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1059 lines
31 KiB
1059 lines
31 KiB
// ---------------------------------------------------------------------------
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// This file is part of reSID, a MOS6581 SID emulator engine.
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// Copyright (C) 2004 Dag Lem <resid@nimrod.no>
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//
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// This program is free software; you can redistribute it and/or modify
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// it under the terms of the GNU General Public License as published by
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// the Free Software Foundation; either version 2 of the License, or
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// (at your option) any later version.
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//
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// This program is distributed in the hope that it will be useful,
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// but WITHOUT ANY WARRANTY; without even the implied warranty of
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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// GNU General Public License for more details.
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//
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// You should have received a copy of the GNU General Public License
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// along with this program; if not, write to the Free Software
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// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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// ---------------------------------------------------------------------------
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#include "sid.h"
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#include <math.h>
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#include <AudioStream.h>
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RESID_NAMESPACE_START
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// Resampling constants.
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// The error in interpolated lookup is bounded by 1.234/L^2,
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// while the error in non-interpolated lookup is bounded by
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// 0.7854/L + 0.4113/L^2, see
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// http://www-ccrma.stanford.edu/~jos/resample/Choice_Table_Size.html
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// For a resolution of 16 bits this yields L >= 285 and L >= 51473,
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// respectively.
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const int SID::FIR_N = 125;
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const int SID::FIR_RES_INTERPOLATE = 285;
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const int SID::FIR_RES_FAST = 51473;
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const int SID::FIR_SHIFT = 15;
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const int SID::RINGSIZE = 16384;
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// Fixpoint constants (16.16 bits).
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const int SID::FIXP_SHIFT = 16;
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const int SID::FIXP_MASK = 0xffff;
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// ----------------------------------------------------------------------------
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// Constructor.
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// ----------------------------------------------------------------------------
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SID::SID()
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{
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// Initialize pointers.
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sample = 0;
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fir = 0;
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voice[0].set_sync_source(&voice[2]);
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voice[1].set_sync_source(&voice[0]);
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voice[2].set_sync_source(&voice[1]);
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set_sampling_parameters(985248, SAMPLE_FAST, AUDIO_SAMPLE_RATE_EXACT);
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bus_value = 0;
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bus_value_ttl = 0;
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ext_in = 0;
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}
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// ----------------------------------------------------------------------------
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// Destructor.
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// ----------------------------------------------------------------------------
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SID::~SID()
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{
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delete[] sample;
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delete[] fir;
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}
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// ----------------------------------------------------------------------------
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// Set chip model.
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// ----------------------------------------------------------------------------
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/*
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void SID::set_chip_model(chip_model model)
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{
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voice[0].set_chip_model(model);
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voice[1].set_chip_model(model);
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voice[2].set_chip_model(model);
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filter.set_chip_model(model);
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extfilt.set_chip_model(model);
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}
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*/
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// ----------------------------------------------------------------------------
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// SID reset.
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// ----------------------------------------------------------------------------
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void SID::reset()
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{
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voice[0].reset();
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voice[1].reset();
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voice[2].reset();
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filter.reset();
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extfilt.reset();
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bus_value = 0;
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bus_value_ttl = 0;
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}
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// ----------------------------------------------------------------------------
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// Write 16-bit sample to audio input.
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// NB! The caller is responsible for keeping the value within 16 bits.
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// Note that to mix in an external audio signal, the signal should be
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// resampled to 1MHz first to avoid sampling noise.
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// ----------------------------------------------------------------------------
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void SID::input(int sample)
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{
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// Voice outputs are 20 bits. Scale up to match three voices in order
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// to facilitate simulation of the MOS8580 "digi boost" hardware hack.
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ext_in = (sample << 4)*3;
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}
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// ----------------------------------------------------------------------------
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// Read sample from audio output.
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// Both 16-bit and n-bit output is provided.
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// ----------------------------------------------------------------------------
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int SID::output()
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{
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const int range = 1 << 16;
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//const int half = range >> 1;
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int sample = extfilt.output()/((4095*255 >> 7)*3*15*2/range);
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asm ("ssat %0, #16, %1" : "=r" (sample) : "r" (sample));
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/*
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if (sample >= half) {
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return half - 1;
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}
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if (sample < -half) {
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return -half;
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}
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*/
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return sample;
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}
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int SID::output(int bits)
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{
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const int range = 1 << bits;
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const int half = range >> 1;
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int sample = extfilt.output()/((4095*255 >> 7)*3*15*2/range);
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if (sample >= half) {
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return half - 1;
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}
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if (sample < -half) {
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return -half;
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}
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return sample;
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}
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// ----------------------------------------------------------------------------
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// Read registers.
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//
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// Reading a write only register returns the last byte written to any SID
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// register. The individual bits in this value start to fade down towards
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// zero after a few cycles. All bits reach zero within approximately
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// $2000 - $4000 cycles.
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// It has been claimed that this fading happens in an orderly fashion, however
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// sampling of write only registers reveals that this is not the case.
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// NB! This is not correctly modeled.
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// The actual use of write only registers has largely been made in the belief
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// that all SID registers are readable. To support this belief the read
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// would have to be done immediately after a write to the same register
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// (remember that an intermediate write to another register would yield that
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// value instead). With this in mind we return the last value written to
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// any SID register for $2000 cycles without modeling the bit fading.
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// ----------------------------------------------------------------------------
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reg8 SID::read(reg8 offset)
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{
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switch (offset) {
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case 0x19:
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return potx.readPOT();
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case 0x1a:
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return poty.readPOT();
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case 0x1b:
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return voice[2].wave.readOSC();
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case 0x1c:
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return voice[2].envelope.readENV();
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default:
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return bus_value;
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}
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}
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// ----------------------------------------------------------------------------
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// Write registers.
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// ----------------------------------------------------------------------------
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void SID::write(reg8 offset, reg8 value)
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{
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bus_value = value;
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bus_value_ttl = 0x2000;
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switch (offset) {
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case 0x00:
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voice[0].wave.writeFREQ_LO(value);
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break;
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case 0x01:
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voice[0].wave.writeFREQ_HI(value);
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break;
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case 0x02:
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voice[0].wave.writePW_LO(value);
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break;
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case 0x03:
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voice[0].wave.writePW_HI(value);
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break;
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case 0x04:
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voice[0].writeCONTROL_REG(value);
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break;
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case 0x05:
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voice[0].envelope.writeATTACK_DECAY(value);
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break;
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case 0x06:
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voice[0].envelope.writeSUSTAIN_RELEASE(value);
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break;
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case 0x07:
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voice[1].wave.writeFREQ_LO(value);
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break;
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case 0x08:
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voice[1].wave.writeFREQ_HI(value);
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break;
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case 0x09:
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voice[1].wave.writePW_LO(value);
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break;
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case 0x0a:
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voice[1].wave.writePW_HI(value);
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break;
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case 0x0b:
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voice[1].writeCONTROL_REG(value);
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break;
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case 0x0c:
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voice[1].envelope.writeATTACK_DECAY(value);
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break;
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case 0x0d:
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voice[1].envelope.writeSUSTAIN_RELEASE(value);
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break;
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case 0x0e:
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voice[2].wave.writeFREQ_LO(value);
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break;
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case 0x0f:
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voice[2].wave.writeFREQ_HI(value);
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break;
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case 0x10:
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voice[2].wave.writePW_LO(value);
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break;
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case 0x11:
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voice[2].wave.writePW_HI(value);
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break;
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case 0x12:
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voice[2].writeCONTROL_REG(value);
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break;
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case 0x13:
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voice[2].envelope.writeATTACK_DECAY(value);
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break;
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case 0x14:
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voice[2].envelope.writeSUSTAIN_RELEASE(value);
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break;
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case 0x15:
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filter.writeFC_LO(value);
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break;
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case 0x16:
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filter.writeFC_HI(value);
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break;
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case 0x17:
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filter.writeRES_FILT(value);
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break;
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case 0x18:
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filter.writeMODE_VOL(value);
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break;
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default:
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break;
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}
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}
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// ----------------------------------------------------------------------------
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// SID voice muting.
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// ----------------------------------------------------------------------------
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void SID::mute(reg8 channel, bool enable)
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{
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// Only have 3 voices!
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if (channel >= 3)
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return;
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voice[channel].mute (enable);
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}
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// ----------------------------------------------------------------------------
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// Constructor.
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// ----------------------------------------------------------------------------
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SID::State::State()
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{
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int i;
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for (i = 0; i < 0x20; i++) {
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sid_register[i] = 0;
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}
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bus_value = 0;
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bus_value_ttl = 0;
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for (i = 0; i < 3; i++) {
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accumulator[i] = 0;
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shift_register[i] = 0x7ffff8;
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rate_counter[i] = 0;
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rate_counter_period[i] = 9;
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exponential_counter[i] = 0;
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exponential_counter_period[i] = 1;
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envelope_counter[i] = 0;
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envelope_state[i] = EnvelopeGenerator::RELEASE;
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hold_zero[i] = true;
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}
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}
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// ----------------------------------------------------------------------------
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// Read state.
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// ----------------------------------------------------------------------------
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SID::State SID::read_state()
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{
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State state;
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int i, j;
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for (i = 0, j = 0; i < 3; i++, j += 7) {
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WaveformGenerator& wave = voice[i].wave;
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EnvelopeGenerator& envelope = voice[i].envelope;
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state.sid_register[j + 0] = wave.freq & 0xff;
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state.sid_register[j + 1] = wave.freq >> 8;
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state.sid_register[j + 2] = wave.pw & 0xff;
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state.sid_register[j + 3] = wave.pw >> 8;
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state.sid_register[j + 4] =
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(wave.waveform << 4)
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| (wave.test ? 0x08 : 0)
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| (wave.ring_mod ? 0x04 : 0)
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| (wave.sync ? 0x02 : 0)
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| (envelope.gate ? 0x01 : 0);
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state.sid_register[j + 5] = (envelope.attack << 4) | envelope.decay;
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state.sid_register[j + 6] = (envelope.sustain << 4) | envelope.release;
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}
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state.sid_register[j++] = filter.fc & 0x007;
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state.sid_register[j++] = filter.fc >> 3;
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state.sid_register[j++] = (filter.res << 4) | filter.filt;
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state.sid_register[j++] =
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(filter.voice3off ? 0x80 : 0)
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| (filter.hp_bp_lp << 4)
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| filter.vol;
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// These registers are superfluous, but included for completeness.
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for (; j < 0x1d; j++) {
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state.sid_register[j] = read(j);
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}
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for (; j < 0x20; j++) {
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state.sid_register[j] = 0;
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}
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state.bus_value = bus_value;
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state.bus_value_ttl = bus_value_ttl;
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for (i = 0; i < 3; i++) {
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state.accumulator[i] = voice[i].wave.accumulator;
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state.shift_register[i] = voice[i].wave.shift_register;
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state.rate_counter[i] = voice[i].envelope.rate_counter;
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state.rate_counter_period[i] = voice[i].envelope.rate_period;
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state.exponential_counter[i] = voice[i].envelope.exponential_counter;
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state.exponential_counter_period[i] = voice[i].envelope.exponential_counter_period;
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state.envelope_counter[i] = voice[i].envelope.envelope_counter;
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state.envelope_state[i] = voice[i].envelope.state;
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state.hold_zero[i] = voice[i].envelope.hold_zero;
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}
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return state;
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}
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// ----------------------------------------------------------------------------
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// Write state.
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// ----------------------------------------------------------------------------
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void SID::write_state(const State& state)
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{
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int i;
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for (i = 0; i <= 0x18; i++) {
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write(i, state.sid_register[i]);
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}
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bus_value = state.bus_value;
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bus_value_ttl = state.bus_value_ttl;
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for (i = 0; i < 3; i++) {
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voice[i].wave.accumulator = state.accumulator[i];
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voice[i].wave.shift_register = state.shift_register[i];
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voice[i].envelope.rate_counter = state.rate_counter[i];
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voice[i].envelope.rate_period = state.rate_counter_period[i];
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voice[i].envelope.exponential_counter = state.exponential_counter[i];
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voice[i].envelope.exponential_counter_period = state.exponential_counter_period[i];
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voice[i].envelope.envelope_counter = state.envelope_counter[i];
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voice[i].envelope.state = state.envelope_state[i];
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voice[i].envelope.hold_zero = state.hold_zero[i];
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}
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}
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// ----------------------------------------------------------------------------
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// Enable filter.
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// ----------------------------------------------------------------------------
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void SID::enable_filter(bool enable)
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{
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filter.enable_filter(enable);
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}
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// ----------------------------------------------------------------------------
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// Enable external filter.
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// ----------------------------------------------------------------------------
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void SID::enable_external_filter(bool enable)
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{
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extfilt.enable_filter(enable);
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}
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// ----------------------------------------------------------------------------
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// I0() computes the 0th order modified Bessel function of the first kind.
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// This function is originally from resample-1.5/filterkit.c by J. O. Smith.
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// ----------------------------------------------------------------------------
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float SID::I0(float x)
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{
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// Max error acceptable in I0.
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const float I0e = 1e-6;
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float sum, u, halfx, temp;
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int n;
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sum = u = n = 1;
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halfx = x/2.0;
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do {
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temp = halfx/n++;
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u *= temp*temp;
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sum += u;
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} while (u >= I0e*sum);
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return sum;
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}
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// ----------------------------------------------------------------------------
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// Setting of SID sampling parameters.
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//
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// Use a clock freqency of 985248Hz for PAL C64, 1022730Hz for NTSC C64.
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// The default end of passband frequency is pass_freq = 0.9*sample_freq/2
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// for sample frequencies up to ~ 44.1kHz, and 20kHz for higher sample
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// frequencies.
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//
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// For resampling, the ratio between the clock frequency and the sample
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// frequency is limited as follows:
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// 125*clock_freq/sample_freq < 16384
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// E.g. provided a clock frequency of ~ 1MHz, the sample frequency can not
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// be set lower than ~ 8kHz. A lower sample frequency would make the
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// resampling code overfill its 16k sample ring buffer.
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//
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// The end of passband frequency is also limited:
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// pass_freq <= 0.9*sample_freq/2
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// E.g. for a 44.1kHz sampling rate the end of passband frequency is limited
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// to slightly below 20kHz. This constraint ensures that the FIR table is
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// not overfilled.
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// ----------------------------------------------------------------------------
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bool SID::set_sampling_parameters(float clock_freq, sampling_method method,
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float sample_freq, float pass_freq,
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float filter_scale)
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{
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// Check resampling constraints.
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if (method == SAMPLE_RESAMPLE_INTERPOLATE || method == SAMPLE_RESAMPLE_FAST)
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{
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// Check whether the sample ring buffer would overfill.
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if (FIR_N*clock_freq/sample_freq >= RINGSIZE) {
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return false;
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}
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}
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// The default passband limit is 0.9*sample_freq/2 for sample
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// frequencies below ~ 44.1kHz, and 20kHz for higher sample frequencies.
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if (pass_freq < 0) {
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pass_freq = 20000;
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if (2.0*pass_freq/sample_freq >= 0.9) {
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pass_freq = 0.9*sample_freq/2.0;
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}
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}
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// Check whether the FIR table would overfill.
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else if (pass_freq > 0.9*sample_freq/2.0) {
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return false;
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}
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// The filter scaling is only included to avoid clipping, so keep
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// it sane.
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if (filter_scale < 0.9 || filter_scale > 1.0) {
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return false;
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}
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// Set the external filter to the pass freq
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extfilt.set_sampling_parameter (pass_freq);
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clock_frequency = clock_freq;
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sampling = method;
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cycles_per_sample =
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cycle_count(clock_freq/sample_freq*(1 << FIXP_SHIFT) + 0.5);
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sample_offset = 0;
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sample_prev = 0;
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// FIR initialization is only necessary for resampling.
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if (method != SAMPLE_RESAMPLE_INTERPOLATE && method != SAMPLE_RESAMPLE_FAST)
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{
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delete[] sample;
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delete[] fir;
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sample = 0;
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fir = 0;
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return true;
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}
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|
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const float pi = 3.1415926535897932385;
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|
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// 16 bits -> -96dB stopband attenuation.
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const float A = -20.0*log10(1.0/(1 << 16));
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// A fraction of the bandwidth is allocated to the transition band,
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float dw = (1.0 - 2.0*pass_freq/sample_freq)*pi;
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|
// 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<sound_sample> 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
|
|
|