mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-22 17:51:16 +00:00
74 lines
2.7 KiB
C
74 lines
2.7 KiB
C
/* 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 "fwd.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__ */
|