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

webrtc/modules/audio_processing/ns/noise_suppression.c

 /*
  *  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 23 matching lines @@
 int WebRtcNs_set_policy(NsHandle* NS_inst, int mode) {
   return WebRtcNs_set_policy_core((NoiseSuppressionC*)NS_inst, mode);
 }
 
 void WebRtcNs_Analyze(NsHandle* NS_inst, const float* spframe) {
   WebRtcNs_AnalyzeCore((NoiseSuppressionC*)NS_inst, spframe);
 }
 
 void WebRtcNs_Process(NsHandle* NS_inst,
                       const float* const* spframe,
-                      int num_bands,
+                      size_t num_bands,
                       float* const* outframe) {
   WebRtcNs_ProcessCore((NoiseSuppressionC*)NS_inst, spframe, num_bands,
                        outframe);
 }
 
 float WebRtcNs_prior_speech_probability(NsHandle* handle) {
   NoiseSuppressionC* self = (NoiseSuppressionC*)handle;
   if (handle == NULL) {
     return -1;
   }
   if (self->initFlag == 0) {
     return -1;
   }
   return self->priorSpeechProb;
 }