gstreamer/ext/webrtc/webrtcdatachannel.h

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/* 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__ */