Adding full initial version of delay estimation functionality in echo canceller 3

webrtc/modules/audio_processing/aec3/matched_filter.h

Index: webrtc/modules/audio_processing/aec3/matched_filter.h
diff --git a/webrtc/modules/audio_processing/aec3/matched_filter.h b/webrtc/modules/audio_processing/aec3/matched_filter.h
new file mode 100644
index 0000000000000000000000000000000000000000..3e09d4b97157ccfa64f6ccf8c3e6725507276755
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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 WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_H_
+
+#include <array>
+#include <memory>
+#include <vector>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/modules/audio_processing/aec3/aec3_constants.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+
+// Produces recursively updated cross-correlation estimates for several signal
+// shifts where the intra-shift spacing is uniform.
+class MatchedFilter {
+ public:
+  // Stores properties for the lag estimate corresponding to a particular signal
+  // shift.
+  struct LagEstimate {
+    LagEstimate() = default;
+    LagEstimate(float accuracy, bool reliable, size_t lag, bool updated)
+        : accuracy(accuracy), reliable(reliable), lag(lag), updated(updated) {}
+
+    float accuracy = 0.f;
+    bool reliable = false;
+    size_t lag = 0;
+    bool updated = false;
+  };
+
+  MatchedFilter(ApmDataDumper* data_dumper,
+                size_t window_size_sub_blocks,
+                int num_matched_filters,
+                size_t alignment_shift_sub_blocks);
+
+  ~MatchedFilter();
+
+  // Updates the correlation with the values in render and capture.
+  void Update(const std::array<float, kSubBlockSize>& render,
+              const std::array<float, kSubBlockSize>& capture);
+
+  // Returns the current lag estimates.
+  rtc::ArrayView<const MatchedFilter::LagEstimate> GetLagEstimates() const {
+    return lag_estimates_;
+  }
+
+  // Returns the number of lag estimates produced using the shifted signals.
+  size_t NumLagEstimates() const { return filters_.size(); }
+
+ private:
+  // Provides buffer with a related index.
+  struct IndexedBuffer {
+    explicit IndexedBuffer(size_t size);
+    ~IndexedBuffer();
+
+    std::vector<float> data;
+    int index = 0;
+    RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(IndexedBuffer);
+  };
+
+  ApmDataDumper* const data_dumper_;
+  const size_t filter_intra_lag_shift_;
+  std::vector<std::vector<float>> filters_;
+  std::vector<LagEstimate> lag_estimates_;
+  IndexedBuffer x_buffer_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MatchedFilter);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_H_