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https://github.com/web-platform-tests/wpt
26 December 2018, 22:31:15 UTC
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Tip revision: 26e6ef3f13680c4d02b33bf1235884119f9e26e7 authored by James Graham on 04 April 2018, 18:33:40 UTC
Add long timeout to unstable CSP tests, a=testonly on a CLOSED TREE
Tip revision: 26e6ef3
RTCRtpReceiver-getStats.https.html 
<!doctype html>
<meta charset=utf-8>
<title>RTCRtpReceiver.prototype.getStats</title>
<script src="/resources/testharness.js"></script>
<script src="/resources/testharnessreport.js"></script>
<script src="RTCPeerConnection-helper.js"></script>
<script src="dictionary-helper.js"></script>
<script src="RTCStats-helper.js"></script>
<script>
  'use strict';

  // Test is based on the following editor draft:
  // https://w3c.github.io/webrtc-pc/archives/20170605/webrtc.html
  // https://w3c.github.io/webrtc-stats/archives/20170614/webrtc-stats.html

  // The following helper functions are called from RTCPeerConnection-helper.js:
  //   doSignalingHandshake

  // The following helper function is called from RTCStats-helper.js
  //   validateStatsReport
  //   assert_stats_report_has_stats

  /*
    5.3.  RTCRtpReceiver Interface
      interface RTCRtpReceiver {
         Promise<RTCStatsReport> getStats();
          ...
      };

      getStats
        1.  Let selector be the RTCRtpReceiver object on which the method was invoked.
        2.  Let p be a new promise, and run the following steps in parallel:
          1.  Gather the stats indicated by selector according to the stats selection
              algorithm.
          2.  Resolve p with the resulting RTCStatsReport object, containing the
              gathered stats.
        3.  Return p.

    8.5. The stats selection algorithm
      4.  If selector is an RTCRtpReceiver, gather stats for and add the following objects
          to result:
        - All RTCInboundRTPStreamStats objects corresponding to selector.
        - All stats objects referenced directly or indirectly by the RTCInboundRTPStreamStats
          added.
   */

  promise_test(async () => {
    const caller = new RTCPeerConnection();
    const callee = new RTCPeerConnection();
    const { receiver } = caller.addTransceiver('audio');

    await doSignalingHandshake(caller, callee);
    const statsReport = await receiver.getStats();
    validateStatsReport(statsReport);
    assert_stats_report_has_stats(statsReport, ['inbound-rtp']);
  }, 'receiver.getStats() via addTransceiver should return stats report containing inbound-rtp stats');

  promise_test(async () => {
    const caller = new RTCPeerConnection();
    const callee = new RTCPeerConnection();
    const stream = await navigator.mediaDevices.getUserMedia({audio:true});
    const [track] = stream.getTracks();
    caller.addTrack(track, stream);

    await doSignalingHandshake(caller, callee);
    const receiver = callee.getReceivers()[0];
    const statsReport = await receiver.getStats();
    validateStatsReport(statsReport);
    assert_stats_report_has_stats(statsReport, ['inbound-rtp']);
  }, 'receiver.getStats() via addTrack should return stats report containing inbound-rtp stats');

</script>
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