mirror of https://github.com/g4klx/MMDVM.git
Add FM filtering
This commit is contained in:
Geoffrey Merck
parent
507a4bf64c
commit
f8d082e2dd
14
FM.cpp
14
FM.cpp
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@ -23,9 +23,9 @@
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// 3 stage IIR Butterworth filter generated (if you change the order change the size of m_filterState). Also change the ordre in init call below
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// 0.2db band pass ripple
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// 300 - 2700Hz
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q15_t FILTER_COEFFS[] = { 362, 724, 362,16384,-18947,10676, 14,//1st stage
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16384, 0, -16384,16384,-25170, 9526, 14,//2nd stage
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16384,-32768, 16384,16384,-32037,15730,14};//3rd stage
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q15_t FILTER_COEFFS[] = {362,724,362,16384,-18947,10676,
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16384,0,-16384,16384,-25170,9526,
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16384,-32768,16384,16384,-32037,15730};
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CFM::CFM() :
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@ -44,14 +44,14 @@ m_kerchunkTimer(),
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m_ackMinTimer(),
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m_ackDelayTimer(),
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m_hangTimer(),
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m_filter()
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m_filterStage1( 724, 1448, 724, 32768, -37895, 21352),
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m_filterStage2(32768, 0,-32768, 32768, -50339, 19052),
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m_filterStage3(32768, -65536, 32768, 32768, -64075, 31460)
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{
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arm_biquad_cascade_df1_init_q15(&m_filter, 3, FILTER_COEFFS, m_filterState, 0);
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}
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void CFM::samples(q15_t* samples, uint8_t length)
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{
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arm_biquad_casd_df1_inst_q15* filterPtr = &m_filter;
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uint8_t i = 0;
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for (; i < length; i++) {
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q15_t currentSample = samples[i];//save to a local variable to avoid indirection on every access
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@ -97,7 +97,7 @@ void CFM::samples(q15_t* samples, uint8_t length)
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if (!m_callsign.isRunning() && !m_rfAck.isRunning())
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currentSample += m_timeoutTone.getAudio();
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arm_biquad_cascade_df1_fast_q15(filterPtr, samples +i, ¤tSample, 1);
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currentSample = q15_t(m_filterStage3.filter(m_filterStage2.filter(m_filterStage1.filter(currentSample))));
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currentSample += m_ctcssTX.getAudio();
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6
FM.h
6
FM.h
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@ -26,6 +26,7 @@
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#include "FMTimeout.h"
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#include "FMKeyer.h"
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#include "FMTimer.h"
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#include "FMDirectForm1.h"
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enum FM_STATE {
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FS_LISTENING,
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@ -70,8 +71,9 @@ private:
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CFMTimer m_ackMinTimer;
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CFMTimer m_ackDelayTimer;
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CFMTimer m_hangTimer;
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arm_biquad_casd_df1_inst_q15 m_filter;
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q15_t m_filterState[12];//must be filterOrder * 4 long
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CFMDirectFormI m_filterStage1;
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CFMDirectFormI m_filterStage2;
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CFMDirectFormI m_filterStage3;
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void stateMachine(bool validSignal, uint8_t length);
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void listeningState(bool validSignal);
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@ -0,0 +1,114 @@
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/*******************************************************************************
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This header file has been taken from:
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"A Collection of Useful C++ Classes for Digital Signal Processing"
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By Vinnie Falco
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Bernd Porr adapted it for Linux and turned it into a filter using
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fixed point arithmetic.
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--------------------------------------------------------------------------------
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License: MIT License (http://www.opensource.org/licenses/mit-license.php)
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Copyright (c) 2009 by Vinnie Falco
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Copyright (C) 2013-2017, Bernd Porr, mail@berndporr.me.uk
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Copyright (C) 2020 , Mario Molitor , mario_molitor@web.de
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Permission is hereby granted, free of charge, to any person obtaining a copy
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of this software and associated documentation files (the "Software"), to deal
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in the Software without restriction, including without limitation the rights
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to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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copies of the Software, and to permit persons to whom the Software is
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furnished to do so, subject to the following conditions:
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The above copyright notice and this permission notice shall be included in
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all copies or substantial portions of the Software.
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THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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THE SOFTWARE.
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*******************************************************************************/
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// based on https://raw.githubusercontent.com/berndporr/iir_fixed_point/master/DirectFormI.h
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#include "Globals.h"
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#ifndef DIRECTFORMI_HPP_
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#define DIRECTFORMI_HPP_
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class CFMDirectFormI
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{
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public:
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// convenience function which takes the a0 argument but ignores it!
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CFMDirectFormI(const q31_t b0, const q31_t b1, const q31_t b2,
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const q31_t, const q31_t a1, const q31_t a2)
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{
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// FIR coefficients
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c_b0 = b0;
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c_b1 = b1;
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c_b2 = b2;
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// IIR coefficients
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c_a1 = a1;
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c_a2 = a2;
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reset();
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}
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CFMDirectFormI(const CFMDirectFormI &my)
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{
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// delay line
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m_x2 = my.m_x2; // x[n-2]
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m_y2 = my.m_y2; // y[n-2]
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m_x1 = my.m_x1; // x[n-1]
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m_y1 = my.m_y1; // y[n-1]
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// coefficients
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c_b0 = my.c_b0;
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c_b1 = my.c_b1;
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c_b2 = my.c_b2; // FIR
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c_a1 = my.c_a1;
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c_a2 = my.c_a2; // IIR
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// scaling factor
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q_scaling = my.q_scaling; // 2^q_scaling
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}
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void reset ()
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{
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m_x1 = 0;
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m_x2 = 0;
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m_y1 = 0;
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m_y2 = 0;
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}
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// filtering operation: one sample in and one out
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inline q15_t filter(const q15_t in)
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{
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// calculate the output
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register q31_t out_upscaled = c_b0 * in //F4FXL puting stauration here made everything quiet, not sure why
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+ c_b1 * m_x1
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+ c_b2 * m_x2
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- c_a1 * m_y1
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- c_a2 * m_y2;
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q15_t out = __SSAT(out_upscaled >> 15, 15);
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// update the delay lines
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m_x2 = m_x1;
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m_y2 = m_y1;
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m_x1 = in;
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m_y1 = out;
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return out;
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}
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private:
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// delay line
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q31_t m_x2; // x[n-2]
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q31_t m_y2; // y[n-2]
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q31_t m_x1; // x[n-1]
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q31_t m_y1; // y[n-1]
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// coefficients
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q31_t c_b0,c_b1,c_b2; // FIR
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q31_t c_a1,c_a2; // IIR
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};
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#endif /* DIRECTFORMI_HPP_ */
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@ -0,0 +1,45 @@
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# based on https://github.com/berndporr/iir_fixed_point/blob/master/gen_coeff.py
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import numpy as np
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import scipy.signal as signal
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import pylab as pl
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# Calculate the coefficients for a pure fixed point
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# integer filter
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# sampling rate
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fs = 24000
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# cutoffs
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f1 = 300
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f2 = 2700
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# ripple
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rp = 0.2
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# scaling factor in bits, do not change !
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q = 15
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# scaling factor as facor...
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scaling_factor = 2**q
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# let's generate a sequence of 2nd order IIR filters
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#sos = signal.butter(2,[f1/fs*2,f2/fs*2],'pass',output='sos')
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sos = signal.cheby1(3,rp,[f1/fs*2,f2/fs*2],'bandpass', output='sos')
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sos = np.round((sos) * scaling_factor)
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# print coefficients
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for biquad in sos:
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for coeff in biquad:
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print(int(coeff),",",sep="",end="")
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#print((coeff),",",sep="",end="")
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print("")
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# plot the frequency response
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b,a = signal.sos2tf(sos)
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w,h = signal.freqz(b,a)
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pl.plot(w/np.pi/2*fs,20*np.log(np.abs(h)))
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pl.xlabel('frequency/Hz');
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pl.ylabel('gain/dB');
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pl.ylim(top=1,bottom=-20);
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pl.xlim(left=250, right=12000);
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pl.show()
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