1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168
|
/*
* 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.
*/
#include "rtc_base/rate_statistics.h"
#include <algorithm>
#include <cstdint>
#include <limits>
#include <optional>
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_compare.h"
#include "rtc_base/numerics/safe_conversions.h"
namespace webrtc {
RateStatistics::Bucket::Bucket(int64_t timestamp)
: sum(0), num_samples(0), timestamp(timestamp) {}
RateStatistics::RateStatistics(int64_t window_size_ms, float scale)
: accumulated_count_(0),
first_timestamp_(-1),
num_samples_(0),
scale_(scale),
max_window_size_ms_(window_size_ms),
current_window_size_ms_(max_window_size_ms_) {}
RateStatistics::RateStatistics(const RateStatistics& other)
: buckets_(other.buckets_),
accumulated_count_(other.accumulated_count_),
first_timestamp_(other.first_timestamp_),
overflow_(other.overflow_),
num_samples_(other.num_samples_),
scale_(other.scale_),
max_window_size_ms_(other.max_window_size_ms_),
current_window_size_ms_(other.current_window_size_ms_) {}
RateStatistics::RateStatistics(RateStatistics&& other) = default;
RateStatistics::~RateStatistics() {}
void RateStatistics::Reset() {
accumulated_count_ = 0;
overflow_ = false;
num_samples_ = 0;
first_timestamp_ = -1;
current_window_size_ms_ = max_window_size_ms_;
buckets_.clear();
}
void RateStatistics::Update(int64_t count, int64_t now_ms) {
RTC_DCHECK_GE(count, 0);
// Don't reset `first_timestamp_` if the last sample removed by EraseOld() was
// recent. This ensures that the window maintains its intended duration even
// when samples are received near the boundary. Use a margin of 50% of the
// current window size.
const int64_t recent_sample_time_margin = 1.5 * current_window_size_ms_;
bool last_sample_is_recent =
!buckets_.empty() &&
buckets_.back().timestamp > now_ms - recent_sample_time_margin;
EraseOld(now_ms);
if (first_timestamp_ == -1 || (num_samples_ == 0 && !last_sample_is_recent)) {
first_timestamp_ = now_ms;
}
if (buckets_.empty() || now_ms != buckets_.back().timestamp) {
if (!buckets_.empty() && now_ms < buckets_.back().timestamp) {
RTC_LOG(LS_WARNING) << "Timestamp " << now_ms
<< " is before the last added "
"timestamp in the rate window: "
<< buckets_.back().timestamp << ", aligning to that.";
now_ms = buckets_.back().timestamp;
}
buckets_.emplace_back(now_ms);
}
Bucket& last_bucket = buckets_.back();
last_bucket.sum += count;
++last_bucket.num_samples;
if (std::numeric_limits<int64_t>::max() - accumulated_count_ > count) {
accumulated_count_ += count;
} else {
overflow_ = true;
}
++num_samples_;
}
std::optional<int64_t> RateStatistics::Rate(int64_t now_ms) const {
// Yeah, this const_cast ain't pretty, but the alternative is to declare most
// of the members as mutable...
const_cast<RateStatistics*>(this)->EraseOld(now_ms);
int active_window_size = 0;
if (first_timestamp_ != -1) {
if (first_timestamp_ <= now_ms - current_window_size_ms_) {
// Count window as full even if no data points currently in view, if the
// data stream started before the window.
active_window_size = current_window_size_ms_;
} else {
// Size of a single bucket is 1ms, so even if now_ms == first_timestmap_
// the window size should be 1.
active_window_size = now_ms - first_timestamp_ + 1;
}
}
// If window is a single bucket or there is only one sample in a data set that
// has not grown to the full window size, or if the accumulator has
// overflowed, treat this as rate unavailable.
if (num_samples_ == 0 || active_window_size <= 1 ||
(num_samples_ <= 1 &&
SafeLt(active_window_size, current_window_size_ms_)) ||
overflow_) {
return std::nullopt;
}
float scale = static_cast<float>(scale_) / active_window_size;
float result = accumulated_count_ * scale + 0.5f;
// Better return unavailable rate than garbage value (undefined behavior).
if (result > static_cast<float>(std::numeric_limits<int64_t>::max())) {
return std::nullopt;
}
return dchecked_cast<int64_t>(result);
}
void RateStatistics::EraseOld(int64_t now_ms) {
// New oldest time that is included in data set.
const int64_t new_oldest_time = now_ms - current_window_size_ms_ + 1;
// Loop over buckets and remove too old data points.
while (!buckets_.empty() && buckets_.front().timestamp < new_oldest_time) {
const Bucket& oldest_bucket = buckets_.front();
RTC_DCHECK_GE(accumulated_count_, oldest_bucket.sum);
RTC_DCHECK_GE(num_samples_, oldest_bucket.num_samples);
accumulated_count_ -= oldest_bucket.sum;
num_samples_ -= oldest_bucket.num_samples;
buckets_.pop_front();
// This does not clear overflow_ even when counter is empty.
// TODO(https://bugs.webrtc.org/11247): Consider if overflow_ can be reset.
}
}
bool RateStatistics::SetWindowSize(int64_t window_size_ms, int64_t now_ms) {
if (window_size_ms <= 0 || window_size_ms > max_window_size_ms_)
return false;
if (first_timestamp_ != -1) {
// If the window changes (e.g. decreases - removing data point, then
// increases again) we need to update the first timestamp mark as
// otherwise it indicates the window coveres a region of zeros, suddenly
// under-estimating the rate.
first_timestamp_ = std::max(first_timestamp_, now_ms - window_size_ms + 1);
}
current_window_size_ms_ = window_size_ms;
EraseOld(now_ms);
return true;
}
} // namespace webrtc
|