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/audio_coding/codecs/isac/locked_bandwidth_info.h

@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2015 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_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
+
+#include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
+#include "webrtc/rtc_base/atomicops.h"
+#include "webrtc/rtc_base/criticalsection.h"
+#include "webrtc/rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+// An IsacBandwidthInfo that's safe to access from multiple threads because
+// it's protected by a mutex.
+class LockedIsacBandwidthInfo final {
+ public:
+  LockedIsacBandwidthInfo();
+  ~LockedIsacBandwidthInfo();
+
+  IsacBandwidthInfo Get() const {
+    rtc::CritScope lock(&lock_);
+    return bwinfo_;
+  }
+
+  void Set(const IsacBandwidthInfo& bwinfo) {
+    rtc::CritScope lock(&lock_);
+    bwinfo_ = bwinfo;
+  }
+
+  int AddRef() const { return rtc::AtomicOps::Increment(&ref_count_); }
+
+  int Release() const {
+    const int count = rtc::AtomicOps::Decrement(&ref_count_);
+    if (count == 0) {
+      delete this;
+    }
+    return count;
+  }
+
+ private:
+  mutable volatile int ref_count_;
+  rtc::CriticalSection lock_;
+  IsacBandwidthInfo bwinfo_ RTC_GUARDED_BY(lock_);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_