Update audio code to use size_t more correctly, webrtc/voice_engine/ portion.

webrtc/voice_engine/voice_engine_defines.h

 /*
  *  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.
  */
 
@@ skipping 10 matching lines @@
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 
 // ----------------------------------------------------------------------------
 //  Enumerators
 // ----------------------------------------------------------------------------
 
 namespace webrtc {
 
 // Internal buffer size required for mono audio, based on the highest sample
 // rate voice engine supports (10 ms of audio at 192 kHz).
-static const int kMaxMonoDataSizeSamples = 1920;
+static const size_t kMaxMonoDataSizeSamples = 1920;
 
 // VolumeControl
 enum { kMinVolumeLevel = 0 };
 enum { kMaxVolumeLevel = 255 };
 // Min scale factor for per-channel volume scaling
 const float kMinOutputVolumeScaling = 0.0f;
 // Max scale factor for per-channel volume scaling
 const float kMaxOutputVolumeScaling = 10.0f;
 // Min scale factor for output volume panning
 const float kMinOutputVolumePanning = 0.0f;
@@ skipping 269 matching lines @@
 #define __cdecl
 #define LPSOCKADDR struct sockaddr *
 #define LPCSTR const char *
 #define ULONG unsigned long
 
 // Default device for Mac and iPhone
 #define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0
 #endif  // #ifdef WEBRTC_MAC
 
 #endif  // WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H