| Index: third_party/WebKit/LayoutTests/imported/web-platform-tests/webrtc/simplecall.html
|
| diff --git a/third_party/WebKit/LayoutTests/imported/web-platform-tests/webrtc/simplecall.html b/third_party/WebKit/LayoutTests/imported/web-platform-tests/webrtc/simplecall.html
|
| deleted file mode 100644
|
| index 24cc23c07e94219dbdc3803251a4c772d014472b..0000000000000000000000000000000000000000
|
| --- a/third_party/WebKit/LayoutTests/imported/web-platform-tests/webrtc/simplecall.html
|
| +++ /dev/null
|
| @@ -1,118 +0,0 @@
|
| -<!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"}]'
|
| - 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);
|
| - 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);
|
| - 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();
|
| - };
|
| -
|
| - var onIceCandidateToFirst = test.step_func(function(event) {
|
| - // If event.candidate is null = no more candidates.
|
| - if (event.candidate) {
|
| - gSecondConnection.addIceCandidate(event.candidate);
|
| - }
|
| - });
|
| -
|
| - var onIceCandidateToSecond = test.step_func(function(event) {
|
| - if (event.candidate) {
|
| - gFirstConnection.addIceCandidate(event.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 1979363002:
Moved web-platform-tests to wpt. (Closed)
Base URL: https://chromium.googlesource.com/chromium/src.git@master