Matthew Waters | 1894293 | 2017-01-31 20:56:59 +1100 | [diff] [blame] | 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 |
| 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 | |
Sebastian Dröge | 950ead9 | 2018-03-15 17:31:50 +0200 | [diff] [blame^] | 69 | GST_OBJECT_LOCK (sender); |
Matthew Waters | 1894293 | 2017-01-31 20:56:59 +1100 | [diff] [blame] | 70 | gst_object_replace ((GstObject **) & sender->transport, |
| 71 | GST_OBJECT (transport)); |
Sebastian Dröge | 950ead9 | 2018-03-15 17:31:50 +0200 | [diff] [blame^] | 72 | GST_OBJECT_UNLOCK (sender); |
Matthew Waters | 1894293 | 2017-01-31 20:56:59 +1100 | [diff] [blame] | 73 | } |
| 74 | |
| 75 | void |
| 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 | |
Sebastian Dröge | 950ead9 | 2018-03-15 17:31:50 +0200 | [diff] [blame^] | 82 | GST_OBJECT_LOCK (sender); |
Matthew Waters | 1894293 | 2017-01-31 20:56:59 +1100 | [diff] [blame] | 83 | gst_object_replace ((GstObject **) & sender->rtcp_transport, |
| 84 | GST_OBJECT (transport)); |
Sebastian Dröge | 950ead9 | 2018-03-15 17:31:50 +0200 | [diff] [blame^] | 85 | GST_OBJECT_UNLOCK (sender); |
Matthew Waters | 1894293 | 2017-01-31 20:56:59 +1100 | [diff] [blame] | 86 | } |
| 87 | |
| 88 | static void |
| 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 |
| 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 |
| 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 |
| 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 |
| 137 | gst_webrtc_rtp_sender_init (GstWebRTCRTPSender * webrtc) |
| 138 | { |
| 139 | } |
| 140 | |
| 141 | GstWebRTCRTPSender * |
| 142 | gst_webrtc_rtp_sender_new (GArray * send_encodings /* FIXME */ ) |
| 143 | { |
| 144 | return g_object_new (GST_TYPE_WEBRTC_RTP_SENDER, NULL); |
| 145 | } |