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: http://svn.webkit.org/repository/webkit/trunk@295773 268f45cc-cd09-0410-ab3c-d52691b4dbfc
diff --git a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-mandatory-getStats.https-expected.txt b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-mandatory-getStats.https-expected.txt
index 0a8ae96..067d58c 100644
--- a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-mandatory-getStats.https-expected.txt
+++ b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-mandatory-getStats.https-expected.txt
@@ -9,7 +9,7 @@
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
diff --git a/LayoutTests/webrtc/video-stats.html b/LayoutTests/webrtc/video-stats.html
index c79bec8..a2604ce 100644
--- a/LayoutTests/webrtc/video-stats.html
+++ b/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))
diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp
index 5466115..02c1447 100644
--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp
+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp
@@ -80,12 +80,15 @@
{
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 @@
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;
