RTCInboundRtpStreamStats should have a framesDropped property

https://bugs.webkit.org/show_bug.cgi?id=241844
rdar://problem/95669239

Patch by Youenn Fablet <youennf@gmail.com> on 2022-06-23
Reviewed by Eric Carlson.

We should add framesDropped to RTCReceivedRtpStreamStats as per spec but our WebRC backend only supports it in RTCInboundRtpStreamStats.
Let's add this property there since this is used by some applications.

* LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-mandatory-getStats.https.html:
* LayoutTests/webrtc/video-stats.html:
* Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp:
(WebCore::fillReceivedRtpStreamStats):
(WebCore::fillInboundRtpStreamStats):

Canonical link: https://commits.webkit.org/251778@main
git-svn-id: https://svn.webkit.org/repository/webkit/trunk@295773 268f45cc-cd09-0410-ab3c-d52691b4dbfc

LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-mandatory-getStats.https-expected.txt

@@ -9,7 +9,7 @@ PASS RTCReceivedRtpStreamStats's packetsReceived
 PASS RTCReceivedRtpStreamStats's packetsLost
 PASS RTCReceivedRtpStreamStats's jitter
 PASS RTCReceivedRtpStreamStats's packetsDiscarded
-FAIL RTCReceivedRtpStreamStats's framesDropped assert_true: Is framesDropped present expected true got false
+PASS RTCReceivedRtpStreamStats's framesDropped
 FAIL RTCInboundRtpStreamStats's receiverId assert_true: Is receiverId present expected true got false
 FAIL RTCInboundRtpStreamStats's remoteId assert_true: Is remoteId present expected true got false
 PASS RTCInboundRtpStreamStats's framesDecoded

LayoutTests/webrtc/video-stats.html

@@ -122,6 +122,7 @@
 function checkInboundFramesNumberIncreased(secondConnection, statsSecondConnection, count)
 {
     return getInboundRTPStats(secondConnection).then((stats) => {
+        assert_true("framesDropped" in stats);
         assert_true(!!stats.kind);
         if (stats.framesDecoded > statsSecondConnection.framesDecoded) {
             if (testTimestampDifference(stats.timestamp - statsSecondConnection.timestamp, stats.framesDecoded - statsSecondConnection.framesDecoded))

Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp

@@ -80,12 +80,15 @@ static inline void fillReceivedRtpStreamStats(RTCStatsReport::ReceivedRtpStreamS
 {
     fillRtpStreamStats(stats, rtcStats);
 
+    // packetsReceived
     if (rtcStats.packets_lost.is_defined())
         stats.packetsLost = *rtcStats.packets_lost;
     if (rtcStats.jitter.is_defined())
         stats.jitter = *rtcStats.jitter;
+    // FIXME: Remove packets_discarded
     if (rtcStats.packets_discarded.is_defined())
         stats.packetsDiscarded = *rtcStats.packets_discarded;
+    // framesDropped
 }
 
 static inline void fillInboundRtpStreamStats(RTCStatsReport::InboundRtpStreamStats& stats, const webrtc::RTCInboundRTPStreamStats& rtcStats)
@@ -114,7 +117,9 @@ static inline void fillInboundRtpStreamStats(RTCStatsReport::InboundRtpStreamSta
         stats.gapLossRate = *rtcStats.gap_loss_rate;
     if (rtcStats.gap_discard_rate.is_defined())
         stats.gapDiscardRate = *rtcStats.gap_discard_rate;
-    // Add frames_dropped and full_frames_lost.
+    if (rtcStats.frames_dropped.is_defined())
+        stats.framesDropped = *rtcStats.frames_dropped;
+    // full_frames_lost.
 
     if (rtcStats.frames_decoded.is_defined())
         stats.framesDecoded = *rtcStats.frames_decoded;