e10dfc9fa9
using find modules/audio_processing/ -name "*.h" -o -name "*.cc" | grep -v "aecm_core_neon.cc" | grep -v aecm_core_mips.cc | xargs tools_webrtc/iwyu/apply-include-cleaner followed by tools_webrtc/gn_check_autofix.py -C out/Default/ and git cl format Manual changes: - changes to AECM caused linker errors. Reverted aecm/ - manual modifications in PS 9..12: https://webrtc-review.googlesource.com/c/src/+/387680/9..12 - rebase and follow-up in https://webrtc-review.googlesource.com/c/src/+/387680/14..18 - rebase and follow-up in https://webrtc-review.googlesource.com/c/src/+/387680/19..24 - moved generated debug.pb.h to protobuf_utils.h and exported - agc2 target visibility changes - stringstream TODOs Bug: webrtc:42226242 Change-Id: I9b87208b90f3a7e8f179c1038e6b4dabbcbbbf3c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/387680 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#44809}
50 lines
1.4 KiB
C++
50 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 <cstddef>
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#include <cstdint>
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#include <optional>
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#include <vector>
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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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std::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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