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

webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_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 95 matching lines @@
   EXPECT_CALL(mock_encoder_, SampleRateHz()).WillOnce(Return(17));
   EXPECT_EQ(17, red_->SampleRateHz());
 }
 
 TEST_F(AudioEncoderCopyRedTest, CheckNumChannelsPropagation) {
   EXPECT_CALL(mock_encoder_, NumChannels()).WillOnce(Return(17));
   EXPECT_EQ(17, red_->NumChannels());
 }
 
 TEST_F(AudioEncoderCopyRedTest, CheckFrameSizePropagation) {
-  EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket()).WillOnce(Return(17));
-  EXPECT_EQ(17, red_->Num10MsFramesInNextPacket());
+  EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket()).WillOnce(Return(17U));
+  EXPECT_EQ(17U, red_->Num10MsFramesInNextPacket());
 }
 
 TEST_F(AudioEncoderCopyRedTest, CheckMaxFrameSizePropagation) {
-  EXPECT_CALL(mock_encoder_, Max10MsFramesInAPacket()).WillOnce(Return(17));
-  EXPECT_EQ(17, red_->Max10MsFramesInAPacket());
+  EXPECT_CALL(mock_encoder_, Max10MsFramesInAPacket()).WillOnce(Return(17U));
+  EXPECT_EQ(17U, red_->Max10MsFramesInAPacket());
 }
 
 TEST_F(AudioEncoderCopyRedTest, CheckSetBitratePropagation) {
   EXPECT_CALL(mock_encoder_, SetTargetBitrate(4711));
   red_->SetTargetBitrate(4711);
 }
 
 TEST_F(AudioEncoderCopyRedTest, CheckProjectedPacketLossRatePropagation) {
   EXPECT_CALL(mock_encoder_, SetProjectedPacketLossRate(0.5));
   red_->SetProjectedPacketLossRate(0.5);
@@ skipping 192 matching lines @@
   config.speech_encoder = NULL;
   EXPECT_DEATH(red = new AudioEncoderCopyRed(config),
                "Speech encoder not provided.");
   // The delete operation is needed to avoid leak reports from memcheck.
   delete red;
 }
 
 #endif  // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
 
 }  // namespace webrtc