Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc

 /*
  *  Copyright (c) 2013 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.
  */
 
@@ skipping 14 matching lines @@
 #include "webrtc/modules/audio_coding/main/test/utility.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
 namespace {
 
 double FrameRms(AudioFrame& frame) {
-  int samples = frame.num_channels_ * frame.samples_per_channel_;
+  size_t samples = frame.num_channels_ * frame.samples_per_channel_;
   double rms = 0;
-  for (int n = 0; n < samples; ++n)
+  for (size_t n = 0; n < samples; ++n)
     rms += frame.data_[n] * frame.data_[n];
   rms /= samples;
   rms = sqrt(rms);
   return rms;
 }
 
 }
 
 class InitialPlayoutDelayTest : public ::testing::Test {
  protected:
@@ skipping 77 matching lines @@
 
  private:
   void Run(CodecInst codec, int initial_delay_ms) {
     AudioFrame in_audio_frame;
     AudioFrame out_audio_frame;
     int num_frames = 0;
     const int kAmp = 10000;
     in_audio_frame.sample_rate_hz_ = codec.plfreq;
     in_audio_frame.num_channels_ = codec.channels;
     in_audio_frame.samples_per_channel_ = codec.plfreq / 100;  // 10 ms.
-    int samples = in_audio_frame.num_channels_ *
+    size_t samples = in_audio_frame.num_channels_ *
         in_audio_frame.samples_per_channel_;
-    for (int n = 0; n < samples; ++n) {
+    for (size_t n = 0; n < samples; ++n) {
       in_audio_frame.data_[n] = kAmp;
     }
 
     uint32_t timestamp = 0;
     double rms = 0;
     ASSERT_EQ(0, acm_a_->RegisterSendCodec(codec));
     acm_b_->SetInitialPlayoutDelay(initial_delay_ms);
     while (rms < kAmp / 2) {
       in_audio_frame.timestamp_ = timestamp;
       timestamp += static_cast<uint32_t>(in_audio_frame.samples_per_channel_);
@@ skipping 18 matching lines @@
 
 TEST_F(InitialPlayoutDelayTest, SwbMono) { SwbMono(); }
 
 TEST_F(InitialPlayoutDelayTest, NbStereo) { NbStereo(); }
 
 TEST_F(InitialPlayoutDelayTest, WbStereo) { WbStereo(); }
 
 TEST_F(InitialPlayoutDelayTest, SwbStereo) { SwbStereo(); }
 
 }  // namespace webrtc