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diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h
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+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h
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+/*
+ *  Copyright (c) 2019 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_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_
+
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+
+namespace webrtc {
+
+struct RobustThroughputEstimatorSettings {
+  static constexpr char kKey[] = "WebRTC-Bwe-RobustThroughputEstimatorSettings";
+
+  RobustThroughputEstimatorSettings() = delete;
+  explicit RobustThroughputEstimatorSettings(
+      const FieldTrialsView* key_value_config);
+
+  // Set `enabled` to true to use the RobustThroughputEstimator, false to use
+  // the AcknowledgedBitrateEstimator.
+  bool enabled = true;
+
+  // The estimator keeps the smallest window containing at least
+  // `window_packets` and at least the packets received during the last
+  // `min_window_duration` milliseconds.
+  // (This means that it may store more than `window_packets` at high bitrates,
+  // and a longer duration than `min_window_duration` at low bitrates.)
+  // However, if will never store more than kMaxPackets (for performance
+  // reasons), and never longer than max_window_duration (to avoid very old
+  // packets influencing the estimate for example when sending is paused).
+  unsigned window_packets = 20;
+  unsigned max_window_packets = 500;
+  TimeDelta min_window_duration = TimeDelta::Seconds(1);
+  TimeDelta max_window_duration = TimeDelta::Seconds(5);
+
+  // The estimator window requires at least `required_packets` packets
+  // to produce an estimate.
+  unsigned required_packets = 10;
+
+  // If audio packets aren't included in allocation (i.e. the
+  // estimated available bandwidth is divided only among the video
+  // streams), then `unacked_weight` should be set to 0.
+  // If audio packets are included in allocation, but not in bandwidth
+  // estimation (i.e. they don't have transport-wide sequence numbers,
+  // but we nevertheless divide the estimated available bandwidth among
+  // both audio and video streams), then `unacked_weight` should be set to 1.
+  // If all packets have transport-wide sequence numbers, then the value
+  // of `unacked_weight` doesn't matter.
+  double unacked_weight = 1.0;
+
+  std::unique_ptr<StructParametersParser> Parser();
+};
+
+class AcknowledgedBitrateEstimatorInterface {
+ public:
+  static std::unique_ptr<AcknowledgedBitrateEstimatorInterface> Create(
+      const FieldTrialsView* key_value_config);
+  virtual ~AcknowledgedBitrateEstimatorInterface();
+
+  virtual void IncomingPacketFeedbackVector(
+      const std::vector<PacketResult>& packet_feedback_vector) = 0;
+  virtual absl::optional<DataRate> bitrate() const = 0;
+  virtual absl::optional<DataRate> PeekRate() const = 0;
+  virtual void SetAlr(bool in_alr) = 0;
+  virtual void SetAlrEndedTime(Timestamp alr_ended_time) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_