Title: [266052] trunk
- Revision
- 266052
- Author
- commit-qu...@webkit.org
- Date
- 2020-08-24 02:26:58 -0700 (Mon, 24 Aug 2020)
Log Message
RTCRtpSynchronizationSource.rtpTimestamp is not present
https://bugs.webkit.org/show_bug.cgi?id=215722
Patch by Justin Uberti <jus...@uberti.name> 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):
Modified Paths
Diff
Modified: trunk/LayoutTests/imported/w3c/ChangeLog (266051 => 266052)
--- trunk/LayoutTests/imported/w3c/ChangeLog 2020-08-24 01:51:56 UTC (rev 266051)
+++ trunk/LayoutTests/imported/w3c/ChangeLog 2020-08-24 09:26:58 UTC (rev 266052)
@@ -1,3 +1,14 @@
+2020-08-24 Justin Uberti <jus...@uberti.name>
+
+ 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:
+
2020-08-22 Emilio Cobos Álvarez <emi...@crisal.io>
Import css-content tests.
Modified: trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt (266051 => 266052)
--- trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt 2020-08-24 01:51:56 UTC (rev 266051)
+++ trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt 2020-08-24 09:26:58 UTC (rev 266052)
@@ -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: trunk/Source/WebCore/ChangeLog (266051 => 266052)
--- trunk/Source/WebCore/ChangeLog 2020-08-24 01:51:56 UTC (rev 266051)
+++ trunk/Source/WebCore/ChangeLog 2020-08-24 09:26:58 UTC (rev 266052)
@@ -1,3 +1,19 @@
+2020-08-24 Justin Uberti <jus...@uberti.name>
+
+ 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):
+
2020-08-23 Wenson Hsieh <wenson_hs...@apple.com>
Programmatic selection of text in a text field causes the highlight overlay to spill out
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpContributingSource.h (266051 => 266052)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpContributingSource.h 2020-08-24 01:51:56 UTC (rev 266051)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpContributingSource.h 2020-08-24 09:26:58 UTC (rev 266052)
@@ -31,6 +31,7 @@
struct RTCRtpContributingSource {
double timestamp;
+ unsigned long rtpTimestamp;
unsigned long source;
Optional<double> audioLevel;
};
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpContributingSource.idl (266051 => 266052)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpContributingSource.idl 2020-08-24 01:51:56 UTC (rev 266051)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpContributingSource.idl 2020-08-24 09:26:58 UTC (rev 266052)
@@ -31,6 +31,7 @@
JSGenerateToJSObject,
] dictionary RTCRtpContributingSource {
required DOMHighResTimeStamp timestamp;
+ required unsigned long rtpTimestamp;
required unsigned long source;
double audioLevel;
};
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpSynchronizationSource.idl (266051 => 266052)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpSynchronizationSource.idl 2020-08-24 01:51:56 UTC (rev 266051)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpSynchronizationSource.idl 2020-08-24 09:26:58 UTC (rev 266052)
@@ -28,7 +28,7 @@
[
Conditional=WEB_RTC,
EnabledAtRuntime=PeerConnection,
- JSGenerateToJSObject,
+ JSGenerateToJSObject
] dictionary RTCRtpSynchronizationSource : RTCRtpContributingSource {
boolean voiceActivityFlag;
};
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp (266051 => 266052)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp 2020-08-24 01:51:56 UTC (rev 266051)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp 2020-08-24 09:26:58 UTC (rev 266052)
@@ -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);
_______________________________________________
webkit-changes mailing list
webkit-changes@lists.webkit.org
https://lists.webkit.org/mailman/listinfo/webkit-changes
