1 /* 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_LAG_AGGREGATOR_H_ 12 #define MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_LAG_AGGREGATOR_H_ 13 14 #include <vector> 15 16 #include "absl/types/optional.h" 17 #include "api/audio/echo_canceller3_config.h" 18 #include "modules/audio_processing/aec3/delay_estimate.h" 19 #include "modules/audio_processing/aec3/matched_filter.h" 20 21 namespace webrtc { 22 23 class ApmDataDumper; 24 25 // Aggregates lag estimates produced by the MatchedFilter class into a single 26 // reliable combined lag estimate. 27 class MatchedFilterLagAggregator { 28 public: 29 MatchedFilterLagAggregator(ApmDataDumper* data_dumper, 30 size_t max_filter_lag, 31 const EchoCanceller3Config::Delay& delay_config); 32 33 MatchedFilterLagAggregator() = delete; 34 MatchedFilterLagAggregator(const MatchedFilterLagAggregator&) = delete; 35 MatchedFilterLagAggregator& operator=(const MatchedFilterLagAggregator&) = 36 delete; 37 38 ~MatchedFilterLagAggregator(); 39 40 // Resets the aggregator. 41 void Reset(bool hard_reset); 42 43 // Aggregates the provided lag estimates. 44 absl::optional<DelayEstimate> Aggregate( 45 const absl::optional<const MatchedFilter::LagEstimate>& lag_estimate); 46 47 // Returns whether a reliable delay estimate has been found. ReliableDelayFound()48 bool ReliableDelayFound() const { return significant_candidate_found_; } 49 50 // Returns the delay candidate that is computed by looking at the highest peak 51 // on the matched filters. GetDelayAtHighestPeak()52 int GetDelayAtHighestPeak() const { 53 return highest_peak_aggregator_.candidate(); 54 } 55 56 private: 57 class PreEchoLagAggregator { 58 public: 59 PreEchoLagAggregator(size_t max_filter_lag, size_t down_sampling_factor); 60 void Reset(); 61 void Aggregate(int pre_echo_lag); pre_echo_candidate()62 int pre_echo_candidate() const { return pre_echo_candidate_; } 63 void Dump(ApmDataDumper* const data_dumper); 64 65 private: 66 const int block_size_log2_; 67 std::array<int, 250> histogram_data_; 68 std::vector<int> histogram_; 69 int histogram_data_index_ = 0; 70 int pre_echo_candidate_ = 0; 71 }; 72 73 class HighestPeakAggregator { 74 public: 75 explicit HighestPeakAggregator(size_t max_filter_lag); 76 void Reset(); 77 void Aggregate(int lag); candidate()78 int candidate() const { return candidate_; } histogram()79 rtc::ArrayView<const int> histogram() const { return histogram_; } 80 81 private: 82 std::vector<int> histogram_; 83 std::array<int, 250> histogram_data_; 84 int histogram_data_index_ = 0; 85 int candidate_ = -1; 86 }; 87 88 ApmDataDumper* const data_dumper_; 89 bool significant_candidate_found_ = false; 90 const EchoCanceller3Config::Delay::DelaySelectionThresholds thresholds_; 91 const int headroom_; 92 HighestPeakAggregator highest_peak_aggregator_; 93 std::unique_ptr<PreEchoLagAggregator> pre_echo_lag_aggregator_; 94 }; 95 } // namespace webrtc 96 97 #endif // MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_LAG_AGGREGATOR_H_ 98