c20978e581
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL. NOTRY=True NOTREECHECKS=True TBR=kwiberg@webrtc.org, kjellander@webrtc.org Bug: webrtc:7634 Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36 Reviewed-on: https://chromium-review.googlesource.com/562156 Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#18919}
76 lines
2.4 KiB
C++
76 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 "webrtc/modules/audio_processing/test/performance_timer.h"
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#include <math.h>
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#include <numeric>
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#include "webrtc/rtc_base/checks.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_(webrtc::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_ = rtc::Optional<int64_t>(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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