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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& delay_config);
+
+ 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(
+ const absl::optional<const MatchedFilter::LagEstimate>& lag_estimate);
+
+ // Returns whether a reliable delay estimate has been found.
+ bool ReliableDelayFound() const { return significant_candidate_found_; }
+
+ // Returns the delay candidate that is computed by looking at the highest peak
+ // on the matched filters.
+ int GetDelayAtHighestPeak() const {
+ return highest_peak_aggregator_.candidate();
+ }
+
+ private:
+ class PreEchoLagAggregator {
+ public:
+ PreEchoLagAggregator(size_t max_filter_lag, size_t down_sampling_factor);
+ void Reset();
+ void Aggregate(int pre_echo_lag);
+ int pre_echo_candidate() const { return pre_echo_candidate_; }
+ void Dump(ApmDataDumper* const data_dumper);
+
+ private:
+ const int block_size_log2_;
+ const bool penalize_high_delays_initial_phase_;
+ std::array<int, 250> histogram_data_;
+ std::vector<int> histogram_;
+ int histogram_data_index_ = 0;
+ int pre_echo_candidate_ = 0;
+ int number_updates_ = 0;
+ };
+
+ class HighestPeakAggregator {
+ public:
+ explicit HighestPeakAggregator(size_t max_filter_lag);
+ void Reset();
+ void Aggregate(int lag);
+ int candidate() const { return candidate_; }
+ rtc::ArrayView<const int> histogram() const { return histogram_; }
+
+ private:
+ std::vector<int> histogram_;
+ std::array<int, 250> histogram_data_;
+ int histogram_data_index_ = 0;
+ int candidate_ = -1;
+ };
+
+ ApmDataDumper* const data_dumper_;
+ bool significant_candidate_found_ = false;
+ const EchoCanceller3Config::Delay::DelaySelectionThresholds thresholds_;
+ const int headroom_;
+ HighestPeakAggregator highest_peak_aggregator_;
+ std::unique_ptr<PreEchoLagAggregator> pre_echo_lag_aggregator_;
+};
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_LAG_AGGREGATOR_H_