mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 06:10:40 +01:00
92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
75 lines
2.4 KiB
C++
75 lines
2.4 KiB
C++
/*
|
|
* 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 "modules/audio_processing/test/performance_timer.h"
|
|
|
|
#include <math.h>
|
|
|
|
#include <numeric>
|
|
|
|
#include "rtc_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 {
|
|
return GetDurationAverage(0);
|
|
}
|
|
|
|
double PerformanceTimer::GetDurationStandardDeviation() const {
|
|
return GetDurationStandardDeviation(0);
|
|
}
|
|
|
|
double PerformanceTimer::GetDurationAverage(
|
|
size_t number_of_warmup_samples) const {
|
|
RTC_DCHECK_GT(timestamps_us_.size(), number_of_warmup_samples);
|
|
const size_t number_of_samples =
|
|
timestamps_us_.size() - number_of_warmup_samples;
|
|
return static_cast<double>(
|
|
std::accumulate(timestamps_us_.begin() + number_of_warmup_samples,
|
|
timestamps_us_.end(), static_cast<int64_t>(0))) /
|
|
number_of_samples;
|
|
}
|
|
|
|
double PerformanceTimer::GetDurationStandardDeviation(
|
|
size_t number_of_warmup_samples) const {
|
|
RTC_DCHECK_GT(timestamps_us_.size(), number_of_warmup_samples);
|
|
const size_t number_of_samples =
|
|
timestamps_us_.size() - number_of_warmup_samples;
|
|
RTC_DCHECK_GT(number_of_samples, 0);
|
|
double average_duration = GetDurationAverage(number_of_warmup_samples);
|
|
|
|
double variance = std::accumulate(
|
|
timestamps_us_.begin() + number_of_warmup_samples, 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 / number_of_samples);
|
|
}
|
|
|
|
} // namespace test
|
|
} // namespace webrtc
|