[webkit-changes] [269518] branches/safari-610.3.7.1-branch

Wed, 11 Nov 2020 07:39:58 -0800

Title: [269518] branches/safari-610.3.7.1-branch
Revision
269518
Author
[email protected]
Date
2020-11-06 10:20:52 -0800 (Fri, 06 Nov 2020)

Log Message

Cherry-pick r266052. rdar://problem/71120601

    RTCRtpSynchronizationSource.rtpTimestamp is not present
    https://bugs.webkit.org/show_bug.cgi?id=215722

    Patch by Justin Uberti <[email protected]> on 2020-08-24
    Reviewed by Youenn Fablet.

    LayoutTests/imported/w3c:

    Updated expectations file to indicate that tests checking for .rtpTimestamp now pass.

    * LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt:

    Source/WebCore:

    Updated expected results in LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt.

    * Modules/mediastream/RTCRtpContributingSource.idl:
    * Modules/mediastream/RTCRtpContributingSource.idl:
    * Modules/mediastream/RTCRtpSynchronizationSource.idl:
    Minor modification to ensure JSRTCRtpSynchronizationSource.cpp gets regenerated.
    * Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp:
    (WebCore::fillRTCRtpContributingSource):

    git-svn-id: https://svn.webkit.org/repository/webkit/trunk@266052 268f45cc-cd09-0410-ab3c-d52691b4dbfc

Modified Paths

Diff

Modified: branches/safari-610.3.7.1-branch/LayoutTests/imported/w3c/ChangeLog (269517 => 269518)


--- branches/safari-610.3.7.1-branch/LayoutTests/imported/w3c/ChangeLog	2020-11-06 17:51:59 UTC (rev 269517)
+++ branches/safari-610.3.7.1-branch/LayoutTests/imported/w3c/ChangeLog	2020-11-06 18:20:52 UTC (rev 269518)
@@ -1,3 +1,43 @@
+2020-11-06  Kocsen Chung  <[email protected]>
+
+        Cherry-pick r266052. rdar://problem/71120601
+
+    RTCRtpSynchronizationSource.rtpTimestamp is not present
+    https://bugs.webkit.org/show_bug.cgi?id=215722
+    
+    Patch by Justin Uberti <[email protected]> on 2020-08-24
+    Reviewed by Youenn Fablet.
+    
+    LayoutTests/imported/w3c:
+    
+    Updated expectations file to indicate that tests checking for .rtpTimestamp now pass.
+    
+    * LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt:
+    
+    Source/WebCore:
+    
+    Updated expected results in LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt.
+    
+    * Modules/mediastream/RTCRtpContributingSource.idl:
+    * Modules/mediastream/RTCRtpContributingSource.idl:
+    * Modules/mediastream/RTCRtpSynchronizationSource.idl:
+    Minor modification to ensure JSRTCRtpSynchronizationSource.cpp gets regenerated.
+    * Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp:
+    (WebCore::fillRTCRtpContributingSource):
+    
+    git-svn-id: https://svn.webkit.org/repository/webkit/trunk@266052 268f45cc-cd09-0410-ab3c-d52691b4dbfc
+
+    2020-08-24  Justin Uberti  <[email protected]>
+
+            RTCRtpSynchronizationSource.rtpTimestamp is not present
+            https://bugs.webkit.org/show_bug.cgi?id=215722
+
+            Reviewed by Youenn Fablet.
+
+            Updated expectations file to indicate that tests checking for .rtpTimestamp now pass.
+
+            * LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt:
+
 2020-10-06  Alan Coon  <[email protected]>
 
         Cherry-pick r267833. rdar://problem/70024626

Modified: branches/safari-610.3.7.1-branch/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt (269517 => 269518)


--- branches/safari-610.3.7.1-branch/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt	2020-11-06 17:51:59 UTC (rev 269517)
+++ branches/safari-610.3.7.1-branch/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt	2020-11-06 18:20:52 UTC (rev 269518)
@@ -1,13 +1,13 @@
 
 PASS [audio] getSynchronizationSources() eventually returns a non-empty list 
 PASS [audio] RTCRtpSynchronizationSource.timestamp is a number 
-FAIL [audio] RTCRtpSynchronizationSource.rtpTimestamp is a number [0, 2^32-1] assert_equals: expected "number" but got "undefined"
+PASS [audio] RTCRtpSynchronizationSource.rtpTimestamp is a number [0, 2^32-1] 
 PASS [audio] getSynchronizationSources() does not contain SSRCs older than 10 seconds 
 FAIL [audio] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now() assert_true: expected true got false
 PASS [audio] RTCRtpSynchronizationSource.source is a number 
 PASS [video] getSynchronizationSources() eventually returns a non-empty list 
 PASS [video] RTCRtpSynchronizationSource.timestamp is a number 
-FAIL [video] RTCRtpSynchronizationSource.rtpTimestamp is a number [0, 2^32-1] assert_equals: expected "number" but got "undefined"
+PASS [video] RTCRtpSynchronizationSource.rtpTimestamp is a number [0, 2^32-1] 
 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

Modified: branches/safari-610.3.7.1-branch/Source/WebCore/ChangeLog (269517 => 269518)


--- branches/safari-610.3.7.1-branch/Source/WebCore/ChangeLog	2020-11-06 17:51:59 UTC (rev 269517)
+++ branches/safari-610.3.7.1-branch/Source/WebCore/ChangeLog	2020-11-06 18:20:52 UTC (rev 269518)
@@ -1,3 +1,48 @@
+2020-11-06  Kocsen Chung  <[email protected]>
+
+        Cherry-pick r266052. rdar://problem/71120601
+
+    RTCRtpSynchronizationSource.rtpTimestamp is not present
+    https://bugs.webkit.org/show_bug.cgi?id=215722
+    
+    Patch by Justin Uberti <[email protected]> on 2020-08-24
+    Reviewed by Youenn Fablet.
+    
+    LayoutTests/imported/w3c:
+    
+    Updated expectations file to indicate that tests checking for .rtpTimestamp now pass.
+    
+    * LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt:
+    
+    Source/WebCore:
+    
+    Updated expected results in LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt.
+    
+    * Modules/mediastream/RTCRtpContributingSource.idl:
+    * Modules/mediastream/RTCRtpContributingSource.idl:
+    * Modules/mediastream/RTCRtpSynchronizationSource.idl:
+    Minor modification to ensure JSRTCRtpSynchronizationSource.cpp gets regenerated.
+    * Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp:
+    (WebCore::fillRTCRtpContributingSource):
+    
+    git-svn-id: https://svn.webkit.org/repository/webkit/trunk@266052 268f45cc-cd09-0410-ab3c-d52691b4dbfc
+
+    2020-08-24  Justin Uberti  <[email protected]>
+
+            RTCRtpSynchronizationSource.rtpTimestamp is not present
+            https://bugs.webkit.org/show_bug.cgi?id=215722
+
+            Reviewed by Youenn Fablet.
+
+            Updated expected results in LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt.
+
+            * Modules/mediastream/RTCRtpContributingSource.idl:
+            * Modules/mediastream/RTCRtpContributingSource.idl:
+            * Modules/mediastream/RTCRtpSynchronizationSource.idl:
+            Minor modification to ensure JSRTCRtpSynchronizationSource.cpp gets regenerated.
+            * Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp:
+            (WebCore::fillRTCRtpContributingSource):
+
 2020-11-05  Alan Coon  <[email protected]>
 
         Cherry-pick r269384. rdar://problem/71084559

Modified: branches/safari-610.3.7.1-branch/Source/WebCore/Modules/mediastream/RTCRtpContributingSource.h (269517 => 269518)


--- branches/safari-610.3.7.1-branch/Source/WebCore/Modules/mediastream/RTCRtpContributingSource.h	2020-11-06 17:51:59 UTC (rev 269517)
+++ branches/safari-610.3.7.1-branch/Source/WebCore/Modules/mediastream/RTCRtpContributingSource.h	2020-11-06 18:20:52 UTC (rev 269518)
@@ -31,6 +31,7 @@
 
 struct RTCRtpContributingSource {
     double timestamp;
+    unsigned long rtpTimestamp;
     unsigned long source;
     Optional<double> audioLevel;
 };

Modified: branches/safari-610.3.7.1-branch/Source/WebCore/Modules/mediastream/RTCRtpContributingSource.idl (269517 => 269518)


--- branches/safari-610.3.7.1-branch/Source/WebCore/Modules/mediastream/RTCRtpContributingSource.idl	2020-11-06 17:51:59 UTC (rev 269517)
+++ branches/safari-610.3.7.1-branch/Source/WebCore/Modules/mediastream/RTCRtpContributingSource.idl	2020-11-06 18:20:52 UTC (rev 269518)
@@ -31,6 +31,7 @@
     JSGenerateToJSObject,
 ] dictionary RTCRtpContributingSource {
     required DOMHighResTimeStamp timestamp;
+    required unsigned long rtpTimestamp;
     required unsigned long source;
     double audioLevel;
 };

Modified: branches/safari-610.3.7.1-branch/Source/WebCore/Modules/mediastream/RTCRtpSynchronizationSource.idl (269517 => 269518)


--- branches/safari-610.3.7.1-branch/Source/WebCore/Modules/mediastream/RTCRtpSynchronizationSource.idl	2020-11-06 17:51:59 UTC (rev 269517)
+++ branches/safari-610.3.7.1-branch/Source/WebCore/Modules/mediastream/RTCRtpSynchronizationSource.idl	2020-11-06 18:20:52 UTC (rev 269518)
@@ -28,7 +28,7 @@
 [
     Conditional=WEB_RTC,
     EnabledAtRuntime=PeerConnection,
-    JSGenerateToJSObject,
+    JSGenerateToJSObject
 ] dictionary RTCRtpSynchronizationSource : RTCRtpContributingSource {
     boolean voiceActivityFlag;
 };

Modified: branches/safari-610.3.7.1-branch/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp (269517 => 269518)


--- branches/safari-610.3.7.1-branch/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp	2020-11-06 17:51:59 UTC (rev 269517)
+++ branches/safari-610.3.7.1-branch/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp	2020-11-06 18:20:52 UTC (rev 269518)
@@ -41,6 +41,7 @@
 static inline void fillRTCRtpContributingSource(RTCRtpContributingSource& source, const webrtc::RtpSource& rtcSource)
 {
     source.timestamp = rtcSource.timestamp_ms();
+    source.rtpTimestamp = rtcSource.rtp_timestamp();
     source.source = rtcSource.source_id();
     if (rtcSource.audio_level())
         source.audioLevel = (*rtcSource.audio_level() == 127) ? 0 : pow(10, -*rtcSource.audio_level() / 20);

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