gstreamer/subprojects/gst-plugins-bad/ext/webrtc/webrtctransceiver.h
Matthew Waters 2377f8b3f2 webrtcbin: initial support for sending and receiving simulcast streams
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>
2022-03-29 23:55:40 +00:00

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3 KiB
C

/* GStreamer
* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifndef __WEBRTC_TRANSCEIVER_H__
#define __WEBRTC_TRANSCEIVER_H__
#include "fwd.h"
#include <gst/webrtc/rtptransceiver.h>
#include "gst/webrtc/webrtc-priv.h"
#include "transportstream.h"
G_BEGIN_DECLS
GType webrtc_transceiver_get_type(void);
#define WEBRTC_TYPE_TRANSCEIVER (webrtc_transceiver_get_type())
#define WEBRTC_TRANSCEIVER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),WEBRTC_TYPE_TRANSCEIVER,WebRTCTransceiver))
#define WEBRTC_IS_TRANSCEIVER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),WEBRTC_TYPE_TRANSCEIVER))
#define WEBRTC_TRANSCEIVER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,WEBRTC_TYPE_TRANSCEIVER,WebRTCTransceiverClass))
#define WEBRTC_TRANSCEIVER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,WEBRTC_TYPE_TRANSCEIVER,WebRTCTransceiverClass))
struct _WebRTCTransceiver
{
GstWebRTCRTPTransceiver parent;
TransportStream *stream;
GstStructure *local_rtx_ssrc_map;
GstEvent *tos_event;
/* Properties */
GstWebRTCFECType fec_type;
guint fec_percentage;
gboolean do_nack;
/* The last caps that we put into to a SDP media section */
GstCaps *last_retrieved_caps;
/* The last caps that we successfully configured from a valid
* set_local/remote description call.
*/
GstCaps *last_send_configured_caps;
gboolean mline_locked;
GstElement *ulpfecdec;
GstElement *ulpfecenc;
GstElement *redenc;
};
struct _WebRTCTransceiverClass
{
GstWebRTCRTPTransceiverClass parent_class;
};
WebRTCTransceiver * webrtc_transceiver_new (GstWebRTCBin * webrtc,
GstWebRTCRTPSender * sender,
GstWebRTCRTPReceiver * receiver);
void webrtc_transceiver_set_transport (WebRTCTransceiver * trans,
TransportStream * stream);
GstWebRTCDTLSTransport * webrtc_transceiver_get_dtls_transport (GstWebRTCRTPTransceiver * trans);
G_END_DECLS
#endif /* __WEBRTC_TRANSCEIVER_H__ */