/* 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 __GST_WEBRTC_SCTP_TRANSPORT_H__
#define __GST_WEBRTC_SCTP_TRANSPORT_H__

#include <gst/gst.h>
/* libnice */
#include <agent.h>
#include <gst/webrtc/webrtc.h>
#include "gstwebrtcice.h"

G_BEGIN_DECLS

GType gst_webrtc_sctp_transport_get_type(void);
#define GST_TYPE_WEBRTC_SCTP_TRANSPORT            (gst_webrtc_sctp_transport_get_type())
#define GST_WEBRTC_SCTP_TRANSPORT(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransport))
#define GST_IS_WEBRTC_SCTP_TRANSPORT(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_SCTP_TRANSPORT))
#define GST_WEBRTC_SCTP_TRANSPORT_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransportClass))
#define GST_IS_WEBRTC_SCTP_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT))
#define GST_WEBRTC_SCTP_TRANSPORT_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransportClass))

struct _GstWebRTCSCTPTransport
{
  GstObject                     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 _GstWebRTCSCTPTransportClass
{
  GstObjectClass                parent_class;
};

GstWebRTCSCTPTransport *    gst_webrtc_sctp_transport_new               (void);

void
gst_webrtc_sctp_transport_set_priority (GstWebRTCSCTPTransport *sctp,
                                        GstWebRTCPriorityType priority);

G_END_DECLS

#endif /* __GST_WEBRTC_SCTP_TRANSPORT_H__ */