dirkwhoffmann
commented
1 year ago For reference: Here is the setup code for all three filters in the filter pipeline:
double R, C, R1, R2, C1, C2, cutoff, qfactor;
if (amigaModel == MODEL_A500)
{
// A500 1-pole (6db/oct) RC low-pass filter:
R = 360.0; // R321 (360 ohm)
C = 1e-7; // C321 (0.1uF)
cutoff = 1.0 / (PT2_TWO_PI * R * C); // ~4420.971Hz
setupOnePoleFilter(dPaulaOutputFreq, cutoff, &filterLo);
// A500 1-pole (6dB/oct) RC high-pass filter:
R = 1390.0; // R324 (1K ohm) + R325 (390 ohm)
C = 2.233e-5; // C334 (22uF) + C335 (0.33uF)
cutoff = 1.0 / (PT2_TWO_PI * R * C); // ~5.128Hz
setupOnePoleFilter(dPaulaOutputFreq, cutoff, &filterHi);
}
else
{
/* Don't use the A1200 low-pass filter since its cutoff
** is well above human hearable range anyway (~34.4kHz).
** We don't do volume PWM, so we have nothing we need to
** filter away.
*/
useLowpassFilter = false;
// A1200 1-pole (6dB/oct) RC high-pass filter:
R = 1360.0; // R324 (1K ohm resistor) + R325 (360 ohm resistor)
C = 2.2e-5; // C334 (22uF capacitor)
cutoff = 1.0 / (PT2_TWO_PI * R * C); // ~5.319Hz
setupOnePoleFilter(dPaulaOutputFreq, cutoff, &filterHi);
}
// 2-pole (12dB/oct) RC low-pass filter ("LED" filter, same values on A500/A1200):
R1 = 10000.0; // R322 (10K ohm)
R2 = 10000.0; // R323 (10K ohm)
C1 = 6.8e-9; // C322 (6800pF)
C2 = 3.9e-9; // C323 (3900pF)
cutoff = 1.0 / (PT2_TWO_PI * pt2_sqrt(R1 * R2 * C1 * C2)); // ~3090.533Hz
qfactor = pt2_sqrt(R1 * R2 * C1 * C2) / (C2 * (R1 + R2)); // ~0.660225
setupTwoPoleFilter(dPaulaOutputFreq, cutoff, qfactor, &filterLED);Here are the implementations of the one pole and the two pole filter:
#include "pt2_math.h"
#include "pt2_rcfilters.h"
#define SMALL_NUMBER (1E-4)
/* 1-pole RC low-pass/high-pass filter, based on:
** https://www.musicdsp.org/en/latest/Filters/116-one-pole-lp-and-hp.html
*/
void setupOnePoleFilter(double audioRate, double cutOff, onePoleFilter_t *f)
{
if (cutOff >= audioRate/2.0)
cutOff = (audioRate/2.0) - SMALL_NUMBER;
const double a = 2.0 - pt2_cos((PT2_TWO_PI * cutOff) / audioRate);
const double b = a - pt2_sqrt((a * a) - 1.0);
f->a1 = 1.0 - b;
f->a2 = b;
}
void clearOnePoleFilterState(onePoleFilter_t *f)
{
f->tmpL = f->tmpR = 0.0;
}
void onePoleLPFilter(onePoleFilter_t *f, const double in, double *out)
{
f->tmpL = (f->a1 * in) + (f->a2 * f->tmpL);
*out = f->tmpL;
}
void onePoleLPFilterStereo(onePoleFilter_t *f, const double *in, double *out)
{
// left channel
f->tmpL = (f->a1 * in[0]) + (f->a2 * f->tmpL);
out[0] = f->tmpL;
// right channel
f->tmpR = (f->a1 * in[1]) + (f->a2 * f->tmpR);
out[1] = f->tmpR;
}
void onePoleHPFilter(onePoleFilter_t *f, const double in, double *out)
{
f->tmpL = (f->a1 * in) + (f->a2 * f->tmpL);
*out = in - f->tmpL;
}
void onePoleHPFilterStereo(onePoleFilter_t *f, const double *in, double *out)
{
// left channel
f->tmpL = (f->a1 * in[0]) + (f->a2 * f->tmpL);
out[0] = in[0] - f->tmpL;
// right channel
f->tmpR = (f->a1 * in[1]) + (f->a2 * f->tmpR);
out[1] = in[1] - f->tmpR;
}
/* 2-pole RC low-pass filter with Q factor, based on:
** https://www.musicdsp.org/en/latest/Filters/38-lp-and-hp-filter.html
*/
void setupTwoPoleFilter(double audioRate, double cutOff, double qFactor, twoPoleFilter_t *f)
{
if (cutOff >= audioRate/2.0)
cutOff = (audioRate/2.0) - SMALL_NUMBER;
const double a = 1.0 / pt2_tan((PT2_PI * cutOff) / audioRate);
const double b = 1.0 / qFactor;
f->a1 = 1.0 / (1.0 + b * a + a * a);
f->a2 = 2.0 * f->a1;
f->b1 = 2.0 * (1.0 - a*a) * f->a1;
f->b2 = (1.0 - b * a + a * a) * f->a1;
}
void clearTwoPoleFilterState(twoPoleFilter_t *f)
{
f->tmpL[0] = f->tmpL[1] = f->tmpL[2] = f->tmpL[3] = 0.0;
f->tmpR[0] = f->tmpR[1] = f->tmpR[2] = f->tmpR[3] = 0.0;
}
void twoPoleLPFilter(twoPoleFilter_t *f, const double in, double *out)
{
const double LOut = (f->a1 * in) + (f->a2 * f->tmpL[0]) + (f->a1 * f->tmpL[1]) - (f->b1 * f->tmpL[2]) - (f->b2 * f->tmpL[3]);
// shift states
f->tmpL[1] = f->tmpL[0];
f->tmpL[0] = in;
f->tmpL[3] = f->tmpL[2];
f->tmpL[2] = LOut;
// set output
*out = LOut;
}
void twoPoleLPFilterStereo(twoPoleFilter_t *f, const double *in, double *out)
{
const double LOut = (f->a1 * in[0]) + (f->a2 * f->tmpL[0]) + (f->a1 * f->tmpL[1]) - (f->b1 * f->tmpL[2]) - (f->b2 * f->tmpL[3]);
const double ROut = (f->a1 * in[1]) + (f->a2 * f->tmpR[0]) + (f->a1 * f->tmpR[1]) - (f->b1 * f->tmpR[2]) - (f->b2 * f->tmpR[3]);
// shift states
f->tmpL[1] = f->tmpL[0];
f->tmpL[0] = in[0];
f->tmpL[3] = f->tmpL[2];
f->tmpL[2] = LOut;
f->tmpR[1] = f->tmpR[0];
f->tmpR[0] = in[1];
f->tmpR[3] = f->tmpR[2];
f->tmpR[2] = ROut;
// set output
out[0] = LOut;
out[1] = ROut;
}
mras0
Now that you mention filtering, one thing I noticed is there's no filtering applied when the LED is off in vAmiga (which is the case for this test). According to https://eab.abime.net/showthread.php?t=112931 (and UAE source code) there are always low- and high-pass filter applied on A500. WinUAE doesn't seem to implement the high pass filter, but pt2-clone does (and has more information).
This is just info for you consideration, not something I expect you to do anything about :)
For reference the first couple of seconds form WinUAE & vAmigaSDL. For the latter I set the samplerate to 48000 and saved the audio output to a raw file containing (32-bit float stereo). In both cases I've normalized the output with audacity to make sure volume is the same. BeeMoved_samples.zip
Originally posted by @mras0 in https://github.com/dirkwhoffmann/vAmiga/issues/788#issuecomment-1485517848