mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 06:10:40 +01:00
988cc0870b
This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
39 lines
1.2 KiB
C++
39 lines
1.2 KiB
C++
/*
|
|
* 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_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<float>> buf_;
|
|
size_t last_insert_ = 0;
|
|
};
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_AEC3_BLOCK_DELAY_BUFFER_H_
|