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Update web-platform-tests to revision 2b7dace05fc1869398ee24f84fda4c0e4c0455ae

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WPT Sync Bot 2018-08-31 21:37:12 +00:00 • committed by Tom Servo
parent b23125d590
commit 6c901de216
844 changed files with 19802 additions and 3093 deletions
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43
tests/wpt/web-platform-tests/webrtc/RTCQuicStream.https.html Normal file
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@ -0,0 +1,43 @@
<!doctype html>
<meta charset=utf-8>
<title>RTCQuicStream.https.html</title>
<script src="/resources/testharness.js"></script>
<script src="/resources/testharnessreport.js"></script>
<script src="RTCQuicTransport-helper.js"></script>
<script>
'use strict';
// These tests are based on the following specification:
// https://w3c.github.io/webrtc-quic/
// The following helper functions are called from RTCQuicTransport-helper.js:
// makeQuicTransport
test(t => {
const quicTransport = makeQuicTransport(t, []);
const quicStream = quicTransport.createStream();
assert_equals(quicStream.transport, quicTransport,
'Expect transport to be set to the creating RTCQuicTransport.');
assert_equals(quicStream.state, 'new', `Expect state to be 'new'.`);
assert_equals(quicStream.readBufferedAmount, 0,
'Expect read buffered amount to be 0.');
assert_equals(quicStream.writeBufferedAmount, 0,
'Expect write buffered amount to be 0.');
}, 'createStream() returns an RTCQuicStream with initial properties set.');
test(t => {
const quicTransport = makeQuicTransport(t, []);
quicTransport.stop();
assert_throws('InvalidStateError', () => quicTransport.createStream());
}, 'createStream() throws if the transport is closed.');
test(t => {
const quicTransport = makeQuicTransport(t, []);
const firstQuicStream = quicTransport.createStream();
const secondQuicStream = quicTransport.createStream();
quicTransport.stop();
assert_equals(firstQuicStream.state, 'closed');
assert_equals(secondQuicStream.state, 'closed');
}, 'RTCQuicTransport.stop() closes all streams.');
</script>

10
tests/wpt/web-platform-tests/webrtc/RTCQuicTransport-helper.js Normal file
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@ -0,0 +1,10 @@
'use strict';
function makeQuicTransport(t, certificates) {
const iceTransport = new RTCIceTransport();
t.add_cleanup(() => iceTransport.stop());
const quicTransport = new RTCQuicTransport(iceTransport, certificates);
t.add_cleanup(() => quicTransport.stop());
return quicTransport;
}

10
tests/wpt/web-platform-tests/webrtc/RTCQuicTransport.https.html
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@ -3,19 +3,15 @@
<title>RTCQuicTransport.https.html</title>
<script src="/resources/testharness.js"></script>
<script src="/resources/testharnessreport.js"></script>
<script src="RTCQuicTransport-helper.js"></script>
<script>
'use strict';
// These tests are based on the following specification:
// https://w3c.github.io/webrtc-quic/
function makeQuicTransport(t, certificates) {
const iceTransport = new RTCIceTransport();
t.add_cleanup(() => iceTransport.stop());
const quicTransport = new RTCQuicTransport(iceTransport, certificates);
t.add_cleanup(() => quicTransport.stop());
return quicTransport;
}
// The following helper functions are called from RTCQuicTransport-helper.js:
// makeQuicTransport
function generateCertificate(keygenAlgorithm) {
return RTCPeerConnection.generateCertificate({

10
tests/wpt/web-platform-tests/webrtc/no-media-call.html
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@ -37,9 +37,9 @@ This test uses the legacy callback API with no media, and thus does not require
var parsedOffer = new RTCSessionDescription({ type: 'offer',
sdp: offerSdp });
// These functions use the legacy interface extensions to RTCPeerConnection.
gSecondConnection.setRemoteDescription(parsedOffer,
gSecondConnection.setRemoteDescription(parsedOffer).then(
function() {
gSecondConnection.createAnswer(onAnswerCreated,
gSecondConnection.createAnswer().then(onAnswerCreated,
failed('createAnswer'));
},
failed('setRemoteDescription second'));
@ -56,7 +56,7 @@ This test uses the legacy callback API with no media, and thus does not require
function handleAnswer(answerSdp) {
var parsedAnswer = new RTCSessionDescription({ type: 'answer',
sdp: answerSdp });
gFirstConnection.setRemoteDescription(parsedAnswer, ignoreSuccess,
gFirstConnection.setRemoteDescription(parsedAnswer).then(ignoreSuccess,
failed('setRemoteDescription first'));
};
@ -125,8 +125,8 @@ This test uses the legacy callback API with no media, and thus does not require
// The offerToReceiveVideo is necessary and sufficient to make
// an actual connection.
gFirstConnection.createOffer(onOfferCreated, failed('createOffer'),
{offerToReceiveVideo: true});
gFirstConnection.createOffer({offerToReceiveVideo: true})
.then(onOfferCreated, failed('createOffer'));
});
</script>

4
tests/wpt/web-platform-tests/webrtc/simplecall.https.html
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@ -36,7 +36,7 @@
localStream.getTracks().forEach(function(track) {
gFirstConnection.addTrack(track, localStream);
});
gFirstConnection.createOffer(onOfferCreated, failed('createOffer'));
gFirstConnection.createOffer().then(onOfferCreated, failed('createOffer'));
var videoTag = document.getElementById('local-view');
videoTag.srcObject = localStream;
@ -59,7 +59,7 @@
sdp: offerSdp });
gSecondConnection.setRemoteDescription(parsedOffer);
gSecondConnection.createAnswer(onAnswerCreated,
gSecondConnection.createAnswer().then(onAnswerCreated,
failed('createAnswer'));
};