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

webrtc/modules/audio_device/android/audio_record_jni.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 117 matching lines @@
 
   // Delay estimate of the total round-trip delay (input + output).
   // Fixed value set once in AttachAudioBuffer() and it can take one out of two
   // possible values. See audio_common.h for details.
   int total_delay_in_milliseconds_;
 
   // Cached copy of address to direct audio buffer owned by |j_audio_record_|.
   void* direct_buffer_address_;
 
   // Number of bytes in the direct audio buffer owned by |j_audio_record_|.
-  int direct_buffer_capacity_in_bytes_;
+  size_t direct_buffer_capacity_in_bytes_;
 
   // Number audio frames per audio buffer. Each audio frame corresponds to
   // one sample of PCM mono data at 16 bits per sample. Hence, each audio
   // frame contains 2 bytes (given that the Java layer only supports mono).
   // Example: 480 for 48000 Hz or 441 for 44100 Hz.
-  int frames_per_buffer_;
+  size_t frames_per_buffer_;
 
   bool initialized_;
 
   bool recording_;
 
   // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
   // AudioDeviceModuleImpl class and called by AudioDeviceModuleImpl::Create().
   AudioDeviceBuffer* audio_device_buffer_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_