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84 lines
3.4 KiB
C
84 lines
3.4 KiB
C
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/* GStreamer
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* Copyright (C) 2018 Matthew Waters <matthew@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef __GST_WEBRTC_DATA_CHANNEL_H__
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#define __GST_WEBRTC_DATA_CHANNEL_H__
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#include <gst/gst.h>
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#include <gst/webrtc/webrtc_fwd.h>
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#include <gst/webrtc/dtlstransport.h>
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#include "sctptransport.h"
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G_BEGIN_DECLS
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GST_WEBRTC_API
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GType gst_webrtc_data_channel_get_type(void);
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#define GST_TYPE_WEBRTC_DATA_CHANNEL (gst_webrtc_data_channel_get_type())
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#define GST_WEBRTC_DATA_CHANNEL(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_DATA_CHANNEL,GstWebRTCDataChannel))
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#define GST_IS_WEBRTC_DATA_CHANNEL(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_DATA_CHANNEL))
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#define GST_WEBRTC_DATA_CHANNEL_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_DATA_CHANNEL,GstWebRTCDataChannelClass))
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#define GST_IS_WEBRTC_DATA_CHANNEL_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_DATA_CHANNEL))
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#define GST_WEBRTC_DATA_CHANNEL_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_DATA_CHANNEL,GstWebRTCDataChannelClass))
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typedef struct _GstWebRTCDataChannel GstWebRTCDataChannel;
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typedef struct _GstWebRTCDataChannelClass GstWebRTCDataChannelClass;
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struct _GstWebRTCDataChannel
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{
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GstObject parent;
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GstWebRTCSCTPTransport *sctp_transport;
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GstElement *appsrc;
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GstElement *appsink;
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gchar *label;
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gboolean ordered;
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guint max_packet_lifetime;
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guint max_retransmits;
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gchar *protocol;
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gboolean negotiated;
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gint id;
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GstWebRTCPriorityType priority;
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GstWebRTCDataChannelState ready_state;
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guint64 buffered_amount;
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guint64 buffered_amount_low_threshold;
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GstWebRTCBin *webrtcbin;
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gboolean opened;
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gulong src_probe;
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GError *stored_error;
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gpointer _padding[GST_PADDING];
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};
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struct _GstWebRTCDataChannelClass
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{
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GstObjectClass parent_class;
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gpointer _padding[GST_PADDING];
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};
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void gst_webrtc_data_channel_start_negotiation (GstWebRTCDataChannel *channel);
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void gst_webrtc_data_channel_set_sctp_transport (GstWebRTCDataChannel *channel,
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GstWebRTCSCTPTransport *sctp);
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G_END_DECLS
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#endif /* __GST_WEBRTC_DATA_CHANNEL_H__ */
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