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release-0.1
Frank Bösing 9 years ago
parent 458ee52f57
commit 00654c5814
7 changed files with 92 additions and 199 deletions
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  1. 4
      reSID/extfilt.cc
  2. 2
      reSID/extfilt.h
  3. 10
      reSID/filter.cc
  4. 2
      reSID/filter.h
  5. 120
      reSID/reSID.ino
  6. 135
      reSID/sid.cc
  7. 18
      reSID/sid.h

4
reSID/extfilt.cc
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@ -46,9 +46,9 @@ void ExternalFilter::enable_filter(bool enable)
// ----------------------------------------------------------------------------
// Setup of the external filter sampling parameters.
// ----------------------------------------------------------------------------
void ExternalFilter::set_sampling_parameter(double pass_freq)
void ExternalFilter::set_sampling_parameter(float pass_freq)
{
static const double pi = 3.1415926535897932385;
static const float pi = 3.1415926535897932385;
// Low-pass: R = 10kOhm, C = 1000pF; w0l = 1/RC = 1/(1e4*1e-9) = 100000
// High-pass: R = 1kOhm, C = 10uF; w0h = 1/RC = 1/(1e3*1e-5) = 100

2
reSID/extfilt.h
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@ -43,7 +43,7 @@ public:
ExternalFilter();
void enable_filter(bool enable);
void set_sampling_parameter(double pass_freq);
void set_sampling_parameter(float pass_freq);
void set_chip_model(chip_model model);
RESID_INLINE void clock(sound_sample Vi);

10
reSID/filter.cc
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@ -252,18 +252,18 @@ void Filter::writeMODE_VOL(reg8 mode_vol)
// Set filter cutoff frequency.
void Filter::set_w0()
{
const double pi = 3.1415926535897932385;
const float pi = 3.1415926535897932385;
// Multiply with 1.048576 to facilitate division by 1 000 000 by right-
// shifting 20 times (2 ^ 20 = 1048576).
w0 = static_cast<sound_sample>(2*pi*f0[fc]*1.048576);
w0 = static_cast<sound_sample>(2.0*pi*f0[fc]*1.048576);
// Limit f0 to 16kHz to keep 1 cycle filter stable.
const sound_sample w0_max_1 = static_cast<sound_sample>(2*pi*16000*1.048576);
const sound_sample w0_max_1 = static_cast<sound_sample>(2.0*pi*16000.0*1.048576);
w0_ceil_1 = w0 <= w0_max_1 ? w0 : w0_max_1;
// Limit f0 to 4kHz to keep delta_t cycle filter stable.
const sound_sample w0_max_dt = static_cast<sound_sample>(2*pi*4000*1.048576);
const sound_sample w0_max_dt = static_cast<sound_sample>(2.0*pi*4000.0*1.048576);
w0_ceil_dt = w0 <= w0_max_dt ? w0 : w0_max_dt;
}
@ -276,7 +276,7 @@ void Filter::set_Q()
// The coefficient 1024 is dispensed of later by right-shifting 10 times
// (2 ^ 10 = 1024).
_1024_div_Q = static_cast<sound_sample>(1024.0/(0.707 + 1.0*res/0x0f));
_1024_div_Q = static_cast<sound_sample>(1024.0/(0.707 + 1.0*res/15.0));
}
// ----------------------------------------------------------------------------

2
reSID/filter.h
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@ -460,7 +460,7 @@ void Filter::clock(cycle_count delta_t,
// 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 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);

120
reSID/reSID.ino
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@ -1,120 +0,0 @@
//#include "tower.h"
#include <Audio.h>
#include <Wire.h>
#include <SPI.h>
#include <SD.h>
#include <SerialFlash.h>
#include "reSID.h"
#define DMPDIR "/dmp"
//#define DMP "/dmp/GHOSTB~1.DMP"
//#define DMP "/dmp/LAX_UP.DMP"
#define DMP "/dmp/MR_MAR~1.DMP"
// GUItool: begin automatically generated code
AudioPlaySID playSID; //xy=189,110
AudioOutputI2S i2s1; //xy=366,111
AudioConnection patchCord1(playSID, 0, i2s1, 0);
AudioConnection patchCord2(playSID, 0, i2s1, 1);
AudioControlSGTL5000 sgtl5000_1; //xy=354,176
// GUItool: end automatically generated code
const int SDchipSelect = 10;
File directory;
File myfile;
char buffer[26];
char oldbuffer[26];
void printDirectory(File dir, int numTabs) {
while(true) {
File entry = dir.openNextFile();
if (! entry) {
// no more files
//Serial.println("**nomorefiles**");
break;
}
for (uint8_t i=0; i<numTabs; i++) {
Serial.print('\t');
}
Serial.print(entry.name());
if (entry.isDirectory()) {
Serial.println("/");
printDirectory(entry, numTabs+1);
} else {
// files have sizes, directories do not
Serial.print("\t\t");
Serial.println(entry.size(), DEC);
}
entry.close();
}
}
void setup() {
#if USETOWER
initTower();
#endif
AudioMemory(10);
SPI.setSCK(14);
SPI.setMOSI(7);
SPI.setMISO(12);
SPI.begin();
sgtl5000_1.enable();
sgtl5000_1.volume(0.9);
sgtl5000_1.enhanceBassEnable();
sgtl5000_1.enhanceBass(0.5, 2.5);
memset(buffer,0,sizeof(buffer));
memset(oldbuffer,0,sizeof(oldbuffer));
int ml = millis();
while (!Serial && (millis()-ml < 1000)) ; // wait for Arduino Serial Monitor
Serial.print("Initializing SD card...");
if (!SD.begin(SDchipSelect)) {
Serial.println("initialization failed!");
return;
}
Serial.println("initialization done.");
//directory = SD.open(DMPDIR);
//printDirectory(directory, 1);
myfile = SD.open(DMP, FILE_READ);
}
void loop() {
static int m = millis();
if (millis()-m < 20) return;
m = millis();
if (!myfile.available()) return;
for(int i=0;i<25;i++) {
if(buffer[i] != oldbuffer[i]) {
playSID.setreg(i, buffer[i]);
oldbuffer[i] = buffer[i];
}
}
myfile.read(buffer, 25);
}

135
reSID/sid.cc
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@ -19,7 +19,7 @@
#include "sid.h"
#include <math.h>
#include <AudioStream.h>
RESID_NAMESPACE_START
// Resampling constants.
@ -52,7 +52,7 @@ SID::SID()
voice[1].set_sync_source(&voice[0]);
voice[2].set_sync_source(&voice[1]);
set_sampling_parameters(985248, SAMPLE_FAST, 44100);
set_sampling_parameters(985248, SAMPLE_FAST, AUDIO_SAMPLE_RATE_EXACT);
bus_value = 0;
bus_value_ttl = 0;
@ -76,9 +76,9 @@ SID::~SID()
// ----------------------------------------------------------------------------
void SID::set_chip_model(chip_model model)
{
for (int i = 0; i < 3; i++) {
voice[i].set_chip_model(model);
}
voice[0].set_chip_model(model);
voice[1].set_chip_model(model);
voice[2].set_chip_model(model);
filter.set_chip_model(model);
extfilt.set_chip_model(model);
@ -90,9 +90,11 @@ void SID::set_chip_model(chip_model model)
// ----------------------------------------------------------------------------
void SID::reset()
{
for (int i = 0; i < 3; i++) {
voice[i].reset();
}
voice[0].reset();
voice[1].reset();
voice[2].reset();
filter.reset();
extfilt.reset();
@ -121,14 +123,18 @@ void SID::input(int sample)
int SID::output()
{
const int range = 1 << 16;
const int half = range >> 1;
//const int half = range >> 1;
int sample = extfilt.output()/((4095*255 >> 7)*3*15*2/range);
asm ("ssat %0, #16, %1" : "=r" (sample) : "r" (sample));
/*
if (sample >= half) {
return half - 1;
}
if (sample < -half) {
return -half;
}
*/
return sample;
}
@ -422,12 +428,12 @@ void SID::enable_external_filter(bool enable)
// I0() computes the 0th order modified Bessel function of the first kind.
// This function is originally from resample-1.5/filterkit.c by J. O. Smith.
// ----------------------------------------------------------------------------
double SID::I0(double x)
float SID::I0(float x)
{
// Max error acceptable in I0.
const double I0e = 1e-6;
const float I0e = 1e-6;
double sum, u, halfx, temp;
float sum, u, halfx, temp;
int n;
sum = u = n = 1;
@ -465,9 +471,9 @@ double SID::I0(double x)
// to slightly below 20kHz. This constraint ensures that the FIR table is
// not overfilled.
// ----------------------------------------------------------------------------
bool SID::set_sampling_parameters(double clock_freq, sampling_method method,
double sample_freq, double pass_freq,
double filter_scale)
bool SID::set_sampling_parameters(float clock_freq, sampling_method method,
float sample_freq, float pass_freq,
float filter_scale)
{
// Check resampling constraints.
if (method == SAMPLE_RESAMPLE_INTERPOLATE || method == SAMPLE_RESAMPLE_FAST)
@ -482,12 +488,12 @@ bool SID::set_sampling_parameters(double clock_freq, sampling_method method,
// frequencies below ~ 44.1kHz, and 20kHz for higher sample frequencies.
if (pass_freq < 0) {
pass_freq = 20000;
if (2*pass_freq/sample_freq >= 0.9) {
pass_freq = 0.9*sample_freq/2;
if (2.0*pass_freq/sample_freq >= 0.9) {
pass_freq = 0.9*sample_freq/2.0;
}
}
// Check whether the FIR table would overfill.
else if (pass_freq > 0.9*sample_freq/2) {
else if (pass_freq > 0.9*sample_freq/2.0) {
return false;
}
@ -520,20 +526,20 @@ bool SID::set_sampling_parameters(double clock_freq, sampling_method method,
return true;
}
const double pi = 3.1415926535897932385;
const float pi = 3.1415926535897932385;
// 16 bits -> -96dB stopband attenuation.
const double A = -20*log10(1.0/(1 << 16));
const float A = -20.0*log10(1.0/(1 << 16));
// A fraction of the bandwidth is allocated to the transition band,
double dw = (1 - 2*pass_freq/sample_freq)*pi;
float dw = (1.0 - 2.0*pass_freq/sample_freq)*pi;
// The cutoff frequency is midway through the transition band.
double wc = (2*pass_freq/sample_freq + 1)*pi/2;
float wc = (2.0*pass_freq/sample_freq + 1.0)*pi/2.0;
// For calculation of beta and N see the reference for the kaiserord
// function in the MATLAB Signal Processing Toolbox:
// http://www.mathworks.com/access/helpdesk/help/toolbox/signal/kaiserord.html
const double beta = 0.1102*(A - 8.7);
const double I0beta = I0(beta);
const float beta = 0.1102*(A - 8.7);
const float I0beta = I0(beta);
// The filter order will maximally be 124 with the current constraints.
// N >= (96.33 - 7.95)/(2.285*0.1*pi) -> N >= 123
@ -542,8 +548,8 @@ bool SID::set_sampling_parameters(double clock_freq, sampling_method method,
int N = int((A - 7.95)/(2.285*dw) + 0.5);
N += N & 1;
double f_samples_per_cycle = sample_freq/clock_freq;
double f_cycles_per_sample = clock_freq/sample_freq;
float f_samples_per_cycle = sample_freq/clock_freq;
float f_cycles_per_sample = clock_freq/sample_freq;
// The filter length is equal to the filter order + 1.
// The filter length must be an odd number (sinc is symmetric about x = 0).
@ -564,18 +570,18 @@ bool SID::set_sampling_parameters(double clock_freq, sampling_method method,
// Calculate fir_RES FIR tables for linear interpolation.
for (int i = 0; i < fir_RES; i++) {
int fir_offset = i*fir_N + fir_N/2;
double j_offset = double(i)/fir_RES;
float j_offset = float(i)/fir_RES;
// Calculate FIR table. This is the sinc function, weighted by the
// Kaiser window.
for (int j = -fir_N/2; j <= fir_N/2; j++) {
double jx = j - j_offset;
double wt = wc*jx/f_cycles_per_sample;
double temp = jx/(fir_N/2);
double Kaiser =
float jx = j - j_offset;
float wt = wc*jx/f_cycles_per_sample;
float temp = jx/(fir_N/2);
float Kaiser =
fabs(temp) <= 1 ? I0(beta*sqrt(1 - temp*temp))/I0beta : 0;
double sincwt =
float sincwt =
fabs(wt) >= 1e-6 ? sin(wt)/wt : 1;
double val =
float val =
(1 << FIR_SHIFT)*filter_scale*f_samples_per_cycle*wc/pi*sincwt*Kaiser;
fir[fir_offset + j] = short(val + 0.5);
}
@ -607,7 +613,7 @@ bool SID::set_sampling_parameters(double clock_freq, sampling_method method,
// that any adjustment of the sampling frequency will change the
// characteristics of the resampling filter, since the filter is not rebuilt.
// ----------------------------------------------------------------------------
void SID::adjust_sampling_frequency(double sample_freq)
void SID::adjust_sampling_frequency(float sample_freq)
{
cycles_per_sample =
cycle_count(clock_frequency/sample_freq*(1 << FIXP_SHIFT) + 0.5);
@ -638,7 +644,6 @@ PointPlotter<sound_sample> SID::fc_plotter()
// ----------------------------------------------------------------------------
void SID::clock()
{
int i;
// Age bus value.
if (--bus_value_ttl <= 0) {
@ -647,20 +652,20 @@ void SID::clock()
}
// Clock amplitude modulators.
for (i = 0; i < 3; i++) {
voice[i].envelope.clock();
}
voice[0].envelope.clock();
voice[1].envelope.clock();
voice[2].envelope.clock();
// Clock oscillators.
for (i = 0; i < 3; i++) {
voice[i].wave.clock();
}
voice[0].wave.clock();
voice[1].wave.clock();
voice[2].wave.clock();
// Synchronize oscillators.
for (i = 0; i < 3; i++) {
voice[i].wave.synchronize();
}
voice[0].wave.synchronize();
voice[1].wave.synchronize();
voice[2].wave.synchronize();
// Clock filter.
filter.clock(voice[0].output(), voice[1].output(), voice[2].output(), ext_in);
@ -688,10 +693,11 @@ void SID::clock(cycle_count delta_t)
}
// Clock amplitude modulators.
for (i = 0; i < 3; i++) {
voice[i].envelope.clock(delta_t);
}
voice[0].envelope.clock(delta_t);
voice[1].envelope.clock(delta_t);
voice[2].envelope.clock(delta_t);
// Clock and synchronize oscillators.
// Loop until we reach the current cycle.
cycle_count delta_t_osc = delta_t;
@ -715,27 +721,28 @@ void SID::clock(cycle_count delta_t)
// Clock on MSB off if MSB is on, clock on MSB on if MSB is off.
reg24 delta_accumulator =
(accumulator & 0x800000 ? 0x1000000 : 0x800000) - accumulator;
(accumulator & 0x800000 ? 0x1000000 : 0x800000) - accumulator;
cycle_count delta_t_next = delta_accumulator/freq;
if (delta_accumulator%freq) {
++delta_t_next;
++delta_t_next;
}
if (delta_t_next < delta_t_min) {
delta_t_min = delta_t_next;
delta_t_min = delta_t_next;
}
}
// Clock oscillators.
for (i = 0; i < 3; i++) {
voice[i].wave.clock(delta_t_min);
}
voice[0].wave.clock(delta_t_min);
voice[1].wave.clock(delta_t_min);
voice[2].wave.clock(delta_t_min);
// Synchronize oscillators.
for (i = 0; i < 3; i++) {
voice[i].wave.synchronize();
}
voice[0].wave.synchronize();
voice[1].wave.synchronize();
voice[2].wave.synchronize();
delta_t_osc -= delta_t_min;
}
@ -957,6 +964,9 @@ int SID::clock_resample_interpolate(cycle_count& delta_t, short* buf, int n,
v >>= FIR_SHIFT;
// Saturated arithmetics to guard against 16 bit sample overflow.
//(fb)
asm ("ssat %0, #16, %1" : "=r" (v) : "r" (v));
/*
const int half = 1 << 15;
if (v >= half) {
v = half - 1;
@ -964,7 +974,7 @@ int SID::clock_resample_interpolate(cycle_count& delta_t, short* buf, int n,
else if (v < -half) {
v = -half;
}
*/
buf[s++*interleave] = v;
}
@ -1020,6 +1030,9 @@ int SID::clock_resample_fast(cycle_count& delta_t, short* buf, int n,
v >>= FIR_SHIFT;
// Saturated arithmetics to guard against 16 bit sample overflow.
//(fb)
asm ("ssat %0, #16, %1" : "=r" (v) : "r" (v));
/*
const int half = 1 << 15;
if (v >= half) {
v = half - 1;
@ -1027,7 +1040,7 @@ int SID::clock_resample_fast(cycle_count& delta_t, short* buf, int n,
else if (v < -half) {
v = -half;
}
*/
buf[s++*interleave] = v;
}

18
reSID/sid.h
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@ -2,9 +2,9 @@
// 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
// This program is free float; 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
// the Free float 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,
@ -13,7 +13,7 @@
// 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
// along with this program; if not, write to the Free float
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
// ---------------------------------------------------------------------------
@ -37,10 +37,10 @@ public:
void set_chip_model(chip_model model);
void enable_filter(bool enable);
void enable_external_filter(bool enable);
bool set_sampling_parameters(double clock_freq, sampling_method method,
double sample_freq, double pass_freq = -1,
double filter_scale = 0.97);
void adjust_sampling_frequency(double sample_freq);
bool set_sampling_parameters(float clock_freq, sampling_method method,
float sample_freq, float pass_freq = -1,
float filter_scale = 0.97);
void adjust_sampling_frequency(float sample_freq);
void fc_default(const fc_point*& points, int& count);
PointPlotter<sound_sample> fc_plotter();
@ -89,7 +89,7 @@ public:
int output(int bits);
protected:
static double I0(double x);
static float I0(float x);
RESID_INLINE int clock_fast(cycle_count& delta_t, short* buf, int n,
int interleave);
RESID_INLINE int clock_interpolate(cycle_count& delta_t, short* buf, int n,
@ -108,7 +108,7 @@ protected:
reg8 bus_value;
cycle_count bus_value_ttl;
double clock_frequency;
float clock_frequency;
// External audio input.
int ext_in;

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