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/agc.h

@@ -0,0 +1,58 @@
+/*
+ *  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_AGC_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_processing/vad/voice_activity_detector.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class AudioFrame;
+class Histogram;
+
+class Agc {
+ public:
+  Agc();
+  virtual ~Agc();
+
+  // Returns the proportion of samples in the buffer which are at full-scale
+  // (and presumably clipped).
+  virtual float AnalyzePreproc(const int16_t* audio, size_t length);
+  // |audio| must be mono; in a multi-channel stream, provide the first (usually
+  // left) channel.
+  virtual int Process(const int16_t* audio, size_t length, int sample_rate_hz);
+
+  // Retrieves the difference between the target RMS level and the current
+  // signal RMS level in dB. Returns true if an update is available and false
+  // otherwise, in which case |error| should be ignored and no action taken.
+  virtual bool GetRmsErrorDb(int* error);
+  virtual void Reset();
+
+  virtual int set_target_level_dbfs(int level);
+  virtual int target_level_dbfs() const { return target_level_dbfs_; }
+
+  virtual float voice_probability() const {
+    return vad_.last_voice_probability();
+  }
+
+ private:
+  double target_level_loudness_;
+  int target_level_dbfs_;
+  rtc::scoped_ptr<Histogram> histogram_;
+  rtc::scoped_ptr<Histogram> inactive_histogram_;
+  VoiceActivityDetector vad_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_H_