Convert some more things to size_t.

webrtc/modules/audio_device/android/fine_audio_buffer.cc

 /*
  *  Copyright (c) 2013 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/audio_device/android/fine_audio_buffer.h"
 
 #include <memory.h>
 #include <stdio.h>
 #include <algorithm>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_device/audio_device_buffer.h"
 
 namespace webrtc {
 
 FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* device_buffer,
-                                 int desired_frame_size_bytes,
+                                 size_t desired_frame_size_bytes,
                                  int sample_rate)
     : device_buffer_(device_buffer),
       desired_frame_size_bytes_(desired_frame_size_bytes),
       sample_rate_(sample_rate),
-      samples_per_10_ms_(sample_rate_ * 10 / 1000),
+      samples_per_10_ms_(static_cast<size_t>(sample_rate_ * 10 / 1000)),
       bytes_per_10_ms_(samples_per_10_ms_ * sizeof(int16_t)),
       cached_buffer_start_(0),
       cached_bytes_(0) {
   cache_buffer_.reset(new int8_t[bytes_per_10_ms_]);
 }
 
 FineAudioBuffer::~FineAudioBuffer() {
 }
 
-int FineAudioBuffer::RequiredBufferSizeBytes() {
+size_t FineAudioBuffer::RequiredBufferSizeBytes() {
   // It is possible that we store the desired frame size - 1 samples. Since new
   // audio frames are pulled in chunks of 10ms we will need a buffer that can
   // hold desired_frame_size - 1 + 10ms of data. We omit the - 1.
   return desired_frame_size_bytes_ + bytes_per_10_ms_;
 }
 
 void FineAudioBuffer::GetBufferData(int8_t* buffer) {
   if (desired_frame_size_bytes_ <= cached_bytes_) {
     memcpy(buffer, &cache_buffer_.get()[cached_buffer_start_],
            desired_frame_size_bytes_);
     cached_buffer_start_ += desired_frame_size_bytes_;
     cached_bytes_ -= desired_frame_size_bytes_;
     CHECK_LT(cached_buffer_start_ + cached_bytes_, bytes_per_10_ms_);
     return;
   }
   memcpy(buffer, &cache_buffer_.get()[cached_buffer_start_], cached_bytes_);
   // Push another n*10ms of audio to |buffer|. n > 1 if
   // |desired_frame_size_bytes_| is greater than 10ms of audio. Note that we
   // write the audio after the cached bytes copied earlier.
   int8_t* unwritten_buffer = &buffer[cached_bytes_];
-  int bytes_left = desired_frame_size_bytes_ - cached_bytes_;
+  int bytes_left = static_cast<int>(desired_frame_size_bytes_ - cached_bytes_);
   // Ceiling of integer division: 1 + ((x - 1) / y)
-  int number_of_requests = 1 + (bytes_left - 1) / (bytes_per_10_ms_);
-  for (int i = 0; i < number_of_requests; ++i) {
+  size_t number_of_requests = 1 + (bytes_left - 1) / (bytes_per_10_ms_);
+  for (size_t i = 0; i < number_of_requests; ++i) {
     device_buffer_->RequestPlayoutData(samples_per_10_ms_);
     int num_out = device_buffer_->GetPlayoutData(unwritten_buffer);
-    if (num_out != samples_per_10_ms_) {
+    if (static_cast<size_t>(num_out) != samples_per_10_ms_) {
       CHECK_EQ(num_out, 0);
       cached_bytes_ = 0;
       return;
     }
     unwritten_buffer += bytes_per_10_ms_;
     CHECK_GE(bytes_left, 0);
     bytes_left -= bytes_per_10_ms_;
   }
   CHECK_LE(bytes_left, 0);
   // Put the samples that were written to |buffer| but are not used in the
   // cache.
-  int cache_location = desired_frame_size_bytes_;
+  size_t cache_location = desired_frame_size_bytes_;
   int8_t* cache_ptr = &buffer[cache_location];
   cached_bytes_ = number_of_requests * bytes_per_10_ms_ -
       (desired_frame_size_bytes_ - cached_bytes_);
   // If cached_bytes_ is larger than the cache buffer, uninitialized memory
   // will be read.
   CHECK_LE(cached_bytes_, bytes_per_10_ms_);
-  CHECK_EQ(-bytes_left, cached_bytes_);
+  CHECK_EQ(static_cast<size_t>(-bytes_left), cached_bytes_);
   cached_buffer_start_ = 0;
   memcpy(cache_buffer_.get(), cache_ptr, cached_bytes_);
 }
 
 }  // namespace webrtc