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

webrtc/modules/audio_coding/main/acm2/audio_coding_module_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.
  */
 
@@ skipping 254 matching lines @@
   EXPECT_EQ(kNumPlcCng, stats.decoded_plc_cng);
 }
 
 TEST_F(AudioCodingModuleTest, VerifyOutputFrame) {
   CreateAcm();
   AudioFrame audio_frame;
   const int kSampleRateHz = 32000;
   EXPECT_TRUE(acm_->Get10MsAudio(&audio_frame));
   EXPECT_EQ(0u, audio_frame.timestamp_);
   EXPECT_GT(audio_frame.num_channels_, 0);
-  EXPECT_EQ(kSampleRateHz / 100, audio_frame.samples_per_channel_);
+  EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100),
+            audio_frame.samples_per_channel_);
   EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
 }
 
 // A multi-threaded test for ACM. This base class is using the PCM16b 16 kHz
 // codec, while the derive class AcmIsacMtTest is using iSAC.
 class AudioCodingModuleMtTest : public AudioCodingModuleTest {
  protected:
   static const int kNumPackets = 500;
   static const int kNumPullCalls = 500;
 
@@ skipping 677 matching lines @@
           "d781cce1ab986b618d0da87226cdde30",
           "1a1fe04dd12e755949987c8d729fb3e0",
           "d781cce1ab986b618d0da87226cdde30"),
       50,
       test::AcmReceiveTest::kStereoOutput);
 }
 
 #endif
 
 }  // namespace webrtc