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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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182 lines
4.8 KiB
C
182 lines
4.8 KiB
C
/* GStreamer
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* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:gstwebrtc-sender
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* @short_description: RTCRtpSender object
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* @title: GstWebRTCRTPSender
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* @see_also: #GstWebRTCRTPReceiver, #GstWebRTCRTPTransceiver
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*
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* <https://www.w3.org/TR/webrtc/#rtcrtpsender-interface>
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include "rtpsender.h"
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#include "rtptransceiver.h"
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#include "webrtc-priv.h"
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#define GST_CAT_DEFAULT gst_webrtc_rtp_sender_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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#define gst_webrtc_rtp_sender_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstWebRTCRTPSender, gst_webrtc_rtp_sender,
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GST_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_rtp_sender_debug,
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"webrtcsender", 0, "webrtcsender");
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);
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enum
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{
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SIGNAL_0,
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LAST_SIGNAL,
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};
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enum
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{
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PROP_0,
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PROP_PRIORITY,
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PROP_TRANSPORT,
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};
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//static guint gst_webrtc_rtp_sender_signals[LAST_SIGNAL] = { 0 };
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/**
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* gst_webrtc_rtp_sender_set_priority:
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* @sender: a #GstWebRTCRTPSender
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* @priority: The priority of this sender
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*
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* Sets the content of the IPv4 Type of Service (ToS), also known as DSCP
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* (Differentiated Services Code Point).
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* This also sets the Traffic Class field of IPv6.
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*
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* Since: 1.20
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*/
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void
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gst_webrtc_rtp_sender_set_priority (GstWebRTCRTPSender * sender,
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GstWebRTCPriorityType priority)
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{
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GST_OBJECT_LOCK (sender);
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sender->priority = priority;
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GST_OBJECT_UNLOCK (sender);
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g_object_notify (G_OBJECT (sender), "priority");
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}
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static void
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gst_webrtc_rtp_sender_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstWebRTCRTPSender *sender = GST_WEBRTC_RTP_SENDER (object);
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switch (prop_id) {
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case PROP_PRIORITY:
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gst_webrtc_rtp_sender_set_priority (sender, g_value_get_uint (value));
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_webrtc_rtp_sender_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstWebRTCRTPSender *sender = GST_WEBRTC_RTP_SENDER (object);
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switch (prop_id) {
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case PROP_PRIORITY:
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GST_OBJECT_LOCK (sender);
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g_value_set_uint (value, sender->priority);
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GST_OBJECT_UNLOCK (sender);
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break;
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case PROP_TRANSPORT:
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GST_OBJECT_LOCK (sender);
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g_value_set_object (value, sender->transport);
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GST_OBJECT_UNLOCK (sender);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_webrtc_rtp_sender_finalize (GObject * object)
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{
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GstWebRTCRTPSender *sender = GST_WEBRTC_RTP_SENDER (object);
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if (sender->transport)
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gst_object_unref (sender->transport);
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sender->transport = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_webrtc_rtp_sender_class_init (GstWebRTCRTPSenderClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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gobject_class->get_property = gst_webrtc_rtp_sender_get_property;
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gobject_class->set_property = gst_webrtc_rtp_sender_set_property;
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gobject_class->finalize = gst_webrtc_rtp_sender_finalize;
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/**
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* GstWebRTCRTPSender:priority:
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*
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* The priority from which to set the DSCP field on packets
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*
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* Since: 1.20
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*/
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g_object_class_install_property (gobject_class,
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PROP_PRIORITY,
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g_param_spec_enum ("priority",
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"Priority",
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"The priority from which to set the DSCP field on packets",
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GST_TYPE_WEBRTC_PRIORITY_TYPE, GST_WEBRTC_PRIORITY_TYPE_LOW,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstWebRTCRTPSender:transport:
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*
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* The DTLS transport for this sender
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*
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* Since: 1.20
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*/
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g_object_class_install_property (gobject_class,
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PROP_TRANSPORT,
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g_param_spec_object ("transport", "Transport",
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"The DTLS transport for this sender",
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GST_TYPE_WEBRTC_DTLS_TRANSPORT,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_webrtc_rtp_sender_init (GstWebRTCRTPSender * webrtc)
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{
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}
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GstWebRTCRTPSender *
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gst_webrtc_rtp_sender_new (void)
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{
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return g_object_new (GST_TYPE_WEBRTC_RTP_SENDER, NULL);
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}
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