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

webrtc/modules/audio_processing/agc/agc.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/agc/agc.h"
 
 #include <cmath>
 #include <cstdlib>
 
 #include <algorithm>
 #include <vector>
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_processing/agc/histogram.h"
+#include "webrtc/modules/audio_processing/agc/loudness_histogram.h"
 #include "webrtc/modules/audio_processing/agc/utility.h"
 #include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace {
 
 const int kDefaultLevelDbfs = -18;
 const int kNumAnalysisFrames = 100;
 const double kActivityThreshold = 0.3;
 
 }  // namespace
 
 Agc::Agc()
     : target_level_loudness_(Dbfs2Loudness(kDefaultLevelDbfs)),
       target_level_dbfs_(kDefaultLevelDbfs),
-      histogram_(Histogram::Create(kNumAnalysisFrames)),
-      inactive_histogram_(Histogram::Create()) {
-  }
+      histogram_(LoudnessHistogram::Create(kNumAnalysisFrames)),
+      inactive_histogram_(LoudnessHistogram::Create()) {}
 
 Agc::~Agc() {}
 
 float Agc::AnalyzePreproc(const int16_t* audio, size_t length) {
   assert(length > 0);
   size_t num_clipped = 0;
   for (size_t i = 0; i < length; ++i) {
     if (audio[i] == 32767 || audio[i] == -32768)
       ++num_clipped;
   }
@@ skipping 43 matching lines @@
   // limits. The upper limit should be chosen such that the risk of clipping is
   // low. The lower limit should not result in a too quiet signal.
   if (level >= 0 || level <= -100)
     return -1;
   target_level_dbfs_ = level;
   target_level_loudness_ = Dbfs2Loudness(level);
   return 0;
 }
 
 }  // namespace webrtc