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This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
91 lines
2.9 KiB
C++
91 lines
2.9 KiB
C++
/*
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* Copyright (c) 2013 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 RTC_BASE_RATE_STATISTICS_H_
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#define RTC_BASE_RATE_STATISTICS_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <memory>
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#include "absl/types/optional.h"
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namespace webrtc {
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class RateStatistics {
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public:
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static constexpr float kBpsScale = 8000.0f;
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// max_window_size_ms = Maximum window size in ms for the rate estimation.
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// Initial window size is set to this, but may be changed
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// to something lower by calling SetWindowSize().
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// scale = coefficient to convert counts/ms to desired unit
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// ex: kBpsScale (8000) for bits/s if count represents bytes.
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RateStatistics(int64_t max_window_size_ms, float scale);
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RateStatistics(const RateStatistics& other);
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RateStatistics(RateStatistics&& other);
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~RateStatistics();
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// Reset instance to original state.
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void Reset();
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// Update rate with a new data point, moving averaging window as needed.
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void Update(size_t count, int64_t now_ms);
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// Note that despite this being a const method, it still updates the internal
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// state (moves averaging window), but it doesn't make any alterations that
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// are observable from the other methods, as long as supplied timestamps are
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// from a monotonic clock. Ie, it doesn't matter if this call moves the
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// window, since any subsequent call to Update or Rate would still have moved
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// the window as much or more.
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absl::optional<uint32_t> Rate(int64_t now_ms) const;
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// Update the size of the averaging window. The maximum allowed value for
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// window_size_ms is max_window_size_ms as supplied in the constructor.
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bool SetWindowSize(int64_t window_size_ms, int64_t now_ms);
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private:
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void EraseOld(int64_t now_ms);
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bool IsInitialized() const;
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// Counters are kept in buckets (circular buffer), with one bucket
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// per millisecond.
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struct Bucket {
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size_t sum; // Sum of all samples in this bucket.
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size_t samples; // Number of samples in this bucket.
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};
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std::unique_ptr<Bucket[]> buckets_;
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// Total count recorded in buckets.
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size_t accumulated_count_;
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// The total number of samples in the buckets.
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size_t num_samples_;
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// Oldest time recorded in buckets.
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int64_t oldest_time_;
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// Bucket index of oldest counter recorded in buckets.
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uint32_t oldest_index_;
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// To convert counts/ms to desired units
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const float scale_;
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// The window sizes, in ms, over which the rate is calculated.
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const int64_t max_window_size_ms_;
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int64_t current_window_size_ms_;
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};
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} // namespace webrtc
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#endif // RTC_BASE_RATE_STATISTICS_H_
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