mirror of
https://github.com/f4exb/sdrangel.git
synced 2024-11-25 17:28:50 -05:00
77 lines
3.3 KiB
C++
77 lines
3.3 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
|
|
// Copyright (C) 2019 Edouard Griffiths, F4EXB //
|
|
// //
|
|
// 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 //
|
|
// (at your option) any later version. //
|
|
// //
|
|
// 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_AUDIO_AUDIOFILTER_H_
|
|
#define _SDRBASE_AUDIO_AUDIOFILTER_H_
|
|
|
|
#include "export.h"
|
|
#include "dsp/iirfilter.h"
|
|
|
|
/**
|
|
* By default 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
|
|
*
|
|
* It can be generalized using the program found in tables 20-4 and 20-5 of the same document. This form is used as a
|
|
* decimation filter and can be set with the setDecimFilters method
|
|
*/
|
|
|
|
class SDRBASE_API AudioFilter {
|
|
public:
|
|
AudioFilter();
|
|
~AudioFilter();
|
|
|
|
void useHP(bool useHP) { m_useHP = useHP; }
|
|
bool usesHP() const { return m_useHP; }
|
|
void setDecimFilters(int srHigh, int srLow, float fcHigh, float fcLow, float gain = 1.0f);
|
|
float run(const float& sample);
|
|
float runHP(const float& sample);
|
|
float runLP(const float& sample);
|
|
|
|
private:
|
|
void calculate2(bool highPass, double fc, float *a, float *b, float fgain); // two pole Chebyshev calculation
|
|
void cheby(bool highPass, double fc, float pr, int np, double *a, double *b, float fgain);
|
|
void cheby_sub(bool highPass, double fc, float pr, int np, int stage,
|
|
double& a0, double& a1, double& a2, double& b1, double& b2);
|
|
|
|
IIRFilter<float, 2> m_filterLP;
|
|
IIRFilter<float, 2> m_filterHP;
|
|
bool m_useHP;
|
|
float m_lpva[3];
|
|
float m_lpvb[3];
|
|
float m_hpva[3];
|
|
float m_hpvb[3];
|
|
static const float m_lpa[3];
|
|
static const float m_lpb[3];
|
|
static const float m_hpa[3];
|
|
static const float m_hpb[3];
|
|
|
|
};
|
|
|
|
#endif // _SDRBASE_AUDIO_AUDIOFILTER_H_
|