forked from webrtc/samples
-
Notifications
You must be signed in to change notification settings - Fork 0
/
index.html
148 lines (91 loc) · 6.52 KB
/
index.html
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
<!DOCTYPE html>
<html>
<head>
<meta name="keywords" content="WebRTC, HTML5, JavaScript" />
<meta name="description" content="Client-side WebRTC code samples." />
<meta http-equiv="Content-Type" content="text/html; charset=utf-8" />
<meta name="viewport" content="width=device-width, initial-scale=1.0, minimum-scale=1.0">
<base target="_blank">
<title>WebRTC samples</title>
<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: ;
}
</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/interop">webrtc.org/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>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!</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 (based on Google App Engine)</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>