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

webrtc/common_audio/vad/vad_sp.c

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/common_audio/vad/vad_sp.h"
 
 #include <assert.h>
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_audio/vad/vad_core.h"
 #include "webrtc/typedefs.h"
 
 // Allpass filter coefficients, upper and lower, in Q13.
 // Upper: 0.64, Lower: 0.17.
 static const int16_t kAllPassCoefsQ13[2] = { 5243, 1392 };  // Q13.
 static const int16_t kSmoothingDown = 6553;  // 0.2 in Q15.
 static const int16_t kSmoothingUp = 32439;  // 0.99 in Q15.
 
 // TODO(bjornv): Move this function to vad_filterbank.c.
 // Downsampling filter based on splitting filter and allpass functions.
 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++) {
     // All-pass filtering upper branch.
     tmp16_1 = (int16_t) ((tmp32_1 >> 1) +
         ((kAllPassCoefsQ13[0] * *signal_in) >> 14));
     *signal_out = tmp16_1;
     tmp32_1 = (int32_t)(*signal_in++) - ((kAllPassCoefsQ13[0] * tmp16_1) >> 12);
 
     // All-pass filtering lower branch.
@@ skipping 122 matching lines @@
       alpha = kSmoothingUp;  // 0.99 in Q15.
     }
   }
   tmp32 = (alpha + 1) * self->mean_value[channel];
   tmp32 += (WEBRTC_SPL_WORD16_MAX - alpha) * current_median;
   tmp32 += 16384;
   self->mean_value[channel] = (int16_t) (tmp32 >> 15);
 
   return self->mean_value[channel];
 }