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
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
|
/* GStreamer
* Copyright (C) 2018 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.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <stdio.h>
#include "webrtcsctptransport.h"
#include "gstwebrtcbin.h"
#define GST_CAT_DEFAULT webrtc_sctp_transport_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
enum
{
SIGNAL_0,
ON_STREAM_RESET_SIGNAL,
LAST_SIGNAL,
};
enum
{
PROP_0,
PROP_TRANSPORT,
PROP_STATE,
PROP_MAX_MESSAGE_SIZE,
PROP_MAX_CHANNELS,
};
static guint webrtc_sctp_transport_signals[LAST_SIGNAL] = { 0 };
#define webrtc_sctp_transport_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (WebRTCSCTPTransport, webrtc_sctp_transport,
GST_TYPE_WEBRTC_SCTP_TRANSPORT,
GST_DEBUG_CATEGORY_INIT (webrtc_sctp_transport_debug,
"webrtcsctptransport", 0, "webrtcsctptransport"););
typedef void (*SCTPTask) (WebRTCSCTPTransport * sctp, gpointer user_data);
struct task
{
WebRTCSCTPTransport *sctp;
SCTPTask func;
gpointer user_data;
GDestroyNotify notify;
};
static GstStructure *
_execute_task (GstWebRTCBin * webrtc, struct task *task)
{
if (task->func)
task->func (task->sctp, task->user_data);
return NULL;
}
static void
_free_task (struct task *task)
{
gst_object_unref (task->sctp);
if (task->notify)
task->notify (task->user_data);
g_free (task);
}
static void
_sctp_enqueue_task (WebRTCSCTPTransport * sctp, SCTPTask func,
gpointer user_data, GDestroyNotify notify)
{
struct task *task = g_new0 (struct task, 1);
task->sctp = gst_object_ref (sctp);
task->func = func;
task->user_data = user_data;
task->notify = notify;
gst_webrtc_bin_enqueue_task (sctp->webrtcbin,
(GstWebRTCBinFunc) _execute_task, task, (GDestroyNotify) _free_task,
NULL);
}
static void
_emit_stream_reset (WebRTCSCTPTransport * sctp, gpointer user_data)
{
guint stream_id = GPOINTER_TO_UINT (user_data);
g_signal_emit (sctp,
webrtc_sctp_transport_signals[ON_STREAM_RESET_SIGNAL], 0, stream_id);
}
static void
_on_sctp_dec_pad_removed (GstElement * sctpdec, GstPad * pad,
WebRTCSCTPTransport * sctp)
{
guint stream_id;
if (sscanf (GST_PAD_NAME (pad), "src_%u", &stream_id) != 1)
return;
_sctp_enqueue_task (sctp, (SCTPTask) _emit_stream_reset,
GUINT_TO_POINTER (stream_id), NULL);
}
static void
_on_sctp_association_established (GstElement * sctpenc, gboolean established,
WebRTCSCTPTransport * sctp)
{
GST_OBJECT_LOCK (sctp);
if (established)
sctp->state = GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED;
else
sctp->state = GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED;
sctp->association_established = established;
GST_OBJECT_UNLOCK (sctp);
g_object_notify (G_OBJECT (sctp), "state");
}
void
webrtc_sctp_transport_set_priority (WebRTCSCTPTransport * sctp,
GstWebRTCPriorityType priority)
{
GstPad *pad;
pad = gst_element_get_static_pad (sctp->sctpenc, "src");
gst_pad_push_event (pad,
gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_STICKY,
gst_structure_new ("GstWebRtcBinUpdateTos", "sctp-priority",
GST_TYPE_WEBRTC_PRIORITY_TYPE, priority, NULL)));
gst_object_unref (pad);
}
static void
webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
WebRTCSCTPTransport *sctp = WEBRTC_SCTP_TRANSPORT (object);
switch (prop_id) {
case PROP_TRANSPORT:
g_value_set_object (value, sctp->transport);
break;
case PROP_STATE:
g_value_set_enum (value, sctp->state);
break;
case PROP_MAX_MESSAGE_SIZE:
g_value_set_uint64 (value, sctp->max_message_size);
break;
case PROP_MAX_CHANNELS:
g_value_set_uint (value, sctp->max_channels);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
webrtc_sctp_transport_finalize (GObject * object)
{
WebRTCSCTPTransport *sctp = WEBRTC_SCTP_TRANSPORT (object);
g_signal_handlers_disconnect_by_data (sctp->sctpdec, sctp);
g_signal_handlers_disconnect_by_data (sctp->sctpenc, sctp);
gst_object_unref (sctp->sctpdec);
gst_object_unref (sctp->sctpenc);
g_clear_object (&sctp->transport);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
webrtc_sctp_transport_constructed (GObject * object)
{
WebRTCSCTPTransport *sctp = WEBRTC_SCTP_TRANSPORT (object);
guint association_id;
association_id = g_random_int_range (0, G_MAXUINT16);
sctp->sctpdec =
g_object_ref_sink (gst_element_factory_make ("sctpdec", NULL));
g_object_set (sctp->sctpdec, "sctp-association-id", association_id, NULL);
sctp->sctpenc =
g_object_ref_sink (gst_element_factory_make ("sctpenc", NULL));
g_object_set (sctp->sctpenc, "sctp-association-id", association_id, NULL);
g_object_set (sctp->sctpenc, "use-sock-stream", TRUE, NULL);
g_signal_connect (sctp->sctpdec, "pad-removed",
G_CALLBACK (_on_sctp_dec_pad_removed), sctp);
g_signal_connect (sctp->sctpenc, "sctp-association-established",
G_CALLBACK (_on_sctp_association_established), sctp);
G_OBJECT_CLASS (parent_class)->constructed (object);
}
static void
webrtc_sctp_transport_class_init (WebRTCSCTPTransportClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
gobject_class->constructed = webrtc_sctp_transport_constructed;
gobject_class->get_property = webrtc_sctp_transport_get_property;
gobject_class->finalize = webrtc_sctp_transport_finalize;
g_object_class_override_property (gobject_class, PROP_TRANSPORT, "transport");
g_object_class_override_property (gobject_class, PROP_STATE, "state");
g_object_class_override_property (gobject_class,
PROP_MAX_MESSAGE_SIZE, "max-message-size");
g_object_class_override_property (gobject_class,
PROP_MAX_CHANNELS, "max-channels");
/**
* WebRTCSCTPTransport::stream-reset:
* @object: the #WebRTCSCTPTransport
* @stream_id: the SCTP stream that was reset
*/
webrtc_sctp_transport_signals[ON_STREAM_RESET_SIGNAL] =
g_signal_new ("stream-reset", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT);
}
static void
webrtc_sctp_transport_init (WebRTCSCTPTransport * nice)
{
}
WebRTCSCTPTransport *
webrtc_sctp_transport_new (void)
{
return g_object_new (TYPE_WEBRTC_SCTP_TRANSPORT, NULL);
}
|
