mirror of
https://github.com/f4exb/sdrangel.git
synced 2024-11-25 01:18:38 -05:00
92 lines
3.5 KiB
C++
92 lines
3.5 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
|
|
// Copyright (C) 2012 maintech GmbH, Otto-Hahn-Str. 15, 97204 Hoechberg, Germany //
|
|
// written by Christian Daniel //
|
|
// Copyright (C) 2015-2019 Edouard Griffiths, F4EXB <f4exb06@gmail.com> //
|
|
// Copyright (C) 2015 John Greb <hexameron@spam.no> //
|
|
// //
|
|
// 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_DSP_FILTERMBE_H_
|
|
#define SDRBASE_DSP_FILTERMBE_H_
|
|
|
|
/**
|
|
* Uses the generic IIR filter internally
|
|
*
|
|
* Low pass / High pass:
|
|
*
|
|
* This is a 2 pole Chebyshev (recursive) filter using coefficients found in table 20-1 (low pass)
|
|
* or table 20-2 (high pass) of http://www.analog.com/media/en/technical-documentation/dsp-book/dsp_book_Ch20.pdf
|
|
*
|
|
* For low pass fc = 0.075
|
|
* For high oass fc = 0.01
|
|
*
|
|
* Convention taken here exchanges A and B coefficients as shown in this image:
|
|
* https://cdn.mikroe.com/ebooks/img/8/2016/02/digital-filter-design-chapter-03-image-2-9.gif
|
|
* So A applies to Y and B to X
|
|
*
|
|
* At the interpolated sampling frequency of 48 kHz the -3 dB corner is at 48 * .075 = 3.6 kHz which is perfect for voice
|
|
* The high pass has a 3 dB corner of 48 * 0.01 = 0.48 kHz
|
|
*
|
|
* Low pass:
|
|
*
|
|
* b0 = 3.869430E-02 (a0 = 1.0)
|
|
* b1 = 7.738860E-02 a1 = 1.392667E+00
|
|
* b2 = 3.869430E-02 a2 = -5.474446E-01
|
|
*
|
|
* High pass:
|
|
*
|
|
* b0 = 9.567529E-01 (a0 = 1.0)
|
|
* b1 = -1.913506E+00 a1 = 1.911437E+00
|
|
* b2 = 9.567529E-01 a2 = -9.155749E-01
|
|
*
|
|
* given x[n] is the new input sample and y[n] the returned output sample:
|
|
*
|
|
* y[n] = b0*x[n] + b1*x[n] + b2*x[n] + a1*y[n-1] + a2*y[n-2]
|
|
*
|
|
* This one works directly with floats
|
|
*
|
|
*
|
|
*/
|
|
|
|
#include "iirfilter.h"
|
|
#include "export.h"
|
|
|
|
class SDRBASE_API MBEAudioInterpolatorFilter
|
|
{
|
|
public:
|
|
MBEAudioInterpolatorFilter();
|
|
~MBEAudioInterpolatorFilter();
|
|
|
|
void useHP(bool useHP) { m_useHP = useHP; }
|
|
bool usesHP() const { return m_useHP; }
|
|
float run(const float& sample);
|
|
float runHP(const float& sample);
|
|
float runLP(const float& sample);
|
|
|
|
private:
|
|
IIRFilter<float, 2> m_filterLP;
|
|
IIRFilter<float, 2> m_filterHP;
|
|
bool m_useHP;
|
|
// low pass coefficients
|
|
static const float m_lpa[3];
|
|
static const float m_lpb[3];
|
|
// band pass coefficients
|
|
static const float m_hpa[3];
|
|
static const float m_hpb[3];
|
|
};
|
|
|
|
|
|
#endif /* SDRBASE_DSP_FILTERMBE_H_ */
|