ref: 6fe2950c2bca6718a6d876d6fb626bb07a98c79d
parent: f94686bf4da0bc46cd4f08ca9aa420cc2d75779f
author: Olav Sørensen <olav.sorensen@live.no>
date: Fri Dec 18 14:56:04 EST 2020
A500 low-pass filter is now much closer to a real typical A500
--- a/src/pt2_audio.c
+++ b/src/pt2_audio.c
@@ -35,6 +35,7 @@
#include "pt2_structs.h"
#include "pt2_rcfilter.h"
#include "pt2_ledfilter.h"
+#include "pt2_downsamplers2x.h"
#define INITIAL_DITHER_SEED 0x12345000
@@ -46,9 +47,9 @@
static uint32_t audLatencyPerfValInt, audLatencyPerfValFrac;
static uint64_t tickTime64, tickTime64Frac;
static double *dMixBufferL, *dMixBufferR, *dMixBufferLUnaligned, *dMixBufferRUnaligned, dOldVoiceDelta, dOldVoiceDeltaMul;
-static double dPrngStateL, dPrngStateR;
+static double dPrngStateL, dPrngStateR, dLState[2], dRState[2];
static blep_t blep[AMIGA_VOICES], blepVol[AMIGA_VOICES];
-static rcFilter_t filterLoA500, filterLoA1200, filterHiA500, filterHiA1200;
+static rcFilter_t filterLoA500, filterHiA500, filterHiA1200;
static ledFilter_t filterLED;
static SDL_AudioDeviceID dev;
@@ -188,7 +189,6 @@
mixerKillVoice(i);
clearRCFilterState(&filterLoA500);
- clearRCFilterState(&filterLoA1200);
clearRCFilterState(&filterHiA500);
clearRCFilterState(&filterHiA1200);
clearLEDFilterState(&filterLED);
@@ -372,7 +372,6 @@
lockAudio();
clearRCFilterState(&filterLoA500);
- clearRCFilterState(&filterLoA1200);
clearRCFilterState(&filterHiA500);
clearRCFilterState(&filterHiA1200);
clearLEDFilterState(&filterLED);
@@ -465,6 +464,12 @@
dPrngStateR = 0.0;
}
+void resetAudioDownsamplingStates(void)
+{+ dLState[0] = dLState[1] = 0.0;
+ dRState[0] = dRState[1] = 0.0;
+}
+
static inline int32_t random32(void)
{// LCG random 32-bit generator (quite good and fast)
@@ -473,44 +478,56 @@
return randSeed;
}
-static inline void processMixedSamples(int32_t i, int16_t *out)
+static void processMixedSamples(int32_t i, int16_t *out)
{int32_t smp32;
- double dOut[2], dPrng;
+ double dPrng, dOut[2], dMixL[2], dMixR[2];
- dOut[0] = dMixBufferL[i];
- dOut[1] = dMixBufferR[i];
-
- if (filterModel == FILTERMODEL_A500)
+ // we run the filters at 2x the audio output rate for more precision
+ for (int32_t j = 0; j < 2; j++)
{- // A500 low-pass RC filter
- RCLowPassFilterStereo(&filterLoA500, dOut, dOut);
+ // zero-padding (yes, this makes sense)
+ dOut[0] = (j == 0) ? dMixBufferL[i] : 0.0;
+ dOut[1] = (j == 0) ? dMixBufferR[i] : 0.0;
- // "LED" Sallen-Key filter
- if (ledFilterEnabled)
- LEDFilter(&filterLED, dOut, dOut);
+ if (filterModel == FILTERMODEL_A500)
+ {+ // A500 low-pass RC filter
+ RCLowPassFilterStereo(&filterLoA500, dOut, dOut);
- // A500 high-pass RC filter
- RCHighPassFilterStereo(&filterHiA500, dOut, dOut);
- }
- else
- {- // A1200 low-pass RC filter
- if (audio.outputRate >= 96000) // cutoff is too high for 44.1kHz/48kHz
- RCLowPassFilterStereo(&filterLoA1200, dOut, dOut);
+ // "LED" Sallen-Key filter
+ if (ledFilterEnabled)
+ LEDFilter(&filterLED, dOut, dOut);
- // "LED" Sallen-Key filter
- if (ledFilterEnabled)
- LEDFilter(&filterLED, dOut, dOut);
+ // A500 high-pass RC filter
+ RCHighPassFilterStereo(&filterHiA500, dOut, dOut);
+ }
+ else
+ {+ // A1200 low-pass filter is ignored (we don't want it)
- // A1200 high-pass RC filter
- RCHighPassFilterStereo(&filterHiA1200, dOut, dOut);
+ // "LED" Sallen-Key filter
+ if (ledFilterEnabled)
+ LEDFilter(&filterLED, dOut, dOut);
+
+ // A1200 high-pass RC filter
+ RCHighPassFilterStereo(&filterHiA1200, dOut, dOut);
+ }
+
+ dMixL[j] = dOut[0];
+ dMixR[j] = dOut[1];
}
- // normalize and flip phase (A500/A1200 has a phase-inverted audio signal)
- dOut[0] *= -INT16_MAX / (double)AMIGA_VOICES;
- dOut[1] *= -INT16_MAX / (double)AMIGA_VOICES;
+#define NORMALIZE_DOWNSAMPLE 2.0
+ // 2x "all-pass halfband" downsampling
+ dOut[0] = d2x(dMixL, dLState);
+ dOut[1] = d2x(dMixR, dRState);
+
+ // normalize and invert phase (A500/A1200 has a phase-inverted audio signal)
+ dOut[0] *= NORMALIZE_DOWNSAMPLE * (-INT16_MAX / (double)AMIGA_VOICES);
+ dOut[1] *= NORMALIZE_DOWNSAMPLE * (-INT16_MAX / (double)AMIGA_VOICES);
+
// left channel - 1-bit triangular dithering (high-pass filtered)
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dOut[0] = (dOut[0] + dPrng) - dPrngStateL;
@@ -564,6 +581,7 @@
for (i = 0; i < numSamples; i++)
{processMixedSamples(i, out);
+
*outStream++ = out[0];
*outStream++ = out[1];
}
@@ -647,7 +665,7 @@
samplesToMix = remainingTick;
outputAudio(streamOut, samplesToMix);
- streamOut += samplesToMix << 1;
+ streamOut += samplesToMix<<1;
samplesLeft -= samplesToMix;
audio.dTickSampleCounter -= samplesToMix;
@@ -711,24 +729,48 @@
** - RC 6dB/oct high-pass: R=1390 ohm (1000+390), C=22uF (f=5.204Hz)
*/
+ // we run the filters at twice the frequency for improved precision (zero-padding)
+ const uint32_t audioFreq = audio.outputRate * 2;
+
double R, C, R1, R2, C1, C2, fc, fb;
const double pi = 4.0 * atan(1.0); // M_PI can not be trusted
+ /*
+ ** 8bitbubsy:
+ ** Hackish low-pass cutoff compensation to better match Amiga 500 when
+ ** we use "lower" audio output rates. This has been loosely hand-picked
+ ** after looking at many frequency analyses on a sine-sweep test module
+ ** rendered on 7 different Amiga 500 machines (and taking the average).
+ ** Don't try to make sense of this magic constant, and it should only be
+ ** used within this very specific application!
+ **
+ ** The reason we want this bias is because our digital RC filter is not
+ ** that precise at lower audio output rates. It would otherwise lead to a
+ ** slight unwanted cut of treble near the cutoff we aim for. It was easily
+ ** audible, and especially visible on a plotted frequency spectrum.
+ **
+ ** 1100Hz is the magic value I found that seems to be good. Higher than that
+ ** would allow too much treble to pass.
+ **
+ ** Scaling it like this is 'acceptable' (confirmed with further frequency analyses
+ ** at output rates of 48, 96 and 192).
+ */
+ double dLPCutoffBias = 1100.0 * (44100.0 / audio.outputRate);
+
// A500 1-pole (6db/oct) static RC low-pass filter:
R = 360.0; // R321 (360 ohm resistor)
C = 1e-7; // C321 (0.1uF capacitor)
- fc = 1.0 / (2.0 * pi * R * C);
- calcRCFilterCoeffs(audio.outputRate, fc, &filterLoA500);
+ fc = (1.0 / (2.0 * pi * R * C)) + dLPCutoffBias;
+ calcRCFilterCoeffs(audioFreq, fc, &filterLoA500);
+
+ /*
+ ** 8bitbubsy:
+ ** We don't handle Amiga 1200's ~34kHz low-pass filter as it's not really
+ ** needed. The reason it was still present in the A1200 (despite its high
+ ** non-audible cutoff) was to filter away high-frequency noise from Paula's
+ ** PWM (volume modulation). We don't do PWM for volume in the PT2 clone.
+ */
- // A1200 1-pole (6dB/oct) static RC low-pass filter:
- if (audio.outputRate >= 96000) // cutoff is too high for 44.1kHz/48kHz
- {- R = 680.0; // R321 (680 ohm resistor)
- C = 6.8e-9; // C321 (6800pf capacitor)
- fc = 1.0 / (2.0 * pi * R * C);
- calcRCFilterCoeffs(audio.outputRate, fc, &filterLoA1200);
- }
-
// Sallen-Key filter ("LED" filter, same RC values on A500 and A1200):R1 = 10000.0; // R322 (10K ohm resistor)
R2 = 10000.0; // R323 (10K ohm resistor)
@@ -736,19 +778,19 @@
C2 = 3.9e-9; // C323 (3900pF capacitor)
fc = 1.0 / (2.0 * pi * sqrt(R1 * R2 * C1 * C2));
fb = 0.125; // Fb = 0.125 : Q ~= 1/sqrt(2)
- calcLEDFilterCoeffs(audio.outputRate, fc, fb, &filterLED);
+ calcLEDFilterCoeffs(audioFreq, fc, fb, &filterLED);
// A500 1-pole (6dB/oct) static RC high-pass filter:
R = 1390.0; // R324 (1K ohm resistor) + R325 (390 ohm resistor)
C = 2.233e-5; // C334 (22uF capacitor) + C335 (0.33�F capacitor)
fc = 1.0 / (2.0 * pi * R * C);
- calcRCFilterCoeffs(audio.outputRate, fc, &filterHiA500);
+ calcRCFilterCoeffs(audioFreq, fc, &filterHiA500);
// A1200 1-pole (6dB/oct) static RC high-pass filter:
R = 1390.0; // R324 (1K ohm resistor) + R325 (390 ohm resistor)
C = 2.2e-5; // C334 (22uF capacitor)
fc = 1.0 / (2.0 * pi * R * C);
- calcRCFilterCoeffs(audio.outputRate, fc, &filterHiA1200);
+ calcRCFilterCoeffs(audioFreq, fc, &filterHiA1200);
}
void recalcFilterCoeffs(int32_t outputRate) // for MOD2WAV
@@ -761,7 +803,6 @@
audio.outputRate = outputRate;
clearRCFilterState(&filterLoA500);
- clearRCFilterState(&filterLoA1200);
clearRCFilterState(&filterHiA500);
clearRCFilterState(&filterHiA1200);
clearLEDFilterState(&filterLED);
@@ -904,8 +945,8 @@
const int32_t maxSamplesToMix = MAX(pat2SmpMaxSamples, MAX(mod2WavMaxSamples, renderMaxSamples));
- dMixBufferLUnaligned = (double *)MALLOC_PAD(maxSamplesToMix * sizeof (double), 256);
- dMixBufferRUnaligned = (double *)MALLOC_PAD(maxSamplesToMix * sizeof (double), 256);
+ dMixBufferLUnaligned = (double *)MALLOC_PAD(maxSamplesToMix * sizeof (double) * 8, 256);
+ dMixBufferRUnaligned = (double *)MALLOC_PAD(maxSamplesToMix * sizeof (double) * 8, 256);
if (dMixBufferLUnaligned == NULL || dMixBufferRUnaligned == NULL)
{@@ -928,6 +969,7 @@
calcAudioLatencyVars(audio.audioBufferSize, audio.outputRate);
+ resetAudioDownsamplingStates();
audio.resetSyncTickTimeFlag = true;
SDL_PauseAudioDevice(dev, false);
return true;
--- a/src/pt2_audio.h
+++ b/src/pt2_audio.h
@@ -72,6 +72,7 @@
void turnOffVoices(void);
void mixerCalcVoicePans(uint8_t stereoSeparation);
void outputAudio(int16_t *target, int32_t numSamples);
+void resetAudioDownsamplingStates(void);
extern audio_t audio; // pt2_audio.c
extern paulaVoice_t paula[AMIGA_VOICES]; // pt2_audio.c
--- a/src/pt2_downsample2x.c
+++ b/src/pt2_downsample2x.c
@@ -25,7 +25,7 @@
return out;
}
-static double d2x(const double *input, double *b)
+double d2x(const double *input, double *b)
{return (f(input[0], &b[0], 0.150634765625) + f(input[1], &b[1], -0.3925628662109375)) * 0.5;
}
--- a/src/pt2_downsamplers2x.h
+++ b/src/pt2_downsamplers2x.h
@@ -4,6 +4,8 @@
// all-pass halfband filters
+double d2x(const double *input, double *b);
+
// Warning: These can exceed -1.0 .. 1.0 because of undershoot/overshoot!
void downsample2xFloat(float *buffer, int32_t originalLength);
--- a/src/pt2_mod2wav.c
+++ b/src/pt2_mod2wav.c
@@ -123,6 +123,8 @@
fwrite(&wavHeader, sizeof (wavHeader_t), 1, f);
fclose(f);
+ resetAudioDownsamplingStates();
+
ui.mod2WavFinished = true;
ui.updateMod2WavDialog = true;