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

webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc

Index: webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 3983c077010383185a1de8ae1ff280840aa11eb8..a2ef9d1cb1e87f0ee05647301b7806d279637c7e 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -141,7 +141,7 @@ class AudioDecoderTest : public ::testing::Test {
              input_len_samples);
     rtc::scoped_ptr<int16_t[]> interleaved_input(
         new int16_t[channels_ * samples_per_10ms]);
-    for (int i = 0; i < audio_encoder_->Num10MsFramesInNextPacket(); ++i) {
+    for (size_t i = 0; i < audio_encoder_->Num10MsFramesInNextPacket(); ++i) {
       EXPECT_EQ(0u, encoded_info_.encoded_bytes);
 
       // Duplicate the mono input signal to however many channels the test
@@ -348,7 +348,7 @@ class AudioDecoderIlbcTest : public AudioDecoderTest {
                                       output.get(), &speech_type);
     EXPECT_EQ(frame_size_, dec_len);
     // Simply call DecodePlc and verify that we get 0 as return value.
-    EXPECT_EQ(0, decoder_->DecodePlc(1, output.get()));
+    EXPECT_EQ(0U, decoder_->DecodePlc(1, output.get()));
   }
 };