICE candidate gathering from STUN/TURN servers</a></p>
<p><a href="src/content/peerconnection/webaudio-input">Web Audio output as input to peer connection</a></p>
<h3 id="datachannel">RTCDataChannel</h3>
<p><a href="src/content/datachannel/basic">Transmit text</a></p>
<p><a href="src/content/datachannel/filetransfer">Transfer a file</a></p>
<h3 id="videoChat">Video chat</h3>
<p><a href="//apprtc.appspot.com">AppRTC video chat client</a> powered by Google App Engine</p>
<p><a href="//apprtc.appspot.com/html/params.html">AppRTC URL parameters</a></p>
</section>
<a href="//github.com/webrtc/samples" title="View the repository" id="viewSource">github.com/webrtc/samples</a>
</div>
<script src="src/js/lib/ga.js"></script>
</body>
</html>