RTCRtpSynchronizationSource-senderCaptureTimeOffset.html

<!doctype html>
<meta charset=utf-8>
<!-- This file contains a test that waits for 2 seconds. -->
<meta name="timeout" content="long">
<title>senderCaptureTimeOffset attribute in RTCRtpSynchronizationSource</title>
<div><video id="remote" width="124" height="124" autoplay></video></div>
<script src="/resources/testharness.js"></script>
<script src="/resources/testharnessreport.js"></script>
<script src="/webrtc/RTCPeerConnection-helper.js"></script>
<script src="/webrtc-extensions/RTCRtpSynchronizationSource-helper.js"></script>
<script>
'use strict';

function listenForSenderCaptureTimeOffset(t, receiver) {
  return new Promise((resolve) => {
    function listen() {
      const ssrcs = receiver.getSynchronizationSources();
      assert_true(ssrcs != undefined);
      if (ssrcs.length > 0) {
        assert_equals(ssrcs.length, 1);
        if (ssrcs[0].captureTimestamp != undefined) {
          resolve(ssrcs[0].senderCaptureTimeOffset);
          return true;
        }
      }
      return false;
    };
    t.step_wait(listen, 'No abs-capture-time capture time header extension.');
  });
}

// Passes if `getSynchronizationSources()` contains `senderCaptureTimeOffset` if
// and only if expected.
for (const kind of ['audio', 'video']) {
  promise_test(async t => {
    const [caller, callee] = await initiateSingleTrackCall(
        t, /* caps=  */{[kind]: true}, /* absCaptureTimeOffered= */false,
        /* absCaptureTimeAnswered= */false);
    const receiver = callee.getReceivers()[0];

    for (const ssrc of await listenForSSRCs(t, receiver)) {
      assert_equals(typeof ssrc.senderCaptureTimeOffset, 'undefined');
    }
  }, '[' + kind + '] getSynchronizationSources() should not contain ' +
      'senderCaptureTimeOffset if absolute capture time RTP header extension ' +
      'is not offered');

  promise_test(async t => {
    const [caller, callee] = await initiateSingleTrackCall(
        t, /* caps=  */{[kind]: true}, /* absCaptureTimeOffered= */false,
        /* absCaptureTimeAnswered= */false);
    const receiver = callee.getReceivers()[0];

    for (const ssrc of await listenForSSRCs(t, receiver)) {
      assert_equals(typeof ssrc.senderCaptureTimeOffset, 'undefined');
    }
  }, '[' + kind + '] getSynchronizationSources() should not contain ' +
      'senderCaptureTimeOffset if absolute capture time RTP header extension ' +
      'is offered, but not answered');

  promise_test(async t => {
    const [caller, callee] = await initiateSingleTrackCall(
        t, /* caps=  */{[kind]: true}, /* absCaptureTimeOffered= */true,
        /* absCaptureTimeAnswered= */true);
    const receiver = callee.getReceivers()[0];
    let senderCaptureTimeOffset = await listenForSenderCaptureTimeOffset(
        t, receiver);
    assert_true(senderCaptureTimeOffset != undefined);
  }, '[' + kind + '] getSynchronizationSources() should contain ' +
      'senderCaptureTimeOffset if absolute capture time RTP header extension ' +
      'is negotiated');
}

// Passes if `senderCaptureTimeOffset` is zero, which is expected since the test
// creates a local peer connection between `caller` and `callee`.
promise_test(async t => {
  const [caller, callee] = await initiateSingleTrackCall(
      t, /* caps=  */{audio: true, video: true},
      /* absCaptureTimeOffered= */true, /* absCaptureTimeAnswered= */true);
  const receivers = callee.getReceivers();
  assert_equals(receivers.length, 2);

  for (let i = 0; i < 2; ++i) {
    let senderCaptureTimeOffset = await listenForSenderCaptureTimeOffset(
        t, receivers[i]);
    assert_equals(senderCaptureTimeOffset, 0);
  }
}, 'Audio and video RTCRtpSynchronizationSource.senderCaptureTimeOffset must ' +
   'be zero');

</script>
chromium/third_party/blink/web_tests/external/wpt/webrtc-extensions/RTCRtpSynchronizationSource-senderCaptureTimeOffset.html
