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