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_RTP_SENDER_H__
21 #define __GST_WEBRTC_RTP_SENDER_H__
22 
23 #include <gst/gst.h>
24 #include <gst/webrtc/webrtc_fwd.h>
25 #include <gst/webrtc/dtlstransport.h>
26 
27 G_BEGIN_DECLS
28 
29 GST_WEBRTC_API
30 GType gst_webrtc_rtp_sender_get_type(void);
31 #define GST_TYPE_WEBRTC_RTP_SENDER            (gst_webrtc_rtp_sender_get_type())
32 #define GST_WEBRTC_RTP_SENDER(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_RTP_SENDER,GstWebRTCRTPSender))
33 #define GST_IS_WEBRTC_RTP_SENDER(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_RTP_SENDER))
34 #define GST_WEBRTC_RTP_SENDER_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_RTP_SENDER,GstWebRTCRTPSenderClass))
35 #define GST_IS_WEBRTC_RTP_SENDER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_RTP_SENDER))
36 #define GST_WEBRTC_RTP_SENDER_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_SENDER,GstWebRTCRTPSenderClass))
37 
38 struct _GstWebRTCRTPSender
39 {
40   GstObject                          parent;
41 
42   /* The MediStreamTrack is represented by the stream and is output into @transport/@rtcp_transport as necessary */
43   GstWebRTCDTLSTransport            *transport;
44   GstWebRTCDTLSTransport            *rtcp_transport;
45 
46   GArray                            *send_encodings;
47 
48   gpointer                          _padding[GST_PADDING];
49 };
50 
51 struct _GstWebRTCRTPSenderClass
52 {
53   GstObjectClass        parent_class;
54 
55   gpointer              _padding[GST_PADDING];
56 };
57 
58 GST_WEBRTC_API
59 GstWebRTCRTPSender *        gst_webrtc_rtp_sender_new                   (void);
60 
61 GST_WEBRTC_API
62 void                        gst_webrtc_rtp_sender_set_transport         (GstWebRTCRTPSender * sender,
63                                                                          GstWebRTCDTLSTransport * transport);
64 GST_WEBRTC_API
65 void                        gst_webrtc_rtp_sender_set_rtcp_transport    (GstWebRTCRTPSender * sender,
66                                                                          GstWebRTCDTLSTransport * transport);
67 
68 
69 #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
70 G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCRTPSender, gst_object_unref)
71 #endif
72 
73 G_END_DECLS
74 
75 #endif /* __GST_WEBRTC_RTP_SENDER_H__ */
76