Refactor RMSLevel and give it new functionality

webrtc/modules/audio_processing/level_estimator_impl.cc

 /*
  *  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/base/array_view.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
 LevelEstimatorImpl::LevelEstimatorImpl(rtc::CriticalSection* crit)
-    : crit_(crit), rms_(new RMSLevel()) {
+    : 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_->Process(audio->channels_const()[i], audio->num_frames());
+    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_->RMS();
+  return rms_->Average();
 }
 }  // namespace webrtc