| <!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> |