According to the GStreamer checks, the required GStreamer version is 1.18.4 (with 1.20 for USE_GSTREAMER_WEBRTC, ENABLE_MEDIA_RECORDER and USE_GSTREAMER_TRANSCODER). But the build enabling libwebrtc fails due to the usage of gst_buffer_new_memdup in Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingAudioSourceLibWebRTC.cpp, which is not available on 1.18.
in downstream wpe-2.38 this is fixed by commit 5f7400aee390ab9d964a0f2ede34f2008e75eaf8
(In reply to Philippe Normand from comment #1) > in downstream wpe-2.38 this is fixed by commit > 5f7400aee390ab9d964a0f2ede34f2008e75eaf8 Shouldn't this upstreamed? Or are we keeping this downstream?
It's fine to upstream... Wasn't needed until now because the SDK ships a recent version, and I suppose you're not using it.
Pull request: https://github.com/WebKit/WebKit/pull/29979
Committed 280247@main (1b86aba5013c): <https://commits.webkit.org/280247@main> Reviewed commits have been landed. Closing PR #29979 and removing active labels.