Diff
Modified: trunk/LayoutTests/ChangeLog (290864 => 290865)
--- trunk/LayoutTests/ChangeLog 2022-03-05 11:18:55 UTC (rev 290864)
+++ trunk/LayoutTests/ChangeLog 2022-03-05 11:33:24 UTC (rev 290865)
@@ -1,3 +1,13 @@
+2022-03-05 Youenn Fablet <you...@apple.com>
+
+ Implement remote-inbound-rtp packetsLost
+ https://bugs.webkit.org/show_bug.cgi?id=237443
+
+ Reviewed by Eric Carlson.
+
+ * platform/mac/TestExpectations:
+ * webrtc/video-stats.html:
+
2022-03-04 Tyler Wilcock <tyle...@apple.com>
AX: [WebAccessibilityObjectWrapperMac AXAttributeStringSetFont] crashes when given a font with a nil postscript name, font family, or display name
Modified: trunk/LayoutTests/platform/mac/TestExpectations (290864 => 290865)
--- trunk/LayoutTests/platform/mac/TestExpectations 2022-03-05 11:18:55 UTC (rev 290864)
+++ trunk/LayoutTests/platform/mac/TestExpectations 2022-03-05 11:33:24 UTC (rev 290865)
@@ -2097,7 +2097,6 @@
[ arm64 ] webrtc/captureCanvas-webrtc-software-h264-high.html [ Pass Failure ]
#These two are failing on intel as well for ews
webrtc/h264-baseline.html [ Failure Timeout ]
-webrtc/video-stats.html [ Pass Failure ]
webkit.org/b/223043 [ BigSur ] webrtc/multi-audio.html [ Pass Failure ]
Modified: trunk/LayoutTests/webrtc/video-stats.html (290864 => 290865)
--- trunk/LayoutTests/webrtc/video-stats.html 2022-03-05 11:18:55 UTC (rev 290864)
+++ trunk/LayoutTests/webrtc/video-stats.html 2022-03-05 11:33:24 UTC (rev 290865)
@@ -130,7 +130,7 @@
assert_not_equals(Object.keys(stats).indexOf("trackId"), -1, "trackId");
return;
}
- if (++count === 20)
+ if (++count === 50)
return Promise.reject("checking inbound stats frame number increasing timed out");
return waitFor(50).then(() => {
return checkInboundFramesNumberIncreased(secondConnection, statsSecondConnection, count)
@@ -148,7 +148,7 @@
assert_not_equals(Object.keys(stats).indexOf("trackId"), -1, "trackId");
return;
}
- if (++count === 20)
+ if (++count === 50)
return Promise.reject("checking outbound stats frame number increasing timed out");
return waitFor(50).then(() => {
return checkOutboundFramesNumberIncreased(firstConnection, statsFirstConnection, count)
@@ -220,6 +220,7 @@
const remoteInboundStats = await getRemoteInboundRTPStats(firstConnection);
assert_true(remoteInboundStats.kind === "audio"|| remoteInboundStats.kind === "video", "kind is present");
+ assert_not_equals(remoteInboundStats.packetsLost, undefined);
}, "Basic video stats");
promise_test(async (test) => {
Modified: trunk/Source/WebCore/ChangeLog (290864 => 290865)
--- trunk/Source/WebCore/ChangeLog 2022-03-05 11:18:55 UTC (rev 290864)
+++ trunk/Source/WebCore/ChangeLog 2022-03-05 11:33:24 UTC (rev 290865)
@@ -1,3 +1,18 @@
+2022-03-05 Youenn Fablet <you...@apple.com>
+
+ Implement remote-inbound-rtp packetsLost
+ https://bugs.webkit.org/show_bug.cgi?id=237443
+
+ Reviewed by Eric Carlson.
+
+ Take benefit of latest backend to expose RemoteInboundRtpStreamStats values inherited from ReceivedRtpStreamStats.
+ Covered by updated test.
+
+ * Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp:
+ (WebCore::fillReceivedRtpStreamStats):
+ (WebCore::fillInboundRtpStreamStats):
+ (WebCore::fillRemoteInboundRtpStreamStats):
+
2022-03-05 Oriol Brufau <obru...@igalia.com>
[css-cascade] Let 'revert-layer' in lowest layer roll back to user styles
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp (290864 => 290865)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp 2022-03-05 11:18:55 UTC (rev 290864)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp 2022-03-05 11:33:24 UTC (rev 290865)
@@ -76,12 +76,10 @@
}
}
-static inline void fillReceivedRtpStreamStats(RTCStatsReport::ReceivedRtpStreamStats& stats, const webrtc::RTCInboundRTPStreamStats& rtcStats)
+static inline void fillReceivedRtpStreamStats(RTCStatsReport::ReceivedRtpStreamStats& stats, const webrtc::RTCReceivedRtpStreamStats& rtcStats)
{
fillRtpStreamStats(stats, rtcStats);
- if (rtcStats.packets_received.is_defined())
- stats.packetsReceived = *rtcStats.packets_received;
if (rtcStats.packets_lost.is_defined())
stats.packetsLost = *rtcStats.packets_lost;
if (rtcStats.jitter.is_defined())
@@ -88,6 +86,16 @@
stats.jitter = *rtcStats.jitter;
if (rtcStats.packets_discarded.is_defined())
stats.packetsDiscarded = *rtcStats.packets_discarded;
+}
+
+static inline void fillInboundRtpStreamStats(RTCStatsReport::InboundRtpStreamStats& stats, const webrtc::RTCInboundRTPStreamStats& rtcStats)
+{
+ fillReceivedRtpStreamStats(stats, rtcStats);
+
+ // receiverId
+ // remoteId
+ if (rtcStats.packets_received.is_defined())
+ stats.packetsReceived = *rtcStats.packets_received;
if (rtcStats.packets_repaired.is_defined())
stats.packetsRepaired = *rtcStats.packets_repaired;
if (rtcStats.burst_packets_lost.is_defined())
@@ -107,14 +115,7 @@
if (rtcStats.gap_discard_rate.is_defined())
stats.gapDiscardRate = *rtcStats.gap_discard_rate;
// Add frames_dropped and full_frames_lost.
-}
-static inline void fillInboundRtpStreamStats(RTCStatsReport::InboundRtpStreamStats& stats, const webrtc::RTCInboundRTPStreamStats& rtcStats)
-{
- fillReceivedRtpStreamStats(stats, rtcStats);
-
- // receiverId
- // remoteId
if (rtcStats.frames_decoded.is_defined())
stats.framesDecoded = *rtcStats.frames_decoded;
if (rtcStats.key_frames_decoded.is_defined())
@@ -188,7 +189,7 @@
static inline void fillRemoteInboundRtpStreamStats(RTCStatsReport::RemoteInboundRtpStreamStats& stats, const webrtc::RTCRemoteInboundRtpStreamStats& rtcStats)
{
- fillRTCStats(stats, rtcStats);
+ fillReceivedRtpStreamStats(stats, rtcStats);
// FIXME: this should be filled in fillRtpStreamStats.
if (rtcStats.ssrc.is_defined())
