Bug 1526512 - remote-inbound-rtp stat roundTripTime should be express…

…ed in seconds r=jib,smaug Changing the units roundTripTime is reported in from milliseconds to seconds Differential Revision: https://phabricator.services.mozilla.com/D19544 --HG-- extra : moz-landing-system : lando
dothq · Feb 12, 2019 · c838f88 · c838f88
1 parent e4b8830
commit c838f88
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Showing 5 changed files with 7 additions and 7 deletions.
diff --git a/dom/media/tests/mochitest/pc.js b/dom/media/tests/mochitest/pc.js
@@ -1854,9 +1854,9 @@ PeerConnectionWrapper.prototype = {
               }
               ok(rem.jitter !== undefined, "Rtcp jitter");
               if (rem.roundTripTime) {
-                ok(rem.roundTripTime > 0,
+                ok(rem.roundTripTime >= 0,
                    "Rtcp rtt " + rem.roundTripTime + " >= 0");
-                ok(rem.roundTripTime < 60000,
+                ok(rem.roundTripTime < 60,
                    "Rtcp rtt " + rem.roundTripTime + " < 1 min");
               }
             } else {

diff --git a/dom/webidl/RTCStatsReport.webidl b/dom/webidl/RTCStatsReport.webidl
@@ -57,7 +57,7 @@ dictionary RTCInboundRTPStreamStats : RTCRtpStreamStats {
   unsigned long long bytesReceived;
   double jitter;
   unsigned long packetsLost;
-  long roundTripTime;
+  double roundTripTime;
 
   // Video decoder measurement, not present in RTCP case
   unsigned long discardedPackets;

diff --git a/media/webrtc/signaling/src/peerconnection/PeerConnectionCtx.cpp b/media/webrtc/signaling/src/peerconnection/PeerConnectionCtx.cpp
@@ -239,7 +239,7 @@ void PeerConnectionCtx::DeliverStats(RTCStatsQuery& aQuery) {
         MOZ_ASSERT(isRemote);
         HistogramID id = isAudio ? WEBRTC_AUDIO_QUALITY_OUTBOUND_RTT
                                  : WEBRTC_VIDEO_QUALITY_OUTBOUND_RTT;
-        Accumulate(id, s.mRoundTripTime.Value());
+        Accumulate(id, s.mRoundTripTime.Value() * 1000);
       }
       if (lastReport && lastReport->mInboundRTPStreamStats.WasPassed() &&
           s.mBytesReceived.WasPassed()) {

diff --git a/media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp b/media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp
@@ -2842,8 +2842,8 @@ RefPtr<RTCStatsQueryPromise> PeerConnectionImpl::ExecuteStatsQuery_s(
             s.mPacketsReceived.Construct(packetsReceived);
             s.mBytesReceived.Construct(bytesReceived);
             s.mPacketsLost.Construct(packetsLost);
-            if (rtt > 0) {
-              s.mRoundTripTime.Construct(rtt);
+            if (rtt > 0) { // RTT is not reported when it is zero
+              s.mRoundTripTime.Construct(static_cast<double>(rtt) / 1000);
             }
             query->report->mInboundRTPStreamStats.Value().AppendElement(
                 s, fallible);

diff --git a/toolkit/content/aboutwebrtc/aboutWebrtc.js b/toolkit/content/aboutwebrtc/aboutWebrtc.js
@@ -564,7 +564,7 @@ RTPStats.prototype = {
       statsString += ` ${getString("lost_label")}: ${stats.packetsLost} ${getString("jitter_label")}: ${stats.jitter}`;
 
       if (stats.roundTripTime) {
-        statsString += ` RTT: ${stats.roundTripTime} ms`;
+        statsString += ` RTT: ${stats.roundTripTime * 1000} ms`;
       }
     } else if (stats.packetsSent) {
       statsString += ` ${getString("sent_label")}: ${stats.packetsSent} ${getString("packets")}`;