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-rw-r--r--ext/webrtc/sctptransport.c270
1 files changed, 270 insertions, 0 deletions
diff --git a/ext/webrtc/sctptransport.c b/ext/webrtc/sctptransport.c
new file mode 100644
index 000000000..f5643e9fe
--- /dev/null
+++ b/ext/webrtc/sctptransport.c
@@ -0,0 +1,270 @@
+/* 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 "sctptransport.h"
+#include "gstwebrtcbin.h"
+
+#define GST_CAT_DEFAULT gst_webrtc_sctp_transport_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+enum
+{
+ SIGNAL_0,
+ ON_RESET_STREAM_SIGNAL,
+ LAST_SIGNAL,
+};
+
+enum
+{
+ PROP_0,
+ PROP_TRANSPORT,
+ PROP_STATE,
+ PROP_MAX_MESSAGE_SIZE,
+ PROP_MAX_CHANNELS,
+};
+
+static guint gst_webrtc_sctp_transport_signals[LAST_SIGNAL] = { 0 };
+
+#define gst_webrtc_sctp_transport_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (GstWebRTCSCTPTransport, gst_webrtc_sctp_transport,
+ GST_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_sctp_transport_debug,
+ "webrtcsctptransport", 0, "webrtcsctptransport"););
+
+typedef void (*SCTPTask) (GstWebRTCSCTPTransport * sctp, gpointer user_data);
+
+struct task
+{
+ GstWebRTCSCTPTransport *sctp;
+ SCTPTask func;
+ gpointer user_data;
+ GDestroyNotify notify;
+};
+
+static void
+_execute_task (GstWebRTCBin * webrtc, struct task *task)
+{
+ if (task->func)
+ task->func (task->sctp, task->user_data);
+}
+
+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 (GstWebRTCSCTPTransport * 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);
+}
+
+static void
+_emit_stream_reset (GstWebRTCSCTPTransport * sctp, gpointer user_data)
+{
+ guint stream_id = GPOINTER_TO_UINT (user_data);
+
+ g_signal_emit (sctp,
+ gst_webrtc_sctp_transport_signals[ON_RESET_STREAM_SIGNAL], 0, stream_id);
+}
+
+static void
+_on_sctp_dec_pad_removed (GstElement * sctpdec, GstPad * pad,
+ GstWebRTCSCTPTransport * 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,
+ GstWebRTCSCTPTransport * 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");
+}
+
+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;
+ }
+}
+
+static void
+gst_webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstWebRTCSCTPTransport *sctp = GST_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
+gst_webrtc_sctp_transport_finalize (GObject * object)
+{
+ GstWebRTCSCTPTransport *sctp = GST_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
+gst_webrtc_sctp_transport_constructed (GObject * object)
+{
+ GstWebRTCSCTPTransport *sctp = GST_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_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
+gst_webrtc_sctp_transport_class_init (GstWebRTCSCTPTransportClass * klass)
+{
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+
+ gobject_class->constructed = gst_webrtc_sctp_transport_constructed;
+ gobject_class->get_property = gst_webrtc_sctp_transport_get_property;
+ gobject_class->set_property = gst_webrtc_sctp_transport_set_property;
+ gobject_class->finalize = gst_webrtc_sctp_transport_finalize;
+
+ g_object_class_install_property (gobject_class,
+ PROP_TRANSPORT,
+ 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,
+ 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::reset-stream:
+ * @object: the #GstWebRTCSCTPTransport
+ * @stream_id: the SCTP stream that was reset
+ */
+ gst_webrtc_sctp_transport_signals[ON_RESET_STREAM_SIGNAL] =
+ g_signal_new ("stream-reset", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 1, G_TYPE_UINT);
+}
+
+static void
+gst_webrtc_sctp_transport_init (GstWebRTCSCTPTransport * nice)
+{
+}
+
+GstWebRTCSCTPTransport *
+gst_webrtc_sctp_transport_new (void)
+{
+ return g_object_new (GST_TYPE_WEBRTC_SCTP_TRANSPORT, NULL);
+}
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