Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.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 44 matching lines @@
  *
  * Return value            : 0 if everything went fine,
  *                           -1 otherwise
  */
 
 int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr       *bwest_str,
                                          const uint16_t        rtp_number,
                                          const int16_t         frameSize,
                                          const uint32_t        send_ts,
                                          const uint32_t        arr_ts,
-                                         const int16_t         pksize,
+                                         const size_t          pksize,
                                          const uint16_t        Index);
 
 /* Update receiving estimates. Used when we only receive BWE index, no iSAC data packet. */
 int16_t WebRtcIsacfix_UpdateUplinkBwRec(BwEstimatorstr *bwest_str,
                                         const int16_t Index);
 
 /****************************************************************************
  * WebRtcIsacfix_GetDownlinkBwIndexImpl(...)
  *
  * This function calculates and returns the bandwidth/jitter estimation code
@@ skipping 50 matching lines @@
 
 /* Returns the new framelength value (input argument: bottle_neck) */
 int16_t WebRtcIsacfix_GetNewFrameLength(int16_t bottle_neck, int16_t current_framelength);
 
 /* Returns the new SNR value (input argument: bottle_neck) */
 //returns snr in Q10
 int16_t WebRtcIsacfix_GetSnr(int16_t bottle_neck, int16_t framesamples);
 
 
 #endif /*  WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_ */