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1 /* GStreamer
2  * Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
3  *
4  * This library is free software; you can redistribute it and/or
5  * modify it under the terms of the GNU Library General Public
6  * License as published by the Free Software Foundation; either
7  * version 2 of the License, or (at your option) any later version.
8  *
9  * This library is distributed in the hope that it will be useful,
10  * but WITHOUT ANY WARRANTY; without even the implied warranty of
11  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
12  * Library General Public License for more details.
13  *
14  * You should have received a copy of the GNU Library General Public
15  * License along with this library; if not, write to the
16  * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17  * Boston, MA 02110-1301, USA.
18  */
19 
20 #ifndef __GST_WEBRTC_SESSION_DESCRIPTION_H__
21 #define __GST_WEBRTC_SESSION_DESCRIPTION_H__
22 
23 #include <gst/gst.h>
24 #include <gst/sdp/sdp.h>
25 #include <gst/webrtc/webrtc_fwd.h>
26 
27 G_BEGIN_DECLS
28 
29 GST_WEBRTC_API
30 const gchar *       gst_webrtc_sdp_type_to_string (GstWebRTCSDPType type);
31 
32 #define GST_TYPE_WEBRTC_SESSION_DESCRIPTION (gst_webrtc_session_description_get_type())
33 GST_WEBRTC_API
34 GType gst_webrtc_session_description_get_type (void);
35 
36 /**
37  * GstWebRTCSessionDescription:
38  * @type: the #GstWebRTCSDPType of the description
39  * @sdp: the #GstSDPMessage of the description
40  *
41  * See <https://www.w3.org/TR/webrtc/#rtcsessiondescription-class>
42  */
43 struct _GstWebRTCSessionDescription
44 {
45   GstWebRTCSDPType       type;
46   GstSDPMessage         *sdp;
47 };
48 
49 GST_WEBRTC_API
50 GstWebRTCSessionDescription *       gst_webrtc_session_description_new      (GstWebRTCSDPType type, GstSDPMessage *sdp);
51 GST_WEBRTC_API
52 GstWebRTCSessionDescription *       gst_webrtc_session_description_copy     (const GstWebRTCSessionDescription * src);
53 GST_WEBRTC_API
54 void                                gst_webrtc_session_description_free     (GstWebRTCSessionDescription * desc);
55 
56 
57 G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCSessionDescription, gst_webrtc_session_description_free)
58 
59 G_END_DECLS
60 
61 #endif /* __GST_WEBRTC_PEERCONNECTION_H__ */
62