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Update web-platform-tests to revision bd951736b4d7dbfd862dfa0f27271b544e35ca30

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WPT Sync Bot 2019-01-22 20:47:08 -05:00 • committed by Josh Matthews
parent 461bce1241
commit 74f3bef690
110 changed files with 2556 additions and 363 deletions
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96
tests/wpt/web-platform-tests/webrtc/RTCRtpReceiver-getContributingSources.https.html
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@ -5,85 +5,31 @@
<script src="/resources/testharnessreport.js"></script>
<script src="RTCPeerConnection-helper.js"></script>
<script>
'use strict';
'use strict';
// Test is based on the following editor draft:
// https://w3c.github.io/webrtc-pc/archives/20170605/webrtc.html
async function connectAndExpectNoCsrcs(t, kind) {
const pc1 = new RTCPeerConnection();
t.add_cleanup(() => pc1.close());
const pc2 = new RTCPeerConnection();
t.add_cleanup(() => pc2.close());
// The following helper function is called from RTCPeerConnection-helper.js
// getTrackFromUserMedia
// exchangeIceCandidates
// doSignalingHandshake
const stream = await navigator.mediaDevices.getUserMedia({[kind]:true});
const [track] = stream.getTracks();
t.add_cleanup(() => track.stop());
pc1.addTrack(track, stream);
/*
5.3. RTCRtpReceiver Interface
interface RTCRtpReceiver {
...
sequence<RTCRtpContributingSource> getContributingSources();
};
exchangeIceCandidates(pc1, pc2);
const trackEvent = await exchangeOfferAndListenToOntrack(t, pc1, pc2);
await exchangeAnswer(pc1, pc2);
interface RTCRtpContributingSource {
readonly attribute DOMHighResTimeStamp timestamp;
readonly attribute unsigned long source;
readonly attribute byte? audioLevel;
};
assert_array_equals(trackEvent.receiver.getContributingSources(), []);
}
audioLevel
The audio level contained in the last RTP packet played from this source.
audioLevel will be the level value defined in [RFC6465] if the RFC 6465
header extension is present, and otherwise null. RFC 6465 defines the
level as a integral value from 0 to 127 representing the audio level in
negative decibels relative to the loudest signal that the system could
possibly encode. Thus, 0 represents the loudest signal the system could
possibly encode, and 127 represents silence.
*/
promise_test(t => {
const pc1 = new RTCPeerConnection();
t.add_cleanup(() => pc1.close());
const pc2 = new RTCPeerConnection();
t.add_cleanup(() => pc2.close());
const ontrackPromise = new Promise(resolve => {
pc2.addEventListener('track', trackEvent => {
const { receiver } = trackEvent;
assert_true(receiver instanceof RTCRtpReceiver,
'Expect trackEvent.receiver to be instance of RTCRtpReceiver');
resolve(receiver);
});
});
return getTrackFromUserMedia('audio')
.then(([track, mediaStream]) => {
pc1.addTrack(track, mediaStream);
exchangeIceCandidates(pc1, pc2);
return doSignalingHandshake(pc1, pc2);
})
.then(() => ontrackPromise)
.then(receiver => {
const contributingSources = receiver.getContributingSources();
assert_greater_than(contributingSources.length, 0,
'Expect CSRCs to be available after RTP connection is established');
for(const csrc of contributingSources) {
assert_true(csrc instanceof RTCRtpContributingSource,
'Expect contributingSources elements to be instance of RTCRtpContributingSource');
assert_equals(typeof csrc.timestamp, 'number',
'Expect csrc.timestamp attribute to be DOMHighResTimeStamp');
assert_true(Number.isInteger(csrc.source) && csrc.source > 0,
'Expect CSRC identifier to be unsigned long');
if(csrc.audioLevel !== null) {
assert_true(Number.isInteger(csrc.audioLevel) &&
csrc.audioLevel >= 0 && csrc.audioLevel <= 127,
'Expect csrc.audioLevel to be either null or byte value from 0-127.');
}
}
});
}, `getContributingSources() should return list of CSRC after connection is established`);
promise_test(async t => {
await connectAndExpectNoCsrcs(t, 'audio');
}, '[audio] getContributingSources() returns an empty list in loopback call');
promise_test(async t => {
await connectAndExpectNoCsrcs(t, 'video');
}, '[video] getContributingSources() returns an empty list in loopback call');
</script>