| Index: third_party/WebKit/LayoutTests/imported/web-platform-tests/webrtc/no-media-call.html
|
| diff --git a/third_party/WebKit/LayoutTests/imported/web-platform-tests/webrtc/no-media-call.html b/third_party/WebKit/LayoutTests/imported/web-platform-tests/webrtc/no-media-call.html
|
| deleted file mode 100644
|
| index f38a457dcdd8ad98853219275e0d9409d55b8483..0000000000000000000000000000000000000000
|
| --- a/third_party/WebKit/LayoutTests/imported/web-platform-tests/webrtc/no-media-call.html
|
| +++ /dev/null
|
| @@ -1,139 +0,0 @@
|
| -<!doctype html>
|
| -<!--
|
| -This test uses the legacy callback API with no media, and thus does not require fake media devices.
|
| - -->
|
| -
|
| -<html>
|
| -<head>
|
| - <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
|
| - <title>RTCPeerConnection No-Media Connection Test</title>
|
| -</head>
|
| -<body>
|
| - <div id="log"></div>
|
| - <h2>iceConnectionState info</h2>
|
| - <div id="stateinfo">
|
| - </div>
|
| -
|
| - <!-- These files are in place when executing on W3C. -->
|
| - <script src="../../../resources/testharness.js"></script>
|
| - <script src="../../../resources/testharnessreport.js"></script>
|
| - <script type="text/javascript">
|
| - var test = async_test('Can set up a basic WebRTC call with no data.');
|
| -
|
| - var gFirstConnection = null;
|
| - var gSecondConnection = null;
|
| -
|
| - var onOfferCreated = test.step_func(function(offer) {
|
| - gFirstConnection.setLocalDescription(offer, ignoreSuccess,
|
| - failed('setLocalDescription first'));
|
| -
|
| - // 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) {
|
| -
|
| - var parsedOffer = new RTCSessionDescription({ type: 'offer',
|
| - sdp: offerSdp });
|
| - // These functions use the legacy interface extensions to RTCPeerConnection.
|
| - gSecondConnection.setRemoteDescription(parsedOffer,
|
| - function() {
|
| - gSecondConnection.createAnswer(onAnswerCreated,
|
| - failed('createAnswer'));
|
| - },
|
| - failed('setRemoteDescription second'));
|
| - };
|
| -
|
| - var onAnswerCreated = test.step_func(function(answer) {
|
| - gSecondConnection.setLocalDescription(answer, ignoreSuccess,
|
| - failed('setLocalDescription second'));
|
| -
|
| - // 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, ignoreSuccess,
|
| - failed('setRemoteDescription first'));
|
| - };
|
| -
|
| - 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) {
|
| - assert_unreached('WebRTC received a stream when there was none');
|
| - });
|
| -
|
| - var onIceConnectionStateChange = test.step_func(function(event) {
|
| - assert_equals(event.type, 'iceconnectionstatechange');
|
| - assert_not_equals(gFirstConnection.iceConnectionState, "failed", "iceConnectionState of first connection");
|
| - assert_not_equals(gSecondConnection.iceConnectionState, "failed", "iceConnectionState of second connection");
|
| - var stateinfo = document.getElementById('stateinfo');
|
| - stateinfo.innerHTML = 'First: ' + gFirstConnection.iceConnectionState
|
| - + '<br>Second: ' + gSecondConnection.iceConnectionState;
|
| - // Note: All these combinations are legal states indicating that the
|
| - // call has connected. All browsers should end up in completed/completed,
|
| - // but as of this moment, we've chosen to terminate the test early.
|
| - // TODO: Revise test to ensure completed/completed is reached.
|
| - if (gFirstConnection.iceConnectionState == 'connected' &&
|
| - gSecondConnection.iceConnectionState == 'connected') {
|
| - test.done()
|
| - }
|
| - if (gFirstConnection.iceConnectionState == 'connected' &&
|
| - gSecondConnection.iceConnectionState == 'completed') {
|
| - test.done()
|
| - }
|
| - if (gFirstConnection.iceConnectionState == 'completed' &&
|
| - gSecondConnection.iceConnectionState == 'connected') {
|
| - test.done()
|
| - }
|
| - if (gFirstConnection.iceConnectionState == 'completed' &&
|
| - gSecondConnection.iceConnectionState == 'completed') {
|
| - test.done()
|
| - }
|
| - });
|
| -
|
| - // Returns a suitable error callback.
|
| - function failed(function_name) {
|
| - return test.step_func(function() {
|
| - assert_unreached('WebRTC called error callback for ' + function_name);
|
| - });
|
| - }
|
| -
|
| - // Returns a suitable do-nothing.
|
| - function ignoreSuccess(function_name) {
|
| - }
|
| -
|
| - // This function starts the test.
|
| - test.step(function() {
|
| - gFirstConnection = new RTCPeerConnection(null);
|
| - gFirstConnection.onicecandidate = onIceCandidateToFirst;
|
| - gFirstConnection.oniceconnectionstatechange = onIceConnectionStateChange;
|
| -
|
| - gSecondConnection = new RTCPeerConnection(null);
|
| - gSecondConnection.onicecandidate = onIceCandidateToSecond;
|
| - gSecondConnection.onaddstream = onRemoteStream;
|
| - gSecondConnection.oniceconnectionstatechange = onIceConnectionStateChange;
|
| -
|
| - // The offerToReceiveVideo is necessary and sufficient to make
|
| - // an actual connection.
|
| - gFirstConnection.createOffer(onOfferCreated, failed('createOffer'),
|
| - {offerToReceiveVideo: true});
|
| - });
|
| -</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