| Index: LayoutTests/imported/web-platform-tests/webrtc/simplecall.html
|
| diff --git a/LayoutTests/imported/web-platform-tests/webrtc/simplecall.html b/LayoutTests/imported/web-platform-tests/webrtc/simplecall.html
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..595af80e95bc9b4717b443c6678c636ac37ebb38
|
| --- /dev/null
|
| +++ b/LayoutTests/imported/web-platform-tests/webrtc/simplecall.html
|
| @@ -0,0 +1,126 @@
|
| +<!doctype html>
|
| +<!--
|
| +To quickly iterate when developing this test, use --use-fake-ui-for-media-stream
|
| +for Chrome and set the media.navigator.permission.disabled property to true in
|
| +Firefox. You must either have a webcam/mic available on the system or use for
|
| +instance --use-fake-device-for-media-stream for Chrome.
|
| + -->
|
| +
|
| +<html>
|
| +<head>
|
| + <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
|
| + <title>RTCPeerConnection Connection Test</title>
|
| +</head>
|
| +<body>
|
| + <div id="log"></div>
|
| + <div>
|
| + <video id="local-view" autoplay="autoplay"></video>
|
| + <video id="remote-view" autoplay="autoplay"/>
|
| + </video>
|
| + </div>
|
| +
|
| + <!-- These files are in place when executing on W3C. -->
|
| + <script src="../../../resources/testharness.js"></script>
|
| + <script src="../../../resources/testharnessreport.js"></script>
|
| + <script src="../../../resources/vendor-prefix.js"
|
| + data-prefixed-objects=
|
| + '[{"ancestors":["navigator"], "name":"getUserMedia"},
|
| + {"ancestors":["window"], "name":"RTCPeerConnection"},
|
| + {"ancestors":["window"], "name":"RTCSessionDescription"},
|
| + {"ancestors":["window"], "name":"RTCIceCandidate"}]'
|
| + data-prefixed-prototypes=
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| + '[{"ancestors":["HTMLMediaElement"],"name":"srcObject"}]'>
|
| + </script>
|
| + <script type="text/javascript">
|
| + var test = async_test('Can set up a basic WebRTC call.', {timeout: 5000});
|
| +
|
| + var gFirstConnection = null;
|
| + var gSecondConnection = null;
|
| +
|
| + function getUserMediaOkCallback(localStream) {
|
| + gFirstConnection = new RTCPeerConnection(null, null);
|
| + gFirstConnection.onicecandidate = onIceCandidateToFirst;
|
| + gFirstConnection.addStream(localStream);
|
| + gFirstConnection.createOffer(onOfferCreated, failed('createOffer'));
|
| +
|
| + var videoTag = document.getElementById('local-view');
|
| + videoTag.srcObject = localStream;
|
| + };
|
| +
|
| + var onOfferCreated = test.step_func(function(offer) {
|
| + gFirstConnection.setLocalDescription(offer);
|
| +
|
| + // This would normally go across the application's signaling solution.
|
| + // In our case, the "signaling" is to call this function.
|
| + receiveCall(offer.sdp);
|
| + });
|
| +
|
| + function receiveCall(offerSdp) {
|
| + gSecondConnection = new RTCPeerConnection(null, null);
|
| + gSecondConnection.onicecandidate = onIceCandidateToSecond;
|
| + gSecondConnection.onaddstream = onRemoteStream;
|
| +
|
| + var parsedOffer = new RTCSessionDescription({ type: 'offer',
|
| + sdp: offerSdp });
|
| + gSecondConnection.setRemoteDescription(parsedOffer);
|
| +
|
| + gSecondConnection.createAnswer(onAnswerCreated,
|
| + failed('createAnswer'));
|
| + };
|
| +
|
| + var onAnswerCreated = test.step_func(function(answer) {
|
| + gSecondConnection.setLocalDescription(answer);
|
| +
|
| + // Similarly, this would go over the application's signaling solution.
|
| + handleAnswer(answer.sdp);
|
| + });
|
| +
|
| + function handleAnswer(answerSdp) {
|
| + var parsedAnswer = new RTCSessionDescription({ type: 'answer',
|
| + sdp: answerSdp });
|
| + gFirstConnection.setRemoteDescription(parsedAnswer);
|
| +
|
| + // Call negotiated: done.
|
| + test.done();
|
| + };
|
| +
|
| + // Note: the ice candidate handlers are special. We can not wrap them in test
|
| + // steps since that seems to cause some kind of starvation that prevents the
|
| + // call of being set up. Unfortunately we cannot report errors in here.
|
| + var onIceCandidateToFirst = function(event) {
|
| + // If event.candidate is null = no more candidates.
|
| + if (event.candidate) {
|
| + var candidate = new RTCIceCandidate(event.candidate);
|
| + gSecondConnection.addIceCandidate(candidate);
|
| + }
|
| + };
|
| +
|
| + var onIceCandidateToSecond = function(event) {
|
| + if (event.candidate) {
|
| + var candidate = new RTCIceCandidate(event.candidate);
|
| + gFirstConnection.addIceCandidate(candidate);
|
| + }
|
| + };
|
| +
|
| + var onRemoteStream = test.step_func(function(event) {
|
| + var videoTag = document.getElementById('remote-view');
|
| + videoTag.srcObject = event.stream;
|
| + });
|
| +
|
| + // Returns a suitable error callback.
|
| + function failed(function_name) {
|
| + return test.step_func(function() {
|
| + assert_unreached('WebRTC called error callback for ' + function_name);
|
| + });
|
| + }
|
| +
|
| + // This function starts the test.
|
| + test.step(function() {
|
| + navigator.getUserMedia({ video: true, audio: true },
|
| + getUserMediaOkCallback,
|
| + failed('getUserMedia'));
|
| + });
|
| +</script>
|
| +
|
| +</body>
|
| +</html>
|
|
|
Chromium Code Reviews
Issue 560893005:
First checked-in import of the W3C's test suites. (Closed)
Base URL: svn://svn.chromium.org/blink/trunk