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

webrtc/modules/audio_processing/splitting_filter_unittest.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.
  */
 
 // MSVC++ requires this to be set before any other includes to get M_PI.
 #define _USE_MATH_DEFINES
 
 #include <cmath>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/modules/audio_processing/splitting_filter.h"
 
 namespace webrtc {
 namespace {
 
-const int kSamplesPer16kHzChannel = 160;
-const int kSamplesPer48kHzChannel = 480;
+const size_t kSamplesPer16kHzChannel = 160;
+const size_t kSamplesPer48kHzChannel = 480;
 
 }  // namespace
 
 // Generates a signal from presence or absence of sine waves of different
 // frequencies.
 // Splits into 3 bands and checks their presence or absence.
 // Recombines the bands.
 // Calculates the delay.
 // Checks that the cross correlation of input and output is high enough at the
 // calculated delay.
 TEST(SplittingFilterTest, SplitsIntoThreeBandsAndReconstructs) {
   static const int kChannels = 1;
   static const int kSampleRateHz = 48000;
-  static const int kNumBands = 3;
+  static const size_t kNumBands = 3;
   static const int kFrequenciesHz[kNumBands] = {1000, 12000, 18000};
   static const float kAmplitude = 8192.f;
-  static const int kChunks = 8;
+  static const size_t kChunks = 8;
   SplittingFilter splitting_filter(kChannels,
                                    kNumBands,
                                    kSamplesPer48kHzChannel);
   IFChannelBuffer in_data(kSamplesPer48kHzChannel, kChannels, kNumBands);
   IFChannelBuffer bands(kSamplesPer48kHzChannel, kChannels, kNumBands);
   IFChannelBuffer out_data(kSamplesPer48kHzChannel, kChannels, kNumBands);
-  for (int i = 0; i < kChunks; ++i) {
+  for (size_t i = 0; i < kChunks; ++i) {
     // Input signal generation.
     bool is_present[kNumBands];
     memset(in_data.fbuf()->channels()[0],
            0,
            kSamplesPer48kHzChannel * sizeof(in_data.fbuf()->channels()[0][0]));
-    for (int j = 0; j < kNumBands; ++j) {
-      is_present[j] = i & (1 << j);
+    for (size_t j = 0; j < kNumBands; ++j) {
+      is_present[j] = i & (static_cast<size_t>(1) << j);
       float amplitude = is_present[j] ? kAmplitude : 0.f;
-      for (int k = 0; k < kSamplesPer48kHzChannel; ++k) {
+      for (size_t k = 0; k < kSamplesPer48kHzChannel; ++k) {
         in_data.fbuf()->channels()[0][k] +=
             amplitude * sin(2.f * M_PI * kFrequenciesHz[j] *
                 (i * kSamplesPer48kHzChannel + k) / kSampleRateHz);
       }
     }
     // Three band splitting filter.
     splitting_filter.Analysis(&in_data, &bands);
     // Energy calculation.
     float energy[kNumBands];
-    for (int j = 0; j < kNumBands; ++j) {
+    for (size_t j = 0; j < kNumBands; ++j) {
       energy[j] = 0.f;
-      for (int k = 0; k < kSamplesPer16kHzChannel; ++k) {
+      for (size_t k = 0; k < kSamplesPer16kHzChannel; ++k) {
         energy[j] += bands.fbuf_const()->channels(j)[0][k] *
                      bands.fbuf_const()->channels(j)[0][k];
       }
       energy[j] /= kSamplesPer16kHzChannel;
       if (is_present[j]) {
         EXPECT_GT(energy[j], kAmplitude * kAmplitude / 4);
       } else {
         EXPECT_LT(energy[j], kAmplitude * kAmplitude / 4);
       }
     }
     // Three band merge.
     splitting_filter.Synthesis(&bands, &out_data);
     // Delay and cross correlation estimation.
     float xcorr = 0.f;
-    for (int delay = 0; delay < kSamplesPer48kHzChannel; ++delay) {
+    for (size_t delay = 0; delay < kSamplesPer48kHzChannel; ++delay) {
       float tmpcorr = 0.f;
-      for (int j = delay; j < kSamplesPer48kHzChannel; ++j) {
+      for (size_t j = delay; j < kSamplesPer48kHzChannel; ++j) {
         tmpcorr += in_data.fbuf_const()->channels()[0][j - delay] *
                    out_data.fbuf_const()->channels()[0][j];
       }
       tmpcorr /= kSamplesPer48kHzChannel;
       if (tmpcorr > xcorr) {
         xcorr = tmpcorr;
       }
     }
     // High cross correlation check.
     bool any_present = false;
-    for (int j = 0; j < kNumBands; ++j) {
+    for (size_t j = 0; j < kNumBands; ++j) {
       any_present |= is_present[j];
     }
     if (any_present) {
       EXPECT_GT(xcorr, kAmplitude * kAmplitude / 4);
     }
   }
 }
 
 }  // namespace webrtc