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Teensy-reSID/reSID/filter.h

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// ---------------------------------------------------------------------------
// This file is part of reSID, a MOS6581 SID emulator engine.
// Copyright (C) 2004 Dag Lem <resid@nimrod.no>
//
// 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
// ---------------------------------------------------------------------------
#ifndef __FILTER_H__
#define __FILTER_H__
#include "siddefs.h"
//#include "spline.h"
#include "filter6581.h"
RESID_NAMESPACE_START
// ----------------------------------------------------------------------------
// The SID filter is modeled with a two-integrator-loop biquadratic filter,
// which has been confirmed by Bob Yannes to be the actual circuit used in
// the SID chip.
//
// Measurements show that excellent emulation of the SID filter is achieved,
// except when high resonance is combined with high sustain levels.
// In this case the SID op-amps are performing less than ideally and are
// causing some peculiar behavior of the SID filter. This however seems to
// have more effect on the overall amplitude than on the color of the sound.
//
// The theory for the filter circuit can be found in "Microelectric Circuits"
// by Adel S. Sedra and Kenneth C. Smith.
// The circuit is modeled based on the explanation found there except that
// an additional inverter is used in the feedback from the bandpass output,
// allowing the summer op-amp to operate in single-ended mode. This yields
// inverted filter outputs with levels independent of Q, which corresponds with
// the results obtained from a real SID.
//
// We have been able to model the summer and the two integrators of the circuit
// to form components of an IIR filter.
// Vhp is the output of the summer, Vbp is the output of the first integrator,
// and Vlp is the output of the second integrator in the filter circuit.
//
// According to Bob Yannes, the active stages of the SID filter are not really
// op-amps. Rather, simple NMOS inverters are used. By biasing an inverter
// into its region of quasi-linear operation using a feedback resistor from
// input to output, a MOS inverter can be made to act like an op-amp for
// small signals centered around the switching threshold.
//
// Qualified guesses at SID filter schematics are depicted below.
//
// SID filter
// ----------
//
// -----------------------------------------------
// | |
// | ---Rq-- |
// | | | |
// | ------------<A]-----R1--------- |
// | | | |
// | | ---C---| ---C---|
// | | | | | |
// | --R1-- ---R1-- |---Rs--| |---Rs--|
// | | | | | | | |
// ----R1--|-----[A>--|--R-----[A>--|--R-----[A>--|
// | | | |
// vi -----R1-- | | |
//
// vhp vbp vlp
//
//
// vi - input voltage
// vhp - highpass output
// vbp - bandpass output
// vlp - lowpass output
// [A> - op-amp
// R1 - summer resistor
// Rq - resistor array controlling resonance (4 resistors)
// R - NMOS FET voltage controlled resistor controlling cutoff frequency
// Rs - shunt resitor
// C - capacitor
//
//
//
// SID integrator
// --------------
//
// V+
//
// |
// |
// -----|
// | |
// | ||--
// -||
// ---C--- ||->
// | | |
// |---Rs-----------|---- vo
// | |
// | ||--
// vi ---- -----|------------||
// | ^ | ||->
// |___| | |
// ----- | |
// | | |
// |---R2-- |
// |
// R1 V-
// |
// |
//
// Vw
//
// ----------------------------------------------------------------------------
class Filter
{
public:
Filter();
void enable_filter(bool enable);
// void set_chip_model(chip_model model);
RESID_INLINE
void clock(sound_sample voice1, sound_sample voice2, sound_sample voice3,
sound_sample ext_in);
RESID_INLINE
void clock(cycle_count delta_t,
sound_sample voice1, sound_sample voice2, sound_sample voice3,
sound_sample ext_in);
void reset();
// Write registers.
void writeFC_LO(reg8);
void writeFC_HI(reg8);
void writeRES_FILT(reg8);
void writeMODE_VOL(reg8);
// SID audio output (16 bits).
sound_sample output();
// Spline functions.
// void fc_default(const fc_point*& points, int& count);
// PointPlotter<sound_sample> fc_plotter();
protected:
void set_w0();
void set_Q();
// Filter enabled.
bool enabled;
// Filter cutoff frequency.
reg12 fc;
// Filter resonance.
reg8 res;
// Selects which inputs to route through filter.
reg8 filt;
// Switch voice 3 off.
reg8 voice3off;
// Highpass, bandpass, and lowpass filter modes.
reg8 hp_bp_lp;
// Output master volume.
reg4 vol;
// Mixer DC offset.
sound_sample mixer_DC;
// State of filter.
sound_sample Vhp; // highpass
sound_sample Vbp; // bandpass
sound_sample Vlp; // lowpass
sound_sample Vnf; // not filtered
// Cutoff frequency, resonance.
sound_sample w0, w0_ceil_1, w0_ceil_dt;
sound_sample _1024_div_Q;
// Cutoff frequency tables.
// FC is an 11 bit register.
//sound_sample f0_6581[2048];
//sound_sample f0_8580[2048];
//sound_sample* f0;
//const sound_sample* f0 = filter6581;
const short* f0 = filter6581;
//const static fc_point f0_points_6581[];
//const static fc_point f0_points_8580[];
//const fc_point* f0_points;
//int f0_count;
friend class SID;
};
// ----------------------------------------------------------------------------
// Inline functions.
// The following functions are defined inline because they are called every
// time a sample is calculated.
// ----------------------------------------------------------------------------
#if RESID_INLINING || defined(__FILTER_CC__)
// ----------------------------------------------------------------------------
// SID clocking - 1 cycle.
// ----------------------------------------------------------------------------
RESID_INLINE
void Filter::clock(sound_sample voice1,
sound_sample voice2,
sound_sample voice3,
sound_sample ext_in)
{
// Scale each voice down from 20 to 13 bits.
voice1 >>= 7;
voice2 >>= 7;
// NB! Voice 3 is not silenced by voice3off if it is routed through
// the filter.
if (voice3off && !(filt & 0x04)) {
voice3 = 0;
}
else {
voice3 >>= 7;
}
ext_in >>= 7;
// This is handy for testing.
if (!enabled) {
Vnf = voice1 + voice2 + voice3 + ext_in;
Vhp = Vbp = Vlp = 0;
return;
}
// Route voices into or around filter.
// The code below is expanded to a switch for faster execution.
// (filt1 ? Vi : Vnf) += voice1;
// (filt2 ? Vi : Vnf) += voice2;
// (filt3 ? Vi : Vnf) += voice3;
sound_sample Vi;
switch (filt) {
default:
case 0x0:
Vi = 0;
Vnf = voice1 + voice2 + voice3 + ext_in;
break;
case 0x1:
Vi = voice1;
Vnf = voice2 + voice3 + ext_in;
break;
case 0x2:
Vi = voice2;
Vnf = voice1 + voice3 + ext_in;
break;
case 0x3:
Vi = voice1 + voice2;
Vnf = voice3 + ext_in;
break;
case 0x4:
Vi = voice3;
Vnf = voice1 + voice2 + ext_in;
break;
case 0x5:
Vi = voice1 + voice3;
Vnf = voice2 + ext_in;
break;
case 0x6:
Vi = voice2 + voice3;
Vnf = voice1 + ext_in;
break;
case 0x7:
Vi = voice1 + voice2 + voice3;
Vnf = ext_in;
break;
case 0x8:
Vi = ext_in;
Vnf = voice1 + voice2 + voice3;
break;
case 0x9:
Vi = voice1 + ext_in;
Vnf = voice2 + voice3;
break;
case 0xa:
Vi = voice2 + ext_in;
Vnf = voice1 + voice3;
break;
case 0xb:
Vi = voice1 + voice2 + ext_in;
Vnf = voice3;
break;
case 0xc:
Vi = voice3 + ext_in;
Vnf = voice1 + voice2;
break;
case 0xd:
Vi = voice1 + voice3 + ext_in;
Vnf = voice2;
break;
case 0xe:
Vi = voice2 + voice3 + ext_in;
Vnf = voice1;
break;
case 0xf:
Vi = voice1 + voice2 + voice3 + ext_in;
Vnf = 0;
break;
}
// delta_t = 1 is converted to seconds given a 1MHz clock by dividing
// with 1 000 000.
// Calculate filter outputs.
// Vhp = Vbp/Q - Vlp - Vi;
// dVbp = -w0*Vhp*dt;
// dVlp = -w0*Vbp*dt;
sound_sample dVbp = (w0_ceil_1*Vhp >> 20);
sound_sample dVlp = (w0_ceil_1*Vbp >> 20);
Vbp -= dVbp;
Vlp -= dVlp;
Vhp = (Vbp*_1024_div_Q >> 10) - Vlp - Vi;
}
// ----------------------------------------------------------------------------
// SID clocking - delta_t cycles.
// ----------------------------------------------------------------------------
RESID_INLINE
void Filter::clock(cycle_count delta_t,
sound_sample voice1,
sound_sample voice2,
sound_sample voice3,
sound_sample ext_in)
{
// Scale each voice down from 20 to 13 bits.
voice1 >>= 7;
voice2 >>= 7;
// NB! Voice 3 is not silenced by voice3off if it is routed through
// the filter.
if (voice3off && !(filt & 0x04)) {
voice3 = 0;
}
else {
voice3 >>= 7;
}
ext_in >>= 7;
// Enable filter on/off.
// This is not really part of SID, but is useful for testing.
// On slow CPUs it may be necessary to bypass the filter to lower the CPU
// load.
if (!enabled) {
Vnf = voice1 + voice2 + voice3 + ext_in;
Vhp = Vbp = Vlp = 0;
return;
}
// Route voices into or around filter.
// The code below is expanded to a switch for faster execution.
// (filt1 ? Vi : Vnf) += voice1;
// (filt2 ? Vi : Vnf) += voice2;
// (filt3 ? Vi : Vnf) += voice3;
sound_sample Vi;
switch (filt) {
default:
case 0x0:
Vi = 0;
Vnf = voice1 + voice2 + voice3 + ext_in;
break;
case 0x1:
Vi = voice1;
Vnf = voice2 + voice3 + ext_in;
break;
case 0x2:
Vi = voice2;
Vnf = voice1 + voice3 + ext_in;
break;
case 0x3:
Vi = voice1 + voice2;
Vnf = voice3 + ext_in;
break;
case 0x4:
Vi = voice3;
Vnf = voice1 + voice2 + ext_in;
break;
case 0x5:
Vi = voice1 + voice3;
Vnf = voice2 + ext_in;
break;
case 0x6:
Vi = voice2 + voice3;
Vnf = voice1 + ext_in;
break;
case 0x7:
Vi = voice1 + voice2 + voice3;
Vnf = ext_in;
break;
case 0x8:
Vi = ext_in;
Vnf = voice1 + voice2 + voice3;
break;
case 0x9:
Vi = voice1 + ext_in;
Vnf = voice2 + voice3;
break;
case 0xa:
Vi = voice2 + ext_in;
Vnf = voice1 + voice3;
break;
case 0xb:
Vi = voice1 + voice2 + ext_in;
Vnf = voice3;
break;
case 0xc:
Vi = voice3 + ext_in;
Vnf = voice1 + voice2;
break;
case 0xd:
Vi = voice1 + voice3 + ext_in;
Vnf = voice2;
break;
case 0xe:
Vi = voice2 + voice3 + ext_in;
Vnf = voice1;
break;
case 0xf:
Vi = voice1 + voice2 + voice3 + ext_in;
Vnf = 0;
break;
}
// Maximum delta cycles for the filter to work satisfactorily under current
// cutoff frequency and resonance constraints is approximately 8.
cycle_count delta_t_flt = 8;
while (delta_t) {
if (delta_t < delta_t_flt) {
delta_t_flt = delta_t;
}
// delta_t is converted to seconds given a 1MHz clock by dividing
// with 1 000 000. This is done in two operations to avoid integer
// multiplication overflow.
// Calculate filter outputs.
// Vhp = Vbp/Q - Vlp - Vi;
// dVbp = -w0*Vhp*dt;
// dVlp = -w0*Vbp*dt;
sound_sample w0_delta_t = w0_ceil_dt * delta_t_flt >> 6;
sound_sample dVbp = (w0_delta_t*Vhp >> 14);
sound_sample dVlp = (w0_delta_t*Vbp >> 14);
Vbp -= dVbp;
Vlp -= dVlp;
Vhp = (Vbp*_1024_div_Q >> 10) - Vlp - Vi;
delta_t -= delta_t_flt;
}
}
// ----------------------------------------------------------------------------
// SID audio output (20 bits).
// ----------------------------------------------------------------------------
RESID_INLINE
sound_sample Filter::output()
{
// This is handy for testing.
if (!enabled) {
return (Vnf + mixer_DC)*static_cast<sound_sample>(vol);
}
// Mix highpass, bandpass, and lowpass outputs. The sum is not
// weighted, this can be confirmed by sampling sound output for
// e.g. bandpass, lowpass, and bandpass+lowpass from a SID chip.
// The code below is expanded to a switch for faster execution.
// if (hp) Vf += Vhp;
// if (bp) Vf += Vbp;
// if (lp) Vf += Vlp;
sound_sample Vf;
switch (hp_bp_lp) {
default:
case 0x0:
Vf = 0;
break;
case 0x1:
Vf = Vlp;
break;
case 0x2:
Vf = Vbp;
break;
case 0x3:
Vf = Vlp + Vbp;
break;
case 0x4:
Vf = Vhp;
break;
case 0x5:
Vf = Vlp + Vhp;
break;
case 0x6:
Vf = Vbp + Vhp;
break;
case 0x7:
Vf = Vlp + Vbp + Vhp;
break;
}
// Sum non-filtered and filtered output.
// Multiply the sum with volume.
return (Vnf + Vf + mixer_DC)*static_cast<sound_sample>(vol);
}
#endif // RESID_INLINING || defined(__FILTER_CC__)
RESID_NAMESPACE_STOP
#endif // not __FILTER_H__