Branch: refs/heads/webkitglib/2.46
Home: https://github.com/WebKit/WebKit
Commit: 6ca154e7038f7a1e43121bbc8c4123843d36acc7
https://github.com/WebKit/WebKit/commit/6ca154e7038f7a1e43121bbc8c4123843d36acc7
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
A LayoutTests/platform/glib/http/wpt/mediarecorder/mimeType-expected.txt
M Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.cpp
M Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.h
M Source/WebCore/platform/mediarecorder/MediaRecorderPrivateGStreamer.cpp
M Source/WebCore/platform/mediarecorder/MediaRecorderPrivateGStreamer.h
Log Message:
-----------
Cherry-pick 286080@main (da9240878292).
https://bugs.webkit.org/show_bug.cgi?id=281295
[GStreamer] MediaRecorder enhancements
https://bugs.webkit.org/show_bug.cgi?id=281295
Reviewed by Xabier Rodriguez-Calvar.
This patch adds WebM (for GStreamer versions >= 1.24.9) and audio bitrate
configuration support to
the GStreamer MediaRecorder backend. As we also handle vorbis encoding, add
platform baselines for
the mimeType test which expects that codec to not be supported.
* LayoutTests/platform/glib/TestExpectations:
*
LayoutTests/platform/glib/http/wpt/mediarecorder/MediaRecorder-first-frame-expected.txt:
Added.
* LayoutTests/platform/glib/http/wpt/mediarecorder/mimeType-expected.txt:
Added.
* Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.cpp:
(WebCore::GStreamerRegistryScanner::isCodecSupported const):
* Source/WebCore/platform/mediarecorder/MediaRecorderPrivateGStreamer.cpp:
(WebCore::MediaRecorderPrivateBackend::MediaRecorderPrivateBackend):
(WebCore::MediaRecorderPrivateBackend::fetchData):
(WebCore::MediaRecorderPrivateBackend::containerProfile):
(WebCore::MediaRecorderPrivateBackend::configureAudioEncoder):
(WebCore::MediaRecorderPrivateBackend::preparePipeline):
(WebCore::MediaRecorderPrivateBackend::mimeType const): Deleted.
* Source/WebCore/platform/mediarecorder/MediaRecorderPrivateGStreamer.h:
(WebCore::MediaRecorderPrivateBackend::mimeType const):
Canonical link: https://commits.webkit.org/286080@main
Canonical link: https://commits.webkit.org/282416.296@webkitglib/2.46
Commit: b56c2a0c578029ccf93b325cf5ede884c1189ce0
https://github.com/WebKit/WebKit/commit/b56c2a0c578029ccf93b325cf5ede884c1189ce0
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M LayoutTests/platform/glib/TestExpectations
M LayoutTests/platform/gtk/TestExpectations
M LayoutTests/platform/wpe/TestExpectations
M
Source/WebCore/platform/mediastream/gstreamer/GStreamerIncomingTrackProcessor.cpp
Log Message:
-----------
Cherry-pick 286322@main (55837daba8e1).
https://bugs.webkit.org/show_bug.cgi?id=276587
[GStreamer][WebRTC] webrtc/video-av1.html is flaky
https://bugs.webkit.org/show_bug.cgi?id=276587
<rdar://problem/131697366>
Reviewed by Xabier Rodriguez-Calvar.
The test was consistently timing out since 281394@main, due to two issues:
- After disabling the local track, the buffers on the remote track no
longer had video metas,
because the allocation query on the incoming track was no longer active.
- The frames-decoded stats were no set in case there was no dropped frame
on the video sink.
* LayoutTests/platform/gtk/TestExpectations:
* LayoutTests/platform/wpe/TestExpectations:
*
Source/WebCore/platform/mediastream/gstreamer/GStreamerIncomingTrackProcessor.cpp:
(WebCore::GStreamerIncomingTrackProcessor::configure):
(WebCore::GStreamerIncomingTrackProcessor::stats):
Canonical link: https://commits.webkit.org/286322@main
Canonical link: https://commits.webkit.org/282416.297@webkitglib/2.46
Commit: f9abcba36ec2ef2a94acdf454320327d82aef00b
https://github.com/WebKit/WebKit/commit/f9abcba36ec2ef2a94acdf454320327d82aef00b
Author: Théo Maillart <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M
Source/WebCore/platform/graphics/gstreamer/mse/WebKitMediaSourceGStreamer.cpp
Log Message:
-----------
Cherry-pick 286347@main (9e9ea966373d).
https://bugs.webkit.org/show_bug.cgi?id=282749
[GStreamer] Video dimensions are wrong since GStreamer 1.24.9
https://bugs.webkit.org/show_bug.cgi?id=282749
Reviewed by Philippe Normand.
With the latest version of GStreamer, if the source is not selectable,
uridecodebin3 will drop the stream collection emitted from this element
As we only consider stream collection from the source element, we will
never set the stream collection internally, this will produce faulty
behaviour such as using wrong video dimensions
To avoid that, we reply true to the selectable query
*
Source/WebCore/platform/graphics/gstreamer/mse/WebKitMediaSourceGStreamer.cpp:
(webKitMediaSrcQuery):
Canonical link: https://commits.webkit.org/286347@main
Canonical link: https://commits.webkit.org/282416.298@webkitglib/2.46
Commit: e07e9654fe826eed3327fb588e0b420967849cfc
https://github.com/WebKit/WebKit/commit/e07e9654fe826eed3327fb588e0b420967849cfc
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M LayoutTests/platform/glib/TestExpectations
A
LayoutTests/platform/glib/http/wpt/mediarecorder/MediaRecorder-audio-samplingrate-change-expected.txt
M Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.cpp
M Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.h
M
Source/WebCore/platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.cpp
M
Source/WebCore/platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.h
Log Message:
-----------
Cherry-pick 286528@main (803a89518b4e).
https://bugs.webkit.org/show_bug.cgi?id=282534
[GStreamer]
http/wpt/mediarecorder/MediaRecorder-audio-samplingrate-change.html is a
permanent failure
https://bugs.webkit.org/show_bug.cgi?id=282534
Reviewed by Xabier Rodriguez-Calvar.
Reconfigure the capturer when the realtime media source settings have
changed. Platform baselines
are added for this test because matroskademux fails to seek in push mode on
the resulting WebM blob.
The MP4 test is passing though.
* LayoutTests/platform/glib/TestExpectations:
*
LayoutTests/platform/glib/http/wpt/mediarecorder/MediaRecorder-audio-samplingrate-change-expected.txt:
Added.
* Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.cpp:
(WebCore::GStreamerCapturer::isStopped const):
* Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.h:
*
Source/WebCore/platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.cpp:
(WebCore::MockRealtimeAudioSourceGStreamer::render):
(WebCore::MockRealtimeAudioSourceGStreamer::settingsDidChange):
*
Source/WebCore/platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.h:
Canonical link: https://commits.webkit.org/286528@main
Canonical link: https://commits.webkit.org/282416.299@webkitglib/2.46
Commit: 41a49ba82f3658c9242067d11ca836ace64e7796
https://github.com/WebKit/WebKit/commit/41a49ba82f3658c9242067d11ca836ace64e7796
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp
M Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp
M Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp
M Source/WebCore/platform/mediarecorder/MediaRecorderPrivateGStreamer.cpp
M Source/WebCore/platform/mediastream/RealtimeMediaSource.cpp
M Source/WebCore/platform/mediastream/RealtimeMediaSource.h
M
Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCaptureSource.cpp
M
Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCaptureSource.h
M
Source/WebCore/platform/mediastream/gstreamer/GStreamerCaptureDeviceManager.cpp
M
Source/WebCore/platform/mediastream/gstreamer/GStreamerCaptureDeviceManager.h
M Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.cpp
M Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.h
M
Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp
M Source/WebCore/platform/mediastream/gstreamer/GStreamerMockDevice.cpp
M
Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoCaptureSource.cpp
M
Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoCaptureSource.h
M
Source/WebCore/platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.cpp
M
Source/WebCore/platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.h
M
Source/WebCore/platform/mediastream/gstreamer/MockRealtimeVideoSourceGStreamer.cpp
M
Source/WebCore/platform/mediastream/gstreamer/MockRealtimeVideoSourceGStreamer.h
Log Message:
-----------
Cherry-pick 286479@main (a83ef3f05200).
https://bugs.webkit.org/show_bug.cgi?id=282756
[GStreamer][WebRTC] Latency improvements
https://bugs.webkit.org/show_bug.cgi?id=282756
<rdar://problem/139497628>
Reviewed by Xabier Rodriguez-Calvar.
When sharing data between pipelines it is recommended to use a shared base
time of 0 and -1 start
time on each pipeline. If appsrc is used its handle-segment-change property
should be enabled as
well. Then latency queries need to be plumbed as well, this was partly done
already for incoming
sources, but wasn't done yet for capture sources. Finally the RTP latency
for incoming streams was
reduced from 200ms to 40ms in webrtcbin.
With this patch the end-to-end latency on the pc1 demo went from ~10 frames
to 3 (or 4 frames).
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp:
(WebCore::GStreamerMediaEndpoint::initializePipeline):
* Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp:
(WebCore::connectSimpleBusMessageCallback):
*
Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp:
(WebCore::MediaPlayerPrivateGStreamer::handleMessage):
(WebCore::MediaPlayerPrivateGStreamer::createGSTPlayBin):
* Source/WebCore/platform/mediarecorder/MediaRecorderPrivateGStreamer.cpp:
(WebCore::MediaRecorderPrivateBackend::preparePipeline):
* Source/WebCore/platform/mediastream/RealtimeMediaSource.h:
*
Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCaptureSource.cpp:
(WebCore::GStreamerAudioCaptureSource::GStreamerAudioCaptureSource):
(WebCore::GStreamerAudioCaptureSource::queryCaptureLatency const):
*
Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCaptureSource.h:
*
Source/WebCore/platform/mediastream/gstreamer/GStreamerCaptureDeviceManager.cpp:
(WebCore::GStreamerCaptureDeviceManager::registerCapturer):
*
Source/WebCore/platform/mediastream/gstreamer/GStreamerCaptureDeviceManager.h:
* Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.cpp:
(WebCore::GStreamerCapturer::setupPipeline):
(WebCore::GStreamerCapturer::queryLatency):
* Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.h:
*
Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp:
* Source/WebCore/platform/mediastream/gstreamer/GStreamerMockDevice.cpp:
(webkitMockDeviceCreateElement):
*
Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoCaptureSource.cpp:
(WebCore::m_deviceType):
(WebCore::GStreamerVideoCaptureSource::queryCaptureLatency const):
*
Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoCaptureSource.h:
*
Source/WebCore/platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.cpp:
(WebCore::MockRealtimeAudioSourceGStreamer::queryCaptureLatency const):
*
Source/WebCore/platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.h:
*
Source/WebCore/platform/mediastream/gstreamer/MockRealtimeVideoSourceGStreamer.cpp:
(WebCore::MockRealtimeVideoSourceGStreamer::queryCaptureLatency const):
*
Source/WebCore/platform/mediastream/gstreamer/MockRealtimeVideoSourceGStreamer.h:
Canonical link: https://commits.webkit.org/286479@main
Canonical link: https://commits.webkit.org/282416.300@webkitglib/2.46
Commit: 5046e6f857498e1aada9f739baf1a4355df00703
https://github.com/WebKit/WebKit/commit/5046e6f857498e1aada9f739baf1a4355df00703
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
A LayoutTests/webrtc/stats-timestamp-increases-expected.txt
A LayoutTests/webrtc/stats-timestamp-increases.html
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerStatsCollector.cpp
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.cpp
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.h
Log Message:
-----------
Cherry-pick 283693@main (fa7d13583e7f).
https://bugs.webkit.org/show_bug.cgi?id=279508
[GStreamer][WebRTC] Rewrite stats timestamps
https://bugs.webkit.org/show_bug.cgi?id=279508
Reviewed by Xabier Rodriguez-Calvar.
The stats provided by webrtcbin are timestamped using the system monotonic
time, which is not
exactly following the spec which stipulates that we should use
Performance.timeOrigin +
Performance.now(). By applying the monotonic offset to the epoch we can use
Performance::reduceTimeResolution() and get a bit closer to spec compliance.
Covered by a new test, inspired from
imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-getStats.https.html
that checks the
timestamp remain close to window.performance.timeOrigin +
window.performance.now().
* LayoutTests/webrtc/stats-timestamp-increases-expected.txt: Added.
* LayoutTests/webrtc/stats-timestamp-increases.html: Added.
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp:
(WebCore::GStreamerMediaEndpoint::preprocessStats):
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerStatsCollector.cpp:
(WebCore::RTCStatsReport::Stats::Stats):
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.cpp:
(WebCore::StatsTimestampConverter::singleton):
(WebCore::StatsTimestampConverter::convertFromMonotonicTime const):
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.h:
Canonical link: https://commits.webkit.org/283693@main
Canonical link: https://commits.webkit.org/282416.301@webkitglib/2.46
Commit: 39af06eb8ff0824fb82ce4fbcc3b06f5821047a8
https://github.com/WebKit/WebKit/commit/39af06eb8ff0824fb82ce4fbcc3b06f5821047a8
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerStatsCollector.cpp
M
Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp
Log Message:
-----------
Cherry-pick 285185@main (ea312518dd5e).
https://bugs.webkit.org/show_bug.cgi?id=281308
[GStreamer][WebRTC] Fill trackIdentifier in incoming stream stats
https://bugs.webkit.org/show_bug.cgi?id=281308
Reviewed by Xabier Rodriguez-Calvar.
The MediaStreamTrack is associated at the GStreamer MediaStream source
element level, so fill it in
and plumb the value back to the stats collector.
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp:
(WebCore::GStreamerMediaEndpoint::preprocessStats):
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerStatsCollector.cpp:
(WebCore::RTCStatsReport::InboundRtpStreamStats::InboundRtpStreamStats):
*
Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp:
Canonical link: https://commits.webkit.org/285185@main
Canonical link: https://commits.webkit.org/282416.302@webkitglib/2.46
Commit: a4d2116d57a8827f430496e273db25179469be5b
https://github.com/WebKit/WebKit/commit/a4d2116d57a8827f430496e273db25179469be5b
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerStatsCollector.cpp
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.cpp
M Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp
M Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.h
M Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp
M Source/WebCore/platform/gstreamer/GStreamerElementHarness.cpp
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeIncomingVideoSourceGStreamer.cpp
Log Message:
-----------
Cherry-pick 285186@main (0cf4ab0811ed).
https://bugs.webkit.org/show_bug.cgi?id=281321
[GStreamer] Fix 1.25 deprecation warnings
https://bugs.webkit.org/show_bug.cgi?id=281321
Reviewed by Xabier Rodriguez-Calvar.
The GQuark-based GStreamer APIs are going to be deprecated in GStreamer
1.26, in favor of the
GstIdStr-based APIs, so introduce wrapper functions handling both cases in
GStreamerCommon.
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp:
(WebCore::GStreamerMediaEndpoint::preprocessStats):
(WebCore::GStreamerMediaEndpoint::onStatsDelivered):
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerStatsCollector.cpp:
(WebCore::fillReportCallback):
(WebCore::GStreamerStatsCollector::getStats):
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.cpp:
(WebCore::capsFromSDPMedia):
* Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp:
(WebCore::gstStructureToJSON):
(WebCore::gstStructureForeach):
(WebCore::gstStructureIdSetValue):
(WebCore::gstStructureMapInPlace):
(WebCore::gstIdToString):
(WebCore::gstStructureFilterAndMapInPlace):
(WebCore::parseGstStructureValue): Deleted.
* Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.h:
* Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp:
(webKitWebSrcSetExtraHeader):
(webKitWebSrcMakeRequest):
(webKitWebSrcProcessExtraHeaders): Deleted.
* Source/WebCore/platform/gstreamer/GStreamerElementHarness.cpp:
(WebCore::MermaidBuilder::describeCaps):
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeIncomingVideoSourceGStreamer.cpp:
(WebCore::RealtimeIncomingVideoSourceGStreamer::stats):
Canonical link: https://commits.webkit.org/285186@main
Canonical link: https://commits.webkit.org/282416.303@webkitglib/2.46
Commit: 746bc62be45860b640a4b70757dfa5888f4b9b41
https://github.com/WebKit/WebKit/commit/746bc62be45860b640a4b70757dfa5888f4b9b41
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp
Log Message:
-----------
Cherry-pick 285187@main (e93fff94a58a). <bug>
[GStreamer] Fix build after 285186@main
Unreviewed.
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp:
(WebCore::GStreamerMediaEndpoint::preprocessStats):
Canonical link: https://commits.webkit.org/285187@main
Canonical link: https://commits.webkit.org/282416.304@webkitglib/2.46
Commit: eeb9c84642b1ce935c87530858c6d84d507dc37b
https://github.com/WebKit/WebKit/commit/eeb9c84642b1ce935c87530858c6d84d507dc37b
Author: Youenn Fablet <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M
LayoutTests/imported/w3c/web-platform-tests/webrtc/simulcast/negotiation-encodings.https-expected.txt
M LayoutTests/webrtc/video-getParameters-expected.txt
M LayoutTests/webrtc/video-getParameters.html
M Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp
M Source/WebCore/Modules/mediastream/RTCPeerConnection.h
M Source/WebCore/Modules/mediastream/RTCRtpEncodingParameters.h
M Source/WebCore/Modules/mediastream/RTCRtpEncodingParameters.idl
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.cpp
M Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp
Log Message:
-----------
Cherry-pick 283750@main (cbd18cef4633).
https://bugs.webkit.org/show_bug.cgi?id=279193
RTCRtpEncodingParameters has wrong default values
https://bugs.webkit.org/show_bug.cgi?id=279193
rdar://135345629
Reviewed by Philippe Normand.
We update the WebIDL according the spec.
The default values, in particular scaleResolutionDownBy, triggered an
interop issue:
- Safari would return 1 on all layers
- Chrome would return undefined (meaning default value 1,2, 4 is to be
applied).
- Firefox would return 1, 2, 4 as defined in the spec.
By updating the WebIDL and binding code to libwebrtc, we align with Chrome.
This includes aligning on failures in
LayoutTests/imported/w3c/web-platform-tests/webrtc/simulcast/negotiation-encodings.https.html.
This is probably the safest approach for now, until we reach out consensus
within the WebRTC WG.
*
LayoutTests/imported/w3c/web-platform-tests/webrtc/simulcast/negotiation-encodings.https-expected.txt:
* LayoutTests/webrtc/video-getParameters-expected.txt:
* LayoutTests/webrtc/video-getParameters.html:
* Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp:
(WebCore::isAudioTransceiver):
(WebCore::validateSendEncodings):
(WebCore::RTCPeerConnection::addTransceiver):
* Source/WebCore/Modules/mediastream/RTCPeerConnection.h:
* Source/WebCore/Modules/mediastream/RTCRtpEncodingParameters.h:
* Source/WebCore/Modules/mediastream/RTCRtpEncodingParameters.idl:
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.cpp:
(WebCore::fromRTCEncodingParameters):
* Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp:
(WebCore::updateRTCRtpSendParameters):
Canonical link: https://commits.webkit.org/283750@main
Canonical link: https://commits.webkit.org/282416.305@webkitglib/2.46
Commit: 7aa3542b39ce3086749158f74a96b477af78cf5e
https://github.com/WebKit/WebKit/commit/7aa3542b39ce3086749158f74a96b477af78cf5e
Author: Youenn Fablet <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M Source/WTF/Scripts/Preferences/UnifiedWebPreferences.yaml
M Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
M
Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
M
Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h
M Tools/WebKitTestRunner/TestController.cpp
Log Message:
-----------
Cherry-pick 284477@main (20616a6e98e2).
https://bugs.webkit.org/show_bug.cgi?id=280571
Add an experimental flag for WebRTC L4S support
rdar://136896745
https://bugs.webkit.org/show_bug.cgi?id=280571
Reviewed by Jean-Yves Avenard and Eric Carlson.
We enable this flag to allow experimenting L4S support.
We mark it as testable for now.
We wire libwebrtc internal flag according our WebKit flag.
We disable this flag by default on WebKitTestRunner until we have SDP
negoatiation support.
* Source/WTF/Scripts/Preferences/UnifiedWebPreferences.yaml:
* Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
(WebCore::LibWebRTCMediaEndpoint::LibWebRTCMediaEndpoint):
*
Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp:
(WebCore::LibWebRTCPeerConnectionBackend::shouldEnableWebRTCL4S const):
*
Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h:
* Tools/WebKitTestRunner/TestController.cpp:
(WTR::TestController::resetPreferencesToConsistentValues):
Canonical link: https://commits.webkit.org/284477@main
Canonical link: https://commits.webkit.org/282416.306@webkitglib/2.46
Commit: ed64f58d99b0ee140039514825a6cc06ee2ea948
https://github.com/WebKit/WebKit/commit/ed64f58d99b0ee140039514825a6cc06ee2ea948
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M LayoutTests/platform/glib/TestExpectations
M
LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-answer-expected.txt
M
LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-offer-expected.txt
A
LayoutTests/platform/glib/imported/w3c/web-platform-tests/webrtc/RTCRtpParameters-encodings-expected.txt
M Source/WebCore/Modules/mediastream/MediaStream.cpp
M Source/WebCore/Modules/mediastream/MediaStreamTrack.h
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.h
M
Source/WebCore/Modules/mediastream/gstreamer/GStreamerPeerConnectionBackend.cpp
M
Source/WebCore/Modules/mediastream/gstreamer/GStreamerPeerConnectionBackend.h
M
Source/WebCore/Modules/mediastream/gstreamer/GStreamerRtpReceiverBackend.cpp
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerRtpReceiverBackend.h
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerRtpSenderBackend.cpp
M
Source/WebCore/Modules/mediastream/gstreamer/GStreamerRtpTransceiverBackend.cpp
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerStatsCollector.cpp
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.cpp
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.h
M Source/WebCore/platform/SourcesGStreamer.txt
A
Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioRTPPacketizer.cpp
A
Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioRTPPacketizer.h
M
Source/WebCore/platform/mediastream/gstreamer/GStreamerIncomingTrackProcessor.cpp
M
Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp
A Source/WebCore/platform/mediastream/gstreamer/GStreamerRTPPacketizer.cpp
A Source/WebCore/platform/mediastream/gstreamer/GStreamerRTPPacketizer.h
A
Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoRTPPacketizer.cpp
A
Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoRTPPacketizer.h
M Source/WebCore/platform/mediastream/gstreamer/GStreamerWebRTCCommon.h
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.cpp
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.h
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.cpp
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.h
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.cpp
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.h
Log Message:
-----------
Cherry-pick 286800@main (a5509c9f1bd7).
https://bugs.webkit.org/show_bug.cgi?id=282926
[GStreamer][WebRTC] Pipeline revamp in preparation for simulcast support
https://bugs.webkit.org/show_bug.cgi?id=282926
Reviewed by Xabier Rodriguez-Calvar.
Outgoing sources can now multiplex multiple RTP streams, this is going to
be required for simulcast
support. Encoding/payloading are now done in a separate sub-bin, managed by
new
GStreamer{Audio,Video}RTPPacketizer classes. Encoding RTP parameters
validation support was improved
as well.
* LayoutTests/platform/glib/TestExpectations:
*
LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-answer-expected.txt:
*
LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-offer-expected.txt:
* Source/WebCore/Modules/mediastream/MediaStream.cpp:
(WebCore::MediaStream::MediaStream):
* Source/WebCore/Modules/mediastream/MediaStreamTrack.h:
(WebCore::MediaStreamTrack::setMediaStreamId):
(WebCore::MediaStreamTrack::mediaStreamId const):
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp:
(WebCore::GStreamerMediaEndpoint::initializePipeline):
(WebCore::GStreamerMediaEndpoint::setConfiguration):
(WebCore::getMediaStreamIdsFromSDPMedia):
(WebCore::toGStreamerMediaEndpointTransceiverState):
(WebCore::transceiverStatesFromWebRTCBin):
(WebCore::GStreamerMediaEndpoint::doSetLocalDescription):
(WebCore::GStreamerMediaEndpoint::doSetRemoteDescription):
(WebCore::GStreamerMediaEndpoint::setDescription):
(WebCore::GStreamerMediaEndpoint::processSDPMessage):
(WebCore::GStreamerMediaEndpoint::configureSource):
(WebCore::GStreamerMediaEndpoint::requestPad):
(WebCore::GStreamerMediaEndpoint::addTrack):
(WebCore::GStreamerMediaEndpoint::initiate):
(WebCore::GStreamerMediaEndpoint::trackIdFromSDPMedia):
(WebCore::GStreamerMediaEndpoint::connectIncomingTrack):
(WebCore::GStreamerMediaEndpoint::connectPad):
(WebCore::GStreamerMediaEndpoint::createTransceiverBackends):
(WebCore::GStreamerMediaEndpoint::addTransceiver):
(WebCore::GStreamerMediaEndpoint::createMutedSource):
(WebCore::GStreamerMediaEndpoint::createSourceForTrack):
(WebCore::GStreamerMediaEndpoint::close):
(WebCore::GStreamerMediaEndpoint::suspend):
(WebCore::GStreamerMediaEndpoint::onNegotiationNeeded):
(WebCore::GStreamerMediaEndpoint::collectTransceivers):
(WebCore::GStreamerMediaEndpoint::preprocessStats):
(WebCore::GStreamerMediaEndpoint::configureAndLinkSource): Deleted.
(WebCore::GStreamerMediaEndpoint::createLinkedSourceForTrack): Deleted.
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.h:
*
Source/WebCore/Modules/mediastream/gstreamer/GStreamerPeerConnectionBackend.cpp:
(WebCore::GStreamerPeerConnectionBackend::addTrack):
(WebCore::GStreamerPeerConnectionBackend::createSourceForTrack):
(WebCore::GStreamerPeerConnectionBackend::existingTransceiverForTrackId):
(WebCore::GStreamerPeerConnectionBackend::createLinkedSourceForTrack):
Deleted.
*
Source/WebCore/Modules/mediastream/gstreamer/GStreamerPeerConnectionBackend.h:
*
Source/WebCore/Modules/mediastream/gstreamer/GStreamerRtpReceiverBackend.cpp:
(WebCore::GStreamerRtpReceiverBackend::GStreamerRtpReceiverBackend):
(WebCore::GStreamerRtpReceiverBackend::getParameters):
*
Source/WebCore/Modules/mediastream/gstreamer/GStreamerRtpReceiverBackend.h:
*
Source/WebCore/Modules/mediastream/gstreamer/GStreamerRtpSenderBackend.cpp:
(WebCore::m_initData):
(WebCore::GStreamerRtpSenderBackend::setSource):
(WebCore::GStreamerRtpSenderBackend::replaceTrack):
(WebCore::GStreamerRtpSenderBackend::setParameters):
*
Source/WebCore/Modules/mediastream/gstreamer/GStreamerRtpTransceiverBackend.cpp:
(WebCore::GStreamerRtpTransceiverBackend::GStreamerRtpTransceiverBackend):
(WebCore::GStreamerRtpTransceiverBackend::createReceiverBackend):
(WebCore::toRtpCodecCapability):
(WebCore::GStreamerRtpTransceiverBackend::setCodecPreferences):
(WebCore::getMsidFromCurrentCodecPreferences): Deleted.
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerStatsCollector.cpp:
(WebCore::RTCStatsReport::ReceivedRtpStreamStats::ReceivedRtpStreamStats):
(WebCore::RTCStatsReport::OutboundRtpStreamStats::OutboundRtpStreamStats):
(WebCore::GStreamerStatsCollector::getStats):
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.cpp:
(WebCore::fromRTCEncodingParameters):
(WebCore::toRTCEncodingParameters):
(WebCore::toRTCCodecParameters):
(WebCore::toRTCRtpSendParameters):
(WebCore::fromRTCCodecParameters):
(WebCore::fromRTCSendParameters):
(WebCore::capsFromRtpCapabilities):
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.h:
(WebCore::webrtcKindFromCaps):
* Source/WebCore/platform/SourcesGStreamer.txt:
*
Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioRTPPacketizer.cpp:
Added.
(WebCore::GStreamerAudioRTPPacketizer::create):
(WebCore::GStreamerAudioRTPPacketizer::GStreamerAudioRTPPacketizer):
*
Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioRTPPacketizer.h:
Copied from
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.h.
*
Source/WebCore/platform/mediastream/gstreamer/GStreamerIncomingTrackProcessor.cpp:
(WebCore::GStreamerIncomingTrackProcessor::configure):
(WebCore::GStreamerIncomingTrackProcessor::trackReady):
*
Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp:
* Source/WebCore/platform/mediastream/gstreamer/GStreamerRTPPacketizer.cpp:
Added.
(WebCore::GStreamerRTPPacketizer::GStreamerRTPPacketizer):
(WebCore::GStreamerRTPPacketizer::configureExtensions):
(WebCore::GStreamerRTPPacketizer::ensureMidExtension):
(WebCore::GStreamerRTPPacketizer::rtpParameters const):
(WebCore::GStreamerRTPPacketizer::rtpStreamId const):
(WebCore::GStreamerRTPPacketizer::payloadType const):
(WebCore::GStreamerRTPPacketizer::currentSequenceNumberOffset const):
(WebCore::GStreamerRTPPacketizer::setSequenceNumberOffset):
(WebCore::GStreamerRTPPacketizer::findLastExtensionId):
(WebCore::GStreamerRTPPacketizer::stats const):
(WebCore::GStreamerRTPPacketizer::startUpdatingStats):
(WebCore::GStreamerRTPPacketizer::updateStatsFromRTPExtensions):
(WebCore::GStreamerRTPPacketizer::stopUpdatingStats):
(WebCore::GStreamerRTPPacketizer::applyEncodingParameters const):
(WebCore::GStreamerRTPPacketizer::reconfigure):
* Source/WebCore/platform/mediastream/gstreamer/GStreamerRTPPacketizer.h:
Added.
(WebCore::GStreamerRTPPacketizer::bin const):
(WebCore::GStreamerRTPPacketizer::payloader const):
(WebCore::GStreamerRTPPacketizer::updateStats):
(WebCore::GStreamerRTPPacketizer::configure const):
*
Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoRTPPacketizer.cpp:
Added.
(WebCore::GStreamerVideoRTPPacketizer::create):
(WebCore::GStreamerVideoRTPPacketizer::GStreamerVideoRTPPacketizer):
(WebCore::GStreamerVideoRTPPacketizer::configure const):
(WebCore::GStreamerVideoRTPPacketizer::updateStats):
*
Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoRTPPacketizer.h:
Copied from
Source/WebCore/platform/mediastream/gstreamer/GStreamerWebRTCCommon.h.
* Source/WebCore/platform/mediastream/gstreamer/GStreamerWebRTCCommon.h:
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.cpp:
(WebCore::RealtimeOutgoingAudioSourceGStreamer::RealtimeOutgoingAudioSourceGStreamer):
(WebCore::RealtimeOutgoingAudioSourceGStreamer::initialize):
(WebCore::RealtimeOutgoingAudioSourceGStreamer::linkTee):
(WebCore::RealtimeOutgoingAudioSourceGStreamer::outgoingSourcePad const):
(WebCore::RealtimeOutgoingAudioSourceGStreamer::createPacketizer):
(WebCore::RealtimeOutgoingAudioSourceGStreamer::setPayloadType): Deleted.
(WebCore::RealtimeOutgoingAudioSourceGStreamer::setParameters): Deleted.
(WebCore::RealtimeOutgoingAudioSourceGStreamer::teardown): Deleted.
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.h:
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.cpp:
(WebCore::RealtimeOutgoingMediaSourceGStreamer::RealtimeOutgoingMediaSourceGStreamer):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::~RealtimeOutgoingMediaSourceGStreamer):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::initialize):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::allowedCaps const):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::rtpCaps const):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::start):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::stopOutgoingSource):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::sourceEnabledChanged):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::initializeSourceFromTrackPrivate):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::link):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::setSinkPad):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::checkMid):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::parameters):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::codecPreferencesChanged):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::replaceTrack):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::setInitialParameters):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::configure):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::setParameters):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::getPacketizerForRid):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::linkPacketizer):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::configurePacketizers):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::lookupRtpExtensions):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::stats):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::startUpdatingStats):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::stopUpdatingStats):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::teardown):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::flush): Deleted.
(WebCore::RealtimeOutgoingMediaSourceGStreamer::initializeFromTrack):
Deleted.
(WebCore::RealtimeOutgoingMediaSourceGStreamer::unlinkPayloader): Deleted.
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.h:
(WebCore::RealtimeOutgoingMediaSourceGStreamer::setPayloadType): Deleted.
(WebCore::RealtimeOutgoingMediaSourceGStreamer::fillEncodingParameters):
Deleted.
(WebCore::RealtimeOutgoingMediaSourceGStreamer::setParameters): Deleted.
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.cpp:
(WebCore::RealtimeOutgoingVideoSourceGStreamer::RealtimeOutgoingVideoSourceGStreamer):
(WebCore::RealtimeOutgoingVideoSourceGStreamer::initializePreProcessor):
(WebCore::RealtimeOutgoingVideoSourceGStreamer::teardown):
(WebCore::RealtimeOutgoingVideoSourceGStreamer::rtpCapabilities const):
(WebCore::RealtimeOutgoingVideoSourceGStreamer::linkTee):
(WebCore::RealtimeOutgoingVideoSourceGStreamer::outgoingSourcePad const):
(WebCore::RealtimeOutgoingVideoSourceGStreamer::createPacketizer):
(WebCore::RealtimeOutgoingVideoSourceGStreamer::updateStats): Deleted.
(WebCore::RealtimeOutgoingVideoSourceGStreamer::setPayloadType): Deleted.
(WebCore::RealtimeOutgoingVideoSourceGStreamer::startUpdatingStats):
Deleted.
(WebCore::RealtimeOutgoingVideoSourceGStreamer::stopUpdatingStats): Deleted.
(WebCore::RealtimeOutgoingVideoSourceGStreamer::sourceEnabledChanged):
Deleted.
(WebCore::RealtimeOutgoingVideoSourceGStreamer::flush): Deleted.
(WebCore::RealtimeOutgoingVideoSourceGStreamer::setParameters): Deleted.
(WebCore::RealtimeOutgoingVideoSourceGStreamer::fillEncodingParameters):
Deleted.
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.h:
Canonical link: https://commits.webkit.org/286800@main
Canonical link: https://commits.webkit.org/282416.307@webkitglib/2.46
Commit: 74ed13858f1bfe02c0d69e422592e13250ed88e3
https://github.com/WebKit/WebKit/commit/74ed13858f1bfe02c0d69e422592e13250ed88e3
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M LayoutTests/platform/glib/TestExpectations
M Source/WebCore/platform/mediarecorder/MediaRecorderPrivateGStreamer.cpp
M Source/WebCore/platform/mediarecorder/MediaRecorderPrivateGStreamer.h
Log Message:
-----------
Cherry-pick 286587@main (237403719d8f).
https://bugs.webkit.org/show_bug.cgi?id=282911
[GStreamer]
imported/w3c/web-platform-tests/mediacapture-record/MediaRecorder-blob-timecode.https.html
is a permanent failure
https://bugs.webkit.org/show_bug.cgi?id=282911
Reviewed by Xabier Rodriguez-Calvar.
Ensure the first blob event has a time code set to 0. Following ones use
the position reported by
the transcoder.
* LayoutTests/platform/glib/TestExpectations:
* Source/WebCore/platform/mediarecorder/MediaRecorderPrivateGStreamer.cpp:
(WebCore::MediaRecorderPrivateBackend::stopRecording):
(WebCore::MediaRecorderPrivateBackend::fetchData):
(WebCore::MediaRecorderPrivateBackend::notifyPosition):
* Source/WebCore/platform/mediarecorder/MediaRecorderPrivateGStreamer.h:
(WebCore::MediaRecorderPrivateBackend::WTF_GUARDED_BY_LOCK):
(WebCore::MediaRecorderPrivateBackend::notifyPosition): Deleted.
Canonical link: https://commits.webkit.org/286587@main
Canonical link: https://commits.webkit.org/282416.308@webkitglib/2.46
Commit: 2b66d679071f5180c3a8e921f1dfd5db2f673b09
https://github.com/WebKit/WebKit/commit/2b66d679071f5180c3a8e921f1dfd5db2f673b09
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M
Source/WebCore/platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.cpp
Log Message:
-----------
Cherry-pick 286718@main (aaf43cf52732).
https://bugs.webkit.org/show_bug.cgi?id=283170
REGRESSION(286528@main): Triggers asserts in Mock audio source
https://bugs.webkit.org/show_bug.cgi?id=283170
Reviewed by Xabier Rodriguez-Calvar.
*
Source/WebCore/platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.cpp:
(WebCore::MockRealtimeAudioSourceGStreamer::render): Make sure the stream
format is set before
attempting to access it. Also simplify the loop exit condition.
Canonical link: https://commits.webkit.org/286718@main
Canonical link: https://commits.webkit.org/282416.309@webkitglib/2.46
Commit: 7dc318887c7ca2088013c47044e81749b27267dc
https://github.com/WebKit/WebKit/commit/7dc318887c7ca2088013c47044e81749b27267dc
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp
M Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.h
M Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoCapturer.cpp
Log Message:
-----------
Cherry-pick 286790@main (34beac2e3606).
https://bugs.webkit.org/show_bug.cgi?id=283298
[GStreamer][PipeWire] getDisplayMedia() broken
https://bugs.webkit.org/show_bug.cgi?id=283298
Reviewed by Xabier Rodriguez-Calvar.
Move the DMABuf caps builder utility function to GStreamerCommon and use it
from the VideoCapturer,
fixing caps negotiation issues with PipeWire.
* Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp:
(initializeDMABufAvailability):
(drmFourccToGstVideoFormat): Deleted.
(buildDMABufCaps): Deleted.
* Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp:
(WebCore::drmFourccToGstVideoFormat):
(WebCore::buildDMABufCaps):
* Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.h:
* Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoCapturer.cpp:
(WebCore::GStreamerVideoCapturer::createConverter):
Canonical link: https://commits.webkit.org/286790@main
Canonical link: https://commits.webkit.org/282416.310@webkitglib/2.46
Commit: a4fa68a3f7fd944ebc0ecec51d5ecb42ed71522b
https://github.com/WebKit/WebKit/commit/a4fa68a3f7fd944ebc0ecec51d5ecb42ed71522b
Author: Carlos Bentzen <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M
Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoDecoderFactory.cpp
Log Message:
-----------
Cherry-pick 286848@main (20f5df0feb7d).
https://bugs.webkit.org/show_bug.cgi?id=283359
[GStreamer][LibWebRTC] No H.264 decoder found when using broadcom and
westeros quirks and no FFmpeg installed
https://bugs.webkit.org/show_bug.cgi?id=283359
Reviewed by Philippe Normand.
Use GStreamerRegistryScanner::isCodecSupport() instead of the
hand-written implementation of GstDecoderFactory(), because this is more
robust to detect decoders in hardware-accelerated scenarios with custom
elements and GStreamer quirks.
This is cherry-pick of
https://github.com/WebPlatformForEmbedded/WPEWebKit/commit/d7a7509280358127a6ac75d9ba2242aa9b88150e
Original author: Vivek.A <[email protected]>
*
Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoDecoderFactory.cpp:
(WebCore::GStreamerWebRTCVideoDecoder::GstDecoderFactory):
Canonical link: https://commits.webkit.org/286848@main
Canonical link: https://commits.webkit.org/282416.311@webkitglib/2.46
Commit: 52919c8dab086a2f784d9268fc452f236f59fc70
https://github.com/WebKit/WebKit/commit/52919c8dab086a2f784d9268fc452f236f59fc70
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.cpp
Log Message:
-----------
Cherry-pick 286845@main (618f62718940).
https://bugs.webkit.org/show_bug.cgi?id=283361
[GStreamer][WebRTC] Use autovideoflip for outgoing video stream rotation
handling
https://bugs.webkit.org/show_bug.cgi?id=283361
Reviewed by Xabier Rodriguez-Calvar.
The autovideoflip element provides more flexibility regarding input formats
and underlying memory.
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.cpp:
(WebCore::RealtimeOutgoingVideoSourceGStreamer::initializePreProcessor):
Canonical link: https://commits.webkit.org/286845@main
Canonical link: https://commits.webkit.org/282416.312@webkitglib/2.46
Commit: 0948ea98c0b80551e6dc1cfe415612f742644db4
https://github.com/WebKit/WebKit/commit/0948ea98c0b80551e6dc1cfe415612f742644db4
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M LayoutTests/platform/glib/TestExpectations
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.h
M
Source/WebCore/Modules/mediastream/gstreamer/GStreamerPeerConnectionBackend.cpp
M
Source/WebCore/Modules/mediastream/gstreamer/GStreamerPeerConnectionBackend.h
Log Message:
-----------
Cherry-pick 286894@main (dddc53d20d11).
https://bugs.webkit.org/show_bug.cgi?id=283354
[GStreamer][WebRTC] webrtc/video-lowercase-media-subtype.html is timing out
https://bugs.webkit.org/show_bug.cgi?id=283354
Reviewed by Xabier Rodriguez-Calvar.
The test was failing because the incoming track was associated with new
transceiver instead of
reusing the one created during the addTransceiver() call. There was also a
mix-up of MediaStream and
Track IDs.
* LayoutTests/platform/glib/TestExpectations:
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp:
(WebCore::GStreamerMediaEndpoint::connectIncomingTrack):
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.h:
*
Source/WebCore/Modules/mediastream/gstreamer/GStreamerPeerConnectionBackend.cpp:
(WebCore::GStreamerPeerConnectionBackend::existingTransceiverForTrackId):
Deleted.
*
Source/WebCore/Modules/mediastream/gstreamer/GStreamerPeerConnectionBackend.h:
Canonical link: https://commits.webkit.org/286894@main
Canonical link: https://commits.webkit.org/282416.313@webkitglib/2.46
Commit: 1a55f44f87ea9c85244707dd7f742c401dac1c0a
https://github.com/WebKit/WebKit/commit/1a55f44f87ea9c85244707dd7f742c401dac1c0a
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.cpp
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.h
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.cpp
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.h
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.cpp
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.h
Log Message:
-----------
Cherry-pick 286895@main (0b855998667a).
https://bugs.webkit.org/show_bug.cgi?id=283411
REGRESSION(286479@main): [GStreamer][WebRTC] Broke
webrtc/captureCanvas-webrtc.html
https://bugs.webkit.org/show_bug.cgi?id=283411
Reviewed by Xabier Rodriguez-Calvar.
The outgoing tracks no longer require a livesync element since the WebRTC
pipeline and playback
pipeline share the same clock.
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.cpp:
(WebCore::RealtimeOutgoingAudioSourceGStreamer::linkTee): Deleted.
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.h:
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.cpp:
(WebCore::RealtimeOutgoingMediaSourceGStreamer::RealtimeOutgoingMediaSourceGStreamer):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::start):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::stopOutgoingSource):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::linkSource):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::configurePacketizers):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::teardown):
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.h:
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.cpp:
(WebCore::RealtimeOutgoingVideoSourceGStreamer::linkTee): Deleted.
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.h:
Canonical link: https://commits.webkit.org/286895@main
Canonical link: https://commits.webkit.org/282416.314@webkitglib/2.46
Commit: 2163d0eaabc874b205fd465787b4cb6ca1a40cef
https://github.com/WebKit/WebKit/commit/2163d0eaabc874b205fd465787b4cb6ca1a40cef
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M Source/WebCore/platform/graphics/gstreamer/TrackPrivateBaseGStreamer.cpp
M
Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp
Log Message:
-----------
Cherry-pick 286896@main (4b3a7f836f5a).
https://bugs.webkit.org/show_bug.cgi?id=283414
[GStreamer][WebRTC] Canvas to PeerConnection produces flipped video frames
on receiving side
https://bugs.webkit.org/show_bug.cgi?id=283414
Reviewed by Xabier Rodriguez-Calvar.
The issue was that the pad probe in the mediastreamsrc element was pushing
a tag event without
orientation information, after the video frame observer notified its first
frame (with video
rotation), so a tag event with rotation was pushed from the
videoFrameAvailable callback and then
another one without video rotation info was pushed from the pad probe.
*
Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp:
(webkitMediaStreamSrcPadProbeCb):
(webkitMediaStreamSrcAddTrack):
Canonical link: https://commits.webkit.org/286896@main
Canonical link: https://commits.webkit.org/282416.315@webkitglib/2.46
Commit: 3c3b9d8cc2da042017bb0ff2e00357809d075fa9
https://github.com/WebKit/WebKit/commit/3c3b9d8cc2da042017bb0ff2e00357809d075fa9
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M
Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoRTPPacketizer.cpp
Log Message:
-----------
Cherry-pick 287056@main (6a6659e8614f).
https://bugs.webkit.org/show_bug.cgi?id=283557
[GStreamer][WebRTC] Fallback to constrained-baseline when H.264 is
negotiated for outgoing video source
https://bugs.webkit.org/show_bug.cgi?id=283557
Reviewed by Xabier Rodriguez-Calvar.
According to RFC 7742, endpoints MUST support Constrained Baseline Profile
Level 1.2 and SHOULD
support H.264 Constrained High Profile Level 1.3. So if no profile is
specified, use
constrained-baseline.
The AVC constraint flags were also wrongly set for this profile, so fix
that as well, by setting the
second most significant bit to 1, corresponding to the constraint_set1_flag.
*
Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoRTPPacketizer.cpp:
(WebCore::GStreamerVideoRTPPacketizer::create):
Canonical link: https://commits.webkit.org/287056@main
Canonical link: https://commits.webkit.org/282416.316@webkitglib/2.46
Commit: bf31036436930fcb03e4c118e7e749b50d9a3fc2
https://github.com/WebKit/WebKit/commit/bf31036436930fcb03e4c118e7e749b50d9a3fc2
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M
Source/WebCore/platform/graphics/gstreamer/mse/MediaPlayerPrivateGStreamerMSE.cpp
Log Message:
-----------
Cherry-pick 287055@main (33adf31f1891).
https://bugs.webkit.org/show_bug.cgi?id=283622
REGRESSION(279943@main): [GStreamer][MSE] Infinite recursion when pausing
the player
https://bugs.webkit.org/show_bug.cgi?id=283622
Reviewed by Jean-Yves Avenard.
Exit early from the pause method in case the player is already paused, in
order to prevent an
infinite recursion triggered by the call to player->playbackStateChanged()
that would make the media
element call pause() again, and so on.
*
Source/WebCore/platform/graphics/gstreamer/mse/MediaPlayerPrivateGStreamerMSE.cpp:
(WebCore::MediaPlayerPrivateGStreamerMSE::pause):
Canonical link: https://commits.webkit.org/287055@main
Canonical link: https://commits.webkit.org/282416.317@webkitglib/2.46
Commit: 42878c03b353da5da4a7435316f4b8873de0581a
https://github.com/WebKit/WebKit/commit/42878c03b353da5da4a7435316f4b8873de0581a
Author: Philippe Normand <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M Source/WebCore/platform/mediarecorder/MediaRecorderPrivateGStreamer.cpp
Log Message:
-----------
Cherry-pick 287057@main (06c5d71ff724).
https://bugs.webkit.org/show_bug.cgi?id=283646
[GStreamer][MediaRecorder] Restrict audio samplerate to original value
https://bugs.webkit.org/show_bug.cgi?id=283646
Reviewed by Xabier Rodriguez-Calvar.
This effectively ignores sampleRate changes on the input audio track. See
also bug 282489. Tested
manually.
* Source/WebCore/platform/mediarecorder/MediaRecorderPrivateGStreamer.cpp:
(WebCore::MediaRecorderPrivateBackend::containerProfile):
Canonical link: https://commits.webkit.org/287057@main
Canonical link: https://commits.webkit.org/282416.318@webkitglib/2.46
Commit: 847af600d25f5de30f0f89df506c7138d02c348e
https://github.com/WebKit/WebKit/commit/847af600d25f5de30f0f89df506c7138d02c348e
Author: Alicia Boya Garcia <[email protected]>
Date: 2024-11-25 (Mon, 25 Nov 2024)
Changed paths:
M Source/WebCore/platform/graphics/TrackPrivateBase.h
M
Source/WebCore/platform/graphics/gstreamer/InbandTextTrackPrivateGStreamer.cpp
M Source/WebCore/platform/graphics/gstreamer/TrackPrivateBaseGStreamer.cpp
Log Message:
-----------
Cherry-pick 286324@main (d6348a3fcc70).
https://bugs.webkit.org/show_bug.cgi?id=282396
[GStreamer] Fix invalid AtomString usage in TrackPrivateBaseGStreamer
https://bugs.webkit.org/show_bug.cgi?id=282396
Reviewed by Philippe Normand and Xabier Rodriguez-Calvar.
This fixes a flaky assertion crash in
imported/w3c/web-platform-tests/media-source/mediasource-sequencemode-append-buffer.html
AtomString values must be always created, read, written and destroyed
from the same thread, as not only they're not thread-safe, but they also
depend on thread-local storage.
Fortunately, this is already the case for most usages I've seen in
TrackPrivateBaseGStreamer and derived. This patch fixes the one case of
cross-thread access I was able to identify as the culprit behind the
crash.
This patch also adds a number of thread assertions and comments so that
it is more clear what functions should be called only in main thread and
which ones may be called from any thread.
* Source/WebCore/platform/graphics/TrackPrivateBase.h:
*
Source/WebCore/platform/graphics/gstreamer/InbandTextTrackPrivateGStreamer.cpp:
(WebCore::InbandTextTrackPrivateGStreamer::tagsChanged):
* Source/WebCore/platform/graphics/gstreamer/TrackPrivateBaseGStreamer.cpp:
(WebCore::TrackPrivateBaseGStreamer::setPad):
(WebCore::TrackPrivateBaseGStreamer::tagsChanged):
(WebCore::TrackPrivateBaseGStreamer::notifyTrackOfTagsChanged):
(WebCore::TrackPrivateBaseGStreamer::notifyTrackOfStreamChanged):
(WebCore::TrackPrivateBaseGStreamer::updateTrackIDFromTags):
Canonical link: https://commits.webkit.org/286324@main
Canonical link: https://commits.webkit.org/282416.319@webkitglib/2.46
Commit: dcdf85fc4f0ccd85cf3f20f7ea4fefbfcc765ca3
https://github.com/WebKit/WebKit/commit/dcdf85fc4f0ccd85cf3f20f7ea4fefbfcc765ca3
Author: Vivienne Watermeier <[email protected]>
Date: 2024-11-26 (Tue, 26 Nov 2024)
Changed paths:
M Source/WebCore/platform/graphics/gstreamer/AudioTrackPrivateGStreamer.cpp
M Source/WebCore/platform/graphics/gstreamer/AudioTrackPrivateGStreamer.h
M Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp
M Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.h
M
Source/WebCore/platform/graphics/gstreamer/InbandTextTrackPrivateGStreamer.cpp
M
Source/WebCore/platform/graphics/gstreamer/InbandTextTrackPrivateGStreamer.h
M Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp
M Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h
M Source/WebCore/platform/graphics/gstreamer/TextSinkGStreamer.cpp
M Source/WebCore/platform/graphics/gstreamer/TrackPrivateBaseGStreamer.cpp
M Source/WebCore/platform/graphics/gstreamer/TrackPrivateBaseGStreamer.h
M Source/WebCore/platform/graphics/gstreamer/VideoTrackPrivateGStreamer.cpp
M Source/WebCore/platform/graphics/gstreamer/VideoTrackPrivateGStreamer.h
M Source/WebCore/platform/graphics/gstreamer/mse/AppendPipeline.cpp
M Source/WebCore/platform/graphics/gstreamer/mse/AppendPipeline.h
M
Source/WebCore/platform/graphics/gstreamer/mse/MediaPlayerPrivateGStreamerMSE.cpp
M
Source/WebCore/platform/graphics/gstreamer/mse/MediaSourcePrivateGStreamer.cpp
M
Source/WebCore/platform/graphics/gstreamer/mse/MediaSourcePrivateGStreamer.h
M
Source/WebCore/platform/graphics/gstreamer/mse/MediaSourceTrackGStreamer.cpp
M Source/WebCore/platform/graphics/gstreamer/mse/MediaSourceTrackGStreamer.h
M
Source/WebCore/platform/graphics/gstreamer/mse/SourceBufferPrivateGStreamer.cpp
M
Source/WebCore/platform/graphics/gstreamer/mse/SourceBufferPrivateGStreamer.h
M Source/WebCore/platform/graphics/gstreamer/mse/TrackQueue.cpp
M Source/WebCore/platform/graphics/gstreamer/mse/TrackQueue.h
M
Source/WebCore/platform/graphics/gstreamer/mse/WebKitMediaSourceGStreamer.cpp
M
Source/WebCore/platform/graphics/gstreamer/mse/WebKitMediaSourceGStreamer.h
Log Message:
-----------
Cherry-pick 286527@main (7a7367fb8f43).
https://bugs.webkit.org/show_bug.cgi?id=270638
[GStreamer] Use integers as track ID instead of AtomString
https://bugs.webkit.org/show_bug.cgi?id=270638
Reviewed by Philippe Normand.
Switch to using integers for track IDs internally instead of AtomString.
Some notable special cases:
- as MediaStream tracks are recommended by the spec to use UUID strings,
we just truncate it to 64 bits for the internal TrackID,
and return the full ID as String (stored in m_gstStreamId) in `trackUID()`
- In MSE, there is the potential for multiple SourceBuffers providing tracks
with identical IDs. Previously, this was less of an issue since IDs were
also based on track type, so two tracks "A0" and "V0" would still have
distinct IDs. However, now they would both get ID 0, and one would get
dropped in `MediaSourceTrackGStreamer::filterOutRepeatingTracks`
To work around this, AppendPipeline will register any IDs it creates
with `MediaSourcePrivateGStreamer::m_trackIdRegistry`, which will
assign different IDs in case of a collision.
There has been some consideration as to where to place this registry,
and how to assign fallback IDs, which is both explained in comments.
One test that produced these collisions is:
imported/w3c/web-platform-tests/encrypted-media/clearkey-mp4-playback-temporary-waitingforkey.https.html
* Source/WebCore/platform/graphics/gstreamer/AudioTrackPrivateGStreamer.cpp:
(WebCore::AudioTrackPrivateGStreamer::AudioTrackPrivateGStreamer):
Add constructor with an explicit TrackID argument.
(WebCore::AudioTrackPrivateGStreamer::capsChanged):
* Source/WebCore/platform/graphics/gstreamer/AudioTrackPrivateGStreamer.h:
(WebCore::AudioTrackPrivateGStreamer::create):
* Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp:
(WebCore::getTrackIdFromPad): wrapper around gst_pad_get_stream_id, parses
stream-id into a TrackID
(WebCore::getTrackIdFromStream): ditto, wraps around
gst_stream_get_stream_id
(WebCore::trackIdFromString): parses a stream-id.
Can handle plain integers, the qtdemux-style ".../<ID>" format, and UUIDs.
* Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.h:
*
Source/WebCore/platform/graphics/gstreamer/InbandTextTrackPrivateGStreamer.cpp:
(WebCore::InbandTextTrackPrivateGStreamer::InbandTextTrackPrivateGStreamer):
Add constructor with an explicit TrackID argument.
*
Source/WebCore/platform/graphics/gstreamer/InbandTextTrackPrivateGStreamer.h:
(WebCore::InbandTextTrackPrivateGStreamer::create):
*
Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp:
(WebCore::MediaPlayerPrivateGStreamer::notifyPlayerOfTrack):
(WebCore::MediaPlayerPrivateGStreamer::handleTextSample):
(WebCore::MediaPlayerPrivateGStreamer::updateEnabledVideoTrack):
(WebCore::MediaPlayerPrivateGStreamer::updateEnabledAudioTrack):
(WebCore::MediaPlayerPrivateGStreamer::playbin3SendSelectStreamsIfAppropriate):
(WebCore::MediaPlayerPrivateGStreamer::updateTracks):
(WebCore::MediaPlayerPrivateGStreamer::setupCodecProbe):
(WebCore::MediaPlayerPrivateGStreamer::codecForStreamId):
* Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h:
* Source/WebCore/platform/graphics/gstreamer/TextSinkGStreamer.cpp:
(webkitTextSinkHandleSample):
* Source/WebCore/platform/graphics/gstreamer/TrackPrivateBaseGStreamer.cpp:
(WebCore::TrackPrivateBaseGStreamer::TrackPrivateBaseGStreamer):
Add constructor with an explicit TrackID argument.
Sets the new m_shouldUsePadStreamId field false, to avoid the ID being
changed elsewhere.
(WebCore::TrackPrivateBaseGStreamer::setPad):
(WebCore::TrackPrivateBaseGStreamer::notifyTrackOfStreamChanged):
(WebCore::TrackPrivateBaseGStreamer::installUpdateConfigurationHandlers):
(WebCore::TrackPrivateBaseGStreamer::updateTrackIDFromTags):
(WebCore::TrackPrivateBaseGStreamer::prefixForType): Deleted.
(WebCore::trimStreamId): Deleted.
(WebCore::TrackPrivateBaseGStreamer::generateUniquePlaybin2StreamID):
Deleted.
(WebCore::TrackPrivateBaseGStreamer::trackIdFromPadStreamStartOrUniqueID):
Deleted.
(WebCore::TrackPrivateBaseGStreamer::trackIdFromStringIdOrIndex): Deleted.
* Source/WebCore/platform/graphics/gstreamer/TrackPrivateBaseGStreamer.h:
(WebCore::TrackPrivateBaseGStreamer::streamId const):
Added, returns m_id. This is different to TrackPrivateBase::id(), since
its implementations first try to return the ID derived from the
container-specific-track-id tag if available.
(WebCore::TrackPrivateBaseGStreamer::gstStreamId const): Added.
(WebCore::TrackPrivateBaseGStreamer::capsChanged):
(WebCore::TrackPrivateBaseGStreamer::stringId const): Deleted.
- Removed m_stringId, instead using m_id everywhere.
- Added m_gstStreamId, storing the stream-id string as reported by
GStreamer.
* Source/WebCore/platform/graphics/gstreamer/VideoTrackPrivateGStreamer.cpp:
(WebCore::VideoTrackPrivateGStreamer::VideoTrackPrivateGStreamer):
(WebCore::VideoTrackPrivateGStreamer::capsChanged):
* Source/WebCore/platform/graphics/gstreamer/VideoTrackPrivateGStreamer.h:
Added constructor with an explicit TrackID argument.
* Source/WebCore/platform/graphics/gstreamer/mse/AppendPipeline.cpp:
(WebCore::AppendPipeline::~AppendPipeline):
(WebCore::AppendPipeline::didReceiveInitializationSegment):
(WebCore::createOptionalParserForFormat):
(WebCore::AppendPipeline::recycleTrackForPad):
(WebCore::AppendPipeline::linkPadWithTrack):
(WebCore::AppendPipeline::makeWebKitTrack):
Explicitly construct tracks with our IDs, as relying on
TrackPrivateBaseGStreamer to pick one up from stream-id risks collisions.
(WebCore::AppendPipeline::Track::emplaceOptionalParserForFormat):
(WebCore::AppendPipeline::generateTrackId): Deleted.
* Source/WebCore/platform/graphics/gstreamer/mse/AppendPipeline.h:
(WebCore::AppendPipeline::Track::Track):
(WebCore::AppendPipeline::tryCreateTrackFromPad):
Instead of generating an ID based on track type and index, parse stream-id,
and add it to the registry so potential ID collisions can be caught and
avoided.
*
Source/WebCore/platform/graphics/gstreamer/mse/MediaPlayerPrivateGStreamerMSE.cpp:
(WebCore::filterOutRepeatingTracks):
*
Source/WebCore/platform/graphics/gstreamer/mse/MediaSourcePrivateGStreamer.cpp:
(WebCore::MediaSourcePrivateGStreamer::registerTrackId): Added.
(WebCore::MediaSourcePrivateGStreamer::unregisterTrackId): Added.
*
Source/WebCore/platform/graphics/gstreamer/mse/MediaSourcePrivateGStreamer.h:
- Added m_trackIdRegistry, to be used by AppendPipeline.
*
Source/WebCore/platform/graphics/gstreamer/mse/MediaSourceTrackGStreamer.cpp:
(WebCore::MediaSourceTrackGStreamer::MediaSourceTrackGStreamer):
(WebCore::MediaSourceTrackGStreamer::create):
*
Source/WebCore/platform/graphics/gstreamer/mse/MediaSourceTrackGStreamer.h:
*
Source/WebCore/platform/graphics/gstreamer/mse/SourceBufferPrivateGStreamer.cpp:
(WebCore::SourceBufferPrivateGStreamer::flush):
(WebCore::SourceBufferPrivateGStreamer::allSamplesInTrackEnqueued):
(WebCore::SourceBufferPrivateGStreamer::precheckInitializationSegment):
(WebCore::SourceBufferPrivateGStreamer::tryRegisterTrackId):
Added, calls `MediaSourcePrivateGStreamer` to try to register an ID.
(WebCore::SourceBufferPrivateGStreamer::tryUnregisterTrackId): ditto
*
Source/WebCore/platform/graphics/gstreamer/mse/SourceBufferPrivateGStreamer.h:
* Source/WebCore/platform/graphics/gstreamer/mse/TrackQueue.cpp:
(WebCore::TrackQueue::TrackQueue):
(WebCore::TrackQueue::enqueueObject):
(WebCore::TrackQueue::clear):
(WebCore::TrackQueue::flush):
(WebCore::TrackQueue::notifyWhenLowLevel):
(WebCore::TrackQueue::pop):
(WebCore::TrackQueue::notifyWhenNotEmpty):
(WebCore::TrackQueue::resetNotEmptyHandler):
(WebCore::TrackQueue::checkLowLevel):
* Source/WebCore/platform/graphics/gstreamer/mse/TrackQueue.h:
*
Source/WebCore/platform/graphics/gstreamer/mse/WebKitMediaSourceGStreamer.cpp:
(WebKitMediaSrcPrivate::streamById): Renamed from streamByName.
(dumpPipeline):
(webKitMediaSrcEmitStreams):
(webKitMediaSrcTearDownStream):
(webKitMediaSrcLoop):
(webKitMediaSrcStreamFlush):
(webKitMediaSrcFlush):
(WebKitMediaSrcPrivate::streamByName): Deleted.
*
Source/WebCore/platform/graphics/gstreamer/mse/WebKitMediaSourceGStreamer.h:
Canonical link: https://commits.webkit.org/286527@main
Canonical link: https://commits.webkit.org/282416.320@webkitglib/2.46
Commit: f08fb2ed07a55e64d8b071e2267c42c6b4f8b163
https://github.com/WebKit/WebKit/commit/f08fb2ed07a55e64d8b071e2267c42c6b4f8b163
Author: Vivienne Watermeier <[email protected]>
Date: 2024-11-26 (Tue, 26 Nov 2024)
Changed paths:
M Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp
Log Message:
-----------
Cherry-pick 286598@main (8330471282e8).
https://bugs.webkit.org/show_bug.cgi?id=283104
[GStreamer] Fix webrtc/utf8-sdp.html regression
https://bugs.webkit.org/show_bug.cgi?id=283104
Reviewed by Philippe Normand.
getStreamIdFromStream() would sometimes returned nullopt in updateTracks(),
fixed by providing a fallback value
*
Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp:
(WebCore::MediaPlayerPrivateGStreamer::updateTracks):
Canonical link: https://commits.webkit.org/286598@main
Canonical link: https://commits.webkit.org/282416.321@webkitglib/2.46
Commit: c937f77e243f9dc2c774d69f60bc372eef48103d
https://github.com/WebKit/WebKit/commit/c937f77e243f9dc2c774d69f60bc372eef48103d
Author: Vivienne Watermeier <[email protected]>
Date: 2024-11-26 (Tue, 26 Nov 2024)
Changed paths:
M
Source/WebCore/platform/graphics/gstreamer/mse/MediaSourcePrivateGStreamer.h
Log Message:
-----------
Cherry-pick 286794@main (6ecba7116c9c).
https://bugs.webkit.org/show_bug.cgi?id=283352
[GTK] ASSERTION FAILED: !isHashTraitsEmptyValue<KeyTraits>(key)
https://bugs.webkit.org/show_bug.cgi?id=283352
Reviewed by Philippe Normand.
By default, HashSet disallows 0 as key, see:
https://github.com/WebKit/WebKit/blob/3261dcc32730cb8765e216fa45401618366a5fd2/Source/WTF/wtf/HashTraits.h#L107
Instead, use WTF::UnsignedWithZeroKeyHashTraits.
*
Source/WebCore/platform/graphics/gstreamer/mse/MediaSourcePrivateGStreamer.h:
Canonical link: https://commits.webkit.org/286794@main
Canonical link: https://commits.webkit.org/282416.322@webkitglib/2.46
Commit: 0c60a2686f3d429886ebb0edbee9e18658f93b68
https://github.com/WebKit/WebKit/commit/0c60a2686f3d429886ebb0edbee9e18658f93b68
Author: Vivienne Watermeier <[email protected]>
Date: 2024-11-26 (Tue, 26 Nov 2024)
Changed paths:
M Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp
Log Message:
-----------
Cherry-pick 286797@main (a127f8f44e5d).
https://bugs.webkit.org/show_bug.cgi?id=283357
[GStreamer] Fix hang in webaudio
https://bugs.webkit.org/show_bug.cgi?id=283357
Reviewed by Philippe Normand.
During a PAUSED_TO_READY state change, there is a race condition between
webKitWebAudioSrcRenderAndPushFrames setting dispatchDone true, and
the state change resetting it to false, so that the renderer thread
will block on dispatchCondition.
To fix this, the state transition no longer sets dispatchDone to false,
which is already done on every renderer thread entry.
Original author: Marcin Mielczarczyk <[email protected]>
See: https://github.com/WebPlatformForEmbedded/WPEWebKit/pull/1426
* Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:
(webKitWebAudioSrcChangeState):
Canonical link: https://commits.webkit.org/286797@main
Canonical link: https://commits.webkit.org/282416.323@webkitglib/2.46
Compare: https://github.com/WebKit/WebKit/compare/53e7f27d2622...0c60a2686f3d
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