Update a ton of audio code to use size_t more correctly and in general reduce

webrtc/modules/audio_processing/agc/agc.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_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, int length);
+  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, int length, int sample_rate_hz);
+  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_