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https://github.com/danog/libtgvoip.git
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5caaaafa42
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.
45 lines
1.3 KiB
C++
45 lines
1.3 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_MOVING_AVERAGE_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_MOVING_AVERAGE_H_
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#include <stddef.h>
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#include <vector>
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#include "api/array_view.h"
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namespace webrtc {
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namespace aec3 {
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class MovingAverage {
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public:
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// Creates an instance of MovingAverage that accepts inputs of length num_elem
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// and averages over mem_len inputs.
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MovingAverage(size_t num_elem, size_t mem_len);
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~MovingAverage();
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// Computes the average of input and mem_len-1 previous inputs and stores the
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// result in output.
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void Average(rtc::ArrayView<const float> input, rtc::ArrayView<float> output);
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private:
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const size_t num_elem_;
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const size_t mem_len_;
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const float scaling_;
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std::vector<float> memory_;
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size_t mem_index_;
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};
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} // namespace aec3
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_MOVING_AVERAGE_H_
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