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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-sessiondescription
* @short_description: RTCSessionDescription object
* @title: GstWebRTCSessionDescription
*
* <https://www.w3.org/TR/webrtc/#rtcsessiondescription-class>
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "rtcsessiondescription.h"
#define GST_CAT_DEFAULT gst_webrtc_peerconnection_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
/**
* gst_webrtc_sdp_type_to_string:
* @type: a #GstWebRTCSDPType
*
* Returns: the string representation of @type or "unknown" when @type is not
* recognized.
*/
const gchar *
gst_webrtc_sdp_type_to_string (GstWebRTCSDPType type)
{
switch (type) {
case GST_WEBRTC_SDP_TYPE_OFFER:
return "offer";
case GST_WEBRTC_SDP_TYPE_PRANSWER:
return "pranswer";
case GST_WEBRTC_SDP_TYPE_ANSWER:
return "answer";
case GST_WEBRTC_SDP_TYPE_ROLLBACK:
return "rollback";
default:
return "unknown";
}
}
/**
* gst_webrtc_session_description_copy:
* @src: (transfer none): a #GstWebRTCSessionDescription
*
* Returns: (transfer full): a new copy of @src
*/
GstWebRTCSessionDescription *
gst_webrtc_session_description_copy (const GstWebRTCSessionDescription * src)
{
GstWebRTCSessionDescription *ret;
if (!src)
return NULL;
ret = g_new0 (GstWebRTCSessionDescription, 1);
ret->type = src->type;
gst_sdp_message_copy (src->sdp, &ret->sdp);
return ret;
}
/**
* gst_webrtc_session_description_free:
* @desc: (transfer full): a #GstWebRTCSessionDescription
*
* Free @desc and all associated resources
*/
void
gst_webrtc_session_description_free (GstWebRTCSessionDescription * desc)
{
g_return_if_fail (desc != NULL);
gst_sdp_message_free (desc->sdp);
g_free (desc);
}
/**
* gst_webrtc_session_description_new:
* @type: a #GstWebRTCSDPType
* @sdp: (transfer full): a #GstSDPMessage
*
* Returns: (transfer full): a new #GstWebRTCSessionDescription from @type
* and @sdp
*/
GstWebRTCSessionDescription *
gst_webrtc_session_description_new (GstWebRTCSDPType type, GstSDPMessage * sdp)
{
GstWebRTCSessionDescription *ret;
ret = g_new0 (GstWebRTCSessionDescription, 1);
ret->type = type;
ret->sdp = sdp;
return ret;
}
G_DEFINE_BOXED_TYPE_WITH_CODE (GstWebRTCSessionDescription,
gst_webrtc_session_description, gst_webrtc_session_description_copy,
gst_webrtc_session_description_free,
GST_DEBUG_CATEGORY_INIT (gst_webrtc_peerconnection_debug,
"webrtcsessiondescription", 0, "webrtcsessiondescription"));
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