mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-27 02:30:35 +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.
87 lines
3.3 KiB
C
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__ */
|