[webkit-changes] [WebKit/WebKit] 157122: Cherry-pick 282780 at main (d90a119ac846). https://bu...

[Philippe Normand]

  Branch: refs/heads/webkitglib/2.46
  Home:   https://github.com/WebKit/WebKit
  Commit: 157122f4560d9bf97c8c97b253ea4b6179aa2d8b
      https://github.com/WebKit/WebKit/commit/157122f4560d9bf97c8c97b253ea4b6179aa2d8b
  Author: Philippe Normand <philn at igalia.com>
  Date:   2024-09-02 (Mon, 02 Sep 2024)

  Changed paths:
    M Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp

  Log Message:
  -----------
  Cherry-pick 282780 at main (d90a119ac846). https://bugs.webkit.org/show_bug.cgi?id=278705

    [GStreamer][WebRTC] Fix getStats() call leading to crash
    https://bugs.webkit.org/show_bug.cgi?id=278705

    Reviewed by Xabier Rodriguez-Calvar.

    Check the GstBin associated to the receiver's RealtimeIncomingSource is not null before attempting
    to get its sink pad. This fixes crashes in the webrtc/video-stats.html and
    imported/w3c/web-platform-tests/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats.html tests.

    * Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp:
    (WebCore::GStreamerMediaEndpoint::getStats):

    Canonical link: https://commits.webkit.org/282780@main


  Commit: 2a93dd31c1d827b20497d017d70d42fe89d1d252
      https://github.com/WebKit/WebKit/commit/2a93dd31c1d827b20497d017d70d42fe89d1d252
  Author: Philippe Normand <philn at igalia.com>
  Date:   2024-09-02 (Mon, 02 Sep 2024)

  Changed paths:
    M Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp
    M Tools/TestWebKitAPI/Tests/WebCore/gstreamer/GStreamerTest.cpp

  Log Message:
  -----------
  Cherry-pick 282782 at main (afb9961648af). https://bugs.webkit.org/show_bug.cgi?id=278706

    [GStreamer] Support for int64 structure field serialization to JSON
    https://bugs.webkit.org/show_bug.cgi?id=278706

    Reviewed by Xabier Rodriguez-Calvar.

    Also driving-by, fixing up support for WebRTCStats fields which was broken.

    * Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp:
    (WebCore::gstStructureValueToJSON):
    * Tools/TestWebKitAPI/Tests/WebCore/gstreamer/GStreamerTest.cpp:
    (TestWebKitAPI::TEST_F(GStreamerTest, gstStructureJSONSerializing)):

    Canonical link: https://commits.webkit.org/282782@main


  Commit: 621617204a3e4f9b048781d8e35693592587096c
      https://github.com/WebKit/WebKit/commit/621617204a3e4f9b048781d8e35693592587096c
  Author: Philippe Normand <philn at igalia.com>
  Date:   2024-09-02 (Mon, 02 Sep 2024)

  Changed paths:
    M Source/WebCore/platform/mediastream/gstreamer/RealtimeIncomingSourceGStreamer.cpp

  Log Message:
  -----------
  Cherry-pick 282783 at main (2067d74636c7). https://bugs.webkit.org/show_bug.cgi?id=278707

    [GStreamer][WebRTC] Remove a couple early returns in RealtimeIncomingSourceGStreamer
    https://bugs.webkit.org/show_bug.cgi?id=278707

    Reviewed by Xabier Rodriguez-Calvar.

    Replace early returns with release asserts.

    * Source/WebCore/platform/mediastream/gstreamer/RealtimeIncomingSourceGStreamer.cpp:
    (WebCore::RealtimeIncomingSourceGStreamer::handleUpstreamEvent):
    (WebCore::RealtimeIncomingSourceGStreamer::handleUpstreamQuery):

    Canonical link: https://commits.webkit.org/282783@main


  Commit: 8eee825cf48ef2e1b0e21c076e3acbc1a864ed72
      https://github.com/WebKit/WebKit/commit/8eee825cf48ef2e1b0e21c076e3acbc1a864ed72
  Author: Philippe Normand <philn at igalia.com>
  Date:   2024-09-02 (Mon, 02 Sep 2024)

  Changed paths:
    M Source/WebCore/platform/audio/FFTFrame.h
    M Source/WebCore/platform/audio/gstreamer/FFTFrameGStreamer.cpp
    M Source/WebCore/platform/graphics/gstreamer/GUniquePtrGStreamer.h

  Log Message:
  -----------
  Cherry-pick 282836 at main (a315a6d8a294). https://bugs.webkit.org/show_bug.cgi?id=278712

    [GStreamer][WebAudio] Convert GstFFT raw pointers to GUniquePtrs
    https://bugs.webkit.org/show_bug.cgi?id=278712

    Reviewed by Xabier Rodriguez-Calvar.

    * Source/WebCore/platform/audio/FFTFrame.h:
    * Source/WebCore/platform/audio/gstreamer/FFTFrameGStreamer.cpp:
    (WebCore::FFTFrame::FFTFrame):
    (WebCore::FFTFrame::doFFT):
    (WebCore::FFTFrame::doInverseFFT):
    (WebCore::FFTFrame::~FFTFrame): Deleted.
    * Source/WebCore/platform/graphics/gstreamer/GUniquePtrGStreamer.h:

    Canonical link: https://commits.webkit.org/282836@main


  Commit: 0250fbfe2f87e04bd3ab81ed41a8ae961ba8820f
      https://github.com/WebKit/WebKit/commit/0250fbfe2f87e04bd3ab81ed41a8ae961ba8820f
  Author: Philippe Normand <philn at igalia.com>
  Date:   2024-09-02 (Mon, 02 Sep 2024)

  Changed paths:
    M Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.cpp
    M Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.h

  Log Message:
  -----------
  Cherry-pick 282898 at main (69a99fe5db4f). https://bugs.webkit.org/show_bug.cgi?id=278792

    [GStreamer][WebRTC] Initialize all audio/video RTP header extension capabilities
    https://bugs.webkit.org/show_bug.cgi?id=278792

    Reviewed by Xabier Rodriguez-Calvar.

    In some cases a newly added audio transceiver would induce an SDP offer including only audio RTP
    header extensions, without the common extensions. Ditto for video transceivers.

    Driving-by, make use of ASCIILiteral in more places in the registry scanner.

    * Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.cpp:
    (WebCore::probeRtpExtensions):
    (WebCore::GStreamerRegistryScanner::fillAudioRtpCapabilities):
    (WebCore::GStreamerRegistryScanner::fillVideoRtpCapabilities):
    (WebCore::GStreamerRegistryScanner::audioRtpExtensions):
    (WebCore::GStreamerRegistryScanner::videoRtpExtensions):
    * Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.h:

    Canonical link: https://commits.webkit.org/282898@main


Compare: https://github.com/WebKit/WebKit/compare/ad990fb0b458...0250fbfe2f87

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