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BFM demod: apply de-emphasis
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2f8fda7137
commit
e533997dbe
@ -79,6 +79,7 @@ set(sdrbase_SOURCES
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sdrbase/dsp/fftengine.cpp
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sdrbase/dsp/fftfilt.cxx
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sdrbase/dsp/fftwindow.cpp
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sdrbase/dsp/filterrc.cpp
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sdrbase/dsp/filesink.cpp
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sdrbase/dsp/interpolator.cpp
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sdrbase/dsp/inthalfbandfilter.cpp
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@ -155,6 +156,7 @@ set(sdrbase_HEADERS
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include/dsp/fftfilt.h
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include/dsp/fftwengine.h
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include/dsp/fftwindow.h
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include/dsp/filterrc.h
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include/dsp/filesink.h
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include/dsp/gfft.h
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include/dsp/interpolator.h
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50
include/dsp/filterrc.h
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50
include/dsp/filterrc.h
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@ -0,0 +1,50 @@
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///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2015 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 INCLUDE_DSP_FILTERRC_H_
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#define INCLUDE_DSP_FILTERRC_H_
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#include "dsp/dsptypes.h"
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/** First order low-pass IIR filter for real-valued signals. */
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class LowPassFilterRC
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{
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public:
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/**
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* Construct 1st order low-pass IIR filter.
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*
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* timeconst :: RC time constant in seconds (1 / (2 * PI * cutoff_freq)
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*/
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LowPassFilterRC(Real timeconst);
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/**
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* Reconfigure filter with new time constant
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*/
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void configure(Real timeout);
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/** Process samples. */
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void process(const Real& sample_in, Real& sample_out);
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private:
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Real m_timeconst;
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Real m_y1;
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Real m_a1;
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Real m_b0;
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};
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#endif /* INCLUDE_DSP_FILTERRC_H_ */
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@ -32,7 +32,9 @@ BFMDemod::BFMDemod(SampleSink* sampleSink) :
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m_sampleSink(sampleSink),
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m_audioFifo(4, 250000),
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m_settingsMutex(QMutex::Recursive),
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m_pilotPLL(19000/384000, 50/384000, 0.01)
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m_pilotPLL(19000/384000, 50/384000, 0.01),
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m_deemphasisFilterX(default_deemphasis * 48000 * 1.0e-6),
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m_deemphasisFilterY(default_deemphasis * 48000 * 1.0e-6)
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{
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setObjectName("BFMDemod");
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@ -43,6 +45,8 @@ BFMDemod::BFMDemod(SampleSink* sampleSink) :
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m_config.m_squelch = -60.0;
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m_config.m_volume = 2.0;
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m_config.m_audioSampleRate = DSPEngine::instance()->getAudioSampleRate(); // normally 48 kHz
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m_deemphasisFilterX.configure(default_deemphasis * m_config.m_audioSampleRate * 1.0e-6);
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m_deemphasisFilterY.configure(default_deemphasis * m_config.m_audioSampleRate * 1.0e-6);
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m_rfFilter = new fftfilt(-50000.0 / 384000.0, 50000.0 / 384000.0, rfFilterFftLength);
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apply();
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@ -120,7 +124,7 @@ void BFMDemod::feed(const SampleVector::const_iterator& begin, const SampleVecto
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m_m1Sample = rf[i];
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m_sampleBuffer.push_back(Sample(demod * (1<<15), 0.0));
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quint16 sampleStereo;
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Real sampleStereo;
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// Process stereo if stereo mode is selected
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@ -135,7 +139,7 @@ void BFMDemod::feed(const SampleVector::const_iterator& begin, const SampleVecto
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if (m_interpolatorStereo.interpolate(&m_interpolatorStereoDistanceRemain, s, &cs))
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{
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sampleStereo = (qint16)(cs.real() * 3000 * m_running.m_volume);
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sampleStereo = cs.real();
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}
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}
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@ -143,15 +147,19 @@ void BFMDemod::feed(const SampleVector::const_iterator& begin, const SampleVecto
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if (m_interpolator.interpolate(&m_interpolatorDistanceRemain, e, &ci))
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{
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quint16 sample = (qint16)(ci.real() * 3000 * m_running.m_volume);
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if (m_running.m_audioStereo)
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{
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m_audioBuffer[m_audioBufferFill].l = sample + sampleStereo;
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m_audioBuffer[m_audioBufferFill].r = sample - sampleStereo;
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Real deemph_l, deemph_r; // Pre-emphasis is applied on each channel before multiplexing
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m_deemphasisFilterX.process(ci.real() + sampleStereo, deemph_l);
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m_deemphasisFilterY.process(ci.real() - sampleStereo, deemph_r);
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m_audioBuffer[m_audioBufferFill].l = (qint16)(deemph_l * 3000 * m_running.m_volume);
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m_audioBuffer[m_audioBufferFill].r = (qint16)(deemph_r * 3000 * m_running.m_volume);
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}
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else
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{
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Real deemph;
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m_deemphasisFilterX.process(ci.real() + sampleStereo, deemph);
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quint16 sample = (qint16)(deemph * 3000 * m_running.m_volume);
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m_audioBuffer[m_audioBufferFill].l = sample;
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m_audioBuffer[m_audioBufferFill].r = sample;
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}
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@ -319,6 +327,12 @@ void BFMDemod::apply()
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m_squelchLevel *= m_squelchLevel;
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}
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if (m_config.m_audioSampleRate != m_running.m_audioSampleRate)
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{
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m_deemphasisFilterX.configure(default_deemphasis * m_config.m_audioSampleRate * 1.0e-6);
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m_deemphasisFilterY.configure(default_deemphasis * m_config.m_audioSampleRate * 1.0e-6);
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}
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m_running.m_inputSampleRate = m_config.m_inputSampleRate;
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m_running.m_inputFrequencyOffset = m_config.m_inputFrequencyOffset;
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m_running.m_rfBandwidth = m_config.m_rfBandwidth;
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@ -27,6 +27,7 @@
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#include "dsp/movingaverage.h"
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#include "dsp/fftfilt.h"
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#include "dsp/phaselock.h"
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#include "dsp/filterrc.h"
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#include "audio/audiofifo.h"
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#include "util/message.h"
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@ -147,6 +148,10 @@ private:
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StereoPhaseLock m_pilotPLL;
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Real m_pilotPLLSamples[2];
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LowPassFilterRC m_deemphasisFilterX;
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LowPassFilterRC m_deemphasisFilterY;
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static const Real default_deemphasis = 50.0; // 50 us
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void apply();
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};
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58
sdrbase/dsp/filterrc.cpp
Normal file
58
sdrbase/dsp/filterrc.cpp
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@ -0,0 +1,58 @@
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///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2015 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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#include <QDebug>
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#include "dsp/filterrc.h"
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// Construct 1st order low-pass IIR filter.
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LowPassFilterRC::LowPassFilterRC(Real timeconst) :
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m_timeconst(timeconst),
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m_y1(0)
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{
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m_a1 = - exp(-1/m_timeconst);
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m_b0 = 1 + m_a1;
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}
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// Reconfigure
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void LowPassFilterRC::configure(Real timeconst)
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{
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m_timeconst = timeconst;
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m_y1 = 0;
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m_a1 = - exp(-1/m_timeconst);
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m_b0 = 1 + m_a1;
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qDebug() << "LowPassFilterRC::configure: t: " << m_timeconst
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<< " a1: " << m_a1
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<< " b0: " << m_b0;
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}
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// Process samples.
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void LowPassFilterRC::process(const Real& sample_in, Real& sample_out)
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{
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/*
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* Continuous domain:
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* H(s) = 1 / (1 - s * timeconst)
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*
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* Discrete domain:
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* H(z) = (1 - exp(-1/timeconst)) / (1 - exp(-1/timeconst) / z)
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*/
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m_y1 = (sample_in * m_b0) - (m_y1 * m_a1);
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sample_out = m_y1;
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}
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@ -81,6 +81,7 @@ PhaseLock::PhaseLock(Real freq, Real bandwidth, Real minsignal)
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m_sample_cnt = 0;
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}
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void PhaseLock::configure(Real freq, Real bandwidth, Real minsignal)
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{
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qDebug("PhaseLock::configure: freq: %f bandwidth: %f minsignal: %f", freq, bandwidth, minsignal);
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@ -142,6 +143,7 @@ void PhaseLock::configure(Real freq, Real bandwidth, Real minsignal)
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m_sample_cnt = 0;
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}
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// Process samples. Bufferized version
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void PhaseLock::process(const std::vector<Real>& samples_in, std::vector<Real>& samples_out)
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{
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@ -247,21 +249,6 @@ void PhaseLock::process(const std::vector<Real>& samples_in, std::vector<Real>&
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m_sample_cnt += n;
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}
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/*
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void PhaseLock::process(const Real& sample_in, Real& sample_out)
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{
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m_phase += m_freq;
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if (m_phase > 2.0 * M_PI) {
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m_phase -= 2.0 * M_PI;
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}
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Real psin = sin(m_phase);
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Real pcos = cos(m_phase);
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sample_out = 2 * psin * pcos;
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}*/
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// Process samples. Multiple output
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void PhaseLock::process(const Real& sample_in, Real *samples_out)
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