Branch: refs/heads/main
Home: https://github.com/WebKit/WebKit
Commit: ad280f2c9623b9c5f88d513578b6c5bec8db33a8
https://github.com/WebKit/WebKit/commit/ad280f2c9623b9c5f88d513578b6c5bec8db33a8
Author: Philippe Normand <[email protected]>
Date: 2026-09-17 (Thu, 17 Sep 2026)
Changed paths:
M LayoutTests/platform/glib/TestExpectations
M
Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp
M
Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.h
Log Message:
-----------
[GTK][WPE] webrtc/audio-replace-track-to-webaudio.html crashes
https://bugs.webkit.org/show_bug.cgi?id=324340
Reviewed by Xabier Rodriguez-Calvar.
The crash was happening when attempting to grow a Vector with a new size
inferior to the current
one, so use the resize method instead. Then another flaky crash showed up, when
the source was
reconfigured the sample converter was cleared, introducing crashes in the
signaling thread. The
converter is now protected by a lock because it is used from multiple threads.
* LayoutTests/platform/glib/TestExpectations:
*
Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp:
(WebCore::RealtimeOutgoingAudioSourceLibWebRTC::audioSamplesAvailable):
(WebCore::RealtimeOutgoingAudioSourceLibWebRTC::gstAudioConverterInputFramesForOutput):
(WebCore::RealtimeOutgoingAudioSourceLibWebRTC::pullAudioData):
(WebCore::gstAudioConverterInputFramesForOutput): Deleted.
*
Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.h:
Canonical link: https://commits.webkit.org/321317@main
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