mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-07-16 03:18:29 +00:00
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>
81 lines
3 KiB
C
81 lines
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__ */
|