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

 /*
  *  Copyright (c) 2012 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 123 matching lines @@
   // implementations are gone.
   virtual int EncodeFrame(const int16_t* input,
                           size_t input_len_samples,
                           uint8_t* output) {
     encoded_info_.encoded_bytes = 0;
     const size_t samples_per_10ms = audio_encoder_->SampleRateHz() / 100;
     CHECK_EQ(samples_per_10ms * audio_encoder_->Num10MsFramesInNextPacket(),
              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
       // wants.
       test::InputAudioFile::DuplicateInterleaved(input + i * samples_per_10ms,
                                                  samples_per_10ms, channels_,
                                                  interleaved_input.get());
 
       encoded_info_ = audio_encoder_->Encode(
           0, interleaved_input.get(), audio_encoder_->SampleRateHz() / 100,
@@ skipping 186 matching lines @@
         input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
     size_t enc_len = EncodeFrame(input.get(), frame_size_, encoded_);
     AudioDecoder::SpeechType speech_type;
     EXPECT_EQ(0, decoder_->Init());
     rtc::scoped_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
     size_t dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
                                       frame_size_ * channels_ * sizeof(int16_t),
                                       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()));
   }
 };
 
 class AudioDecoderIsacFloatTest : public AudioDecoderTest {
  protected:
   AudioDecoderIsacFloatTest() : AudioDecoderTest() {
     codec_input_rate_hz_ = 16000;
     frame_size_ = 480;
     data_length_ = 10 * frame_size_;
     AudioEncoderIsac::Config config;
@@ skipping 358 matching lines @@
   EXPECT_TRUE(CodecSupported(kDecoderCNGnb));
   EXPECT_TRUE(CodecSupported(kDecoderCNGwb));
   EXPECT_TRUE(CodecSupported(kDecoderCNGswb32kHz));
   EXPECT_TRUE(CodecSupported(kDecoderCNGswb48kHz));
   EXPECT_TRUE(CodecSupported(kDecoderArbitrary));
   EXPECT_TRUE(CodecSupported(kDecoderOpus));
   EXPECT_TRUE(CodecSupported(kDecoderOpus_2ch));
 }
 
 }  // namespace webrtc