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

webrtc/modules/audio_coding/neteq/audio_decoder_impl.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 104 matching lines @@
 
  private:
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcm16B);
 };
 
 // This class handles all four types (i.e., sample rates) of PCM16B codecs.
 // The type is specified in the constructor parameter |type|, and the number
 // of channels is derived from the type.
 class AudioDecoderPcm16BMultiCh : public AudioDecoderPcm16B {
  public:
-  explicit AudioDecoderPcm16BMultiCh(int num_channels);
+  explicit AudioDecoderPcm16BMultiCh(size_t num_channels);
   size_t Channels() const override;
 
  private:
   const size_t channels_;
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcm16BMultiCh);
 };
 #endif
 
 #ifdef WEBRTC_CODEC_ILBC
 class AudioDecoderIlbc : public AudioDecoder {
  public:
   AudioDecoderIlbc();
   ~AudioDecoderIlbc() override;
   bool HasDecodePlc() const override;
-  int DecodePlc(int num_frames, int16_t* decoded) override;
+  size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
   int Init() override;
   size_t Channels() const override;
 
  protected:
   int DecodeInternal(const uint8_t* encoded,
                      size_t encoded_len,
                      int sample_rate_hz,
                      int16_t* decoded,
                      SpeechType* speech_type) override;
 
@@ skipping 51 matching lines @@
   G722DecInst* dec_state_left_;
   G722DecInst* dec_state_right_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722Stereo);
 };
 #endif
 
 #ifdef WEBRTC_CODEC_OPUS
 class AudioDecoderOpus : public AudioDecoder {
  public:
-  explicit AudioDecoderOpus(int num_channels);
+  explicit AudioDecoderOpus(size_t num_channels);
   ~AudioDecoderOpus() override;
 
   int Init() override;
   int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
   int PacketDurationRedundant(const uint8_t* encoded,
                               size_t encoded_len) const override;
   bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
   size_t Channels() const override;
 
  protected:
@@ skipping 84 matching lines @@
 // Returns the sample rate for |codec_type|.
 int CodecSampleRateHz(NetEqDecoder codec_type);
 
 // Creates an AudioDecoder object of type |codec_type|. Returns NULL for for
 // unsupported codecs, and when creating an AudioDecoder is not applicable
 // (e.g., for RED and DTMF/AVT types).
 AudioDecoder* CreateAudioDecoder(NetEqDecoder codec_type);
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_DECODER_IMPL_H_