Misc. small cleanups

webrtc/common_audio/audio_converter_unittest.cc

Index: webrtc/common_audio/audio_converter_unittest.cc
diff --git a/webrtc/common_audio/audio_converter_unittest.cc b/webrtc/common_audio/audio_converter_unittest.cc
index c85b96e28589bbc92ac424879aed402260de0b41..e373d78b46316c5b0ede89f447746914f4857810 100644
--- a/webrtc/common_audio/audio_converter_unittest.cc
+++ b/webrtc/common_audio/audio_converter_unittest.cc
@@ -13,6 +13,7 @@
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/audio_converter.h"
@@ -24,11 +25,11 @@ namespace webrtc {
 typedef rtc::scoped_ptr<ChannelBuffer<float>> ScopedBuffer;
 
 // Sets the signal value to increase by |data| with every sample.
-ScopedBuffer CreateBuffer(const std::vector<float>& data, int frames) {
+ScopedBuffer CreateBuffer(const std::vector<float>& data, size_t frames) {
   const int num_channels = static_cast<int>(data.size());

[tlegrand-webrtc, 2016/01/07 12:47:52]

size_t?

   ScopedBuffer sb(new ChannelBuffer<float>(frames, num_channels));
   for (int i = 0; i < num_channels; ++i)
-    for (int j = 0; j < frames; ++j)
+    for (size_t j = 0; j < frames; ++j)
       sb->channels()[i][j] = data[i] * j;
   return sb;
 }
@@ -96,8 +97,8 @@ void RunAudioConverterTest(int src_channels,
   const float dst_left = resampling_factor * kSrcLeft;
   const float dst_right = resampling_factor * kSrcRight;
   const float dst_mono = (dst_left + dst_right) / 2;
-  const int src_frames = src_sample_rate_hz / 100;
-  const int dst_frames = dst_sample_rate_hz / 100;
+  const size_t src_frames = static_cast<size_t>(src_sample_rate_hz / 100);
+  const size_t dst_frames = static_cast<size_t>(dst_sample_rate_hz / 100);
 
   std::vector<float> src_data(1, kSrcLeft);
   if (src_channels == 2)
@@ -141,13 +142,13 @@ void RunAudioConverterTest(int src_channels,
 
 TEST(AudioConverterTest, ConversionsPassSNRThreshold) {
   const int kSampleRates[] = {8000, 16000, 32000, 44100, 48000};
-  const int kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
   const int kChannels[] = {1, 2};
-  const int kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
-  for (int src_rate = 0; src_rate < kSampleRatesSize; ++src_rate) {
-    for (int dst_rate = 0; dst_rate < kSampleRatesSize; ++dst_rate) {
-      for (int src_channel = 0; src_channel < kChannelsSize; ++src_channel) {
-        for (int dst_channel = 0; dst_channel < kChannelsSize; ++dst_channel) {
+  for (size_t src_rate = 0; src_rate < arraysize(kSampleRates); ++src_rate) {
+    for (size_t dst_rate = 0; dst_rate < arraysize(kSampleRates); ++dst_rate) {
+      for (size_t src_channel = 0; src_channel < arraysize(kChannels);
+           ++src_channel) {
+        for (size_t dst_channel = 0; dst_channel < arraysize(kChannels);
+             ++dst_channel) {
           RunAudioConverterTest(kChannels[src_channel], kSampleRates[src_rate],
                                 kChannels[dst_channel], kSampleRates[dst_rate]);
         }