Add FM filtering

FM.cpp

@@ -23,9 +23,9 @@
 // 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
 // 0.2db band pass ripple
 // 300 - 2700Hz
-q15_t FILTER_COEFFS[] = {  362,   724,    362,16384,-18947,10676, 14,//1st stage
+q15_t FILTER_COEFFS[] = {362,724,362,16384,-18947,10676,
-                         16384,     0, -16384,16384,-25170, 9526, 14,//2nd stage
+                          16384,0,-16384,16384,-25170,9526,
-                         16384,-32768,  16384,16384,-32037,15730,14};//3rd stage
+                          16384,-32768,16384,16384,-32037,15730};
 
 
 CFM::CFM() :
@@ -44,14 +44,14 @@ m_kerchunkTimer(),
 m_ackMinTimer(),
 m_ackDelayTimer(),
 m_hangTimer(),
-m_filter()
+m_filterStage1(  724,   1448,   724, 32768, -37895, 21352),
+m_filterStage2(32768,      0,-32768, 32768, -50339, 19052),
+m_filterStage3(32768, -65536, 32768, 32768, -64075, 31460)
 {
-  arm_biquad_cascade_df1_init_q15(&m_filter, 3, FILTER_COEFFS, m_filterState, 0);
 }
 
 void CFM::samples(q15_t* samples, uint8_t length)
 {
-  arm_biquad_casd_df1_inst_q15* filterPtr = &m_filter;
   uint8_t i = 0;
   for (; i < length; i++) {
     q15_t currentSample = samples[i];//save to a local variable to avoid indirection on every access
@@ -97,7 +97,7 @@ void CFM::samples(q15_t* samples, uint8_t length)
     if (!m_callsign.isRunning() && !m_rfAck.isRunning())
       currentSample += m_timeoutTone.getAudio();
 
-    arm_biquad_cascade_df1_fast_q15(filterPtr, samples +i, &currentSample, 1);
+    currentSample = q15_t(m_filterStage3.filter(m_filterStage2.filter(m_filterStage1.filter(currentSample))));
 
     currentSample += m_ctcssTX.getAudio();
 

FM.h

@@ -26,6 +26,7 @@
 #include "FMTimeout.h"
 #include "FMKeyer.h"
 #include "FMTimer.h"
+#include "FMDirectForm1.h"
 
 enum FM_STATE {
   FS_LISTENING,
@@ -70,8 +71,9 @@ private:
   CFMTimer             m_ackMinTimer;
   CFMTimer             m_ackDelayTimer;
   CFMTimer             m_hangTimer;
-  arm_biquad_casd_df1_inst_q15 m_filter;
+  CFMDirectFormI          m_filterStage1;
-  q15_t                m_filterState[12];//must be filterOrder * 4 long
+  CFMDirectFormI          m_filterStage2;
+  CFMDirectFormI          m_filterStage3;
 
   void stateMachine(bool validSignal, uint8_t length);
   void listeningState(bool validSignal);

FMDirectForm1.h

@@ -0,0 +1,114 @@
+
+/*******************************************************************************
+This header file has been taken from:
+"A Collection of Useful C++ Classes for Digital Signal Processing"
+By Vinnie Falco 
+Bernd Porr adapted it for Linux and turned it into a filter using
+fixed point arithmetic.
+--------------------------------------------------------------------------------
+License: MIT License (http://www.opensource.org/licenses/mit-license.php)
+Copyright (c) 2009 by Vinnie Falco
+Copyright (C) 2013-2017, Bernd Porr, mail@berndporr.me.uk
+Copyright (C) 2020 , Mario Molitor , mario_molitor@web.de
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+The above copyright notice and this permission notice shall be included in
+all copies or substantial portions of the Software.
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+THE SOFTWARE.
+*******************************************************************************/
+
+// based on https://raw.githubusercontent.com/berndporr/iir_fixed_point/master/DirectFormI.h
+
+#include "Globals.h"
+
+#ifndef DIRECTFORMI_HPP_
+#define DIRECTFORMI_HPP_
+class CFMDirectFormI
+{
+public:
+
+	// convenience function which takes the a0 argument but ignores it!
+	CFMDirectFormI(const q31_t b0, const q31_t b1, const q31_t b2,
+			const q31_t, const q31_t a1, const q31_t a2)
+	{
+		// FIR coefficients
+		c_b0 = b0;
+		c_b1 = b1;
+		c_b2 = b2;
+		// IIR coefficients
+		c_a1 = a1;
+		c_a2 = a2;
+		reset();
+	}
+
+	CFMDirectFormI(const CFMDirectFormI &my)
+	{
+	// delay line
+		m_x2 = my.m_x2; // x[n-2]
+		m_y2 = my.m_y2; // y[n-2]
+		m_x1 = my.m_x1; // x[n-1]
+		m_y1 = my.m_y1; // y[n-1]
+
+	// coefficients
+		c_b0 = my.c_b0;
+		c_b1 = my.c_b1;
+		c_b2 = my.c_b2; // FIR
+		c_a1 = my.c_a1;
+		c_a2 = my.c_a2; // IIR
+
+	// scaling factor
+		q_scaling = my.q_scaling; // 2^q_scaling
+    }
+
+	void reset ()
+	{
+		m_x1 = 0;
+		m_x2 = 0;
+		m_y1 = 0;
+		m_y2 = 0;
+	}
+
+	// filtering operation: one sample in and one out
+	inline q15_t filter(const q15_t in)
+	{
+		// calculate the output
+		register q31_t out_upscaled = c_b0 * in    //F4FXL puting stauration here made everything quiet, not sure why
+									+ c_b1 * m_x1
+									+ c_b2 * m_x2
+									- c_a1 * m_y1
+									- c_a2 * m_y2;
+
+		q15_t out = __SSAT(out_upscaled >> 15, 15);
+
+		// update the delay lines
+		m_x2 = m_x1;
+		m_y2 = m_y1;
+		m_x1 = in;
+		m_y1 = out;
+
+		return out;
+	}
+
+private:
+	// delay line
+	q31_t m_x2; // x[n-2]
+	q31_t m_y2; // y[n-2]
+	q31_t m_x1; // x[n-1]
+	q31_t m_y1; // y[n-1]
+
+	// coefficients
+	q31_t c_b0,c_b1,c_b2; // FIR
+	q31_t c_a1,c_a2; // IIR
+};
+
+#endif /* DIRECTFORMI_HPP_ */

FMGenerateFilterCoefficients.py

@@ -0,0 +1,45 @@
+# based on https://github.com/berndporr/iir_fixed_point/blob/master/gen_coeff.py
+
+import numpy as np
+import scipy.signal as signal
+import pylab as pl
+
+# Calculate the coefficients for a pure fixed point
+# integer filter
+
+# sampling rate
+fs = 24000
+
+# cutoffs
+f1 = 300
+f2 = 2700
+# ripple
+rp = 0.2
+
+# scaling factor in bits, do not change !
+q = 15
+# scaling factor as facor...
+scaling_factor = 2**q
+
+# let's generate a sequence of 2nd order IIR filters
+#sos = signal.butter(2,[f1/fs*2,f2/fs*2],'pass',output='sos')
+sos = signal.cheby1(3,rp,[f1/fs*2,f2/fs*2],'bandpass', output='sos')
+
+sos = np.round((sos) * scaling_factor)
+
+# print coefficients
+for biquad in sos:
+    for coeff in biquad:
+        print(int(coeff),",",sep="",end="")
+        #print((coeff),",",sep="",end="")
+    print("")
+
+# plot the frequency response
+b,a = signal.sos2tf(sos)
+w,h = signal.freqz(b,a)
+pl.plot(w/np.pi/2*fs,20*np.log(np.abs(h)))
+pl.xlabel('frequency/Hz');
+pl.ylabel('gain/dB');
+pl.ylim(top=1,bottom=-20);
+pl.xlim(left=250, right=12000);
+pl.show()