GstClockTime gst_planar_audio_adapter_prev_dts (GstPlanarAudioAdapter * adapter,
guint64 * distance);
-#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstPlanarAudioAdapter, gst_object_unref)
-#endif
G_END_DECLS
GstInsertBinCallback callback, gpointer user_data);
-#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstInsertBin, gst_object_unref)
-#endif
G_END_DECLS
#endif /* __GST_INSERT_BIN_H__ */
void gst_webrtc_dtls_transport_set_transport (GstWebRTCDTLSTransport * transport,
GstWebRTCICETransport * ice);
-#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCDTLSTransport, gst_object_unref)
-#endif
G_END_DECLS
GST_WEBRTC_API
void gst_webrtc_ice_transport_new_candidate (GstWebRTCICETransport * ice, guint stream_id, GstWebRTCICEComponent component, gchar * attr);
-#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCICETransport, gst_object_unref)
-#endif
G_END_DECLS
void gst_webrtc_session_description_free (GstWebRTCSessionDescription * desc);
-#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCSessionDescription, gst_webrtc_session_description_free)
-#endif
G_END_DECLS
void gst_webrtc_rtp_receiver_set_rtcp_transport (GstWebRTCRTPReceiver * receiver,
GstWebRTCDTLSTransport * transport);
-#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCRTPReceiver, gst_object_unref)
-#endif
G_END_DECLS
GstWebRTCDTLSTransport * transport);
-#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCRTPSender, gst_object_unref)
-#endif
G_END_DECLS
gpointer _padding[GST_PADDING-1];
};
-#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCRTPTransceiver, gst_object_unref)
-#endif
GST_WEBRTC_API
void gst_webrtc_rtp_transceiver_set_direction (GstWebRTCRTPTransceiver * trans,