modules/audio_coding/neteq/statistics_calculator.cc - src - Git at Google

 /*
  *  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.
  */

 #include "modules/audio_coding/neteq/statistics_calculator.h"

 #include <string.h>  // memset

 #include <algorithm>

 #include "modules/audio_coding/neteq/delay_manager.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "system_wrappers/include/metrics.h"

 namespace webrtc {

 namespace {
 size_t AddIntToSizeTWithLowerCap(int a, size_t b) {
   const size_t ret = b + a;
   // If a + b is negative, resulting in a negative wrap, cap it to zero instead.
   static_assert(sizeof(size_t) >= sizeof(int),
                 "int must not be wider than size_t for this to work");
   return (a < 0 && ret > b) ? 0 : ret;
 }

 constexpr int kInterruptionLenMs = 150;
 }  // namespace

 // Allocating the static const so that it can be passed by reference to
 // RTC_DCHECK.
 const size_t StatisticsCalculator::kLenWaitingTimes;

 StatisticsCalculator::PeriodicUmaLogger::PeriodicUmaLogger(
     const std::string& uma_name,
     int report_interval_ms,
     int max_value)
     : uma_name_(uma_name),
       report_interval_ms_(report_interval_ms),
       max_value_(max_value),
       timer_(0) {}

 StatisticsCalculator::PeriodicUmaLogger::~PeriodicUmaLogger() = default;

 void StatisticsCalculator::PeriodicUmaLogger::AdvanceClock(int step_ms) {
   timer_ += step_ms;
   if (timer_ < report_interval_ms_) {
     return;
   }
   LogToUma(Metric());
   Reset();
   timer_ -= report_interval_ms_;
   RTC_DCHECK_GE(timer_, 0);
 }

 void StatisticsCalculator::PeriodicUmaLogger::LogToUma(int value) const {
   RTC_HISTOGRAM_COUNTS_SPARSE(uma_name_, value, 1, max_value_, 50);
 }

 StatisticsCalculator::PeriodicUmaCount::PeriodicUmaCount(
     const std::string& uma_name,
     int report_interval_ms,
     int max_value)
     : PeriodicUmaLogger(uma_name, report_interval_ms, max_value) {}

 StatisticsCalculator::PeriodicUmaCount::~PeriodicUmaCount() {
   // Log the count for the current (incomplete) interval.
   LogToUma(Metric());
 }

 void StatisticsCalculator::PeriodicUmaCount::RegisterSample() {
   ++counter_;
 }

 int StatisticsCalculator::PeriodicUmaCount::Metric() const {
   return counter_;
 }

 void StatisticsCalculator::PeriodicUmaCount::Reset() {
   counter_ = 0;
 }

 StatisticsCalculator::PeriodicUmaAverage::PeriodicUmaAverage(
     const std::string& uma_name,
     int report_interval_ms,
     int max_value)
     : PeriodicUmaLogger(uma_name, report_interval_ms, max_value) {}

 StatisticsCalculator::PeriodicUmaAverage::~PeriodicUmaAverage() {
   // Log the average for the current (incomplete) interval.
   LogToUma(Metric());
 }

 void StatisticsCalculator::PeriodicUmaAverage::RegisterSample(int value) {
   sum_ += value;
   ++counter_;
 }

 int StatisticsCalculator::PeriodicUmaAverage::Metric() const {
   return counter_ == 0 ? 0 : static_cast<int>(sum_ / counter_);
 }

 void StatisticsCalculator::PeriodicUmaAverage::Reset() {
   sum_ = 0.0;
   counter_ = 0;
 }

 StatisticsCalculator::StatisticsCalculator()
     : preemptive_samples_(0),
       accelerate_samples_(0),
       expanded_speech_samples_(0),
       expanded_noise_samples_(0),
       timestamps_since_last_report_(0),
       secondary_decoded_samples_(0),
       discarded_secondary_packets_(0),
       delayed_packet_outage_counter_(
           "WebRTC.Audio.DelayedPacketOutageEventsPerMinute",
           60000,  // 60 seconds report interval.
           100),
       excess_buffer_delay_("WebRTC.Audio.AverageExcessBufferDelayMs",
                            60000,  // 60 seconds report interval.
                            1000),
       buffer_full_counter_("WebRTC.Audio.JitterBufferFullPerMinute",
                            60000,  // 60 seconds report interval.
                            100) {}

 StatisticsCalculator::~StatisticsCalculator() = default;

 void StatisticsCalculator::Reset() {
   preemptive_samples_ = 0;
   accelerate_samples_ = 0;
   expanded_speech_samples_ = 0;
   expanded_noise_samples_ = 0;
   secondary_decoded_samples_ = 0;
   discarded_secondary_packets_ = 0;
   waiting_times_.clear();
 }

 void StatisticsCalculator::ResetMcu() {
   timestamps_since_last_report_ = 0;
 }

 void StatisticsCalculator::ExpandedVoiceSamples(size_t num_samples,
                                                 bool is_new_concealment_event) {
   expanded_speech_samples_ += num_samples;
   ConcealedSamplesCorrection(rtc::dchecked_cast<int>(num_samples), true);
   lifetime_stats_.concealment_events += is_new_concealment_event;
 }

 void StatisticsCalculator::ExpandedNoiseSamples(size_t num_samples,
                                                 bool is_new_concealment_event) {
   expanded_noise_samples_ += num_samples;
   ConcealedSamplesCorrection(rtc::dchecked_cast<int>(num_samples), false);
   lifetime_stats_.concealment_events += is_new_concealment_event;
 }

 void StatisticsCalculator::ExpandedVoiceSamplesCorrection(int num_samples) {
   expanded_speech_samples_ =
       AddIntToSizeTWithLowerCap(num_samples, expanded_speech_samples_);
   ConcealedSamplesCorrection(num_samples, true);
 }

 void StatisticsCalculator::ExpandedNoiseSamplesCorrection(int num_samples) {
   expanded_noise_samples_ =
       AddIntToSizeTWithLowerCap(num_samples, expanded_noise_samples_);
   ConcealedSamplesCorrection(num_samples, false);
 }

 void StatisticsCalculator::DecodedOutputPlayed() {
   decoded_output_played_ = true;
 }

 void StatisticsCalculator::EndExpandEvent(int fs_hz) {
   RTC_DCHECK_GE(lifetime_stats_.concealed_samples,
                 concealed_samples_at_event_end_);
   const int event_duration_ms =
       1000 *
       (lifetime_stats_.concealed_samples - concealed_samples_at_event_end_) /
       fs_hz;
   if (event_duration_ms >= kInterruptionLenMs && decoded_output_played_) {
     lifetime_stats_.interruption_count++;
     lifetime_stats_.total_interruption_duration_ms += event_duration_ms;
     RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AudioInterruptionMs", event_duration_ms,
                          /*min=*/150, /*max=*/5000, /*bucket_count=*/50);
   }
   concealed_samples_at_event_end_ = lifetime_stats_.concealed_samples;
 }

 void StatisticsCalculator::ConcealedSamplesCorrection(int num_samples,
                                                       bool is_voice) {
   if (num_samples < 0) {
     // Store negative correction to subtract from future positive additions.
     // See also the function comment in the header file.
     concealed_samples_correction_ -= num_samples;
     if (!is_voice) {
       silent_concealed_samples_correction_ -= num_samples;
     }
     return;
   }

   const size_t canceled_out =
       std::min(static_cast<size_t>(num_samples), concealed_samples_correction_);
   concealed_samples_correction_ -= canceled_out;
   lifetime_stats_.concealed_samples += num_samples - canceled_out;

   if (!is_voice) {
     const size_t silent_canceled_out = std::min(
         static_cast<size_t>(num_samples), silent_concealed_samples_correction_);
     silent_concealed_samples_correction_ -= silent_canceled_out;
     lifetime_stats_.silent_concealed_samples +=
         num_samples - silent_canceled_out;
   }
 }

 void StatisticsCalculator::PreemptiveExpandedSamples(size_t num_samples) {
   preemptive_samples_ += num_samples;
   operations_and_state_.preemptive_samples += num_samples;
   lifetime_stats_.inserted_samples_for_deceleration += num_samples;
 }

 void StatisticsCalculator::AcceleratedSamples(size_t num_samples) {
   accelerate_samples_ += num_samples;
   operations_and_state_.accelerate_samples += num_samples;
   lifetime_stats_.removed_samples_for_acceleration += num_samples;
 }

 void StatisticsCalculator::PacketsDiscarded(size_t num_packets) {
   operations_and_state_.discarded_primary_packets += num_packets;
 }

 void StatisticsCalculator::SecondaryPacketsDiscarded(size_t num_packets) {
   discarded_secondary_packets_ += num_packets;
   lifetime_stats_.fec_packets_discarded += num_packets;
 }

 void StatisticsCalculator::SecondaryPacketsReceived(size_t num_packets) {
   lifetime_stats_.fec_packets_received += num_packets;
 }

 void StatisticsCalculator::IncreaseCounter(size_t num_samples, int fs_hz) {
   const int time_step_ms =
       rtc::CheckedDivExact(static_cast<int>(1000 * num_samples), fs_hz);
   delayed_packet_outage_counter_.AdvanceClock(time_step_ms);
   excess_buffer_delay_.AdvanceClock(time_step_ms);
   buffer_full_counter_.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)) {
     timestamps_since_last_report_ = 0;
   }
   lifetime_stats_.total_samples_received += num_samples;
 }

 void StatisticsCalculator::JitterBufferDelay(size_t num_samples,
                                              uint64_t waiting_time_ms,
                                              uint64_t target_delay_ms) {
   lifetime_stats_.jitter_buffer_delay_ms += waiting_time_ms * num_samples;
   lifetime_stats_.jitter_buffer_target_delay_ms +=
       target_delay_ms * num_samples;
   lifetime_stats_.jitter_buffer_emitted_count += num_samples;
 }

 void StatisticsCalculator::SecondaryDecodedSamples(int num_samples) {
   secondary_decoded_samples_ += num_samples;
 }

 void StatisticsCalculator::FlushedPacketBuffer() {
   operations_and_state_.packet_buffer_flushes++;
   buffer_full_counter_.RegisterSample();
 }

 void StatisticsCalculator::ReceivedPacket() {
   ++lifetime_stats_.jitter_buffer_packets_received;
 }

 void StatisticsCalculator::RelativePacketArrivalDelay(size_t delay_ms) {
   lifetime_stats_.relative_packet_arrival_delay_ms += delay_ms;
 }

 void StatisticsCalculator::LogDelayedPacketOutageEvent(int num_samples,
                                                        int fs_hz) {
   int outage_duration_ms = num_samples / (fs_hz / 1000);
   RTC_HISTOGRAM_COUNTS("WebRTC.Audio.DelayedPacketOutageEventMs",
                        outage_duration_ms, 1 /* min */, 2000 /* max */,
                        100 /* bucket count */);
   delayed_packet_outage_counter_.RegisterSample();
   lifetime_stats_.delayed_packet_outage_samples += num_samples;
 }

 void StatisticsCalculator::StoreWaitingTime(int waiting_time_ms) {
   excess_buffer_delay_.RegisterSample(waiting_time_ms);
   RTC_DCHECK_LE(waiting_times_.size(), kLenWaitingTimes);
   if (waiting_times_.size() == kLenWaitingTimes) {
     // Erase first value.
     waiting_times_.pop_front();
   }
   waiting_times_.push_back(waiting_time_ms);
   operations_and_state_.last_waiting_time_ms = waiting_time_ms;
 }

 void StatisticsCalculator::GetNetworkStatistics(size_t samples_per_packet,
                                                 NetEqNetworkStatistics* stats) {
   RTC_DCHECK(stats);

   stats->accelerate_rate =
       CalculateQ14Ratio(accelerate_samples_, timestamps_since_last_report_);

   stats->preemptive_rate =
       CalculateQ14Ratio(preemptive_samples_, timestamps_since_last_report_);

   stats->expand_rate =
       CalculateQ14Ratio(expanded_speech_samples_ + expanded_noise_samples_,
                         timestamps_since_last_report_);

   stats->speech_expand_rate = CalculateQ14Ratio(expanded_speech_samples_,
                                                 timestamps_since_last_report_);

   stats->secondary_decoded_rate = CalculateQ14Ratio(
       secondary_decoded_samples_, timestamps_since_last_report_);

   const size_t discarded_secondary_samples =
       discarded_secondary_packets_ * samples_per_packet;
   stats->secondary_discarded_rate =
       CalculateQ14Ratio(discarded_secondary_samples,
                         static_cast<uint32_t>(discarded_secondary_samples +
                                               secondary_decoded_samples_));

   if (waiting_times_.size() == 0) {
     stats->mean_waiting_time_ms = -1;
     stats->median_waiting_time_ms = -1;
     stats->min_waiting_time_ms = -1;
     stats->max_waiting_time_ms = -1;
   } else {
     std::sort(waiting_times_.begin(), waiting_times_.end());
     // Find mid-point elements. If the size is odd, the two values
     // `middle_left` and `middle_right` will both be the one middle element; if
     // the size is even, they will be the the two neighboring elements at the
     // middle of the list.
     const int middle_left = waiting_times_[(waiting_times_.size() - 1) / 2];
     const int middle_right = waiting_times_[waiting_times_.size() / 2];
     // Calculate the average of the two. (Works also for odd sizes.)
     stats->median_waiting_time_ms = (middle_left + middle_right) / 2;
     stats->min_waiting_time_ms = waiting_times_.front();
     stats->max_waiting_time_ms = waiting_times_.back();
     double sum = 0;
     for (auto time : waiting_times_) {
       sum += time;
     }
     stats->mean_waiting_time_ms = static_cast<int>(sum / waiting_times_.size());
   }

   // Reset counters.
   ResetMcu();
   Reset();
 }

 NetEqLifetimeStatistics StatisticsCalculator::GetLifetimeStatistics() const {
   return lifetime_stats_;
 }

 NetEqOperationsAndState StatisticsCalculator::GetOperationsAndState() const {
   return operations_and_state_;
 }

 uint16_t StatisticsCalculator::CalculateQ14Ratio(size_t numerator,
                                                  uint32_t denominator) {
   if (numerator == 0) {
     return 0;
   } else if (numerator < denominator) {
     // Ratio must be smaller than 1 in Q14.
     RTC_DCHECK_LT((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
	/*
	* 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.
	*/

	#include "modules/audio_coding/neteq/statistics_calculator.h"

	#include <string.h> // memset

	#include <algorithm>

	#include "modules/audio_coding/neteq/delay_manager.h"
	#include "rtc_base/checks.h"
	#include "rtc_base/numerics/safe_conversions.h"
	#include "system_wrappers/include/metrics.h"

	namespace webrtc {

	namespace {
	size_t AddIntToSizeTWithLowerCap(int a, size_t b) {
	const size_t ret = b + a;
	// If a + b is negative, resulting in a negative wrap, cap it to zero instead.
	static_assert(sizeof(size_t) >= sizeof(int),
	"int must not be wider than size_t for this to work");
	return (a < 0 && ret > b) ? 0 : ret;
	}

	constexpr int kInterruptionLenMs = 150;
	} // namespace

	// Allocating the static const so that it can be passed by reference to
	// RTC_DCHECK.
	const size_t StatisticsCalculator::kLenWaitingTimes;

	StatisticsCalculator::PeriodicUmaLogger::PeriodicUmaLogger(
	const std::string& uma_name,
	int report_interval_ms,
	int max_value)
	: uma_name_(uma_name),
	report_interval_ms_(report_interval_ms),
	max_value_(max_value),
	timer_(0) {}

	StatisticsCalculator::PeriodicUmaLogger::~PeriodicUmaLogger() = default;

	void StatisticsCalculator::PeriodicUmaLogger::AdvanceClock(int step_ms) {
	timer_ += step_ms;
	if (timer_ < report_interval_ms_) {
	return;
	}
	LogToUma(Metric());
	Reset();
	timer_ -= report_interval_ms_;
	RTC_DCHECK_GE(timer_, 0);
	}

	void StatisticsCalculator::PeriodicUmaLogger::LogToUma(int value) const {
	RTC_HISTOGRAM_COUNTS_SPARSE(uma_name_, value, 1, max_value_, 50);
	}

	StatisticsCalculator::PeriodicUmaCount::PeriodicUmaCount(
	const std::string& uma_name,
	int report_interval_ms,
	int max_value)
	: PeriodicUmaLogger(uma_name, report_interval_ms, max_value) {}

	StatisticsCalculator::PeriodicUmaCount::~PeriodicUmaCount() {
	// Log the count for the current (incomplete) interval.
	LogToUma(Metric());
	}

	void StatisticsCalculator::PeriodicUmaCount::RegisterSample() {
	++counter_;
	}

	int StatisticsCalculator::PeriodicUmaCount::Metric() const {
	return counter_;
	}

	void StatisticsCalculator::PeriodicUmaCount::Reset() {
	counter_ = 0;
	}

	StatisticsCalculator::PeriodicUmaAverage::PeriodicUmaAverage(
	const std::string& uma_name,
	int report_interval_ms,
	int max_value)
	: PeriodicUmaLogger(uma_name, report_interval_ms, max_value) {}

	StatisticsCalculator::PeriodicUmaAverage::~PeriodicUmaAverage() {
	// Log the average for the current (incomplete) interval.
	LogToUma(Metric());
	}

	void StatisticsCalculator::PeriodicUmaAverage::RegisterSample(int value) {
	sum_ += value;
	++counter_;
	}

	int StatisticsCalculator::PeriodicUmaAverage::Metric() const {
	return counter_ == 0 ? 0 : static_cast<int>(sum_ / counter_);
	}

	void StatisticsCalculator::PeriodicUmaAverage::Reset() {
	sum_ = 0.0;
	counter_ = 0;
	}

	StatisticsCalculator::StatisticsCalculator()
	: preemptive_samples_(0),
	accelerate_samples_(0),
	expanded_speech_samples_(0),
	expanded_noise_samples_(0),
	timestamps_since_last_report_(0),
	secondary_decoded_samples_(0),
	discarded_secondary_packets_(0),
	delayed_packet_outage_counter_(
	"WebRTC.Audio.DelayedPacketOutageEventsPerMinute",
	60000, // 60 seconds report interval.
	100),
	excess_buffer_delay_("WebRTC.Audio.AverageExcessBufferDelayMs",
	60000, // 60 seconds report interval.
	1000),
	buffer_full_counter_("WebRTC.Audio.JitterBufferFullPerMinute",
	60000, // 60 seconds report interval.
	100) {}

	StatisticsCalculator::~StatisticsCalculator() = default;

	void StatisticsCalculator::Reset() {
	preemptive_samples_ = 0;
	accelerate_samples_ = 0;
	expanded_speech_samples_ = 0;
	expanded_noise_samples_ = 0;
	secondary_decoded_samples_ = 0;
	discarded_secondary_packets_ = 0;
	waiting_times_.clear();
	}

	void StatisticsCalculator::ResetMcu() {
	timestamps_since_last_report_ = 0;
	}

	void StatisticsCalculator::ExpandedVoiceSamples(size_t num_samples,
	bool is_new_concealment_event) {
	expanded_speech_samples_ += num_samples;
	ConcealedSamplesCorrection(rtc::dchecked_cast<int>(num_samples), true);
	lifetime_stats_.concealment_events += is_new_concealment_event;
	}

	void StatisticsCalculator::ExpandedNoiseSamples(size_t num_samples,
	bool is_new_concealment_event) {
	expanded_noise_samples_ += num_samples;
	ConcealedSamplesCorrection(rtc::dchecked_cast<int>(num_samples), false);
	lifetime_stats_.concealment_events += is_new_concealment_event;
	}

	void StatisticsCalculator::ExpandedVoiceSamplesCorrection(int num_samples) {
	expanded_speech_samples_ =
	AddIntToSizeTWithLowerCap(num_samples, expanded_speech_samples_);
	ConcealedSamplesCorrection(num_samples, true);
	}

	void StatisticsCalculator::ExpandedNoiseSamplesCorrection(int num_samples) {
	expanded_noise_samples_ =
	AddIntToSizeTWithLowerCap(num_samples, expanded_noise_samples_);
	ConcealedSamplesCorrection(num_samples, false);
	}

	void StatisticsCalculator::DecodedOutputPlayed() {
	decoded_output_played_ = true;
	}

	void StatisticsCalculator::EndExpandEvent(int fs_hz) {
	RTC_DCHECK_GE(lifetime_stats_.concealed_samples,
	concealed_samples_at_event_end_);
	const int event_duration_ms =
	1000 *
	(lifetime_stats_.concealed_samples - concealed_samples_at_event_end_) /
	fs_hz;
	if (event_duration_ms >= kInterruptionLenMs && decoded_output_played_) {
	lifetime_stats_.interruption_count++;
	lifetime_stats_.total_interruption_duration_ms += event_duration_ms;
	RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AudioInterruptionMs", event_duration_ms,
	/min=/150, /max=/5000, /bucket_count=/50);
	}
	concealed_samples_at_event_end_ = lifetime_stats_.concealed_samples;
	}

	void StatisticsCalculator::ConcealedSamplesCorrection(int num_samples,
	bool is_voice) {
	if (num_samples < 0) {
	// Store negative correction to subtract from future positive additions.
	// See also the function comment in the header file.
	concealed_samples_correction_ -= num_samples;
	if (!is_voice) {
	silent_concealed_samples_correction_ -= num_samples;
	}
	return;
	}

	const size_t canceled_out =
	std::min(static_cast<size_t>(num_samples), concealed_samples_correction_);
	concealed_samples_correction_ -= canceled_out;
	lifetime_stats_.concealed_samples += num_samples - canceled_out;

	if (!is_voice) {
	const size_t silent_canceled_out = std::min(
	static_cast<size_t>(num_samples), silent_concealed_samples_correction_);
	silent_concealed_samples_correction_ -= silent_canceled_out;
	lifetime_stats_.silent_concealed_samples +=
	num_samples - silent_canceled_out;
	}
	}

	void StatisticsCalculator::PreemptiveExpandedSamples(size_t num_samples) {
	preemptive_samples_ += num_samples;
	operations_and_state_.preemptive_samples += num_samples;
	lifetime_stats_.inserted_samples_for_deceleration += num_samples;
	}

	void StatisticsCalculator::AcceleratedSamples(size_t num_samples) {
	accelerate_samples_ += num_samples;
	operations_and_state_.accelerate_samples += num_samples;
	lifetime_stats_.removed_samples_for_acceleration += num_samples;
	}

	void StatisticsCalculator::PacketsDiscarded(size_t num_packets) {
	operations_and_state_.discarded_primary_packets += num_packets;
	}

	void StatisticsCalculator::SecondaryPacketsDiscarded(size_t num_packets) {
	discarded_secondary_packets_ += num_packets;
	lifetime_stats_.fec_packets_discarded += num_packets;
	}

	void StatisticsCalculator::SecondaryPacketsReceived(size_t num_packets) {
	lifetime_stats_.fec_packets_received += num_packets;
	}

	void StatisticsCalculator::IncreaseCounter(size_t num_samples, int fs_hz) {
	const int time_step_ms =
	rtc::CheckedDivExact(static_cast<int>(1000 * num_samples), fs_hz);
	delayed_packet_outage_counter_.AdvanceClock(time_step_ms);
	excess_buffer_delay_.AdvanceClock(time_step_ms);
	buffer_full_counter_.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)) {
	timestamps_since_last_report_ = 0;
	}
	lifetime_stats_.total_samples_received += num_samples;
	}

	void StatisticsCalculator::JitterBufferDelay(size_t num_samples,
	uint64_t waiting_time_ms,
	uint64_t target_delay_ms) {
	lifetime_stats_.jitter_buffer_delay_ms += waiting_time_ms * num_samples;
	lifetime_stats_.jitter_buffer_target_delay_ms +=
	target_delay_ms * num_samples;
	lifetime_stats_.jitter_buffer_emitted_count += num_samples;
	}

	void StatisticsCalculator::SecondaryDecodedSamples(int num_samples) {
	secondary_decoded_samples_ += num_samples;
	}

	void StatisticsCalculator::FlushedPacketBuffer() {
	operations_and_state_.packet_buffer_flushes++;
	buffer_full_counter_.RegisterSample();
	}

	void StatisticsCalculator::ReceivedPacket() {
	++lifetime_stats_.jitter_buffer_packets_received;
	}

	void StatisticsCalculator::RelativePacketArrivalDelay(size_t delay_ms) {
	lifetime_stats_.relative_packet_arrival_delay_ms += delay_ms;
	}

	void StatisticsCalculator::LogDelayedPacketOutageEvent(int num_samples,
	int fs_hz) {
	int outage_duration_ms = num_samples / (fs_hz / 1000);
	RTC_HISTOGRAM_COUNTS("WebRTC.Audio.DelayedPacketOutageEventMs",
	outage_duration_ms, 1 /* min /, 2000 / max */,
	100 /* bucket count */);
	delayed_packet_outage_counter_.RegisterSample();
	lifetime_stats_.delayed_packet_outage_samples += num_samples;
	}

	void StatisticsCalculator::StoreWaitingTime(int waiting_time_ms) {
	excess_buffer_delay_.RegisterSample(waiting_time_ms);
	RTC_DCHECK_LE(waiting_times_.size(), kLenWaitingTimes);
	if (waiting_times_.size() == kLenWaitingTimes) {
	// Erase first value.
	waiting_times_.pop_front();
	}
	waiting_times_.push_back(waiting_time_ms);
	operations_and_state_.last_waiting_time_ms = waiting_time_ms;
	}

	void StatisticsCalculator::GetNetworkStatistics(size_t samples_per_packet,
	NetEqNetworkStatistics* stats) {
	RTC_DCHECK(stats);

	stats->accelerate_rate =
	CalculateQ14Ratio(accelerate_samples_, timestamps_since_last_report_);

	stats->preemptive_rate =
	CalculateQ14Ratio(preemptive_samples_, timestamps_since_last_report_);

	stats->expand_rate =
	CalculateQ14Ratio(expanded_speech_samples_ + expanded_noise_samples_,
	timestamps_since_last_report_);

	stats->speech_expand_rate = CalculateQ14Ratio(expanded_speech_samples_,
	timestamps_since_last_report_);

	stats->secondary_decoded_rate = CalculateQ14Ratio(
	secondary_decoded_samples_, timestamps_since_last_report_);

	const size_t discarded_secondary_samples =
	discarded_secondary_packets_ * samples_per_packet;
	stats->secondary_discarded_rate =
	CalculateQ14Ratio(discarded_secondary_samples,
	static_cast<uint32_t>(discarded_secondary_samples +
	secondary_decoded_samples_));

	if (waiting_times_.size() == 0) {
	stats->mean_waiting_time_ms = -1;
	stats->median_waiting_time_ms = -1;
	stats->min_waiting_time_ms = -1;
	stats->max_waiting_time_ms = -1;
	} else {
	std::sort(waiting_times_.begin(), waiting_times_.end());
	// Find mid-point elements. If the size is odd, the two values
	// `middle_left` and `middle_right` will both be the one middle element; if
	// the size is even, they will be the the two neighboring elements at the
	// middle of the list.
	const int middle_left = waiting_times_[(waiting_times_.size() - 1) / 2];
	const int middle_right = waiting_times_[waiting_times_.size() / 2];
	// Calculate the average of the two. (Works also for odd sizes.)
	stats->median_waiting_time_ms = (middle_left + middle_right) / 2;
	stats->min_waiting_time_ms = waiting_times_.front();
	stats->max_waiting_time_ms = waiting_times_.back();
	double sum = 0;
	for (auto time : waiting_times_) {
	sum += time;
	}
	stats->mean_waiting_time_ms = static_cast<int>(sum / waiting_times_.size());
	}

	// Reset counters.
	ResetMcu();
	Reset();
	}

	NetEqLifetimeStatistics StatisticsCalculator::GetLifetimeStatistics() const {
	return lifetime_stats_;
	}

	NetEqOperationsAndState StatisticsCalculator::GetOperationsAndState() const {
	return operations_and_state_;
	}

	uint16_t StatisticsCalculator::CalculateQ14Ratio(size_t numerator,
	uint32_t denominator) {
	if (numerator == 0) {
	return 0;
	} else if (numerator < denominator) {
	// Ratio must be smaller than 1 in Q14.
	RTC_DCHECK_LT((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