Bug 1759931 - Implement some missing outbound-rtp stats. r=jib,emilio,ng

Differential Revision: https://phabricator.services.mozilla.com/D141287

dom/media/webrtc/jsapi/PeerConnectionImpl.cpp

@@ -2717,14 +2717,17 @@ nsTArray<RefPtr<dom::RTCStatsPromise>> PeerConnectionImpl::GetSenderStats(
             local.mBytesSent.Construct(audioStats->payload_bytes_sent);
             local.mNackCount.Construct(
                 audioStats->rtcp_packet_type_counts.nack_packets);
-            /*
-             * Potential new stats that are now available upstream.
             local.mHeaderBytesSent.Construct(
                 audioStats->header_and_padding_bytes_sent);
             local.mRetransmittedPacketsSent.Construct(
                 audioStats->retransmitted_packets_sent);
             local.mRetransmittedBytesSent.Construct(
                 audioStats->retransmitted_bytes_sent);
+            /*
+             * Potential new stats that are now available upstream.
+             * Note: when we last tried exposing this we were getting
+             * targetBitrate for audio was ending up as 0. We did not
+             * investigate why.
             local.mTargetBitrate.Construct(audioStats->target_bitrate_bps);
              */
             if (!report->mOutboundRtpStreamStats.AppendElement(std::move(local),
@@ -2805,8 +2808,6 @@ nsTArray<RefPtr<dom::RTCStatsPromise>> PeerConnectionImpl::GetSenderStats(
             if (streamStats->qp_sum) {
               local.mQpSum.Construct(*streamStats->qp_sum);
             }
-            /*
-             * Potential new stats that are now available upstream.
             local.mHeaderBytesSent.Construct(
                 streamStats->rtp_stats.transmitted.header_bytes +
                 streamStats->rtp_stats.transmitted.padding_bytes);
@@ -2814,6 +2815,8 @@ nsTArray<RefPtr<dom::RTCStatsPromise>> PeerConnectionImpl::GetSenderStats(
                 streamStats->rtp_stats.retransmitted.packets);
             local.mRetransmittedBytesSent.Construct(
                 streamStats->rtp_stats.retransmitted.payload_bytes);
+            /*
+             * Potential new stats that are now available upstream.
             local.mTargetBitrate.Construct(videoStats->target_media_bitrate_bps);
             local.mTotalEncodedBytesTarget.Construct(
                 videoStats->total_encoded_bytes_target);

dom/media/webrtc/tests/mochitests/stats.js

@@ -56,6 +56,9 @@ const statsExpectedByType = {
       "packetsSent",
       "bytesSent",
       "remoteId",
+      "headerBytesSent",
+      "retransmittedPacketsSent",
+      "retransmittedBytesSent",
     ],
     optional: ["nackCount", "qpSum"],
     localVideoOnly: ["framesEncoded", "firCount", "pliCount"],
@@ -567,6 +570,32 @@ function pedanticChecks(report) {
           `${stat.kind} test. value=${stat.bytesSent}`
       );
 
+      // headerBytesSent
+      ok(
+        stat.headerBytesSent > 0 &&
+          stat.headerBytesSent < (stat.kind == "video" ? video1Min : audio1Min),
+        `${stat.type}.headerBytesSent is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.headerBytesSent}`
+      );
+
+      // retransmittedPacketsSent
+      ok(
+        stat.retransmittedPacketsSent >= 0 &&
+          stat.retransmittedPacketsSent <
+            (stat.kind == "video" ? video1Min : audio1Min),
+        `${stat.type}.retransmittedPacketsSent is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.retransmittedPacketsSent}`
+      );
+
+      // retransmittedBytesSent
+      ok(
+        stat.retransmittedBytesSent >= 0 &&
+          stat.retransmittedBytesSent <
+            (stat.kind == "video" ? video1Min : audio1Min),
+        `${stat.type}.retransmittedBytesSent is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.retransmittedBytesSent}`
+      );
+
       //
       // Optional fields
       //

dom/webidl/RTCStatsReport.webidl

@@ -87,6 +87,9 @@ dictionary RTCOutboundRtpStreamStats : RTCSentRtpStreamStats {
   unsigned long nackCount;
   unsigned long firCount;
   unsigned long pliCount;
+  unsigned long long headerBytesSent;
+  unsigned long long retransmittedPacketsSent;
+  unsigned long long retransmittedBytesSent;
 };
 
 dictionary RTCRemoteOutboundRtpStreamStats : RTCSentRtpStreamStats {

testing/web-platform/meta/webrtc-stats/supported-stats.html.ini

@@ -172,9 +172,6 @@
   [outbound-rtp's lastPacketSentTimestamp]
     expected: FAIL
 
-  [outbound-rtp's headerBytesSent]
-    expected: FAIL
-
   [outbound-rtp's packetsDiscardedOnSend]
     expected: FAIL
 
@@ -184,12 +181,6 @@
   [outbound-rtp's fecPacketsSent]
     expected: FAIL
 
-  [outbound-rtp's retransmittedPacketsSent]
-    expected: FAIL
-
-  [outbound-rtp's retransmittedBytesSent]
-    expected: FAIL
-
   [outbound-rtp's targetBitrate]
     expected: FAIL