Update audio code to use size_t more correctly, webrtc/modules/audio_processing/

webrtc/modules/audio_processing/splitting_filter_unittest.cc

Index: webrtc/modules/audio_processing/splitting_filter_unittest.cc
diff --git a/webrtc/modules/audio_processing/splitting_filter_unittest.cc b/webrtc/modules/audio_processing/splitting_filter_unittest.cc
index 0498cc688a5427b169b8ac32f8d116cc13ea8f48..e7af65115c69319dafc733974f30311aee2bbe6e 100644
--- a/webrtc/modules/audio_processing/splitting_filter_unittest.cc
+++ b/webrtc/modules/audio_processing/splitting_filter_unittest.cc
@@ -20,8 +20,8 @@
 namespace webrtc {
 namespace {
 
-const int kSamplesPer16kHzChannel = 160;
-const int kSamplesPer48kHzChannel = 480;
+const size_t kSamplesPer16kHzChannel = 160;
+const size_t kSamplesPer48kHzChannel = 480;
 
 }  // namespace
 
@@ -35,26 +35,26 @@ const int kSamplesPer48kHzChannel = 480;
 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);

[Andrew MacDonald, 2015/07/24 04:01:44]

Literal?

[Peter Kasting, 2015/07/24 06:44:22]

That might work here, I can't remember.  I'll try it.

[Peter Kasting, 2015/07/27 23:09:54]

Just using "1" seems to work on Windows, not sure if I changed this for other
platforms or what.  I'll try "1", and if that won't compile on all the trybots,
I'll use "1u", and only use the cast version if neither works.

[Andrew MacDonald, 2015/07/28 07:05:43]

Sounds good.

[Peter Kasting, 2015/07/28 22:34:32]

Update: the static_cast really is required for the Win x64 bot:

e:\b\build\slave\win\build\src\webrtc\modules\audio_processing\splitting_filter_unittest.cc(55)
: warning C4334: '<<' : result of 32-bit shift implicitly converted to 64 bits
(was 64-bit shift intended?)

[Andrew MacDonald, 2015/07/29 01:10:56]

Thanks for checking.

       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);
@@ -64,9 +64,9 @@ TEST(SplittingFilterTest, SplitsIntoThreeBandsAndReconstructs) {
     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];
       }
@@ -81,9 +81,9 @@ TEST(SplittingFilterTest, SplitsIntoThreeBandsAndReconstructs) {
     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];
       }
@@ -94,7 +94,7 @@ TEST(SplittingFilterTest, SplitsIntoThreeBandsAndReconstructs) {
     }
     // 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) {