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
|
/* 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.
*/
/**
* SECTION:gstwebrtc-sender
* @short_description: RTCRtpSender object
* @title: GstWebRTCRTPSender
* @see_also: #GstWebRTCRTPReceiver, #GstWebRTCRTPTransceiver
*
* <ulink url="https://www.w3.org/TR/webrtc/#rtcrtpsender-interface">https://www.w3.org/TR/webrtc/#rtcrtpsender-interface</ulink>
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "rtpsender.h"
#include "rtptransceiver.h"
#define GST_CAT_DEFAULT gst_webrtc_rtp_sender_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
#define gst_webrtc_rtp_sender_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstWebRTCRTPSender, gst_webrtc_rtp_sender,
GST_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_rtp_sender_debug,
"webrtcsender", 0, "webrtcsender");
);
enum
{
SIGNAL_0,
LAST_SIGNAL,
};
enum
{
PROP_0,
PROP_MID,
PROP_SENDER,
PROP_STOPPED,
PROP_DIRECTION,
};
//static guint gst_webrtc_rtp_sender_signals[LAST_SIGNAL] = { 0 };
void
gst_webrtc_rtp_sender_set_transport (GstWebRTCRTPSender * sender,
GstWebRTCDTLSTransport * transport)
{
g_return_if_fail (GST_IS_WEBRTC_RTP_SENDER (sender));
g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
gst_object_replace ((GstObject **) & sender->transport,
GST_OBJECT (transport));
}
void
gst_webrtc_rtp_sender_set_rtcp_transport (GstWebRTCRTPSender * sender,
GstWebRTCDTLSTransport * transport)
{
g_return_if_fail (GST_IS_WEBRTC_RTP_SENDER (sender));
g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
gst_object_replace ((GstObject **) & sender->rtcp_transport,
GST_OBJECT (transport));
}
static void
gst_webrtc_rtp_sender_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_webrtc_rtp_sender_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_webrtc_rtp_sender_finalize (GObject * object)
{
GstWebRTCRTPSender *webrtc = GST_WEBRTC_RTP_SENDER (object);
if (webrtc->transport)
gst_object_unref (webrtc->transport);
webrtc->transport = NULL;
if (webrtc->rtcp_transport)
gst_object_unref (webrtc->rtcp_transport);
webrtc->rtcp_transport = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_webrtc_rtp_sender_class_init (GstWebRTCRTPSenderClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
gobject_class->get_property = gst_webrtc_rtp_sender_get_property;
gobject_class->set_property = gst_webrtc_rtp_sender_set_property;
gobject_class->finalize = gst_webrtc_rtp_sender_finalize;
}
static void
gst_webrtc_rtp_sender_init (GstWebRTCRTPSender * webrtc)
{
}
GstWebRTCRTPSender *
gst_webrtc_rtp_sender_new (GArray * send_encodings /* FIXME */ )
{
return g_object_new (GST_TYPE_WEBRTC_RTP_SENDER, NULL);
}
|
