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

webrtc/modules/audio_coding/neteq/buffer_level_filter.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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_
 
+#include <stddef.h>
+
 #include "webrtc/base/constructormagic.h"
 
 namespace webrtc {
 
 class BufferLevelFilter {
  public:
   BufferLevelFilter();
   virtual ~BufferLevelFilter() {}
   virtual void Reset();
 
   // Updates the filter. Current buffer size is |buffer_size_packets| (Q0).
   // If |time_stretched_samples| is non-zero, the value is converted to the
   // corresponding number of packets, and is subtracted from the filtered
   // value (thus bypassing the filter operation). |packet_len_samples| is the
   // number of audio samples carried in each incoming packet.
-  virtual void Update(int buffer_size_packets, int time_stretched_samples,
-                      int packet_len_samples);
+  virtual void Update(size_t buffer_size_packets, int time_stretched_samples,
+                      size_t packet_len_samples);
 
   // Set the current target buffer level (obtained from
   // DelayManager::base_target_level()). Used to select the appropriate
   // filter coefficient.
   virtual void SetTargetBufferLevel(int target_buffer_level);
 
   virtual int filtered_current_level() const;
 
  private:
   int level_factor_;  // Filter factor for the buffer level filter in Q8.
   int filtered_current_level_;  // Filtered current buffer level in Q8.
 
   DISALLOW_COPY_AND_ASSIGN(BufferLevelFilter);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_