Revision b55e29597d92ab89063673f031151e2bdb4b9479 authored by Joe Downing on 22 March 2018, 07:35:35 UTC, committed by Chromium WPT Sync on 22 March 2018, 07:35:35 UTC
This change moves the KeyboardLock API methods to a 'keyboard' namespace on the Navigator object. We are doing this work now as there has been a request for additional keyboard functionality that would also be placed on the new keyboard object and we wanted to move the KeyboardLock methods there for consistency before we launch. KeyboardLock API Spec is here: https://w3c.github.io/keyboard-lock/#API Old calling pattern: Navigator.keyboardLock(); Navigator.keyboardUnlock(); New calling pattern: Navigator.keyboard.lock(); Navigator.keyboard.unlock(); Note: The main logic in the KeyboardLock.cpp class and tests is the same as it was, however the file changed enough that git does not recognize it as a file move. BUG=680809 Change-Id: I234b2ab12d5ecd44c894ed5103863fd96fd548d4 Reviewed-on: https://chromium-review.googlesource.com/969656 Reviewed-by: Philip Jägenstedt <foolip@chromium.org> Reviewed-by: Gary Kacmarcik <garykac@chromium.org> Reviewed-by: Daniel Cheng <dcheng@chromium.org> Commit-Queue: Daniel Cheng <dcheng@chromium.org> Cr-Commit-Position: refs/heads/master@{#544996}
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RTCRtpReceiver-getSynchronizationSources.https.html
<!doctype html>
<meta charset=utf-8>
<title>RTCRtpReceiver.prototype.getSynchronizationSources</title>
<script src="/resources/testharness.js"></script>
<script src="/resources/testharnessreport.js"></script>
<script src="RTCPeerConnection-helper.js"></script>
<script>
'use strict';
// Test is based on the following editor draft:
// https://w3c.github.io/webrtc-pc/archives/20170605/webrtc.html
/*
5.3. RTCRtpReceiver Interface
interface RTCRtpReceiver {
...
sequence<RTCRtpSynchronizationSource> getSynchronizationSources();
};
interface RTCRtpSynchronizationSource {
readonly attribute DOMHighResTimeStamp timestamp;
readonly attribute unsigned long source;
readonly attribute byte audioLevel;
readonly attribute boolean? voiceActivityFlag;
};
*/
promise_test(() => {
const pc1 = new RTCPeerConnection();
const pc2 = new RTCPeerConnection();
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 syncSources = receiver.getSynchronizationSources();
assert_greater_than(syncSources.length, 0,
'Expect SSRCs to be available after RTP connection is established');
for(const ssrc of syncSources) {
assert_true(ssrc instanceof RTCRtpSynchronizationSource,
'Expect synchronizationSources elements to be instance of RTCSynchronizationSource');
assert_equals(typeof ssrc.timestamp, 'number',
'Expect ssrc.timestamp attribute to be DOMHighResTimeStamp');
assert_true(Number.isInteger(ssrc.source) && ssrc.source > 0,
'Expect SSRC identifier to be unsigned long');
assert_true(Number.isInteger(ssrc.audioLevel) &&
ssrc.audioLevel >= 0 && ssrc.audioLevel <= 127,
'Expect ssrc.audioLevel to be byte value from 0-127.');
if(ssrc.voiceActivityFlag !== null) {
assert_equals(typeof ssrc.voiceActivityFlag, 'boolean',
'Expect ssrc.voiceActivity to be either null or boolean');
}
}
});
}, `getContributingSources() should return list of CSRC after connection is established`);
</script>
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