Update audio code to use size_t more correctly,

webrtc/common_audio/signal_processing/filter_ar_fast_q12_armv7.S

Index: webrtc/common_audio/signal_processing/filter_ar_fast_q12_armv7.S
diff --git a/webrtc/common_audio/signal_processing/filter_ar_fast_q12_armv7.S b/webrtc/common_audio/signal_processing/filter_ar_fast_q12_armv7.S
index b72396e1c308eb75a8985b4c8bc56e00294913e3..76c8eee7fa09123fb1bc09f4bbfe2f0ad867c22a 100644
--- a/webrtc/common_audio/signal_processing/filter_ar_fast_q12_armv7.S
+++ b/webrtc/common_audio/signal_processing/filter_ar_fast_q12_armv7.S
@@ -155,10 +155,10 @@ END:
 @void WebRtcSpl_FilterARFastQ12(int16_t* data_in,
 @                               int16_t* data_out,
 @                               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);