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Update web-platform-tests to revision 6087baf4a83e7953112242be9fd6e719797ebdf6

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This commit is contained in:
WPT Sync Bot 2018-12-07 20:30:51 -05:00
parent 3c19cd49ec
commit 6c2d26eb4b
76 changed files with 3020 additions and 345 deletions
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10
tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-track-stats.https.html
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@ -155,11 +155,6 @@
}));
}, 'addTrack(): Media stream stats references track stats');
// TODO(hbos): addStream() is legacy API not in the spec. Based on discussion
// whether to standardize in legacy section, consider removing this test or
// keeping it until addTrack() has wide support.
// https://github.com/w3c/webrtc-pc/issues/1705
// https://github.com/w3c/webrtc-pc/issues/1125
async_test(t => {
const pc = new RTCPeerConnection();
t.add_cleanup(() => pc.close());
@ -170,7 +165,8 @@
track = tracks[0];
stream = streams[0];
stream.addTrack(track);
pc.addStream(stream);
for (const track of stream.getTracks())
pc.addTrack(track, stream);
return pc.createOffer();
}))
.then(t.step_func(offer => {
@ -193,7 +189,7 @@
.catch(t.step_func(reason => {
assert_unreached(reason);
}));
}, 'Legacy addStream(): Media stream stats references track stats');
}, 'Media stream stats references track stats');
async_test(t => {
const caller = new RTCPeerConnection();

6
tests/wpt/web-platform-tests/webrtc/RTCRtpParameters-encodings.html
View file

@ -48,6 +48,7 @@
RTCDtxStatus dtx;
boolean active;
RTCPriorityType priority;
RTCPriorityType networkPriority;
unsigned long ptime;
unsigned long maxBitrate;
double maxFramerate;
@ -146,6 +147,7 @@
dtx: 'enabled',
active: false,
priority: 'low',
networkPriority: 'low',
ptime: 5,
maxBitrate: 8,
maxFramerate: 25,
@ -161,6 +163,7 @@
assert_equals(encoding.dtx, 'enabled');
assert_equals(encoding.active, false);
assert_equals(encoding.priority, 'low');
assert_equals(encoding.networkPriority, 'low');
assert_equals(encoding.ptime, 5);
assert_equals(encoding.maxBitrate, 8);
assert_equals(encoding.maxFramerate, 25);
@ -339,6 +342,9 @@
test_modified_encoding('audio', 'priority', 'very-low', 'high',
'setParameters() with modified encoding.priority should succeed');
test_modified_encoding('audio', 'networkPriority', 'very-low', 'high',
'setParameters() with modified encoding.networkPriority should succeed');
test_modified_encoding('audio', 'ptime', 2, 4,
'setParameters() with modified encoding.ptime should succeed');

3
tests/wpt/web-platform-tests/webrtc/RTCRtpParameters-helper.js
View file

@ -142,6 +142,7 @@ function validateRtpParameters(param) {
RTCDtxStatus dtx;
boolean active;
RTCPriorityType priority;
RTCPriorityType networkPriority;
unsigned long ptime;
unsigned long maxBitrate;
double maxFramerate;
@ -171,6 +172,8 @@ function validateEncodingParameters(encoding) {
assert_optional_boolean_field(encoding, 'active');
assert_optional_enum_field(encoding, 'priority',
['very-low', 'low', 'medium', 'high']);
assert_optional_enum_field(encoding, 'networkPriority',
['very-low', 'low', 'medium', 'high']);
assert_optional_unsigned_int_field(encoding, 'ptime');
assert_optional_unsigned_int_field(encoding, 'maxBitrate');

74
tests/wpt/web-platform-tests/webrtc/legacy/RTCPeerConnection-addStream.https.html Normal file
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@ -0,0 +1,74 @@
<!doctype html>
<meta charset=utf-8>
<meta name="timeout" content="long">
<title>RTCPeerConnection legacy addStream</title>
<script src="/resources/testharness.js"></script>
<script src="/resources/testharnessreport.js"></script>
<script src="../RTCPeerConnection-helper.js"></script>
<script src="../RTCStats-helper.js"></script>
<script src="../dictionary-helper.js"></script>
<script>
'use strict';
// The following helper functions are called from RTCPeerConnection-helper.js:
// getUserMediaTracksAndStreams
// The following helper functions are called from RTCStats-helper.js
// (depends on dictionary-helper.js):
// validateRtcStats
// TODO(hbos): addStream() is legacy API not in the spec. Based on discussion
// whether to standardize in legacy section, consider removing this test or
// keeping it until addTrack() has wide support.
// https://github.com/w3c/webrtc-pc/issues/1705
// https://github.com/w3c/webrtc-pc/issues/1125
async_test(t => {
const pc = new RTCPeerConnection();
t.add_cleanup(() => pc.close());
let track;
let stream;
return getUserMediaTracksAndStreams(1)
.then(t.step_func(([tracks, streams]) => {
track = tracks[0];
stream = streams[0];
stream.addTrack(track);
pc.addStream(stream);
return pc.createOffer();
}))
.then(t.step_func(offer => {
return pc.setLocalDescription(offer);
}))
.then(t.step_func(() => {
return pc.getStats();
}))
.then(t.step_func(report => {
let trackStats = findStatsByTypeAndId(report, 'track', track.id);
let streamStats = findStatsByTypeAndId(report, 'stream', stream.id);
assert_true(trackStats != null && streamStats != null,
'Has stats for track and stream');
assert_array_equals(streamStats.trackIds, [ trackStats.id ],
'streamStats.trackIds == [ trackStats.id ]');
validateRtcStats(report, trackStats);
validateRtcStats(report, streamStats);
t.done();
}))
.catch(t.step_func(reason => {
assert_unreached(reason);
}));
}, 'Legacy addStream(): Media stream stats references track stats');
function findStatsByTypeAndId(report, type, identifier) {
return findStats(report, stats => {
return stats.type == type && stats[type + 'Identifier'] == identifier;
});
}
function findStats(report, findFunc) {
for (let it = report.values(), n = it.next(); !n.done; n = it.next()) {
if (findFunc(n.value))
return n.value;
}
return null;
}
</script>

4
tests/wpt/web-platform-tests/webrtc/no-media-call.html
View file

@ -1,7 +1,4 @@
<!doctype html>
<!--
This test uses the legacy callback API with no media, and thus does not require fake media devices.
-->
<html>
<head>
@ -37,7 +34,6 @@ 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).then(
function() {
gSecondConnection.createAnswer().then(onAnswerCreated,