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+/*
+ * Copyright (c) 2018 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_BLOCK_DELAY_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_DELAY_BUFFER_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "modules/audio_processing/audio_buffer.h"
+
+namespace webrtc {
+
+// Class for applying a fixed delay to the samples in a signal partitioned using
+// the audiobuffer band-splitting scheme.
+class BlockDelayBuffer {
+ public:
+ BlockDelayBuffer(size_t num_channels,
+ size_t num_bands,
+ size_t frame_length,
+ size_t delay_samples);
+ ~BlockDelayBuffer();
+
+ // Delays the samples by the specified delay.
+ void DelaySignal(AudioBuffer* frame);
+
+ private:
+ const size_t frame_length_;
+ const size_t delay_;
+ std::vector<std::vector<std::vector<float>>> buf_;
+ size_t last_insert_ = 0;
+};
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_PROCESSING_AEC3_BLOCK_DELAY_BUFFER_H_