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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_
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