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diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_audioContributingSources.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioContributingSources.html new file mode 100644 index 0000000000..333b40a888 --- /dev/null +++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioContributingSources.html @@ -0,0 +1,144 @@ +<!DOCTYPE HTML> +<html> +<head> + <script type="application/javascript" src="pc.js"></script> +</head> +<body> +<pre id="test"> +<script type="application/javascript"> + createHTML({ + bug: "1363667", + title: "Test audio receiver getContributingSources" + }); + + // test_peerConnection_audioSynchronizationSources.html tests + // much of the functionality of getContributingSources as the implementation + // is shared. + var testGetContributingSources = async (test) => { + const remoteReceiver = test.pcRemote.getReceivers()[0]; + const localReceiver = test.pcLocal.getReceivers()[0]; + + // Check that getContributingSources is empty as there is no MCU + is(remoteReceiver.getContributingSources().length, 0, + "remote contributing sources is empty"); + is(localReceiver.getContributingSources().length, 0, + "local contributing sources is empty"); + // Wait for the next JS event loop iteration, to clear the cache + await Promise.resolve().then(); + // Insert new entries as if there were an MCU + const csrc0 = 124756; + const timestamp0 = performance.now() + performance.timeOrigin; + const rtpTimestamp0 = 11111; + const hasAudioLevel0 = true; + // Audio level as expected to be received in RTP + const audioLevel0 = 34; + // Audio level as expected to be returned + const expectedAudioLevel0 = 10 ** (-audioLevel0 / 20); + + SpecialPowers.wrap(remoteReceiver).mozInsertAudioLevelForContributingSource( + csrc0, + timestamp0, + rtpTimestamp0, + hasAudioLevel0, + audioLevel0); + + const csrc1 = 5786; + const timestamp1 = timestamp0 - 200; + const rtpTimestamp1 = 22222; + const hasAudioLevel1 = false; + const audioLevel1 = 0; + + SpecialPowers.wrap(remoteReceiver).mozInsertAudioLevelForContributingSource( + csrc1, + timestamp1, + rtpTimestamp1, + hasAudioLevel1, + audioLevel1); + + const csrc2 = 93487; + const timestamp2 = timestamp0 - 200; + const rtpTimestamp2 = 333333; + const hasAudioLevel2 = true; + const audioLevel2 = 127; + + SpecialPowers.wrap(remoteReceiver).mozInsertAudioLevelForContributingSource( + csrc2, + timestamp2, + rtpTimestamp2, + hasAudioLevel2, + audioLevel2); + + const contributingSources = remoteReceiver.getContributingSources(); + is(contributingSources.length, 3, + "Expected number of contributing sources"); + + // Check that both inserted were returned + const source0 = contributingSources.find(c => c.source == csrc0); + ok(source0, "first csrc was found"); + + const source1 = contributingSources.find(c => c.source == csrc1); + ok(source1, "second csrsc was found"); + + // Add a small margin of error in the timestamps + const compareTimestamps = (ts1, ts2) => Math.abs(ts1 - ts2) < 100; + + // Check the CSRC with audioLevel + const isWithinErr = Math.abs(source0.audioLevel - expectedAudioLevel0) + < expectedAudioLevel0 / 50; + ok(isWithinErr, + `Contributing source has correct audio level. (${source0.audioLevel})`); + ok(compareTimestamps(source0.timestamp, timestamp0), + `Contributing source has correct timestamp (got ${source0.timestamp}), expected ${timestamp0}`); + is(source0.rtpTimestamp, rtpTimestamp0, + `Contributing source has correct RTP timestamp (${source0.rtpTimestamp}`); + // Check the CSRC without audioLevel + is(source1.audioLevel, undefined, + `Contributing source has no audio level. (${source1.audioLevel})`); + ok(compareTimestamps(source1.timestamp, timestamp1), + `Contributing source has correct timestamp (got ${source1.timestamp}, expected ${timestamp1})`); + is(source1.rtpTimestamp, rtpTimestamp1, + `Contributing source has correct RTP timestamp (${source1.rtpTimestamp}`); + // Check that a received RTP audio level 127 is exactly 0 + const source2 = contributingSources.find(c => c.source == csrc2); + ok(source2, "third csrc was found"); + is(source2.audioLevel, 0, + `Contributing source has audio level of 0 when RTP audio level is 127`); + // Check caching + is(JSON.stringify(contributingSources), + JSON.stringify(remoteReceiver.getContributingSources()), + "getContributingSources is cached"); + // Check that sources are sorted in descending order by time stamp + const timestamp3 = performance.now() + performance.timeOrigin; + const rtpTimestamp3 = 44444; + // Larger offsets are further back in time + const testOffsets = [3, 7, 5, 6, 1, 4]; + for (const offset of testOffsets) { + SpecialPowers.wrap(localReceiver).mozInsertAudioLevelForContributingSource( + offset, // Using offset for SSRC for convenience + timestamp3 - offset, + rtpTimestamp3, + true, + offset); + } + const sources = localReceiver.getContributingSources(); + const sourceOffsets = sources.map(s => s.source); + is(JSON.stringify(sourceOffsets), + JSON.stringify([...testOffsets].sort((a, b) => a - b)), + `Contributing sources are sorted in descending order by timestamp:` + + ` ${JSON.stringify(sources)}`); + }; + + var test; + runNetworkTest(async function(options) { + test = new PeerConnectionTest(options); + test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW", + [testGetContributingSources]); + test.setMediaConstraints([{audio: true}], [{audio: true}]); + test.pcLocal.audioElementsOnly = true; + await pushPrefs(["privacy.reduceTimerPrecision", false]); + await test.run(); + }); +</script> +</pre> +</body> +</html> |