| /* |
| * Copyright (c) 2016 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 "webrtc/modules/audio_processing/test/performance_timer.h" |
| |
| #include <math.h> |
| |
| #include <numeric> |
| |
| #include "webrtc/base/checks.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| PerformanceTimer::PerformanceTimer(int num_frames_to_process) |
| : clock_(webrtc::Clock::GetRealTimeClock()) { |
| timestamps_us_.reserve(num_frames_to_process); |
| } |
| |
| PerformanceTimer::~PerformanceTimer() = default; |
| |
| void PerformanceTimer::StartTimer() { |
| start_timestamp_us_ = rtc::Optional<int64_t>(clock_->TimeInMicroseconds()); |
| } |
| |
| void PerformanceTimer::StopTimer() { |
| RTC_DCHECK(start_timestamp_us_); |
| timestamps_us_.push_back(clock_->TimeInMicroseconds() - *start_timestamp_us_); |
| } |
| |
| double PerformanceTimer::GetDurationAverage() const { |
| RTC_DCHECK(!timestamps_us_.empty()); |
| return static_cast<double>( |
| std::accumulate(timestamps_us_.begin(), timestamps_us_.end(), 0)) / |
| timestamps_us_.size(); |
| } |
| |
| double PerformanceTimer::GetDurationStandardDeviation() const { |
| RTC_DCHECK(!timestamps_us_.empty()); |
| double average_duration = GetDurationAverage(); |
| |
| double variance = std::accumulate( |
| timestamps_us_.begin(), timestamps_us_.end(), 0.0, |
| [average_duration](const double& a, const int64_t& b) { |
| return a + (b - average_duration) * (b - average_duration); |
| }); |
| |
| return sqrt(variance / timestamps_us_.size()); |
| } |
| |
| } // namespace test |
| } // namespace webrtc |