| /* 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. |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| # include "config.h" |
| #endif |
| |
| #include "gstwebrtcbin.h" |
| #include "utils.h" |
| #include "webrtctransceiver.h" |
| |
| #define webrtc_transceiver_parent_class parent_class |
| G_DEFINE_TYPE (WebRTCTransceiver, webrtc_transceiver, |
| GST_TYPE_WEBRTC_RTP_TRANSCEIVER); |
| |
| enum |
| { |
| PROP_0, |
| PROP_WEBRTC, |
| }; |
| |
| void |
| webrtc_transceiver_set_transport (WebRTCTransceiver * trans, |
| TransportStream * stream) |
| { |
| GstWebRTCRTPTransceiver *rtp_trans; |
| |
| g_return_if_fail (WEBRTC_IS_TRANSCEIVER (trans)); |
| |
| rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans); |
| |
| gst_object_replace ((GstObject **) & trans->stream, (GstObject *) stream); |
| |
| if (rtp_trans->sender) |
| gst_object_replace ((GstObject **) & rtp_trans->sender->transport, |
| (GstObject *) stream->transport); |
| if (rtp_trans->receiver) |
| gst_object_replace ((GstObject **) & rtp_trans->receiver->transport, |
| (GstObject *) stream->transport); |
| |
| if (rtp_trans->sender) |
| gst_object_replace ((GstObject **) & rtp_trans->sender->rtcp_transport, |
| (GstObject *) stream->rtcp_transport); |
| if (rtp_trans->receiver) |
| gst_object_replace ((GstObject **) & rtp_trans->receiver->rtcp_transport, |
| (GstObject *) stream->rtcp_transport); |
| } |
| |
| static void |
| webrtc_transceiver_set_property (GObject * object, guint prop_id, |
| const GValue * value, GParamSpec * pspec) |
| { |
| WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (object); |
| |
| switch (prop_id) { |
| case PROP_WEBRTC: |
| gst_object_set_parent (GST_OBJECT (trans), g_value_get_object (value)); |
| break; |
| } |
| |
| GST_OBJECT_LOCK (trans); |
| switch (prop_id) { |
| case PROP_WEBRTC: |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
| break; |
| } |
| GST_OBJECT_UNLOCK (trans); |
| } |
| |
| static void |
| webrtc_transceiver_get_property (GObject * object, guint prop_id, |
| GValue * value, GParamSpec * pspec) |
| { |
| WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (object); |
| |
| GST_OBJECT_LOCK (trans); |
| switch (prop_id) { |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
| break; |
| } |
| GST_OBJECT_UNLOCK (trans); |
| } |
| |
| static void |
| webrtc_transceiver_finalize (GObject * object) |
| { |
| WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (object); |
| |
| if (trans->stream) |
| gst_object_unref (trans->stream); |
| trans->stream = NULL; |
| |
| G_OBJECT_CLASS (parent_class)->finalize (object); |
| } |
| |
| static void |
| webrtc_transceiver_class_init (WebRTCTransceiverClass * klass) |
| { |
| GObjectClass *gobject_class = (GObjectClass *) klass; |
| |
| gobject_class->get_property = webrtc_transceiver_get_property; |
| gobject_class->set_property = webrtc_transceiver_set_property; |
| gobject_class->finalize = webrtc_transceiver_finalize; |
| |
| /* some acrobatics are required to set the parent before _constructed() |
| * has been called */ |
| g_object_class_install_property (gobject_class, |
| PROP_WEBRTC, |
| g_param_spec_object ("webrtc", "Parent webrtcbin", |
| "Parent webrtcbin", |
| GST_TYPE_WEBRTC_BIN, |
| G_PARAM_WRITABLE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS)); |
| } |
| |
| static void |
| webrtc_transceiver_init (WebRTCTransceiver * trans) |
| { |
| } |
| |
| WebRTCTransceiver * |
| webrtc_transceiver_new (GstWebRTCBin * webrtc, GstWebRTCRTPSender * sender, |
| GstWebRTCRTPReceiver * receiver) |
| { |
| WebRTCTransceiver *trans; |
| |
| trans = g_object_new (webrtc_transceiver_get_type (), "sender", sender, |
| "receiver", receiver, "webrtc", webrtc, NULL); |
| |
| return trans; |
| } |