1 files changed, 135 insertions, 0 deletions
diff --git a/gst-libs/gst/webrtc/rtpreceiver.c b/gst-libs/gst/webrtc/rtpreceiver.c
new file mode 100644
index 000000000..edf6e201b
--- /dev/null
+++ b/gst-libs/gst/webrtc/rtpreceiver.c
@@ -0,0 +1,135 @@
+/* GStreamer
+ * Copyright (C) 2017 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.
+ */
+
+/**
+ * SECTION:gstwebrtc-receiver
+ * @short_description: RTCRtpReceiver object
+ * @title: GstWebRTCRTPReceiver
+ * @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPTransceiver
+ *
+ * <ulink url="https://www.w3.org/TR/webrtc/#rtcrtpreceiver-interface">https://www.w3.org/TR/webrtc/#rtcrtpreceiver-interface</ulink>
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include "rtpreceiver.h"
+
+#define GST_CAT_DEFAULT gst_webrtc_rtp_receiver_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+#define gst_webrtc_rtp_receiver_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (GstWebRTCRTPReceiver, gst_webrtc_rtp_receiver,
+    GST_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_rtp_receiver_debug,
+        "webrtcreceiver", 0, "webrtcreceiver"););
+
+enum
+{
+  SIGNAL_0,
+  LAST_SIGNAL,
+};
+
+enum
+{
+  PROP_0,
+};
+
+//static guint gst_webrtc_rtp_receiver_signals[LAST_SIGNAL] = { 0 };
+
+void
+gst_webrtc_rtp_receiver_set_transport (GstWebRTCRTPReceiver * receiver,
+    GstWebRTCDTLSTransport * transport)
+{
+  g_return_if_fail (GST_IS_WEBRTC_RTP_RECEIVER (receiver));
+  g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
+
+  gst_object_replace ((GstObject **) & receiver->transport,
+      GST_OBJECT (transport));
+}
+
+void
+gst_webrtc_rtp_receiver_set_rtcp_transport (GstWebRTCRTPReceiver * receiver,
+    GstWebRTCDTLSTransport * transport)
+{
+  g_return_if_fail (GST_IS_WEBRTC_RTP_RECEIVER (receiver));
+  g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
+
+  gst_object_replace ((GstObject **) & receiver->rtcp_transport,
+      GST_OBJECT (transport));
+}
+
+static void
+gst_webrtc_rtp_receiver_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  switch (prop_id) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_webrtc_rtp_receiver_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  switch (prop_id) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_webrtc_rtp_receiver_finalize (GObject * object)
+{
+  GstWebRTCRTPReceiver *webrtc = GST_WEBRTC_RTP_RECEIVER (object);
+
+  if (webrtc->transport)
+    gst_object_unref (webrtc->transport);
+  webrtc->transport = NULL;
+
+  if (webrtc->rtcp_transport)
+    gst_object_unref (webrtc->rtcp_transport);
+  webrtc->rtcp_transport = NULL;
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static void
+gst_webrtc_rtp_receiver_class_init (GstWebRTCRTPReceiverClass * klass)
+{
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+
+  gobject_class->get_property = gst_webrtc_rtp_receiver_get_property;
+  gobject_class->set_property = gst_webrtc_rtp_receiver_set_property;
+  gobject_class->finalize = gst_webrtc_rtp_receiver_finalize;
+}
+
+static void
+gst_webrtc_rtp_receiver_init (GstWebRTCRTPReceiver * webrtc)
+{
+}
+
+GstWebRTCRTPReceiver *
+gst_webrtc_rtp_receiver_new (void)
+{
+  return g_object_new (GST_TYPE_WEBRTC_RTP_RECEIVER, NULL);
+}