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diff --git a/third_party/libwebrtc/modules/video_coding/timing/rtt_filter.h b/third_party/libwebrtc/modules/video_coding/timing/rtt_filter.h
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+/*
+ *  Copyright (c) 2011 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 MODULES_VIDEO_CODING_TIMING_RTT_FILTER_H_
+#define MODULES_VIDEO_CODING_TIMING_RTT_FILTER_H_
+
+#include <stdint.h>
+
+#include "absl/container/inlined_vector.h"
+#include "api/units/time_delta.h"
+
+namespace webrtc {
+
+class RttFilter {
+ public:
+  RttFilter();
+  RttFilter(const RttFilter&) = delete;
+  RttFilter& operator=(const RttFilter&) = delete;
+
+  // Resets the filter.
+  void Reset();
+  // Updates the filter with a new sample.
+  void Update(TimeDelta rtt);
+  // A getter function for the current RTT level.
+  TimeDelta Rtt() const;
+
+ private:
+  // The size of the drift and jump memory buffers
+  // and thus also the detection threshold for these
+  // detectors in number of samples.
+  static constexpr int kMaxDriftJumpCount = 5;
+  using BufferList = absl::InlinedVector<TimeDelta, kMaxDriftJumpCount>;
+
+  // Detects RTT jumps by comparing the difference between
+  // samples and average to the standard deviation.
+  // Returns true if the long time statistics should be updated
+  // and false otherwise
+  bool JumpDetection(TimeDelta rtt);
+
+  // Detects RTT drifts by comparing the difference between
+  // max and average to the standard deviation.
+  // Returns true if the long time statistics should be updated
+  // and false otherwise
+  bool DriftDetection(TimeDelta rtt);
+
+  // Computes the short time average and maximum of the vector buf.
+  void ShortRttFilter(const BufferList& buf);
+
+  bool got_non_zero_update_;
+  TimeDelta avg_rtt_;
+  // Variance units are TimeDelta^2. Store as ms^2.
+  int64_t var_rtt_;
+  TimeDelta max_rtt_;
+  uint32_t filt_fact_count_;
+  bool last_jump_positive_ = false;
+  BufferList jump_buf_;
+  BufferList drift_buf_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_TIMING_RTT_FILTER_H_