blob: 4dc6deca6cd55efa8624de40e191baf20e4d4184 [file] [log] [blame]
Matthew Waters18942932017-01-31 20:56:59 +11001/* 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
37GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
38
39#define gst_webrtc_rtp_sender_parent_class parent_class
40G_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
45enum
46{
47 SIGNAL_0,
48 LAST_SIGNAL,
49};
50
51enum
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
62void
63gst_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
Sebastian Dröge950ead92018-03-15 17:31:50 +020069 GST_OBJECT_LOCK (sender);
Matthew Waters18942932017-01-31 20:56:59 +110070 gst_object_replace ((GstObject **) & sender->transport,
71 GST_OBJECT (transport));
Sebastian Dröge950ead92018-03-15 17:31:50 +020072 GST_OBJECT_UNLOCK (sender);
Matthew Waters18942932017-01-31 20:56:59 +110073}
74
75void
76gst_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
Sebastian Dröge950ead92018-03-15 17:31:50 +020082 GST_OBJECT_LOCK (sender);
Matthew Waters18942932017-01-31 20:56:59 +110083 gst_object_replace ((GstObject **) & sender->rtcp_transport,
84 GST_OBJECT (transport));
Sebastian Dröge950ead92018-03-15 17:31:50 +020085 GST_OBJECT_UNLOCK (sender);
Matthew Waters18942932017-01-31 20:56:59 +110086}
87
88static void
89gst_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
99static void
100gst_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
110static void
111gst_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
126static void
127gst_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
136static void
137gst_webrtc_rtp_sender_init (GstWebRTCRTPSender * webrtc)
138{
139}
140
141GstWebRTCRTPSender *
142gst_webrtc_rtp_sender_new (GArray * send_encodings /* FIXME */ )
143{
144 return g_object_new (GST_TYPE_WEBRTC_RTP_SENDER, NULL);
145}