Title: [244689] trunk
Revision
244689
Author
[email protected]
Date
2019-04-26 09:29:15 -0700 (Fri, 26 Apr 2019)

Log Message

[Mac WK2 iOS Sim] Layout Test imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html is a flaky failure
https://bugs.webkit.org/show_bug.cgi?id=196633
<rdar://problem/49627667>

Reviewed by Alex Christensen.

LayoutTests/imported/w3c:

* web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt:

Source/WebCore:

Use formula defined in https://w3c.github.io/webrtc-stats/#dom-rtcrtpcontributingsourcestats-audiolevel
to compute the audio level from the RTP header information.
Covered by rebased test.

* Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp:
(WebCore::fillRTCRtpContributingSource):

LayoutTests:

* platform/mac-wk2/TestExpectations:

Modified Paths

Diff

Modified: trunk/LayoutTests/ChangeLog (244688 => 244689)


--- trunk/LayoutTests/ChangeLog	2019-04-26 16:25:27 UTC (rev 244688)
+++ trunk/LayoutTests/ChangeLog	2019-04-26 16:29:15 UTC (rev 244689)
@@ -1,3 +1,13 @@
+2019-04-26  Youenn Fablet  <[email protected]>
+
+        [Mac WK2 iOS Sim] Layout Test imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html is a flaky failure
+        https://bugs.webkit.org/show_bug.cgi?id=196633
+        <rdar://problem/49627667>
+
+        Reviewed by Alex Christensen.
+
+        * platform/mac-wk2/TestExpectations:
+
 2019-04-26  Per Arne Vollan  <[email protected]>
 
         Layout Test fast/harness/render-tree-as-text-options.html is failing

Modified: trunk/LayoutTests/imported/w3c/ChangeLog (244688 => 244689)


--- trunk/LayoutTests/imported/w3c/ChangeLog	2019-04-26 16:25:27 UTC (rev 244688)
+++ trunk/LayoutTests/imported/w3c/ChangeLog	2019-04-26 16:29:15 UTC (rev 244689)
@@ -1,3 +1,13 @@
+2019-04-26  Youenn Fablet  <[email protected]>
+
+        [Mac WK2 iOS Sim] Layout Test imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html is a flaky failure
+        https://bugs.webkit.org/show_bug.cgi?id=196633
+        <rdar://problem/49627667>
+
+        Reviewed by Alex Christensen.
+
+        * web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt:
+
 2019-04-22  Youenn Fablet  <[email protected]>
 
         Cache API should return Abort error in case of putting an aborted fetch

Modified: trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt (244688 => 244689)


--- trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt	2019-04-26 16:25:27 UTC (rev 244688)
+++ trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt	2019-04-26 16:29:15 UTC (rev 244689)
@@ -9,6 +9,6 @@
 PASS [video] getSynchronizationSources() does not contain SSRCs older than 10 seconds 
 FAIL [video] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now() assert_true: expected true got false
 PASS [video] RTCRtpSynchronizationSource.source is a number 
-FAIL [audio-only] RTCRtpSynchronizationSource.audioLevel is a number [0, 1] assert_less_than_equal: expected a number less than or equal to 1 but got 23
+PASS [audio-only] RTCRtpSynchronizationSource.audioLevel is a number [0, 1] 
 FAIL [audio-only] RTCRtpSynchronizationSource.voiceActivityFlag is a boolean assert_equals: expected "boolean" but got "undefined"
 

Modified: trunk/LayoutTests/platform/mac-wk2/TestExpectations (244688 => 244689)


--- trunk/LayoutTests/platform/mac-wk2/TestExpectations	2019-04-26 16:25:27 UTC (rev 244688)
+++ trunk/LayoutTests/platform/mac-wk2/TestExpectations	2019-04-26 16:29:15 UTC (rev 244689)
@@ -938,8 +938,6 @@
 
 webkit.org/b/194164 http/tests/storageAccess/request-and-grant-access-cross-origin-sandboxed-iframe-from-prevalent-domain-with-user-interaction-but-access-from-wrong-frame.html [ Pass Timeout ]
 
-webkit.org/b/196633 imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html [ Pass Failure ]
-
 webkit.org/b/195961 [ Debug ] imported/w3c/IndexedDB-private-browsing/idbobjectstore_createIndex7-event_order.html [ Pass Failure ]
 webkit.org/b/195961 [ Debug ] imported/w3c/IndexedDB-private-browsing/idbobjectstore_createIndex6-event_order.html [ Pass Failure ]
 

Modified: trunk/Source/WebCore/ChangeLog (244688 => 244689)


--- trunk/Source/WebCore/ChangeLog	2019-04-26 16:25:27 UTC (rev 244688)
+++ trunk/Source/WebCore/ChangeLog	2019-04-26 16:29:15 UTC (rev 244689)
@@ -1,3 +1,18 @@
+2019-04-26  Youenn Fablet  <[email protected]>
+
+        [Mac WK2 iOS Sim] Layout Test imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html is a flaky failure
+        https://bugs.webkit.org/show_bug.cgi?id=196633
+        <rdar://problem/49627667>
+
+        Reviewed by Alex Christensen.
+
+        Use formula defined in https://w3c.github.io/webrtc-stats/#dom-rtcrtpcontributingsourcestats-audiolevel
+        to compute the audio level from the RTP header information.
+        Covered by rebased test.
+
+        * Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp:
+        (WebCore::fillRTCRtpContributingSource):
+
 2019-04-26  Sihui Liu  <[email protected]>
 
         Stop IDB transactions to release locked database files when network process is ready to suspend

Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp (244688 => 244689)


--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp	2019-04-26 16:25:27 UTC (rev 244688)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp	2019-04-26 16:29:15 UTC (rev 244689)
@@ -41,7 +41,7 @@
     source.timestamp = rtcSource.timestamp_ms();
     source.source = rtcSource.source_id();
     if (rtcSource.audio_level())
-        source.audioLevel = *rtcSource.audio_level();
+        source.audioLevel = (*rtcSource.audio_level() == 127) ? 0 : pow(10, -*rtcSource.audio_level() / 20);
 }
 
 static inline RTCRtpContributingSource toRTCRtpContributingSource(const webrtc::RtpSource& rtcSource)
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