Repo created

TMessagesProj/jni/voip/webrtc/rtc_base/rate_statistics.h

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+/*
+ *  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_