webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc
Danil Chapovalov db9f7ab9f9 Replace rtc::Optional with absl::optional in modules/audio processing
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameter 'modules/audio_processing'

find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: Id29f8de59dba704787c2c38a3d05c60827c181b0
Reviewed-on: https://webrtc-review.googlesource.com/83982
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23653}
2018-06-19 10:38:56 +00:00

139 lines
4.3 KiB
C++

/*
* 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 "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
};
constexpr int kMaxSkewShiftCount = 20;
} // namespace
RenderDelayControllerMetrics::RenderDelayControllerMetrics() = default;
void RenderDelayControllerMetrics::Update(
absl::optional<size_t> delay_samples,
size_t buffer_delay_blocks,
absl::optional<int> skew_shift_blocks) {
++call_counter_;
if (!initial_update) {
size_t delay_blocks;
if (delay_samples) {
++reliable_delay_estimate_counter_;
delay_blocks = (*delay_samples) / kBlockSize + 2;
} else {
delay_blocks = 0;
}
if (delay_blocks != delay_blocks_) {
++delay_change_counter_;
delay_blocks_ = delay_blocks;
}
if (skew_shift_blocks) {
skew_shift_count_ = std::min(kMaxSkewShiftCount, skew_shift_count_);
}
} else if (++initial_call_counter_ == 5 * kNumBlocksPerSecond) {
initial_update = false;
}
if (call_counter_ == kMetricsReportingIntervalBlocks) {
int value_to_report = static_cast<int>(delay_blocks_);
value_to_report = std::min(124, value_to_report >> 1);
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.EchoPathDelay",
value_to_report, 0, 124, 125);
value_to_report = static_cast<int>(buffer_delay_blocks + 2);
value_to_report = std::min(124, value_to_report >> 1);
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));
metrics_reported_ = true;
call_counter_ = 0;
ResetMetrics();
} else {
metrics_reported_ = false;
}
if (!initial_update && ++skew_report_timer_ == 60 * kNumBlocksPerSecond) {
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.MaxSkewShiftCount",
skew_shift_count_, 0, kMaxSkewShiftCount,
kMaxSkewShiftCount + 1);
skew_shift_count_ = 0;
skew_report_timer_ = 0;
}
}
void RenderDelayControllerMetrics::ResetMetrics() {
delay_change_counter_ = 0;
reliable_delay_estimate_counter_ = 0;
}
} // namespace webrtc