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89 lines
3.2 KiB
HTML
89 lines
3.2 KiB
HTML
<!doctype html>
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<meta charset=utf-8>
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<title>RTCRtpReceiver.prototype.getContributingSources</title>
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<script src="/resources/testharness.js"></script>
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<script src="/resources/testharnessreport.js"></script>
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<script src="RTCPeerConnection-helper.js"></script>
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<script>
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'use strict';
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// Test is based on the following editor draft:
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// https://w3c.github.io/webrtc-pc/archives/20170605/webrtc.html
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// The following helper function is called from RTCPeerConnection-helper.js
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// getTrackFromUserMedia
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// exchangeIceCandidates
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// doSignalingHandshake
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/*
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5.3. RTCRtpReceiver Interface
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interface RTCRtpReceiver {
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...
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sequence<RTCRtpContributingSource> getContributingSources();
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};
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interface RTCRtpContributingSource {
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readonly attribute DOMHighResTimeStamp timestamp;
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readonly attribute unsigned long source;
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readonly attribute byte? audioLevel;
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};
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audioLevel
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The audio level contained in the last RTP packet played from this source.
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audioLevel will be the level value defined in [RFC6465] if the RFC 6465
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header extension is present, and otherwise null. RFC 6465 defines the
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level as a integral value from 0 to 127 representing the audio level in
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negative decibels relative to the loudest signal that the system could
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possibly encode. Thus, 0 represents the loudest signal the system could
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possibly encode, and 127 represents silence.
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*/
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promise_test(t => {
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const pc1 = new RTCPeerConnection();
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t.add_cleanup(() => pc1.close());
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const pc2 = new RTCPeerConnection();
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t.add_cleanup(() => pc2.close());
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const ontrackPromise = new Promise(resolve => {
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pc2.addEventListener('track', trackEvent => {
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const { receiver } = trackEvent;
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assert_true(receiver instanceof RTCRtpReceiver,
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'Expect trackEvent.receiver to be instance of RTCRtpReceiver');
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resolve(receiver);
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});
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});
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return getTrackFromUserMedia('audio')
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.then(([track, mediaStream]) => {
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pc1.addTrack(track, mediaStream);
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exchangeIceCandidates(pc1, pc2);
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return doSignalingHandshake(pc1, pc2);
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})
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.then(() => ontrackPromise)
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.then(receiver => {
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const contributingSources = receiver.getContributingSources();
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assert_greater_than(contributingSources.length, 0,
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'Expect CSRCs to be available after RTP connection is established');
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for(const csrc of contributingSources) {
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assert_true(csrc instanceof RTCRtpContributingSource,
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'Expect contributingSources elements to be instance of RTCRtpContributingSource');
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assert_equals(typeof csrc.timestamp, 'number',
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'Expect csrc.timestamp attribute to be DOMHighResTimeStamp');
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assert_true(Number.isInteger(csrc.source) && csrc.source > 0,
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'Expect CSRC identifier to be unsigned long');
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if(csrc.audioLevel !== null) {
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assert_true(Number.isInteger(csrc.audioLevel) &&
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csrc.audioLevel >= 0 && csrc.audioLevel <= 127,
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'Expect csrc.audioLevel to be either null or byte value from 0-127.');
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}
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}
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});
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}, `getContributingSources() should return list of CSRC after connection is established`);
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</script>
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