/* GStreamer * Copyright (C) 2018 Matthew Waters <matthew@centricular.com> * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #ifndef __WEBRTC_SCTP_TRANSPORT_H__ #define __WEBRTC_SCTP_TRANSPORT_H__ #include <gst/gst.h> #include <gst/webrtc/webrtc.h> #include <gst/webrtc/sctptransport.h> #include "gstwebrtcice.h" #include "gst/webrtc/webrtc-priv.h" G_BEGIN_DECLS GType webrtc_sctp_transport_get_type(void); #define TYPE_WEBRTC_SCTP_TRANSPORT (webrtc_sctp_transport_get_type()) #define WEBRTC_SCTP_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),TYPE_WEBRTC_SCTP_TRANSPORT,WebRTCSCTPTransport)) #define WEBRTC_IS_SCTP_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),TYPE_WEBRTC_SCTP_TRANSPORT)) #define WEBRTC_SCTP_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,TYPE_WEBRTC_SCTP_TRANSPORT,WebRTCSCTPTransportClass)) #define WEBRTC_SCTP_IS_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,TYPE_WEBRTC_SCTP_TRANSPORT)) #define WEBRTC_SCTP_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,TYPE_WEBRTC_SCTP_TRANSPORT,WebRTCSCTPTransportClass)) typedef struct _WebRTCSCTPTransport WebRTCSCTPTransport; typedef struct _WebRTCSCTPTransportClass WebRTCSCTPTransportClass; struct _WebRTCSCTPTransport { GstWebRTCSCTPTransport parent; GstWebRTCDTLSTransport *transport; GstWebRTCSCTPTransportState state; guint64 max_message_size; guint max_channels; gboolean association_established; gulong sctpdec_block_id; GstElement *sctpdec; GstElement *sctpenc; GstWebRTCBin *webrtcbin; }; struct _WebRTCSCTPTransportClass { GstWebRTCSCTPTransportClass parent_class; }; WebRTCSCTPTransport * webrtc_sctp_transport_new (void); void webrtc_sctp_transport_set_priority (WebRTCSCTPTransport *sctp, GstWebRTCPriorityType priority); G_END_DECLS #endif /* __WEBRTC_SCTP_TRANSPORT_H__ */