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

webrtc/common_audio/resampler/resampler_unittest.cc

 /*
  *  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 80 matching lines @@
 // with number of channels and ResamplerType as input.
 TEST_F(ResamplerTest, Mono) {
   const int kChannels = 1;
   for (size_t i = 0; i < kRatesSize; ++i) {
     for (size_t j = 0; j < kRatesSize; ++j) {
       std::ostringstream ss;
       ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j];
       SCOPED_TRACE(ss.str());
 
       if (ValidRates(kRates[i], kRates[j])) {
-        int in_length = kRates[i] / 100;
-        int out_length = 0;
+        size_t in_length = static_cast<size_t>(kRates[i] / 100);
+        size_t out_length = 0;
         EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kChannels));
         EXPECT_EQ(0, rs_.Push(data_in_, in_length, data_out_, kDataSize,
                               out_length));
-        EXPECT_EQ(kRates[j] / 100, out_length);
+        EXPECT_EQ(static_cast<size_t>(kRates[j] / 100), out_length);
       } else {
         EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kChannels));
       }
     }
   }
 }
 
 TEST_F(ResamplerTest, Stereo) {
   const int kChannels = 2;
   for (size_t i = 0; i < kRatesSize; ++i) {
     for (size_t j = 0; j < kRatesSize; ++j) {
       std::ostringstream ss;
       ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j];
       SCOPED_TRACE(ss.str());
 
       if (ValidRates(kRates[i], kRates[j])) {
-        int in_length = kChannels * kRates[i] / 100;
-        int out_length = 0;
+        size_t in_length = static_cast<size_t>(kChannels * kRates[i] / 100);
+        size_t out_length = 0;
         EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j],
                                kChannels));
         EXPECT_EQ(0, rs_.Push(data_in_, in_length, data_out_, kDataSize,
                               out_length));
-        EXPECT_EQ(kChannels * kRates[j] / 100, out_length);
+        EXPECT_EQ(static_cast<size_t>(kChannels * kRates[j] / 100), out_length);
       } else {
         EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j],
                                 kChannels));
       }
     }
   }
 }
 
 }  // namespace
 }  // namespace webrtc