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webrtc: fix documentation error in GstWebRTCKind
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7407>
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2 changed files with 2 additions and 2 deletions
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@ -2327,7 +2327,7 @@ for more information.</doc>
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<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/webrtc/webrtc_fwd.h">Kind is audio</doc>
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</member>
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<member name="video" value="2" c:identifier="GST_WEBRTC_KIND_VIDEO" glib:nick="video">
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<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/webrtc/webrtc_fwd.h">Kind is audio</doc>
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<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/webrtc/webrtc_fwd.h">Kind is video</doc>
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</member>
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</enumeration>
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<enumeration name="WebRTCPeerConnectionState" glib:type-name="GstWebRTCPeerConnectionState" glib:get-type="gst_webrtc_peer_connection_state_get_type" c:type="GstWebRTCPeerConnectionState">
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@ -449,7 +449,7 @@ typedef enum /*<underscore_name=gst_webrtc_ice_transport_policy>*/
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* GstWebRTCKind:
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* @GST_WEBRTC_KIND_UNKNOWN: Kind has not yet been set
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* @GST_WEBRTC_KIND_AUDIO: Kind is audio
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* @GST_WEBRTC_KIND_VIDEO: Kind is audio
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* @GST_WEBRTC_KIND_VIDEO: Kind is video
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*
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* https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind
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*
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