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authorMichele Calgaro <michele.calgaro@yahoo.it>2020-12-08 22:26:17 +0900
committerMichele Calgaro <michele.calgaro@yahoo.it>2020-12-08 22:26:17 +0900
commitfce86b22a2367f1be1f9aae5e1ba3d18d1371b74 (patch)
tree707fe84fef0569a152e632ce1e16407f9d19a3d2 /arts/modules/synth/synth_std_equalizer_impl.cpp
parent41fa1afc2c571b909acd0312e4eebb4a0b21e3c2 (diff)
downloadtdemultimedia-fce86b22a2367f1be1f9aae5e1ba3d18d1371b74.tar.gz
tdemultimedia-fce86b22a2367f1be1f9aae5e1ba3d18d1371b74.zip
Renaming of files in preparation for code style tools.
Signed-off-by: Michele Calgaro <michele.calgaro@yahoo.it>
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+/*
+
+ Copyright (C) 2000 Jeff Tranter
+ tranter@pobox.com
+
+ (C) 1999 Stefan Westerfeld
+ stefan@space.twc.de
+
+ (C) 1999 Martin Lorenz
+ lorenz@ch.tum.de
+
+ 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; either version 2 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 for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+
+ */
+
+#include <math.h>
+#include "artsmodulessynth.h"
+#include "stdsynthmodule.h"
+
+using namespace Arts;
+
+class Synth_STD_EQUALIZER_impl : virtual public Synth_STD_EQUALIZER_skel,
+ virtual public StdSynthModule
+{
+protected:
+ float _low, _mid, _high, _frequency, _q;
+ float tlow, tmid, thigh, tfrequency;
+ float a1, a2, b0, b1, b2, x_0, x_1, x_2, y_1, y_2;
+ unsigned long all;
+
+public:
+ float low() { return _low; }
+ void low(float newLow)
+ {
+ if(newLow != _low)
+ {
+ _low = newLow;
+ calcParameters();
+ high_changed(newLow);
+ }
+ }
+
+
+ float mid() { return _mid; }
+ void mid(float newMid)
+ {
+ if(newMid != _mid)
+ {
+ _mid = newMid;
+ calcParameters();
+ mid_changed(newMid);
+ }
+ }
+
+ float high() { return _high; }
+ void high(float newHigh)
+ {
+ if(newHigh != _high)
+ {
+ _high = newHigh;
+ calcParameters();
+ high_changed(newHigh);
+ }
+ }
+
+
+ float frequency() { return _frequency; }
+ void frequency(float newFrequency)
+ {
+ if(newFrequency != _frequency)
+ {
+ _frequency = newFrequency;
+ calcParameters();
+ frequency_changed(newFrequency);
+ }
+ }
+
+ float q() { return _q; }
+ void q(float newQ)
+ {
+ if(newQ != _q)
+ {
+ _q = newQ;
+ calcParameters();
+ q_changed(newQ);
+ }
+ }
+
+ Synth_STD_EQUALIZER_impl() {
+ _low = _mid = _high = 0; _q = 0.5;
+ _frequency = 300;
+ }
+
+ void calcParameters()
+ {
+ /*
+
+ * _low, _mid, _high are in dB, transform them to tlow, tmid,
+ * thigh using:
+ * -6dB => 0.5 ; 0dB => 1 ; 6dB = 2.0 ; ...
+ */
+
+ tlow = exp(_low * 0.115524530093324); // exp(p[LOW]*ln(2)/6)
+ tmid = exp(_mid * 0.115524530093324);
+ thigh = exp(_high * 0.115524530093324);
+
+ // _frequency is given in Hz, we need the w-value (and do clipping if
+ // it exceeds SR/2)
+ const float SAMPLING_RATE = 44100.0;
+ tfrequency = _frequency;
+ if (tfrequency > SAMPLING_RATE / 2.01)
+ tfrequency = SAMPLING_RATE / 2.01;
+ float w = 2 * M_PI * tfrequency / SAMPLING_RATE;
+
+ // Calculations:
+ float t = 1/tan(w/2);
+ float tq = t/_q;
+ float t2 = t*t;
+
+ float a0 = 1+tq+t2;
+ float a0r = 1/a0;
+
+ // and now the real filter values:
+ a1 = (2 - 2 * t2) * a0r;
+ a2 = (1 - tq + t2) * a0r;
+ b0 = (tlow + tmid * tq + thigh * t2) * a0r;
+ b1 = (2 * tlow -2 * thigh * t2) * a0r;
+ b2 = (tlow - tmid * tq + thigh * t2) * a0r;
+
+ // TODO: try if we need that here, or if we can change filter
+ // coefficients without setting the state to 0
+ x_0 = x_1 = x_2 = y_1 = y_2 = 0.0;
+ all = 0;
+ }
+
+ void streamInit()
+ {
+ calcParameters();
+ }
+
+ void calculateBlock(unsigned long samples)
+ {
+ all += samples;
+
+ if (all > 1024)
+ {
+ /* The _problem_: (observed on a PII-350)
+ *
+ * I am not quite sure what happens here, but it seems to be like that:
+ *
+ * If an ordinary signal (a mp3 for instance) is sent through the
+ * equalizer, and then no more input is given (zeros as input),
+ * the y_1 and y_2 values oscillate for some time, coming closer and
+ * close to zero.
+ *
+ * But before the reach zero, they reach the smallest negative number
+ * (or smallest positive, or whatever), and stay there
+ * (because 0.005*smallest_negative will remain smallest_negative).
+ *
+ * Since then, the CPU usage for all operations on these floats
+ * increases, (since handling of smallest_negative seems to be a rare
+ * case).
+ *
+ * The _fix_:
+ *
+ * We observe the value of y_1. If it's very close to zero (may be as
+ * well smallest_positive/smallest_negative), we set it to zero,
+ * together with y_2. This shouldn't significantly influence
+ * correctness of the filter, but effectively solves the problem.
+ *
+ * If you don't believe me, try without this fix and tell me what
+ * happens on your computer.
+ */
+ const float zero_lower =-0.00000001;
+ const float zero_upper = 0.00000001;
+ all = 0;
+
+ if(zero_lower < y_1 && y_1 < zero_upper)
+ y_1 = y_2 = 0.0;
+ }
+
+ unsigned long i;
+ float tmp;
+ for (i=0; i<samples; i++)
+ {
+ x_0 = invalue[i];
+ tmp = x_0 * b0 + x_1 * b1 + x_2 * b2 - y_1 * a1 - y_2 * a2;
+ x_2 = x_1; x_1 = x_0; y_2 = y_1; y_1 = tmp;
+ outvalue[i] = tmp;
+ }
+ }
+
+};
+
+REGISTER_IMPLEMENTATION(Synth_STD_EQUALIZER_impl);