Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/neteq/delay_manager.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 14 matching lines @@
 class DelayPeakDetector;
 
 class DelayManager {
  public:
   typedef std::vector<int> IATVector;
 
   // Create a DelayManager object. Notify the delay manager that the packet
   // buffer can hold no more than |max_packets_in_buffer| packets (i.e., this
   // is the number of packet slots in the buffer). Supply a PeakDetector
   // object to the DelayManager.
-  DelayManager(int max_packets_in_buffer, DelayPeakDetector* peak_detector);
+  DelayManager(size_t max_packets_in_buffer, DelayPeakDetector* peak_detector);
 
   virtual ~DelayManager();
 
   // Read the inter-arrival time histogram. Mainly for testing purposes.
   virtual const IATVector& iat_vector() const;
 
   // Updates the delay manager with a new incoming packet, with
   // |sequence_number| and |timestamp| from the RTP header. This updates the
   // inter-arrival time histogram and other statistics, as well as the
   // associated DelayPeakDetector. A new target buffer level is calculated.
@@ skipping 79 matching lines @@
   // equal to |iat_packets| (in integer packets) is increased slightly, while
   // all other entries are decreased. This method is called by Update().
   void UpdateHistogram(size_t iat_packets);
 
   // Makes sure that |target_level_| is not too large, taking
   // |max_packets_in_buffer_| and |extra_delay_ms_| into account. This method is
   // called by Update().
   void LimitTargetLevel();
 
   bool first_packet_received_;
-  const int max_packets_in_buffer_;  // Capacity of the packet buffer.
+  const size_t max_packets_in_buffer_;  // Capacity of the packet buffer.
   IATVector iat_vector_;  // Histogram of inter-arrival times.
   int iat_factor_;  // Forgetting factor for updating the IAT histogram (Q15).
   int packet_iat_count_ms_;  // Milliseconds elapsed since last packet.
   int base_target_level_;   // Currently preferred buffer level before peak
                             // detection and streaming mode (Q0).
   // TODO(turajs) change the comment according to the implementation of
   // minimum-delay.
   int target_level_;  // Currently preferred buffer level in (fractions)
                       // of packets (Q8), before adding any extra delay.
   int packet_len_ms_;  // Length of audio in each incoming packet [ms].
   bool streaming_mode_;
   uint16_t last_seq_no_;  // Sequence number for last received packet.
   uint32_t last_timestamp_;  // Timestamp for the last received packet.
   int minimum_delay_ms_;  // Externally set minimum delay.
   int least_required_delay_ms_;  // Smallest preferred buffer level (same unit
                               // as |target_level_|), before applying
                               // |minimum_delay_ms_| and/or |maximum_delay_ms_|.
   int maximum_delay_ms_;  // Externally set maximum allowed delay.
   int iat_cumulative_sum_;  // Cumulative sum of delta inter-arrival times.
   int max_iat_cumulative_sum_;  // Max of |iat_cumulative_sum_|.
   int max_timer_ms_;  // Time elapsed since maximum was observed.
   DelayPeakDetector& peak_detector_;
   int last_pack_cng_or_dtmf_;
 
   DISALLOW_COPY_AND_ASSIGN(DelayManager);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_