Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002)
Reland the base->rtc_base without adding stub headers (will be done in follow-up CL). This preserves git blame history of all files. BUG=webrtc:7634 NOTRY=True TBR=kwiberg@webrtc.org Change-Id: Iea3bb6f3f67b8374c96337b63e8f5aa3e6181012 Reviewed-on: https://chromium-review.googlesource.com/554611 Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#18821}
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/*
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* Copyright 2013 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_BASE_DSCP_H_
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#define WEBRTC_BASE_DSCP_H_
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namespace rtc {
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// Differentiated Services Code Point.
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// See http://tools.ietf.org/html/rfc2474 for details.
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enum DiffServCodePoint {
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DSCP_NO_CHANGE = -1,
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DSCP_DEFAULT = 0, // Same as DSCP_CS0
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DSCP_CS0 = 0, // The default
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DSCP_CS1 = 8, // Bulk/background traffic
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DSCP_AF11 = 10,
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DSCP_AF12 = 12,
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DSCP_AF13 = 14,
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DSCP_CS2 = 16,
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DSCP_AF21 = 18,
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DSCP_AF22 = 20,
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DSCP_AF23 = 22,
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DSCP_CS3 = 24,
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DSCP_AF31 = 26,
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DSCP_AF32 = 28,
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DSCP_AF33 = 30,
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DSCP_CS4 = 32,
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DSCP_AF41 = 34, // Video
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DSCP_AF42 = 36, // Video
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DSCP_AF43 = 38, // Video
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DSCP_CS5 = 40, // Video
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DSCP_EF = 46, // Voice
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DSCP_CS6 = 48, // Voice
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DSCP_CS7 = 56, // Control messages
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};
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} // namespace rtc
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#endif // WEBRTC_BASE_DSCP_H_
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