| /* |
| * 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 <assert.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; |
| } |
| } // 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), |
| added_zero_samples_(0), |
| expanded_speech_samples_(0), |
| expanded_noise_samples_(0), |
| discarded_packets_(0), |
| lost_timestamps_(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; |
| added_zero_samples_ = 0; |
| expanded_speech_samples_ = 0; |
| expanded_noise_samples_ = 0; |
| secondary_decoded_samples_ = 0; |
| discarded_secondary_packets_ = 0; |
| waiting_times_.clear(); |
| } |
| |
| void StatisticsCalculator::ResetMcu() { |
| discarded_packets_ = 0; |
| lost_timestamps_ = 0; |
| 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::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) { |
| voice_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 voice_canceled_out = std::min( |
| static_cast<size_t>(num_samples), voice_concealed_samples_correction_); |
| voice_concealed_samples_correction_ -= voice_canceled_out; |
| lifetime_stats_.voice_concealed_samples += num_samples - voice_canceled_out; |
| } |
| } |
| |
| void StatisticsCalculator::PreemptiveExpandedSamples(size_t num_samples) { |
| preemptive_samples_ += num_samples; |
| operations_and_state_.preemptive_samples += num_samples; |
| } |
| |
| void StatisticsCalculator::AcceleratedSamples(size_t num_samples) { |
| accelerate_samples_ += num_samples; |
| operations_and_state_.accelerate_samples += num_samples; |
| } |
| |
| void StatisticsCalculator::AddZeros(size_t num_samples) { |
| added_zero_samples_ += num_samples; |
| } |
| |
| void StatisticsCalculator::PacketsDiscarded(size_t num_packets) { |
| discarded_packets_ += num_packets; |
| } |
| |
| void StatisticsCalculator::SecondaryPacketsDiscarded(size_t num_packets) { |
| discarded_secondary_packets_ += num_packets; |
| } |
| |
| void StatisticsCalculator::LostSamples(size_t num_samples) { |
| lost_timestamps_ += num_samples; |
| } |
| |
| 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)) { |
| lost_timestamps_ = 0; |
| timestamps_since_last_report_ = 0; |
| discarded_packets_ = 0; |
| } |
| lifetime_stats_.total_samples_received += num_samples; |
| } |
| |
| void StatisticsCalculator::JitterBufferDelay(size_t num_samples, |
| uint64_t waiting_time_ms) { |
| lifetime_stats_.jitter_buffer_delay_ms += waiting_time_ms * 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::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_.RegisterSample(); |
| } |
| |
| 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(int fs_hz, |
| size_t num_samples_in_buffers, |
| size_t samples_per_packet, |
| NetEqNetworkStatistics* stats) { |
| RTC_DCHECK_GT(fs_hz, 0); |
| RTC_DCHECK(stats); |
| |
| stats->added_zero_samples = added_zero_samples_; |
| stats->current_buffer_size_ms = |
| static_cast<uint16_t>(num_samples_in_buffers * 1000 / fs_hz); |
| |
| stats->packet_loss_rate = |
| CalculateQ14Ratio(lost_timestamps_, timestamps_since_last_report_); |
| |
| 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(); |
| } |
| |
| void StatisticsCalculator::PopulateDelayManagerStats( |
| int ms_per_packet, |
| const DelayManager& delay_manager, |
| NetEqNetworkStatistics* stats) { |
| RTC_DCHECK(stats); |
| stats->preferred_buffer_size_ms = |
| (delay_manager.TargetLevel() >> 8) * ms_per_packet; |
| stats->jitter_peaks_found = delay_manager.PeakFound(); |
| stats->clockdrift_ppm = |
| rtc::saturated_cast<int32_t>(delay_manager.EstimatedClockDriftPpm()); |
| } |
| |
| 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. |
| 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 |