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diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h
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+/*
+ *  Copyright (c) 2022 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_AUDIO_CODING_NETEQ_PACKET_ARRIVAL_HISTORY_H_
+#define MODULES_AUDIO_CODING_NETEQ_PACKET_ARRIVAL_HISTORY_H_
+
+#include <cstdint>
+#include <deque>
+
+#include "absl/types/optional.h"
+#include "api/neteq/tick_timer.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+
+namespace webrtc {
+
+// Stores timing information about previously received packets.
+// The history has a fixed window size beyond which old data is automatically
+// pruned.
+class PacketArrivalHistory {
+ public:
+  explicit PacketArrivalHistory(int window_size_ms);
+
+  // Insert packet with `rtp_timestamp` and `arrival_time_ms` into the history.
+  void Insert(uint32_t rtp_timestamp, int64_t arrival_time_ms);
+
+  // The delay for `rtp_timestamp` at `time_ms` is calculated as
+  // `(time_ms - p.arrival_time_ms) - (rtp_timestamp - p.rtp_timestamp)`
+  // where `p` is chosen as the packet arrival in the history that maximizes the
+  // delay.
+  int GetDelayMs(uint32_t rtp_timestamp, int64_t time_ms) const;
+
+  // Get the maximum packet arrival delay observed in the history.
+  int GetMaxDelayMs() const;
+
+  bool IsNewestRtpTimestamp(uint32_t rtp_timestamp) const;
+
+  void Reset();
+
+  void set_sample_rate(int sample_rate) {
+    sample_rate_khz_ = sample_rate / 1000;
+  }
+
+  size_t size() const { return history_.size(); }
+
+ private:
+  struct PacketArrival {
+    PacketArrival(int64_t rtp_timestamp_ms, int64_t arrival_time_ms)
+        : rtp_timestamp_ms(rtp_timestamp_ms),
+          arrival_time_ms(arrival_time_ms) {}
+    int64_t rtp_timestamp_ms;
+    int64_t arrival_time_ms;
+    bool operator<=(const PacketArrival& other) const {
+      return arrival_time_ms - rtp_timestamp_ms <=
+             other.arrival_time_ms - other.rtp_timestamp_ms;
+    }
+    bool operator>=(const PacketArrival& other) const {
+      return arrival_time_ms - rtp_timestamp_ms >=
+             other.arrival_time_ms - other.rtp_timestamp_ms;
+    }
+  };
+  std::deque<PacketArrival> history_;
+  int GetPacketArrivalDelayMs(const PacketArrival& packet_arrival) const;
+  // Updates `min_packet_arrival_` and `max_packet_arrival_`.
+  void MaybeUpdateCachedArrivals(const PacketArrival& packet);
+  const PacketArrival* min_packet_arrival_ = nullptr;
+  const PacketArrival* max_packet_arrival_ = nullptr;
+  const int window_size_ms_;
+  RtpTimestampUnwrapper timestamp_unwrapper_;
+  absl::optional<int64_t> newest_rtp_timestamp_;
+  int sample_rate_khz_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_PACKET_ARRIVAL_HISTORY_H_