Update audio code to use size_t more correctly, webrtc/modules/audio_processing/

webrtc/modules/audio_processing/aec/aec_resampler.h

 /*
  *  Copyright (c) 2012 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 13 matching lines @@
 void* WebRtcAec_CreateResampler();  // Returns NULL on error.
 int WebRtcAec_InitResampler(void* resampInst, int deviceSampleRateHz);
 void WebRtcAec_FreeResampler(void* resampInst);
 
 // Estimates skew from raw measurement.
 int WebRtcAec_GetSkew(void* resampInst, int rawSkew, float* skewEst);
 
 // Resamples input using linear interpolation.
 void WebRtcAec_ResampleLinear(void* resampInst,
                               const float* inspeech,
-                              int size,
+                              size_t size,
                               float skew,
                               float* outspeech,
-                              int* size_out);
+                              size_t* size_out);
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_