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

webrtc/modules/audio_processing/aec/aec_core_internal.h

 /*
  *  Copyright (c) 2013 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 83 matching lines @@
   short divergeState;
 
   int xfBufBlockPos;
 
   RingBuffer* far_buf;
   RingBuffer* far_buf_windowed;
   int system_delay;  // Current system delay buffered in AEC.
 
   int mult;  // sampling frequency multiple
   int sampFreq;
-  int num_bands;
+  size_t num_bands;
   uint32_t seed;
 
   float normal_mu;               // stepsize
   float normal_error_threshold;  // error threshold
 
   int noiseEstCtr;
 
   PowerLevel farlevel;
   PowerLevel nearlevel;
   PowerLevel linoutlevel;
@@ skipping 77 matching lines @@
 
 typedef void (*WebRtcAecSubBandCoherence)(AecCore* aec,
                                           float efw[2][PART_LEN1],
                                           float xfw[2][PART_LEN1],
                                           float* fft,
                                           float* cohde,
                                           float* cohxd);
 extern WebRtcAecSubBandCoherence WebRtcAec_SubbandCoherence;
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_INTERNAL_H_