Populate “fractionLost” stats for remote inbound rtp streams

Tests:
./out/Default/peerconnection_unittests

Manually tested with Chromium to see the data populated

Spec: https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*
Bug: webrtc:12506
Change-Id: I60ef8061fb31deab06ca5f115246ceb5a8cdc5ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208960
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33361}

api/stats/rtcstats_objects.h

@@ -555,8 +555,7 @@ class RTC_EXPORT RTCRemoteInboundRtpStreamStats final : public RTCStats {
   // RTCRemoteInboundRtpStreamStats
   RTCStatsMember<std::string> local_id;
   RTCStatsMember<double> round_trip_time;
-  // TODO(hbos): The following RTCRemoteInboundRtpStreamStats metric should also
+  RTCStatsMember<double> fraction_lost;
-  // be implemented: fractionLost.
 };
 
 // https://w3c.github.io/webrtc-stats/#dom-rtcmediasourcestats

pc/rtc_stats_collector.cc

@@ -541,6 +541,8 @@ ProduceRemoteInboundRtpStreamStatsFromReportBlockData(
   remote_inbound->kind =
       media_type == cricket::MEDIA_TYPE_AUDIO ? "audio" : "video";
   remote_inbound->packets_lost = report_block.packets_lost;
+  remote_inbound->fraction_lost =
+      static_cast<double>(report_block.fraction_lost) / (1 << 8);
   remote_inbound->round_trip_time =
       static_cast<double>(report_block_data.last_rtt_ms()) /
       rtc::kNumMillisecsPerSec;

pc/rtc_stats_collector_unittest.cc

@@ -2678,6 +2678,7 @@ TEST_P(RTCStatsCollectorTestWithParamKind,
   const int64_t kReportBlockTimestampUtcUs = 123456789;
   const int64_t kRoundTripTimeMs = 13000;
   const double kRoundTripTimeSeconds = 13.0;
+  const uint8_t kFractionLost = 12;
 
   // The report block's timestamp cannot be from the future, set the fake clock
   // to match.
@@ -2690,6 +2691,7 @@ TEST_P(RTCStatsCollectorTestWithParamKind,
     // |source_ssrc|, "SSRC of the RTP packet sender".
     report_block.source_ssrc = ssrc;
     report_block.packets_lost = 7;
+    report_block.fraction_lost = kFractionLost;
     ReportBlockData report_block_data;
     report_block_data.SetReportBlock(report_block, kReportBlockTimestampUtcUs);
     report_block_data.AddRoundTripTimeSample(1234);
@@ -2708,6 +2710,8 @@ TEST_P(RTCStatsCollectorTestWithParamKind,
         "RTCRemoteInboundRtp" + MediaTypeUpperCase() + stream_id,
         kReportBlockTimestampUtcUs);
     expected_remote_inbound_rtp.ssrc = ssrc;
+    expected_remote_inbound_rtp.fraction_lost =
+        static_cast<double>(kFractionLost) / (1 << 8);
     expected_remote_inbound_rtp.kind = MediaTypeLowerCase();
     expected_remote_inbound_rtp.transport_id =
         "RTCTransport_TransportName_1";  // 1 for RTP (we have no RTCP

pc/rtc_stats_integrationtest.cc

@@ -1014,6 +1014,7 @@ class RTCStatsReportVerifier {
     verifier.TestMemberIsIDReference(remote_inbound_stream.codec_id,
                                      RTCCodecStats::kType);
     verifier.TestMemberIsDefined(remote_inbound_stream.packets_lost);
+    verifier.TestMemberIsDefined(remote_inbound_stream.fraction_lost);
     // Note that the existance of RTCCodecStats is needed for |codec_id| and
     // |jitter| to be present.
     verifier.TestMemberIsNonNegative<double>(remote_inbound_stream.jitter);

stats/rtcstats_objects.cc

@@ -848,7 +848,8 @@ WEBRTC_RTCSTATS_IMPL(
     &packets_lost,
     &jitter,
     &local_id,
-    &round_trip_time)
+    &round_trip_time,
+    &fraction_lost)
 // clang-format on
 
 RTCRemoteInboundRtpStreamStats::RTCRemoteInboundRtpStreamStats(
@@ -867,7 +868,8 @@ RTCRemoteInboundRtpStreamStats::RTCRemoteInboundRtpStreamStats(
       packets_lost("packetsLost"),
       jitter("jitter"),
       local_id("localId"),
-      round_trip_time("roundTripTime") {}
+      round_trip_time("roundTripTime"),
+      fraction_lost("fractionLost") {}
 
 RTCRemoteInboundRtpStreamStats::RTCRemoteInboundRtpStreamStats(
     const RTCRemoteInboundRtpStreamStats& other)
@@ -879,7 +881,8 @@ RTCRemoteInboundRtpStreamStats::RTCRemoteInboundRtpStreamStats(
       packets_lost(other.packets_lost),
       jitter(other.jitter),
       local_id(other.local_id),
-      round_trip_time(other.round_trip_time) {}
+      round_trip_time(other.round_trip_time),
+      fraction_lost(other.fraction_lost) {}
 
 RTCRemoteInboundRtpStreamStats::~RTCRemoteInboundRtpStreamStats() {}