Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/neteq/buffer_level_filter.cc

 /*
  *  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.
  */
 
 #include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
 
 #include <algorithm>  // Provide access to std::max.
 
 namespace webrtc {
 
 BufferLevelFilter::BufferLevelFilter() {
   Reset();
 }
 
 void BufferLevelFilter::Reset() {
   filtered_current_level_ = 0;
   level_factor_ = 253;
 }
 
-void BufferLevelFilter::Update(int buffer_size_packets,
+void BufferLevelFilter::Update(size_t buffer_size_packets,
                                int time_stretched_samples,
-                               int packet_len_samples) {
+                               size_t packet_len_samples) {
   // Filter:
   // |filtered_current_level_| = |level_factor_| * |filtered_current_level_| +
   //                            (1 - |level_factor_|) * |buffer_size_packets|
   // |level_factor_| and |filtered_current_level_| are in Q8.
   // |buffer_size_packets| is in Q0.
   filtered_current_level_ = ((level_factor_ * filtered_current_level_) >> 8) +
-      ((256 - level_factor_) * buffer_size_packets);
+      ((256 - level_factor_) * static_cast<int>(buffer_size_packets));
 
   // Account for time-scale operations (accelerate and pre-emptive expand).
   if (time_stretched_samples && packet_len_samples > 0) {
     // Time-scaling has been performed since last filter update. Subtract the
     // value of |time_stretched_samples| from |filtered_current_level_| after
     // converting |time_stretched_samples| from samples to packets in Q8.
     // Make sure that the filtered value remains non-negative.
     filtered_current_level_ = std::max(0,
         filtered_current_level_ -
-        (time_stretched_samples << 8) / packet_len_samples);
+        (time_stretched_samples << 8) / static_cast<int>(packet_len_samples));
   }
 }
 
 void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level) {
   if (target_buffer_level <= 1) {
     level_factor_ = 251;
   } else if (target_buffer_level <= 3) {
     level_factor_ = 252;
   } else if (target_buffer_level <= 7) {
     level_factor_ = 253;
   } else {
     level_factor_ = 254;
   }
 }
 
 int BufferLevelFilter::filtered_current_level() const {
   return filtered_current_level_;
 }
 
 }  // namespace webrtc