Update audio code to use size_t more correctly, webrtc/ portion.

webrtc/tools/agc/agc_manager_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 15 matching lines @@
 using ::testing::Mock;
 using ::testing::Return;
 using ::testing::SetArgPointee;
 using ::testing::SetArgReferee;
 
 namespace webrtc {
 namespace {
 
 const int kSampleRateHz = 32000;
 const int kNumChannels = 1;
-const int kSamplesPerChannel = kSampleRateHz / 100;
+const size_t kSamplesPerChannel = static_cast<size_t>(kSampleRateHz / 100);
 const float kAboveClippedThreshold = 0.2f;
 
 }  // namespace
 
 class AgcManagerUnitTest : public ::testing::Test {
  protected:
   AgcManagerUnitTest()
       : media_(),
         volume_(),
         agc_(new MockAgc),
@@ skipping 685 matching lines @@
   SetVolumeAndProcess(128u);
   EXPECT_CALL(*agc_, GetRmsErrorDb(_))
       .WillRepeatedly(DoAll(SetArgPointee<0>(30), Return(true)));
   EXPECT_CALL(volume_, GetMicVolume(_))
       .WillRepeatedly(DoAll(SetArgReferee<0>(0), Return(0)));
   EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
   PostProcCallback(10);
 }
 
 }  // namespace webrtc