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

webrtc/voice_engine/utility_unittest.cc

 /*
  *  Copyright (c) 2012 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 <math.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/voice_engine/utility.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 namespace webrtc {
 namespace voe {
 namespace {
 
 enum FunctionToTest {
@@ skipping 22 matching lines @@
 };
 
 // Sets the signal value to increase by |data| with every sample. Floats are
 // used so non-integer values result in rounding error, but not an accumulating
 // error.
 void SetMonoFrame(AudioFrame* frame, float data, int sample_rate_hz) {
   memset(frame->data_, 0, sizeof(frame->data_));
   frame->num_channels_ = 1;
   frame->sample_rate_hz_ = sample_rate_hz;
   frame->samples_per_channel_ = sample_rate_hz / 100;
-  for (int i = 0; i < frame->samples_per_channel_; i++) {
+  for (size_t i = 0; i < frame->samples_per_channel_; i++) {
     frame->data_[i] = static_cast<int16_t>(data * i);
   }
 }
 
 // Keep the existing sample rate.
 void SetMonoFrame(AudioFrame* frame, float data) {
   SetMonoFrame(frame, data, frame->sample_rate_hz_);
 }
 
 // Sets the signal value to increase by |left| and |right| with every sample in
 // each channel respectively.
 void SetStereoFrame(AudioFrame* frame, float left, float right,
                     int sample_rate_hz) {
   memset(frame->data_, 0, sizeof(frame->data_));
   frame->num_channels_ = 2;
   frame->sample_rate_hz_ = sample_rate_hz;
   frame->samples_per_channel_ = sample_rate_hz / 100;
-  for (int i = 0; i < frame->samples_per_channel_; i++) {
+  for (size_t i = 0; i < frame->samples_per_channel_; i++) {
     frame->data_[i * 2] = static_cast<int16_t>(left * i);
     frame->data_[i * 2 + 1] = static_cast<int16_t>(right * i);
   }
 }
 
 // Keep the existing sample rate.
 void SetStereoFrame(AudioFrame* frame, float left, float right) {
   SetStereoFrame(frame, left, right, frame->sample_rate_hz_);
 }
 
 void VerifyParams(const AudioFrame& ref_frame, const AudioFrame& test_frame) {
   EXPECT_EQ(ref_frame.num_channels_, test_frame.num_channels_);
   EXPECT_EQ(ref_frame.samples_per_channel_, test_frame.samples_per_channel_);
   EXPECT_EQ(ref_frame.sample_rate_hz_, test_frame.sample_rate_hz_);
 }
 
 // Computes the best SNR based on the error between |ref_frame| and
 // |test_frame|. It allows for up to a |max_delay| in samples between the
 // signals to compensate for the resampling delay.
 float ComputeSNR(const AudioFrame& ref_frame, const AudioFrame& test_frame,
-                 int max_delay) {
+                 size_t max_delay) {
   VerifyParams(ref_frame, test_frame);
   float best_snr = 0;
-  int best_delay = 0;
-  for (int delay = 0; delay <= max_delay; delay++) {
+  size_t best_delay = 0;
+  for (size_t delay = 0; delay <= max_delay; delay++) {
     float mse = 0;
     float variance = 0;
-    for (int i = 0; i < ref_frame.samples_per_channel_ *
+    for (size_t i = 0; i < ref_frame.samples_per_channel_ *
         ref_frame.num_channels_ - delay; i++) {
       int error = ref_frame.data_[i] - test_frame.data_[i + delay];
       mse += error * error;
       variance += ref_frame.data_[i] * ref_frame.data_[i];
     }
     float snr = 100;  // We assign 100 dB to the zero-error case.
     if (mse > 0)
       snr = 10 * log10(variance / mse);
     if (snr > best_snr) {
       best_snr = snr;
       best_delay = delay;
     }
   }
-  printf("SNR=%.1f dB at delay=%d\n", best_snr, best_delay);
+  printf("SNR=%.1f dB at delay=%" PRIuS "\n", best_snr, best_delay);
   return best_snr;
 }
 
 void VerifyFramesAreEqual(const AudioFrame& ref_frame,
                           const AudioFrame& test_frame) {
   VerifyParams(ref_frame, test_frame);
-  for (int i = 0; i < ref_frame.samples_per_channel_ * ref_frame.num_channels_;
-      i++) {
+  for (size_t i = 0;
+       i < ref_frame.samples_per_channel_ * ref_frame.num_channels_; i++) {
     EXPECT_EQ(ref_frame.data_[i], test_frame.data_[i]);
   }
 }
 
 void UtilityTest::RunResampleTest(int src_channels,
                                   int src_sample_rate_hz,
                                   int dst_channels,
                                   int dst_sample_rate_hz,
                                   FunctionToTest function) {
   PushResampler<int16_t> resampler;  // Create a new one with every test.
@@ skipping 19 matching lines @@
     SetStereoFrame(&dst_frame_, 0, 0, dst_sample_rate_hz);
     if (src_channels == 1)
       SetStereoFrame(&golden_frame_, dst_left, dst_left, dst_sample_rate_hz);
     else
       SetStereoFrame(&golden_frame_, dst_left, dst_right, dst_sample_rate_hz);
   }
 
   // The sinc resampler has a known delay, which we compute here. Multiplying by
   // two gives us a crude maximum for any resampling, as the old resampler
   // typically (but not always) has lower delay.
-  static const int kInputKernelDelaySamples = 16;
-  const int max_delay = static_cast<double>(dst_sample_rate_hz)
-      / src_sample_rate_hz * kInputKernelDelaySamples * dst_channels * 2;
+  static const size_t kInputKernelDelaySamples = 16;
+  const size_t max_delay = static_cast<size_t>(
+      static_cast<double>(dst_sample_rate_hz) / src_sample_rate_hz *
+      kInputKernelDelaySamples * dst_channels * 2);
   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);
   if (function == TestRemixAndResample) {
     RemixAndResample(src_frame_, &resampler, &dst_frame_);
   } else {
     int16_t mono_buffer[kMaxMonoDataSizeSamples];
     DownConvertToCodecFormat(src_frame_.data_,
                              src_frame_.samples_per_channel_,
                              src_frame_.num_channels_,
                              src_frame_.sample_rate_hz_,
@@ skipping 77 matching lines @@
           }
         }
       }
     }
   }
 }
 
 }  // namespace
 }  // namespace voe
 }  // namespace webrtc