| <!doctype html> | |
| <meta charset=utf-8> | |
| <title>RTCRtpReceiver.prototype.getSynchronizationSources</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 | |
| /* | |
| 5.3. RTCRtpReceiver Interface | |
| interface RTCRtpReceiver { | |
| ... | |
| sequence<RTCRtpSynchronizationSource> getSynchronizationSources(); | |
| }; | |
| interface RTCRtpSynchronizationSource { | |
| readonly attribute DOMHighResTimeStamp timestamp; | |
| readonly attribute unsigned long source; | |
| readonly attribute byte audioLevel; | |
| readonly attribute boolean? voiceActivityFlag; | |
| }; | |
| */ | |
| 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 syncSources = receiver.getSynchronizationSources(); | |
| assert_greater_than(syncSources.length, 0, | |
| 'Expect SSRCs to be available after RTP connection is established'); | |
| for(const ssrc of syncSources) { | |
| assert_true(ssrc instanceof RTCRtpSynchronizationSource, | |
| 'Expect synchronizationSources elements to be instance of RTCSynchronizationSource'); | |
| assert_equals(typeof ssrc.timestamp, 'number', | |
| 'Expect ssrc.timestamp attribute to be DOMHighResTimeStamp'); | |
| assert_true(Number.isInteger(ssrc.source) && ssrc.source > 0, | |
| 'Expect SSRC identifier to be unsigned long'); | |
| assert_true(Number.isInteger(ssrc.audioLevel) && | |
| ssrc.audioLevel >= 0 && ssrc.audioLevel <= 127, | |
| 'Expect ssrc.audioLevel to be byte value from 0-127.'); | |
| if(ssrc.voiceActivityFlag !== null) { | |
| assert_equals(typeof ssrc.voiceActivityFlag, 'boolean', | |
| 'Expect ssrc.voiceActivity to be either null or boolean'); | |
| } | |
| } | |
| }); | |
| }, `getContributingSources() should return list of CSRC after connection is established`); | |
| </script> |