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

webrtc/modules/audio_processing/rms_level.cc

 /*
  *  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.
  */
 
@@ skipping 10 matching lines @@
     : sum_square_(0),
       sample_count_(0) {}
 
 RMSLevel::~RMSLevel() {}
 
 void RMSLevel::Reset() {
   sum_square_ = 0;
   sample_count_ = 0;
 }
 
-void RMSLevel::Process(const int16_t* data, int length) {
-  for (int i = 0; i < length; ++i) {
+void RMSLevel::Process(const int16_t* data, size_t length) {
+  for (size_t i = 0; i < length; ++i) {
     sum_square_ += data[i] * data[i];
   }
   sample_count_ += length;
 }
 
-void RMSLevel::ProcessMuted(int length) {
+void RMSLevel::ProcessMuted(size_t length) {
   sample_count_ += length;
 }
 
 int RMSLevel::RMS() {
   if (sample_count_ == 0 || sum_square_ == 0) {
     Reset();
     return kMinLevel;
   }
 
   // Normalize by the max level.
   float rms = sum_square_ / (sample_count_ * kMaxSquaredLevel);
   // 20log_10(x^0.5) = 10log_10(x)
   rms = 10 * log10(rms);
   assert(rms <= 0);
   if (rms < -kMinLevel)
     rms = -kMinLevel;
 
   rms = -rms;
   Reset();
   return static_cast<int>(rms + 0.5);
 }
 
 }  // namespace webrtc