mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-25 03:01:03 +00:00
84 lines
3.4 KiB
C
84 lines
3.4 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_DATA_CHANNEL_H__
|
||
|
|
#define __GST_WEBRTC_DATA_CHANNEL_H__
|
||
|
|
|
||
|
|
#include <gst/gst.h>
|
||
|
|
#include <gst/webrtc/webrtc_fwd.h>
|
||
|
|
#include <gst/webrtc/dtlstransport.h>
|
||
|
|
#include "sctptransport.h"
|
||
|
|
|
||
|
|
G_BEGIN_DECLS
|
||
|
|
|
||
|
|
GST_WEBRTC_API
|
||
|
|
GType gst_webrtc_data_channel_get_type(void);
|
||
|
|
#define GST_TYPE_WEBRTC_DATA_CHANNEL (gst_webrtc_data_channel_get_type())
|
||
|
|
#define GST_WEBRTC_DATA_CHANNEL(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_DATA_CHANNEL,GstWebRTCDataChannel))
|
||
|
|
#define GST_IS_WEBRTC_DATA_CHANNEL(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_DATA_CHANNEL))
|
||
|
|
#define GST_WEBRTC_DATA_CHANNEL_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_DATA_CHANNEL,GstWebRTCDataChannelClass))
|
||
|
|
#define GST_IS_WEBRTC_DATA_CHANNEL_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_DATA_CHANNEL))
|
||
|
|
#define GST_WEBRTC_DATA_CHANNEL_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_DATA_CHANNEL,GstWebRTCDataChannelClass))
|
||
|
|
|
||
|
|
typedef struct _GstWebRTCDataChannel GstWebRTCDataChannel;
|
||
|
|
typedef struct _GstWebRTCDataChannelClass GstWebRTCDataChannelClass;
|
||
|
|
|
||
|
|
struct _GstWebRTCDataChannel
|
||
|
|
{
|
||
|
|
GstObject parent;
|
||
|
|
|
||
|
|
GstWebRTCSCTPTransport *sctp_transport;
|
||
|
|
GstElement *appsrc;
|
||
|
|
GstElement *appsink;
|
||
|
|
|
||
|
|
gchar *label;
|
||
|
|
gboolean ordered;
|
||
|
|
guint max_packet_lifetime;
|
||
|
|
guint max_retransmits;
|
||
|
|
gchar *protocol;
|
||
|
|
gboolean negotiated;
|
||
|
|
gint id;
|
||
|
|
GstWebRTCPriorityType priority;
|
||
|
|
GstWebRTCDataChannelState ready_state;
|
||
|
|
guint64 buffered_amount;
|
||
|
|
guint64 buffered_amount_low_threshold;
|
||
|
|
|
||
|
|
GstWebRTCBin *webrtcbin;
|
||
|
|
gboolean opened;
|
||
|
|
gulong src_probe;
|
||
|
|
GError *stored_error;
|
||
|
|
|
||
|
|
gpointer _padding[GST_PADDING];
|
||
|
|
};
|
||
|
|
|
||
|
|
struct _GstWebRTCDataChannelClass
|
||
|
|
{
|
||
|
|
GstObjectClass parent_class;
|
||
|
|
|
||
|
|
gpointer _padding[GST_PADDING];
|
||
|
|
};
|
||
|
|
|
||
|
|
void gst_webrtc_data_channel_start_negotiation (GstWebRTCDataChannel *channel);
|
||
|
|
void gst_webrtc_data_channel_set_sctp_transport (GstWebRTCDataChannel *channel,
|
||
|
|
GstWebRTCSCTPTransport *sctp);
|
||
|
|
|
||
|
|
G_END_DECLS
|
||
|
|
|
||
|
|
#endif /* __GST_WEBRTC_DATA_CHANNEL_H__ */
|