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Update web-platform-tests to revision 9c2bea6dac36e36ba1f489d10c2be42160d8f34f

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This commit is contained in:
WPT Sync Bot 2018-11-27 21:07:27 -05:00
parent 482923cec2
commit 5c371dd958
459 changed files with 10717 additions and 834 deletions
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2
tests/wpt/web-platform-tests/webrtc/RTCDTMFSender-insertDTMF.https.html
View file

@ -122,7 +122,7 @@
const stream = await navigator.mediaDevices.getUserMedia({audio: true});
t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
const [track] = stream.getTracks();
callee.addTrack(track);
callee.addTrack(track, stream);
const answer = await callee.createAnswer();
await callee.setLocalDescription(answer);
await caller.setRemoteDescription(answer);

4
tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-removeTrack.https.html
View file

@ -175,7 +175,7 @@
const offer = await caller.createOffer();
await caller.setLocalDescription(offer);
await callee.setRemoteDescription(offer);
callee.addTrack(track);
callee.addTrack(track, stream);
const answer = await callee.createAnswer();
await callee.setLocalDescription(answer);
await caller.setRemoteDescription(answer);
@ -244,7 +244,7 @@
const offer = await caller.createOffer();
await caller.setLocalDescription(offer);
await callee.setRemoteDescription(offer);
callee.addTrack(track);
callee.addTrack(track, stream);
const answer = await callee.createAnswer();
await callee.setLocalDescription(answer);
await caller.setRemoteDescription(answer);

4
tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html
View file

@ -228,7 +228,7 @@
const localStream =
await getNoiseStream({audio: true});
t.add_cleanup(() => localStream.getTracks().forEach(track => track.stop()));
caller.addTrack(localStream.getTracks()[0]);
caller.addTrack(localStream.getTracks()[0], localStream);
const ontrackPromise = addEventListenerPromise(t, callee, 'track', e => {
assert_array_equals(callee.getReceivers(), [e.receiver],
'getReceivers() == [e.receiver].');
@ -245,7 +245,7 @@
const localStream =
await getNoiseStream({audio: true});
t.add_cleanup(() => localStream.getTracks().forEach(track => track.stop()));
const sender = caller.addTrack(localStream.getTracks()[0]);
const sender = caller.addTrack(localStream.getTracks()[0], localStream);
const ontrackPromise = addEventListenerPromise(t, callee, 'track');
await exchangeOfferAnswer(caller, callee);
await ontrackPromise;

20
tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-track-stats.https.html
View file

@ -26,7 +26,7 @@
return getUserMediaTracksAndStreams(1)
.then(t.step_func(([tracks, streams]) => {
track = tracks[0];
pc.addTrack(track);
pc.addTrack(track, streams[0]);
return pc.getStats();
}))
.then(t.step_func(report => {
@ -73,7 +73,7 @@
return getUserMediaTracksAndStreams(1)
.then(t.step_func(([tracks, streams]) => {
track = tracks[0];
pc.addTrack(track);
pc.addTrack(track, streams[0]);
return pc.createOffer();
}))
.then(t.step_func(offer => {
@ -204,7 +204,7 @@
return getUserMediaTracksAndStreams(1)
.then(t.step_func(([tracks, streams]) => {
sendingTrack = tracks[0];
caller.addTrack(sendingTrack);
caller.addTrack(sendingTrack, streams[0]);
return doSignalingHandshake(caller, callee);
}))
.then(t.step_func(() => {
@ -237,7 +237,7 @@
};
return getUserMediaTracksAndStreams(1)
.then(t.step_func(([tracks, streams]) => {
caller.addTrack(tracks[0]);
caller.addTrack(tracks[0], streams[0]);
return doSignalingHandshake(caller, callee);
}))
.then(t.step_func(() => {
@ -271,7 +271,7 @@
.then(t.step_func(([tracks, streams]) => {
sendingTrack1 = tracks[0];
sendingTrack2 = tracks[1];
sender = caller.addTrack(sendingTrack1);
sender = caller.addTrack(sendingTrack1, streams[0]);
return sender.replaceTrack(sendingTrack2);
}))
.then(t.step_func(() => {
@ -300,7 +300,7 @@
.then(t.step_func(([tracks, streams]) => {
sendingTrack1 = tracks[0];
sendingTrack2 = tracks[1];
sender = caller.addTrack(sendingTrack1);
sender = caller.addTrack(sendingTrack1, streams[0]);
return exchangeOffer(caller, callee);
}))
.then(t.step_func(() => {
@ -337,7 +337,7 @@
.then(t.step_func(([tracks, streams]) => {
sendingTrack1 = tracks[0];
sendingTrack2 = tracks[1];
sender = caller.addTrack(sendingTrack1);
sender = caller.addTrack(sendingTrack1, streams[0]);
return doSignalingHandshake(caller, callee);
}))
.then(t.step_func(() => {
@ -373,7 +373,7 @@
.then(t.step_func(([tracks, streams]) => {
sendingTrack1 = tracks[0];
sendingTrack2 = tracks[1];
sender = caller.addTrack(sendingTrack1);
sender = caller.addTrack(sendingTrack1, streams[0]);
return doSignalingHandshake(caller, callee);
}))
.then(t.step_func(() => {
@ -569,8 +569,8 @@
const pc = new RTCPeerConnection();
t.add_cleanup(() => pc.close());
let [tracks, streams] = await getUserMediaTracksAndStreams(2);
let sender1 = pc.addTrack(tracks[0]);
let sender2 = pc.addTrack(tracks[1]);
let sender1 = pc.addTrack(tracks[0], streams[0]);
let sender2 = pc.addTrack(tracks[1], streams[1]);
await sender2.replaceTrack(sender1.track);
await promise_rejects(t, 'InvalidAccessError', pc.getStats(sender1.track));
}, 'RTCPeerConnection.getStats(track) throws InvalidAccessError when there ' +

2
tests/wpt/web-platform-tests/webrtc/RTCRtpReceiver-getStats.https.html
View file

@ -53,7 +53,7 @@
const stream = await getNoiseStream({audio:true});
t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
const [track] = stream.getTracks();
callee.addTrack(track);
callee.addTrack(track, stream);
const { receiver } = caller.addTransceiver('audio');

2
tests/wpt/web-platform-tests/webrtc/simplecall-no-ssrcs.https.html
View file

@ -17,7 +17,7 @@
<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 without announcing ssrcs.', {timeout: 5000});
var test = async_test('Can set up a basic WebRTC call without announcing ssrcs.');
var gFirstConnection = null;
var gSecondConnection = null;

2
tests/wpt/web-platform-tests/webrtc/simplecall.https.html
View file

@ -17,7 +17,7 @@
<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.', {timeout: 5000});
var test = async_test('Can set up a basic WebRTC call.');
var gFirstConnection = null;
var gSecondConnection = null;