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

webrtc/modules/audio_device/audio_device_buffer.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.
  */
 
 #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
 
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/system_wrappers/interface/file_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 class CriticalSectionWrapper;
 
 const uint32_t kPulsePeriodMs = 1000;
-const uint32_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
+const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
 
 class AudioDeviceObserver;
 
 class AudioDeviceBuffer
 {
 public:
     AudioDeviceBuffer();
     virtual ~AudioDeviceBuffer();
 
     void SetId(uint32_t id);
@@ skipping 10 matching lines @@
     virtual int32_t SetRecordingChannels(uint8_t channels);
     virtual int32_t SetPlayoutChannels(uint8_t channels);
     uint8_t RecordingChannels() const;
     uint8_t PlayoutChannels() const;
     int32_t SetRecordingChannel(
         const AudioDeviceModule::ChannelType channel);
     int32_t RecordingChannel(
         AudioDeviceModule::ChannelType& channel) const;
 
     virtual int32_t SetRecordedBuffer(const void* audioBuffer,
-                                      uint32_t nSamples);
+                                      size_t nSamples);
     int32_t SetCurrentMicLevel(uint32_t level);
     virtual void SetVQEData(int playDelayMS,
                             int recDelayMS,
                             int clockDrift);
     virtual int32_t DeliverRecordedData();
     uint32_t NewMicLevel() const;
 
-    virtual int32_t RequestPlayoutData(uint32_t nSamples);
+    virtual int32_t RequestPlayoutData(size_t nSamples);
     virtual int32_t GetPlayoutData(void* audioBuffer);
 
     int32_t StartInputFileRecording(
         const char fileName[kAdmMaxFileNameSize]);
     int32_t StopInputFileRecording();
     int32_t StartOutputFileRecording(
         const char fileName[kAdmMaxFileNameSize]);
     int32_t StopOutputFileRecording();
 
     int32_t SetTypingStatus(bool typingStatus);
 
 private:
     int32_t                   _id;
     CriticalSectionWrapper&         _critSect;
     CriticalSectionWrapper&         _critSectCb;
 
     AudioTransport*                 _ptrCbAudioTransport;
 
     uint32_t                  _recSampleRate;
     uint32_t                  _playSampleRate;
 
     uint8_t                   _recChannels;
     uint8_t                   _playChannels;
 
     // selected recording channel (left/right/both)
     AudioDeviceModule::ChannelType _recChannel;
 
     // 2 or 4 depending on mono or stereo
-    uint8_t                   _recBytesPerSample;
-    uint8_t                   _playBytesPerSample;
+    size_t                   _recBytesPerSample;
+    size_t                   _playBytesPerSample;
 
     // 10ms in stereo @ 96kHz
     int8_t                          _recBuffer[kMaxBufferSizeBytes];
 
     // one sample <=> 2 or 4 bytes
-    uint32_t                  _recSamples;
-    uint32_t                  _recSize;           // in bytes
+    size_t                    _recSamples;
+    size_t                    _recSize;           // in bytes
 
     // 10ms in stereo @ 96kHz
     int8_t                          _playBuffer[kMaxBufferSizeBytes];
 
     // one sample <=> 2 or 4 bytes
-    uint32_t                  _playSamples;
-    uint32_t                  _playSize;          // in bytes
+    size_t                    _playSamples;
+    size_t                    _playSize;          // in bytes
 
     FileWrapper&                    _recFile;
     FileWrapper&                    _playFile;
 
     uint32_t                  _currentMicLevel;
     uint32_t                  _newMicLevel;
 
     bool                      _typingStatus;
 
     int _playDelayMS;
     int _recDelayMS;
     int _clockDrift;
     int high_delay_counter_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H