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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.
41 lines
1.2 KiB
C++
41 lines
1.2 KiB
C++
/*
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* Copyright (c) 2017 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_DECIMATOR_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_DECIMATOR_H_
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#include <array>
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#include "api/array_view.h"
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#include "modules/audio_processing/aec3/aec3_common.h"
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#include "modules/audio_processing/aec3/cascaded_biquad_filter.h"
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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// Provides functionality for decimating a signal.
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class Decimator {
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public:
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explicit Decimator(size_t down_sampling_factor);
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// Downsamples the signal.
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void Decimate(rtc::ArrayView<const float> in, rtc::ArrayView<float> out);
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private:
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const size_t down_sampling_factor_;
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CascadedBiQuadFilter anti_aliasing_filter_;
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CascadedBiQuadFilter noise_reduction_filter_;
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RTC_DISALLOW_COPY_AND_ASSIGN(Decimator);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_DECIMATOR_H_
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