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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
69 lines
2.4 KiB
C
69 lines
2.4 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 __TRANSPORT_STREAM_H__
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#define __TRANSPORT_STREAM_H__
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#include "fwd.h"
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#include <gst/webrtc/rtptransceiver.h>
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G_BEGIN_DECLS
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GType transport_stream_get_type(void);
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#define GST_TYPE_WEBRTC_TRANSPORT_STREAM (transport_stream_get_type())
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#define TRANSPORT_STREAM(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_TRANSPORT_STREAM,TransportStream))
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#define TRANSPORT_STREAM_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_TRANSPORT_STREAM,TransportStreamClass))
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#define TRANSPORT_STREAM_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_TRANSPORT_STREAM,TransportStreamClass))
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typedef struct
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{
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guint8 pt;
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GstCaps *caps;
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} PtMapItem;
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struct _TransportStream
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{
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GstObject parent;
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guint session_id; /* session_id */
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gboolean rtcp;
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gboolean rtcp_mux;
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gboolean rtcp_rsize;
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gboolean dtls_client;
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TransportSendBin *send_bin; /* bin containing all the sending transport elements */
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TransportReceiveBin *receive_bin; /* bin containing all the receiving transport elements */
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GstWebRTCICEStream *stream;
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GstWebRTCDTLSTransport *transport;
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GstWebRTCDTLSTransport *rtcp_transport;
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GArray *ptmap; /* array of PtMapItem's */
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};
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struct _TransportStreamClass
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{
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GstObjectClass parent_class;
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};
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TransportStream * transport_stream_new (GstWebRTCBin * webrtc,
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guint session_id);
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G_END_DECLS
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#endif /* __TRANSPORT_STREAM_H__ */
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