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/*
* Copyright 2015 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_tracker.h"
#include <algorithm>
#include <cstddef>
#include <cstdint>
#include "rtc_base/checks.h"
#include "rtc_base/time_utils.h"
namespace webrtc {
static const int64_t kTimeUnset = -1;
RateTracker::RateTracker(int64_t bucket_milliseconds, size_t bucket_count)
: bucket_milliseconds_(bucket_milliseconds),
bucket_count_(bucket_count),
sample_buckets_(new int64_t[bucket_count + 1]),
total_sample_count_(0u),
bucket_start_time_milliseconds_(kTimeUnset) {
RTC_CHECK(bucket_milliseconds > 0);
RTC_CHECK(bucket_count > 0);
}
RateTracker::~RateTracker() {
delete[] sample_buckets_;
}
double RateTracker::ComputeRateForInterval(
int64_t interval_milliseconds) const {
if (bucket_start_time_milliseconds_ == kTimeUnset) {
return 0.0;
}
int64_t current_time = Time();
// Calculate which buckets to sum up given the current time. If the time
// has passed to a new bucket then we have to skip some of the oldest buckets.
int64_t available_interval_milliseconds =
std::min(interval_milliseconds,
bucket_milliseconds_ * static_cast<int64_t>(bucket_count_));
// number of old buckets (i.e. after the current bucket in the ring buffer)
// that are expired given our current time interval.
size_t buckets_to_skip;
// Number of milliseconds of the first bucket that are not a portion of the
// current interval.
int64_t milliseconds_to_skip;
if (current_time >
initialization_time_milliseconds_ + available_interval_milliseconds) {
int64_t time_to_skip =
current_time - bucket_start_time_milliseconds_ +
static_cast<int64_t>(bucket_count_) * bucket_milliseconds_ -
available_interval_milliseconds;
buckets_to_skip = time_to_skip / bucket_milliseconds_;
milliseconds_to_skip = time_to_skip % bucket_milliseconds_;
} else {
buckets_to_skip = bucket_count_ - current_bucket_;
milliseconds_to_skip = 0;
available_interval_milliseconds =
TimeDiff(current_time, initialization_time_milliseconds_);
// Let one bucket interval pass after initialization before reporting.
if (available_interval_milliseconds < bucket_milliseconds_) {
return 0.0;
}
}
// If we're skipping all buckets that means that there have been no samples
// within the sampling interval so report 0.
if (buckets_to_skip > bucket_count_ || available_interval_milliseconds == 0) {
return 0.0;
}
size_t start_bucket = NextBucketIndex(current_bucket_ + buckets_to_skip);
// Only count a portion of the first bucket according to how much of the
// first bucket is within the current interval.
int64_t total_samples = ((sample_buckets_[start_bucket] *
(bucket_milliseconds_ - milliseconds_to_skip)) +
(bucket_milliseconds_ >> 1)) /
bucket_milliseconds_;
// All other buckets in the interval are counted in their entirety.
for (size_t i = NextBucketIndex(start_bucket);
i != NextBucketIndex(current_bucket_); i = NextBucketIndex(i)) {
total_samples += sample_buckets_[i];
}
// Convert to samples per second.
return static_cast<double>(total_samples * 1000) /
static_cast<double>(available_interval_milliseconds);
}
double RateTracker::ComputeTotalRate() const {
if (bucket_start_time_milliseconds_ == kTimeUnset) {
return 0.0;
}
int64_t current_time = Time();
if (current_time <= initialization_time_milliseconds_) {
return 0.0;
}
return static_cast<double>(total_sample_count_ * 1000) /
static_cast<double>(
TimeDiff(current_time, initialization_time_milliseconds_));
}
int64_t RateTracker::TotalSampleCount() const {
return total_sample_count_;
}
void RateTracker::AddSamples(int64_t sample_count) {
AddSamplesAtTime(Time(), sample_count);
}
void RateTracker::AddSamplesAtTime(int64_t current_time_ms,
int64_t sample_count) {
RTC_DCHECK_LE(0, sample_count);
EnsureInitialized();
// Advance the current bucket as needed for the current time, and reset
// bucket counts as we advance.
for (size_t i = 0; i <= bucket_count_ &&
current_time_ms >=
bucket_start_time_milliseconds_ + bucket_milliseconds_;
++i) {
bucket_start_time_milliseconds_ += bucket_milliseconds_;
current_bucket_ = NextBucketIndex(current_bucket_);
sample_buckets_[current_bucket_] = 0;
}
// Ensure that bucket_start_time_milliseconds_ is updated appropriately if
// the entire buffer of samples has been expired.
bucket_start_time_milliseconds_ +=
bucket_milliseconds_ *
((current_time_ms - bucket_start_time_milliseconds_) /
bucket_milliseconds_);
// Add all samples in the bucket that includes the current time.
sample_buckets_[current_bucket_] += sample_count;
total_sample_count_ += sample_count;
}
int64_t RateTracker::Time() const {
return TimeMillis();
}
void RateTracker::EnsureInitialized() {
if (bucket_start_time_milliseconds_ == kTimeUnset) {
initialization_time_milliseconds_ = Time();
bucket_start_time_milliseconds_ = initialization_time_milliseconds_;
current_bucket_ = 0;
// We only need to initialize the first bucket because we reset buckets when
// current_bucket_ increments.
sample_buckets_[current_bucket_] = 0;
}
}
size_t RateTracker::NextBucketIndex(size_t bucket_index) const {
return (bucket_index + 1u) % (bucket_count_ + 1u);
}
} // namespace webrtc
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