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libtgvoip/webrtc_dsp/modules/audio_processing/aec3/block_delay_buffer.h
Grishka 5caaaafa42 Updated WebRTC APM
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
2018-11-23 04:02:53 +03:00

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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_