kjellander c8fa692ec4 Update includes for webrtc/{base => rtc_base} rename (1/3)
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`

The only manual edit is to add an include of webrtc/rtc_base/checks.h in
webrtc/modules/audio_device/android/opensles_common.h, which likely
was needed due to changed include paths due to 'git cl format'.

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2969653002
Cr-Commit-Position: refs/heads/master@{#18871}
2017-06-30 21:02:00 +00:00

76 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 "webrtc/modules/audio_processing/test/performance_timer.h"
#include <math.h>
#include <numeric>
#include "webrtc/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