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/* 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.
*/
/**
* SECTION:gstwebrtc-receiver
* @short_description: RTCRtpReceiver object
* @title: GstWebRTCRTPReceiver
* @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPTransceiver
*
* <ulink url="https://www.w3.org/TR/webrtc/#rtcrtpreceiver-interface">https://www.w3.org/TR/webrtc/#rtcrtpreceiver-interface</ulink>
*/
#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 };
void
gst_webrtc_rtp_receiver_set_transport (GstWebRTCRTPReceiver * receiver,
GstWebRTCDTLSTransport * transport)
{
g_return_if_fail (GST_IS_WEBRTC_RTP_RECEIVER (receiver));
g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
gst_object_replace ((GstObject **) & receiver->transport,
GST_OBJECT (transport));
}
void
gst_webrtc_rtp_receiver_set_rtcp_transport (GstWebRTCRTPReceiver * receiver,
GstWebRTCDTLSTransport * transport)
{
g_return_if_fail (GST_IS_WEBRTC_RTP_RECEIVER (receiver));
g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
gst_object_replace ((GstObject **) & receiver->rtcp_transport,
GST_OBJECT (transport));
}
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);
}
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