webrtc/protocol/rtp-clockrate.html

<!doctype html>
<meta charset=utf-8>
<!-- This file contains a test that waits for two seconds. -->
<meta name="timeout" content="long">
<title>RTP clockrate</title>
<script src="/resources/testharness.js"></script>
<script src="/resources/testharnessreport.js"></script>
<script src="../RTCPeerConnection-helper.js"></script>
<script>
'use strict';

async function initiateSingleTrackCallAndReturnReceiver(t, kind) {
  const pc1 = new RTCPeerConnection();
  t.add_cleanup(() => pc1.close());
  const pc2 = new RTCPeerConnection();
  t.add_cleanup(() => pc2.close());

  const stream = await getNoiseStream({[kind]:true});
  const [track] = stream.getTracks();
  t.add_cleanup(() => track.stop());
  pc1.addTrack(track, stream);

  exchangeIceCandidates(pc1, pc2);
  const trackEvent = await exchangeOfferAndListenToOntrack(t, pc1, pc2);
  await exchangeAnswer(pc1, pc2);
  await waitForConnectionStateChange(pc2, ['connected']);
  return trackEvent.receiver;
}

promise_test(async t => {
  // the getSynchronizationSources API exposes the rtp timestamp.
  const receiver = await initiateSingleTrackCallAndReturnReceiver(t, 'video');
  const first = await listenForSSRCs(t, receiver);
  await new Promise(resolve => t.step_timeout(resolve, 2000));
  const second = await listenForSSRCs(t, receiver);
  // rtpTimestamp may wrap at 0xffffffff, take care of that.
  const actualClockRate = ((second[0].rtpTimestamp - first[0].rtpTimestamp + 0xffffffff) % 0xffffffff) / (second[0].timestamp - first[0].timestamp) * 1000;
  assert_approx_equals(actualClockRate, 90000, 9000, 'Video clockrate is approximately 90000');
}, 'video rtp timestamps increase by approximately 90000 per second');
</script>