webrtc/modules/audio_processing/test/performance_timer.h
Florent Castelli 8037fc6ffa Migrate absl::optional to std::optional
Bug: webrtc:342905193
No-Try: True
Change-Id: Icc968be43b8830038ea9a1f5f604307220457807
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361021
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42911}
2024-09-02 12:16:47 +00:00

47 lines
1.4 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.
*/
#ifndef MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_
#define MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_
#include <optional>
#include <vector>
#include "system_wrappers/include/clock.h"
namespace webrtc {
namespace test {
class PerformanceTimer {
public:
explicit PerformanceTimer(int num_frames_to_process);
~PerformanceTimer();
void StartTimer();
void StopTimer();
double GetDurationAverage() const;
double GetDurationStandardDeviation() const;
// These methods are the same as those above, but they ignore the first
// `number_of_warmup_samples` measurements.
double GetDurationAverage(size_t number_of_warmup_samples) const;
double GetDurationStandardDeviation(size_t number_of_warmup_samples) const;
private:
webrtc::Clock* clock_;
std::optional<int64_t> start_timestamp_us_;
std::vector<int64_t> timestamps_us_;
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_