Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/codecs/isac/main/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 77 matching lines @@
   /* arr_ts        - value given by timeGetTime() time of arrival in samples of packet from NetEq      */
   /* pksize        - size of packet in bytes, from NetEq                                               */
   /* Index         - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */
   /* returns 0 if everything went fine, -1 otherwise                                                   */
   int16_t WebRtcIsac_UpdateBandwidthEstimator(
       BwEstimatorstr* bwest_str,
       const uint16_t rtp_number,
       const int32_t frame_length,
       const uint32_t send_ts,
       const uint32_t arr_ts,
-      const int32_t pksize);
+      const size_t pksize);
 
   /* Update receiving estimates. Used when we only receive BWE index, no iSAC data packet. */
   int16_t WebRtcIsac_UpdateUplinkBwImpl(
       BwEstimatorstr*           bwest_str,
       int16_t               Index,
       enum IsacSamplingRate encoderSamplingFreq);
 
   /* Returns the bandwidth/jitter estimation code (integer 0...23) to put in the sending iSAC payload */
   void WebRtcIsac_GetDownlinkBwJitIndexImpl(
       BwEstimatorstr* bwest_str,
@@ skipping 66 matching lines @@
   int16_t WebRtcIsac_UpdateUplinkJitter(
       BwEstimatorstr*              bwest_str,
       int32_t                  index);
 
 #if defined(__cplusplus)
 }
 #endif
 
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_ */