Branch: refs/heads/main
Home: https://github.com/WebKit/WebKit
Commit: c32395413d947c28aed871d04c9cdf9536d7f0d5
https://github.com/WebKit/WebKit/commit/c32395413d947c28aed871d04c9cdf9536d7f0d5
Author: Philippe Normand <[email protected]>
Date: 2026-08-27 (Thu, 27 Aug 2026)
Changed paths:
M LayoutTests/platform/glib/TestExpectations
M
Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp
Log Message:
-----------
[GStreamer]
imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-connectionState.https.html
crashes
https://bugs.webkit.org/show_bug.cgi?id=320653
Reviewed by Adrian Perez de Castro.
The audio converted created in the outgoing audio source expects its input
buffer to be writable
because it is created with the GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE flag. So
when doing actual audio
conversion the input buffer has to be mapped in read-write mode.
* LayoutTests/platform/glib/TestExpectations:
*
Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp:
(WebCore::RealtimeOutgoingAudioSourceLibWebRTC::pullAudioData):
Canonical link: https://commits.webkit.org/319995@main
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