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

webrtc/modules/audio_device/audio_device_buffer.cc

Index: webrtc/modules/audio_device/audio_device_buffer.cc
diff --git a/webrtc/modules/audio_device/audio_device_buffer.cc b/webrtc/modules/audio_device/audio_device_buffer.cc
index 3cfbc7d09c4da6d7f9fb88850e76ae9db104af05..cc6d6bb1f780cf5c2ae2e5e80979c2986fa565c3 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.cc
+++ b/webrtc/modules/audio_device/audio_device_buffer.cc
@@ -13,6 +13,7 @@
 #include <assert.h>
 #include <string.h>
 
+#include "webrtc/base/format_macros.h"
 #include "webrtc/modules/audio_device/audio_device_config.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
@@ -380,7 +381,7 @@ int32_t AudioDeviceBuffer::StopOutputFileRecording()
 // ----------------------------------------------------------------------------
 
 int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audioBuffer,
-                                             uint32_t nSamples)
+                                             size_t nSamples)
 {
     CriticalSectionScoped lock(&_critSect);
 
@@ -414,7 +415,7 @@ int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audioBuffer,
         }
 
         // exctract left or right channel from input buffer to the local buffer
-        for (uint32_t i = 0; i < _recSamples; i++)
+        for (size_t i = 0; i < _recSamples; i++)
         {
             *ptr16Out = *ptr16In;
             ptr16Out++;
@@ -482,10 +483,10 @@ int32_t AudioDeviceBuffer::DeliverRecordedData()
 //  RequestPlayoutData
 // ----------------------------------------------------------------------------
 
-int32_t AudioDeviceBuffer::RequestPlayoutData(uint32_t nSamples)
+int32_t AudioDeviceBuffer::RequestPlayoutData(size_t nSamples)
 {
     uint32_t playSampleRate = 0;
-    uint8_t playBytesPerSample = 0;
+    size_t playBytesPerSample = 0;
     uint8_t playChannels = 0;
     {
         CriticalSectionScoped lock(&_critSect);
@@ -520,7 +521,7 @@ int32_t AudioDeviceBuffer::RequestPlayoutData(uint32_t nSamples)
         }
     }
 
-    uint32_t nSamplesOut(0);
+    size_t nSamplesOut(0);
 
     CriticalSectionScoped lock(&_critSectCb);
 
@@ -563,7 +564,7 @@ int32_t AudioDeviceBuffer::GetPlayoutData(void* audioBuffer)
     if (_playSize > kMaxBufferSizeBytes)
     {
        WEBRTC_TRACE(kTraceError, kTraceUtility, _id,
-                    "_playSize %i exceeds kMaxBufferSizeBytes in "
+                    "_playSize %" PRIuS " exceeds kMaxBufferSizeBytes in "
                     "AudioDeviceBuffer::GetPlayoutData", _playSize);
        assert(false);
        return -1;