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

webrtc/common_audio/signal_processing/downsample_fast_mips.c

Index: webrtc/common_audio/signal_processing/downsample_fast_mips.c
diff --git a/webrtc/common_audio/signal_processing/downsample_fast_mips.c b/webrtc/common_audio/signal_processing/downsample_fast_mips.c
index dbde43d30b506f69915d2a745f497356a59a9d73..ac39401abba01118f0bc516ef3bbc9e4270e144c 100644
--- a/webrtc/common_audio/signal_processing/downsample_fast_mips.c
+++ b/webrtc/common_audio/signal_processing/downsample_fast_mips.c
@@ -12,18 +12,18 @@
 
 // Version of WebRtcSpl_DownsampleFast() for MIPS platforms.
 int WebRtcSpl_DownsampleFast_mips(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) {
+                                  size_t delay) {
   int i;
   int j;
   int k;
   int32_t out_s32 = 0;
-  int endpos = delay + factor * (data_out_length - 1) + 1;
+  size_t endpos = delay + factor * (data_out_length - 1) + 1;
 
   int32_t  tmp1, tmp2, tmp3, tmp4, factor_2;
   int16_t* p_coefficients;
@@ -36,7 +36,7 @@ int WebRtcSpl_DownsampleFast_mips(const int16_t* data_in,
 #endif  // #if !defined(MIPS_DSP_R1_LE)
 
   // 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;
   }