1 files changed, 83 insertions, 0 deletions
diff --git a/ext/webrtc/webrtcdatachannel.h b/ext/webrtc/webrtcdatachannel.h
new file mode 100644
index 000000000..769844171
--- /dev/null
+++ b/ext/webrtc/webrtcdatachannel.h
@@ -0,0 +1,83 @@
+/* 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 __GST_WEBRTC_DATA_CHANNEL_H__
+#define __GST_WEBRTC_DATA_CHANNEL_H__
+
+#include <gst/gst.h>
+#include <gst/webrtc/webrtc_fwd.h>
+#include <gst/webrtc/dtlstransport.h>
+#include "sctptransport.h"
+
+G_BEGIN_DECLS
+
+GST_WEBRTC_API
+GType gst_webrtc_data_channel_get_type(void);
+#define GST_TYPE_WEBRTC_DATA_CHANNEL            (gst_webrtc_data_channel_get_type())
+#define GST_WEBRTC_DATA_CHANNEL(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_DATA_CHANNEL,GstWebRTCDataChannel))
+#define GST_IS_WEBRTC_DATA_CHANNEL(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_DATA_CHANNEL))
+#define GST_WEBRTC_DATA_CHANNEL_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_DATA_CHANNEL,GstWebRTCDataChannelClass))
+#define GST_IS_WEBRTC_DATA_CHANNEL_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_DATA_CHANNEL))
+#define GST_WEBRTC_DATA_CHANNEL_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_DATA_CHANNEL,GstWebRTCDataChannelClass))
+
+typedef struct _GstWebRTCDataChannel GstWebRTCDataChannel;
+typedef struct _GstWebRTCDataChannelClass GstWebRTCDataChannelClass;
+
+struct _GstWebRTCDataChannel
+{
+  GstObject                         parent;
+
+  GstWebRTCSCTPTransport           *sctp_transport;
+  GstElement                       *appsrc;
+  GstElement                       *appsink;
+
+  gchar                            *label;
+  gboolean                          ordered;
+  guint                             max_packet_lifetime;
+  guint                             max_retransmits;
+  gchar                            *protocol;
+  gboolean                          negotiated;
+  gint                              id;
+  GstWebRTCPriorityType             priority;
+  GstWebRTCDataChannelState         ready_state;
+  guint64                           buffered_amount;
+  guint64                           buffered_amount_low_threshold;
+
+  GstWebRTCBin                     *webrtcbin;
+  gboolean                          opened;
+  gulong                            src_probe;
+  GError                           *stored_error;
+
+  gpointer                          _padding[GST_PADDING];
+};
+
+struct _GstWebRTCDataChannelClass
+{
+  GstObjectClass            parent_class;
+
+  gpointer                  _padding[GST_PADDING];
+};
+
+void    gst_webrtc_data_channel_start_negotiation   (GstWebRTCDataChannel       *channel);
+void    gst_webrtc_data_channel_set_sctp_transport  (GstWebRTCDataChannel       *channel,
+                                                     GstWebRTCSCTPTransport     *sctp);
+
+G_END_DECLS
+
+#endif /* __GST_WEBRTC_DATA_CHANNEL_H__ */