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

webrtc/modules/audio_processing/splitting_filter.cc

 /*
  *  Copyright (c) 2014 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/modules/audio_processing/splitting_filter.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_audio/channel_buffer.h"
 
 namespace webrtc {
 
 SplittingFilter::SplittingFilter(int num_channels,
-                                 int num_bands,
-                                 int num_frames)
+                                 size_t num_bands,
+                                 size_t num_frames)
     : num_bands_(num_bands) {
   CHECK(num_bands_ == 2 || num_bands_ == 3);
   if (num_bands_ == 2) {
     two_bands_states_.resize(num_channels);
   } else if (num_bands_ == 3) {
     for (int i = 0; i < num_channels; ++i) {
       three_band_filter_banks_.push_back(new ThreeBandFilterBank(num_frames));
     }
   }
 }
@@ skipping 66 matching lines @@
   DCHECK_EQ(static_cast<int>(three_band_filter_banks_.size()),
             data->num_channels());
   for (size_t i = 0; i < three_band_filter_banks_.size(); ++i) {
     three_band_filter_banks_[i]->Synthesis(bands->fbuf_const()->bands(i),
                                            bands->num_frames_per_band(),
                                            data->fbuf()->channels()[i]);
   }
 }
 
 }  // namespace webrtc