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Diffstat (limited to 'sid/sidplay-libs-2.1.0/resid/sid.cc')
| -rw-r--r-- | sid/sidplay-libs-2.1.0/resid/sid.cc | 856 |
1 files changed, 0 insertions, 856 deletions
diff --git a/sid/sidplay-libs-2.1.0/resid/sid.cc b/sid/sidplay-libs-2.1.0/resid/sid.cc deleted file mode 100644 index f3cd02a3..00000000 --- a/sid/sidplay-libs-2.1.0/resid/sid.cc +++ /dev/null @@ -1,856 +0,0 @@ -// --------------------------------------------------------------------------- -// This file is part of reSID, a MOS6581 SID emulator engine. -// Copyright (C) 2002 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> - -RESID_NAMESPACE_START - -const int SID::FIR_ORDER = RESID_FIR_ORDER; -const int SID::FIR_N = RESID_FIR_N; -const int SID::FIR_RES = RESID_FIR_RES; -const int SID::FIR_SHIFT = RESID_FIR_SHIFT; - -// ---------------------------------------------------------------------------- -// Constructor. -// ---------------------------------------------------------------------------- -SID::SID() -{ - 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_FAST, 44100); -} - - -// ---------------------------------------------------------------------------- -// Set chip model. -// ---------------------------------------------------------------------------- -void SID::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); -} - - -// ---------------------------------------------------------------------------- -// SID reset. -// ---------------------------------------------------------------------------- -void SID::reset() -{ - for (int i = 0; i < 3; i++) { - voice[i].reset(); - } - filter.reset(); - extfilt.reset(); - - bus_value = 0; - bus_value_ttl = 0; -} - - -// ---------------------------------------------------------------------------- -// Read sample of audio output. -// Both 16-bit and n-bit output is provided. -// ---------------------------------------------------------------------------- -int SID::output() -{ - const int range = 1 << 16; - const int half = range >> 1; - int sample = extfilt.output()/((4095*255 >> 7)*3*15*2/range); - if (sample >= half) { - return half - 1; - } - if (sample < -half) { - return -half; - } - return sample; -} - -int SID::output(int bits) -{ - const int range = 1 << bits; - const int half = range >> 1; - int sample = extfilt.output()/((4095*255 >> 7)*3*15*2/range); - if (sample >= half) { - return half - 1; - } - if (sample < -half) { - return -half; - } - return sample; -} - - -// ---------------------------------------------------------------------------- -// 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 SID::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 SID::write(reg8 offset, reg8 value) -{ - bus_value = value; - bus_value_ttl = 0x2000; - - 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; - } -} - - -// ---------------------------------------------------------------------------- -// SID voice muting. -// ---------------------------------------------------------------------------- -void SID::mute(reg8 channel, bool enable) -{ - // Only have 3 voices! - if (channel >= 3) - return; - - voice[channel].mute (enable); -} - - -// ---------------------------------------------------------------------------- -// Constructor. -// ---------------------------------------------------------------------------- -SID::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] = 0; - rate_counter[i] = 0; - exponential_counter[i] = 0; - envelope_counter[i] = 0; - hold_zero[i] = 0; - } -} - - -// ---------------------------------------------------------------------------- -// Read state. -// ---------------------------------------------------------------------------- -SID::State SID::read_state() -{ - State state; - int i, j; - - for (i = 0, j = 0; i < 3; i++, j += 7) { - WaveformGenerator& wave = voice[i].wave; - EnvelopeGenerator& 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.decay << 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.filtex ? 0x08 : 0) - | filter.filt3_filt2_filt1; - 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.exponential_counter[i] = voice[i].envelope.exponential_counter; - state.envelope_counter[i] = voice[i].envelope.envelope_counter; - state.hold_zero[i] = voice[i].envelope.hold_zero; - } - - return state; -} - - -// ---------------------------------------------------------------------------- -// Write state. -// ---------------------------------------------------------------------------- -void SID::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.exponential_counter = state.exponential_counter[i]; - voice[i].envelope.envelope_counter = state.envelope_counter[i]; - voice[i].envelope.hold_zero = state.hold_zero[i]; - } -} - - -// ---------------------------------------------------------------------------- -// Enable filter. -// ---------------------------------------------------------------------------- -void SID::enable_filter(bool enable) -{ - filter.enable_filter(enable); -} - - -// ---------------------------------------------------------------------------- -// Enable external filter. -// ---------------------------------------------------------------------------- -void SID::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 SID::I0(double x) -{ - // Max error acceptable in I0. - const double I0e = 1E-21; - - 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: -// 123*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 SID::set_sampling_parameters(double clock_freq, sampling_method method, - double sample_freq, double pass_freq) -{ - // Check resampling constraints. - if (method == SAMPLE_RESAMPLE) { - // Check whether the sample ring buffer would overfill. - if (FIR_ORDER*clock_freq/sample_freq >= 16384) { - return false; - } - } - - // The default passband limit is 0.9*sample_freq/2 for sample - // 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; - } - } - // Check whether the FIR table would overfill. - else if (pass_freq > 0.9*sample_freq/2) { - return false; - } - - // Set the external filter to the pass freq - extfilt.set_sampling_parameter (pass_freq); - clock_frequency = clock_freq; - sampling = method; - - cycles_per_sample = - cycle_count(clock_freq/sample_freq*(1 << 10) + 0.5); - - sample_offset = 0; - sample_prev = 0; - - // FIR initialization is only necessary for resampling. - if (sampling != SAMPLE_RESAMPLE) { - return true; - } - - const double pi = 3.1415926535897932385; - - // 16 bits -> -96dB stopband attenuation. - const double A = -20*log10(1.0/(1 << 16)); - const double beta = 0.1102*(A - 8.7); - const double I0beta = I0(beta); - - // A fraction of the bandwidth is allocated to the transition band, - double dw = (1 - 2*pass_freq/sample_freq)*pi; - - // The filter order will maximally be 123 with the current constraints. - // N >= (A - 8)/(2.285*0.1*pi) -> N >= 123 - int N = int((A - 8)/(2.285*dw) + 0.5); - fir_N = 1 + N/2; - foffset_max = fir_N*FIR_RES << 10; - - // The cutoff frequency is midway through the transition band. - double wc = (2*pass_freq/sample_freq + 1)*pi/2; - - // Calculate FIR table. This is the right wing of the sinc function, - // weighted by the Kaiser window. - double samples_per_cycle = sample_freq/clock_freq; - double val1, val2 = 0; - for (int i = fir_N*FIR_RES; i > 0; i--) { - double wt = wc*i/FIR_RES; - double temp = double(i)/(fir_N*FIR_RES); - val1 = (1 << FIR_SHIFT)*samples_per_cycle*wc/pi*sin(wt)/wt*I0(beta*sqrt(1.0 - temp*temp))/I0beta; - fir[i] = short(val1 + 0.5); - fir_diff[i] = short(val2 - val1 + 0.5); - val2 = val1; - } - val1 = (1 << FIR_SHIFT)*samples_per_cycle*wc/pi; - fir[0] = short(val1 + 0.5); - fir_diff[0] = short(val2 - val1 + 0.5); - - // Calculate FIR constants. - fstep_per_cycle = - cycle_count(FIR_RES*sample_freq/clock_freq*(1 << 10) + 0.5); - sample_delay = cycle_count(fir_N*clock_freq/sample_freq + 0.5); - - // Clear sample buffer. - for (int j = 0; j < 4096; 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 SID::adjust_sampling_frequency(double sample_freq) -{ - cycles_per_sample = - cycle_count(clock_frequency/sample_freq*(1 << 10) + 0.5); -} - - -// ---------------------------------------------------------------------------- -// Return array of default spline interpolation points to map FC to -// filter cutoff frequency. -// ---------------------------------------------------------------------------- -void SID::fc_default(const fc_point*& points, int& count) -{ - filter.fc_default(points, count); -} - - -// ---------------------------------------------------------------------------- -// Return FC spline plotter object. -// ---------------------------------------------------------------------------- -PointPlotter<sound_sample> SID::fc_plotter() -{ - return filter.fc_plotter(); -} - - -// ---------------------------------------------------------------------------- -// SID clocking - 1 cycle. -// ---------------------------------------------------------------------------- -void SID::clock() -{ - int i; - - // Age bus value. - if (--bus_value_ttl <= 0) { - bus_value = 0; - bus_value_ttl = 0; - } - - // 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. - filter.clock(voice[0].output(), voice[1].output(), voice[2].output()); - - // Clock external filter. - extfilt.clock(filter.output()); -} - - -// ---------------------------------------------------------------------------- -// SID clocking - delta_t cycles. -// ---------------------------------------------------------------------------- -void SID::clock(cycle_count delta_t) -{ - int i; - - if (delta_t <= 0) { - return; - } - - // Age bus value. - bus_value_ttl -= delta_t; - if (bus_value_ttl <= 0) { - bus_value = 0; - bus_value_ttl = 0; - } - - // Clock amplitude modulators. - for (i = 0; i < 3; i++) { - voice[i].envelope.clock(delta_t); - } - - // Clock and synchronize oscillators. - // Loop until we reach the current cycle. - cycle_count delta_t_osc = delta_t; - while (delta_t_osc) { - cycle_count delta_t_min = delta_t_osc; - - // Find minimum number of cycles to an oscillator accumulator MSB toggle. - // We have to clock on each MSB on / MSB off for hard sync to operate - // correctly. - for (i = 0; i < 3; i++) { - WaveformGenerator& wave = voice[i].wave; - - // It is only necessary to clock on the MSB of an oscillator that is - // a sync source and has freq != 0. - if (!(wave.sync_dest->sync && wave.freq)) { - continue; - } - - reg16 freq = wave.freq; - reg24 accumulator = wave.accumulator; - - // Clock on MSB off if MSB is on, clock on MSB on if MSB is off. - reg24 delta_accumulator = - (accumulator & 0x800000 ? 0x1000000 : 0x800000) - accumulator; - - cycle_count delta_t_next = delta_accumulator/freq; - if (delta_accumulator%freq) { - ++delta_t_next; - } - - if (delta_t_next < delta_t_min) { - delta_t_min = delta_t_next; - } - } - - // Clock oscillators. - for (i = 0; i < 3; i++) { - voice[i].wave.clock(delta_t_min); - } - - // Synchronize oscillators. - for (i = 0; i < 3; i++) { - voice[i].wave.synchronize(); - } - - delta_t_osc -= delta_t_min; - } - - // Clock filter. - filter.clock(delta_t, - voice[0].output(), voice[1].output(), voice[2].output()); - - // Clock external filter. - extfilt.clock(delta_t, filter.output()); -} - - -// ---------------------------------------------------------------------------- -// SID clocking with audio sampling. -// Fixpoint arithmetic (22.10 bits) 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 SID::clock(cycle_count& delta_t, short* buf, int n, int interleave) -{ - switch (sampling) { - default: - case SAMPLE_FAST: - return clock_fast(delta_t, buf, n, interleave); - case SAMPLE_INTERPOLATE: - return clock_interpolate(delta_t, buf, n, interleave); - case SAMPLE_RESAMPLE: - return clock_resample(delta_t, buf, n, interleave); - } -} - -// ---------------------------------------------------------------------------- -// SID clocking with audio sampling - delta clocking picking nearest sample. -// ---------------------------------------------------------------------------- -RESID_INLINE -int SID::clock_fast(cycle_count& delta_t, short* buf, int n, - int interleave) -{ - int s = 0; - - for (;;) { - cycle_count next_sample_offset = sample_offset + cycles_per_sample + (1 << 9); - cycle_count delta_t_sample = next_sample_offset >> 10; - if (delta_t_sample > delta_t) { - break; - } - if (s >= n) { - return s; - } - clock(delta_t_sample); - delta_t -= delta_t_sample; - sample_offset = (next_sample_offset & 0x3ff) - (1 << 9); - buf[s++*interleave] = output(); - } - - clock(delta_t); - sample_offset -= delta_t << 10; - delta_t = 0; - return s; -} - - -// ---------------------------------------------------------------------------- -// 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 SID::clock_interpolate(cycle_count& delta_t, short* buf, int n, - int interleave) -{ - int s = 0; - int i; - - for (;;) { - cycle_count next_sample_offset = sample_offset + cycles_per_sample; - cycle_count delta_t_sample = next_sample_offset >> 10; - 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 & 0x3ff; - - short sample_now = output(); - buf[s++*interleave] = - sample_prev + (sample_offset*(sample_now - sample_prev) >> 10); - 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 << 10; - 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/ -// -// NB! The sample ring buffer requires two's complement integer, and -// the result of right shifting negative numbers is really implementation -// dependent in the C++ standard. It is crucial for speed, however. -// ---------------------------------------------------------------------------- -RESID_INLINE -int SID::clock_resample(cycle_count& delta_t, short* buf, int n, - int interleave) -{ - int s = 0; - - for (;;) { - cycle_count next_sample_offset = sample_offset + cycles_per_sample; - cycle_count delta_t_sample = next_sample_offset >> 10; - if (delta_t_sample > delta_t) { - break; - } - if (s >= n) { - return s; - } - for (int i = 0; i < delta_t_sample; i++) { - clock(); - sample[sample_index++] = output(); - sample_index &= 0x3fff; - } - delta_t -= delta_t_sample; - sample_offset = next_sample_offset & 0x3ff; - - int v = 0; - int filter_offset = sample_offset*fstep_per_cycle >> 10; - int foffset; - - // Convolution with right wing of filter impulse response. - unsigned int j = (sample_index - sample_delay - 1) & 0x3fff; - for (foffset = filter_offset; - foffset <= foffset_max; - foffset += fstep_per_cycle) - { - int findex = foffset >> 10; - int frmd = foffset & 0x3ff; - v += sample[j--]*(fir[findex] + (frmd*fir_diff[findex] >> 10)); - j &= 0x3fff; - } - - // Convolution with left wing of filter impulse response. - j = (sample_index - sample_delay) & 0x3fff; - for (foffset = fstep_per_cycle - filter_offset; - foffset <= foffset_max; - foffset += fstep_per_cycle) - { - int findex = foffset >> 10; - int frmd = foffset & 0x3ff; - v += sample[j++]*(fir[findex] + (frmd*fir_diff[findex] >> 10)); - j &= 0x3fff; - } - - buf[s++*interleave] = v >> FIR_SHIFT; - } - - for (int i = 0; i < delta_t; i++) { - clock(); - sample[sample_index++] = output(); - sample_index &= 0x3fff; - } - sample_offset -= delta_t << 10; - delta_t = 0; - return s; -} - -RESID_NAMESPACE_STOP |