Convert some more things to size_t.

webrtc/modules/audio_device/android/fine_audio_buffer.h

 /*
  *  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.
  */
 
@@ skipping 14 matching lines @@
 // samples.
 class FineAudioBuffer {
  public:
   // |device_buffer| is a buffer that provides 10ms of audio data.
   // |desired_frame_size_bytes| is the number of bytes of audio data
   // (not samples) |GetBufferData| should return on success.
   // |sample_rate| is the sample rate of the audio data. This is needed because
   // |device_buffer| delivers 10ms of data. Given the sample rate the number
   // of samples can be calculated.
   FineAudioBuffer(AudioDeviceBuffer* device_buffer,
-                  int desired_frame_size_bytes,
+                  size_t desired_frame_size_bytes,
                   int sample_rate);
   ~FineAudioBuffer();
 
   // Returns the required size of |buffer| when calling GetBufferData. If the
   // buffer is smaller memory trampling will happen.
   // |desired_frame_size_bytes| and |samples_rate| are as described in the
   // constructor.
-  int RequiredBufferSizeBytes();
+  size_t RequiredBufferSizeBytes();
 
   // |buffer| must be of equal or greater size than what is returned by
   // RequiredBufferSize. This is to avoid unnecessary memcpy.
   void GetBufferData(int8_t* buffer);
 
  private:
   // Device buffer that provides 10ms chunks of data.
   AudioDeviceBuffer* device_buffer_;
   // Number of bytes delivered per GetBufferData
-  int desired_frame_size_bytes_;
+  size_t desired_frame_size_bytes_;
   int sample_rate_;
-  int samples_per_10_ms_;
+  size_t samples_per_10_ms_;
   // Convenience parameter to avoid converting from samples
-  int bytes_per_10_ms_;
+  size_t bytes_per_10_ms_;
 
   // Storage for samples that are not yet asked for.
   rtc::scoped_ptr<int8_t[]> cache_buffer_;
   // Location of first unread sample.
-  int cached_buffer_start_;
+  size_t cached_buffer_start_;
   // Number of bytes stored in cache.
-  int cached_bytes_;
+  size_t cached_bytes_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_FINE_AUDIO_BUFFER_H_