[NOT FOR REVIEW] Convert channel counts to size_t.

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..dace0bdccf59b3e612bad16d09aca2fed964192c 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) {
-  const int num_channels = static_cast<int>(data.size());
+ScopedBuffer CreateBuffer(const std::vector<float>& data, size_t frames) {
+  const size_t num_channels = data.size();
   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 i = 0; i < num_channels; ++i)
+    for (size_t j = 0; j < frames; ++j)
       sb->channels()[i][j] = data[i] * j;
   return sb;
 }
@@ -56,7 +57,7 @@ float ComputeSNR(const ChannelBuffer<float>& ref,
     float mse = 0;
     float variance = 0;
     float mean = 0;
-    for (int i = 0; i < ref.num_channels(); ++i) {
+    for (size_t i = 0; i < ref.num_channels(); ++i) {
       for (size_t j = 0; j < ref.num_frames() - delay; ++j) {
         float error = ref.channels()[i][j] - test.channels()[i][j + delay];
         mse += error * error;
@@ -85,9 +86,9 @@ float ComputeSNR(const ChannelBuffer<float>& ref,
 // Sets the source to a linearly increasing signal for which we can easily
 // generate a reference. Runs the AudioConverter and ensures the output has
 // sufficiently high SNR relative to the reference.
-void RunAudioConverterTest(int src_channels,
+void RunAudioConverterTest(size_t src_channels,
                            int src_sample_rate_hz,
-                           int dst_channels,
+                           size_t dst_channels,
                            int dst_sample_rate_hz) {
   const float kSrcLeft = 0.0002f;
   const float kSrcRight = 0.0001f;
@@ -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)
@@ -127,8 +128,9 @@ void RunAudioConverterTest(int src_channels,
       static_cast<size_t>(
           PushSincResampler::AlgorithmicDelaySeconds(src_sample_rate_hz) *
           dst_sample_rate_hz);
-  printf("(%d, %d Hz) -> (%d, %d Hz) ",  // SNR reported on the same line later.
-      src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
+  // SNR reported on the same line later.
+  printf("(%" PRIuS ", %d Hz) -> (%" PRIuS ", %d Hz) ",
+         src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
 
   rtc::scoped_ptr<AudioConverter> converter = AudioConverter::Create(
       src_channels, src_frames, dst_channels, dst_frames);
@@ -141,13 +143,13 @@ 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) {
+  const size_t kChannels[] = {1, 2};
+  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]);
         }