Branch: refs/heads/main
Home: https://github.com/WebKit/WebKit
Commit: 921d8dd036b83497f2748bec96ad57604d50ac3d
https://github.com/WebKit/WebKit/commit/921d8dd036b83497f2748bec96ad57604d50ac3d
Author: Philippe Normand <[email protected]>
Date: 2024-04-10 (Wed, 10 Apr 2024)
Changed paths:
M
LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-answer-expected.txt
M
LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-offer-expected.txt
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp
M Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.cpp
M Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.h
Log Message:
-----------
[GStreamer][WebRTC] RTP extensions advertizing clean-up
https://bugs.webkit.org/show_bug.cgi?id=272389
Reviewed by Xabier Rodriguez-Calvar.
The sdes:repaired-rtp-stream-id and sdes:rtp-stream-id RTP header extensions
are now listed in SDP
audio sections. The registry scanner code handling this was refactored a bit to
avoid code
duplication in the various extension identifiers.
*
LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-answer-expected.txt:
*
LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-offer-expected.txt:
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp:
(WebCore::fetchDescription):
* Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.cpp:
(WebCore::GStreamerRegistryScanner::fillAudioRtpCapabilities):
(WebCore::GStreamerRegistryScanner::fillVideoRtpCapabilities):
* Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.h:
Canonical link: https://commits.webkit.org/277295@main
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