webrtc/modules/audio_processing/test/performance_timer.h
Danil Chapovalov db9f7ab9f9 Replace rtc::Optional with absl::optional in modules/audio processing
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameter 'modules/audio_processing'

find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: Id29f8de59dba704787c2c38a3d05c60827c181b0
Reviewed-on: https://webrtc-review.googlesource.com/83982
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23653}
2018-06-19 10:38:56 +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 <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_