1
0
mirror of https://github.com/f4exb/sdrangel.git synced 2024-12-12 09:55:07 -05:00
sdrangel/sdrbase/dsp/filtermbe.h

55 lines
2.3 KiB
C++

///////////////////////////////////////////////////////////////////////////////////
// Copyright (C) 2016 F4EXB //
// written by Edouard Griffiths //
// //
// This program is free software; you can redistribute it and/or modify //
// it under the terms of the GNU General Public License as published by //
// the Free Software Foundation as version 3 of the License, or //
// //
// This program is distributed in the hope that it will be useful, //
// but WITHOUT ANY WARRANTY; without even the implied warranty of //
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
// GNU General Public License V3 for more details. //
// //
// You should have received a copy of the GNU General Public License //
// along with this program. If not, see <http://www.gnu.org/licenses/>. //
///////////////////////////////////////////////////////////////////////////////////
#ifndef SDRBASE_DSP_FILTERMBE_H_
#define SDRBASE_DSP_FILTERMBE_H_
/**
* This is a 2 pole lowpass Chebyshev (recursive) filter at fc=0.075 using coefficients found in table 20-1 of
* http://www.analog.com/media/en/technical-documentation/dsp-book/dsp_book_Ch20.pdf
*
* At the interpolated sampling frequency of 48 kHz the -3 dB corner is at 48 * .075 = 3.6 kHz which is perfect for voice
*
* a0= 3.869430E-02
* a1= 7.738860E-02 b1= 1.392667E+00
* a2= 3.869430E-02 b2= -5.474446E-01
*
* given x[n] is the new input sample and y[n] the returned output sample:
*
* y[n] = a0*x[n] + a1*x[n] + a2*x[n] + b1*y[n-1] + b2*y[n-2]
*
* This one works directly with floats
*
*/
class MBEAudioInterpolatorFilter
{
public:
MBEAudioInterpolatorFilter();
~MBEAudioInterpolatorFilter();
void init();
float run(float sample);
private:
float m_x[2];
float m_y[2];
static const float m_a0, m_a1, m_a2, m_b1, m_b2;
};
#endif /* SDRBASE_DSP_FILTERMBE_H_ */