kopia lustrzana https://github.com/f4exb/sdrangel
82 wiersze
3.1 KiB
C++
82 wiersze
3.1 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2016 F4EXB //
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// written by Edouard Griffiths //
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// //
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// This program is free software; you can redistribute it and/or modify //
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// it under the terms of the GNU General Public License as published by //
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// the Free Software Foundation as version 3 of the License, or //
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// //
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// This program is distributed in the hope that it will be useful, //
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// but WITHOUT ANY WARRANTY; without even the implied warranty of //
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
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// GNU General Public License V3 for more details. //
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// //
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// You should have received a copy of the GNU General Public License //
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// along with this program. If not, see <http://www.gnu.org/licenses/>. //
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///////////////////////////////////////////////////////////////////////////////////
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#ifndef SDRBASE_DSP_FILTERMBE_H_
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#define SDRBASE_DSP_FILTERMBE_H_
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/**
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* Uses the generic IIR filter internally
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*
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* Low pass / High pass:
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*
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* This is a 2 pole Chebyshev (recursive) filter using coefficients found in table 20-1 (low pass)
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* or table 20-2 (high pass) of http://www.analog.com/media/en/technical-documentation/dsp-book/dsp_book_Ch20.pdf
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*
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* For low pass fc = 0.075
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* For high oass fc = 0.01
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*
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* Convention taken here exchanges A and B coefficients as shown in this image:
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* https://cdn.mikroe.com/ebooks/img/8/2016/02/digital-filter-design-chapter-03-image-2-9.gif
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* So A applies to Y and B to X
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*
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* At the interpolated sampling frequency of 48 kHz the -3 dB corner is at 48 * .075 = 3.6 kHz which is perfect for voice
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* The high pass has a 3 dB corner of 48 * 0.01 = 0.48 kHz
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*
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* Low pass:
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*
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* b0 = 3.869430E-02 (a0 = 1.0)
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* b1 = 7.738860E-02 a1 = 1.392667E+00
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* b2 = 3.869430E-02 a2 = -5.474446E-01
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*
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* High pass:
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*
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* b0 = 9.567529E-01 (a0 = 1.0)
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* b1 = -1.913506E+00 a1 = 1.911437E+00
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* b2 = 9.567529E-01 a2 = -9.155749E-01
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*
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* given x[n] is the new input sample and y[n] the returned output sample:
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*
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* y[n] = b0*x[n] + b1*x[n] + b2*x[n] + a1*y[n-1] + a2*y[n-2]
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*
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* This one works directly with floats
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*
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*
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*/
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#include "iirfilter.h"
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#include "export.h"
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class SDRBASE_API MBEAudioInterpolatorFilter
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{
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public:
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MBEAudioInterpolatorFilter();
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~MBEAudioInterpolatorFilter();
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void useHP(bool useHP) { m_useHP = useHP; }
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float run(const float& sample);
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private:
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IIRFilter<float, 2> m_filterLP;
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IIRFilter<float, 2> m_filterHP;
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bool m_useHP;
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static const float m_lpa[3], m_lpb[3]; // low pass coefficients
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static const float m_hpa[3], m_hpb[3]; // band pass coefficients
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};
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#endif /* SDRBASE_DSP_FILTERMBE_H_ */
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