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

webrtc/modules/audio_coding/codecs/audio_decoder.cc

Index: webrtc/modules/audio_coding/codecs/audio_decoder.cc
diff --git a/webrtc/modules/audio_coding/codecs/audio_decoder.cc b/webrtc/modules/audio_coding/codecs/audio_decoder.cc
index 0a4a6a964bd30140836e3963d7dc956a5c56e796..08d101c5ae4355fd41beae0dbee6d628ea456b04 100644
--- a/webrtc/modules/audio_coding/codecs/audio_decoder.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_decoder.cc
@@ -56,7 +56,9 @@ int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded,
 
 bool AudioDecoder::HasDecodePlc() const { return false; }
 
-int AudioDecoder::DecodePlc(int num_frames, int16_t* decoded) { return 0; }
+size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) {
+  return 0;
+}
 
 int AudioDecoder::IncomingPacket(const uint8_t* payload,
                                  size_t payload_len,