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Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/test/performance_timer.h | 47 |
1 files changed, 47 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/performance_timer.h b/third_party/libwebrtc/modules/audio_processing/test/performance_timer.h new file mode 100644 index 0000000000..5375ba74e8 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/performance_timer.h @@ -0,0 +1,47 @@ +/* + * 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_ |