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

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

 /*
  *  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 38 matching lines @@
 #ifdef WEBRTC_AGC_DEBUG_DUMP
     FILE* logFile;
     int frameCounter;
 #endif
 } DigitalAgc;
 
 int32_t WebRtcAgc_InitDigital(DigitalAgc* digitalAgcInst, int16_t agcMode);
 
 int32_t WebRtcAgc_ProcessDigital(DigitalAgc* digitalAgcInst,
                                  const int16_t* const* inNear,
-                                 int16_t num_bands,
+                                 size_t num_bands,
                                  int16_t* const* out,
                                  uint32_t FS,
                                  int16_t lowLevelSignal);
 
 int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* digitalAgcInst,
                                      const int16_t* inFar,
-                                     int16_t nrSamples);
+                                     size_t nrSamples);
 
 void WebRtcAgc_InitVad(AgcVad* vadInst);
 
 int16_t WebRtcAgc_ProcessVad(AgcVad* vadInst,     // (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 WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
                                      int16_t compressionGaindB, // Q0 (in dB)
                                      int16_t targetLevelDbfs,// Q0 (in dB)
                                      uint8_t limiterEnable,
                                      int16_t analogTarget);
 
 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_