webrtc: Split sctptransport into lib and implementation parts

GstWebRTCSCTPTransport is now made into into an abstract base class
that only contains property specifications matching the
RTCSctpTransport interface of the W3C WebRTC specification, see
https://w3c.github.io/webrtc-pc/#rtcsctptransport-interface. This
class is put into the WebRTC library to expose it for applications and
to allow for generation of bindings for non-dynamic languages using
GObject introspection.

The actual implementation is moved to the subclass WebRTCSCTPTransport
located in the WebRTC plugin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2214>

ext/webrtc/gstwebrtcbin.c

@@ -29,7 +29,7 @@
 #include "webrtcsdp.h"
 #include "webrtctransceiver.h"
 #include "webrtcdatachannel.h"
-#include "sctptransport.h"
+#include "webrtcsctptransport.h"
 
 #include "gst/webrtc/webrtc-priv.h"
 
@@ -1986,7 +1986,7 @@ gst_webrtc_bin_update_sctp_priority (GstWebRTCBin * webrtc)
   /* If one stream has a non-default priority, then everyone else does too */
   gst_webrtc_bin_attach_tos (webrtc);
 
-  gst_webrtc_sctp_transport_set_priority (webrtc->priv->sctp_transport,
+  webrtc_sctp_transport_set_priority (webrtc->priv->sctp_transport,
       sctp_priority);
 }
 
@@ -2201,7 +2201,7 @@ _on_sctpdec_pad_added (GstElement * sctpdec, GstPad * pad,
 }
 
 static void
-_on_sctp_state_notify (GstWebRTCSCTPTransport * sctp, GParamSpec * pspec,
+_on_sctp_state_notify (WebRTCSCTPTransport * sctp, GParamSpec * pspec,
     GstWebRTCBin * webrtc)
 {
   GstWebRTCSCTPTransportState state;
@@ -2238,7 +2238,7 @@ _sctp_check_dtls_state_task (GstWebRTCBin * webrtc, gpointer unused)
   TransportStream *stream;
   GstWebRTCDTLSTransport *transport;
   GstWebRTCDTLSTransportState dtls_state;
-  GstWebRTCSCTPTransport *sctp_transport;
+  WebRTCSCTPTransport *sctp_transport;
 
   stream = webrtc->priv->data_channel_transport;
   transport = stream->transport;
@@ -2326,7 +2326,7 @@ _get_or_create_data_channel_transports (GstWebRTCBin * webrtc, guint session_id)
 {
   if (!webrtc->priv->data_channel_transport) {
     TransportStream *stream;
-    GstWebRTCSCTPTransport *sctp_transport;
+    WebRTCSCTPTransport *sctp_transport;
 
     stream = _find_transport_for_session (webrtc, session_id);
 
@@ -2336,7 +2336,7 @@ _get_or_create_data_channel_transports (GstWebRTCBin * webrtc, guint session_id)
     webrtc->priv->data_channel_transport = stream;
 
     if (!(sctp_transport = webrtc->priv->sctp_transport)) {
-      sctp_transport = gst_webrtc_sctp_transport_new ();
+      sctp_transport = webrtc_sctp_transport_new ();
       sctp_transport->transport =
           g_object_ref (webrtc->priv->data_channel_transport->transport);
       sctp_transport->webrtcbin = webrtc;

ext/webrtc/gstwebrtcbin.h

@@ -24,6 +24,7 @@
 #include "fwd.h"
 #include "gstwebrtcice.h"
 #include "transportstream.h"
+#include "webrtcsctptransport.h"
 
 G_BEGIN_DECLS
 
@@ -106,7 +107,7 @@ struct _GstWebRTCBinPrivate
 
   guint jb_latency;
 
-  GstWebRTCSCTPTransport *sctp_transport;
+  WebRTCSCTPTransport *sctp_transport;
   TransportStream *data_channel_transport;
 
   GstWebRTCICE *ice;

ext/webrtc/meson.build

@@ -4,7 +4,7 @@ webrtc_sources = [
   'gstwebrtcstats.c',
   'icestream.c',
   'nicetransport.c',
-  'sctptransport.c',
+  'webrtcsctptransport.c',
   'gstwebrtcbin.c',
   'transportreceivebin.c',
   'transportsendbin.c',

ext/webrtc/webrtcdatachannel.c

@@ -356,7 +356,7 @@ _close_procedure (WebRTCDataChannel * channel, gpointer user_data)
 }
 
 static void
-_on_sctp_stream_reset (GstWebRTCSCTPTransport * sctp, guint stream_id,
+_on_sctp_stream_reset (WebRTCSCTPTransport * sctp, guint stream_id,
     WebRTCDataChannel * channel)
 {
   if (channel->parent.id == stream_id) {
@@ -1003,7 +1003,7 @@ webrtc_data_channel_init (WebRTCDataChannel * channel)
 
 static void
 _data_channel_set_sctp_transport (WebRTCDataChannel * channel,
-    GstWebRTCSCTPTransport * sctp)
+    WebRTCSCTPTransport * sctp)
 {
   g_return_if_fail (GST_IS_WEBRTC_DATA_CHANNEL (channel));
   g_return_if_fail (GST_IS_WEBRTC_SCTP_TRANSPORT (sctp));
@@ -1026,7 +1026,7 @@ _data_channel_set_sctp_transport (WebRTCDataChannel * channel,
 
 void
 webrtc_data_channel_link_to_sctp (WebRTCDataChannel * channel,
-    GstWebRTCSCTPTransport * sctp_transport)
+    WebRTCSCTPTransport * sctp_transport)
 {
   if (sctp_transport && !channel->sctp_transport) {
     gint id;

ext/webrtc/webrtcdatachannel.h

@@ -24,7 +24,7 @@
 #include <gst/webrtc/webrtc_fwd.h>
 #include <gst/webrtc/dtlstransport.h>
 #include <gst/webrtc/datachannel.h>
-#include "sctptransport.h"
+#include "webrtcsctptransport.h"
 
 #include "gst/webrtc/webrtc-priv.h"
 
@@ -45,7 +45,7 @@ struct _WebRTCDataChannel
 {
   GstWebRTCDataChannel              parent;
 
-  GstWebRTCSCTPTransport           *sctp_transport;
+  WebRTCSCTPTransport              *sctp_transport;
   GstElement                       *appsrc;
   GstElement                       *appsink;
 
@@ -68,7 +68,7 @@ struct _WebRTCDataChannelClass
 void    webrtc_data_channel_start_negotiation   (WebRTCDataChannel       *channel);
 G_GNUC_INTERNAL
 void    webrtc_data_channel_link_to_sctp (WebRTCDataChannel                 *channel,
-                                          GstWebRTCSCTPTransport            *sctp_transport);
+                                          WebRTCSCTPTransport               *sctp_transport);
 
 G_END_DECLS
 

ext/webrtc/webrtcsctptransport.c

@@ -23,10 +23,10 @@
 
 #include <stdio.h>
 
-#include "sctptransport.h"
+#include "webrtcsctptransport.h"
 #include "gstwebrtcbin.h"
 
-#define GST_CAT_DEFAULT gst_webrtc_sctp_transport_debug
+#define GST_CAT_DEFAULT webrtc_sctp_transport_debug
 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
 
 enum
@@ -45,18 +45,19 @@ enum
   PROP_MAX_CHANNELS,
 };
 
-static guint gst_webrtc_sctp_transport_signals[LAST_SIGNAL] = { 0 };
+static guint webrtc_sctp_transport_signals[LAST_SIGNAL] = { 0 };
 
-#define gst_webrtc_sctp_transport_parent_class parent_class
+#define webrtc_sctp_transport_parent_class parent_class
-G_DEFINE_TYPE_WITH_CODE (GstWebRTCSCTPTransport, gst_webrtc_sctp_transport,
+G_DEFINE_TYPE_WITH_CODE (WebRTCSCTPTransport, webrtc_sctp_transport,
-    GST_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_sctp_transport_debug,
+    GST_TYPE_WEBRTC_SCTP_TRANSPORT,
+    GST_DEBUG_CATEGORY_INIT (webrtc_sctp_transport_debug,
         "webrtcsctptransport", 0, "webrtcsctptransport"););
 
-typedef void (*SCTPTask) (GstWebRTCSCTPTransport * sctp, gpointer user_data);
+typedef void (*SCTPTask) (WebRTCSCTPTransport * sctp, gpointer user_data);
 
 struct task
 {
-  GstWebRTCSCTPTransport *sctp;
+  WebRTCSCTPTransport *sctp;
   SCTPTask func;
   gpointer user_data;
   GDestroyNotify notify;
@@ -81,7 +82,7 @@ _free_task (struct task *task)
 }
 
 static void
-_sctp_enqueue_task (GstWebRTCSCTPTransport * sctp, SCTPTask func,
+_sctp_enqueue_task (WebRTCSCTPTransport * sctp, SCTPTask func,
     gpointer user_data, GDestroyNotify notify)
 {
   struct task *task = g_new0 (struct task, 1);
@@ -97,17 +98,17 @@ _sctp_enqueue_task (GstWebRTCSCTPTransport * sctp, SCTPTask func,
 }
 
 static void
-_emit_stream_reset (GstWebRTCSCTPTransport * sctp, gpointer user_data)
+_emit_stream_reset (WebRTCSCTPTransport * sctp, gpointer user_data)
 {
   guint stream_id = GPOINTER_TO_UINT (user_data);
 
   g_signal_emit (sctp,
-      gst_webrtc_sctp_transport_signals[ON_STREAM_RESET_SIGNAL], 0, stream_id);
+      webrtc_sctp_transport_signals[ON_STREAM_RESET_SIGNAL], 0, stream_id);
 }
 
 static void
 _on_sctp_dec_pad_removed (GstElement * sctpdec, GstPad * pad,
-    GstWebRTCSCTPTransport * sctp)
+    WebRTCSCTPTransport * sctp)
 {
   guint stream_id;
 
@@ -120,7 +121,7 @@ _on_sctp_dec_pad_removed (GstElement * sctpdec, GstPad * pad,
 
 static void
 _on_sctp_association_established (GstElement * sctpenc, gboolean established,
-    GstWebRTCSCTPTransport * sctp)
+    WebRTCSCTPTransport * sctp)
 {
   GST_OBJECT_LOCK (sctp);
   if (established)
@@ -133,21 +134,8 @@ _on_sctp_association_established (GstElement * sctpenc, gboolean established,
   g_object_notify (G_OBJECT (sctp), "state");
 }
 
-static void
-gst_webrtc_sctp_transport_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec)
-{
-//  GstWebRTCSCTPTransport *sctp = GST_WEBRTC_SCTP_TRANSPORT (object);
-
-  switch (prop_id) {
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
 void
-gst_webrtc_sctp_transport_set_priority (GstWebRTCSCTPTransport * sctp,
+webrtc_sctp_transport_set_priority (WebRTCSCTPTransport * sctp,
     GstWebRTCPriorityType priority)
 {
   GstPad *pad;
@@ -161,10 +149,10 @@ gst_webrtc_sctp_transport_set_priority (GstWebRTCSCTPTransport * sctp,
 }
 
 static void
-gst_webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
+webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec)
 {
-  GstWebRTCSCTPTransport *sctp = GST_WEBRTC_SCTP_TRANSPORT (object);
+  WebRTCSCTPTransport *sctp = WEBRTC_SCTP_TRANSPORT (object);
 
   switch (prop_id) {
     case PROP_TRANSPORT:
@@ -186,9 +174,9 @@ gst_webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
 }
 
 static void
-gst_webrtc_sctp_transport_finalize (GObject * object)
+webrtc_sctp_transport_finalize (GObject * object)
 {
-  GstWebRTCSCTPTransport *sctp = GST_WEBRTC_SCTP_TRANSPORT (object);
+  WebRTCSCTPTransport *sctp = WEBRTC_SCTP_TRANSPORT (object);
 
   g_signal_handlers_disconnect_by_data (sctp->sctpdec, sctp);
   g_signal_handlers_disconnect_by_data (sctp->sctpenc, sctp);
@@ -202,9 +190,9 @@ gst_webrtc_sctp_transport_finalize (GObject * object)
 }
 
 static void
-gst_webrtc_sctp_transport_constructed (GObject * object)
+webrtc_sctp_transport_constructed (GObject * object)
 {
-  GstWebRTCSCTPTransport *sctp = GST_WEBRTC_SCTP_TRANSPORT (object);
+  WebRTCSCTPTransport *sctp = WEBRTC_SCTP_TRANSPORT (object);
   guint association_id;
 
   association_id = g_random_int_range (0, G_MAXUINT16);
@@ -226,61 +214,38 @@ gst_webrtc_sctp_transport_constructed (GObject * object)
 }
 
 static void
-gst_webrtc_sctp_transport_class_init (GstWebRTCSCTPTransportClass * klass)
+webrtc_sctp_transport_class_init (WebRTCSCTPTransportClass * klass)
 {
   GObjectClass *gobject_class = (GObjectClass *) klass;
 
-  gobject_class->constructed = gst_webrtc_sctp_transport_constructed;
+  gobject_class->constructed = webrtc_sctp_transport_constructed;
-  gobject_class->get_property = gst_webrtc_sctp_transport_get_property;
+  gobject_class->get_property = webrtc_sctp_transport_get_property;
-  gobject_class->set_property = gst_webrtc_sctp_transport_set_property;
+  gobject_class->finalize = webrtc_sctp_transport_finalize;
-  gobject_class->finalize = gst_webrtc_sctp_transport_finalize;
 
-  g_object_class_install_property (gobject_class,
+  g_object_class_override_property (gobject_class, PROP_TRANSPORT, "transport");
-      PROP_TRANSPORT,
+  g_object_class_override_property (gobject_class, PROP_STATE, "state");
-      g_param_spec_object ("transport",
+  g_object_class_override_property (gobject_class,
-          "WebRTC DTLS Transport",
+      PROP_MAX_MESSAGE_SIZE, "max-message-size");
-          "DTLS transport used for this SCTP transport",
+  g_object_class_override_property (gobject_class,
-          GST_TYPE_WEBRTC_DTLS_TRANSPORT,
+      PROP_MAX_CHANNELS, "max-channels");
-          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
-
-  g_object_class_install_property (gobject_class,
-      PROP_STATE,
-      g_param_spec_enum ("state",
-          "WebRTC SCTP Transport state", "WebRTC SCTP Transport state",
-          GST_TYPE_WEBRTC_SCTP_TRANSPORT_STATE,
-          GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW,
-          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
-
-  g_object_class_install_property (gobject_class,
-      PROP_MAX_MESSAGE_SIZE,
-      g_param_spec_uint64 ("max-message-size",
-          "Maximum message size",
-          "Maximum message size as reported by the transport", 0, G_MAXUINT64,
-          0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
-
-  g_object_class_install_property (gobject_class,
-      PROP_MAX_CHANNELS,
-      g_param_spec_uint ("max-channels",
-          "Maximum number of channels", "Maximum number of channels",
-          0, G_MAXUINT16, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
 
   /**
-   * GstWebRTCSCTPTransport::stream-reset:
+   * WebRTCSCTPTransport::stream-reset:
-   * @object: the #GstWebRTCSCTPTransport
+   * @object: the #WebRTCSCTPTransport
    * @stream_id: the SCTP stream that was reset
    */
-  gst_webrtc_sctp_transport_signals[ON_STREAM_RESET_SIGNAL] =
+  webrtc_sctp_transport_signals[ON_STREAM_RESET_SIGNAL] =
       g_signal_new ("stream-reset", G_TYPE_FROM_CLASS (klass),
       G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT);
 }
 
 static void
-gst_webrtc_sctp_transport_init (GstWebRTCSCTPTransport * nice)
+webrtc_sctp_transport_init (WebRTCSCTPTransport * nice)
 {
 }
 
-GstWebRTCSCTPTransport *
+WebRTCSCTPTransport *
-gst_webrtc_sctp_transport_new (void)
+webrtc_sctp_transport_new (void)
 {
-  return g_object_new (GST_TYPE_WEBRTC_SCTP_TRANSPORT, NULL);
+  return g_object_new (TYPE_WEBRTC_SCTP_TRANSPORT, NULL);
 }

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__ */

gst-libs/gst/webrtc/meson.build

@@ -6,6 +6,7 @@ webrtc_sources = [
   'rtpsender.c',
   'rtptransceiver.c',
   'datachannel.c',
+  'sctptransport.c',
 ]
 
 webrtc_headers = [
@@ -18,6 +19,7 @@ webrtc_headers = [
   'datachannel.h',
   'webrtc_fwd.h',
   'webrtc.h',
+  'sctptransport.h',
 ]
 
 webrtc_enumtypes_headers = [

gst-libs/gst/webrtc/sctptransport.c

@@ -0,0 +1,79 @@
+/* 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include "sctptransport.h"
+#include "webrtc-priv.h"
+
+G_DEFINE_ABSTRACT_TYPE (GstWebRTCSCTPTransport, gst_webrtc_sctp_transport,
+    GST_TYPE_OBJECT);
+
+static void
+gst_webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  /* all properties should by handled by the plugin class */
+  g_assert_not_reached ();
+}
+
+static void
+gst_webrtc_sctp_transport_class_init (GstWebRTCSCTPTransportClass * klass)
+{
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+  guint property_id_dummy = 0;
+
+  gobject_class->get_property = gst_webrtc_sctp_transport_get_property;
+
+  g_object_class_install_property (gobject_class,
+      ++property_id_dummy,
+      g_param_spec_object ("transport",
+          "WebRTC DTLS Transport",
+          "DTLS transport used for this SCTP transport",
+          GST_TYPE_WEBRTC_DTLS_TRANSPORT,
+          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class,
+      ++property_id_dummy,
+      g_param_spec_enum ("state",
+          "WebRTC SCTP Transport state", "WebRTC SCTP Transport state",
+          GST_TYPE_WEBRTC_SCTP_TRANSPORT_STATE,
+          GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW,
+          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class,
+      ++property_id_dummy,
+      g_param_spec_uint64 ("max-message-size",
+          "Maximum message size",
+          "Maximum message size as reported by the transport", 0, G_MAXUINT64,
+          0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class,
+      ++property_id_dummy,
+      g_param_spec_uint ("max-channels",
+          "Maximum number of channels", "Maximum number of channels",
+          0, G_MAXUINT16, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+}
+
+static void
+gst_webrtc_sctp_transport_init (GstWebRTCSCTPTransport * nice)
+{
+}

gst-libs/gst/webrtc/sctptransport.h

@@ -21,14 +21,13 @@
 #define __GST_WEBRTC_SCTP_TRANSPORT_H__
 
 #include <gst/gst.h>
-/* libnice */
+#include <gst/webrtc/webrtc_fwd.h>
-#include <agent.h>
-#include <gst/webrtc/webrtc.h>
-#include "gstwebrtcice.h"
 
 G_BEGIN_DECLS
 
+GST_WEBRTC_API
 GType gst_webrtc_sctp_transport_get_type(void);
+
 #define GST_TYPE_WEBRTC_SCTP_TRANSPORT            (gst_webrtc_sctp_transport_get_type())
 #define GST_WEBRTC_SCTP_TRANSPORT(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransport))
 #define GST_IS_WEBRTC_SCTP_TRANSPORT(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_SCTP_TRANSPORT))
@@ -36,34 +35,7 @@ GType gst_webrtc_sctp_transport_get_type(void);
 #define GST_IS_WEBRTC_SCTP_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT))
 #define GST_WEBRTC_SCTP_TRANSPORT_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransportClass))
 
-struct _GstWebRTCSCTPTransport
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCSCTPTransport, gst_object_unref)
-{
-  GstObject                     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 _GstWebRTCSCTPTransportClass
-{
-  GstObjectClass                parent_class;
-};
-
-GstWebRTCSCTPTransport *    gst_webrtc_sctp_transport_new               (void);
-
-void
-gst_webrtc_sctp_transport_set_priority (GstWebRTCSCTPTransport *sctp,
-                                        GstWebRTCPriorityType priority);
 
 G_END_DECLS
 

gst-libs/gst/webrtc/webrtc-priv.h

@@ -289,6 +289,27 @@ GST_WEBRTC_API
 void gst_webrtc_data_channel_on_buffered_amount_low (GstWebRTCDataChannel * channel);
 
 
+/**
+ * GstWebRTCSCTPTransport:
+ *
+ * Since: 1.20
+ */
+struct _GstWebRTCSCTPTransport
+{
+  GstObject                     parent;
+};
+
+/**
+ * GstWebRTCSCTPTransportClass:
+ *
+ * Since: 1.20
+ */
+struct _GstWebRTCSCTPTransportClass
+{
+  GstObjectClass                parent_class;
+};
+
+
 G_END_DECLS
 
 #endif /* __GST_WEBRTC_PRIV_H__ */

gst-libs/gst/webrtc/webrtc_fwd.h

@@ -92,6 +92,9 @@ typedef struct _GstWebRTCRTPTransceiverClass GstWebRTCRTPTransceiverClass;
 typedef struct _GstWebRTCDataChannel GstWebRTCDataChannel;
 typedef struct _GstWebRTCDataChannelClass GstWebRTCDataChannelClass;
 
+typedef struct _GstWebRTCSCTPTransport GstWebRTCSCTPTransport;
+typedef struct _GstWebRTCSCTPTransportClass GstWebRTCSCTPTransportClass;
+
 /**
  * GstWebRTCDTLSTransportState:
  * @GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new