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

webrtc/common_audio/signal_processing/downsample_fast.c

Index: webrtc/common_audio/signal_processing/downsample_fast.c
diff --git a/webrtc/common_audio/signal_processing/downsample_fast.c b/webrtc/common_audio/signal_processing/downsample_fast.c
index 179c36a25c14ce83c520b0ebd771f1ba2d09d1e5..726a88819ac01e47e3dbfb4e41ccd2739d2f3dcd 100644
--- a/webrtc/common_audio/signal_processing/downsample_fast.c
+++ b/webrtc/common_audio/signal_processing/downsample_fast.c
@@ -13,20 +13,20 @@
 // TODO(Bjornv): Change the function parameter order to WebRTC code style.
 // C version of WebRtcSpl_DownsampleFast() for generic platforms.
 int WebRtcSpl_DownsampleFastC(const int16_t* data_in,
-                              int data_in_length,
+                              size_t data_in_length,
                               int16_t* data_out,
-                              int data_out_length,
+                              size_t data_out_length,
                               const int16_t* __restrict coefficients,
-                              int coefficients_length,
+                              size_t coefficients_length,
                               int factor,
-                              int delay) {
-  int i = 0;
-  int j = 0;
+                              size_t delay) {
+  size_t i = 0;
+  size_t j = 0;
   int32_t out_s32 = 0;
-  int endpos = delay + factor * (data_out_length - 1) + 1;
+  size_t endpos = delay + factor * (data_out_length - 1) + 1;
 
   // Return error if any of the running conditions doesn't meet.
-  if (data_out_length <= 0 || coefficients_length <= 0
+  if (data_out_length == 0 || coefficients_length == 0
                            || data_in_length < endpos) {
     return -1;
   }