diff options
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/test/performance_timer.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/test/performance_timer.cc | 75 |
1 files changed, 75 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/performance_timer.cc b/third_party/libwebrtc/modules/audio_processing/test/performance_timer.cc new file mode 100644 index 0000000000..1a82258903 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/performance_timer.cc @@ -0,0 +1,75 @@ +/* + * 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. + */ + +#include "modules/audio_processing/test/performance_timer.h" + +#include <math.h> + +#include <numeric> + +#include "rtc_base/checks.h" + +namespace webrtc { +namespace test { + +PerformanceTimer::PerformanceTimer(int num_frames_to_process) + : clock_(webrtc::Clock::GetRealTimeClock()) { + timestamps_us_.reserve(num_frames_to_process); +} + +PerformanceTimer::~PerformanceTimer() = default; + +void PerformanceTimer::StartTimer() { + start_timestamp_us_ = clock_->TimeInMicroseconds(); +} + +void PerformanceTimer::StopTimer() { + RTC_DCHECK(start_timestamp_us_); + timestamps_us_.push_back(clock_->TimeInMicroseconds() - *start_timestamp_us_); +} + +double PerformanceTimer::GetDurationAverage() const { + return GetDurationAverage(0); +} + +double PerformanceTimer::GetDurationStandardDeviation() const { + return GetDurationStandardDeviation(0); +} + +double PerformanceTimer::GetDurationAverage( + size_t number_of_warmup_samples) const { + RTC_DCHECK_GT(timestamps_us_.size(), number_of_warmup_samples); + const size_t number_of_samples = + timestamps_us_.size() - number_of_warmup_samples; + return static_cast<double>( + std::accumulate(timestamps_us_.begin() + number_of_warmup_samples, + timestamps_us_.end(), static_cast<int64_t>(0))) / + number_of_samples; +} + +double PerformanceTimer::GetDurationStandardDeviation( + size_t number_of_warmup_samples) const { + RTC_DCHECK_GT(timestamps_us_.size(), number_of_warmup_samples); + const size_t number_of_samples = + timestamps_us_.size() - number_of_warmup_samples; + RTC_DCHECK_GT(number_of_samples, 0); + double average_duration = GetDurationAverage(number_of_warmup_samples); + + double variance = std::accumulate( + timestamps_us_.begin() + number_of_warmup_samples, timestamps_us_.end(), + 0.0, [average_duration](const double& a, const int64_t& b) { + return a + (b - average_duration) * (b - average_duration); + }); + + return sqrt(variance / number_of_samples); +} + +} // namespace test +} // namespace webrtc |