/* * 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 "webrtc/modules/audio_coding/neteq/statistics_calculator.h" #include #include // memset #include "webrtc/base/checks.h" #include "webrtc/modules/audio_coding/neteq/decision_logic.h" #include "webrtc/modules/audio_coding/neteq/delay_manager.h" #include "webrtc/system_wrappers/interface/metrics.h" namespace webrtc { 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_; DCHECK_GE(timer_, 0); } void StatisticsCalculator::PeriodicUmaLogger::LogToUma(int value) const { RTC_HISTOGRAM_COUNTS(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 static_cast(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), len_waiting_times_(0), next_waiting_time_index_(0), secondary_decoded_samples_(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) { memset(waiting_times_, 0, kLenWaitingTimes * sizeof(waiting_times_[0])); } 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; } void StatisticsCalculator::ResetMcu() { discarded_packets_ = 0; lost_timestamps_ = 0; timestamps_since_last_report_ = 0; } void StatisticsCalculator::ResetWaitingTimeStatistics() { memset(waiting_times_, 0, kLenWaitingTimes * sizeof(waiting_times_[0])); len_waiting_times_ = 0; next_waiting_time_index_ = 0; } void StatisticsCalculator::ExpandedVoiceSamples(int num_samples) { expanded_speech_samples_ += num_samples; } void StatisticsCalculator::ExpandedNoiseSamples(int num_samples) { expanded_noise_samples_ += num_samples; } void StatisticsCalculator::PreemptiveExpandedSamples(int num_samples) { preemptive_samples_ += num_samples; } void StatisticsCalculator::AcceleratedSamples(int num_samples) { accelerate_samples_ += num_samples; } void StatisticsCalculator::AddZeros(int num_samples) { added_zero_samples_ += num_samples; } 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) { 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(num_samples); if (timestamps_since_last_report_ > static_cast(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_.RegisterSample(); } void StatisticsCalculator::StoreWaitingTime(int waiting_time_ms) { excess_buffer_delay_.RegisterSample(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_++; } } void StatisticsCalculator::GetNetworkStatistics( int fs_hz, int num_samples_in_buffers, int samples_per_packet, const DelayManager& delay_manager, const DecisionLogic& decision_logic, NetEqNetworkStatistics *stats) { if (fs_hz <= 0 || !stats) { assert(false); return; } stats->added_zero_samples = added_zero_samples_; stats->current_buffer_size_ms = static_cast(num_samples_in_buffers * 1000 / fs_hz); const int ms_per_packet = decision_logic.packet_length_samples() / (fs_hz / 1000); stats->preferred_buffer_size_ms = (delay_manager.TargetLevel() >> 8) * ms_per_packet; stats->jitter_peaks_found = delay_manager.PeakFound(); stats->clockdrift_ppm = delay_manager.AverageIAT(); stats->packet_loss_rate = CalculateQ14Ratio(lost_timestamps_, timestamps_since_last_report_); const unsigned discarded_samples = discarded_packets_ * samples_per_packet; stats->packet_discard_rate = CalculateQ14Ratio(discarded_samples, 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_); // Reset counters. ResetMcu(); Reset(); } void StatisticsCalculator::WaitingTimes(std::vector* waiting_times) { if (!waiting_times) { return; } waiting_times->assign(waiting_times_, waiting_times_ + len_waiting_times_); ResetWaitingTimeStatistics(); } uint16_t StatisticsCalculator::CalculateQ14Ratio(uint32_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((numerator << 14) / denominator); } else { // Will not produce a ratio larger than 1, since this is probably an error. return 1 << 14; } } } // namespace webrtc