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+<!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>