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

webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.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 21 matching lines @@
                                    int frame_size_ms,
                                    int16_t enforce_frame_size) {
     return WebRtcIsac_ControlBwe(inst, rate_bps, frame_size_ms,
                                  enforce_frame_size);
   }
   static inline int16_t Create(instance_type** inst) {
     return WebRtcIsac_Create(inst);
   }
   static inline int DecodeInternal(instance_type* inst,
                                    const uint8_t* encoded,
-                                   int16_t len,
+                                   size_t len,
                                    int16_t* decoded,
                                    int16_t* speech_type) {
     return WebRtcIsac_Decode(inst, encoded, len, decoded, speech_type);
   }
-  static inline int16_t DecodePlc(instance_type* inst,
-                                  int16_t* decoded,
-                                  int16_t num_lost_frames) {
+  static inline size_t DecodePlc(instance_type* inst,
+                                 int16_t* decoded,
+                                 size_t num_lost_frames) {
     return WebRtcIsac_DecodePlc(inst, decoded, num_lost_frames);
   }
 
   static inline int16_t DecoderInit(instance_type* inst) {
     return WebRtcIsac_DecoderInit(inst);
   }
   static inline int Encode(instance_type* inst,
                            const int16_t* speech_in,
                            uint8_t* encoded) {
     return WebRtcIsac_Encode(inst, speech_in, encoded);
@@ skipping 35 matching lines @@
                                                 uint16_t sample_rate_hz) {
     WebRtcIsac_SetEncSampRateInDecoder(inst, sample_rate_hz);
   }
   static inline void SetInitialBweBottleneck(
       instance_type* inst,
       int bottleneck_bits_per_second) {
     WebRtcIsac_SetInitialBweBottleneck(inst, bottleneck_bits_per_second);
   }
   static inline int16_t UpdateBwEstimate(instance_type* inst,
                                          const uint8_t* encoded,
-                                         int32_t packet_size,
+                                         size_t packet_size,
                                          uint16_t rtp_seq_number,
                                          uint32_t send_ts,
                                          uint32_t arr_ts) {
     return WebRtcIsac_UpdateBwEstimate(inst, encoded, packet_size,
                                        rtp_seq_number, send_ts, arr_ts);
   }
   static inline int16_t SetMaxPayloadSize(instance_type* inst,
                                           int16_t max_payload_size_bytes) {
     return WebRtcIsac_SetMaxPayloadSize(inst, max_payload_size_bytes);
   }
@@ skipping 11 matching lines @@
     : public AudioEncoderMutableImpl<AudioEncoderIsac> {
  public:
   AudioEncoderMutableIsacFloat(const CodecInst& codec_inst,
                                LockedIsacBandwidthInfo* bwinfo);
   void SetMaxPayloadSize(int max_payload_size_bytes) override;
   void SetMaxRate(int max_rate_bps) override;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_