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_processing/level_estimator_impl.cc

@@ -0,0 +1,66 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/modules/audio_processing/level_estimator_impl.h"
+
+#include "webrtc/api/array_view.h"
+#include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/modules/audio_processing/rms_level.h"
+
+namespace webrtc {
+
+LevelEstimatorImpl::LevelEstimatorImpl(rtc::CriticalSection* crit)
+    : crit_(crit), rms_(new RmsLevel()) {
+  RTC_DCHECK(crit);
+}
+
+LevelEstimatorImpl::~LevelEstimatorImpl() {}
+
+void LevelEstimatorImpl::Initialize() {
+  rtc::CritScope cs(crit_);
+  rms_->Reset();
+}
+
+void LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) {
+  RTC_DCHECK(audio);
+  rtc::CritScope cs(crit_);
+  if (!enabled_) {
+    return;
+  }
+
+  for (size_t i = 0; i < audio->num_channels(); i++) {
+    rms_->Analyze(rtc::ArrayView<const int16_t>(audio->channels_const()[i],
+                                                audio->num_frames()));
+  }
+}
+
+int LevelEstimatorImpl::Enable(bool enable) {
+  rtc::CritScope cs(crit_);
+  if (enable && !enabled_) {
+    rms_->Reset();
+  }
+  enabled_ = enable;
+  return AudioProcessing::kNoError;
+}
+
+bool LevelEstimatorImpl::is_enabled() const {
+  rtc::CritScope cs(crit_);
+  return enabled_;
+}
+
+int LevelEstimatorImpl::RMS() {
+  rtc::CritScope cs(crit_);
+  if (!enabled_) {
+    return AudioProcessing::kNotEnabledError;
+  }
+
+  return rms_->Average();
+}
+}  // namespace webrtc