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/*
* 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 <vector>
#include "absl/types/optional.h"
#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_;
absl::optional<int64_t> start_timestamp_us_;
std::vector<int64_t> timestamps_us_;
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_
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