mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-22 16:26:39 +00:00
177aa22bcd
Limitations: - No transport changes at all (ICE, DTLS) - Codec changes are untested and probably don't work - Stream removal doesn't remove transports (i.e. non-bundled transports will stay around until webrtcbin is shutdown) - Unified Plan SDP only. No Plan-B support.
68 lines
2.7 KiB
C
68 lines
2.7 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 "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;
|
|
|
|
/* Properties */
|
|
GstWebRTCFECType fec_type;
|
|
guint fec_percentage;
|
|
gboolean do_nack;
|
|
|
|
GstCaps *last_configured_caps;
|
|
};
|
|
|
|
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);
|
|
GstWebRTCDTLSTransport * webrtc_transceiver_get_rtcp_dtls_transport (GstWebRTCRTPTransceiver * trans);
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __WEBRTC_TRANSCEIVER_H__ */
|