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-sender
22 * @short_description: RTCRtpSender object
23 * @title: GstWebRTCRTPSender
24 * @see_also: #GstWebRTCRTPReceiver, #GstWebRTCRTPTransceiver
25 *
26 * <ulink url="https://www.w3.org/TR/webrtc/#rtcrtpsender-interface">https://www.w3.org/TR/webrtc/#rtcrtpsender-interface</ulink>
27 */
28
29 #ifdef HAVE_CONFIG_H
30 # include "config.h"
31 #endif
32
33 #include "rtpsender.h"
34 #include "rtptransceiver.h"
35
36 #define GST_CAT_DEFAULT gst_webrtc_rtp_sender_debug
37 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
38
39 #define gst_webrtc_rtp_sender_parent_class parent_class
40 G_DEFINE_TYPE_WITH_CODE (GstWebRTCRTPSender, gst_webrtc_rtp_sender,
41 GST_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_rtp_sender_debug,
42 "webrtcsender", 0, "webrtcsender");
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_STOPPED,
57 PROP_DIRECTION,
58 };
59
60 //static guint gst_webrtc_rtp_sender_signals[LAST_SIGNAL] = { 0 };
61
62 void
gst_webrtc_rtp_sender_set_transport(GstWebRTCRTPSender * sender,GstWebRTCDTLSTransport * transport)63 gst_webrtc_rtp_sender_set_transport (GstWebRTCRTPSender * sender,
64 GstWebRTCDTLSTransport * transport)
65 {
66 g_return_if_fail (GST_IS_WEBRTC_RTP_SENDER (sender));
67 g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
68
69 GST_OBJECT_LOCK (sender);
70 gst_object_replace ((GstObject **) & sender->transport,
71 GST_OBJECT (transport));
72 GST_OBJECT_UNLOCK (sender);
73 }
74
75 void
gst_webrtc_rtp_sender_set_rtcp_transport(GstWebRTCRTPSender * sender,GstWebRTCDTLSTransport * transport)76 gst_webrtc_rtp_sender_set_rtcp_transport (GstWebRTCRTPSender * sender,
77 GstWebRTCDTLSTransport * transport)
78 {
79 g_return_if_fail (GST_IS_WEBRTC_RTP_SENDER (sender));
80 g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
81
82 GST_OBJECT_LOCK (sender);
83 gst_object_replace ((GstObject **) & sender->rtcp_transport,
84 GST_OBJECT (transport));
85 GST_OBJECT_UNLOCK (sender);
86 }
87
88 static void
gst_webrtc_rtp_sender_set_property(GObject * object,guint prop_id,const GValue * value,GParamSpec * pspec)89 gst_webrtc_rtp_sender_set_property (GObject * object, guint prop_id,
90 const GValue * value, GParamSpec * pspec)
91 {
92 switch (prop_id) {
93 default:
94 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
95 break;
96 }
97 }
98
99 static void
gst_webrtc_rtp_sender_get_property(GObject * object,guint prop_id,GValue * value,GParamSpec * pspec)100 gst_webrtc_rtp_sender_get_property (GObject * object, guint prop_id,
101 GValue * value, GParamSpec * pspec)
102 {
103 switch (prop_id) {
104 default:
105 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
106 break;
107 }
108 }
109
110 static void
gst_webrtc_rtp_sender_finalize(GObject * object)111 gst_webrtc_rtp_sender_finalize (GObject * object)
112 {
113 GstWebRTCRTPSender *webrtc = GST_WEBRTC_RTP_SENDER (object);
114
115 if (webrtc->transport)
116 gst_object_unref (webrtc->transport);
117 webrtc->transport = NULL;
118
119 if (webrtc->rtcp_transport)
120 gst_object_unref (webrtc->rtcp_transport);
121 webrtc->rtcp_transport = NULL;
122
123 G_OBJECT_CLASS (parent_class)->finalize (object);
124 }
125
126 static void
gst_webrtc_rtp_sender_class_init(GstWebRTCRTPSenderClass * klass)127 gst_webrtc_rtp_sender_class_init (GstWebRTCRTPSenderClass * klass)
128 {
129 GObjectClass *gobject_class = (GObjectClass *) klass;
130
131 gobject_class->get_property = gst_webrtc_rtp_sender_get_property;
132 gobject_class->set_property = gst_webrtc_rtp_sender_set_property;
133 gobject_class->finalize = gst_webrtc_rtp_sender_finalize;
134 }
135
136 static void
gst_webrtc_rtp_sender_init(GstWebRTCRTPSender * webrtc)137 gst_webrtc_rtp_sender_init (GstWebRTCRTPSender * webrtc)
138 {
139 }
140
141 GstWebRTCRTPSender *
gst_webrtc_rtp_sender_new(void)142 gst_webrtc_rtp_sender_new (void)
143 {
144 return g_object_new (GST_TYPE_WEBRTC_RTP_SENDER, NULL);
145 }
146