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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
58 lines
2.2 KiB
C
58 lines
2.2 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 __WEBRTC_TRANSCEIVER_H__
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#define __WEBRTC_TRANSCEIVER_H__
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#include "fwd.h"
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#include <gst/webrtc/rtptransceiver.h>
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#include "transportstream.h"
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G_BEGIN_DECLS
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GType webrtc_transceiver_get_type(void);
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#define WEBRTC_TYPE_TRANSCEIVER (webrtc_transceiver_get_type())
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#define WEBRTC_TRANSCEIVER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),WEBRTC_TYPE_TRANSCEIVER,WebRTCTransceiver))
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#define WEBRTC_IS_TRANSCEIVER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),WEBRTC_TYPE_TRANSCEIVER))
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#define WEBRTC_TRANSCEIVER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,WEBRTC_TYPE_TRANSCEIVER,WebRTCTransceiverClass))
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#define WEBRTC_TRANSCEIVER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,WEBRTC_TYPE_TRANSCEIVER,WebRTCTransceiverClass))
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struct _WebRTCTransceiver
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{
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GstWebRTCRTPTransceiver parent;
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TransportStream *stream;
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};
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struct _WebRTCTransceiverClass
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{
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GstWebRTCRTPTransceiverClass parent_class;
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};
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WebRTCTransceiver * webrtc_transceiver_new (GstWebRTCBin * webrtc,
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GstWebRTCRTPSender * sender,
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GstWebRTCRTPReceiver * receiver);
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void webrtc_transceiver_set_transport (WebRTCTransceiver * trans,
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TransportStream * stream);
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G_END_DECLS
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#endif /* __WEBRTC_TRANSCEIVER_H__ */
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