// --------------------------------------------------------------------------- // This file is part of reSID, a MOS6581 SID emulator engine. // Copyright (C) 2004 Dag Lem // // 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 #include RESID_NAMESPACE_START // Resampling constants. // The error in interpolated lookup is bounded by 1.234/L^2, // while the error in non-interpolated lookup is bounded by // 0.7854/L + 0.4113/L^2, see // http://www-ccrma.stanford.edu/~jos/resample/Choice_Table_Size.html // For a resolution of 16 bits this yields L >= 285 and L >= 51473, // respectively. const int SID::FIR_N = 125; const int SID::FIR_RES_INTERPOLATE = 285; const int SID::FIR_RES_FAST = 51473; const int SID::FIR_SHIFT = 15; const int SID::RINGSIZE = 16384; // Fixpoint constants (16.16 bits). const int SID::FIXP_SHIFT = 16; const int SID::FIXP_MASK = 0xffff; // ---------------------------------------------------------------------------- // Constructor. // ---------------------------------------------------------------------------- SID::SID() { // 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_FAST, AUDIO_SAMPLE_RATE_EXACT); bus_value = 0; bus_value_ttl = 0; ext_in = 0; } // ---------------------------------------------------------------------------- // Destructor. // ---------------------------------------------------------------------------- SID::~SID() { delete[] sample; delete[] fir; } // ---------------------------------------------------------------------------- // Set chip model. // ---------------------------------------------------------------------------- /* void SID::set_chip_model(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); } */ // ---------------------------------------------------------------------------- // SID reset. // ---------------------------------------------------------------------------- void SID::reset() { voice[0].reset(); voice[1].reset(); voice[2].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 SID::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 = (sample << 4)*3; } // ---------------------------------------------------------------------------- // Read sample from 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); asm ("ssat %0, #16, %1" : "=r" (sample) : "r" (sample)); /* 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] = 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] = EnvelopeGenerator::RELEASE; hold_zero[i] = true; } } // ---------------------------------------------------------------------------- // 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.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 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.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 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. // ---------------------------------------------------------------------------- float SID::I0(float x) { // Max error acceptable in I0. const float I0e = 1e-6; float 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 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) { // Check whether the sample ring buffer would overfill. if (FIR_N*clock_freq/sample_freq >= RINGSIZE) { 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 (2.0*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/2.0) { return false; } // The filter scaling is only included to avoid clipping, so keep // it sane. if (filter_scale < 0.9 || filter_scale > 1.0) { 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 << FIXP_SHIFT) + 0.5); sample_offset = 0; sample_prev = 0; // FIR initialization is only necessary for resampling. if (method != SAMPLE_RESAMPLE_INTERPOLATE && method != SAMPLE_RESAMPLE_FAST) { delete[] sample; delete[] fir; sample = 0; fir = 0; return true; } const float pi = 3.1415926535897932385; // 16 bits -> -96dB stopband attenuation. const float A = -20.0*log10(1.0/(1 << 16)); // A fraction of the bandwidth is allocated to the transition band, float dw = (1.0 - 2.0*pass_freq/sample_freq)*pi; // The cutoff frequency is midway through the transition band. 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 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 // The filter order is equal to the number of zero crossings, i.e. // it should be an even number (sinc is symmetric about x = 0). int N = int((A - 7.95)/(2.285*dw) + 0.5); N += N & 1; 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). fir_N = int(N*f_cycles_per_sample) + 1; fir_N |= 1; // We clamp the filter table resolution to 2^n, making the fixpoint // sample_offset a whole multiple of the filter table resolution. int res = method == SAMPLE_RESAMPLE_INTERPOLATE ? FIR_RES_INTERPOLATE : FIR_RES_FAST; int n = (int)ceil(log(res/f_cycles_per_sample)/log(2)); fir_RES = 1 << n; // Allocate memory for FIR tables. delete[] fir; fir = new short[fir_N*fir_RES]; // 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; 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++) { 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; float sincwt = fabs(wt) >= 1e-6 ? sin(wt)/wt : 1; float val = (1 << FIR_SHIFT)*filter_scale*f_samples_per_cycle*wc/pi*sincwt*Kaiser; fir[fir_offset + j] = short(val + 0.5); } } // Allocate sample buffer. if (!sample) { sample = new short[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 SID::adjust_sampling_frequency(float sample_freq) { cycles_per_sample = cycle_count(clock_frequency/sample_freq*(1 << FIXP_SHIFT) + 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 SID::fc_plotter() { return filter.fc_plotter(); } // ---------------------------------------------------------------------------- // SID clocking - 1 cycle. // ---------------------------------------------------------------------------- void SID::clock() { // Age bus value. if (--bus_value_ttl <= 0) { bus_value = 0; bus_value_ttl = 0; } // Clock amplitude modulators. voice[0].envelope.clock(); voice[1].envelope.clock(); voice[2].envelope.clock(); // Clock oscillators. voice[0].wave.clock(); voice[1].wave.clock(); voice[2].wave.clock(); // Synchronize oscillators. 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); // 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. 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; 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. voice[0].wave.clock(delta_t_min); voice[1].wave.clock(delta_t_min); voice[2].wave.clock(delta_t_min); // Synchronize oscillators. voice[0].wave.synchronize(); voice[1].wave.synchronize(); voice[2].wave.synchronize(); delta_t_osc -= delta_t_min; } // Clock filter. filter.clock(delta_t, voice[0].output(), voice[1].output(), voice[2].output(), ext_in); // Clock external filter. extfilt.clock(delta_t, filter.output()); } // ---------------------------------------------------------------------------- // 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 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_INTERPOLATE: return clock_resample_interpolate(delta_t, buf, n, interleave); case SAMPLE_RESAMPLE_FAST: return clock_resample_fast(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 << (FIXP_SHIFT - 1)); cycle_count delta_t_sample = next_sample_offset >> FIXP_SHIFT; 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 & FIXP_MASK) - (1 << (FIXP_SHIFT - 1)); buf[s++*interleave] = output(); } clock(delta_t); sample_offset -= delta_t << FIXP_SHIFT; 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 >> FIXP_SHIFT; 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 & FIXP_MASK; short sample_now = output(); buf[s++*interleave] = sample_prev + (sample_offset*(sample_now - sample_prev) >> FIXP_SHIFT); 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 << FIXP_SHIFT; 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 SID::clock_resample_interpolate(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 >> FIXP_SHIFT; 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] = sample[sample_index + RINGSIZE] = output(); ++sample_index; sample_index &= 0x3fff; } delta_t -= delta_t_sample; sample_offset = next_sample_offset & FIXP_MASK; int fir_offset = sample_offset*fir_RES >> FIXP_SHIFT; int fir_offset_rmd = sample_offset*fir_RES & FIXP_MASK; short* fir_start = fir + fir_offset*fir_N; short* sample_start = sample + sample_index - fir_N + RINGSIZE; // Convolution with filter impulse response. int v1 = 0; for (int j = 0; j < fir_N; j++) { v1 += sample_start[j]*fir_start[j]; } // Use next FIR table, wrap around to first FIR table using // previous sample. if (++fir_offset == fir_RES) { fir_offset = 0; --sample_start; } fir_start = fir + fir_offset*fir_N; // Convolution with filter impulse response. int v2 = 0; for (int j = 0; j < fir_N; j++) { v2 += sample_start[j]*fir_start[j]; } // Linear interpolation. // fir_offset_rmd is equal for all samples, it can thus be factorized out: // sum(v1 + rmd*(v2 - v1)) = sum(v1) + rmd*(sum(v2) - sum(v1)) int v = v1 + (fir_offset_rmd*(v2 - v1) >> FIXP_SHIFT); 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; } else if (v < -half) { v = -half; } */ buf[s++*interleave] = v; } for (int i = 0; i < delta_t; i++) { clock(); sample[sample_index] = sample[sample_index + RINGSIZE] = output(); ++sample_index; sample_index &= 0x3fff; } sample_offset -= delta_t << FIXP_SHIFT; delta_t = 0; return s; } // ---------------------------------------------------------------------------- // SID clocking with audio sampling - cycle based with audio resampling. // ---------------------------------------------------------------------------- RESID_INLINE int SID::clock_resample_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; cycle_count delta_t_sample = next_sample_offset >> FIXP_SHIFT; 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] = sample[sample_index + RINGSIZE] = output(); ++sample_index; sample_index &= 0x3fff; } delta_t -= delta_t_sample; sample_offset = next_sample_offset & FIXP_MASK; int fir_offset = sample_offset*fir_RES >> FIXP_SHIFT; short* fir_start = fir + fir_offset*fir_N; short* sample_start = sample + sample_index - fir_N + RINGSIZE; // Convolution with filter impulse response. int v = 0; for (int j = 0; j < fir_N; j++) { v += sample_start[j]*fir_start[j]; } 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; } else if (v < -half) { v = -half; } */ buf[s++*interleave] = v; } for (int i = 0; i < delta_t; i++) { clock(); sample[sample_index] = sample[sample_index + RINGSIZE] = output(); ++sample_index; sample_index &= 0x3fff; } sample_offset -= delta_t << FIXP_SHIFT; delta_t = 0; return s; } RESID_NAMESPACE_STOP