Convert some more things to size_t.

webrtc/modules/audio_device/include/audio_device_defines.h

 /*
  *  Copyright (c) 2011 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 125 matching lines @@
 };
 
 // Helper class for storage of fundamental audio parameters such as sample rate,
 // number of channels, native buffer size etc.
 // Note that one audio frame can contain more than one channel sample and each
 // sample is assumed to be a 16-bit PCM sample. Hence, one audio frame in
 // stereo contains 2 * (16/8) = 4 bytes of data.
 class AudioParameters {
  public:
   // This implementation does only support 16-bit PCM samples.
-  enum { kBitsPerSample = 16 };
+  static const size_t kBitsPerSample = 16;
   AudioParameters()
       : sample_rate_(0),
         channels_(0),
         frames_per_buffer_(0),
         frames_per_10ms_buffer_(0) {}
-  AudioParameters(int sample_rate, int channels, int frames_per_buffer)
+  AudioParameters(int sample_rate, int channels, size_t frames_per_buffer)
       : sample_rate_(sample_rate),
         channels_(channels),
         frames_per_buffer_(frames_per_buffer),
         frames_per_10ms_buffer_(static_cast<size_t>(sample_rate / 100)) {}
-  void reset(int sample_rate, int channels, int frames_per_buffer) {
+  void reset(int sample_rate, int channels, size_t frames_per_buffer) {
     sample_rate_ = sample_rate;
     channels_ = channels;
     frames_per_buffer_ = frames_per_buffer;
     frames_per_10ms_buffer_ = static_cast<size_t>(sample_rate / 100);
   }
-  int bits_per_sample() const { return kBitsPerSample; }
+  size_t bits_per_sample() const { return kBitsPerSample; }
   int sample_rate() const { return sample_rate_; }
   int channels() const { return channels_; }
-  int frames_per_buffer() const { return frames_per_buffer_; }
+  size_t frames_per_buffer() const { return frames_per_buffer_; }
   size_t frames_per_10ms_buffer() const { return frames_per_10ms_buffer_; }
   bool is_valid() const {
     return ((sample_rate_ > 0) && (channels_ > 0) && (frames_per_buffer_ > 0));
   }
-  int GetBytesPerFrame() const { return channels_ * kBitsPerSample / 8; }
-  int GetBytesPerBuffer() const {
+  size_t GetBytesPerFrame() const { return channels_ * kBitsPerSample / 8; }
+  size_t GetBytesPerBuffer() const {
     return frames_per_buffer_ * GetBytesPerFrame();
   }
   size_t GetBytesPer10msBuffer() const {
     return frames_per_10ms_buffer_ * GetBytesPerFrame();
   }
   float GetBufferSizeInMilliseconds() const {
     if (sample_rate_ == 0)
       return 0.0f;
     return frames_per_buffer_ / (sample_rate_ / 1000.0f);
   }
 
  private:
   int sample_rate_;
   int channels_;
-  int frames_per_buffer_;
+  size_t frames_per_buffer_;
   size_t frames_per_10ms_buffer_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H