mirror of https://github.com/f4exb/sdrangel.git
55 lines
2.3 KiB
C++
55 lines
2.3 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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* This is a 2 pole lowpass Chebyshev (recursive) filter at fc=0.075 using coefficients found in table 20-1 of
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* http://www.analog.com/media/en/technical-documentation/dsp-book/dsp_book_Ch20.pdf
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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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*
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* a0= 3.869430E-02
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* a1= 7.738860E-02 b1= 1.392667E+00
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* a2= 3.869430E-02 b2= -5.474446E-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] = a0*x[n] + a1*x[n] + a2*x[n] + b1*y[n-1] + b2*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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class 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 init();
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float run(float sample);
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private:
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float m_x[2];
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float m_y[2];
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static const float m_a0, m_a1, m_a2, m_b1, m_b2;
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};
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#endif /* SDRBASE_DSP_FILTERMBE_H_ */
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