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| author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
|---|---|---|
| committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
| commit | 43a97878ce14b72f0981164f87f2e35e14151312 (patch) | |
| tree | 620249daf56c0258faa40cbdcf9cfba06de2a846 /testing/web-platform/tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html | |
| parent | Initial commit. (diff) | |
| download | firefox-43a97878ce14b72f0981164f87f2e35e14151312.tar.xz firefox-43a97878ce14b72f0981164f87f2e35e14151312.zip | |
Adding upstream version 110.0.1.upstream/110.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'testing/web-platform/tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html')
| -rw-r--r-- | testing/web-platform/tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html | 105 |
1 files changed, 105 insertions, 0 deletions
diff --git a/testing/web-platform/tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html b/testing/web-platform/tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html new file mode 100644 index 0000000000..8436a44ebc --- /dev/null +++ b/testing/web-platform/tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html @@ -0,0 +1,105 @@ +<!doctype html> +<meta charset=utf-8> +<!-- This file contains two tests that wait for 10 seconds each. --> +<meta name="timeout" content="long"> +<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'; + +async function initiateSingleTrackCallAndReturnReceiver(t, kind) { + const pc1 = new RTCPeerConnection(); + t.add_cleanup(() => pc1.close()); + const pc2 = new RTCPeerConnection(); + t.add_cleanup(() => pc2.close()); + + const stream = await getNoiseStream({[kind]:true}); + const [track] = stream.getTracks(); + t.add_cleanup(() => track.stop()); + pc1.addTrack(track, stream); + + exchangeIceCandidates(pc1, pc2); + const trackEvent = await exchangeOfferAndListenToOntrack(t, pc1, pc2); + await exchangeAnswer(pc1, pc2); + return trackEvent.receiver; +} + +for (const kind of ['audio', 'video']) { + promise_test(async t => { + const receiver = await initiateSingleTrackCallAndReturnReceiver(t, kind); + await listenForSSRCs(t, receiver); + }, '[' + kind + '] getSynchronizationSources() eventually returns a ' + + 'non-empty list'); + + promise_test(async t => { + const startTime = performance.now(); + const receiver = await initiateSingleTrackCallAndReturnReceiver(t, kind); + const [ssrc] = await listenForSSRCs(t, receiver); + assert_equals(typeof ssrc.timestamp, 'number'); + assert_true(ssrc.timestamp >= startTime); + }, '[' + kind + '] RTCRtpSynchronizationSource.timestamp is a number'); + + promise_test(async t => { + const receiver = await initiateSingleTrackCallAndReturnReceiver(t, kind); + const [ssrc] = await listenForSSRCs(t, receiver); + assert_equals(typeof ssrc.rtpTimestamp, 'number'); + assert_greater_than_equal(ssrc.rtpTimestamp, 0); + assert_less_than_equal(ssrc.rtpTimestamp, 0xffffffff); + }, '[' + kind + '] RTCRtpSynchronizationSource.rtpTimestamp is a number ' + + '[0, 2^32-1]'); + + promise_test(async t => { + const receiver = await initiateSingleTrackCallAndReturnReceiver(t, kind); + // Wait for packets to start flowing. + await listenForSSRCs(t, receiver); + // Wait for 10 seconds. + await new Promise(resolve => t.step_timeout(resolve, 10000)); + let earliestTimestamp = undefined; + let latestTimestamp = undefined; + for (const ssrc of await listenForSSRCs(t, receiver)) { + if (earliestTimestamp == undefined || earliestTimestamp > ssrc.timestamp) + earliestTimestamp = ssrc.timestamp; + if (latestTimestamp == undefined || latestTimestamp < ssrc.timestamp) + latestTimestamp = ssrc.timestamp; + } + assert_true(latestTimestamp - earliestTimestamp <= 10000); + }, '[' + kind + '] getSynchronizationSources() does not contain SSRCs ' + + 'older than 10 seconds'); + + promise_test(async t => { + const startTime = performance.timeOrigin + performance.now(); + const receiver = await initiateSingleTrackCallAndReturnReceiver(t, kind); + const [ssrc] = await listenForSSRCs(t, receiver); + const endTime = performance.timeOrigin + performance.now(); + assert_true(startTime <= ssrc.timestamp && ssrc.timestamp <= endTime); + }, '[' + kind + '] RTCRtpSynchronizationSource.timestamp is comparable to ' + + 'performance.timeOrigin + performance.now()'); + + promise_test(async t => { + const receiver = await initiateSingleTrackCallAndReturnReceiver(t, kind); + const [ssrc] = await listenForSSRCs(t, receiver); + assert_equals(typeof ssrc.source, 'number'); + }, '[' + kind + '] RTCRtpSynchronizationSource.source is a number'); +} + +promise_test(async t => { + const receiver = await initiateSingleTrackCallAndReturnReceiver(t, 'audio'); + const [ssrc] = await listenForSSRCs(t, receiver); + assert_equals(typeof ssrc.audioLevel, 'number'); + assert_greater_than_equal(ssrc.audioLevel, 0); + assert_less_than_equal(ssrc.audioLevel, 1); +}, '[audio-only] RTCRtpSynchronizationSource.audioLevel is a number [0, 1]'); + +// This test only passes if the implementation is sending the RFC 6464 extension +// header and the "vad" extension attribute is not "off", otherwise +// voiceActivityFlag is absent. TODO: Consider moving this test to an +// optional-to-implement subfolder? +promise_test(async t => { + const receiver = await initiateSingleTrackCallAndReturnReceiver(t, 'audio'); + const [ssrc] = await listenForSSRCs(t, receiver); + assert_equals(typeof ssrc.voiceActivityFlag, 'boolean'); +}, '[audio-only] RTCRtpSynchronizationSource.voiceActivityFlag is a boolean'); + +</script> |