gstreamer/gst-libs/gst/webrtc/datachannel.h

115 lines
4.2 KiB
C

/* GStreamer
* Copyright (C) 2018 Matthew Waters <matthew@centricular.com>
* Copyright (C) 2020 Sebastian Dröge <sebastian@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>
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))
#define GST_WEBRTC_DATA_CHANNEL_LOCK(channel) g_mutex_lock(&((GstWebRTCDataChannel *)(channel))->lock)
#define GST_WEBRTC_DATA_CHANNEL_UNLOCK(channel) g_mutex_unlock(&((GstWebRTCDataChannel *)(channel))->lock)
/**
* GstWebRTCDataChannel:
*
* Since: 1.18
*/
struct _GstWebRTCDataChannel
{
GObject parent;
GMutex lock;
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;
gpointer _padding[GST_PADDING];
};
/**
* GstWebRTCDataChannelClass:
*
* Since: 1.18
*/
struct _GstWebRTCDataChannelClass
{
GObjectClass parent_class;
void (*send_data) (GstWebRTCDataChannel * channel, GBytes *data);
void (*send_string) (GstWebRTCDataChannel * channel, const gchar *str);
void (*close) (GstWebRTCDataChannel * channel);
gpointer _padding[GST_PADDING];
};
GST_WEBRTC_API
void gst_webrtc_data_channel_on_open (GstWebRTCDataChannel * channel);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_close (GstWebRTCDataChannel * channel);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_error (GstWebRTCDataChannel * channel, GError * error);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_message_data (GstWebRTCDataChannel * channel, GBytes * data);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_message_string (GstWebRTCDataChannel * channel, const gchar * str);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_buffered_amount_low (GstWebRTCDataChannel * channel);
GST_WEBRTC_API
void gst_webrtc_data_channel_send_data (GstWebRTCDataChannel * channel, GBytes * data);
GST_WEBRTC_API
void gst_webrtc_data_channel_send_string (GstWebRTCDataChannel * channel, const gchar * str);
GST_WEBRTC_API
void gst_webrtc_data_channel_close (GstWebRTCDataChannel * channel);
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCDataChannel, g_object_unref)
G_END_DECLS
#endif /* __GST_WEBRTC_DATA_CHANNEL_H__ */