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

webrtc/tools/agc/test_utils.cc

 /*
  *  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.
  */
 
 #include "webrtc/tools/agc/test_utils.h"
 
 #include <cmath>
 
 #include <algorithm>
 
 #include "webrtc/modules/interface/module_common_types.h"
 
 namespace webrtc {
 
 float MicLevel2Gain(int gain_range_db, int level) {
   return (level - 127.0f) / 128.0f * gain_range_db / 2;
 }
 
 float Db2Linear(float db) {
   return powf(10.0f, db / 20.0f);
 }
 
 void ApplyGainLinear(float gain, float last_gain, AudioFrame* frame) {
-  const int frame_length = frame->samples_per_channel_ * frame->num_channels_;
+  const size_t frame_length =
+      frame->samples_per_channel_ * frame->num_channels_;
   // Smooth the transition between gain levels across the frame.
   float smoothed_gain = last_gain;
   float gain_step = (gain - last_gain) / (frame_length - 1);
-  for (int i = 0; i < frame_length; ++i) {
+  for (size_t i = 0; i < frame_length; ++i) {
     smoothed_gain += gain_step;
     float sample = std::floor(frame->data_[i] * smoothed_gain + 0.5);
     sample = std::max(std::min(32767.0f, sample), -32768.0f);
     frame->data_[i] = static_cast<int16_t>(sample);
   }
 }
 
 void ApplyGain(float gain_db, float last_gain_db, AudioFrame* frame) {
   ApplyGainLinear(Db2Linear(gain_db), Db2Linear(last_gain_db), frame);
 }
 
 void SimulateMic(int gain_range_db, int mic_level, int last_mic_level,
                  AudioFrame* frame) {
   assert(mic_level >= 0 && mic_level <= 255);
   assert(last_mic_level >= 0 && last_mic_level <= 255);
   ApplyGain(MicLevel2Gain(gain_range_db, mic_level),
             MicLevel2Gain(gain_range_db, last_mic_level),
             frame);
 }
 
 void SimulateMic(int gain_map[255], int mic_level, int last_mic_level,
                  AudioFrame* frame) {
   assert(mic_level >= 0 && mic_level <= 255);
   assert(last_mic_level >= 0 && last_mic_level <= 255);
   ApplyGain(gain_map[mic_level], gain_map[last_mic_level], frame);
 }
 
 }  // namespace webrtc