| /* GStreamer |
| * Copyright (C) 2017 Matthew Waters <matthew@centricular.com> |
| * |
| * This library is free software; you can redistribute it and/or |
| * modify it under the terms of the GNU Library General Public |
| * License as published by the Free Software Foundation; either |
| * version 2 of the License, or (at your option) any later version. |
| * |
| * This library is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| * Library General Public License for more details. |
| * |
| * You should have received a copy of the GNU Library General Public |
| * License along with this library; if not, write to the |
| * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, |
| * Boston, MA 02110-1301, USA. |
| */ |
| |
| #ifndef __GST_WEBRTC_RTP_RECEIVER_H__ |
| #define __GST_WEBRTC_RTP_RECEIVER_H__ |
| |
| #include <gst/gst.h> |
| #include <gst/webrtc/webrtc_fwd.h> |
| #include <gst/webrtc/dtlstransport.h> |
| |
| G_BEGIN_DECLS |
| |
| GST_WEBRTC_API |
| GType gst_webrtc_rtp_receiver_get_type(void); |
| #define GST_TYPE_WEBRTC_RTP_RECEIVER (gst_webrtc_rtp_receiver_get_type()) |
| #define GST_WEBRTC_RTP_RECEIVER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_RTP_RECEIVER,GstWebRTCRTPReceiver)) |
| #define GST_IS_WEBRTC_RTP_RECEIVER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_RTP_RECEIVER)) |
| #define GST_WEBRTC_RTP_RECEIVER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_RTP_RECEIVER,GstWebRTCRTPReceiverClass)) |
| #define GST_IS_WEBRTC_RTP_RECEIVER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_RTP_RECEIVER)) |
| #define GST_WEBRTC_RTP_RECEIVER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_RECEIVER,GstWebRTCRTPReceiverClass)) |
| |
| typedef struct _GstWebRTCRTPReceiver GstWebRTCRTPReceiver; |
| typedef struct _GstWebRTCRTPReceiverClass GstWebRTCRTPReceiverClass; |
| |
| struct _GstWebRTCRTPReceiver |
| { |
| GstObject parent; |
| |
| /* The MediStreamTrack is represented by the stream and is output into @transport/@rtcp_transport as necessary */ |
| GstWebRTCDTLSTransport *transport; |
| GstWebRTCDTLSTransport *rtcp_transport; |
| |
| gpointer _padding[GST_PADDING]; |
| }; |
| |
| struct _GstWebRTCRTPReceiverClass |
| { |
| GstObjectClass parent_class; |
| |
| gpointer _padding[GST_PADDING]; |
| }; |
| |
| GST_WEBRTC_API |
| GstWebRTCRTPReceiver * gst_webrtc_rtp_receiver_new (void); |
| GST_WEBRTC_API |
| void gst_webrtc_rtp_receiver_set_transport (GstWebRTCRTPReceiver * receiver, |
| GstWebRTCDTLSTransport * transport); |
| GST_WEBRTC_API |
| void gst_webrtc_rtp_receiver_set_rtcp_transport (GstWebRTCRTPReceiver * receiver, |
| GstWebRTCDTLSTransport * transport); |
| |
| G_END_DECLS |
| |
| #endif /* __GST_WEBRTC_RTP_RECEIVER_H__ */ |