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Automatic update from web-platform-tests Use getNoiseStream instead of getUserMedia (#20862) * Use getNoiStream instead of getUserMedia Avoid dependency on device permission May enable to switch the tests from https to http (since they should apply generally) * Add dependency to RTCPC-helpers to addTransceiver test to use getNoiseStream -- wpt-commits: 213a7607e8d6ba44d88f1774319e6c1c82ab1ccb wpt-pr: 20862
105 lines
4.5 KiB
HTML
105 lines
4.5 KiB
HTML
<!doctype html>
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<meta charset=utf-8>
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<!-- This file contains two tests that wait for 10 seconds each. -->
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<meta name="timeout" content="long">
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<title>RTCRtpReceiver.prototype.getSynchronizationSources</title>
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<script src="/resources/testharness.js"></script>
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<script src="/resources/testharnessreport.js"></script>
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<script src="RTCPeerConnection-helper.js"></script>
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<script>
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'use strict';
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async function initiateSingleTrackCallAndReturnReceiver(t, kind) {
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const pc1 = new RTCPeerConnection();
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t.add_cleanup(() => pc1.close());
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const pc2 = new RTCPeerConnection();
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t.add_cleanup(() => pc2.close());
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const stream = await getNoiseStream({[kind]:true});
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const [track] = stream.getTracks();
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t.add_cleanup(() => track.stop());
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pc1.addTrack(track, stream);
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exchangeIceCandidates(pc1, pc2);
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const trackEvent = await exchangeOfferAndListenToOntrack(t, pc1, pc2);
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await exchangeAnswer(pc1, pc2);
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return trackEvent.receiver;
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}
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for (const kind of ['audio', 'video']) {
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promise_test(async t => {
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const receiver = await initiateSingleTrackCallAndReturnReceiver(t, kind);
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await listenForSSRCs(t, receiver);
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}, '[' + kind + '] getSynchronizationSources() eventually returns a ' +
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'non-empty list');
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promise_test(async t => {
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const startTime = performance.now();
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const receiver = await initiateSingleTrackCallAndReturnReceiver(t, kind);
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const [ssrc] = await listenForSSRCs(t, receiver);
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assert_equals(typeof ssrc.timestamp, 'number');
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assert_true(ssrc.timestamp >= startTime);
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}, '[' + kind + '] RTCRtpSynchronizationSource.timestamp is a number');
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promise_test(async t => {
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const receiver = await initiateSingleTrackCallAndReturnReceiver(t, kind);
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const [ssrc] = await listenForSSRCs(t, receiver);
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assert_equals(typeof ssrc.rtpTimestamp, 'number');
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assert_greater_than_equal(ssrc.rtpTimestamp, 0);
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assert_less_than_equal(ssrc.rtpTimestamp, 0xffffffff);
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}, '[' + kind + '] RTCRtpSynchronizationSource.rtpTimestamp is a number ' +
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'[0, 2^32-1]');
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promise_test(async t => {
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const receiver = await initiateSingleTrackCallAndReturnReceiver(t, kind);
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// Wait for packets to start flowing.
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await listenForSSRCs(t, receiver);
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// Wait for 10 seconds.
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await new Promise(resolve => t.step_timeout(resolve, 10000));
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let earliestTimestamp = undefined;
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let latestTimestamp = undefined;
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for (const ssrc of await listenForSSRCs(t, receiver)) {
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if (earliestTimestamp == undefined || earliestTimestamp > ssrc.timestamp)
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earliestTimestamp = ssrc.timestamp;
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if (latestTimestamp == undefined || latestTimestamp < ssrc.timestamp)
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latestTimestamp = ssrc.timestamp;
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}
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assert_true(latestTimestamp - earliestTimestamp <= 10000);
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}, '[' + kind + '] getSynchronizationSources() does not contain SSRCs ' +
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'older than 10 seconds');
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promise_test(async t => {
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const startTime = performance.timeOrigin + performance.now();
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const receiver = await initiateSingleTrackCallAndReturnReceiver(t, kind);
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const [ssrc] = await listenForSSRCs(t, receiver);
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const endTime = performance.timeOrigin + performance.now();
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assert_true(startTime <= ssrc.timestamp && ssrc.timestamp <= endTime);
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}, '[' + kind + '] RTCRtpSynchronizationSource.timestamp is comparable to ' +
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'performance.timeOrigin + performance.now()');
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promise_test(async t => {
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const receiver = await initiateSingleTrackCallAndReturnReceiver(t, kind);
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const [ssrc] = await listenForSSRCs(t, receiver);
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assert_equals(typeof ssrc.source, 'number');
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}, '[' + kind + '] RTCRtpSynchronizationSource.source is a number');
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}
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promise_test(async t => {
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const receiver = await initiateSingleTrackCallAndReturnReceiver(t, 'audio');
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const [ssrc] = await listenForSSRCs(t, receiver);
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assert_equals(typeof ssrc.audioLevel, 'number');
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assert_greater_than_equal(ssrc.audioLevel, 0);
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assert_less_than_equal(ssrc.audioLevel, 1);
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}, '[audio-only] RTCRtpSynchronizationSource.audioLevel is a number [0, 1]');
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// This test only passes if the implementation is sending the RFC 6464 extension
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// header and the "vad" extension attribute is not "off", otherwise
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// voiceActivityFlag is absent. TODO: Consider moving this test to an
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// optional-to-implement subfolder?
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promise_test(async t => {
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const receiver = await initiateSingleTrackCallAndReturnReceiver(t, 'audio');
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const [ssrc] = await listenForSSRCs(t, receiver);
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assert_equals(typeof ssrc.voiceActivityFlag, 'boolean');
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}, '[audio-only] RTCRtpSynchronizationSource.voiceActivityFlag is a boolean');
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</script>
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