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>

1 files changed, 251 insertions, 0 deletions

diff --git a/ext/webrtc/webrtcsctptransport.c b/ext/webrtc/webrtcsctptransport.c
new file mode 100644
index 000000000..c65dd1973
--- /dev/null
+++ b/ext/webrtc/webrtcsctptransport.c
@@ -0,0 +1,251 @@
+/* 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 <stdio.h>
+
+#include "webrtcsctptransport.h"
+#include "gstwebrtcbin.h"
+
+#define GST_CAT_DEFAULT webrtc_sctp_transport_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+enum
+{
+  SIGNAL_0,
+  ON_STREAM_RESET_SIGNAL,
+  LAST_SIGNAL,
+};
+
+enum
+{
+  PROP_0,
+  PROP_TRANSPORT,
+  PROP_STATE,
+  PROP_MAX_MESSAGE_SIZE,
+  PROP_MAX_CHANNELS,
+};
+
+static guint webrtc_sctp_transport_signals[LAST_SIGNAL] = { 0 };
+
+#define webrtc_sctp_transport_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (WebRTCSCTPTransport, webrtc_sctp_transport,
+    GST_TYPE_WEBRTC_SCTP_TRANSPORT,
+    GST_DEBUG_CATEGORY_INIT (webrtc_sctp_transport_debug,
+        "webrtcsctptransport", 0, "webrtcsctptransport"););
+
+typedef void (*SCTPTask) (WebRTCSCTPTransport * sctp, gpointer user_data);
+
+struct task
+{
+  WebRTCSCTPTransport *sctp;
+  SCTPTask func;
+  gpointer user_data;
+  GDestroyNotify notify;
+};
+
+static GstStructure *
+_execute_task (GstWebRTCBin * webrtc, struct task *task)
+{
+  if (task->func)
+    task->func (task->sctp, task->user_data);
+  return NULL;
+}
+
+static void
+_free_task (struct task *task)
+{
+  gst_object_unref (task->sctp);
+
+  if (task->notify)
+    task->notify (task->user_data);
+  g_free (task);
+}
+
+static void
+_sctp_enqueue_task (WebRTCSCTPTransport * sctp, SCTPTask func,
+    gpointer user_data, GDestroyNotify notify)
+{
+  struct task *task = g_new0 (struct task, 1);
+
+  task->sctp = gst_object_ref (sctp);
+  task->func = func;
+  task->user_data = user_data;
+  task->notify = notify;
+
+  gst_webrtc_bin_enqueue_task (sctp->webrtcbin,
+      (GstWebRTCBinFunc) _execute_task, task, (GDestroyNotify) _free_task,
+      NULL);
+}
+
+static void
+_emit_stream_reset (WebRTCSCTPTransport * sctp, gpointer user_data)
+{
+  guint stream_id = GPOINTER_TO_UINT (user_data);
+
+  g_signal_emit (sctp,
+      webrtc_sctp_transport_signals[ON_STREAM_RESET_SIGNAL], 0, stream_id);
+}
+
+static void
+_on_sctp_dec_pad_removed (GstElement * sctpdec, GstPad * pad,
+    WebRTCSCTPTransport * sctp)
+{
+  guint stream_id;
+
+  if (sscanf (GST_PAD_NAME (pad), "src_%u", &stream_id) != 1)
+    return;
+
+  _sctp_enqueue_task (sctp, (SCTPTask) _emit_stream_reset,
+      GUINT_TO_POINTER (stream_id), NULL);
+}
+
+static void
+_on_sctp_association_established (GstElement * sctpenc, gboolean established,
+    WebRTCSCTPTransport * sctp)
+{
+  GST_OBJECT_LOCK (sctp);
+  if (established)
+    sctp->state = GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED;
+  else
+    sctp->state = GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED;
+  sctp->association_established = established;
+  GST_OBJECT_UNLOCK (sctp);
+
+  g_object_notify (G_OBJECT (sctp), "state");
+}
+
+void
+webrtc_sctp_transport_set_priority (WebRTCSCTPTransport * sctp,
+    GstWebRTCPriorityType priority)
+{
+  GstPad *pad;
+
+  pad = gst_element_get_static_pad (sctp->sctpenc, "src");
+  gst_pad_push_event (pad,
+      gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_STICKY,
+          gst_structure_new ("GstWebRtcBinUpdateTos", "sctp-priority",
+              GST_TYPE_WEBRTC_PRIORITY_TYPE, priority, NULL)));
+  gst_object_unref (pad);
+}
+
+static void
+webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  WebRTCSCTPTransport *sctp = WEBRTC_SCTP_TRANSPORT (object);
+
+  switch (prop_id) {
+    case PROP_TRANSPORT:
+      g_value_set_object (value, sctp->transport);
+      break;
+    case PROP_STATE:
+      g_value_set_enum (value, sctp->state);
+      break;
+    case PROP_MAX_MESSAGE_SIZE:
+      g_value_set_uint64 (value, sctp->max_message_size);
+      break;
+    case PROP_MAX_CHANNELS:
+      g_value_set_uint (value, sctp->max_channels);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+webrtc_sctp_transport_finalize (GObject * 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);
+
+  gst_object_unref (sctp->sctpdec);
+  gst_object_unref (sctp->sctpenc);
+
+  g_clear_object (&sctp->transport);
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static void
+webrtc_sctp_transport_constructed (GObject * object)
+{
+  WebRTCSCTPTransport *sctp = WEBRTC_SCTP_TRANSPORT (object);
+  guint association_id;
+
+  association_id = g_random_int_range (0, G_MAXUINT16);
+
+  sctp->sctpdec =
+      g_object_ref_sink (gst_element_factory_make ("sctpdec", NULL));
+  g_object_set (sctp->sctpdec, "sctp-association-id", association_id, NULL);
+  sctp->sctpenc =
+      g_object_ref_sink (gst_element_factory_make ("sctpenc", NULL));
+  g_object_set (sctp->sctpenc, "sctp-association-id", association_id, NULL);
+  g_object_set (sctp->sctpenc, "use-sock-stream", TRUE, NULL);
+
+  g_signal_connect (sctp->sctpdec, "pad-removed",
+      G_CALLBACK (_on_sctp_dec_pad_removed), sctp);
+  g_signal_connect (sctp->sctpenc, "sctp-association-established",
+      G_CALLBACK (_on_sctp_association_established), sctp);
+
+  G_OBJECT_CLASS (parent_class)->constructed (object);
+}
+
+static void
+webrtc_sctp_transport_class_init (WebRTCSCTPTransportClass * klass)
+{
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+
+  gobject_class->constructed = webrtc_sctp_transport_constructed;
+  gobject_class->get_property = webrtc_sctp_transport_get_property;
+  gobject_class->finalize = webrtc_sctp_transport_finalize;
+
+  g_object_class_override_property (gobject_class, PROP_TRANSPORT, "transport");
+  g_object_class_override_property (gobject_class, PROP_STATE, "state");
+  g_object_class_override_property (gobject_class,
+      PROP_MAX_MESSAGE_SIZE, "max-message-size");
+  g_object_class_override_property (gobject_class,
+      PROP_MAX_CHANNELS, "max-channels");
+
+  /**
+   * WebRTCSCTPTransport::stream-reset:
+   * @object: the #WebRTCSCTPTransport
+   * @stream_id: the SCTP stream that was reset
+   */
+  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
+webrtc_sctp_transport_init (WebRTCSCTPTransport * nice)
+{
+}
+
+WebRTCSCTPTransport *
+webrtc_sctp_transport_new (void)
+{
+  return g_object_new (TYPE_WEBRTC_SCTP_TRANSPORT, NULL);
+}