NetEq/ACM: Refactor how packet waiting times are calculated

webrtc/modules/audio_coding/neteq/statistics_calculator.h

 /*
  *  Copyright (c) 2013 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_STATISTICS_CALCULATOR_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_
 
+#include <deque>
 #include <string>
-#include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // Forward declarations.
 class DecisionLogic;
 class DelayManager;
 
 // This class handles various network statistics in NetEq.
 class StatisticsCalculator {
  public:
   StatisticsCalculator();
 
-  virtual ~StatisticsCalculator() {}
+  virtual ~StatisticsCalculator();
 
   // Resets most of the counters.
   void Reset();
 
   // Resets the counters that are not handled by Reset().
   void ResetMcu();
 
-  // Resets the waiting time statistics.
-  void ResetWaitingTimeStatistics();
-
   // Reports that |num_samples| samples were produced through expansion, and
   // that the expansion produced other than just noise samples.
   void ExpandedVoiceSamples(size_t num_samples);
 
   // Reports that |num_samples| samples were produced through expansion, and
   // that the expansion produced only noise samples.
   void ExpandedNoiseSamples(size_t num_samples);
 
   // Reports that |num_samples| samples were produced through preemptive
   // expansion.
@@ skipping 31 matching lines @@
   // is |fs_hz|, the total number of samples in packet buffer and sync buffer
   // yet to play out is |num_samples_in_buffers|, and the number of samples per
   // packet is |samples_per_packet|.
   void GetNetworkStatistics(int fs_hz,
                             size_t num_samples_in_buffers,
                             size_t samples_per_packet,
                             const DelayManager& delay_manager,
                             const DecisionLogic& decision_logic,
                             NetEqNetworkStatistics *stats);
 
-  void WaitingTimes(std::vector<int>* waiting_times);
-
  private:
   static const int kMaxReportPeriod = 60;  // Seconds before auto-reset.
-  static const int kLenWaitingTimes = 100;
+  static const size_t kLenWaitingTimes = 100;
 
   class PeriodicUmaLogger {
    public:
     PeriodicUmaLogger(const std::string& uma_name,
                       int report_interval_ms,
                       int max_value);
     virtual ~PeriodicUmaLogger();
     void AdvanceClock(int step_ms);
 
    protected:
@@ skipping 44 matching lines @@
   static uint16_t CalculateQ14Ratio(size_t numerator, uint32_t denominator);
 
   size_t preemptive_samples_;
   size_t accelerate_samples_;
   size_t added_zero_samples_;
   size_t expanded_speech_samples_;
   size_t expanded_noise_samples_;
   size_t discarded_packets_;
   size_t lost_timestamps_;
   uint32_t timestamps_since_last_report_;
-  int waiting_times_[kLenWaitingTimes];  // Used as a circular buffer.
-  int len_waiting_times_;
-  int next_waiting_time_index_;
+  std::deque<int> waiting_times_;
   uint32_t secondary_decoded_samples_;
   PeriodicUmaCount delayed_packet_outage_counter_;
   PeriodicUmaAverage excess_buffer_delay_;
 
   DISALLOW_COPY_AND_ASSIGN(StatisticsCalculator);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_