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In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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 WEBRTC_MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_
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#include <vector>
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#include "webrtc/api/optional.h"
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#include "webrtc/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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rtc::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 // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_
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