| /* 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_EXPORT |
| const gchar * gst_webrtc_sdp_type_to_string (GstWebRTCSDPType type); |
| |
| #define GST_TYPE_WEBRTC_SESSION_DESCRIPTION (gst_webrtc_session_description_get_type()) |
| GST_EXPORT |
| 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_EXPORT |
| GstWebRTCSessionDescription * gst_webrtc_session_description_new (GstWebRTCSDPType type, GstSDPMessage *sdp); |
| GST_EXPORT |
| GstWebRTCSessionDescription * gst_webrtc_session_description_copy (const GstWebRTCSessionDescription * src); |
| GST_EXPORT |
| void gst_webrtc_session_description_free (GstWebRTCSessionDescription * desc); |
| |
| G_END_DECLS |
| |
| #endif /* __GST_WEBRTC_PEERCONNECTION_H__ */ |