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

webrtc/common_audio/vad/vad_sp.c

Index: webrtc/common_audio/vad/vad_sp.c
diff --git a/webrtc/common_audio/vad/vad_sp.c b/webrtc/common_audio/vad/vad_sp.c
index 217ef26566f19c94044a13c65ac6e328eaed46bb..a54be17daacae5eaf97ec09db0c71efc7d8e31e9 100644
--- a/webrtc/common_audio/vad/vad_sp.c
+++ b/webrtc/common_audio/vad/vad_sp.c
@@ -27,12 +27,13 @@ static const int16_t kSmoothingUp = 32439;  // 0.99 in Q15.
 void WebRtcVad_Downsampling(const int16_t* signal_in,
                             int16_t* signal_out,
                             int32_t* filter_state,
-                            int in_length) {
+                            size_t in_length) {
   int16_t tmp16_1 = 0, tmp16_2 = 0;
   int32_t tmp32_1 = filter_state[0];
   int32_t tmp32_2 = filter_state[1];
-  int n = 0;
-  int half_length = (in_length >> 1);  // Downsampling by 2 gives half length.
+  size_t n = 0;
+  // Downsampling by 2 gives half length.
+  size_t half_length = (in_length >> 1);
 
   // Filter coefficients in Q13, filter state in Q0.
   for (n = 0; n < half_length; n++) {