| <!doctype html> |
| <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> |
