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

webrtc/modules/audio_processing/rms_level.h

Index: webrtc/modules/audio_processing/rms_level.h
diff --git a/webrtc/modules/audio_processing/rms_level.h b/webrtc/modules/audio_processing/rms_level.h
index 055d271bb1980c1d0c9f4903a3fc8b24098461c5..12fa2125f08acdd3820971dddbcb370f3d25cdfa 100644
--- a/webrtc/modules/audio_processing/rms_level.h
+++ b/webrtc/modules/audio_processing/rms_level.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
 
+#include <cstddef>
+
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -35,11 +37,11 @@ class RMSLevel {
   void Reset();
 
   // Pass each chunk of audio to Process() to accumulate the level.
-  void Process(const int16_t* data, int length);
+  void Process(const int16_t* data, size_t length);
 
   // If all samples with the given |length| have a magnitude of zero, this is
   // a shortcut to avoid some computation.
-  void ProcessMuted(int length);
+  void ProcessMuted(size_t length);
 
   // Computes the RMS level over all data passed to Process() since the last
   // call to RMS(). The returned value is positive but should be interpreted as
@@ -48,7 +50,7 @@ class RMSLevel {
 
  private:
   float sum_square_;
-  int sample_count_;
+  size_t sample_count_;
 };
 
 }  // namespace webrtc