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83 lines
3.4 KiB
C
83 lines
3.4 KiB
C
/* GStreamer
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* Copyright (C) 2017 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_RTP_SENDER_H__
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#define __GST_WEBRTC_RTP_SENDER_H__
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#include <gst/gst.h>
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#include <gst/webrtc/webrtc_fwd.h>
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#include <gst/webrtc/dtlstransport.h>
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G_BEGIN_DECLS
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GST_WEBRTC_API
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GType gst_webrtc_rtp_sender_get_type(void);
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#define GST_TYPE_WEBRTC_RTP_SENDER (gst_webrtc_rtp_sender_get_type())
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#define GST_WEBRTC_RTP_SENDER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_RTP_SENDER,GstWebRTCRTPSender))
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#define GST_IS_WEBRTC_RTP_SENDER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_RTP_SENDER))
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#define GST_WEBRTC_RTP_SENDER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_RTP_SENDER,GstWebRTCRTPSenderClass))
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#define GST_IS_WEBRTC_RTP_SENDER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_RTP_SENDER))
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#define GST_WEBRTC_RTP_SENDER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_SENDER,GstWebRTCRTPSenderClass))
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/**
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* GstWebRTCRTPSender:
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* @transport: The transport for RTP packets
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* @rtcp_transport: The transport for RTCP packets without rtcp-mux
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* @send_encodings: Unused
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* @priority: The priority of the stream (Since: 1.20)
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*/
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struct _GstWebRTCRTPSender
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{
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GstObject parent;
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/* The MediStreamTrack is represented by the stream and is output into @transport/@rtcp_transport as necessary */
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GstWebRTCDTLSTransport *transport;
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GstWebRTCDTLSTransport *rtcp_transport;
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GArray *send_encodings;
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GstWebRTCPriorityType priority;
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gpointer _padding[GST_PADDING];
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};
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struct _GstWebRTCRTPSenderClass
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{
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GstObjectClass parent_class;
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gpointer _padding[GST_PADDING];
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};
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GST_WEBRTC_API
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GstWebRTCRTPSender * gst_webrtc_rtp_sender_new (void);
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GST_WEBRTC_API
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void gst_webrtc_rtp_sender_set_transport (GstWebRTCRTPSender * sender,
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GstWebRTCDTLSTransport * transport);
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GST_WEBRTC_API
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void gst_webrtc_rtp_sender_set_rtcp_transport (GstWebRTCRTPSender * sender,
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GstWebRTCDTLSTransport * transport);
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GST_WEBRTC_API
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void gst_webrtc_rtp_sender_set_priority (GstWebRTCRTPSender *sender,
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GstWebRTCPriorityType priority);
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G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCRTPSender, gst_object_unref)
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G_END_DECLS
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#endif /* __GST_WEBRTC_RTP_SENDER_H__ */
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