Update audio code to use size_t more correctly,

webrtc/common_audio/signal_processing/filter_ar_fast_q12.c

Index: webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
diff --git a/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c b/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
index cfd82ca8cf3e7020b44b3596bdb8604be0425adb..70001a088272ab450501e7d49637d27f41508f40 100644
--- a/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
+++ b/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
@@ -16,10 +16,10 @@
 void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
                                int16_t* data_out,
                                const int16_t* __restrict coefficients,
-                               int coefficients_length,
-                               int data_length) {
-  int i = 0;
-  int j = 0;
+                               size_t coefficients_length,
+                               size_t data_length) {
+  size_t i = 0;
+  size_t j = 0;
 
   assert(data_length > 0);
   assert(coefficients_length > 1);