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Audio UDP/RTP: added a lowpass filter for decimation

This commit is contained in:
f4exb 2019-02-14 14:27:06 +01:00
parent 5cd7f4a2e6
commit c46bf5b110
4 changed files with 114 additions and 0 deletions
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2
sdrbase/CMakeLists.txt
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@ -7,6 +7,7 @@ set(sdrbase_SOURCES
audio/audiocompressor.cpp
audio/audiodevicemanager.cpp
audio/audiofifo.cpp
audio/audiofilter.cpp
audio/audiooutput.cpp
audio/audioinput.cpp
audio/audionetsink.cpp
@ -99,6 +100,7 @@ set(sdrbase_HEADERS
audio/audiocompressor.h
audio/audiodevicemanager.h
audio/audiofifo.h
audio/audiofilter.h
audio/audiooutput.h
audio/audioinput.h
audio/audionetsink.h

48
sdrbase/audio/audiofilter.cpp Normal file
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@ -0,0 +1,48 @@
///////////////////////////////////////////////////////////////////////////////////
// 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 //
// //
// 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/>. //
///////////////////////////////////////////////////////////////////////////////////
#include "audiofilter.h"
const float AudioFilter::m_lpa[3] = {1.0, 1.392667E+00, -5.474446E-01};
const float AudioFilter::m_lpb[3] = {3.869430E-02, 7.738860E-02, 3.869430E-02};
// f(-3dB) = 300 Hz @ 8000 Hz SR (w = 0.075):
const float AudioFilter::m_hpa[3] = {1.000000e+00, 1.667871e+00, -7.156964e-01};
const float AudioFilter::m_hpb[3] = {8.459039e-01, -1.691760e+00, 8.459039e-01};
AudioFilter::AudioFilter() :
m_filterLP(m_lpa, m_lpb),
m_filterHP(m_hpa, m_hpb),
m_useHP(false)
{
}
AudioFilter::~AudioFilter()
{}
float AudioFilter::run(const float& sample)
{
return m_useHP ? m_filterLP.run(m_filterHP.run(sample)) : m_filterLP.run(sample);
}
float AudioFilter::runHP(const float& sample)
{
return m_filterHP.run(sample);
}
float AudioFilter::runLP(const float& sample)
{
return m_filterLP.run(sample);
}

62
sdrbase/audio/audiofilter.h Normal file
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@ -0,0 +1,62 @@
///////////////////////////////////////////////////////////////////////////////////
// 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 //
// //
// 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"
/**
* 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 SDRBASE_API AudioFilter {
public:
AudioFilter();
~AudioFilter();
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;
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_

2
sdrbase/sdrbase.pro
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@ -56,6 +56,7 @@ MINGW32 || MINGW64 || MSVC {
SOURCES += audio/audiodevicemanager.cpp\
audio/audiocompressor.cpp\
audio/audiofifo.cpp\
audio/audiofilter.cpp\
audio/audiooutput.cpp\
audio/audioinput.cpp\
audio/audionetsink.cpp\
@ -134,6 +135,7 @@ SOURCES += audio/audiodevicemanager.cpp\
HEADERS += audio/audiodevicemanager.h\
audio/audiocompressor.h\
audio/audiofifo.h\
audio/audiofilter.h\
audio/audiooutput.h\
audio/audioinput.h\
audio/audionetsink.h\