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| author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
|---|---|---|
| committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
| commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
| tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /testing/web-platform/tests/webrtc-stats/rtp-stats-creation.html | |
| parent | Initial commit. (diff) | |
| download | firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip | |
Adding upstream version 115.7.0esr.upstream/115.7.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'testing/web-platform/tests/webrtc-stats/rtp-stats-creation.html')
| -rw-r--r-- | testing/web-platform/tests/webrtc-stats/rtp-stats-creation.html | 110 |
1 files changed, 110 insertions, 0 deletions
diff --git a/testing/web-platform/tests/webrtc-stats/rtp-stats-creation.html b/testing/web-platform/tests/webrtc-stats/rtp-stats-creation.html new file mode 100644 index 0000000000..7a6d9df456 --- /dev/null +++ b/testing/web-platform/tests/webrtc-stats/rtp-stats-creation.html @@ -0,0 +1,110 @@ +<!doctype html> +<meta charset=utf-8> +<title>No RTCRtpStreamStats should exist prior to RTP/RTCP packet flow</title> +<script src="/resources/testharness.js"></script> +<script src="/resources/testharnessreport.js"></script> +<script src="../webrtc/RTCPeerConnection-helper.js"></script> +<script> +'use strict'; + +promise_test(async (test) => { + const localPc = createPeerConnectionWithCleanup(test); + const remotePc = createPeerConnectionWithCleanup(test); + + localPc.addTransceiver("audio"); + localPc.addTransceiver("video"); + await exchangeOfferAndListenToOntrack(test, localPc, remotePc); + const report = await remotePc.getStats(); + const rtp = [...report.values()].filter(({type}) => type.endsWith("rtp")); + assert_equals(rtp.length, 0, "no rtp stats with only remote description"); +}, "No RTCRtpStreamStats exist when only remote description is set"); + +promise_test(async (test) => { + const localPc = createPeerConnectionWithCleanup(test); + const remotePc = createPeerConnectionWithCleanup(test); + + localPc.addTrack(...await createTrackAndStreamWithCleanup(test, "audio")); + localPc.addTrack(...await createTrackAndStreamWithCleanup(test, "video")); + await exchangeOfferAndListenToOntrack(test, localPc, remotePc); + const report = await localPc.getStats(); + const rtp = [...report.values()].filter(({type}) => type.endsWith("rtp")); + assert_equals(rtp.length, 0, "no rtp stats with only local description"); +}, "No RTCRtpStreamStats exist when only local description is set"); + +promise_test(async (test) => { + const localPc = createPeerConnectionWithCleanup(test); + const remotePc = createPeerConnectionWithCleanup(test); + + localPc.addTrack(...await createTrackAndStreamWithCleanup(test, "audio")); + localPc.addTrack(...await createTrackAndStreamWithCleanup(test, "video")); + exchangeIceCandidates(localPc, remotePc); + await Promise.all([ + exchangeOfferAnswer(localPc, remotePc), + new Promise(r => remotePc.ontrack = e => e.track.onunmute = r) + ]); + const start = performance.now(); + while (true) { + const report = await localPc.getStats(); + const outbound = + [...report.values()].filter(({type}) => type == "outbound-rtp"); + assert_true(outbound.every(({packetsSent}) => packetsSent > 0), + "no outbound rtp stats before packets sent"); + if (outbound.length == 2) { + // One outbound stat for each track is present. We're done. + break; + } + if (performance.now() > start + 5000) { + assert_unreached("outbound stats should become available"); + } + await new Promise(r => test.step_timeout(r, 100)); + } +}, "No RTCOutboundRtpStreamStats exist until packets have been sent"); + +promise_test(async (test) => { + const localPc = createPeerConnectionWithCleanup(test); + const remotePc = createPeerConnectionWithCleanup(test); + + localPc.addTrack(...await createTrackAndStreamWithCleanup(test, "audio")); + localPc.addTrack(...await createTrackAndStreamWithCleanup(test, "video")); + exchangeIceCandidates(localPc, remotePc); + await exchangeOfferAnswer(localPc, remotePc); + const start = performance.now(); + while (true) { + const report = await remotePc.getStats(); + const inbound = + [...report.values()].filter(({type}) => type == "inbound-rtp"); + assert_true(inbound.every(({packetsReceived}) => packetsReceived > 0), + "no inbound rtp stats before packets received"); + if (inbound.length == 2) { + // One inbound stat for each track is present. We're done. + break; + } + if (performance.now() > start + 5000) { + assert_unreached("inbound stats should become available"); + } + await new Promise(r => test.step_timeout(r, 100)); + } +}, "No RTCInboundRtpStreamStats exist until packets have been received"); + +promise_test(async (test) => { + const localPc = createPeerConnectionWithCleanup(test); + const remotePc = createPeerConnectionWithCleanup(test); + + localPc.addTrack(...await createTrackAndStreamWithCleanup(test, "audio")); + exchangeIceCandidates(localPc, remotePc); + await exchangeOfferAnswer(localPc, remotePc); + const start = performance.now(); + while (true) { + const report = await remotePc.getStats(); + const audioPlayout = + [...report.values()].filter(({type}) => type == "media-playout"); + if (audioPlayout.length == 1) { + break; + } + if (performance.now() > start + 5000) { + assert_unreached("Audio playout stats should become available"); + } + await new Promise(r => test.step_timeout(r, 100)); + } +}, "RTCAudioPlayoutStats should be present"); +</script> |