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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /testing/web-platform/tests/webrtc-stats/supported-stats.https.html | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'testing/web-platform/tests/webrtc-stats/supported-stats.https.html')
-rw-r--r-- | testing/web-platform/tests/webrtc-stats/supported-stats.https.html | 213 |
1 files changed, 213 insertions, 0 deletions
diff --git a/testing/web-platform/tests/webrtc-stats/supported-stats.https.html b/testing/web-platform/tests/webrtc-stats/supported-stats.https.html new file mode 100644 index 0000000000..b886778e51 --- /dev/null +++ b/testing/web-platform/tests/webrtc-stats/supported-stats.https.html @@ -0,0 +1,213 @@ +<!doctype html> +<meta charset=utf-8> +<meta name="timeout" content="long"> +<title>Support for all stats defined in WebRTC Stats</title> +<script src=/resources/testharness.js></script> +<script src=/resources/testharnessreport.js></script> +<script src="../webrtc/RTCPeerConnection-helper.js"></script> +<script src="/resources/WebIDLParser.js"></script> +<script> +'use strict'; + +// inspired from similar test for MTI stats in ../webrtc/RTCPeerConnection-mandatory-getStats.https.html + +// From https://w3c.github.io/webrtc-stats/webrtc-stats.html#rtcstatstype-str* +const dictionaryNames = { + "codec": "RTCCodecStats", + "inbound-rtp": "RTCInboundRtpStreamStats", + "outbound-rtp": "RTCOutboundRtpStreamStats", + "remote-inbound-rtp": "RTCRemoteInboundRtpStreamStats", + "remote-outbound-rtp": "RTCRemoteOutboundRtpStreamStats", + "csrc": "RTCRtpContributingSourceStats", + "peer-connection": "RTCPeerConnectionStats", + "data-channel": "RTCDataChannelStats", + "media-source": { + audio: "RTCAudioSourceStats", + video: "RTCVideoSourceStats" + }, + "media-playout": "RTCAudioPlayoutStats", + "sender": { + audio: "RTCAudioSenderStats", + video: "RTCVideoSenderStats" + }, + "receiver": { + audio: "RTCAudioReceiverStats", + video: "RTCVideoReceiverStats", + }, + "transport": "RTCTransportStats", + "candidate-pair": "RTCIceCandidatePairStats", + "local-candidate": "RTCIceCandidateStats", + "remote-candidate": "RTCIceCandidateStats", + "certificate": "RTCCertificateStats", +}; + +function isPropertyTestable(type, property) { + // List of properties which are not testable by this test. + // When adding something to this list, please explain why. + const untestablePropertiesByType = { + 'candidate-pair': [ + 'availableIncomingBitrate', // requires REMB, no TWCC. + ], + 'certificate': [ + 'issuerCertificateId', // we only use self-signed certificates. + ], + 'local-candidate': [ + 'url', // requires a STUN/TURN server. + 'relayProtocol', // requires a TURN server. + 'relatedAddress', // requires a STUN/TURN server. + 'relatedPort', // requires a STUN/TURN server. + ], + 'remote-candidate': [ + 'url', // requires a STUN/TURN server. + 'relayProtocol', // requires a TURN server. + 'relatedAddress', // requires a STUN/TURN server. + 'relatedPort', // requires a STUN/TURN server. + 'tcpType', // requires ICE-TCP connection. + ], + 'outbound-rtp': [ + 'rid', // requires simulcast. + ], + 'inbound-rtp': [ + 'fecSsrc', // requires FlexFEC to be negotiated. + ], + 'media-source': [ + 'echoReturnLoss', // requires gUM with an audio input device. + 'echoReturnLossEnhancement', // requires gUM with an audio input device. + ] + }; + if (!untestablePropertiesByType[type]) { + return true; + } + return !untestablePropertiesByType[type].includes(property); +} + +async function getAllStats(t, pc) { + // Try to obtain as many stats as possible, waiting up to 20 seconds for + // roundTripTime which can take several RTCP messages to calculate. + let stats; + for (let i = 0; i < 20; i++) { + stats = await pc.getStats(); + const values = [...stats.values()]; + const [remoteInboundAudio, remoteInboundVideo] = + ["audio", "video"].map(kind => + values.find(s => s.type == "remote-inbound-rtp" && s.kind == kind)); + const [remoteOutboundAudio, remoteOutboundVideo] = + ["audio", "video"].map(kind => + values.find(s => s.type == "remote-outbound-rtp" && s.kind == kind)); + // We expect both audio and video remote-inbound-rtp RTT. + const hasRemoteInbound = + remoteInboundAudio && "roundTripTime" in remoteInboundAudio && + remoteInboundVideo && "roundTripTime" in remoteInboundVideo; + // Due to current implementation limitations, we don't put as hard + // requirements on remote-outbound-rtp as remote-inbound-rtp. It's enough if + // it is available for either kind and `roundTripTime` is not required. In + // Chromium, remote-outbound-rtp is only implemented for audio and + // `roundTripTime` is missing in this test, but awaiting for any + // remote-outbound-rtp avoids flaky failures. + const hasRemoteOutbound = remoteOutboundAudio || remoteOutboundVideo; + const hasMediaPlayout = values.find(({type}) => type == "media-playout") != undefined; + if (hasRemoteInbound && hasRemoteOutbound && hasMediaPlayout) { + return stats; + } + await new Promise(r => t.step_timeout(r, 1000)); + } + return stats; +} + +promise_test(async t => { + // load the IDL to know which members to be looking for + const idl = await fetch("/interfaces/webrtc-stats.idl").then(r => r.text()); + // for RTCStats definition + const webrtcIdl = await fetch("/interfaces/webrtc.idl").then(r => r.text()); + const astArray = WebIDL2.parse(idl + webrtcIdl); + + let all = {}; + for (let type in dictionaryNames) { + // TODO: make use of audio/video distinction + let dictionaries = dictionaryNames[type].audio ? Object.values(dictionaryNames[type]) : [dictionaryNames[type]]; + all[type] = []; + let i = 0; + // Recursively collect members from inherited dictionaries + while (i < dictionaries.length) { + const dictName = dictionaries[i]; + const dict = astArray.find(i => i.name === dictName && i.type === "dictionary"); + if (dict && dict.members) { + all[type] = all[type].concat(dict.members.map(m => m.name)); + if (dict.inheritance) { + dictionaries.push(dict.inheritance); + } + } + i++; + } + // Unique-ify + all[type] = [...new Set(all[type])]; + } + + const remaining = JSON.parse(JSON.stringify(all)); + for (const type in remaining) { + remaining[type] = new Set(remaining[type]); + } + + const pc1 = new RTCPeerConnection(); + t.add_cleanup(() => pc1.close()); + const pc2 = new RTCPeerConnection(); + t.add_cleanup(() => pc2.close()); + + const dc1 = pc1.createDataChannel("dummy", {negotiated: true, id: 0}); + const dc2 = pc2.createDataChannel("dummy", {negotiated: true, id: 0}); + // Use a real gUM to ensure that all stats exposing hardware capabilities are + // also exposed. + const stream = await navigator.mediaDevices.getUserMedia( + {video: true, audio: true}); + for (const track of stream.getTracks()) { + pc1.addTrack(track, stream); + pc2.addTrack(track, stream); + t.add_cleanup(() => track.stop()); + } + + // Do a non-trickle ICE handshake to ensure that TCP candidates are gathered. + await pc1.setLocalDescription(); + await waitForIceGatheringState(pc1, ['complete']); + await pc2.setRemoteDescription(pc1.localDescription); + await pc2.setLocalDescription(); + await waitForIceGatheringState(pc2, ['complete']); + await pc1.setRemoteDescription(pc2.localDescription); + // Await the DTLS handshake. + await Promise.all([ + listenToConnected(pc1), + listenToConnected(pc2), + ]); + const stats = await getAllStats(t, pc1); + + // The focus of this test is not API correctness, but rather to provide an + // accessible metric of implementation progress by dictionary member. We count + // whether we've seen each dictionary's members in getStats(). + + test(t => { + for (const stat of stats.values()) { + if (all[stat.type]) { + const memberNames = all[stat.type]; + const remainingNames = remaining[stat.type]; + assert_true(memberNames.length > 0, "Test error. No member found."); + for (const memberName of memberNames) { + if (memberName in stat) { + assert_not_equals(stat[memberName], undefined, "Not undefined"); + remainingNames.delete(memberName); + } + } + } + } + }, "Validating stats"); + + for (const type in all) { + for (const memberName of all[type]) { + test(t => { + assert_implements_optional(isPropertyTestable(type, memberName), + `${type}.${memberName} marked as not testable.`); + assert_true(!remaining[type].has(memberName), + `Is ${memberName} present`); + }, `${type}'s ${memberName}`); + } + } +}, 'getStats succeeds'); +</script> |