Branch: refs/heads/main
Home: https://github.com/WebKit/WebKit
Commit: 3f3c280f8192b7585a204bbcb7f2db49deace035
https://github.com/WebKit/WebKit/commit/3f3c280f8192b7585a204bbcb7f2db49deace035
Author: Carlos Bentzen <cadubent...@igalia.com>
Date: 2024-06-22 (Sat, 22 Jun 2024)
Changed paths:
M
LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-answer-expected.txt
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.cpp
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.h
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.cpp
Log Message:
-----------
[GStreamer][WebRTC] Caps from SDP and RealtimeOutgoingAudioSourceGStreamer
aren't aligned
https://bugs.webkit.org/show_bug.cgi?id=275740
Reviewed by Philippe Normand.
Align codec-preferences of webrtcbin's transceivers with caps from
RealtimeOutgoingAudioSourceGStreamer by setting vad=on in the
ssrc-audio-level extension.
*
LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-answer-expected.txt:
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.cpp:
(WebCore::capsFromSDPMedia):
(WebCore::setSsrcAudioLevelVadOn):
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.h:
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.cpp:
(WebCore::RealtimeOutgoingAudioSourceGStreamer::setPayloadType):
Canonical link: https://commits.webkit.org/280273@main
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