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

webrtc/modules/audio_processing/agc/agc_manager_direct.h

Index: webrtc/modules/audio_processing/agc/agc_manager_direct.h
diff --git a/webrtc/modules/audio_processing/agc/agc_manager_direct.h b/webrtc/modules/audio_processing/agc/agc_manager_direct.h
index d12acf30d377b0c7344cc2c1106d8fcf8993dff9..fae1248b195d0c1a586ae5090fadc6cf2b9b76ec 100644
--- a/webrtc/modules/audio_processing/agc/agc_manager_direct.h
+++ b/webrtc/modules/audio_processing/agc/agc_manager_direct.h
@@ -57,8 +57,8 @@ class AgcManagerDirect {
   int Initialize();
   void AnalyzePreProcess(int16_t* audio,
                          int num_channels,
-                         int samples_per_channel);
-  void Process(const int16_t* audio, int length, int sample_rate_hz);
+                         size_t samples_per_channel);
+  void Process(const int16_t* audio, size_t length, int sample_rate_hz);
 
   // Sets a new microphone level, after first checking that it hasn't been
   // updated by the user, in which case no action is taken.