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/* 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__ */
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