1 files changed, 123 insertions, 0 deletions
diff --git a/gst-libs/gst/webrtc/rtcsessiondescription.c b/gst-libs/gst/webrtc/rtcsessiondescription.c
new file mode 100644
index 000000000..3987ab63f
--- /dev/null
+++ b/gst-libs/gst/webrtc/rtcsessiondescription.c
@@ -0,0 +1,123 @@
+/* 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-sessiondescription
+ * @short_description: RTCSessionDescription object
+ * @title: GstWebRTCSessionDescription
+ *
+ * <ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class">https://www.w3.org/TR/webrtc/#rtcsessiondescription-class</ulink>
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include "rtcsessiondescription.h"
+
+#define GST_CAT_DEFAULT gst_webrtc_peerconnection_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+/**
+ * gst_webrtc_sdp_type_to_string:
+ * @type: a #GstWebRTCSDPType
+ *
+ * Returns: the string representation of @type or "unknown" when @type is not
+ *      recognized.
+ */
+const gchar *
+gst_webrtc_sdp_type_to_string (GstWebRTCSDPType type)
+{
+  switch (type) {
+    case GST_WEBRTC_SDP_TYPE_OFFER:
+      return "offer";
+    case GST_WEBRTC_SDP_TYPE_PRANSWER:
+      return "pranswer";
+    case GST_WEBRTC_SDP_TYPE_ANSWER:
+      return "answer";
+    case GST_WEBRTC_SDP_TYPE_ROLLBACK:
+      return "rollback";
+    default:
+      return "unknown";
+  }
+}
+
+/**
+ * gst_webrtc_session_description_copy:
+ * @src: (transfer none): a #GstWebRTCSessionDescription
+ *
+ * Returns: (transfer full): a new copy of @src
+ */
+GstWebRTCSessionDescription *
+gst_webrtc_session_description_copy (const GstWebRTCSessionDescription * src)
+{
+  GstWebRTCSessionDescription *ret;
+
+  if (!src)
+    return NULL;
+
+  ret = g_new0 (GstWebRTCSessionDescription, 1);
+
+  ret->type = src->type;
+  gst_sdp_message_copy (src->sdp, &ret->sdp);
+
+  return ret;
+}
+
+/**
+ * gst_webrtc_session_description_free:
+ * @desc: (transfer full): a #GstWebRTCSessionDescription
+ *
+ * Free @desc and all associated resources
+ */
+void
+gst_webrtc_session_description_free (GstWebRTCSessionDescription * desc)
+{
+  g_return_if_fail (desc != NULL);
+
+  gst_sdp_message_free (desc->sdp);
+  g_free (desc);
+}
+
+/**
+ * gst_webrtc_session_description_new:
+ * @type: a #GstWebRTCSDPType
+ * @sdp: a #GstSDPMessage
+ *
+ * Returns: (transfer full): a new #GstWebRTCSessionDescription from @type
+ *      and @sdp
+ */
+GstWebRTCSessionDescription *
+gst_webrtc_session_description_new (GstWebRTCSDPType type, GstSDPMessage * sdp)
+{
+  GstWebRTCSessionDescription *ret;
+
+  ret = g_new0 (GstWebRTCSessionDescription, 1);
+
+  ret->type = type;
+  ret->sdp = sdp;
+
+  return ret;
+}
+
+G_DEFINE_BOXED_TYPE_WITH_CODE (GstWebRTCSessionDescription,
+    gst_webrtc_session_description, gst_webrtc_session_description_copy,
+    gst_webrtc_session_description_free,
+    GST_DEBUG_CATEGORY_INIT (gst_webrtc_peerconnection_debug,
+        "webrtcsessiondescription", 0, "webrtcsessiondescription"));