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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.
'fecBytesReceived', // 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 that there are no dangling references,
// i.e. keys ending with `Id` as described in
// https://w3c.github.io/webrtc-stats/#guidelines-for-design-of-stats-objects
test(t => {
for (const stat of stats.values()) {
Object.keys(stat).forEach(key => {
if (!key.endsWith('Id')) return;
assert_true(stats.has(stat[key]), `${stat.type}.${key} can be resolved`);
});
}
}, 'All references resolve');
// 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>
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