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https://github.com/danog/libtgvoip.git
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40 lines
1.2 KiB
C
40 lines
1.2 KiB
C
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/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_AEC3_BLOCK_DELAY_BUFFER_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_DELAY_BUFFER_H_
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#include <stddef.h>
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#include <vector>
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#include "modules/audio_processing/audio_buffer.h"
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namespace webrtc {
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// Class for applying a fixed delay to the samples in a signal partitioned using
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// the audiobuffer band-splitting scheme.
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class BlockDelayBuffer {
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public:
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BlockDelayBuffer(size_t num_bands, size_t frame_length, size_t delay_samples);
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~BlockDelayBuffer();
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// Delays the samples by the specified delay.
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void DelaySignal(AudioBuffer* frame);
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private:
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const size_t frame_length_;
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const size_t delay_;
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std::vector<std::vector<float>> buf_;
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size_t last_insert_ = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_BLOCK_DELAY_BUFFER_H_
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