index.html - external/webrtc-samples - Git at Google

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     <title>WebRTC samples</title>

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     <h1>WebRTC samples</h1>

     <section>

         <p>
             This is a collection of small samples demonstrating various parts of the <a
                 href="https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API">WebRTC APIs</a>. The code for all
             samples are available in the <a href="https://github.com/webrtc/samples">GitHub repository</a>.
         </p>

         <p>Most of the samples use <a href="https://github.com/webrtc/adapter">adapter.js</a>, a shim to insulate apps
             from spec changes and prefix differences.</p>

         <p><a href="http://www.webrtc.org/testing" title="Command-line flags for WebRTC testing">https://webrtc.org/testing</a>
             lists command line flags useful for development and testing with Chrome.</p>

         <p>Patches and issues welcome! See <a href="https://github.com/webrtc/samples/blob/gh-pages/CONTRIBUTING.md">CONTRIBUTING.md</a>
             for instructions.</p>

         <p class="warning"><strong>Warning:</strong> It is highly recommended to use headphones when testing these
             samples, as it will otherwise risk loud audio feedback on your system.</p>
     </section>

     <section>

         <h2 id="getusermedia"><a href="https://developer.mozilla.org/en-US/docs/Web/API/Navigator/getUserMedia">getUserMedia():</a>
         </h2>
         <p class="description">Access media devices</p>
         <ul>
             <li><a href="src/content/getusermedia/gum/">Basic getUserMedia demo</a></li>

             <li><a href="src/content/getusermedia/canvas/">Use getUserMedia with canvas</a></li>

             <li><a href="src/content/getusermedia/filter/">Use getUserMedia with canvas and CSS filters</a></li>

             <li><a href="src/content/getusermedia/resolution/">Choose camera resolution</a></li>

             <li><a href="src/content/getusermedia/audio/">Audio-only getUserMedia() output to local audio element</a>
             </li>

             <li><a href="src/content/getusermedia/volume/">Audio-only getUserMedia() displaying volume</a></li>

             <li><a href="src/content/getusermedia/record/">Record stream</a></li>

             <li><a href="src/content/getusermedia/getdisplaymedia/">Screensharing with getDisplayMedia</a></li>

             <li><a href="src/content/getusermedia/pan-tilt-zoom/">Control camera pan, tilt, and zoom</a></li>
         </ul>
         <h2 id="devices">Devices:</h2>
         <p class="description">Query media devices</p>
         <ul>
             <li><a href="src/content/devices/input-output/">Choose camera, microphone and speaker</a></li>

             <li><a href="src/content/devices/multi/">Choose media source and audio output</a></li>
         </ul>

         <h2 id="capture">Stream capture:</h2>
         <p class="description">Stream from canvas or video elements</p>
         <ul>

             <li><a href="src/content/capture/video-video/">Stream from a video element to a video element</a></li>

             <li><a href="src/content/capture/video-pc/">Stream from a video element to a peer connection</a></li>

             <li><a href="src/content/capture/canvas-video/">Stream from a canvas element to a video element</a></li>

             <li><a href="src/content/capture/canvas-pc/">Stream from a canvas element to a peer connection</a></li>

             <li><a href="src/content/capture/canvas-record/">Record a stream from a canvas element</a></li>

             <li><a href="src/content/capture/video-contenthint/">Guiding video encoding with content hints</a></li>
         </ul>

         <h2 id="peerconnection"><a href="https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection">RTCPeerConnection:</a>
         </h2>
         <p class="description">Controlling peer connectivity</p>
         <ul>
             <li><a href="src/content/peerconnection/pc1/">Basic peer connection demo</a></li>

             <li><a href="src/content/peerconnection/perfect-negotiation/">Peer connection using Perfect Negotiation</a></li>

             <li><a href="src/content/peerconnection/audio/">Audio-only peer connection demo</a></li>

             <li><a href="src/content/peerconnection/bandwidth/">Change bandwidth on the fly</a></li>

             <li><a href="src/content/peerconnection/change-codecs/">Change codecs before the call</a></li>

             <li><a href="src/content/peerconnection/upgrade/">Upgrade a call and turn video on</a></li>

             <li><a href="src/content/peerconnection/multiple/">Multiple peer connections at once</a></li>

             <li><a href="src/content/peerconnection/multiple-relay/">Forward the output of one PC into another</a></li>

             <li><a href="src/content/peerconnection/munge-sdp/">Munge SDP parameters</a></li>

             <li><a href="src/content/peerconnection/pr-answer/">Use pranswer when setting up a peer connection</a></li>

             <li><a href="src/content/peerconnection/constraints/">Constraints and stats</a></li>

             <li><a href="src/content/peerconnection/old-new-stats/">More constraints and stats</a></li>

             <li><a href="src/content/peerconnection/create-offer/">Display createOffer output for various scenarios</a>
             </li>

             <li><a href="src/content/peerconnection/dtmf/">Use RTCDTMFSender</a></li>

             <li><a href="src/content/peerconnection/states/">Display peer connection states</a></li>

             <li><a href="src/content/peerconnection/trickle-ice/">ICE candidate gathering from STUN/TURN servers</a>
             </li>

             <li><a href="src/content/peerconnection/restart-ice/">Do an ICE restart</a></li>

             <li><a href="src/content/peerconnection/webaudio-input/">Web Audio output as input to peer connection</a>
             </li>

             <li><a href="src/content/peerconnection/webaudio-output/">Peer connection as input to Web Audio</a></li>
         </ul>
         <h2 id="datachannel"><a
                 href="https://developer.mozilla.org/en-US/docs/Web/API/RTCDataChannel">RTCDataChannel:</a></h2>
         <p class="description">Send arbitrary data over peer connections</p>
         <ul>
             <li><a href="src/content/datachannel/basic/">Transmit text</a></li>

             <li><a href="src/content/datachannel/filetransfer/">Transfer a file</a></li>

             <li><a href="src/content/datachannel/datatransfer/">Transfer data</a></li>

             <li><a href="src/content/datachannel/messaging/">Messaging</a></li>
         </ul>

         <h2 id="videoChat">Video chat:</h2>
         <p class="description">Full featured WebRTC application</p>
         <ul>

             <li><a href="https://apprtc.appspot.com/">AppRTC video chat client</a> powered by Google App Engine</li>

             <li><a href="https://apprtc.appspot.com/params.html">AppRTC URL parameters</a></li>

         </ul>

         <h2 id="capture">Insertable Streams:</h2>
         <p class="description">API for processing media</p>
         <ul>
             <li><a href="src/content/insertable-streams/endtoend-encryption">End to end encryption using WebRTC Insertable Streams</a></li> (Experimental)
             <li><a href="src/content/insertable-streams/video-analyzer">Video analyzer using WebRTC Insertable Streams</a></li> (Experimental)
             <li><a href="src/content/insertable-streams/video-processing">Video processing using MediaStream Insertable Streams</a></li> (Experimental)
         </ul>

     </section>

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 </html>
	<!DOCTYPE html>
	<!--
	* Copyright (c) 2015 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=yes, 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 collection of small samples demonstrating various parts of the <a
	href="https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API">WebRTC APIs</a>. The code for all
	samples are available in the <a href="https://github.com/webrtc/samples">GitHub repository</a>.
	</p>

	<p>Most of the samples use <a href="https://github.com/webrtc/adapter">adapter.js</a>, a shim to insulate apps
	from spec changes and prefix differences.</p>

	<p><a href="http://www.webrtc.org/testing" title="Command-line flags for WebRTC testing">https://webrtc.org/testing</a>
	lists command line flags useful for development and testing with Chrome.</p>

	<p>Patches and issues welcome! See <a href="https://github.com/webrtc/samples/blob/gh-pages/CONTRIBUTING.md">CONTRIBUTING.md</a>
	for instructions.</p>

	<p class="warning"><strong>Warning:</strong> It is highly recommended to use headphones when testing these
	samples, as it will otherwise risk loud audio feedback on your system.</p>
	</section>

	<section>

	<h2 id="getusermedia"><a href="https://developer.mozilla.org/en-US/docs/Web/API/Navigator/getUserMedia">getUserMedia():</a>
	</h2>
	<p class="description">Access media devices</p>
	<ul>
	<li><a href="src/content/getusermedia/gum/">Basic getUserMedia demo</a></li>

	<li><a href="src/content/getusermedia/canvas/">Use getUserMedia with canvas</a></li>

	<li><a href="src/content/getusermedia/filter/">Use getUserMedia with canvas and CSS filters</a></li>

	<li><a href="src/content/getusermedia/resolution/">Choose camera resolution</a></li>

	<li><a href="src/content/getusermedia/audio/">Audio-only getUserMedia() output to local audio element</a>
	</li>

	<li><a href="src/content/getusermedia/volume/">Audio-only getUserMedia() displaying volume</a></li>

	<li><a href="src/content/getusermedia/record/">Record stream</a></li>

	<li><a href="src/content/getusermedia/getdisplaymedia/">Screensharing with getDisplayMedia</a></li>

	<li><a href="src/content/getusermedia/pan-tilt-zoom/">Control camera pan, tilt, and zoom</a></li>
	</ul>
	<h2 id="devices">Devices:</h2>
	<p class="description">Query media devices</p>
	<ul>
	<li><a href="src/content/devices/input-output/">Choose camera, microphone and speaker</a></li>

	<li><a href="src/content/devices/multi/">Choose media source and audio output</a></li>
	</ul>

	<h2 id="capture">Stream capture:</h2>
	<p class="description">Stream from canvas or video elements</p>
	<ul>

	<li><a href="src/content/capture/video-video/">Stream from a video element to a video element</a></li>

	<li><a href="src/content/capture/video-pc/">Stream from a video element to a peer connection</a></li>

	<li><a href="src/content/capture/canvas-video/">Stream from a canvas element to a video element</a></li>

	<li><a href="src/content/capture/canvas-pc/">Stream from a canvas element to a peer connection</a></li>

	<li><a href="src/content/capture/canvas-record/">Record a stream from a canvas element</a></li>

	<li><a href="src/content/capture/video-contenthint/">Guiding video encoding with content hints</a></li>
	</ul>

	<h2 id="peerconnection"><a href="https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection">RTCPeerConnection:</a>
	</h2>
	<p class="description">Controlling peer connectivity</p>
	<ul>
	<li><a href="src/content/peerconnection/pc1/">Basic peer connection demo</a></li>

	<li><a href="src/content/peerconnection/perfect-negotiation/">Peer connection using Perfect Negotiation</a></li>

	<li><a href="src/content/peerconnection/audio/">Audio-only peer connection demo</a></li>

	<li><a href="src/content/peerconnection/bandwidth/">Change bandwidth on the fly</a></li>

	<li><a href="src/content/peerconnection/change-codecs/">Change codecs before the call</a></li>

	<li><a href="src/content/peerconnection/upgrade/">Upgrade a call and turn video on</a></li>

	<li><a href="src/content/peerconnection/multiple/">Multiple peer connections at once</a></li>

	<li><a href="src/content/peerconnection/multiple-relay/">Forward the output of one PC into another</a></li>

	<li><a href="src/content/peerconnection/munge-sdp/">Munge SDP parameters</a></li>

	<li><a href="src/content/peerconnection/pr-answer/">Use pranswer when setting up a peer connection</a></li>

	<li><a href="src/content/peerconnection/constraints/">Constraints and stats</a></li>

	<li><a href="src/content/peerconnection/old-new-stats/">More constraints and stats</a></li>

	<li><a href="src/content/peerconnection/create-offer/">Display createOffer output for various scenarios</a>
	</li>

	<li><a href="src/content/peerconnection/dtmf/">Use RTCDTMFSender</a></li>

	<li><a href="src/content/peerconnection/states/">Display peer connection states</a></li>

	<li><a href="src/content/peerconnection/trickle-ice/">ICE candidate gathering from STUN/TURN servers</a>
	</li>

	<li><a href="src/content/peerconnection/restart-ice/">Do an ICE restart</a></li>

	<li><a href="src/content/peerconnection/webaudio-input/">Web Audio output as input to peer connection</a>
	</li>

	<li><a href="src/content/peerconnection/webaudio-output/">Peer connection as input to Web Audio</a></li>
	</ul>
	<h2 id="datachannel"><a
	href="https://developer.mozilla.org/en-US/docs/Web/API/RTCDataChannel">RTCDataChannel:</a></h2>
	<p class="description">Send arbitrary data over peer connections</p>
	<ul>
	<li><a href="src/content/datachannel/basic/">Transmit text</a></li>

	<li><a href="src/content/datachannel/filetransfer/">Transfer a file</a></li>

	<li><a href="src/content/datachannel/datatransfer/">Transfer data</a></li>

	<li><a href="src/content/datachannel/messaging/">Messaging</a></li>
	</ul>

	<h2 id="videoChat">Video chat:</h2>
	<p class="description">Full featured WebRTC application</p>
	<ul>

	<li><a href="https://apprtc.appspot.com/">AppRTC video chat client</a> powered by Google App Engine</li>

	<li><a href="https://apprtc.appspot.com/params.html">AppRTC URL parameters</a></li>

	</ul>

	<h2 id="capture">Insertable Streams:</h2>
	<p class="description">API for processing media</p>
	<ul>
	<li><a href="src/content/insertable-streams/endtoend-encryption">End to end encryption using WebRTC Insertable Streams</a></li> (Experimental)
	<li><a href="src/content/insertable-streams/video-analyzer">Video analyzer using WebRTC Insertable Streams</a></li> (Experimental)
	<li><a href="src/content/insertable-streams/video-processing">Video processing using MediaStream Insertable Streams</a></li> (Experimental)
	</ul>

	</section>

	</div>

	<script src="src/js/lib/ga.js"></script>

	</body>
	</html>