gstreamer/ext/webrtc/sctptransport.h
Olivier Crête 80a56c25a6 webrtc: Set the DSCP markings based on the priority
This matches how the WebRTC javascript API works and the Chrome implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
2020-10-30 16:24:40 -04:00

70 lines
2.6 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;
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__ */