47 lines
		
	
	
		
			2.1 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			47 lines
		
	
	
		
			2.1 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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| 
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| #ifndef __GST_WEBRTC_RTP_SENDER_H__
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| #define __GST_WEBRTC_RTP_SENDER_H__
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| 
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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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| 
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| G_BEGIN_DECLS
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| 
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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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| 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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| 
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| G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCRTPSender, gst_object_unref)
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| 
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| G_END_DECLS
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| 
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| #endif /* __GST_WEBRTC_RTP_SENDER_H__ */
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