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

webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h

 /*
  *  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 37 matching lines @@
     // in nonadaptive mode.)
     bool enforce_frame_size;
   };
 
   explicit AudioEncoderIsacT(const Config& config);
   ~AudioEncoderIsacT() override;
 
   int SampleRateHz() const override;
   int NumChannels() const override;
   size_t MaxEncodedBytes() const override;
-  int Num10MsFramesInNextPacket() const override;
-  int Max10MsFramesInAPacket() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
                              const int16_t* audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
 
  private:
   // This value is taken from STREAM_SIZE_MAX_60 for iSAC float (60 ms) and
   // STREAM_MAXW16_60MS for iSAC fix (60 ms).
   static const size_t kSufficientEncodeBufferSizeBytes = 400;
@@ skipping 17 matching lines @@
 };
 
 template <typename T>
 class AudioDecoderIsacT final : public AudioDecoder {
  public:
   AudioDecoderIsacT();
   explicit AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo);
   ~AudioDecoderIsacT() 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;
   int IncomingPacket(const uint8_t* payload,
                      size_t payload_len,
                      uint16_t rtp_sequence_number,
                      uint32_t rtp_timestamp,
                      uint32_t arrival_timestamp) override;
   int ErrorCode() override;
   size_t Channels() const override;
   int DecodeInternal(const uint8_t* encoded,
                      size_t encoded_len,
                      int sample_rate_hz,
                      int16_t* decoded,
                      SpeechType* speech_type) override;
 
  private:
   typename T::instance_type* isac_state_;
   LockedIsacBandwidthInfo* bwinfo_;
   int decoder_sample_rate_hz_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_