home
***
CD-ROM
|
disk
|
FTP
|
other
***
search
/
Large Pack of OldSkool DOS MOD Trackers
/
goattracker_2.68.zip
/
src
/
resid-fp
/
sid.cpp
< prev
next >
Wrap
C/C++ Source or Header
|
2009-01-03
|
29KB
|
937 lines
// ---------------------------------------------------------------------------
// 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
// ---------------------------------------------------------------------------
#include "sid.h"
#include <math.h>
extern float convolve(const float *a, const float *b, int n);
extern float convolve_sse(const float *a, const float *b, int n);
enum host_cpu_feature {
HOST_CPU_MMX=1, HOST_CPU_SSE=2, HOST_CPU_SSE2=4, HOST_CPU_SSE3=8
};
/* This code is appropriate for 32-bit and 64-bit x86 CPUs. */
#if defined(__x86_64__) || defined(__i386__) || defined(_MSC_VER)
struct cpu_x86_regs_s {
unsigned int eax;
unsigned int ebx;
unsigned int ecx;
unsigned int edx;
};
typedef struct cpu_x86_regs_s cpu_x86_regs_t;
static cpu_x86_regs_t get_cpuid_regs(unsigned int index)
{
cpu_x86_regs_t retval;
#if defined(_MSC_VER) /* MSVC assembly */
__asm {
mov eax, [index]
cpuid
mov [retval.eax], eax
mov [retval.ebx], ebx
mov [retval.ecx], ecx
mov [retval.edx], edx
}
#else /* GNU assembly */
asm("movl %4, %%eax; cpuid; movl %%eax, %0; movl %%ebx, %1; movl %%ecx, %2; movl %%edx, %3;"
: "=m" (retval.eax),
"=m" (retval.ebx),
"=m" (retval.ecx),
"=m" (retval.edx)
: "r" (index)
: "eax", "ebx", "ecx", "edx");
#endif
return retval;
}
static int host_cpu_features_by_cpuid(void)
{
cpu_x86_regs_t regs = get_cpuid_regs(1);
int features = 0;
if (regs.edx & (1 << 23))
features |= HOST_CPU_MMX;
if (regs.edx & (1 << 25))
features |= HOST_CPU_SSE;
if (regs.edx & (1 << 26))
features |= HOST_CPU_SSE2;
if (regs.ecx & (1 << 0))
features |= HOST_CPU_SSE3;
return features;
}
static int host_cpu_features(void)
{
static int features = 0;
static int features_detected = 0;
/* 32-bit only */
#if defined(__i386__) || (defined(_MSC_VER) && defined(_WIN32))
unsigned long temp1, temp2;
#endif
if (features_detected)
return features;
features_detected = 1;
#if defined(_MSC_VER) && defined(_WIN32) /* MSVC compatible assembly appropriate for 32-bit Windows */
/* see if we are dealing with a cpu that has the cpuid instruction */
__asm {
pushf
pop eax
mov [temp1], eax
xor eax, 0x200000
push eax
popf
pushf
pop eax
mov [temp2], eax
push [temp1]
popf
}
#endif
#if defined(__i386__) /* GNU assembly */
asm("pushfl; popl %%eax; movl %%eax, %0; xorl $0x200000, %%eax; pushl %%eax; popfl; pushfl; popl %%eax; movl %%eax, %1; pushl %0; popfl "
: "=r" (temp1),
"=r" (temp2)
:
: "eax");
#endif
#if defined(__i386__) || (defined(_MSC_VER) && defined(_WIN32))
temp1 &= 0x200000;
temp2 &= 0x200000;
if (temp1 == temp2) {
/* no cpuid support, so we can't test for SSE availability -> false */
return 0;
}
#endif
/* find the highest supported cpuid function, returned in %eax */
if (get_cpuid_regs(0).eax < 1) {
/* no cpuid 1 function, we can't test for features -> no features */
return 0;
}
features = host_cpu_features_by_cpuid();
return features;
}
#else /* !__x86_64__ && !__i386__ && !_MSC_VER */
static int host_cpu_features(void)
{
return 0;
}
#endif
float SIDFP::kinked_dac(const int x, const float nonlinearity, const int max)
{
float value = 0.f;
int bit = 1;
float weight = 1.f;
const float dir = 2.0f * nonlinearity;
for (int i = 0; i < max; i ++) {
if (x & bit)
value += weight;
bit <<= 1;
weight *= dir;
}
return value / (weight / nonlinearity) * (1 << max);
}
// ----------------------------------------------------------------------------
// Constructor.
// ----------------------------------------------------------------------------
SIDFP::SIDFP()
{
#if (RESID_USE_SSE==1)
can_use_sse = (host_cpu_features() & HOST_CPU_SSE) != 0;
#else
can_use_sse = false;
#endif
// Initialize pointers.
sample = 0;
fir = 0;
voice[0].set_sync_source(&voice[2]);
voice[1].set_sync_source(&voice[0]);
voice[2].set_sync_source(&voice[1]);
set_sampling_parameters(985248, SAMPLE_INTERPOLATE, 44100);
bus_value = 0;
bus_value_ttl = 0;
input(0);
}
// ----------------------------------------------------------------------------
// Destructor.
// ----------------------------------------------------------------------------
SIDFP::~SIDFP()
{
delete[] sample;
delete[] fir;
}
// ----------------------------------------------------------------------------
// Set chip model.
// ----------------------------------------------------------------------------
void SIDFP::set_chip_model(chip_model model)
{
for (int i = 0; i < 3; i++) {
voice[i].set_chip_model(model);
}
filter.set_chip_model(model);
extfilt.set_chip_model(model);
}
/* nonlinear DAC support, set 1 for 8580 / no effect, about 0.96 otherwise */
void SIDFP::set_voice_nonlinearity(float nl)
{
for (int i = 0; i < 3; i++) {
voice[i].set_nonlinearity(nl);
}
}
// ----------------------------------------------------------------------------
// SID reset.
// ----------------------------------------------------------------------------
void SIDFP::reset()
{
for (int i = 0; i < 3; i++) {
voice[i].reset();
}
filter.reset();
extfilt.reset();
bus_value = 0;
bus_value_ttl = 0;
}
// ----------------------------------------------------------------------------
// Write 16-bit sample to audio input.
// NB! The caller is responsible for keeping the value within 16 bits.
// Note that to mix in an external audio signal, the signal should be
// resampled to 1MHz first to avoid sampling noise.
// ----------------------------------------------------------------------------
void SIDFP::input(int sample)
{
// Voice outputs are 20 bits. Scale up to match three voices in order
// to facilitate simulation of the MOS8580 "digi boost" hardware hack.
ext_in = (float) ( (sample << 4) * 3 );
}
float SIDFP::output()
{
const float range = 1 << 15;
return extfilt.output() / (4095.f * 255.f * 3.f * 1.5f / range);
}
// ----------------------------------------------------------------------------
// Read registers.
//
// Reading a write only register returns the last byte written to any SID
// register. The individual bits in this value start to fade down towards
// zero after a few cycles. All bits reach zero within approximately
// $2000 - $4000 cycles.
// It has been claimed that this fading happens in an orderly fashion, however
// sampling of write only registers reveals that this is not the case.
// NB! This is not correctly modeled.
// The actual use of write only registers has largely been made in the belief
// that all SID registers are readable. To support this belief the read
// would have to be done immediately after a write to the same register
// (remember that an intermediate write to another register would yield that
// value instead). With this in mind we return the last value written to
// any SID register for $2000 cycles without modeling the bit fading.
// ----------------------------------------------------------------------------
reg8 SIDFP::read(reg8 offset)
{
switch (offset) {
case 0x19:
return potx.readPOT();
case 0x1a:
return poty.readPOT();
case 0x1b:
return voice[2].wave.readOSC();
case 0x1c:
return voice[2].envelope.readENV();
default:
return bus_value;
}
}
// ----------------------------------------------------------------------------
// Write registers.
// ----------------------------------------------------------------------------
void SIDFP::write(reg8 offset, reg8 value)
{
bus_value = value;
bus_value_ttl = 0x4000;
switch (offset) {
case 0x00:
voice[0].wave.writeFREQ_LO(value);
break;
case 0x01:
voice[0].wave.writeFREQ_HI(value);
break;
case 0x02:
voice[0].wave.writePW_LO(value);
break;
case 0x03:
voice[0].wave.writePW_HI(value);
break;
case 0x04:
voice[0].writeCONTROL_REG(value);
break;
case 0x05:
voice[0].envelope.writeATTACK_DECAY(value);
break;
case 0x06:
voice[0].envelope.writeSUSTAIN_RELEASE(value);
break;
case 0x07:
voice[1].wave.writeFREQ_LO(value);
break;
case 0x08:
voice[1].wave.writeFREQ_HI(value);
break;
case 0x09:
voice[1].wave.writePW_LO(value);
break;
case 0x0a:
voice[1].wave.writePW_HI(value);
break;
case 0x0b:
voice[1].writeCONTROL_REG(value);
break;
case 0x0c:
voice[1].envelope.writeATTACK_DECAY(value);
break;
case 0x0d:
voice[1].envelope.writeSUSTAIN_RELEASE(value);
break;
case 0x0e:
voice[2].wave.writeFREQ_LO(value);
break;
case 0x0f:
voice[2].wave.writeFREQ_HI(value);
break;
case 0x10:
voice[2].wave.writePW_LO(value);
break;
case 0x11:
voice[2].wave.writePW_HI(value);
break;
case 0x12:
voice[2].writeCONTROL_REG(value);
break;
case 0x13:
voice[2].envelope.writeATTACK_DECAY(value);
break;
case 0x14:
voice[2].envelope.writeSUSTAIN_RELEASE(value);
break;
case 0x15:
filter.writeFC_LO(value);
break;
case 0x16:
filter.writeFC_HI(value);
break;
case 0x17:
filter.writeRES_FILT(value);
break;
case 0x18:
filter.writeMODE_VOL(value);
break;
default:
break;
}
}
// ----------------------------------------------------------------------------
// Constructor.
// ----------------------------------------------------------------------------
SIDFP::State::State()
{
int i;
for (i = 0; i < 0x20; i++) {
sid_register[i] = 0;
}
bus_value = 0;
bus_value_ttl = 0;
for (i = 0; i < 3; i++) {
accumulator[i] = 0;
shift_register[i] = 0x7ffff8;
rate_counter[i] = 0;
rate_counter_period[i] = 9;
exponential_counter[i] = 0;
exponential_counter_period[i] = 1;
envelope_counter[i] = 0;
envelope_state[i] = EnvelopeGeneratorFP::RELEASE;
hold_zero[i] = true;
}
}
// ----------------------------------------------------------------------------
// Read state.
// ----------------------------------------------------------------------------
SIDFP::State SIDFP::read_state()
{
State state;
int i, j;
for (i = 0, j = 0; i < 3; i++, j += 7) {
WaveformGeneratorFP& wave = voice[i].wave;
EnvelopeGeneratorFP& envelope = voice[i].envelope;
state.sid_register[j + 0] = wave.freq & 0xff;
state.sid_register[j + 1] = wave.freq >> 8;
state.sid_register[j + 2] = wave.pw & 0xff;
state.sid_register[j + 3] = wave.pw >> 8;
state.sid_register[j + 4] =
(wave.waveform << 4)
| (wave.test ? 0x08 : 0)
| (wave.ring_mod ? 0x04 : 0)
| (wave.sync ? 0x02 : 0)
| (envelope.gate ? 0x01 : 0);
state.sid_register[j + 5] = (envelope.attack << 4) | envelope.decay;
state.sid_register[j + 6] = (envelope.sustain << 4) | envelope.release;
}
state.sid_register[j++] = filter.fc & 0x007;
state.sid_register[j++] = filter.fc >> 3;
state.sid_register[j++] = (filter.res << 4) | filter.filt;
state.sid_register[j++] =
(filter.voice3off ? 0x80 : 0)
| (filter.hp_bp_lp << 4)
| filter.vol;
// These registers are superfluous, but included for completeness.
for (; j < 0x1d; j++) {
state.sid_register[j] = read(j);
}
for (; j < 0x20; j++) {
state.sid_register[j] = 0;
}
state.bus_value = bus_value;
state.bus_value_ttl = bus_value_ttl;
for (i = 0; i < 3; i++) {
state.accumulator[i] = voice[i].wave.accumulator;
state.shift_register[i] = voice[i].wave.shift_register;
state.rate_counter[i] = voice[i].envelope.rate_counter;
state.rate_counter_period[i] = voice[i].envelope.rate_period;
state.exponential_counter[i] = voice[i].envelope.exponential_counter;
state.exponential_counter_period[i] = voice[i].envelope.exponential_counter_period;
state.envelope_counter[i] = voice[i].envelope.envelope_counter;
state.envelope_state[i] = voice[i].envelope.state;
state.hold_zero[i] = voice[i].envelope.hold_zero;
}
return state;
}
// ----------------------------------------------------------------------------
// Write state.
// ----------------------------------------------------------------------------
void SIDFP::write_state(const State& state)
{
int i;
for (i = 0; i <= 0x18; i++) {
write(i, state.sid_register[i]);
}
bus_value = state.bus_value;
bus_value_ttl = state.bus_value_ttl;
for (i = 0; i < 3; i++) {
voice[i].wave.accumulator = state.accumulator[i];
voice[i].wave.shift_register = state.shift_register[i];
voice[i].envelope.rate_counter = state.rate_counter[i];
voice[i].envelope.rate_period = state.rate_counter_period[i];
voice[i].envelope.exponential_counter = state.exponential_counter[i];
voice[i].envelope.exponential_counter_period = state.exponential_counter_period[i];
voice[i].envelope.envelope_counter = state.envelope_counter[i];
voice[i].envelope.state = state.envelope_state[i];
voice[i].envelope.hold_zero = state.hold_zero[i];
}
}
// ----------------------------------------------------------------------------
// Enable filter.
// ----------------------------------------------------------------------------
void SIDFP::enable_filter(bool enable)
{
filter.enable_filter(enable);
}
// ----------------------------------------------------------------------------
// Enable external filter.
// ----------------------------------------------------------------------------
void SIDFP::enable_external_filter(bool enable)
{
extfilt.enable_filter(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 SIDFP::I0(double x)
{
// Max error acceptable in I0 could be 1e-6, which gives that 96 dB already.
// I'm overspecify these errors to get a beautiful FFT dump of the FIR.
const double I0e = 1e-10;
double sum, u, halfx, temp;
int n;
sum = u = n = 1;
halfx = x/2.0;
do {
temp = halfx/n++;
u *= temp*temp;
sum += u;
} while (u >= I0e*sum);
return sum;
}
// ----------------------------------------------------------------------------
// Setting of SID sampling parameters.
//
// Use a clock freqency of 985248Hz for PAL C64, 1022730Hz for NTSC C64.
// The default end of passband frequency is pass_freq = 0.9*sample_freq/2
// for sample frequencies up to ~ 44.1kHz, and 20kHz for higher sample
// frequencies.
//
// For resampling, the ratio between the clock frequency and the sample
// frequency is limited as follows:
// 125*clock_freq/sample_freq < 16384
// E.g. provided a clock frequency of ~ 1MHz, the sample frequency can not
// be set lower than ~ 8kHz. A lower sample frequency would make the
// resampling code overfill its 16k sample ring buffer.
//
// The end of passband frequency is also limited:
// pass_freq <= 0.9*sample_freq/2
// E.g. for a 44.1kHz sampling rate the end of passband frequency is limited
// to slightly below 20kHz. This constraint ensures that the FIR table is
// not overfilled.
// ----------------------------------------------------------------------------
bool SIDFP::set_sampling_parameters(float clock_freq, sampling_method method,
float sample_freq, float pass_freq)
{
clock_frequency = clock_freq;
sampling = method;
filter.set_clock_frequency(clock_freq);
extfilt.set_clock_frequency(clock_freq);
adjust_sampling_frequency(sample_freq);
sample_offset = 0;
sample_prev = 0;
// FIR initialization is only necessary for resampling.
if (method != SAMPLE_RESAMPLE_INTERPOLATE)
{
delete[] sample;
delete[] fir;
sample = 0;
fir = 0;
return true;
}
const int bits = 16;
if (pass_freq > 20000)
pass_freq = 20000;
if (2*pass_freq/sample_freq > 0.9)
pass_freq = 0.9f*sample_freq/2;
// 16 bits -> -96dB stopband attenuation.
const double A = -20*log10(1.0/(1 << bits));
// 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);
double f_samples_per_cycle = sample_freq/clock_freq;
double f_cycles_per_sample = clock_freq/sample_freq;
/* This code utilizes the fact that aliasing back to 20 kHz from
* sample_freq/2 is inaudible. This allows us to define a passband
* wider than normally. We might also consider aliasing back to pass_freq,
* but as this can be less than 20 kHz, it might become audible... */
double aliasing_allowance = sample_freq / 2 - 20000;
if (aliasing_allowance < 0)
aliasing_allowance = 0;
double transition_bandwidth = sample_freq/2 - pass_freq + aliasing_allowance;
{
/* Filter order according to Kaiser's paper. */
int N = (int) ((A - 7.95)/(2 * M_PI * 2.285 * transition_bandwidth/sample_freq) + 0.5);
N += N & 1;
// The filter length is equal to the filter order + 1.
// The filter length must be an odd number (sinc is symmetric about x = 0).
fir_N = int(N*f_cycles_per_sample) + 1;
fir_N |= 1;
// Check whether the sample ring buffer would overfill.
if (fir_N > RINGSIZE - 1)
return false;
/* Error is bound by 1.234 / L^2 */
fir_RES = (int) (sqrt(1.234 * (1 << bits)) / f_cycles_per_sample + 0.5);
}
// Allocate memory for FIR tables.
delete[] fir;
fir = new float[fir_N*fir_RES];
// The cutoff frequency is midway through the transition band.
double wc = (pass_freq + transition_bandwidth/2) / sample_freq * M_PI * 2;
// Calculate fir_RES FIR tables for linear interpolation.
for (int i = 0; i < fir_RES; i++) {
double j_offset = double(i)/fir_RES;
// Calculate FIR table. This is the sinc function, weighted by the
// Kaiser window.
for (int j = 0; j < fir_N; j ++) {
double jx = j - fir_N/2. - j_offset;
double wt = wc*jx/f_cycles_per_sample;
double temp = jx/(fir_N/2);
double Kaiser =
fabs(temp) <= 1 ? I0(beta*sqrt(1 - temp*temp))/I0beta : 0;
double sincwt =
fabs(wt) >= 1e-8 ? sin(wt)/wt : 1;
fir[i * fir_N + j] = (float) (f_samples_per_cycle*wc/M_PI*sincwt*Kaiser);
}
}
// Allocate sample buffer.
if (!sample) {
sample = new float[RINGSIZE*2];
}
// Clear sample buffer.
for (int j = 0; j < RINGSIZE*2; j++) {
sample[j] = 0;
}
sample_index = 0;
return true;
}
// ----------------------------------------------------------------------------
// Adjustment of SID sampling frequency.
//
// In some applications, e.g. a C64 emulator, it can be desirable to
// synchronize sound with a timer source. This is supported by adjustment of
// the SID sampling frequency.
//
// NB! Adjustment of the sampling frequency may lead to noticeable shifts in
// frequency, and should only be used for interactive applications. Note also
// that any adjustment of the sampling frequency will change the
// characteristics of the resampling filter, since the filter is not rebuilt.
// ----------------------------------------------------------------------------
void SIDFP::adjust_sampling_frequency(float sample_freq)
{
cycles_per_sample = clock_frequency/sample_freq;
}
void SIDFP::age_bus_value(cycle_count n) {
if (bus_value_ttl != 0) {
bus_value_ttl -= n;
if (bus_value_ttl <= 0) {
bus_value = 0;
bus_value_ttl = 0;
}
}
}
// ----------------------------------------------------------------------------
// SID clocking - 1 cycle.
// ----------------------------------------------------------------------------
void SIDFP::clock()
{
int i;
// Clock amplitude modulators.
for (i = 0; i < 3; i++) {
voice[i].envelope.clock();
}
// Clock oscillators.
for (i = 0; i < 3; i++) {
voice[i].wave.clock();
}
// Synchronize oscillators.
for (i = 0; i < 3; i++) {
voice[i].wave.synchronize();
}
// Clock filter.
extfilt.clock(filter.clock(voice[0].output(), voice[1].output(), voice[2].output(), ext_in));
}
// ----------------------------------------------------------------------------
// SID clocking with audio sampling.
// Fixpoint arithmetics is used.
//
// The example below shows how to clock the SID a specified amount of cycles
// while producing audio output:
//
// while (delta_t) {
// bufindex += sid.clock(delta_t, buf + bufindex, buflength - bufindex);
// write(dsp, buf, bufindex*2);
// bufindex = 0;
// }
//
// ----------------------------------------------------------------------------
int SIDFP::clock(cycle_count& delta_t, short* buf, int n, int interleave)
{
/* XXX I assume n is generally large enough for delta_t here... */
age_bus_value(delta_t);
int res;
switch (sampling) {
default:
case SAMPLE_INTERPOLATE:
res = clock_interpolate(delta_t, buf, n, interleave);
break;
case SAMPLE_RESAMPLE_INTERPOLATE:
res = clock_resample_interpolate(delta_t, buf, n, interleave);
break;
}
filter.nuke_denormals();
extfilt.nuke_denormals();
return res;
}
// ----------------------------------------------------------------------------
// SID clocking with audio sampling - cycle based with linear sample
// interpolation.
//
// Here the chip is clocked every cycle. This yields higher quality
// sound since the samples are linearly interpolated, and since the
// external filter attenuates frequencies above 16kHz, thus reducing
// sampling noise.
// ----------------------------------------------------------------------------
RESID_INLINE
int SIDFP::clock_interpolate(cycle_count& delta_t, short* buf, int n,
int interleave)
{
int s = 0;
int i;
for (;;) {
float next_sample_offset = sample_offset + cycles_per_sample;
int delta_t_sample = (int) next_sample_offset;
if (delta_t_sample > delta_t) {
break;
}
if (s >= n) {
return s;
}
for (i = 0; i < delta_t_sample - 1; i++) {
clock();
}
if (i < delta_t_sample) {
sample_prev = output();
clock();
}
delta_t -= delta_t_sample;
sample_offset = next_sample_offset - delta_t_sample;
float sample_now = output();
int v = (int)(sample_prev + (sample_offset * (sample_now - sample_prev)));
// Saturated arithmetics to guard against 16 bit sample overflow.
const int half = 1 << 15;
if (v >= half) {
v = half - 1;
}
else if (v < -half) {
v = -half;
}
buf[s++*interleave] = v;
sample_prev = sample_now;
}
for (i = 0; i < delta_t - 1; i++) {
clock();
}
if (i < delta_t) {
sample_prev = output();
clock();
}
sample_offset -= delta_t;
delta_t = 0;
return s;
}
// ----------------------------------------------------------------------------
// SID clocking with audio sampling - cycle based with audio resampling.
//
// This is the theoretically correct (and computationally intensive) audio
// sample generation. The samples are generated by resampling to the specified
// sampling frequency. The work rate is inversely proportional to the
// percentage of the bandwidth allocated to the filter transition band.
//
// This implementation is based on the paper "A Flexible Sampling-Rate
// Conversion Method", by J. O. Smith and P. Gosset, or rather on the
// expanded tutorial on the "Digital Audio Resampling Home Page":
// http://www-ccrma.stanford.edu/~jos/resample/
//
// By building shifted FIR tables with samples according to the
// sampling frequency, this implementation dramatically reduces the
// computational effort in the filter convolutions, without any loss
// of accuracy. The filter convolutions are also vectorizable on
// current hardware.
//
// Further possible optimizations are:
// * An equiripple filter design could yield a lower filter order, see
// http://www.mwrf.com/Articles/ArticleID/7229/7229.html
// * The Convolution Theorem could be used to bring the complexity of
// convolution down from O(n*n) to O(n*log(n)) using the Fast Fourier
// Transform, see http://en.wikipedia.org/wiki/Convolution_theorem
// * Simply resampling in two steps can also yield computational
// savings, since the transition band will be wider in the first step
// and the required filter order is thus lower in this step.
// Laurent Ganier has found the optimal intermediate sampling frequency
// to be (via derivation of sum of two steps):
// 2 * pass_freq + sqrt [ 2 * pass_freq * orig_sample_freq
// * (dest_sample_freq - 2 * pass_freq) / dest_sample_freq ]
//
// NB! the result of right shifting negative numbers is really
// implementation dependent in the C++ standard.
// ----------------------------------------------------------------------------
RESID_INLINE
int SIDFP::clock_resample_interpolate(cycle_count& delta_t, short* buf, int n,
int interleave)
{
int s = 0;
for (;;) {
float next_sample_offset = sample_offset + cycles_per_sample;
/* full clocks left to next sample */
int delta_t_sample = (int) next_sample_offset;
if (delta_t_sample > delta_t || s >= n)
break;
/* clock forward delta_t_sample samples */
for (int i = 0; i < delta_t_sample; i++) {
clock();
sample[sample_index] = sample[sample_index + RINGSIZE] = output();
++ sample_index;
sample_index &= RINGSIZE - 1;
}
delta_t -= delta_t_sample;
/* Phase of the sample in terms of clock, [0 .. 1[. */
sample_offset = next_sample_offset - (float) delta_t_sample;
/* find the first of the nearest fir tables close to the phase */
float fir_offset_rmd = sample_offset * fir_RES;
int fir_offset = (int) fir_offset_rmd;
/* [0 .. 1[ */
fir_offset_rmd -= (float) fir_offset;
/* find fir_N most recent samples, plus one extra in case the FIR wraps. */
float* sample_start = sample + sample_index - fir_N + RINGSIZE - 1;
float v1 =
#if (RESID_USE_SSE==1)
can_use_sse ? convolve_sse(sample_start, fir + fir_offset*fir_N, fir_N) :
#endif
convolve(sample_start, fir + fir_offset*fir_N, fir_N);
// Use next FIR table, wrap around to first FIR table using
// previous sample.
if (++ fir_offset == fir_RES) {
fir_offset = 0;
++ sample_start;
}
float v2 =
#if (RESID_USE_SSE==1)
can_use_sse ? convolve_sse(sample_start, fir + fir_offset*fir_N, fir_N) :
#endif
convolve(sample_start, fir + fir_offset*fir_N, fir_N);
// Linear interpolation between the sinc tables yields good approximation
// for the exact value.
int v = (int) (v1 + fir_offset_rmd * (v2 - v1));
// Saturated arithmetics to guard against 16 bit sample overflow.
const int half = 1 << 15;
if (v >= half) {
v = half - 1;
}
else if (v < -half) {
v = -half;
}
buf[s ++ * interleave] = v;
}
/* clock forward delta_t samples */
for (int i = 0; i < delta_t; i++) {
clock();
sample[sample_index] = sample[sample_index + RINGSIZE] = output();
++ sample_index;
sample_index &= RINGSIZE - 1;
}
sample_offset -= (float) delta_t;
delta_t = 0;
return s;
}