summaryrefslogtreecommitdiff
path: root/ext/webrtc/gstwebrtcbin.h
blob: bbcc5f5079d0620829518b5675a248067a02a1db (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
/* 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 __GST_WEBRTC_BIN_H__
#define __GST_WEBRTC_BIN_H__

#include <gst/sdp/sdp.h>
#include "fwd.h"
#include "gstwebrtcice.h"

G_BEGIN_DECLS

#define GST_WEBRTC_BIN_ERROR gst_webrtc_bin_error_quark ()
GQuark gst_webrtc_bin_error_quark (void);

typedef enum
{
  GST_WEBRTC_BIN_ERROR_FAILED,
  GST_WEBRTC_BIN_ERROR_INVALID_SYNTAX,
  GST_WEBRTC_BIN_ERROR_INVALID_MODIFICATION,
  GST_WEBRTC_BIN_ERROR_INVALID_STATE,
  GST_WEBRTC_BIN_ERROR_BAD_SDP,
  GST_WEBRTC_BIN_ERROR_FINGERPRINT,
} GstWebRTCJSEPSDPError;

GType gst_webrtc_bin_pad_get_type(void);
#define GST_TYPE_WEBRTC_BIN_PAD            (gst_webrtc_bin_pad_get_type())
#define GST_WEBRTC_BIN_PAD(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_BIN_PAD,GstWebRTCBinPad))
#define GST_IS_WEBRTC_BIN_PAD(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_BIN_PAD))
#define GST_WEBRTC_BIN_PAD_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_BIN_PAD,GstWebRTCBinPadClass))
#define GST_IS_WEBRTC_BIN_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_BIN_PAD))
#define GST_WEBRTC_BIN_PAD_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_BIN_PAD,GstWebRTCBinPadClass))

typedef struct _GstWebRTCBinPad GstWebRTCBinPad;
typedef struct _GstWebRTCBinPadClass GstWebRTCBinPadClass;

struct _GstWebRTCBinPad
{
  GstGhostPad           parent;

  guint                 mlineindex;

  GstWebRTCRTPTransceiver *trans;
};

struct _GstWebRTCBinPadClass
{
  GstGhostPadClass      parent_class;
};

GType gst_webrtc_bin_get_type(void);
#define GST_TYPE_WEBRTC_BIN            (gst_webrtc_bin_get_type())
#define GST_WEBRTC_BIN(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_BIN,GstWebRTCBin))
#define GST_IS_WEBRTC_BIN(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_BIN))
#define GST_WEBRTC_BIN_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_BIN,GstWebRTCBinClass))
#define GST_IS_WEBRTC_BIN_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_BIN))
#define GST_WEBRTC_BIN_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_BIN,GstWebRTCBinClass))

struct _GstWebRTCBin
{
  GstBin                            parent;

  GstElement                       *rtpbin;

  GstWebRTCSignalingState           signaling_state;
  GstWebRTCICEGatheringState        ice_gathering_state;
  GstWebRTCICEConnectionState       ice_connection_state;
  GstWebRTCPeerConnectionState      peer_connection_state;

  GstWebRTCSessionDescription      *current_local_description;
  GstWebRTCSessionDescription      *pending_local_description;
  GstWebRTCSessionDescription      *current_remote_description;
  GstWebRTCSessionDescription      *pending_remote_description;

  GstWebRTCBinPrivate              *priv;
};

struct _GstWebRTCBinClass
{
  GstBinClass           parent_class;
};

struct _GstWebRTCBinPrivate
{
  guint max_sink_pad_serial;

  gboolean bundle;
  GArray *transceivers;
  GArray *session_mid_map;
  GArray *transports;

  GstWebRTCICE *ice;
  GArray *ice_stream_map;
  GArray *pending_ice_candidates;

  /* peerconnection variables */
  gboolean is_closed;
  gboolean need_negotiation;
  gpointer sctp_transport;      /* FIXME */

  /* peerconnection helper thread for promises */
  GMainContext *main_context;
  GMainLoop *loop;
  GThread *thread;
  GMutex pc_lock;
  GCond pc_cond;

  gboolean running;
  gboolean async_pending;

  GList *pending_pads;

  /* count of the number of media streams we've offered for uniqueness */
  /* FIXME: overflow? */
  guint media_counter;

  GstStructure *stats;
};

typedef void (*GstWebRTCBinFunc) (GstWebRTCBin * webrtc, gpointer data);

typedef struct
{
  GstWebRTCBin *webrtc;
  GstWebRTCBinFunc op;
  gpointer data;
  GDestroyNotify notify;
//  GstPromise *promise;      /* FIXME */
} GstWebRTCBinTask;

void            gst_webrtc_bin_enqueue_task             (GstWebRTCBin * pc,
                                                         GstWebRTCBinFunc func,
                                                         gpointer data,
                                                         GDestroyNotify notify);

G_END_DECLS

#endif /* __GST_WEBRTC_BIN_H__ */