mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-14 13:21:28 +00:00
123 lines
3.3 KiB
C
123 lines
3.3 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-sessiondescription
|
|
* @short_description: RTCSessionDescription object
|
|
* @title: GstWebRTCSessionDescription
|
|
*
|
|
* <ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class">https://www.w3.org/TR/webrtc/#rtcsessiondescription-class</ulink>
|
|
*/
|
|
|
|
#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"));
|