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

webrtc/modules/audio_processing/agc/legacy/digital_agc.c

 /*
  *  Copyright (c) 2011 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.
  */
 
@@ skipping 265 matching lines @@
 
     // initialize VADs
     WebRtcAgc_InitVad(&stt->vadNearend);
     WebRtcAgc_InitVad(&stt->vadFarend);
 
     return 0;
 }
 
 int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* stt,
                                      const int16_t* in_far,
-                                     int16_t nrSamples) {
+                                     size_t nrSamples) {
     assert(stt != NULL);
     // VAD for far end
     WebRtcAgc_ProcessVad(&stt->vadFarend, in_far, nrSamples);
 
     return 0;
 }
 
 int32_t WebRtcAgc_ProcessDigital(DigitalAgc* stt,
                                  const int16_t* const* in_near,
-                                 int16_t num_bands,
+                                 size_t num_bands,
                                  int16_t* const* out,
                                  uint32_t FS,
                                  int16_t lowlevelSignal) {
     // array for gains (one value per ms, incl start & end)
     int32_t gains[11];
 
     int32_t out_tmp, tmp32;
     int32_t env[10];
     int32_t max_nrg;
     int32_t cur_level;
     int32_t gain32, delta;
     int16_t logratio;
     int16_t lower_thr, upper_thr;
     int16_t zeros = 0, zeros_fast, frac = 0;
     int16_t decay;
     int16_t gate, gain_adj;
-    int16_t k, n, i;
-    int16_t L, L2; // samples/subframe
+    int16_t k;
+    size_t n, i, L;
+    int16_t L2; // samples/subframe
 
     // determine number of samples per ms
     if (FS == 8000)
     {
         L = 8;
         L2 = 3;
     } else if (FS == 16000 || FS == 32000 || FS == 48000)
     {
         L = 16;
         L2 = 4;
@@ skipping 300 matching lines @@
     state->counter = 3; // counts updates
     for (k = 0; k < 8; k++)
     {
         // downsampling filter
         state->downState[k] = 0;
     }
 }
 
 int16_t WebRtcAgc_ProcessVad(AgcVad* state,      // (i) VAD state
                              const int16_t* in,  // (i) Speech signal
-                             int16_t nrSamples)  // (i) number of samples
+                             size_t nrSamples)  // (i) number of samples
 {
     int32_t out, nrg, tmp32, tmp32b;
     uint16_t tmpU16;
     int16_t k, subfr, tmp16;
     int16_t buf1[8];
     int16_t buf2[4];
     int16_t HPstate;
     int16_t zeros, dB;
 
     // process in 10 sub frames of 1 ms (to save on memory)
@@ skipping 116 matching lines @@
     {
         state->logRatio = 2048;
     }
     if (state->logRatio < -2048)
     {
         state->logRatio = -2048;
     }
 
     return state->logRatio; // Q10
 }