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145 lines
3.8 KiB
C
145 lines
3.8 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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#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_MID,
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PROP_SENDER,
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PROP_STOPPED,
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PROP_DIRECTION,
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};
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//static guint gst_webrtc_rtp_sender_signals[LAST_SIGNAL] = { 0 };
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void
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gst_webrtc_rtp_sender_set_transport (GstWebRTCRTPSender * sender,
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GstWebRTCDTLSTransport * transport)
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{
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g_return_if_fail (GST_IS_WEBRTC_RTP_SENDER (sender));
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g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
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GST_OBJECT_LOCK (sender);
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gst_object_replace ((GstObject **) & sender->transport,
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GST_OBJECT (transport));
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GST_OBJECT_UNLOCK (sender);
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}
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void
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gst_webrtc_rtp_sender_set_rtcp_transport (GstWebRTCRTPSender * sender,
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GstWebRTCDTLSTransport * transport)
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{
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g_return_if_fail (GST_IS_WEBRTC_RTP_SENDER (sender));
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g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
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GST_OBJECT_LOCK (sender);
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gst_object_replace ((GstObject **) & sender->rtcp_transport,
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GST_OBJECT (transport));
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GST_OBJECT_UNLOCK (sender);
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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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switch (prop_id) {
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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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switch (prop_id) {
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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 *webrtc = GST_WEBRTC_RTP_SENDER (object);
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if (webrtc->transport)
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gst_object_unref (webrtc->transport);
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webrtc->transport = NULL;
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if (webrtc->rtcp_transport)
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gst_object_unref (webrtc->rtcp_transport);
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webrtc->rtcp_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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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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