132 lines
3.5 KiB
C
132 lines
3.5 KiB
C
/* 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
|
|
*
|
|
* <https://www.w3.org/TR/webrtc/#rtcrtpreceiver-interface>
|
|
*/
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
# include "config.h"
|
|
#endif
|
|
|
|
#include "rtpreceiver.h"
|
|
#include "webrtc-priv.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,
|
|
PROP_TRANSPORT,
|
|
};
|
|
|
|
//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)
|
|
{
|
|
GstWebRTCRTPReceiver *receiver = GST_WEBRTC_RTP_RECEIVER (object);
|
|
switch (prop_id) {
|
|
case PROP_TRANSPORT:
|
|
GST_OBJECT_LOCK (receiver);
|
|
g_value_set_object (value, receiver->transport);
|
|
GST_OBJECT_UNLOCK (receiver);
|
|
break;
|
|
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;
|
|
|
|
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;
|
|
|
|
/**
|
|
* GstWebRTCRTPReceiver:transport:
|
|
*
|
|
* The DTLS transport for this receiver
|
|
*
|
|
* Since: 1.20
|
|
*/
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_TRANSPORT,
|
|
g_param_spec_object ("transport", "Transport",
|
|
"The DTLS transport for this receiver",
|
|
GST_TYPE_WEBRTC_DTLS_TRANSPORT,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
}
|
|
|
|
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);
|
|
}
|