webrtcbin: an element that handles the transport aspects of webrtc connections
[platform/upstream/gst-plugins-bad.git] / gst-libs / gst / webrtc / rtptransceiver.c
1 /* GStreamer
2  * Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
3  *
4  * This library is free software; you can redistribute it and/or
5  * modify it under the terms of the GNU Library General Public
6  * License as published by the Free Software Foundation; either
7  * version 2 of the License, or (at your option) any later version.
8  *
9  * This library is distributed in the hope that it will be useful,
10  * but WITHOUT ANY WARRANTY; without even the implied warranty of
11  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
12  * Library General Public License for more details.
13  *
14  * You should have received a copy of the GNU Library General Public
15  * License along with this library; if not, write to the
16  * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17  * Boston, MA 02110-1301, USA.
18  */
19
20 /**
21  * SECTION:gstwebrtc-transceiver
22  * @short_description: RTCRtpTransceiver object
23  * @title: GstWebRTCRTPTransceiver
24  * @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPReceiver
25  *
26  * <ulink url="https://www.w3.org/TR/webrtc/#rtcrtptransceiver-interface">https://www.w3.org/TR/webrtc/#rtcrtptransceiver-interface</ulink>
27  */
28
29 #ifdef HAVE_CONFIG_H
30 # include "config.h"
31 #endif
32
33 #include "rtptransceiver.h"
34
35 #define GST_CAT_DEFAULT gst_webrtc_rtp_transceiver_debug
36 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
37
38 #define gst_webrtc_rtp_transceiver_parent_class parent_class
39 G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstWebRTCRTPTransceiver,
40     gst_webrtc_rtp_transceiver, GST_TYPE_OBJECT,
41     GST_DEBUG_CATEGORY_INIT (gst_webrtc_rtp_transceiver_debug,
42         "webrtctransceiver", 0, "webrtctransceiver");
43     );
44
45 enum
46 {
47   SIGNAL_0,
48   LAST_SIGNAL,
49 };
50
51 enum
52 {
53   PROP_0,
54   PROP_MID,
55   PROP_SENDER,
56   PROP_RECEIVER,
57   PROP_STOPPED,                 // FIXME
58   PROP_DIRECTION,               // FIXME
59   PROP_MLINE,
60 };
61
62 //static guint gst_webrtc_rtp_transceiver_signals[LAST_SIGNAL] = { 0 };
63
64 static void
65 gst_webrtc_rtp_transceiver_set_property (GObject * object, guint prop_id,
66     const GValue * value, GParamSpec * pspec)
67 {
68   GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
69
70   switch (prop_id) {
71     case PROP_SENDER:
72       webrtc->sender = g_value_dup_object (value);
73       break;
74     case PROP_RECEIVER:
75       webrtc->receiver = g_value_dup_object (value);
76       break;
77     case PROP_MLINE:
78       webrtc->mline = g_value_get_uint (value);
79       break;
80     default:
81       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
82       break;
83   }
84 }
85
86 static void
87 gst_webrtc_rtp_transceiver_get_property (GObject * object, guint prop_id,
88     GValue * value, GParamSpec * pspec)
89 {
90   GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
91
92   switch (prop_id) {
93     case PROP_SENDER:
94       g_value_set_object (value, webrtc->sender);
95       break;
96     case PROP_RECEIVER:
97       g_value_set_object (value, webrtc->receiver);
98       break;
99     case PROP_MLINE:
100       g_value_set_uint (value, webrtc->mline);
101       break;
102     default:
103       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
104       break;
105   }
106 }
107
108 static void
109 gst_webrtc_rtp_transceiver_constructed (GObject * object)
110 {
111   GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
112
113   gst_object_set_parent (GST_OBJECT (webrtc->sender), GST_OBJECT (webrtc));
114   gst_object_set_parent (GST_OBJECT (webrtc->receiver), GST_OBJECT (webrtc));
115
116   G_OBJECT_CLASS (parent_class)->constructed (object);
117 }
118
119 static void
120 gst_webrtc_rtp_transceiver_dispose (GObject * object)
121 {
122   GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
123
124   if (webrtc->sender) {
125     GST_OBJECT_PARENT (webrtc->sender) = NULL;
126     gst_object_unref (webrtc->sender);
127   }
128   webrtc->sender = NULL;
129   if (webrtc->receiver) {
130     GST_OBJECT_PARENT (webrtc->receiver) = NULL;
131     gst_object_unref (webrtc->receiver);
132   }
133   webrtc->receiver = NULL;
134
135   G_OBJECT_CLASS (parent_class)->dispose (object);
136 }
137
138 static void
139 gst_webrtc_rtp_transceiver_finalize (GObject * object)
140 {
141   GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
142
143   g_free (webrtc->mid);
144   if (webrtc->codec_preferences)
145     gst_caps_unref (webrtc->codec_preferences);
146
147   G_OBJECT_CLASS (parent_class)->finalize (object);
148 }
149
150 static void
151 gst_webrtc_rtp_transceiver_class_init (GstWebRTCRTPTransceiverClass * klass)
152 {
153   GObjectClass *gobject_class = (GObjectClass *) klass;
154
155   gobject_class->get_property = gst_webrtc_rtp_transceiver_get_property;
156   gobject_class->set_property = gst_webrtc_rtp_transceiver_set_property;
157   gobject_class->constructed = gst_webrtc_rtp_transceiver_constructed;
158   gobject_class->dispose = gst_webrtc_rtp_transceiver_dispose;
159   gobject_class->finalize = gst_webrtc_rtp_transceiver_finalize;
160
161   g_object_class_install_property (gobject_class,
162       PROP_SENDER,
163       g_param_spec_object ("sender", "Sender",
164           "The RTP sender for this transceiver",
165           GST_TYPE_WEBRTC_RTP_SENDER,
166           G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
167
168   g_object_class_install_property (gobject_class,
169       PROP_RECEIVER,
170       g_param_spec_object ("receiver", "Receiver",
171           "The RTP receiver for this transceiver",
172           GST_TYPE_WEBRTC_RTP_RECEIVER,
173           G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
174
175   g_object_class_install_property (gobject_class,
176       PROP_MLINE,
177       g_param_spec_uint ("mlineindex", "Media Line Index",
178           "Index in the SDP of the Media",
179           0, G_MAXUINT, 0,
180           G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
181 }
182
183 static void
184 gst_webrtc_rtp_transceiver_init (GstWebRTCRTPTransceiver * webrtc)
185 {
186 }