webrtc/modules/audio_coding/neteq/statistics_calculator.cc
Ivo Creusen dc6d5533e1 Add more NetEq information to NetEqState.
Some important NetEq information was not available in NetEqState, which
meant it was not available on the API. This CL adds additional
information.

Bug: webrtc:9667
Change-Id: I702707c7d60472f488047d48fb286f839c5608dc
Reviewed-on: https://webrtc-review.googlesource.com/c/102300
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24985}
2018-10-04 11:50:29 +00:00

373 lines
13 KiB
C++

/*
* 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) {}
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);
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++;
}
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