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333f636555
We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
76 lines
3.3 KiB
C
76 lines
3.3 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_RECEIVER_H__
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#define __GST_WEBRTC_RTP_RECEIVER_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_receiver_get_type(void);
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#define GST_TYPE_WEBRTC_RTP_RECEIVER (gst_webrtc_rtp_receiver_get_type())
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#define GST_WEBRTC_RTP_RECEIVER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_RTP_RECEIVER,GstWebRTCRTPReceiver))
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#define GST_IS_WEBRTC_RTP_RECEIVER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_RTP_RECEIVER))
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#define GST_WEBRTC_RTP_RECEIVER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_RTP_RECEIVER,GstWebRTCRTPReceiverClass))
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#define GST_IS_WEBRTC_RTP_RECEIVER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_RTP_RECEIVER))
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#define GST_WEBRTC_RTP_RECEIVER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_RECEIVER,GstWebRTCRTPReceiverClass))
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typedef struct _GstWebRTCRTPReceiver GstWebRTCRTPReceiver;
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typedef struct _GstWebRTCRTPReceiverClass GstWebRTCRTPReceiverClass;
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struct _GstWebRTCRTPReceiver
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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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gpointer _padding[GST_PADDING];
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};
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struct _GstWebRTCRTPReceiverClass
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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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GstWebRTCRTPReceiver * gst_webrtc_rtp_receiver_new (void);
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GST_WEBRTC_API
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GstStructure * gst_webrtc_rtp_receiver_get_parameters (GstWebRTCRTPReceiver * receiver, gchar * kind);
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/* FIXME: promise? */
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GST_WEBRTC_API
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gboolean gst_webrtc_rtp_receiver_set_parameters (GstWebRTCRTPReceiver * receiver,
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GstStructure * parameters);
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GST_WEBRTC_API
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void gst_webrtc_rtp_receiver_set_transport (GstWebRTCRTPReceiver * receiver,
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GstWebRTCDTLSTransport * transport);
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GST_WEBRTC_API
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void gst_webrtc_rtp_receiver_set_rtcp_transport (GstWebRTCRTPReceiver * receiver,
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GstWebRTCDTLSTransport * transport);
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G_END_DECLS
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#endif /* __GST_WEBRTC_RTP_RECEIVER_H__ */
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