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1894293d63
SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
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_EXPORT
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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_EXPORT
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GstWebRTCRTPReceiver * gst_webrtc_rtp_receiver_new (void);
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GST_EXPORT
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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_EXPORT
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gboolean gst_webrtc_rtp_receiver_set_parameters (GstWebRTCRTPReceiver * receiver,
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GstStructure * parameters);
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GST_EXPORT
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void gst_webrtc_rtp_receiver_set_transport (GstWebRTCRTPReceiver * receiver,
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GstWebRTCDTLSTransport * transport);
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GST_EXPORT
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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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