ddaa5fa96e
Done using
tools/clang/scripts/build_clang_tools_extra.py \
--fetch out/Default clang-tidy clang-apply-replacements
ninja -C out/Default
gn gen out/Default --export-compile-commands
cd out/Default
tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \
-clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \
-clang-apply-replacements-binary \
out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \
-checks='-*,modernize-deprecated-headers' \
-fix modules/
followed by
git cl format
and a round of IWYU followed by more git cl format
Bug: webrtc:424706384
Change-Id: I9277f055c78a16117a5c88a8b77bbb587f0fba82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/401100
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#45187}
78 lines
2.4 KiB
C++
78 lines
2.4 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/test/performance_timer.h"
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#include <cmath>
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#include <cstddef>
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#include <cstdint>
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#include <numeric>
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#include "rtc_base/checks.h"
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#include "system_wrappers/include/clock.h"
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namespace webrtc {
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namespace test {
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PerformanceTimer::PerformanceTimer(int num_frames_to_process)
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: clock_(Clock::GetRealTimeClock()) {
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timestamps_us_.reserve(num_frames_to_process);
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}
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PerformanceTimer::~PerformanceTimer() = default;
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void PerformanceTimer::StartTimer() {
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start_timestamp_us_ = clock_->TimeInMicroseconds();
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}
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void PerformanceTimer::StopTimer() {
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RTC_DCHECK(start_timestamp_us_);
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timestamps_us_.push_back(clock_->TimeInMicroseconds() - *start_timestamp_us_);
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}
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double PerformanceTimer::GetDurationAverage() const {
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return GetDurationAverage(0);
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}
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double PerformanceTimer::GetDurationStandardDeviation() const {
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return GetDurationStandardDeviation(0);
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}
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double PerformanceTimer::GetDurationAverage(
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size_t number_of_warmup_samples) const {
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RTC_DCHECK_GT(timestamps_us_.size(), number_of_warmup_samples);
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const size_t number_of_samples =
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timestamps_us_.size() - number_of_warmup_samples;
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return static_cast<double>(
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std::accumulate(timestamps_us_.begin() + number_of_warmup_samples,
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timestamps_us_.end(), static_cast<int64_t>(0))) /
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number_of_samples;
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}
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double PerformanceTimer::GetDurationStandardDeviation(
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size_t number_of_warmup_samples) const {
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RTC_DCHECK_GT(timestamps_us_.size(), number_of_warmup_samples);
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const size_t number_of_samples =
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timestamps_us_.size() - number_of_warmup_samples;
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RTC_DCHECK_GT(number_of_samples, 0);
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double average_duration = GetDurationAverage(number_of_warmup_samples);
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double variance = std::accumulate(
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timestamps_us_.begin() + number_of_warmup_samples, timestamps_us_.end(),
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0.0, [average_duration](const double& a, const int64_t& b) {
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return a + (b - average_duration) * (b - average_duration);
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});
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return sqrt(variance / number_of_samples);
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}
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} // namespace test
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} // namespace webrtc
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