Branch: refs/heads/main
Home: https://github.com/WebKit/WebKit
Commit: 093e1078e7b60bb5388404e0f4651c97cbaf9113
https://github.com/WebKit/WebKit/commit/093e1078e7b60bb5388404e0f4651c97cbaf9113
Author: Philippe Normand <[email protected]>
Date: 2023-06-22 (Thu, 22 Jun 2023)
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
LayoutTests/platform/glib/imported/w3c/web-platform-tests/media-capabilities/decodingInfo.webrtc-expected.txt
M
LayoutTests/platform/glib/imported/w3c/web-platform-tests/media-capabilities/encodingInfo.webrtc-expected.txt
M Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.cpp
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.cpp
Log Message:
-----------
[GStreamer][WebRTC] Remove iSAC codec support
https://bugs.webkit.org/show_bug.cgi?id=258345
Reviewed by Xabier Rodriguez-Calvar.
This codec was removed from libwebrtc and for a long time already opus is the
recommended alternative.
*
LayoutTests/platform/glib/imported/w3c/web-platform-tests/media-capabilities/decodingInfo.webrtc-expected.txt:
*
LayoutTests/platform/glib/imported/w3c/web-platform-tests/media-capabilities/encodingInfo.webrtc-expected.txt:
* Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.cpp:
(WebCore::GStreamerRegistryScanner::fillAudioRtpCapabilities):
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.cpp:
(WebCore::RealtimeOutgoingAudioSourceGStreamer::setPayloadType):
Canonical link: https://commits.webkit.org/265396@main
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