mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-23 18:21:04 +00:00
61 lines
2.1 KiB
C
61 lines
2.1 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.
|
|
*/
|
|
|
|
#ifndef __GST_WEBRTC_SESSION_DESCRIPTION_H__
|
|
#define __GST_WEBRTC_SESSION_DESCRIPTION_H__
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/sdp/sdp.h>
|
|
#include <gst/webrtc/webrtc_fwd.h>
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
GST_WEBRTC_API
|
|
const gchar * gst_webrtc_sdp_type_to_string (GstWebRTCSDPType type);
|
|
|
|
#define GST_TYPE_WEBRTC_SESSION_DESCRIPTION (gst_webrtc_session_description_get_type())
|
|
GST_WEBRTC_API
|
|
GType gst_webrtc_session_description_get_type (void);
|
|
|
|
/**
|
|
* GstWebRTCSessionDescription:
|
|
* @type: the #GstWebRTCSDPType of the description
|
|
* @sdp: the #GstSDPMessage of the description
|
|
*
|
|
* See <https://www.w3.org/TR/webrtc/#rtcsessiondescription-class>
|
|
*/
|
|
struct _GstWebRTCSessionDescription
|
|
{
|
|
GstWebRTCSDPType type;
|
|
GstSDPMessage *sdp;
|
|
};
|
|
|
|
GST_WEBRTC_API
|
|
GstWebRTCSessionDescription * gst_webrtc_session_description_new (GstWebRTCSDPType type, GstSDPMessage *sdp);
|
|
GST_WEBRTC_API
|
|
GstWebRTCSessionDescription * gst_webrtc_session_description_copy (const GstWebRTCSessionDescription * src);
|
|
GST_WEBRTC_API
|
|
void gst_webrtc_session_description_free (GstWebRTCSessionDescription * desc);
|
|
|
|
|
|
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCSessionDescription, gst_webrtc_session_description_free)
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_WEBRTC_PEERCONNECTION_H__ */
|