53034fb247
BUG= Review URL: https://webrtc-codereview.appspot.com/930031 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3211 4adac7df-926f-26a2-2b94-8c16560cd09d
119 lines
4.0 KiB
HTML
119 lines
4.0 KiB
HTML
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
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<!--
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To quickly iterate when developing this test, make sure you select
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'Always allow this site to use this webcam' option in the dropdown menu of
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Chrome when it's requesting access to your webcam.
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Notice that this requires the site you're browsing to use HTTPS.
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Without that, the test might timeout before you have had the chance to accept
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access to the webcam.
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-->
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<html>
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<head>
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<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
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<title>PeerConnection Connection Test</title>
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<script src="https://w3c-test.org/resources/testharness.js"></script>
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<script type="text/javascript">
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var test = async_test('Can set up a basic WebRTC call.', {timeout: 5000});
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var gFirstConnection = null;
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var gSecondConnection = null;
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function getUserMediaOkCallback(localStream) {
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gFirstConnection = new webkitRTCPeerConnection(null, null);
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gFirstConnection.onicecandidate = onIceCandidateToFirst;
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gFirstConnection.addStream(localStream);
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gFirstConnection.createOffer(onOfferCreated);
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var videoTag = document.getElementById('local-view');
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videoTag.src = webkitURL.createObjectURL(localStream);
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};
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var onOfferCreated = test.step_func(function(offer) {
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gFirstConnection.setLocalDescription(offer);
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// This would normally go across the application's signaling solution.
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// In our case, the "signaling" is to call this function.
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receiveCall(offer.sdp);
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});
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function receiveCall(offerSdp) {
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gSecondConnection = new webkitRTCPeerConnection(null, null);
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gSecondConnection.onicecandidate = onIceCandidateToSecond;
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gSecondConnection.onaddstream = onRemoteStream;
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var parsedOffer = new RTCSessionDescription({ type: 'offer',
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sdp: offerSdp });
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gSecondConnection.setRemoteDescription(parsedOffer);
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gSecondConnection.createAnswer(onAnswerCreated,
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failed('createAnswer'));
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};
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var onAnswerCreated = test.step_func(function(answer) {
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gSecondConnection.setLocalDescription(answer);
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// Similarly, this would go over the application's signaling solution.
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handleAnswer(answer.sdp);
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});
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function handleAnswer(answerSdp) {
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var parsedAnswer = new RTCSessionDescription({ type: 'answer',
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sdp: answerSdp });
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gFirstConnection.setRemoteDescription(parsedAnswer);
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// Call negotiated: done.
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test.done();
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};
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// Note: the ice candidate handlers are special. We can not wrap them in test
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// steps since that seems to cause some kind of starvation that prevents the
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// call of being set up. Unfortunately we cannot report errors in here.
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var onIceCandidateToFirst = function(event) {
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// If event.candidate is null = no more candidates.
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if (event.candidate) {
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var candidate = new RTCIceCandidate(event.candidate);
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gSecondConnection.addIceCandidate(candidate);
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}
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};
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var onIceCandidateToSecond = function(event) {
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if (event.candidate) {
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var candidate = new RTCIceCandidate(event.candidate);
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gFirstConnection.addIceCandidate(candidate);
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}
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};
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var onRemoteStream = test.step_func(function(event) {
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var videoTag = document.getElementById('remote-view');
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videoTag.src = webkitURL.createObjectURL(event.stream);
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});
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// Returns a suitable error callback.
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function failed(function_name) {
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return test.step_func(function() {
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assert_unreached('WebRTC called error callback for ' + function_name);
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});
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}
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// This function starts the test.
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test.step(function() {
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navigator.webkitGetUserMedia({ video: true, audio: true },
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getUserMediaOkCallback,
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failed('getUserMedia'));
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});
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</script>
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</head>
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<body>
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<div>
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<video width="320" height="240" id="remote-view" autoplay="autoplay"></video>
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<video width="320" height="240" id="local-view" autoplay="autoplay"></video>
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</div>
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<div id="log"></div>
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</body>
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</html> |