forked from webrtc/samples
-
Notifications
You must be signed in to change notification settings - Fork 0
Expand file tree
/
Copy pathindex.html
More file actions
169 lines (105 loc) · 7.57 KB
/
index.html
File metadata and controls
169 lines (105 loc) · 7.57 KB
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
<!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="Client-side WebRTC 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="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="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="samples/web/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 client-side WebRTC code samples and the <a href="https://apprtc.appspot.com" title="AppRTC video chat client">AppRTC</a> video chat client. The source for these samples is available at <a href="//github.com/GoogleChrome/webrtc" title="View Github repositry for these files">github.com/GoogleChrome/webrtc</a>.</p>
<p>Some of the samples use new browser features. They may only work in <a href="//www.google.co.uk/intl/en/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/GoogleChrome/webrtc/blob/master/samples/web/js/adapter.js">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/GoogleChrome/webrtc/blob/master/CONTRIBUTING">CONTRIBUTING</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="samples/web/content/getusermedia/gum">Basic getUserMedia demo</a></p>
<p><a href="samples/web/content/getusermedia/canvas">Use getUserMedia with canvas</a></p>
<p><a href="samples/web/content/getusermedia/filter">Use getUserMedia with canvas and CSS filters</a></p>
<p><a href="samples/web/content/getusermedia/resolution">Choose camera resolution</a></p>
<p><a href="samples/web/content/getusermedia/source">Choose camera and microphone</a></p>
<p><a href="samples/web/content/getusermedia/audio">Audio-only getUserMedia() output to local audio element</a></p>
<p><a href="samples/web/content/getusermedia/volume">Audio-only getUserMedia() displaying volume</a></p>
<p><a href="samples/web/content/getusermedia/face">Face tracking, using getUserMedia and canvas</a></p>
<h3 id="peerconnection">RTCPeerConnection</h3>
<p><a href="samples/web/content/peerconnection/pc1">Basic peer connection demo</a></p>
<p><a href="samples/web/content/peerconnection/audio">Audio-only peer connection demo</a></p>
<p><a href="samples/web/content/peerconnection/multiple">Multiple peer connections at once</a></p>
<p><a href="samples/web/content/peerconnection/multiple-relay">Forward the output of one PC into another</a></p>
<p><a href="samples/web/content/peerconnection/munge-sdp">Munge SDP parameters</a></p>
<p><a href="samples/web/content/peerconnection/pr-answer">Use pranswer when setting up a peer connection</a></p>
<p><a href="samples/web/content/peerconnection/constraints">Constraints and stats</a></p>
<p><a href="samples/web/content/peerconnection/create-offer">Display createOffer output for various scenarios</a></p>
<p><a href="samples/web/content/peerconnection/dtmf">Use RTCDTMFSender</a></p>
<p><a href="samples/web/content/peerconnection/states">Display peer connection states</a></p>
<p><a href="samples/web/content/peerconnection/trickle-ice">ICE candidate gathering from STUN/TURN servers</a></p>
<p><a href="samples/web/content/peerconnection/webaudio-input">Web Audio output as input to peer connection</a></p>
<h3 id="datachannel">RTCDataChannel</h3>
<p><a href="samples/web/content/datachannel">Basic data channel demo</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>
<section>
<h2 id="test-pages">Test pages</h2>
<p><a href="samples/web/content/manual-test/audio-and-video">Audio and video streams</a></p>
<p><a href="samples/web/content/manual-test/constraints">Constraints</a></p>
<p><a href="samples/web/content/manual-test/iframe-apprtc">Iframe apprtc</a></p>
<p><a href="samples/web/content/manual-test/iframe-video">Iframe video</a></p>
<p><a href="samples/web/content/manual-test/multiple-audio">Multiple audio streams</a></p>
<p><a href="samples/web/content/manual-test/multiple-peerconnections">Multiple peerconnections</a></p>
<p><a href="samples/web/content/manual-test/multiple-video">Multiple video streams</a></p>
<p><a href="samples/web/content/manual-test/multiple-video-devices">Multiple video devices</a></p>
<p><a href="samples/web/content/manual-test/peer2peer">Peer2peer</a></p>
<p><a href="samples/web/content/manual-test/peer2peer-iframe">Peer2peer iframe</a></p>
<p><a href="samples/web/content/manual-test/single-audio">Single audio stream</a></p>
<p><a href="samples/web/content/manual-test/single-video">Single video stream</a></p>
</section>
<a href="//github.com/GoogleChrome/webrtc" title="View the repository" id="viewSource">github.com/GoogleChrome/webrtc</a>
</div>
<script src="samples/web/js/lib/ga.js"></script>
</body>
</html>