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diff --git a/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.h b/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
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+/*
+ *  Copyright (c) 2017 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_PROCESSING_AEC3_MATCHED_FILTER_LAG_AGGREGATOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_LAG_AGGREGATOR_H_
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/delay_estimate.h"
+#include "modules/audio_processing/aec3/matched_filter.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+
+// Aggregates lag estimates produced by the MatchedFilter class into a single
+// reliable combined lag estimate.
+class MatchedFilterLagAggregator {
+ public:
+  MatchedFilterLagAggregator(
+      ApmDataDumper* data_dumper,
+      size_t max_filter_lag,
+      const EchoCanceller3Config::Delay::DelaySelectionThresholds& thresholds);
+
+  MatchedFilterLagAggregator() = delete;
+  MatchedFilterLagAggregator(const MatchedFilterLagAggregator&) = delete;
+  MatchedFilterLagAggregator& operator=(const MatchedFilterLagAggregator&) =
+      delete;
+
+  ~MatchedFilterLagAggregator();
+
+  // Resets the aggregator.
+  void Reset(bool hard_reset);
+
+  // Aggregates the provided lag estimates.
+  absl::optional<DelayEstimate> Aggregate(
+      rtc::ArrayView<const MatchedFilter::LagEstimate> lag_estimates);
+
+ private:
+  ApmDataDumper* const data_dumper_;
+  std::vector<int> histogram_;
+  std::array<int, 250> histogram_data_;
+  int histogram_data_index_ = 0;
+  bool significant_candidate_found_ = false;
+  const EchoCanceller3Config::Delay::DelaySelectionThresholds thresholds_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_LAG_AGGREGATOR_H_