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0b489303d2
Bug: webrtc:12338 Change-Id: I85bff694dd2ead83c939c4d1945eff82e1296001 No-Presubmit: True Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227161 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34690}
47 lines
1.4 KiB
C++
47 lines
1.4 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_
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#define MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_
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#include <vector>
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#include "absl/types/optional.h"
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#include "system_wrappers/include/clock.h"
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namespace webrtc {
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namespace test {
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class PerformanceTimer {
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public:
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explicit PerformanceTimer(int num_frames_to_process);
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~PerformanceTimer();
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void StartTimer();
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void StopTimer();
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double GetDurationAverage() const;
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double GetDurationStandardDeviation() const;
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// These methods are the same as those above, but they ignore the first
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// `number_of_warmup_samples` measurements.
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double GetDurationAverage(size_t number_of_warmup_samples) const;
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double GetDurationStandardDeviation(size_t number_of_warmup_samples) const;
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private:
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webrtc::Clock* clock_;
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absl::optional<int64_t> start_timestamp_us_;
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std::vector<int64_t> timestamps_us_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_
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