Branch: refs/heads/main
Home: https://github.com/WebKit/WebKit
Commit: f309ba2ca43119c4e558a80f5883fdc19307c8b1
https://github.com/WebKit/WebKit/commit/f309ba2ca43119c4e558a80f5883fdc19307c8b1
Author: Philippe Normand <ph...@igalia.com>
Date: 2024-07-03 (Wed, 03 Jul 2024)
Changed paths:
M Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.cpp
M Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.h
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.cpp
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.cpp
Log Message:
-----------
[GStreamer][WebRTC] Support for RTP2 payloaders
https://bugs.webkit.org/show_bug.cgi?id=276115
Reviewed by Xabier Rodriguez-Calvar.
By relying on the registry scanner for payloader selection, the one with higher
rank will be used,
so by setting GST_PLUGIN_FEATURE_RANK=rtpopuspay2:max (or similar), the
selection can be influenced.
* Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.cpp:
(WebCore::GStreamerRegistryScanner::isRtpPacketizerSupported):
* Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.h:
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.cpp:
(WebCore::RealtimeOutgoingAudioSourceGStreamer::setPayloadType):
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.cpp:
(WebCore::RealtimeOutgoingVideoSourceGStreamer::setPayloadType):
Canonical link: https://commits.webkit.org/280619@main
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