webrtc/rtc_base/rate_statistics.cc
Yun Zhang 4774a9fcb8 Fix rate statistic when time window running out of samples
Current rate statistic tracker has assumption, the tracking window will
always be full after first filled up. This assumption looks not always
true. One example is the input_framerate_ tracker inside
video_stream_sender.cc which is used for setup frame droper and encoder.
Whenever there is a gap in video stream, like mute/unmute,
pacer pause/unpause etc. The fps detected from the rate_statistics
becomes samples_filled_partial_window / full_window_size, which could
be extremely low for a while. This creates a misalignment between the
fps we told encoder/frame dropper, and the real fps we fed into them,
which causes short-term serious overshot and very bad experience on
delay, avsync, congestion etc. This may also depends on how fast
encoder could react to the gap between set fps and real fps, but
libvpx and openh264 at least cannot handle this well.

So propose a fix to update first timestamp after tracker window
drained. This will give more accurate fps estimate similar based on
active window after sample gets drained

Bug: webrtc:13403
Change-Id: I96792c11091fe8bfa63e669f4360a3b3e95593e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237720
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35447}
2021-11-30 23:57:40 +00:00

157 lines
5.4 KiB
C++

/*
* 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 <limits>
#include <memory>
#include "rtc_base/checks.h"
#include "rtc_base/logging.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);
EraseOld(now_ms);
if (first_timestamp_ == -1 || num_samples_ == 0) {
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_;
}
absl::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 &&
rtc::SafeLt(active_window_size, current_window_size_ms_)) ||
overflow_) {
return absl::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 absl::nullopt;
}
return rtc::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