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/*
* Copyright (c) 2017 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_processing/aec3/render_delay_controller_metrics.h"
#include <algorithm>
#include "modules/audio_processing/aec3/aec3_common.h"
#include "rtc_base/checks.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace {
enum class DelayReliabilityCategory {
kNone,
kPoor,
kMedium,
kGood,
kExcellent,
kNumCategories
};
enum class DelayChangesCategory {
kNone,
kFew,
kSeveral,
kMany,
kConstant,
kNumCategories
};
} // namespace
RenderDelayControllerMetrics::RenderDelayControllerMetrics() = default;
void RenderDelayControllerMetrics::Update(
std::optional<size_t> delay_samples,
std::optional<size_t> buffer_delay_blocks,
ClockdriftDetector::Level clockdrift) {
++call_counter_;
if (!initial_update) {
size_t delay_blocks;
if (delay_samples) {
++reliable_delay_estimate_counter_;
// Add an offset by 1 (metric is halved before reporting) to reserve 0 for
// absent delay.
delay_blocks = (*delay_samples) / kBlockSize + 2;
} else {
delay_blocks = 0;
}
if (delay_blocks != delay_blocks_) {
++delay_change_counter_;
delay_blocks_ = delay_blocks;
}
} else if (++initial_call_counter_ == 5 * kNumBlocksPerSecond) {
initial_update = false;
}
if (call_counter_ == kMetricsReportingIntervalBlocks) {
int value_to_report = static_cast<int>(delay_blocks_);
// Divide by 2 to compress metric range.
value_to_report = std::min(124, value_to_report >> 1);
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.EchoPathDelay",
value_to_report, 0, 124, 125);
// Divide by 2 to compress metric range.
// Offset by 1 to reserve 0 for absent delay.
value_to_report = buffer_delay_blocks ? (*buffer_delay_blocks + 2) >> 1 : 0;
value_to_report = std::min(124, value_to_report);
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.BufferDelay",
value_to_report, 0, 124, 125);
DelayReliabilityCategory delay_reliability;
if (reliable_delay_estimate_counter_ == 0) {
delay_reliability = DelayReliabilityCategory::kNone;
} else if (reliable_delay_estimate_counter_ > (call_counter_ >> 1)) {
delay_reliability = DelayReliabilityCategory::kExcellent;
} else if (reliable_delay_estimate_counter_ > 100) {
delay_reliability = DelayReliabilityCategory::kGood;
} else if (reliable_delay_estimate_counter_ > 10) {
delay_reliability = DelayReliabilityCategory::kMedium;
} else {
delay_reliability = DelayReliabilityCategory::kPoor;
}
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.Audio.EchoCanceller.ReliableDelayEstimates",
static_cast<int>(delay_reliability),
static_cast<int>(DelayReliabilityCategory::kNumCategories));
DelayChangesCategory delay_changes;
if (delay_change_counter_ == 0) {
delay_changes = DelayChangesCategory::kNone;
} else if (delay_change_counter_ > 10) {
delay_changes = DelayChangesCategory::kConstant;
} else if (delay_change_counter_ > 5) {
delay_changes = DelayChangesCategory::kMany;
} else if (delay_change_counter_ > 2) {
delay_changes = DelayChangesCategory::kSeveral;
} else {
delay_changes = DelayChangesCategory::kFew;
}
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.Audio.EchoCanceller.DelayChanges",
static_cast<int>(delay_changes),
static_cast<int>(DelayChangesCategory::kNumCategories));
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.Audio.EchoCanceller.Clockdrift", static_cast<int>(clockdrift),
static_cast<int>(ClockdriftDetector::Level::kNumCategories));
call_counter_ = 0;
ResetMetrics();
}
}
void RenderDelayControllerMetrics::ResetMetrics() {
delay_change_counter_ = 0;
reliable_delay_estimate_counter_ = 0;
}
} // namespace webrtc
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