/* GStreamer * Copyright (C) 2017 Matthew Waters * * 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. */ /** * SECTION:gstwebrtc-receiver * @short_description: RTCRtpReceiver object * @title: GstWebRTCRTPReceiver * @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPTransceiver * * */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include "rtpreceiver.h" #define GST_CAT_DEFAULT gst_webrtc_rtp_receiver_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); #define gst_webrtc_rtp_receiver_parent_class parent_class G_DEFINE_TYPE_WITH_CODE (GstWebRTCRTPReceiver, gst_webrtc_rtp_receiver, GST_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_rtp_receiver_debug, "webrtcreceiver", 0, "webrtcreceiver");); enum { SIGNAL_0, LAST_SIGNAL, }; enum { PROP_0, }; //static guint gst_webrtc_rtp_receiver_signals[LAST_SIGNAL] = { 0 }; static void gst_webrtc_rtp_receiver_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { switch (prop_id) { default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_webrtc_rtp_receiver_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { switch (prop_id) { default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_webrtc_rtp_receiver_finalize (GObject * object) { GstWebRTCRTPReceiver *webrtc = GST_WEBRTC_RTP_RECEIVER (object); if (webrtc->transport) gst_object_unref (webrtc->transport); webrtc->transport = NULL; if (webrtc->rtcp_transport) gst_object_unref (webrtc->rtcp_transport); webrtc->rtcp_transport = NULL; G_OBJECT_CLASS (parent_class)->finalize (object); } static void gst_webrtc_rtp_receiver_class_init (GstWebRTCRTPReceiverClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; gobject_class->get_property = gst_webrtc_rtp_receiver_get_property; gobject_class->set_property = gst_webrtc_rtp_receiver_set_property; gobject_class->finalize = gst_webrtc_rtp_receiver_finalize; } static void gst_webrtc_rtp_receiver_init (GstWebRTCRTPReceiver * webrtc) { } GstWebRTCRTPReceiver * gst_webrtc_rtp_receiver_new (void) { return g_object_new (GST_TYPE_WEBRTC_RTP_RECEIVER, NULL); }