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

webrtc/modules/utility/source/audio_frame_operations.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/modules/interface/module_common_types.h"
 #include "webrtc/modules/utility/interface/audio_frame_operations.h"
 
 namespace webrtc {
 
 void AudioFrameOperations::MonoToStereo(const int16_t* src_audio,
-                                        int samples_per_channel,
+                                        size_t samples_per_channel,
                                         int16_t* dst_audio) {
-  for (int i = 0; i < samples_per_channel; i++) {
+  for (size_t i = 0; i < samples_per_channel; i++) {
     dst_audio[2 * i] = src_audio[i];
     dst_audio[2 * i + 1] = src_audio[i];
   }
 }
 
 int AudioFrameOperations::MonoToStereo(AudioFrame* frame) {
   if (frame->num_channels_ != 1) {
     return -1;
   }
   if ((frame->samples_per_channel_ * 2) >= AudioFrame::kMaxDataSizeSamples) {
     // Not enough memory to expand from mono to stereo.
     return -1;
   }
 
   int16_t data_copy[AudioFrame::kMaxDataSizeSamples];
   memcpy(data_copy, frame->data_,
          sizeof(int16_t) * frame->samples_per_channel_);
   MonoToStereo(data_copy, frame->samples_per_channel_, frame->data_);
   frame->num_channels_ = 2;
 
   return 0;
 }
 
 void AudioFrameOperations::StereoToMono(const int16_t* src_audio,
-                                        int samples_per_channel,
+                                        size_t samples_per_channel,
                                         int16_t* dst_audio) {
-  for (int i = 0; i < samples_per_channel; i++) {
+  for (size_t i = 0; i < samples_per_channel; i++) {
     dst_audio[i] = (src_audio[2 * i] + src_audio[2 * i + 1]) >> 1;
   }
 }
 
 int AudioFrameOperations::StereoToMono(AudioFrame* frame) {
   if (frame->num_channels_ != 2) {
     return -1;
   }
 
   StereoToMono(frame->data_, frame->samples_per_channel_, frame->data_);
   frame->num_channels_ = 1;
 
   return 0;
 }
 
 void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) {
   if (frame->num_channels_ != 2) return;
 
-  for (int i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
+  for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
     int16_t temp_data = frame->data_[i];
     frame->data_[i] = frame->data_[i + 1];
     frame->data_[i + 1] = temp_data;
   }
 }
 
 void AudioFrameOperations::Mute(AudioFrame& frame) {
   memset(frame.data_, 0, sizeof(int16_t) *
       frame.samples_per_channel_ * frame.num_channels_);
 }
 
 int AudioFrameOperations::Scale(float left, float right, AudioFrame& frame) {
   if (frame.num_channels_ != 2) {
     return -1;
   }
 
-  for (int i = 0; i < frame.samples_per_channel_; i++) {
+  for (size_t i = 0; i < frame.samples_per_channel_; i++) {
     frame.data_[2 * i] =
         static_cast<int16_t>(left * frame.data_[2 * i]);
     frame.data_[2 * i + 1] =
         static_cast<int16_t>(right * frame.data_[2 * i + 1]);
   }
   return 0;
 }
 
 int AudioFrameOperations::ScaleWithSat(float scale, AudioFrame& frame) {
   int32_t temp_data = 0;
 
   // Ensure that the output result is saturated [-32768, +32767].
-  for (int i = 0; i < frame.samples_per_channel_ * frame.num_channels_;
+  for (size_t i = 0; i < frame.samples_per_channel_ * frame.num_channels_;
        i++) {
     temp_data = static_cast<int32_t>(scale * frame.data_[i]);
     if (temp_data < -32768) {
       frame.data_[i] = -32768;
     } else if (temp_data > 32767) {
       frame.data_[i] = 32767;
     } else {
       frame.data_[i] = static_cast<int16_t>(temp_data);
     }
   }
   return 0;
 }
 
 }  // namespace webrtc