fune/testing/web-platform/tests/webrtc/RTCRtpReceiver-getContributingSources.https.html

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<meta charset=utf-8>
<title>RTCRtpReceiver.prototype.getContributingSources</title>
<script src="/resources/testharness.js"></script>
<script src="/resources/testharnessreport.js"></script>
<script src="RTCPeerConnection-helper.js"></script>
<script>
  'use strict';

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

  // The following helper function is called from RTCPeerConnection-helper.js
  //   getTrackFromUserMedia
  //   exchangeIceCandidates
  //   doSignalingHandshake

  /*
    5.3.  RTCRtpReceiver Interface
      interface RTCRtpReceiver {
        ...
        sequence<RTCRtpContributingSource>    getContributingSources();
      };

      interface RTCRtpContributingSource {
        readonly attribute DOMHighResTimeStamp timestamp;
        readonly attribute unsigned long       source;
        readonly attribute byte?               audioLevel;
      };

      audioLevel
        The audio level contained in the last RTP packet played from this source.
        audioLevel will be the level value defined in [RFC6465] if the RFC 6465
        header extension is present, and otherwise null. RFC 6465 defines the
        level as a integral value from 0 to 127 representing the audio level in
        negative decibels relative to the loudest signal that the system could
        possibly encode. Thus, 0 represents the loudest signal the system could
        possibly encode, and 127 represents silence.
   */

  promise_test(() => {
    const pc1 = new RTCPeerConnection();
    const pc2 = new RTCPeerConnection();

    const ontrackPromise = new Promise(resolve => {
      pc2.addEventListener('track', trackEvent => {
        const { receiver } = trackEvent;
        assert_true(receiver instanceof RTCRtpReceiver,
          'Expect trackEvent.receiver to be instance of RTCRtpReceiver');

        resolve(receiver);
      });
    });

    return getTrackFromUserMedia('audio')
    .then(([track, mediaStream]) => {
      pc1.addTrack(track, mediaStream);
      exchangeIceCandidates(pc1, pc2);
      return doSignalingHandshake(pc1, pc2);
    })
    .then(() => ontrackPromise)
    .then(receiver => {
      const contributingSources = receiver.getContributingSources();
      assert_greater_than(contributingSources.length, 0,
        'Expect CSRCs to be available after RTP connection is established');

      for(const csrc of contributingSources) {
        assert_true(csrc instanceof RTCRtpContributingSource,
          'Expect contributingSources elements to be instance of RTCRtpContributingSource');

        assert_equals(typeof csrc.timestamp, 'number',
          'Expect csrc.timestamp attribute to be DOMHighResTimeStamp');

        assert_true(Number.isInteger(csrc.source) && csrc.source > 0,
          'Expect CSRC identifier to be unsigned long');

        if(csrc.audioLevel !== null) {
          assert_true(Number.isInteger(csrc.audioLevel) &&
             csrc.audioLevel >= 0 && csrc.audioLevel <= 127,
            'Expect csrc.audioLevel to be either null or byte value from 0-127.');
        }
      }
    });
  }, `getContributingSources() should return list of CSRC after connection is established`);

</script>