Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.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 69 matching lines @@
   CriticalSectionScoped cs(encoder_lock_.get());
   return encoder()->DecodeRedundant(encoded, encoded_len, sample_rate_hz,
                                     max_decoded_bytes, decoded, speech_type);
 }
 
 bool AudioEncoderDecoderMutableIsacFloat::HasDecodePlc() const {
   CriticalSectionScoped cs(encoder_lock_.get());
   return encoder()->HasDecodePlc();
 }
 
-int AudioEncoderDecoderMutableIsacFloat::DecodePlc(int num_frames,
-                                                   int16_t* decoded) {
+size_t AudioEncoderDecoderMutableIsacFloat::DecodePlc(size_t num_frames,
+                                                      int16_t* decoded) {
   CriticalSectionScoped cs(encoder_lock_.get());
   return encoder()->DecodePlc(num_frames, decoded);
 }
 
 int AudioEncoderDecoderMutableIsacFloat::Init() {
   CriticalSectionScoped cs(encoder_lock_.get());
   return encoder()->Init();
 }
 
 int AudioEncoderDecoderMutableIsacFloat::IncomingPacket(
@@ skipping 32 matching lines @@
   CriticalSectionScoped cs(encoder_lock_.get());
   return encoder()->PacketHasFec(encoded, encoded_len);
 }
 
 size_t AudioEncoderDecoderMutableIsacFloat::Channels() const {
   CriticalSectionScoped cs(encoder_lock_.get());
   return encoder()->Channels();
 }
 
 }  // namespace webrtc