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80a56c25a6
This matches how the WebRTC javascript API works and the Chrome implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
71 lines
2.6 KiB
C
71 lines
2.6 KiB
C
/* 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_SCTP_TRANSPORT_H__
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#define __GST_WEBRTC_SCTP_TRANSPORT_H__
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#include <gst/gst.h>
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/* libnice */
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#include <agent.h>
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#include <gst/webrtc/webrtc.h>
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#include "gstwebrtcice.h"
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G_BEGIN_DECLS
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GType gst_webrtc_sctp_transport_get_type(void);
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#define GST_TYPE_WEBRTC_SCTP_TRANSPORT (gst_webrtc_sctp_transport_get_type())
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#define GST_WEBRTC_SCTP_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransport))
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#define GST_IS_WEBRTC_SCTP_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_SCTP_TRANSPORT))
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#define GST_WEBRTC_SCTP_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransportClass))
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#define GST_IS_WEBRTC_SCTP_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT))
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#define GST_WEBRTC_SCTP_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransportClass))
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struct _GstWebRTCSCTPTransport
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{
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GstObject parent;
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GstWebRTCDTLSTransport *transport;
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GstWebRTCSCTPTransportState state;
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guint64 max_message_size;
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guint max_channels;
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gboolean association_established;
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gulong sctpdec_block_id;
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GstElement *sctpdec;
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GstElement *sctpenc;
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GstWebRTCBin *webrtcbin;
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};
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struct _GstWebRTCSCTPTransportClass
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{
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GstObjectClass parent_class;
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};
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GstWebRTCSCTPTransport * gst_webrtc_sctp_transport_new (void);
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void
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gst_webrtc_sctp_transport_set_priority (GstWebRTCSCTPTransport *sctp,
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GstWebRTCPriorityType priority);
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G_END_DECLS
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#endif /* __GST_WEBRTC_SCTP_TRANSPORT_H__ */
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