Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/codecs/audio_decoder.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 38 matching lines @@
 int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded,
                                           size_t encoded_len,
                                           int sample_rate_hz, int16_t* decoded,
                                           SpeechType* speech_type) {
   return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
                         speech_type);
 }
 
 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,
                                  uint16_t rtp_sequence_number,
                                  uint32_t rtp_timestamp,
                                  uint32_t arrival_timestamp) {
   return 0;
 }
 
 int AudioDecoder::ErrorCode() { return 0; }
@@ skipping 25 matching lines @@
       return kSpeech;
     case 2:
       return kComfortNoise;
     default:
       assert(false);
       return kSpeech;
   }
 }
 
 }  // namespace webrtc