Convert channel counts to size_t.

webrtc/common_audio/audio_converter_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.
  */
 
 #include <cmath>
 #include <algorithm>
 #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"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
 
 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, size_t frames) {
-  const int num_channels = static_cast<int>(data.size());
+  const size_t num_channels = data.size();
   ScopedBuffer sb(new ChannelBuffer<float>(frames, num_channels));
-  for (int i = 0; i < num_channels; ++i)
+  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;
 }
 
 void VerifyParams(const ChannelBuffer<float>& ref,
                   const ChannelBuffer<float>& test) {
   EXPECT_EQ(ref.num_channels(), test.num_channels());
   EXPECT_EQ(ref.num_frames(), test.num_frames());
 }
 
 // Computes the best SNR based on the error between |ref_frame| and
 // |test_frame|. It searches around |expected_delay| in samples between the
 // signals to compensate for the resampling delay.
 float ComputeSNR(const ChannelBuffer<float>& ref,
                  const ChannelBuffer<float>& test,
                  size_t expected_delay) {
   VerifyParams(ref, test);
   float best_snr = 0;
   size_t best_delay = 0;
 
   // Search within one sample of the expected delay.
   for (size_t delay = std::max(expected_delay, static_cast<size_t>(1)) - 1;
        delay <= std::min(expected_delay + 1, ref.num_frames());
        ++delay) {
     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;
         variance += ref.channels()[i][j] * ref.channels()[i][j];
         mean += ref.channels()[i][j];
       }
     }
 
     const size_t length = ref.num_channels() * (ref.num_frames() - delay);
     mse /= length;
     variance /= length;
     mean /= length;
     variance -= mean * mean;
     float snr = 100;  // We assign 100 dB to the zero-error case.
     if (mse > 0)
       snr = 10 * std::log10(variance / mse);
     if (snr > best_snr) {
       best_snr = snr;
       best_delay = delay;
     }
   }
   printf("SNR=%.1f dB at delay=%" PRIuS "\n", best_snr, best_delay);
   return best_snr;
 }
 
 // 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;
   const float resampling_factor = (1.f * src_sample_rate_hz) /
       dst_sample_rate_hz;
   const float dst_left = resampling_factor * kSrcLeft;
   const float dst_right = resampling_factor * kSrcRight;
   const float dst_mono = (dst_left + dst_right) / 2;
   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);
@@ skipping 19 matching lines @@
       ref_data.push_back(dst_right);
   }
   ScopedBuffer dst_buffer = CreateBuffer(dst_data, dst_frames);
   ScopedBuffer ref_buffer = CreateBuffer(ref_data, dst_frames);
 
   // The sinc resampler has a known delay, which we compute here.
   const size_t delay_frames = src_sample_rate_hz == dst_sample_rate_hz ? 0 :
       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);
   converter->Convert(src_buffer->channels(), src_buffer->size(),
                      dst_buffer->channels(), dst_buffer->size());
 
   EXPECT_LT(43.f,
             ComputeSNR(*ref_buffer.get(), *dst_buffer.get(), delay_frames));
 }
 
 TEST(AudioConverterTest, ConversionsPassSNRThreshold) {
   const int kSampleRates[] = {8000, 16000, 32000, 44100, 48000};
-  const int kChannels[] = {1, 2};
+  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]);
         }
       }
     }
   }
 }
 
 }  // namespace webrtc