chromium/third_party/webrtc/rtc_base/rate_statistics.h

/*
 *  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 { … };
}  // namespace webrtc

#endif  // RTC_BASE_RATE_STATISTICS_H_