1 files changed, 61 insertions, 0 deletions
diff --git a/include/gst/webrtc/rtcsessiondescription.h b/include/gst/webrtc/rtcsessiondescription.h
new file mode 100644
index 0000000000..5308c549a1
--- /dev/null
+++ b/include/gst/webrtc/rtcsessiondescription.h
@@ -0,0 +1,61 @@
+/* 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__ */