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Bug 1344970 - rename mozRtt to roundTripTime r=jib,smaug

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MozReview-Commit-ID: 3kES8JUPd3n

--HG--
extra : rebase_source : e49846845d7cbd43f96d13cb1881e0383517f197
This commit is contained in:
Nico Grunbaum 2017-03-06 15:50:10 -08:00
parent 08d6dac87a
commit 9352ee95aa
9 changed files with 58 additions and 33 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

9
dom/media/tests/mochitest/pc.js
View File

@ -1648,9 +1648,12 @@ PeerConnectionWrapper.prototype = {
ok(rem.bytesReceived <= res.bytesSent, "No more than sent bytes");
}
ok(rem.jitter !== undefined, "Rtcp jitter");
ok(rem.mozRtt !== undefined, "Rtcp rtt");
ok(rem.mozRtt >= 0, "Rtcp rtt " + rem.mozRtt + " >= 0");
ok(rem.mozRtt < 60000, "Rtcp rtt " + rem.mozRtt + " < 1 min");
if (rem.roundTripTime) {
ok(rem.roundTripTime > 0,
"Rtcp rtt " + rem.roundTripTime + " >= 0");
ok(rem.roundTripTime < 60000,
"Rtcp rtt " + rem.roundTripTime + " < 1 min");
}
} else {
ok(rem.type == "outbound-rtp", "Rtcp is outbound");
ok(rem.packetsSent !== undefined, "Rtcp packetsSent");

47
dom/media/tests/mochitest/test_peerConnection_stats.html
View File

@ -14,7 +14,7 @@ var statsExpectedByType = {
"inbound-rtp": {
expected: ["id", "timestamp", "type", "ssrc", "isRemote", "mediaType",
"packetsReceived", "packetsLost", "bytesReceived", "jitter",],
optional: ["mozRtt", "remoteId", "nackCount",],
optional: ["roundTripTime", "remoteId", "nackCount",],
localVideoOnly: ["discardedPackets", "framerateStdDev", "framerateMean",
"bitrateMean", "bitrateStdDev", "firCount", "pliCount",],
unimplemented: ["mediaTrackId", "transportId", "codecId", "framesDecoded",
@ -22,6 +22,7 @@ var statsExpectedByType = {
"sliCount", "qpSum", "packetsRepaired", "fractionLost",
"burstPacketsLost", "burstLossCount", "burstDiscardCount",
"gapDiscardRate", "gapLossRate",],
deprecated: ["mozRtt"],
},
"outbound-rtp": {
expected: ["id", "timestamp", "type", "ssrc", "isRemote", "mediaType",
@ -31,7 +32,8 @@ var statsExpectedByType = {
"framerateMean", "framerateStdDev", "framesEncoded", "firCount",
"pliCount",],
unimplemented: ["mediaTrackId", "transportId", "codecId",
"sliCount", "qpSum", "roundTripTime", "targetBitrate",],
"sliCount", "qpSum", "targetBitrate",],
deprecated: [],
},
"codec": { skip: true },
"peer-connection": { skip: true },
@ -81,6 +83,14 @@ var checkExpectedFields = report => report.forEach(stat => {
ok(!Object.keys(stat).includes(field), "Unimplemented field " + stat.type
+ "." + field + " does not exist.");
});
//
// Ensure that all deprecated fields are not present
//
expectations.deprecated.forEach(field => {
ok(!Object.keys(stat).includes(field), "Deprecated field " + stat.type
+ "." + field + " does not exist.");
});
});
var pedanticChecks = report => {
@ -201,12 +211,13 @@ var pedanticChecks = report => {
// Optional fields
//
// mozRtt
// roundTripTime
if (stat.inner.isRemote) {
ok(stat.mozRtt >= 0, stat.type + ".mozRtt is sane.");
ok(stat.roundTripTime >= 0, stat.type + ".roundTripTime is sane with" +
"value of:" + stat.roundTripTime);
} else {
is(stat.mozRtt, undefined, stat.type
+ ".mozRtt is only set when isRemote is true");
is(stat.roundTripTime, undefined, stat.type
+ ".roundTripTime is only set when isRemote is true");
}
//
@ -362,40 +373,46 @@ var pedanticChecks = report => {
// This MUST be run after PC_*_WAIT_FOR_MEDIA_FLOW to ensure that we have RTP
// before checking for RTCP.
var waitForRtcp = async pc => {
var waitForSyncedRtcp = async pc => {
// Ensures that RTCP is present
let ensureRtcp = async () => pc.getStats().then(stats => {
let ensureSyncedRtcp = async () => {
let stats = await pc.getStats();
for (let [k, v] of stats) {
if (v.type.endsWith("bound-rtp") && !v.remoteId) {
throw new Error(v.id + " is missing remoteId: "
+ JSON.stringify(v));
}
if (v.type == "inbound-rtp" && v.isRemote == true
&& v.roundTripTime === undefined) {
throw new Error(v.id + " is missing roundTripTime: "
+ JSON.stringify(v));
}
}
return stats;
});
}
const waitPeriod = 500;
for (let totalTime = 10000; totalTime > 0; totalTime -= waitPeriod) {
const maxTime = 15000;
for (let totalTime = maxTime; totalTime > 0; totalTime -= waitPeriod) {
try {
return await ensureRtcp();
return await ensureSyncedRtcp();
} catch (e) {
info(e);
await wait(waitPeriod);
}
}
throw new Error("Waiting for RTCP timed out after at least " + totalTime
throw new Error("Waiting for synced RTCP timed out after at least " + maxTime
+ "ms");
}
var PC_LOCAL_TEST_LOCAL_STATS = test => {
return waitForRtcp(test.pcLocal).then(stats => {
return waitForSyncedRtcp(test.pcLocal).then(stats => {
checkExpectedFields(stats);
pedanticChecks(stats);
});
}
var PC_REMOTE_TEST_REMOTE_STATS = test => {
return waitForRtcp(test.pcRemote).then(stats => {
return waitForSyncedRtcp(test.pcRemote).then(stats => {
checkExpectedFields(stats);
pedanticChecks(stats);
});

4
dom/media/webrtc/WebrtcGlobal.h
View File

@ -344,7 +344,7 @@ struct ParamTraits<mozilla::dom::RTCInboundRTPStreamStats>
WriteParam(aMsg, aParam.mJitter);
WriteParam(aMsg, aParam.mMozAvSyncDelay);
WriteParam(aMsg, aParam.mMozJitterBufferDelay);
WriteParam(aMsg, aParam.mMozRtt);
WriteParam(aMsg, aParam.mRoundTripTime);
WriteParam(aMsg, aParam.mPacketsLost);
WriteParam(aMsg, aParam.mPacketsReceived);
WriteRTCRTPStreamStats(aMsg, aParam);
@ -358,7 +358,7 @@ struct ParamTraits<mozilla::dom::RTCInboundRTPStreamStats>
!ReadParam(aMsg, aIter, &(aResult->mJitter)) ||
!ReadParam(aMsg, aIter, &(aResult->mMozAvSyncDelay)) ||
!ReadParam(aMsg, aIter, &(aResult->mMozJitterBufferDelay)) ||
!ReadParam(aMsg, aIter, &(aResult->mMozRtt)) ||
!ReadParam(aMsg, aIter, &(aResult->mRoundTripTime)) ||
!ReadParam(aMsg, aIter, &(aResult->mPacketsLost)) ||
!ReadParam(aMsg, aIter, &(aResult->mPacketsReceived)) ||
!ReadRTCRTPStreamStats(aMsg, aIter, aResult) ||

4
dom/webidl/RTCStatsReport.webidl
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@ -42,7 +42,7 @@ dictionary RTCRTPStreamStats : RTCStats {
// Local only measurements, RTCP related but not communicated via RTCP. Not
// present in RTCP case.
unsigned long firCount;
unsigned long firCount;
unsigned long pliCount;
unsigned long nackCount;
};
@ -54,7 +54,7 @@ dictionary RTCInboundRTPStreamStats : RTCRTPStreamStats {
unsigned long packetsLost;
long mozAvSyncDelay;
long mozJitterBufferDelay;
long mozRtt;
long roundTripTime;
// Video decoder measurement, not present in RTCP case
unsigned long discardedPackets;

3
media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
View File

@ -223,6 +223,9 @@ bool WebrtcAudioConduit::GetRTCPReceiverReport(DOMHighResTimeStamp* timestamp,
fractionLost,
*cumulativeLost,
*rttMs);
// Note: rrtMs is 0 when unavailable before the VoE rework. It is likely
// that after the audio moves to the new Call API that rttMs will be -1
// when unavailable.
if (!result) {
return false;
}

12
media/webrtc/signaling/src/media-conduit/VideoConduit.cpp
View File

@ -876,12 +876,7 @@ bool WebrtcVideoConduit::GetRTCPReceiverReport(DOMHighResTimeStamp* timestamp,
*cumulativeLost = ind->second.rtcp_stats.cumulative_lost;
*bytesReceived = ind->second.rtp_stats.MediaPayloadBytes();
*packetsReceived = ind->second.rtp_stats.transmitted.packets;
int64_t rtt = mSendStream->GetRtt(); // TODO: BUG 1241066, mozRtt is 0 or 1
if (rtt >= 0) {
*rttMs = rtt;
} else {
*rttMs = 0;
}
int64_t rtt = mSendStream->GetRtt();
#ifdef DEBUG
if (rtt > INT32_MAX) {
CSFLogError(logTag,
@ -889,6 +884,11 @@ bool WebrtcVideoConduit::GetRTCPReceiverReport(DOMHighResTimeStamp* timestamp,
" maximum size of an RTCP RTT.", __FUNCTION__, this);
}
#endif
if (rtt > 0) {
*rttMs = rtt;
} else {
*rttMs = 0;
}
// Note: timestamp is not correct per the spec... should be time the rtcp
// was received (remote) or sent (local)
*timestamp = webrtc::Clock::GetRealTimeClock()->TimeInMilliseconds();

4
media/webrtc/signaling/src/peerconnection/PeerConnectionCtx.cpp
View File

@ -267,11 +267,11 @@ EverySecondTelemetryCallback_s(nsAutoPtr<RTCStatsQueries> aQueryList) {
}
Accumulate(id, s.mJitter.Value());
}
if (s.mMozRtt.WasPassed()) {
if (s.mRoundTripTime.WasPassed()) {
MOZ_ASSERT(s.mIsRemote);
HistogramID id = isAudio ? WEBRTC_AUDIO_QUALITY_OUTBOUND_RTT :
WEBRTC_VIDEO_QUALITY_OUTBOUND_RTT;
Accumulate(id, s.mMozRtt.Value());
Accumulate(id, s.mRoundTripTime.Value());
}
if (lastInboundStats && s.mBytesReceived.WasPassed()) {
auto& laststats = *lastInboundStats;

4
media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp
View File

@ -3640,7 +3640,9 @@ PeerConnectionImpl::ExecuteStatsQuery_s(RTCStatsQuery *query) {
s.mPacketsReceived.Construct(packetsReceived);
s.mBytesReceived.Construct(bytesReceived);
s.mPacketsLost.Construct(packetsLost);
s.mMozRtt.Construct(rtt);
if (rtt > 0) {
s.mRoundTripTime.Construct(rtt);
}
query->report->mInboundRTPStreamStats.Value().AppendElement(s,
fallible);
}

4
toolkit/content/aboutwebrtc/aboutWebrtc.js
View File

@ -612,8 +612,8 @@ RTPStats.prototype = {
statsString += ` ${getString("lost_label")}: ${stats.packetsLost} ${getString("jitter_label")}: ${stats.jitter}`;
if (stats.mozRtt) {
statsString += ` RTT: ${stats.mozRtt} ms`;
if (stats.roundTripTime) {
statsString += ` RTT: ${stats.roundTripTime} ms`;
}
} else if (stats.packetsSent) {
statsString += ` ${getString("sent_label")}: ${stats.packetsSent} ${getString("packets")}`;