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webrtc: fix recursive G_BEGIN_DECLS and include missing sctptransport.h in webrtc.h

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8470>
This commit is contained in:
wbartel 2025-02-12 16:51:24 +01:00 committed by GStreamer Marge Bot
parent c7367addb5
commit 41ff7727dc
3 changed files with 2 additions and 2 deletions
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1
subprojects/gst-plugins-bad/gst-libs/gst/webrtc/webrtc-priv.h
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@ -22,6 +22,7 @@
#include <gst/gst.h>
#include <gst/webrtc/webrtc_fwd.h>
#include <gst/webrtc/webrtc-enumtypes.h>
#include <gst/webrtc/rtpsender.h>
#include <gst/webrtc/rtpreceiver.h>

1
subprojects/gst-plugins-bad/gst-libs/gst/webrtc/webrtc.h
View file

@ -31,6 +31,7 @@
#include <gst/webrtc/rtpreceiver.h>
#include <gst/webrtc/rtpsender.h>
#include <gst/webrtc/rtptransceiver.h>
#include <gst/webrtc/sctptransport.h>
#include <gst/webrtc/datachannel.h>
#endif /* __GST_WEBRTC_WEBRTC_H__ */

2
subprojects/gst-plugins-bad/gst-libs/gst/webrtc/webrtc_fwd.h
View file

@ -56,8 +56,6 @@ G_BEGIN_DECLS
#define GST_WEBRTC_DEPRECATED_FOR(f) G_DEPRECATED_FOR(f) GST_WEBRTC_API
#endif
#include <gst/webrtc/webrtc-enumtypes.h>
/**
* GstWebRTCDTLSTransport:
*/