forked from mirrors/gecko-dev
27d69106ae
Automatic update from web-platform-tests webrtc wpt: remove RTCStats-helper.js after removing all usage of the helpers in there. BUG=chromium:1395574 Change-Id: I97a904d12f9447a89bde4db35fb25352b9cc5c77 Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/5246286 Reviewed-by: Henrik Boström <hbos@chromium.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#1253881} -- wpt-commits: 0b1c92d4b748d8cb7e7431d3e5f6bd9817a34d65 wpt-pr: 44285
93 lines
3.6 KiB
HTML
93 lines
3.6 KiB
HTML
<!doctype html>
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<meta charset=utf-8>
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<!-- This file contains a test that waits for 2 seconds. -->
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<meta name="timeout" content="long">
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<title>captureTimestamp attribute in RTCRtpSynchronizationSource</title>
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<div><video id="remote" width="124" height="124" autoplay></video></div>
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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="/webrtc/RTCPeerConnection-helper.js"></script>
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<script src="/webrtc-extensions/RTCRtpSynchronizationSource-helper.js"></script>
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<script>
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'use strict';
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function listenForCaptureTimestamp(t, receiver) {
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return new Promise((resolve) => {
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function listen() {
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const ssrcs = receiver.getSynchronizationSources();
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assert_true(ssrcs != undefined);
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if (ssrcs.length > 0) {
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assert_equals(ssrcs.length, 1);
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if (ssrcs[0].captureTimestamp != undefined) {
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resolve(ssrcs[0].captureTimestamp);
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return true;
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}
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}
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return false;
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};
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t.step_wait(listen, 'No abs-capture-time capture time header extension.');
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});
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}
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// Passes if `getSynchronizationSources()` contains `captureTimestamp` if and
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// only if expected.
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for (const kind of ['audio', 'video']) {
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promise_test(async t => {
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const [caller, callee] = await initiateSingleTrackCall(
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t, /* caps= */{[kind]: true}, /* absCaptureTimeOffered= */false,
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/* absCaptureTimeAnswered= */false);
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const receiver = callee.getReceivers()[0];
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for (const ssrc of await listenForSSRCs(t, receiver)) {
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assert_equals(typeof ssrc.captureTimestamp, 'undefined');
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}
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}, '[' + kind + '] getSynchronizationSources() should not contain ' +
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'captureTimestamp if absolute capture time RTP header extension is not ' +
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'offered');
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promise_test(async t => {
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const [caller, callee] = await initiateSingleTrackCall(
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t, /* caps= */{[kind]: true}, /* absCaptureTimeOffered= */false,
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/* absCaptureTimeAnswered= */false);
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const receiver = callee.getReceivers()[0];
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for (const ssrc of await listenForSSRCs(t, receiver)) {
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assert_equals(typeof ssrc.captureTimestamp, 'undefined');
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}
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}, '[' + kind + '] getSynchronizationSources() should not contain ' +
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'captureTimestamp if absolute capture time RTP header extension is ' +
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'offered, but not answered');
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promise_test(async t => {
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const [caller, callee] = await initiateSingleTrackCall(
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t, /* caps= */{[kind]: true}, /* absCaptureTimeOffered= */true,
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/* absCaptureTimeAnswered= */true);
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const receiver = callee.getReceivers()[0];
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await listenForCaptureTimestamp(t, receiver);
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}, '[' + kind + '] getSynchronizationSources() should contain ' +
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'captureTimestamp if absolute capture time RTP header extension is ' +
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'negotiated');
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}
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// Passes if `captureTimestamp` for audio and video are comparable, which is
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// expected since the test creates a local peer connection between `caller` and
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// `callee`.
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promise_test(async t => {
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const [caller, callee] = await initiateSingleTrackCall(
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t, /* caps= */{audio: true, video: true},
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/* absCaptureTimeOffered= */true, /* absCaptureTimeAnswered= */true);
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const receivers = callee.getReceivers();
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assert_equals(receivers.length, 2);
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let captureTimestamps = [undefined, undefined];
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const t0 = performance.now();
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for (let i = 0; i < 2; ++i) {
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captureTimestamps[i] = await listenForCaptureTimestamp(t, receivers[i]);
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}
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const t1 = performance.now();
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assert_less_than(Math.abs(captureTimestamps[0] - captureTimestamps[1]),
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t1 - t0);
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}, 'Audio and video RTCRtpSynchronizationSource.captureTimestamp are ' +
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'comparable');
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</script>
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