gstreamer/gst-libs/gst/webrtc/rtcsessiondescription.h
Tim-Philipp Müller 333f636555 webrtc: GST_EXPORT -> GST_WEBRTC_API
We need different export decorators for the different libs.
For now no actual change though, just rename before the release,
and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:36:33 +00:00

58 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 <ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class">https://www.w3.org/TR/webrtc/#rtcsessiondescription-class</ulink>
*/
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_END_DECLS
#endif /* __GST_WEBRTC_PEERCONNECTION_H__ */