Update audio code to use size_t more correctly, webrtc/modules/audio_processing/

webrtc/modules/audio_processing/rms_level.h

 /*
  *  Copyright (c) 2014 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_RMS_LEVEL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
 
+#include <cstddef>
+
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // Computes the root mean square (RMS) level in dBFs (decibels from digital
 // full-scale) of audio data. The computation follows RFC 6465:
 // https://tools.ietf.org/html/rfc6465
 // with the intent that it can provide the RTP audio level indication.
 //
 // The expected approach is to provide constant-sized chunks of audio to
 // Process(). When enough chunks have been accumulated to form a packet, call
 // RMS() to get the audio level indicator for the RTP header.
 class RMSLevel {
  public:
   static const int kMinLevel = 127;
 
   RMSLevel();
   ~RMSLevel();
 
   // Can be called to reset internal states, but is not required during normal
   // operation.
   void Reset();
 
   // Pass each chunk of audio to Process() to accumulate the level.
-  void Process(const int16_t* data, int length);
+  void Process(const int16_t* data, size_t length);
 
   // If all samples with the given |length| have a magnitude of zero, this is
   // a shortcut to avoid some computation.
-  void ProcessMuted(int length);
+  void ProcessMuted(size_t length);
 
   // Computes the RMS level over all data passed to Process() since the last
   // call to RMS(). The returned value is positive but should be interpreted as
   // negative as per the RFC. It is constrained to [0, 127].
   int RMS();
 
  private:
   float sum_square_;
-  int sample_count_;
+  size_t sample_count_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_