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 <vector>
+#include <list>
 
 #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(int num_samples);
 
   // Reports that |num_samples| samples were produced through expansion, and
   // that the expansion produced only noise samples.
   void ExpandedNoiseSamples(int num_samples);
 
   // Reports that |num_samples| samples were produced through preemptive
   // expansion.
@@ skipping 25 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,
                             int num_samples_in_buffers,
                             int 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;
 
   // Calculates numerator / denominator, and returns the value in Q14.
   static uint16_t CalculateQ14Ratio(uint32_t numerator, uint32_t denominator);
 
   uint32_t preemptive_samples_;
   uint32_t accelerate_samples_;
   int added_zero_samples_;
   uint32_t expanded_speech_samples_;
   uint32_t expanded_noise_samples_;
   int discarded_packets_;
   uint32_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::list<int> waiting_times_;

[minyue-webrtc, 2015/08/18 12:55:34]

why to change from vector to list?

[hlundin-webrtc, 2015/08/19 08:28:39]

The reasoning behind my choice is that waiting_times_ might in many cases be
acting as a FIFO buffer. In particular, if the list is full
(waiting_times_.size() >= kLenWaitingTimes), every new value being pushed to the
end will result in the oldest value being popped from the beginning of the list.
That is not very efficient to do with vectors, which is why I went with
std:list. Your input on this is more than welcome.

[minyue-webrtc, 2015/08/19 11:21:52]

maybe use std::deque

[hlundin-webrtc, 2015/08/19 14:34:49]

Yes, much better. Thanks!

   uint32_t secondary_decoded_samples_;
 
   DISALLOW_COPY_AND_ASSIGN(StatisticsCalculator);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_