gstreamer/ext/webrtc/transportstream.h
Matthew Waters 177aa22bcd webrtc: Initial support for stream addition/removal
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.
2019-05-30 21:33:09 +10:00

87 lines
3.3 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 __TRANSPORT_STREAM_H__
#define __TRANSPORT_STREAM_H__
#include "fwd.h"
#include <gst/webrtc/rtptransceiver.h>
G_BEGIN_DECLS
GType transport_stream_get_type(void);
#define GST_TYPE_WEBRTC_TRANSPORT_STREAM (transport_stream_get_type())
#define TRANSPORT_STREAM(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_TRANSPORT_STREAM,TransportStream))
#define TRANSPORT_STREAM_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_TRANSPORT_STREAM,TransportStreamClass))
#define TRANSPORT_STREAM_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_TRANSPORT_STREAM,TransportStreamClass))
typedef struct
{
guint8 pt;
GstCaps *caps;
} PtMapItem;
typedef struct
{
guint32 ssrc;
guint media_idx;
} SsrcMapItem;
struct _TransportStream
{
GstObject parent;
guint session_id; /* session_id */
gboolean rtcp;
gboolean rtcp_mux;
gboolean rtcp_rsize;
gboolean dtls_client;
TransportSendBin *send_bin; /* bin containing all the sending transport elements */
TransportReceiveBin *receive_bin; /* bin containing all the receiving transport elements */
GstWebRTCICEStream *stream;
GstWebRTCDTLSTransport *transport;
GstWebRTCDTLSTransport *rtcp_transport;
GArray *ptmap; /* array of PtMapItem's */
GArray *remote_ssrcmap; /* array of SsrcMapItem's */
gboolean output_connected; /* whether receive bin is connected to rtpbin */
GstElement *rtxsend;
GstElement *rtxreceive;
};
struct _TransportStreamClass
{
GstObjectClass parent_class;
};
TransportStream * transport_stream_new (GstWebRTCBin * webrtc,
guint session_id);
int transport_stream_get_pt (TransportStream * stream,
const gchar * encoding_name);
int * transport_stream_get_all_pt (TransportStream * stream,
const gchar * encoding_name,
gsize * pt_len);
GstCaps * transport_stream_get_caps_for_pt (TransportStream * stream,
guint pt);
G_END_DECLS
#endif /* __TRANSPORT_STREAM_H__ */