blob: d16a11bc63f3edd3c4176943ed6dbe558a87068d [file] [log] [blame]
/*
* 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 <assert.h>
#include <string.h> // memset
#include <algorithm>
#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/modules/audio_coding/neteq/decision_logic.h"
#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
#include "webrtc/system_wrappers/include/metrics.h"
namespace webrtc {
// 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),
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) {
}
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;
waiting_times_.clear();
}
void StatisticsCalculator::ResetMcu() {
discarded_packets_ = 0;
lost_timestamps_ = 0;
timestamps_since_last_report_ = 0;
}
void StatisticsCalculator::ExpandedVoiceSamples(size_t num_samples) {
expanded_speech_samples_ += num_samples;
}
void StatisticsCalculator::ExpandedNoiseSamples(size_t num_samples) {
expanded_noise_samples_ += num_samples;
}
void StatisticsCalculator::PreemptiveExpandedSamples(size_t num_samples) {
preemptive_samples_ += num_samples;
}
void StatisticsCalculator::AcceleratedSamples(size_t num_samples) {
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::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);
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_.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);
}
void StatisticsCalculator::GetNetworkStatistics(
int fs_hz,
size_t num_samples_in_buffers,
size_t 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<uint16_t>(num_samples_in_buffers * 1000 / fs_hz);
const int ms_per_packet = rtc::checked_cast<int>(
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 size_t 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_);
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();
}
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