File: RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html

package info (click to toggle)
firefox-esr 140.4.0esr-1~deb13u1
  • links: PTS, VCS
  • area: main
  • in suites: trixie
  • size: 4,539,284 kB
  • sloc: cpp: 7,381,286; javascript: 6,388,710; ansic: 3,710,139; python: 1,393,780; xml: 628,165; asm: 426,916; java: 184,004; sh: 65,742; makefile: 19,302; objc: 13,059; perl: 12,912; yacc: 4,583; cs: 3,846; pascal: 3,352; lex: 1,720; ruby: 1,226; exp: 762; php: 436; lisp: 258; awk: 247; sql: 66; sed: 54; csh: 10
file content (18 lines) | stat: -rw-r--r-- 585 bytes parent folder | download | duplicates (12) 
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
 
<!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/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>