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2377f8b3f2
Input (sink pads) is the already-ssrc-muxed stream with the relevant rtp sdes header extensions already applied: - mid - stream-id - repaired-stream-id Output (src pads) have the pads separated into individual ssrc's as that's what rtpbin gives us. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
80 lines
3 KiB
C
80 lines
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 __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 "gst/webrtc/webrtc-priv.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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GstStructure *local_rtx_ssrc_map;
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GstEvent *tos_event;
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/* Properties */
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GstWebRTCFECType fec_type;
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guint fec_percentage;
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gboolean do_nack;
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/* The last caps that we put into to a SDP media section */
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GstCaps *last_retrieved_caps;
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/* The last caps that we successfully configured from a valid
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* set_local/remote description call.
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*/
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GstCaps *last_send_configured_caps;
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gboolean mline_locked;
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GstElement *ulpfecdec;
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GstElement *ulpfecenc;
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GstElement *redenc;
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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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GstWebRTCDTLSTransport * webrtc_transceiver_get_dtls_transport (GstWebRTCRTPTransceiver * trans);
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G_END_DECLS
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#endif /* __WEBRTC_TRANSCEIVER_H__ */
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