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_header_parser.cc

@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 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.
+ */
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/rtc_base/criticalsection.h"
+
+namespace webrtc {
+
+class RtpHeaderParserImpl : public RtpHeaderParser {
+ public:
+  RtpHeaderParserImpl();
+  virtual ~RtpHeaderParserImpl() {}
+
+  bool Parse(const uint8_t* packet,
+             size_t length,
+             RTPHeader* header) const override;
+
+  bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id) override;
+
+  bool DeregisterRtpHeaderExtension(RTPExtensionType type) override;
+
+ private:
+  rtc::CriticalSection critical_section_;
+  RtpHeaderExtensionMap rtp_header_extension_map_
+      RTC_GUARDED_BY(critical_section_);
+};
+
+RtpHeaderParser* RtpHeaderParser::Create() {
+  return new RtpHeaderParserImpl;
+}
+
+RtpHeaderParserImpl::RtpHeaderParserImpl() {}
+
+bool RtpHeaderParser::IsRtcp(const uint8_t* packet, size_t length) {
+  RtpUtility::RtpHeaderParser rtp_parser(packet, length);
+  return rtp_parser.RTCP();
+}
+
+bool RtpHeaderParserImpl::Parse(const uint8_t* packet,
+                                size_t length,
+                                RTPHeader* header) const {
+  RtpUtility::RtpHeaderParser rtp_parser(packet, length);
+  memset(header, 0, sizeof(*header));
+
+  RtpHeaderExtensionMap map;
+  {
+    rtc::CritScope cs(&critical_section_);
+    map = rtp_header_extension_map_;
+  }
+
+  const bool valid_rtpheader = rtp_parser.Parse(header, &map);
+  if (!valid_rtpheader) {
+    return false;
+  }
+  return true;
+}
+
+bool RtpHeaderParserImpl::RegisterRtpHeaderExtension(RTPExtensionType type,
+                                                     uint8_t id) {
+  rtc::CritScope cs(&critical_section_);
+  return rtp_header_extension_map_.RegisterByType(id, type);
+}
+
+bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RTPExtensionType type) {
+  rtc::CritScope cs(&critical_section_);
+  return rtp_header_extension_map_.Deregister(type) == 0;
+}
+}  // namespace webrtc