Update audio_processing module

Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1

Update notes:

 * Pull in third party license file

 * Replace .gypi files with BUILD.gn to keep track of what changes
   upstream

 * Bunch of new filse pulled in as dependencies

 * Won't build yet due to changes needed on top of these

webrtc/modules/audio_processing/agc/histogram.h

@@ -0,0 +1,91 @@
+/*
+ *  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_
+
+#include <string.h>
+
+#include "webrtc/base/scoped_ptr.h"
+#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 {
+ public:
+  // Create a non-sliding Histogram.
+  static Histogram* Create();
+
+  // Create a sliding Histogram, i.e. the histogram represents the last
+  // |window_size| samples.
+  static Histogram* Create(int window_size);
+  ~Histogram();
+
+  // 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);
+
+  // 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.
+  int64_t bin_count_q10_[kHistSize];
+
+  // Circular buffer for probabilities
+  rtc::scoped_ptr<int[]> activity_probability_;
+  // Circular buffer for histogram-indices of probabilities.
+  rtc::scoped_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_