Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/rtp_rtcp/source/rtp_utility.h

@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2012 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_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
+
+#include <map>
+
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "webrtc/rtc_base/deprecation.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+const uint8_t kRtpMarkerBitMask = 0x80;
+
+RtpFeedback* NullObjectRtpFeedback();
+
+namespace RtpUtility {
+
+struct Payload {
+  char name[RTP_PAYLOAD_NAME_SIZE];
+  bool audio;
+  PayloadUnion typeSpecific;
+};
+
+bool StringCompare(const char* str1, const char* str2, const uint32_t length);
+
+// Round up to the nearest size that is a multiple of 4.
+size_t Word32Align(size_t size);
+
+class RtpHeaderParser {
+ public:
+  RtpHeaderParser(const uint8_t* rtpData, size_t rtpDataLength);
+  ~RtpHeaderParser();
+
+  bool RTCP() const;
+  bool ParseRtcp(RTPHeader* header) const;
+  bool Parse(RTPHeader* parsedPacket,
+             RtpHeaderExtensionMap* ptrExtensionMap = nullptr) const;
+
+ private:
+  void ParseOneByteExtensionHeader(RTPHeader* parsedPacket,
+                                   const RtpHeaderExtensionMap* ptrExtensionMap,
+                                   const uint8_t* ptrRTPDataExtensionEnd,
+                                   const uint8_t* ptr) const;
+
+  const uint8_t* const _ptrRTPDataBegin;
+  const uint8_t* const _ptrRTPDataEnd;
+};
+}  // namespace RtpUtility
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_