зеркало из https://github.com/mozilla/gecko-dev.git
121 строка
3.9 KiB
HTML
121 строка
3.9 KiB
HTML
<!doctype html>
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<!--
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To run this test, you must have a webcam and a microphone or use
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fake devices by specifying
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--use-fake-device-for-media-stream --use-fake-ui-for-media-stream
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for Chrome or by setting the
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media.navigator.streams.fake
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property to true in Firefox.
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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>RTCPeerConnection Connection Test</title>
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</head>
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<body>
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<div id="log"></div>
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<div>
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<video id="local-view" autoplay="autoplay"></video>
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<video id="remote-view" autoplay="autoplay"/>
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</video>
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</div>
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<!-- These files are in place when executing on W3C. -->
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<script src="/resources/testharness.js"></script>
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<script src="/resources/testharnessreport.js"></script>
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<script src="/common/vendor-prefix.js"
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data-prefixed-objects=
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'[{"ancestors":["navigator"], "name":"getUserMedia"}]'
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data-prefixed-prototypes=
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'[{"ancestors":["HTMLMediaElement"],"name":"srcObject"}]'>
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</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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// if the remote video gets video data that implies the negotiation
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// as well as the ICE and DTLS connection are up.
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document.getElementById('remote-view')
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.addEventListener('loadedmetadata', function() {
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// Call negotiated: done.
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test.done();
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});
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function getUserMediaOkCallback(localStream) {
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gFirstConnection = new RTCPeerConnection(null);
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gFirstConnection.onicecandidate = onIceCandidateToFirst;
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gFirstConnection.addStream(localStream);
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gFirstConnection.createOffer(onOfferCreated, failed('createOffer'));
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var videoTag = document.getElementById('local-view');
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videoTag.srcObject = 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 RTCPeerConnection(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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};
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var onIceCandidateToFirst = test.step_func(function(event) {
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// If event.candidate is null = no more candidates.
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gSecondConnection.addIceCandidate(event.candidate);
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});
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var onIceCandidateToSecond = test.step_func(function(event) {
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gFirstConnection.addIceCandidate(event.candidate);
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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.srcObject = 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.mediaDevices.getUserMedia({ video: true, audio: true })
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.then(getUserMediaOkCallback, failed('getUserMedia'));
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});
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</script>
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</body>
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</html>
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