mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-13 21:01:14 +00:00
b42d98ca19
There's no reason for it to inherit from GstObject apart from locking, which is easily replaced, and inheriting from GInitiallyUnowned made introspection awkward and needlessly complicated.
85 lines
3.4 KiB
C
85 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
|
|
{
|
|
GObject 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;
|
|
GMutex lock;
|
|
|
|
gpointer _padding[GST_PADDING];
|
|
};
|
|
|
|
struct _GstWebRTCDataChannelClass
|
|
{
|
|
GObjectClass parent_class;
|
|
|
|
gpointer _padding[GST_PADDING];
|
|
};
|
|
|
|
void gst_webrtc_data_channel_start_negotiation (GstWebRTCDataChannel *channel);
|
|
G_GNUC_INTERNAL
|
|
void gst_webrtc_data_channel_link_to_sctp (GstWebRTCDataChannel *channel,
|
|
GstWebRTCSCTPTransport *sctp_transport);
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_WEBRTC_DATA_CHANNEL_H__ */
|