Change name of files and class in agc/histogram* in order to avoid file-name clash.

webrtc/modules/audio_processing/agc/loudness_histogram.h

 /*
  *  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_AUDIO_PROCESSING_AGC_HISTOGRAM_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_HISTOGRAM_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LOUDNESS_HISTOGRAM_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LOUDNESS_HISTOGRAM_H_
 
 #include <string.h>
 
 #include <memory>
 
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // This class implements the histogram of loudness with circular buffers so that
 // the histogram tracks the last T seconds of the loudness.
-class Histogram {
+class LoudnessHistogram {
  public:
-  // Create a non-sliding Histogram.
-  static Histogram* Create();
+  // Create a non-sliding LoudnessHistogram.
+  static LoudnessHistogram* Create();
 
-  // Create a sliding Histogram, i.e. the histogram represents the last
+  // Create a sliding LoudnessHistogram, i.e. the histogram represents the last
   // |window_size| samples.
-  static Histogram* Create(int window_size);
-  ~Histogram();
+  static LoudnessHistogram* Create(int window_size);
+  ~LoudnessHistogram();
 
   // Insert RMS and the corresponding activity probability.
   void Update(double rms, double activity_probability);
 
   // Reset the histogram, forget the past.
   void Reset();
 
   // Current loudness, which is actually the mean of histogram in loudness
   // domain.
   double CurrentRms() const;
 
   // Sum of the histogram content.
   double AudioContent() const;
 
   // Number of times the histogram has been updated.
   int num_updates() const { return num_updates_; }
 
  private:
-  Histogram();
-  explicit Histogram(int window);
+  LoudnessHistogram();
+  explicit LoudnessHistogram(int window);
 
   // Find the histogram bin associated with the given |rms|.
   int GetBinIndex(double rms);
 
   void RemoveOldestEntryAndUpdate();
   void InsertNewestEntryAndUpdate(int activity_prob_q10, int hist_index);
   void UpdateHist(int activity_prob_q10, int hist_index);
   void RemoveTransient();
 
   // Number of histogram bins.
   static const int kHistSize = 77;
 
   // Number of times the histogram is updated
   int num_updates_;
   // Audio content, this should be equal to the sum of the components of
   // |bin_count_q10_|.
   int64_t audio_content_q10_;
 
-  // Histogram of input RMS in Q10 with |kHistSize_| bins. In each 'Update(),'
-  // we increment the associated histogram-bin with the given probability. The
-  // increment is implemented in Q10 to avoid rounding errors.
+  // LoudnessHistogram of input RMS in Q10 with |kHistSize_| bins. In each
+  // 'Update(),' we increment the associated histogram-bin with the given
+  // probability. The increment is implemented in Q10 to avoid rounding errors.
   int64_t bin_count_q10_[kHistSize];
 
   // Circular buffer for probabilities
   std::unique_ptr<int[]> activity_probability_;
   // Circular buffer for histogram-indices of probabilities.
   std::unique_ptr<int[]> hist_bin_index_;
   // Current index of circular buffer, where the newest data will be written to,
   // therefore, pointing to the oldest data if buffer is full.
   int buffer_index_;
   // Indicating if buffer is full and we had a wrap around.
   int buffer_is_full_;
   // Size of circular buffer.
   int len_circular_buffer_;
   int len_high_activity_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_HISTOGRAM_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LOUDNESS_HISTOGRAM_H_