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/*
* 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 "absl/strings/string_view.h"
#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(
absl::string_view 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(
absl::string_view 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(
absl::string_view 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::GeneratedNoiseSamples(size_t num_samples) {
lifetime_stats_.generated_noise_samples += num_samples;
}
void StatisticsCalculator::PacketsDiscarded(size_t num_packets) {
lifetime_stats_.packets_discarded += 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,
uint64_t unlimited_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_minimum_delay_ms +=
unlimited_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