test_peerConnection_stats_jitter.html

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<head>

  <script type="application/javascript" src="pc.js"></script>

  <script type="application/javascript" src="stats.js"></script>

</head>

<body>

<pre id="test">

<script type="application/javascript">

  createHTML({

    bug: "1672590",

    title: "Jitter sanity check"

});

const checkJitter = stats => {

  stats.forEach((stat, mapKey) => {

    if (stat.type == "remote-inbound-rtp") {

      // This should be much lower for audio, TODO: Bug 1330575

      const expectedJitter = stat.kind == "video" ? 0.5 : 1;

      ok(stat.jitter < expectedJitter,

          stat.type + ".jitter is sane number for a local only test. value="

          + stat.jitter);

});

};

const PC_LOCAL_TEST_LOCAL_JITTER = async test => {

  checkJitter(await waitForSyncedRtcp(test.pcLocal._pc));

const PC_REMOTE_TEST_REMOTE_JITTER = async test => {

  checkJitter(await waitForSyncedRtcp(test.pcRemote._pc));

runNetworkTest(async function (options) {

  // We don't know how to get QP value when using Android system codecs.

  if (navigator.userAgent.includes("Android")) {

    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],

                    ["media.webrtc.hw.h264.enabled", false]);

  const test = new PeerConnectionTest(options);

  test.chain.insertAfter("PC_LOCAL_WAIT_FOR_MEDIA_FLOW",

    [PC_LOCAL_TEST_LOCAL_JITTER]);

  test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",

    [PC_REMOTE_TEST_REMOTE_JITTER]);

  test.setMediaConstraints([{audio: true}, {video: true}],

                           [{audio: true}, {video: true}]);

  await test.run();

});

</script>

</pre>

</body>

</html>