fune/testing/web-platform/tests/webrtc/RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html
Dominique Hazael-Massieux 4f86b36527 Bug 1888632 [wpt PR 45427] - Move jitterBufferTarget tests to main WebRTC spec, a=testonly
Automatic update from web-platform-tests
Move jitterBufferTarget tests to main WebRTC spec (#45427)

Complementary of https://github.com/w3c/webrtc-pc/pull/2953
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wpt-commits: bdb28625c739df71379a8e5deeb3afc376087f99
wpt-pr: 45427
2024-04-23 09:47:41 +00:00

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HTML

<!DOCTYPE html>
<meta charset="utf-8">
<meta name="timeout" content="long">
<title>Tests RTCRtpReceiver-jitterBufferTarget verified with stats</title>
<script src="/resources/testharness.js"></script>
<script src="/resources/testharnessreport.js"></script>
<script src="/webrtc/RTCPeerConnection-helper.js"></script>
<script src="/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats-helper.js"></script>
<body>
<script>
'use strict'
promise_test(async t => {
await applyJitterBufferTarget(t, "audio", 300);
}, `measure raising and lowering audio jitterBufferTarget`);
</script>
</body>