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

webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h

 /*
  *  Copyright (c) 2011 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 74 matching lines @@
                                  IsacSaveEncoderData* encData);
 
 int WebRtcIsacfix_DecodePitchGain(Bitstr_dec *streamdata,
                                   int16_t *PitchGain_Q12);
 
 int WebRtcIsacfix_DecodePitchLag(Bitstr_dec *streamdata,
                                  int16_t *PitchGain_Q12,
                                  int16_t *PitchLagQ7);
 
 int WebRtcIsacfix_DecodeFrameLen(Bitstr_dec *streamdata,
-                                 int16_t *framelength);
+                                 size_t *framelength);
 
 
 int WebRtcIsacfix_EncodeFrameLen(int16_t framelength,
                                  Bitstr_enc *streamdata);
 
 int WebRtcIsacfix_DecodeSendBandwidth(Bitstr_dec *streamdata,
                                       int16_t *BWno);
 
 
 int WebRtcIsacfix_EncodeReceiveBandwidth(int16_t *BWno,
@@ skipping 74 matching lines @@
                                       const int shift);
 
 void WebRtcIsacfix_MatrixProduct2MIPS(const int16_t matrix0[],
                                       const int32_t matrix1[],
                                       int32_t matrix_product[],
                                       const int matrix0_index_factor,
                                       const int matrix0_index_step);
 #endif
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_