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

webrtc/modules/audio_coding/codecs/audio_decoder.h

 /*
  *  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 44 matching lines @@
                               int sample_rate_hz,
                               size_t max_decoded_bytes,
                               int16_t* decoded,
                               SpeechType* speech_type);
 
   // Indicates if the decoder implements the DecodePlc method.
   virtual bool HasDecodePlc() const;
 
   // Calls the packet-loss concealment of the decoder to update the state after
   // one or several lost packets.
-  virtual int DecodePlc(int num_frames, int16_t* decoded);
+  virtual size_t DecodePlc(size_t num_frames, int16_t* decoded);
 
   // Initializes the decoder.
   virtual int Init() = 0;
 
   // Notifies the decoder of an incoming packet to NetEQ.
   virtual int IncomingPacket(const uint8_t* payload,
                              size_t payload_len,
                              uint16_t rtp_sequence_number,
                              uint32_t rtp_timestamp,
                              uint32_t arrival_timestamp);
@@ skipping 37 matching lines @@
                                       int sample_rate_hz,
                                       int16_t* decoded,
                                       SpeechType* speech_type);
 
  private:
   DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_AUDIO_DECODER_H_