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398689f581
This CL adds functionality for applying an optional fixed delay in AEC3 to the capture signal Bug: webrtc:9647 Change-Id: Id3b3f896bcf203e6611298dc804c3c80da9f1883 Reviewed-on: https://webrtc-review.googlesource.com/95142 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24399}
38 lines
1.2 KiB
C++
38 lines
1.2 KiB
C++
/*
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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 <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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