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_oldapi.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 296 matching lines @@
   EXPECT_EQ(kNumPlcCng, stats.decoded_plc_cng);
 }
 
 TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) {
   AudioFrame audio_frame;
   const int kSampleRateHz = 32000;
   EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame));
   EXPECT_EQ(id_, audio_frame.id_);
   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_);
 }
 
 TEST_F(AudioCodingModuleTestOldApi, FailOnZeroDesiredFrequency) {
   AudioFrame audio_frame;
   EXPECT_EQ(-1, acm_->PlayoutData10Ms(0, &audio_frame));
 }
 
 // Checks that the transport callback is invoked once for each speech packet.
 // Also checks that the frame type is kAudioFrameSpeech.
@@ skipping 1414 matching lines @@
   Run(16000, 8000, 1000);
 }
 
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
   Run(8000, 16000, 1000);
 }
 
 #endif
 
 }  // namespace webrtc