mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-08-01 19:25:10 +00:00
07e9374eff
Mostly follows the W3C specification https://www.w3.org/TR/webrtc/#peer-to-peer-data-api With contributions from: Mathieu Duponchelle <mathieu@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=794351
66 lines
2.4 KiB
C
66 lines
2.4 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 __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;
|
|
|
|
GstElement *sctpdec;
|
|
GstElement *sctpenc;
|
|
|
|
GstWebRTCBin *webrtcbin;
|
|
};
|
|
|
|
struct _GstWebRTCSCTPTransportClass
|
|
{
|
|
GstObjectClass parent_class;
|
|
};
|
|
|
|
GstWebRTCSCTPTransport * gst_webrtc_sctp_transport_new (void);
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_WEBRTC_SCTP_TRANSPORT_H__ */
|