NetEq: Implement Average Excess Buffer Delay stats

webrtc/modules/audio_coding/neteq/statistics_calculator.cc

 /*
  *  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.
  */
 
@@ skipping 25 matching lines @@
   timer_ -= kReportIntervalMs;
   DCHECK_GE(timer_, 0);
 }
 
 void StatisticsCalculator::DelayedPacketOutagesPerMinuteCounter::LogToUma()
     const {
   RTC_HISTOGRAM_COUNTS_100("WebRTC.Audio.DelayedPacketOutageEventsPerMinute",
                            counter_);
 }
 
+void StatisticsCalculator::AverageExcessBufferDelayMs::
+    RegisterPacketWaitingTime(int time_ms) {
+  sum_ += time_ms;
+  ++num_packets_;
+}
+
+void StatisticsCalculator::AverageExcessBufferDelayMs::AdvanceClock(
+    int step_ms) {
+  timer_ += step_ms;
+  if (timer_ < kReportIntervalMs) {
+    return;
+  }
+  LogToUma();
+  sum_ = 0.0;
+  num_packets_ = 0;
+  timer_ -= kReportIntervalMs;
+  DCHECK_GE(timer_, 0);
+}
+
+void StatisticsCalculator::AverageExcessBufferDelayMs::LogToUma() {
+  if (num_packets_ > 0) {
+    RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.AverageExcessBufferDelayMs",
+                              static_cast<int>(sum_ / num_packets_));
+  }
+}
+
 StatisticsCalculator::StatisticsCalculator()
     : preemptive_samples_(0),
       accelerate_samples_(0),
       added_zero_samples_(0),
       expanded_speech_samples_(0),
       expanded_noise_samples_(0),
       discarded_packets_(0),
       lost_timestamps_(0),
       timestamps_since_last_report_(0),
       len_waiting_times_(0),
@@ skipping 45 matching lines @@
 
 void StatisticsCalculator::PacketsDiscarded(int num_packets) {
   discarded_packets_ += num_packets;
 }
 
 void StatisticsCalculator::LostSamples(int num_samples) {
   lost_timestamps_ += num_samples;
 }
 
 void StatisticsCalculator::IncreaseCounter(int num_samples, int fs_hz) {
-  delayed_packet_outage_counter_.AdvanceClock(
-      rtc::CheckedDivExact(1000 * num_samples, fs_hz));
+  const int time_step_ms = rtc::CheckedDivExact(1000 * num_samples, fs_hz);
+  delayed_packet_outage_counter_.AdvanceClock(time_step_ms);
+  excess_buffer_delay_.AdvanceClock(time_step_ms);
   timestamps_since_last_report_ += static_cast<uint32_t>(num_samples);
   if (timestamps_since_last_report_ >
       static_cast<uint32_t>(fs_hz * kMaxReportPeriod)) {
     lost_timestamps_ = 0;
     timestamps_since_last_report_ = 0;
     discarded_packets_ = 0;
   }
 }
 
 void StatisticsCalculator::SecondaryDecodedSamples(int num_samples) {
   secondary_decoded_samples_ += num_samples;
 }
 
 void StatisticsCalculator::LogDelayedPacketOutageEvent(int outage_duration_ms) {
   RTC_HISTOGRAM_COUNTS("WebRTC.Audio.DelayedPacketOutageEventMs",
                        outage_duration_ms, 1 /* min */, 2000 /* max */,
                        100 /* bucket count */);
   delayed_packet_outage_counter_.RegisterEvent();
 }
 
 void StatisticsCalculator::StoreWaitingTime(int waiting_time_ms) {
+  excess_buffer_delay_.RegisterPacketWaitingTime(waiting_time_ms);
   assert(next_waiting_time_index_ < kLenWaitingTimes);
   waiting_times_[next_waiting_time_index_] = waiting_time_ms;
   next_waiting_time_index_++;
   if (next_waiting_time_index_ >= kLenWaitingTimes) {
     next_waiting_time_index_ = 0;
   }
   if (len_waiting_times_ < kLenWaitingTimes) {
     len_waiting_times_++;
   }
 }
@@ skipping 66 matching lines @@
     // Ratio must be smaller than 1 in Q14.
     assert((numerator << 14) / denominator < (1 << 14));
     return static_cast<uint16_t>((numerator << 14) / denominator);
   } else {
     // Will not produce a ratio larger than 1, since this is probably an error.
     return 1 << 14;
   }
 }
 
 }  // namespace webrtc