Adds a modified copy of talk/base to webrtc/base. It is the first step in
migrating talk/base to webrtc/base.
BUG=N/A
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17479005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/base/dscp.h b/webrtc/base/dscp.h
new file mode 100644
index 0000000..970ff93
--- /dev/null
+++ b/webrtc/base/dscp.h
@@ -0,0 +1,45 @@
+/*
+ * Copyright 2013 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_DSCP_H_
+#define WEBRTC_BASE_DSCP_H_
+
+namespace rtc {
+// Differentiated Services Code Point.
+// See http://tools.ietf.org/html/rfc2474 for details.
+enum DiffServCodePoint {
+ DSCP_NO_CHANGE = -1,
+ DSCP_DEFAULT = 0, // Same as DSCP_CS0
+ DSCP_CS0 = 0, // The default
+ DSCP_CS1 = 8, // Bulk/background traffic
+ DSCP_AF11 = 10,
+ DSCP_AF12 = 12,
+ DSCP_AF13 = 14,
+ DSCP_CS2 = 16,
+ DSCP_AF21 = 18,
+ DSCP_AF22 = 20,
+ DSCP_AF23 = 22,
+ DSCP_CS3 = 24,
+ DSCP_AF31 = 26,
+ DSCP_AF32 = 28,
+ DSCP_AF33 = 30,
+ DSCP_CS4 = 32,
+ DSCP_AF41 = 34, // Video
+ DSCP_AF42 = 36, // Video
+ DSCP_AF43 = 38, // Video
+ DSCP_CS5 = 40, // Video
+ DSCP_EF = 46, // Voice
+ DSCP_CS6 = 48, // Voice
+ DSCP_CS7 = 56, // Control messages
+};
+
+} // namespace rtc
+
+ #endif // WEBRTC_BASE_DSCP_H_