gstreamer/gst-libs/gst/webrtc/rtptransceiver.c

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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-transceiver
* @short_description: RTCRtpTransceiver object
* @title: GstWebRTCRTPTransceiver
* @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPReceiver
*
* <ulink url="https://www.w3.org/TR/webrtc/#rtcrtptransceiver-interface">https://www.w3.org/TR/webrtc/#rtcrtptransceiver-interface</ulink>
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "rtptransceiver.h"
#define GST_CAT_DEFAULT gst_webrtc_rtp_transceiver_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
#define gst_webrtc_rtp_transceiver_parent_class parent_class
G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstWebRTCRTPTransceiver,
gst_webrtc_rtp_transceiver, GST_TYPE_OBJECT,
GST_DEBUG_CATEGORY_INIT (gst_webrtc_rtp_transceiver_debug,
"webrtctransceiver", 0, "webrtctransceiver");
);
enum
{
SIGNAL_0,
LAST_SIGNAL,
};
enum
{
PROP_0,
PROP_MID,
PROP_SENDER,
PROP_RECEIVER,
PROP_STOPPED, // FIXME
PROP_DIRECTION, // FIXME
PROP_MLINE,
};
//static guint gst_webrtc_rtp_transceiver_signals[LAST_SIGNAL] = { 0 };
static void
gst_webrtc_rtp_transceiver_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
switch (prop_id) {
case PROP_SENDER:
webrtc->sender = g_value_dup_object (value);
break;
case PROP_RECEIVER:
webrtc->receiver = g_value_dup_object (value);
break;
case PROP_MLINE:
webrtc->mline = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_webrtc_rtp_transceiver_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
switch (prop_id) {
case PROP_SENDER:
g_value_set_object (value, webrtc->sender);
break;
case PROP_RECEIVER:
g_value_set_object (value, webrtc->receiver);
break;
case PROP_MLINE:
g_value_set_uint (value, webrtc->mline);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_webrtc_rtp_transceiver_constructed (GObject * object)
{
GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
gst_object_set_parent (GST_OBJECT (webrtc->sender), GST_OBJECT (webrtc));
gst_object_set_parent (GST_OBJECT (webrtc->receiver), GST_OBJECT (webrtc));
G_OBJECT_CLASS (parent_class)->constructed (object);
}
static void
gst_webrtc_rtp_transceiver_dispose (GObject * object)
{
GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
if (webrtc->sender) {
GST_OBJECT_PARENT (webrtc->sender) = NULL;
gst_object_unref (webrtc->sender);
}
webrtc->sender = NULL;
if (webrtc->receiver) {
GST_OBJECT_PARENT (webrtc->receiver) = NULL;
gst_object_unref (webrtc->receiver);
}
webrtc->receiver = NULL;
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_webrtc_rtp_transceiver_finalize (GObject * object)
{
GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
g_free (webrtc->mid);
if (webrtc->codec_preferences)
gst_caps_unref (webrtc->codec_preferences);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_webrtc_rtp_transceiver_class_init (GstWebRTCRTPTransceiverClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
gobject_class->get_property = gst_webrtc_rtp_transceiver_get_property;
gobject_class->set_property = gst_webrtc_rtp_transceiver_set_property;
gobject_class->constructed = gst_webrtc_rtp_transceiver_constructed;
gobject_class->dispose = gst_webrtc_rtp_transceiver_dispose;
gobject_class->finalize = gst_webrtc_rtp_transceiver_finalize;
g_object_class_install_property (gobject_class,
PROP_SENDER,
g_param_spec_object ("sender", "Sender",
"The RTP sender for this transceiver",
GST_TYPE_WEBRTC_RTP_SENDER,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_RECEIVER,
g_param_spec_object ("receiver", "Receiver",
"The RTP receiver for this transceiver",
GST_TYPE_WEBRTC_RTP_RECEIVER,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_MLINE,
g_param_spec_uint ("mlineindex", "Media Line Index",
"Index in the SDP of the Media",
0, G_MAXUINT, 0,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
}
static void
gst_webrtc_rtp_transceiver_init (GstWebRTCRTPTransceiver * webrtc)
{
}