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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/rtc_base/rate_statistics.h | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/rtc_base/rate_statistics.h')
-rw-r--r-- | third_party/libwebrtc/rtc_base/rate_statistics.h | 105 |
1 files changed, 105 insertions, 0 deletions
diff --git a/third_party/libwebrtc/rtc_base/rate_statistics.h b/third_party/libwebrtc/rtc_base/rate_statistics.h new file mode 100644 index 0000000000..e7ce8ad5ed --- /dev/null +++ b/third_party/libwebrtc/rtc_base/rate_statistics.h @@ -0,0 +1,105 @@ +/* + * Copyright (c) 2013 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 RTC_BASE_RATE_STATISTICS_H_ +#define RTC_BASE_RATE_STATISTICS_H_ + +#include <stddef.h> +#include <stdint.h> + +#include <deque> +#include <memory> + +#include "absl/types/optional.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +// Class to estimate rates based on counts in a sequence of 1-millisecond +// intervals. + +// This class uses int64 for all its numbers because some rates can be very +// high; for instance, a 20 Mbit/sec video stream can wrap a 32-bit byte +// counter in 14 minutes. + +// Note that timestamps used in Update(), Rate() and SetWindowSize() must never +// decrease for two consecutive calls. +// TODO(bugs.webrtc.org/11600): Migrate from int64_t to Timestamp. + +class RTC_EXPORT RateStatistics { + public: + static constexpr float kBpsScale = 8000.0f; + + // max_window_size_ms = Maximum window size in ms for the rate estimation. + // Initial window size is set to this, but may be changed + // to something lower by calling SetWindowSize(). + // scale = coefficient to convert counts/ms to desired unit + // ex: kBpsScale (8000) for bits/s if count represents bytes. + RateStatistics(int64_t max_window_size_ms, float scale); + + RateStatistics(const RateStatistics& other); + + RateStatistics(RateStatistics&& other); + + ~RateStatistics(); + + // Reset instance to original state. + void Reset(); + + // Update rate with a new data point, moving averaging window as needed. + void Update(int64_t count, int64_t now_ms); + + // Note that despite this being a const method, it still updates the internal + // state (moves averaging window), but it doesn't make any alterations that + // are observable from the other methods, as long as supplied timestamps are + // from a monotonic clock. Ie, it doesn't matter if this call moves the + // window, since any subsequent call to Update or Rate would still have moved + // the window as much or more. + absl::optional<int64_t> Rate(int64_t now_ms) const; + + // Update the size of the averaging window. The maximum allowed value for + // window_size_ms is max_window_size_ms as supplied in the constructor. + bool SetWindowSize(int64_t window_size_ms, int64_t now_ms); + + private: + void EraseOld(int64_t now_ms); + + struct Bucket { + explicit Bucket(int64_t timestamp); + int64_t sum; // Sum of all samples in this bucket. + int num_samples; // Number of samples in this bucket. + const int64_t timestamp; // Timestamp this bucket corresponds to. + }; + // All buckets within the time window, ordered by time. + std::deque<Bucket> buckets_; + + // Total count recorded in all buckets. + int64_t accumulated_count_; + + // Timestamp of the first data point seen, or -1 of none seen. + int64_t first_timestamp_; + + // True if accumulated_count_ has ever grown too large to be + // contained in its integer type. + bool overflow_ = false; + + // The total number of samples in the buckets. + int num_samples_; + + // To convert counts/ms to desired units + const float scale_; + + // The window sizes, in ms, over which the rate is calculated. + const int64_t max_window_size_ms_; + int64_t current_window_size_ms_; +}; +} // namespace webrtc + +#endif // RTC_BASE_RATE_STATISTICS_H_ |