1 files changed, 74 insertions, 0 deletions
diff --git a/ext/webrtc/webrtcsctptransport.h b/ext/webrtc/webrtcsctptransport.h
new file mode 100644
index 000000000..82a513911
--- /dev/null
+++ b/ext/webrtc/webrtcsctptransport.h
@@ -0,0 +1,74 @@
+/* GStreamer
+ * Copyright (C) 2018 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 __WEBRTC_SCTP_TRANSPORT_H__
+#define __WEBRTC_SCTP_TRANSPORT_H__
+
+#include <gst/gst.h>
+#include <gst/webrtc/webrtc.h>
+#include <gst/webrtc/sctptransport.h>
+#include "gstwebrtcice.h"
+
+#include "gst/webrtc/webrtc-priv.h"
+
+G_BEGIN_DECLS
+
+GType webrtc_sctp_transport_get_type(void);
+#define TYPE_WEBRTC_SCTP_TRANSPORT            (webrtc_sctp_transport_get_type())
+#define WEBRTC_SCTP_TRANSPORT(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),TYPE_WEBRTC_SCTP_TRANSPORT,WebRTCSCTPTransport))
+#define WEBRTC_IS_SCTP_TRANSPORT(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),TYPE_WEBRTC_SCTP_TRANSPORT))
+#define WEBRTC_SCTP_TRANSPORT_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,TYPE_WEBRTC_SCTP_TRANSPORT,WebRTCSCTPTransportClass))
+#define WEBRTC_SCTP_IS_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,TYPE_WEBRTC_SCTP_TRANSPORT))
+#define WEBRTC_SCTP_TRANSPORT_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,TYPE_WEBRTC_SCTP_TRANSPORT,WebRTCSCTPTransportClass))
+
+typedef struct _WebRTCSCTPTransport WebRTCSCTPTransport;
+typedef struct _WebRTCSCTPTransportClass WebRTCSCTPTransportClass;
+
+struct _WebRTCSCTPTransport
+{
+  GstWebRTCSCTPTransport        parent;
+
+  GstWebRTCDTLSTransport       *transport;
+  GstWebRTCSCTPTransportState   state;
+  guint64                       max_message_size;
+  guint                         max_channels;
+
+  gboolean                      association_established;
+
+  gulong                        sctpdec_block_id;
+  GstElement                   *sctpdec;
+  GstElement                   *sctpenc;
+
+  GstWebRTCBin                 *webrtcbin;
+};
+
+struct _WebRTCSCTPTransportClass
+{
+  GstWebRTCSCTPTransportClass   parent_class;
+};
+
+WebRTCSCTPTransport *           webrtc_sctp_transport_new               (void);
+
+void
+webrtc_sctp_transport_set_priority (WebRTCSCTPTransport *sctp,
+                                    GstWebRTCPriorityType priority);
+
+G_END_DECLS
+
+#endif /* __WEBRTC_SCTP_TRANSPORT_H__ */