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Removes the crutch of my first-attempt implementation.
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//
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// Channel.cpp
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// Clock Signal
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//
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// Created by Thomas Harte on 15/04/2020.
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// Copyright © 2020 Thomas Harte. All rights reserved.
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//
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#include "Channel.hpp"
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using namespace Yamaha::OPL;
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void Channel::set_frequency_low(uint8_t value) {
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period_ = (period_ &~0xff) | value;
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}
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void Channel::set_10bit_frequency_octave_key_on(uint8_t value) {
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period_ = (period_ & 0xff) | ((value & 3) << 8);
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octave_ = (value >> 2) & 0x7;
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key_on_ = value & 0x20;
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frequency_shift_ = 0;
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}
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void Channel::set_9bit_frequency_octave_key_on(uint8_t value) {
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period_ = (period_ & 0xff) | ((value & 1) << 8);
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octave_ = (value >> 1) & 0x7;
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key_on_ = value & 0x10;;
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frequency_shift_ = 1;
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}
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void Channel::set_feedback_mode(uint8_t value) {
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feedback_strength_ = (value >> 1) & 0x7;
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use_fm_synthesis_ = value & 1;
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}
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void Channel::update(bool modulator, const LowFrequencyOscillator &oscillator, Operator &op, bool force_key_on, OperatorOverrides *overrides) {
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op.update(states_[int(modulator)], oscillator, key_on_ || force_key_on, period_ << frequency_shift_, octave_, overrides);
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}
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int Channel::melodic_output(const Operator &modulator, const Operator &carrier, const OperatorOverrides *overrides) const {
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if(use_fm_synthesis_) {
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// Get modulator level, use that as a phase-adjusting input to the carrier and then return the carrier level.
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const LogSign modulator_output = modulator.melodic_output(states_[1]);
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return carrier.melodic_output(states_[0], &modulator_output, overrides).level();
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} else {
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// Get modulator and carrier levels separately, return their sum.
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return (carrier.melodic_output(states_[0], nullptr, overrides).level() + modulator.melodic_output(states_[1], nullptr, overrides).level()) >> 1;
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}
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}
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int Channel::tom_tom_output(const Operator &modulator, const OperatorOverrides *overrides) const {
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return modulator.melodic_output(states_[1], nullptr, overrides).level();
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}
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int Channel::snare_output(const Operator &carrier, const OperatorOverrides *overrides) const {
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return carrier.snare_output(states_[0], overrides).level();
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}
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int Channel::cymbal_output(const Operator &modulator, const Operator &carrier, const Channel &channel8, const OperatorOverrides *overrides) const {
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return carrier.cymbal_output(states_[0], channel8.states_[1], overrides).level();
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}
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int Channel::high_hat_output(const Operator &modulator, const Operator &carrier, const Channel &channel8, const OperatorOverrides *overrides) const {
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return carrier.high_hat_output(states_[0], channel8.states_[1], overrides).level();
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}
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bool Channel::is_audible(Operator *carrier, OperatorOverrides *overrides) {
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return carrier->is_audible(states_[0], overrides);
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}
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//
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// Channel.hpp
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// Clock Signal
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//
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// Created by Thomas Harte on 15/04/2020.
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// Copyright © 2020 Thomas Harte. All rights reserved.
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//
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#ifndef Channel_hpp
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#define Channel_hpp
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#include "LowFrequencyOscillator.hpp"
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#include "Operator.hpp"
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namespace Yamaha {
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namespace OPL {
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/*!
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Models an L-type two-operator channel.
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Assuming FM synthesis is enabled, the channel modulates the output of the carrier with that of the modulator.
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TODO: make this a template on period counter size in bits?
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*/
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class Channel {
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public:
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/// Sets the low 8 bits of frequency control.
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void set_frequency_low(uint8_t value);
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/// Sets the high two bits of a 10-bit frequency control, along with this channel's
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/// block/octave, and key on or off.
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void set_10bit_frequency_octave_key_on(uint8_t value);
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/// Sets the high two bits of a 9-bit frequency control, along with this channel's
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/// block/octave, and key on or off.
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void set_9bit_frequency_octave_key_on(uint8_t value);
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/// Sets the amount of feedback provided to the first operator (i.e. the modulator)
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/// associated with this channel, and whether FM synthesis is in use.
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void set_feedback_mode(uint8_t value);
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/// Updates one of this channel's operators.
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void update(bool modulator, const LowFrequencyOscillator &oscillator, Operator &op, bool force_key_on = false, OperatorOverrides *overrides = nullptr);
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/// Gets regular 'melodic' output for this channel, i.e. the output you'd expect from all channels when not in rhythm mode.
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int melodic_output(const Operator &modulator, const Operator &carrier, const OperatorOverrides *overrides = nullptr) const;
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/// Generates tom tom output using this channel's modulator.
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int tom_tom_output(const Operator &modulator, const OperatorOverrides *overrides = nullptr) const;
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/// Generates snare output, using this channel's carrier.
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int snare_output(const Operator &carrier, const OperatorOverrides *overrides = nullptr) const;
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/// Generates cymbal output, using this channel's modulator and @c channel8 's carrier.
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int cymbal_output(const Operator &modulator, const Operator &carrier, const Channel &channel8, const OperatorOverrides *overrides = nullptr) const;
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/// Generates cymbal output, using this channel's modulator and @c channel8 's carrier.
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int high_hat_output(const Operator &modulator, const Operator &carrier, const Channel &channel8, const OperatorOverrides *overrides = nullptr) const;
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/// @returns @c true if this channel is currently producing any audio; @c false otherwise;
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bool is_audible(Operator *carrier, OperatorOverrides *carrier_overrides = nullptr);
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private:
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/// 'F-Num' in the spec; this plus the current octave determines channel frequency.
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int period_ = 0;
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/// Linked with the frequency, determines the channel frequency.
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int octave_ = 0;
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/// Sets sets this channel on or off, as an input to the ADSR envelope,
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bool key_on_ = false;
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/// Sets the degree of feedback applied to the modulator.
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int feedback_strength_ = 0;
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/// Selects between FM synthesis, using the modulator to modulate the carrier, or simple mixing of the two
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/// underlying operators as completely disjoint entities.
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bool use_fm_synthesis_ = true;
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/// Used internally to make both the 10-bit OPL2 frequency selection and 9-bit OPLL/VRC7 frequency
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/// selections look the same when passed to the operators.
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int frequency_shift_ = 0;
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// Operator state is stored distinctly from Operators because
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// carrier/modulator may not be unique per channel —
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// on the OPLL there's an extra level of indirection.
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OperatorState states_[2];
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};
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}
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}
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#endif /* Channel_hpp */
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//
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// Operator.cpp
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// Clock Signal
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//
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// Created by Thomas Harte on 15/04/2020.
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// Copyright © 2020 Thomas Harte. All rights reserved.
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//
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#include "Operator.hpp"
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#include <algorithm>
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#include <cassert>
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using namespace Yamaha::OPL;
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// MARK: - Setters
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void Operator::set_attack_decay(uint8_t value) {
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attack_rate_ = (value & 0xf0) >> 2;
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decay_rate_ = (value & 0x0f) << 2;
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}
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void Operator::set_sustain_release(uint8_t value) {
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sustain_level_ = (value & 0xf0) >> 4;
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release_rate_ = (value & 0x0f) << 2;
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}
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void Operator::set_scaling_output(uint8_t value) {
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key_level_scaling_ = value >> 6;
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attenuation_ = value & 0x3f;
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}
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void Operator::set_waveform(uint8_t value) {
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waveform_ = Operator::Waveform(value & 3);
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}
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void Operator::set_am_vibrato_hold_sustain_ksr_multiple(uint8_t value) {
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apply_amplitude_modulation_ = value & 0x80;
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apply_vibrato_ = value & 0x40;
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use_sustain_level_ = value & 0x20;
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key_rate_scaling_shift_ = (value & 0x10) ? 0 : 2;
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frequency_multiple_ = value & 0xf;
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}
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// MARK: - Getter
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bool Operator::is_audible(OperatorState &state, OperatorOverrides *overrides) {
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// TODO: (i) do I actually want to support this functionality? (ii) if so, fix below.
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if(state.adsr_phase_ == OperatorState::ADSRPhase::Release) {
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if(overrides) {
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if(overrides->attenuation == 0xf) return false;
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} else {
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if(attenuation_ == 0x3f) return false;
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}
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}
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return state.adsr_attenuation_ != 511;
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}
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// MARK: - Update logic.
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void Operator::update_adsr(
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OperatorState &state,
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const LowFrequencyOscillator &oscillator,
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bool key_on,
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int channel_period,
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int channel_octave,
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const OperatorOverrides *overrides) {
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// Key-on logic: any time it is false, be in the release state.
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// On the leading edge of it becoming true, enter the attack state.
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if(!key_on) {
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state.adsr_phase_ = OperatorState::ADSRPhase::Release;
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} else if(!state.last_key_on_) {
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state.adsr_phase_ = OperatorState::ADSRPhase::Attack;
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state.attack_time_ = 0;
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// TODO: should this happen only if current ADSR attenuation is 511?
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state.raw_phase_ = 0;
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}
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state.last_key_on_ = key_on;
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// Adjust the ADSR attenuation appropriately;
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// cf. http://forums.submarine.org.uk/phpBB/viewtopic.php?f=9&t=16 (primarily) for the source of the maths below.
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// "An attack rate value of 52 (AR = 13) has 32 samples in the attack phase, an attack rate value of 48 (AR = 12)
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// has 64 samples in the attack phase, but pairs of samples show the same envelope attenuation. I am however struggling to find a plausible algorithm to match the experimental results.
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const int key_scaling_rate = ((channel_octave << 1) | (channel_period >> 9)) >> key_rate_scaling_shift_;
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assert(key_scaling_rate < 16);
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assert((channel_period >> 9) < 2);
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switch(state.adsr_phase_) {
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case OperatorState::ADSRPhase::Attack: {
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const int attack_rate = attack_rate_ + key_scaling_rate;
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// Rules:
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//
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// An attack rate of '13' has 32 samples in the attack phase; a rate of '12' has the same 32 steps, but spread out over 64 samples, etc.
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// An attack rate of '14' uses a divide by four instead of two.
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// 15 is instantaneous.
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if(attack_rate >= 56) {
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state.adsr_attenuation_ = state.adsr_attenuation_ - (state.adsr_attenuation_ >> 2) - 1;
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} else {
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const int sample_length = 1 << (14 - (attack_rate >> 2)); // TODO: don't throw away KSR bits.
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if(!(state.attack_time_ & (sample_length - 1))) {
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state.adsr_attenuation_ = state.adsr_attenuation_ - (state.adsr_attenuation_ >> 3) - 1;
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}
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}
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// Two possible terminating conditions: (i) the attack rate is 15; (ii) full volume has been reached.
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if(attack_rate > 60 || state.adsr_attenuation_ <= 0) {
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state.adsr_attenuation_ = 0;
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state.adsr_phase_ = OperatorState::ADSRPhase::Decay;
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}
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} break;
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case OperatorState::ADSRPhase::Release:
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case OperatorState::ADSRPhase::Decay:
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{
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// Rules:
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//
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// (relative to a 511 scale)
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//
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// A rate of 0 is no decay at all.
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// A rate of 1 means increase 4 per cycle.
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// A rate of 2 means increase 2 per cycle.
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// A rate of 3 means increase 1 per cycle.
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// A rate of 4 means increase 1 every other cycle.
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// A rate of 5 means increase once every fourth cycle.
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// etc.
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// eighth, sixteenth, 32nd, 64th, 128th, 256th, 512th, 1024th, 2048th, 4096th, 8192th
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const int decrease_rate = key_scaling_rate + ((state.adsr_phase_ == OperatorState::ADSRPhase::Decay) ? decay_rate_ : release_rate_);
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if(decrease_rate) {
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// TODO: don't throw away KSR bits.
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switch(decrease_rate >> 2) {
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case 1: state.adsr_attenuation_ += 32; break;
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case 2: state.adsr_attenuation_ += 16; break;
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default: {
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const int sample_length = 1 << ((decrease_rate >> 2) - 4);
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if(!(oscillator.counter & (sample_length - 1))) {
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state.adsr_attenuation_ += 8;
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}
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} break;
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}
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}
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// Clamp to the proper range.
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state.adsr_attenuation_ = std::min(state.adsr_attenuation_, 511);
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// Check for the decay exit condition.
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if(state.adsr_phase_ == OperatorState::ADSRPhase::Decay && state.adsr_attenuation_ >= (sustain_level_ << 3)) {
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state.adsr_attenuation_ = sustain_level_ << 3;
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state.adsr_phase_ = ((overrides && overrides->use_sustain_level) || use_sustain_level_) ? OperatorState::ADSRPhase::Sustain : OperatorState::ADSRPhase::Release;
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}
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} break;
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case OperatorState::ADSRPhase::Sustain:
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// Nothing to do.
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break;
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}
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++state.attack_time_;
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}
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void Operator::update_phase(OperatorState &state, const LowFrequencyOscillator &oscillator, int channel_period, int channel_octave) {
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// Per the documentation:
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//
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// Delta phase = ( [desired freq] * 2^19 / [input clock / 72] ) / 2 ^ (b - 1)
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//
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// After experimentation, I think this gives rate calculation as formulated below.
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// This encodes the MUL -> multiple table given on page 12,
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// multiplied by two.
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constexpr int multipliers[] = {
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1, 2, 4, 6, 8, 10, 12, 14, 16, 18, 20, 20, 24, 24, 30, 30
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};
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const int top_freq = channel_period >> 7;
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assert(top_freq < 8);
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constexpr int vibrato_shifts[8] = {3, 1, 0, 1, 3, 1, 0, 1};
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constexpr int vibrato_signs[2] = {1, -1};
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const int vibrato = (top_freq >> vibrato_shifts[oscillator.vibrato]) * vibrato_signs[oscillator.vibrato >> 2] * int(apply_vibrato_);
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// Update the raw phase.
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state.raw_phase_ += multipliers[frequency_multiple_] * (channel_period + vibrato) << channel_octave;
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}
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int Operator::key_level_scaling(const OperatorState &state, int channel_period, int channel_octave) const {
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// Calculate key-level scaling. Table is as per p14 of the YM3812 application manual,
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// converted into a fixed-point scheme. Compare with https://www.smspower.org/Development/RE12
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// and apologies for the highly ad hoc indentation.
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constexpr int key_level_scale_shifts[4] = {7, 1, 2, 0}; // '7' is just a number large enough to render all the numbers below as 0.
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constexpr int key_level_scales[8][16] = {
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#define _ 0
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// 6 db attenuations.
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{_, _, _, _, _, _, _, _, _, _, _, _, _, _, _, _},
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{_, _, _, _, _, _, _, _, _, 4, 6, 8, 10, 12, 14, 16},
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{_, _, _, _, _, 6, 10, 14, 16, 20, 22, 24, 26, 28, 30, 32},
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{_, _, _, 10, 16, 22, 26, 30, 32, 36, 38, 40, 42, 44, 46, 48},
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{_, _, 16, 26, 32, 38, 42, 46, 48, 52, 54, 56, 58, 60, 62, 64},
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{_, 16, 32, 42, 48, 54, 58, 62, 64, 68, 70, 72, 74, 76, 78, 80},
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{_, 32, 48, 58, 64, 70, 74, 78, 80, 84, 86, 88, 90, 92, 94, 96},
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{_, 48, 64, 74, 80, 86, 90, 94, 96, 100, 102, 104, 106, 108, 110, 112},
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#undef _
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};
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assert((channel_period >> 6) < 16);
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assert(channel_octave < 8);
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return (key_level_scales[channel_octave][channel_period >> 6] >> key_level_scale_shifts[key_level_scaling_]) << 7;
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}
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int Operator::adsr_tremolo_attenuation(const OperatorState &state, const LowFrequencyOscillator &oscillator) const {
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// Add optional tremolo to the current ADSR attenuation.
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return (state.adsr_attenuation_ << 3) + (int(apply_amplitude_modulation_) * oscillator.tremolo << 4);
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}
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int Operator::fixed_attenuation(const OperatorState &state, const OperatorOverrides *overrides) const {
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if(overrides) {
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// Overrides here represent per-channel volume on an OPLL. The bits are defined to represent
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// attenuations of 24db to 3db; the main envelope generator is stated to have a resolution of
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// 0.325db (which I've assumed is supposed to say 0.375db).
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return overrides->attenuation << 7;
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} else {
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// Overrides here represent per-channel volume on an OPLL. The bits are defined to represent
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// attenuations of 24db to 0.75db.
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return attenuation_ << 5;
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}
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}
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void Operator::update(
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OperatorState &state,
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const LowFrequencyOscillator &oscillator,
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bool key_on,
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int channel_period,
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int channel_octave,
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const OperatorOverrides *overrides) {
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update_adsr(state, oscillator, key_on, channel_period, channel_octave, overrides);
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update_phase(state, oscillator, channel_period, channel_octave);
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state.key_level_scaling_ = key_level_scaling(state, channel_period, channel_octave);
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state.adsr_tremolo_attenuation_ = adsr_tremolo_attenuation(state, oscillator);
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state.lfsr_ = oscillator.lfsr;
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}
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// TODO: both the tremolo and ADSR envelopes should be half-resolution on an OPLL.
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// MARK: - Output Generators.
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// A heavy debt is owed to https://github.com/andete/ym2413/blob/master/results/rhythm/rhythm.md regarding
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// the drum sound generation below.
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LogSign Operator::melodic_output(const OperatorState &state, const LogSign *phase_offset, const OperatorOverrides *overrides) const {
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// Calculate raw attenuation level.
|
||||
constexpr int waveforms[4][4] = {
|
||||
{1023, 1023, 1023, 1023}, // Sine: don't mask in any quadrant.
|
||||
{511, 511, 0, 0}, // Half sine: keep the first half intact, lock to 0 in the second half.
|
||||
{511, 511, 511, 511}, // AbsSine: endlessly repeat the first half of the sine wave.
|
||||
{255, 0, 255, 0}, // PulseSine: act as if the first quadrant is in the first and third; lock the other two to 0.
|
||||
};
|
||||
const int scaled_phase_offset = phase_offset ? phase_offset->level(11) : 0;
|
||||
const int phase = (state.raw_phase_ + scaled_phase_offset) >> 11;
|
||||
|
||||
LogSign result = negative_log_sin(phase & waveforms[int(waveform_)][(phase >> 8) & 3]);
|
||||
result += state.key_level_scaling_;
|
||||
result += state.adsr_tremolo_attenuation_ + fixed_attenuation(state, overrides);
|
||||
return result;
|
||||
}
|
||||
|
||||
LogSign Operator::snare_output(const OperatorState &state, const OperatorOverrides *overrides) const {
|
||||
LogSign result;
|
||||
|
||||
// If noise is 0, output is positive.
|
||||
// If noise is 1, output is negative.
|
||||
// If (noise ^ sign) is 0, output is 0. Otherwise it is max.
|
||||
const int sign = (state.raw_phase_ >> 11) & 0x200;
|
||||
const int level = ((state.raw_phase_ >> 20) & 1) ^ state.lfsr_;
|
||||
result = negative_log_sin(sign + (level << 8));
|
||||
|
||||
result += state.key_level_scaling_;
|
||||
result += state.adsr_tremolo_attenuation_ + fixed_attenuation(state, overrides);
|
||||
return result;
|
||||
}
|
||||
|
||||
LogSign Operator::cymbal_output(const OperatorState &state, const OperatorState &modulator, const OperatorOverrides *overrides) const {
|
||||
const int output =
|
||||
((state.raw_phase_ >> 16) ^ (state.raw_phase_ >> 14)) &
|
||||
((modulator.raw_phase_ >> 18) ^ (modulator.raw_phase_ >> 13)) &
|
||||
((state.raw_phase_ >> 16) ^ (modulator.raw_phase_ >> 14));
|
||||
|
||||
constexpr int angles[] = {256, 768};
|
||||
LogSign result = negative_log_sin(angles[output & 1]);
|
||||
|
||||
result += state.key_level_scaling_;
|
||||
result += state.adsr_tremolo_attenuation_ + fixed_attenuation(state, overrides);
|
||||
return result;
|
||||
}
|
||||
|
||||
LogSign Operator::high_hat_output(const OperatorState &state, const OperatorState &modulator, const OperatorOverrides *overrides) const {
|
||||
const int output =
|
||||
((state.raw_phase_ >> 16) ^ (state.raw_phase_ >> 14)) &
|
||||
((modulator.raw_phase_ >> 18) ^ (modulator.raw_phase_ >> 13)) &
|
||||
((state.raw_phase_ >> 16) ^ (modulator.raw_phase_ >> 14));
|
||||
|
||||
constexpr int angles[] = {0x234, 0xd0, 0x2d0, 0x34};
|
||||
LogSign result = negative_log_sin(angles[(output & 1) | (state.lfsr_ << 1)]);
|
||||
|
||||
result += state.key_level_scaling_;
|
||||
result += state.adsr_tremolo_attenuation_ + fixed_attenuation(state, overrides);
|
||||
return result;
|
||||
}
|
@ -1,176 +0,0 @@
|
||||
//
|
||||
// Operator.hpp
|
||||
// Clock Signal
|
||||
//
|
||||
// Created by Thomas Harte on 15/04/2020.
|
||||
// Copyright © 2020 Thomas Harte. All rights reserved.
|
||||
//
|
||||
|
||||
#ifndef Operator_hpp
|
||||
#define Operator_hpp
|
||||
|
||||
#include <cstdint>
|
||||
#include "Tables.hpp"
|
||||
#include "LowFrequencyOscillator.hpp"
|
||||
|
||||
namespace Yamaha {
|
||||
namespace OPL {
|
||||
|
||||
/*!
|
||||
Opaquely describes the ephemeral state of an operator.
|
||||
*/
|
||||
struct OperatorState {
|
||||
private:
|
||||
friend class Operator;
|
||||
|
||||
int raw_phase_ = 0;
|
||||
enum class ADSRPhase {
|
||||
Attack, Decay, Sustain, Release
|
||||
} adsr_phase_ = ADSRPhase::Attack;
|
||||
int adsr_attenuation_ = 511;
|
||||
int attack_time_ = 0;
|
||||
|
||||
int key_level_scaling_;
|
||||
int adsr_tremolo_attenuation_;
|
||||
int lfsr_;
|
||||
|
||||
bool last_key_on_ = false;
|
||||
};
|
||||
|
||||
/*!
|
||||
Describes parts of an operator that are genuinely stored per-operator on the OPLL;
|
||||
these can be provided to the Operator in order to have it ignore its local values
|
||||
if the host is an OPLL or VRC7.
|
||||
*/
|
||||
struct OperatorOverrides {
|
||||
int attenuation = 0;
|
||||
bool use_sustain_level = false;
|
||||
};
|
||||
|
||||
/*!
|
||||
Models an operator.
|
||||
|
||||
In Yamaha FM terms, an operator is a combination of a few things:
|
||||
|
||||
* an oscillator, producing one of a handful of sine-derived waveforms;
|
||||
* an ADSR output level envelope; and
|
||||
* a bunch of potential adjustments to those two things:
|
||||
* optional tremolo and/or vibrato (the rates of which are global);
|
||||
* the option to skip 'sustain' in ADSR and go straight to release (since no sustain period is supplied,
|
||||
it otherwise runs for as long as the programmer leaves a channel enabled);
|
||||
* an attenuation for the output level; and
|
||||
* a factor by which to speed up the ADSR envelope as a function of frequency.
|
||||
|
||||
Oscillator period isn't set directly, it's a multiple of the owning channel, in which
|
||||
period is set as a combination of f-num and octave.
|
||||
*/
|
||||
class Operator {
|
||||
public:
|
||||
/// Sets this operator's attack rate as the top nibble of @c value, its decay rate as the bottom nibble.
|
||||
void set_attack_decay(uint8_t value);
|
||||
|
||||
/// Sets this operator's sustain level as the top nibble of @c value, its release rate as the bottom nibble.
|
||||
void set_sustain_release(uint8_t value);
|
||||
|
||||
/// Sets this operator's key scale level as the top two bits of @c value, its total output level as the low six bits.
|
||||
void set_scaling_output(uint8_t value);
|
||||
|
||||
/// Sets this operator's waveform using the low two bits of @c value.
|
||||
void set_waveform(uint8_t value);
|
||||
|
||||
/// From the top nibble of @c value sets the AM, vibrato, hold/sustain level and keyboard sampling rate flags;
|
||||
/// uses the bottom nibble to set the period multiplier.
|
||||
void set_am_vibrato_hold_sustain_ksr_multiple(uint8_t value);
|
||||
|
||||
/// Provides one clock tick to the operator, along with the relevant parameters of its channel.
|
||||
void update(
|
||||
OperatorState &state,
|
||||
const LowFrequencyOscillator &oscillator,
|
||||
bool key_on,
|
||||
int channel_period,
|
||||
int channel_octave,
|
||||
const OperatorOverrides *overrides = nullptr);
|
||||
|
||||
/// @returns @c true if this channel currently has a non-zero output; @c false otherwise.
|
||||
bool is_audible(OperatorState &state, OperatorOverrides *overrides = nullptr);
|
||||
|
||||
/// Provides ordinary melodic output, optionally with modulation.
|
||||
LogSign melodic_output(const OperatorState &state, const LogSign *phase_offset = nullptr, const OperatorOverrides *overrides = nullptr) const;
|
||||
|
||||
/// Provides snare drum output, which is a function of phase and the captured LFSR level.
|
||||
LogSign snare_output(const OperatorState &state, const OperatorOverrides *overrides = nullptr) const;
|
||||
|
||||
/// Provides cymbal output, which is a function of the phase given by @c state, ordinarily the carrier of channel 8,
|
||||
/// and the phase of @c modulator, which is ordinarily the modulator of channel 7.
|
||||
LogSign cymbal_output(const OperatorState &state, const OperatorState &modulator, const OperatorOverrides *overrides = nullptr) const;
|
||||
|
||||
/// Provides high-hat output, which is a function of the phase given by @c state, ordinarily the carrier of channel 8,
|
||||
/// and the phase of @c modulator, which is ordinarily the modulator of channel 7.
|
||||
LogSign high_hat_output(const OperatorState &state, const OperatorState &modulator, const OperatorOverrides *overrides = nullptr) const;
|
||||
|
||||
private:
|
||||
/// If true then an amplitude modulation of "3.7Hz" is applied,
|
||||
/// with a depth "determined by the AM-DEPTH of the BD register"?
|
||||
bool apply_amplitude_modulation_ = false;
|
||||
|
||||
/// If true then a vibrato of '6.4 Hz' is applied, with a depth
|
||||
/// "determined by VOB_DEPTH of the BD register"?
|
||||
bool apply_vibrato_ = false;
|
||||
|
||||
/// Selects between an ADSR envelope that holds at the sustain level
|
||||
/// for as long as this key is on, releasing afterwards, and one that
|
||||
/// simply switches straight to the release rate once the sustain
|
||||
/// level is hit, getting back to 0 regardless of an ongoing key-on.
|
||||
bool use_sustain_level_ = false;
|
||||
|
||||
/// Indexes a lookup table to determine what multiple of the channel's frequency
|
||||
/// this operator is advancing at.
|
||||
int frequency_multiple_ = 0;
|
||||
|
||||
/// Sets the current output level of this modulator, as an attenuation.
|
||||
int attenuation_ = 0;
|
||||
|
||||
/// Provides a potential faster step through the ADSR envelope. Cf. p12.
|
||||
int key_rate_scaling_shift_ = 0;
|
||||
|
||||
/// Selects attenuation that is applied as a function of interval. Cf. p14.
|
||||
int key_level_scaling_ = 0;
|
||||
|
||||
/// Sets the ADSR rates. These all provide the top four bits of a six-bit number;
|
||||
/// the bottom two bits... are 'RL'?
|
||||
int attack_rate_ = 0;
|
||||
int decay_rate_ = 0;
|
||||
int sustain_level_ = 0;
|
||||
int release_rate_ = 0;
|
||||
|
||||
/// Selects the generated waveform.
|
||||
enum class Waveform {
|
||||
Sine, HalfSine, AbsSine, PulseSine
|
||||
} waveform_ = Waveform::Sine;
|
||||
|
||||
/// Updates the ADSR envelope.
|
||||
void update_adsr(
|
||||
OperatorState &state,
|
||||
const LowFrequencyOscillator &oscillator,
|
||||
bool key_on,
|
||||
int channel_period,
|
||||
int channel_octave,
|
||||
const OperatorOverrides *overrides);
|
||||
|
||||
/// Updates the phase generator.
|
||||
void update_phase(OperatorState &state, const LowFrequencyOscillator &oscillator, int channel_period, int channel_octave);
|
||||
|
||||
/// Adds key-level scaling to the current output state.
|
||||
int key_level_scaling(const OperatorState &state, int channel_period, int channel_octave) const;
|
||||
|
||||
/// Adds ADSR and general channel attenuations to the output state.
|
||||
int adsr_tremolo_attenuation(const OperatorState &state, const LowFrequencyOscillator &oscillator) const;
|
||||
|
||||
int fixed_attenuation(const OperatorState &state, const OperatorOverrides *overrides) const;
|
||||
|
||||
};
|
||||
|
||||
}
|
||||
}
|
||||
|
||||
#endif /* Operator_hpp */
|
@ -1,398 +0,0 @@
|
||||
//
|
||||
// OPL2.cpp
|
||||
// Clock Signal
|
||||
//
|
||||
// Created by Thomas Harte on 02/04/2020.
|
||||
// Copyright © 2020 Thomas Harte. all rights reserved.
|
||||
//
|
||||
|
||||
#include "OPL2.hpp"
|
||||
|
||||
#include <cassert>
|
||||
#include <cmath>
|
||||
|
||||
#include "Implementation/PhaseGenerator.hpp"
|
||||
#include "Implementation/EnvelopeGenerator.hpp"
|
||||
#include "Implementation/KeyLevelScaler.hpp"
|
||||
#include "Implementation/WaveformGenerator.hpp"
|
||||
|
||||
using namespace Yamaha::OPL;
|
||||
|
||||
/*
|
||||
|
||||
template <typename Child>
|
||||
OPLBase<Child>::OPLBase(Concurrency::DeferringAsyncTaskQueue &task_queue) : task_queue_(task_queue) {}
|
||||
|
||||
template <typename Child>
|
||||
void OPLBase<Child>::write(uint16_t address, uint8_t value) {
|
||||
if(address & 1) {
|
||||
static_cast<Child *>(this)->write_register(selected_register_, value);
|
||||
} else {
|
||||
selected_register_ = value;
|
||||
}
|
||||
}
|
||||
|
||||
template class Yamaha::OPL::OPLBase<Yamaha::OPL::OPLL>;
|
||||
template class Yamaha::OPL::OPLBase<Yamaha::OPL::OPL2>;
|
||||
|
||||
|
||||
OPLL::OPLL(Concurrency::DeferringAsyncTaskQueue &task_queue, int audio_divider, bool is_vrc7): OPLBase(task_queue), audio_divider_(audio_divider) {
|
||||
// Due to the way that sound mixing works on the OPLL, the audio divider may not
|
||||
// be larger than 4.
|
||||
assert(audio_divider <= 4);
|
||||
|
||||
// Install fixed instruments.
|
||||
const uint8_t *patch_set = is_vrc7 ? vrc7_patch_set : opll_patch_set;
|
||||
for(int c = 0; c < 15; ++c) {
|
||||
setup_fixed_instrument(c+1, patch_set);
|
||||
patch_set += 8;
|
||||
}
|
||||
|
||||
// Install rhythm patches.
|
||||
for(int c = 0; c < 3; ++c) {
|
||||
setup_fixed_instrument(c+16, &percussion_patch_set[c * 8]);
|
||||
}
|
||||
|
||||
// Set default modulators.
|
||||
for(int c = 0; c < 9; ++c) {
|
||||
channels_[c].modulator = &operators_[0];
|
||||
}
|
||||
}
|
||||
|
||||
bool OPLL::is_zero_level() {
|
||||
// for(int c = 0; c < 9; ++c) {
|
||||
// if(channels_[c].is_audible()) return false;
|
||||
// }
|
||||
return false;
|
||||
}
|
||||
|
||||
void OPLL::get_samples(std::size_t number_of_samples, std::int16_t *target) {
|
||||
// Both the OPLL and the OPL2 divide the input clock by 72 to get the base tick frequency;
|
||||
// unlike the OPL2 the OPLL time-divides the output for 'mixing'.
|
||||
|
||||
const int update_period = 72 / audio_divider_;
|
||||
const int channel_output_period = 4 / audio_divider_;
|
||||
|
||||
while(number_of_samples--) {
|
||||
if(!audio_offset_) update_all_chanels();
|
||||
|
||||
*target = int16_t(output_levels_[audio_offset_ / channel_output_period]);
|
||||
++target;
|
||||
audio_offset_ = (audio_offset_ + 1) % update_period;
|
||||
}
|
||||
|
||||
// // Fill in any leftover from the previous session.
|
||||
// if(audio_offset_) {
|
||||
// while(audio_offset_ < update_period && number_of_samples) {
|
||||
// *target = int16_t(channels_[audio_offset_ / channel_output_period].level);
|
||||
// ++target;
|
||||
// ++audio_offset_;
|
||||
// --number_of_samples;
|
||||
// }
|
||||
// audio_offset_ = 0;
|
||||
// }
|
||||
//
|
||||
// // End now if that provided everything that was asked for.
|
||||
// if(!number_of_samples) return;
|
||||
//
|
||||
// int total_updates = int(number_of_samples) / update_period;
|
||||
// number_of_samples %= size_t(update_period);
|
||||
// audio_offset_ = int(number_of_samples);
|
||||
//
|
||||
// while(total_updates--) {
|
||||
// update_all_chanels();
|
||||
//
|
||||
// for(int c = 0; c < update_period; ++c) {
|
||||
// *target = int16_t(channels_[c / channel_output_period].level);
|
||||
// ++target;
|
||||
// }
|
||||
// }
|
||||
//
|
||||
// // If there are any other spots remaining, fill them.
|
||||
// if(number_of_samples) {
|
||||
// update_all_chanels();
|
||||
//
|
||||
// for(int c = 0; c < int(number_of_samples); ++c) {
|
||||
// *target = int16_t(channels_[c / channel_output_period].level);
|
||||
// ++target;
|
||||
// }
|
||||
// }
|
||||
}
|
||||
|
||||
void OPLL::set_sample_volume_range(std::int16_t range) {
|
||||
total_volume_ = range;
|
||||
}
|
||||
|
||||
uint8_t OPLL::read(uint16_t address) {
|
||||
// I've seen mention of an undocumented two-bit status register. I don't yet know what is in it.
|
||||
return 0xff;
|
||||
}
|
||||
|
||||
void OPLL::write_register(uint8_t address, uint8_t value) {
|
||||
// The OPLL doesn't have timers or other non-audio functions, so all writes
|
||||
// go to the audio queue.
|
||||
task_queue_.defer([this, address, value] {
|
||||
// The first 8 locations are used to define the custom instrument, and have
|
||||
// exactly the same format as the patch set arrays at the head of this file.
|
||||
if(address < 8) {
|
||||
custom_instrument_[address] = value;
|
||||
|
||||
// Update whatever that did to the instrument.
|
||||
setup_fixed_instrument(0, custom_instrument_);
|
||||
return;
|
||||
}
|
||||
|
||||
// Register 0xe is a cut-down version of the OPLL's register 0xbd.
|
||||
if(address == 0xe) {
|
||||
depth_rhythm_control_ = value & 0x3f;
|
||||
// if(depth_rhythm_control_ & 0x08)
|
||||
// printf("%02x\n", depth_rhythm_control_);
|
||||
return;
|
||||
}
|
||||
|
||||
const auto index = address & 0xf;
|
||||
if(index > 8) return;
|
||||
|
||||
switch(address & 0xf0) {
|
||||
case 0x30:
|
||||
// Select an instrument in the top nibble, set a channel volume in the lower.
|
||||
channels_[index].overrides.attenuation = value & 0xf;
|
||||
channels_[index].modulator = &operators_[(value >> 4) * 2];
|
||||
|
||||
// Also crib volume levels for rhythm mode, possibly.
|
||||
if(index >= 6) {
|
||||
rhythm_overrides_[(index - 6) * 2 + 0].attenuation = value >> 4;
|
||||
rhythm_overrides_[(index - 6) * 2 + 1].attenuation = value & 0xf;
|
||||
}
|
||||
break;
|
||||
|
||||
case 0x10: channels_[index].set_frequency_low(value); break;
|
||||
|
||||
case 0x20:
|
||||
// Set sustain on/off, key on/off, octave and a single extra bit of frequency.
|
||||
// So they're a lot like OPLL registers 0xb0 to 0xb8, but not identical.
|
||||
channels_[index].set_9bit_frequency_octave_key_on(value);
|
||||
channels_[index].overrides.use_sustain_level = value & 0x20;
|
||||
break;
|
||||
|
||||
default: printf("Unknown write to %02x?!?\n", address); break;
|
||||
}
|
||||
});
|
||||
}
|
||||
|
||||
void OPLL::setup_fixed_instrument(int number, const uint8_t *data) {
|
||||
auto modulator = &operators_[number * 2];
|
||||
auto carrier = &operators_[number * 2 + 1];
|
||||
|
||||
modulator->set_am_vibrato_hold_sustain_ksr_multiple(data[0]);
|
||||
carrier->set_am_vibrato_hold_sustain_ksr_multiple(data[1]);
|
||||
modulator->set_scaling_output(data[2]);
|
||||
|
||||
// Set waveforms — only sine and halfsine are available.
|
||||
modulator->set_waveform((data[3] >> 3) & 1);
|
||||
carrier->set_waveform((data[3] >> 4) & 1);
|
||||
|
||||
// TODO: data[3] b0-b2: modulator feedback level
|
||||
// TODO: data[3] b6, b7: carrier key-scale level
|
||||
|
||||
// Set ADSR parameters.
|
||||
modulator->set_attack_decay(data[4]);
|
||||
carrier->set_attack_decay(data[5]);
|
||||
modulator->set_sustain_release(data[6]);
|
||||
carrier->set_sustain_release(data[7]);
|
||||
}
|
||||
|
||||
void OPLL::update_all_chanels() {
|
||||
// Update the LFO and then the channels.
|
||||
oscillator_.update();
|
||||
for(int c = 0; c < 6; ++c) {
|
||||
channels_[c].update(oscillator_);
|
||||
oscillator_.update_lfsr(); // TODO: should update per slot, not per channel? Or even per cycle?
|
||||
}
|
||||
|
||||
output_levels_[8] = output_levels_[12] = 0;
|
||||
|
||||
#define VOLUME(x) ((x) * total_volume_) >> 12
|
||||
|
||||
// Channels that are updated for melodic output regardless;
|
||||
// in rhythm mode the final three channels — 6, 7, and 8 —
|
||||
// are lost as their operators are used for drum noises.
|
||||
output_levels_[3] = VOLUME(channels_[0].melodic_output());
|
||||
output_levels_[4] = VOLUME(channels_[1].melodic_output());
|
||||
output_levels_[5] = VOLUME(channels_[2].melodic_output());
|
||||
|
||||
output_levels_[9] = VOLUME(channels_[3].melodic_output());
|
||||
output_levels_[10] = VOLUME(channels_[4].melodic_output());
|
||||
output_levels_[11] = VOLUME(channels_[5].melodic_output());
|
||||
|
||||
if(depth_rhythm_control_ & 0x20) {
|
||||
// TODO: pervasively, volume. And LFSR updates.
|
||||
|
||||
channels_[6].update(oscillator_, &operators_[32], depth_rhythm_control_ & 0x10);
|
||||
channels_[7].update(true, oscillator_, operators_[34], bool(depth_rhythm_control_ & 0x01));
|
||||
channels_[7].update(false, oscillator_, operators_[35], bool(depth_rhythm_control_ & 0x08));
|
||||
channels_[8].update(true, oscillator_, operators_[36], bool(depth_rhythm_control_ & 0x04));
|
||||
channels_[8].update(false, oscillator_, operators_[37], bool(depth_rhythm_control_ & 0x02));
|
||||
|
||||
// Update channel 6 as if melodic, but with the bass instrument.
|
||||
output_levels_[2] = output_levels_[15] = VOLUME(channels_[6].melodic_output(&rhythm_overrides_[1]));
|
||||
|
||||
// Use the carrier from channel 7 for the snare.
|
||||
output_levels_[6] = output_levels_[16] = VOLUME(channels_[7].snare_output(operators_[35], &rhythm_overrides_[3]));
|
||||
|
||||
// Use the modulator from channel 8 for the tom tom.
|
||||
output_levels_[1] = output_levels_[14] = VOLUME(channels_[8].tom_tom_output(operators_[37], &rhythm_overrides_[4]));
|
||||
|
||||
// Use the channel 7 modulator and the channel 8 carrier for a cymbal.
|
||||
output_levels_[7] = output_levels_[17] = VOLUME(channels_[7].cymbal_output(operators_[36], operators_[35], channels_[8], &rhythm_overrides_[5]));
|
||||
|
||||
// Use the channel 7 modulator and the channel 8 modulator (?) for a high-hat.
|
||||
output_levels_[0] = output_levels_[13] = VOLUME(channels_[7].high_hat_output(operators_[36], operators_[35], channels_[8], &rhythm_overrides_[2]));
|
||||
} else {
|
||||
// Not in rhythm mode; channels 7, 8 and 9 are melodic.
|
||||
for(int c = 6; c < 9; ++ c) {
|
||||
channels_[c].update(oscillator_);
|
||||
oscillator_.update_lfsr(); // TODO: should update per slot, not per channel? Or even per cycle?
|
||||
}
|
||||
|
||||
output_levels_[0] = output_levels_[1] = output_levels_[2] =
|
||||
output_levels_[6] = output_levels_[7] =
|
||||
output_levels_[13] = output_levels_[14] = 0;
|
||||
|
||||
output_levels_[15] = VOLUME(channels_[6].melodic_output());
|
||||
output_levels_[16] = VOLUME(channels_[7].melodic_output());
|
||||
output_levels_[17] = VOLUME(channels_[8].melodic_output());
|
||||
}
|
||||
|
||||
// Test!
|
||||
// for(int c = 0; c < 18; ++c) {
|
||||
// if(c != 6 && c != 16)
|
||||
// output_levels_[c] = 0;
|
||||
// }
|
||||
|
||||
// channels_[2].level = (channels_[2].update() * total_volume_) >> 14;
|
||||
|
||||
#undef VOLUME
|
||||
}
|
||||
|
||||
|
||||
//template <Personality personality>
|
||||
//void OPL2<personality>::get_samples(std::size_t number_of_samples, std::int16_t *target) {
|
||||
// TODO.
|
||||
// out = exp(logsin(phase2 + exp(logsin(phase1) + gain1)) + gain2)
|
||||
|
||||
// Melodic channels are:
|
||||
//
|
||||
// Channel Operator 1 Operator 2
|
||||
// 0 0 3
|
||||
// 1 1 4
|
||||
// 2 2 5
|
||||
// 3 6 9
|
||||
// 4 7 10
|
||||
// 5 8 11
|
||||
// 6 12 15
|
||||
// 7 13 16
|
||||
// 8 14 17
|
||||
//
|
||||
// In percussion mode, only channels 0–5 are use as melodic, with 6, 7 and 8 being
|
||||
// replaced by:
|
||||
//
|
||||
// Bass drum, using operators 12 and 15;
|
||||
// Snare, using operator 16;
|
||||
// Tom tom, using operator 14,
|
||||
// Cymbal, using operator 17; and
|
||||
// Symbol, using operator 13.
|
||||
//}
|
||||
|
||||
|
||||
|
||||
void OPL2::write_register(uint8_t address, uint8_t value) {
|
||||
|
||||
// Deal with timer changes synchronously.
|
||||
switch(address) {
|
||||
case 0x02: timers_[0] = value; return;
|
||||
case 0x03: timers_[1] = value; return;
|
||||
case 0x04: timer_control_ = value; return;
|
||||
// TODO from register 4:
|
||||
// b7 = IRQ reset;
|
||||
// b6/b5 = timer 1/2 mask (irq enabling flags, I think?)
|
||||
// b4/b3 = timer 2/1 start (seemingly the opposite order to b6/b5?)
|
||||
|
||||
default: break;
|
||||
}
|
||||
|
||||
// Enqueue any changes that affect audio output.
|
||||
task_queue_.enqueue([this, address, value] {
|
||||
//
|
||||
// Modal modifications.
|
||||
//
|
||||
|
||||
switch(address) {
|
||||
case 0x01: waveform_enable_ = value & 0x20; break;
|
||||
case 0x08:
|
||||
// b7: "composite sine wave mode on/off"?
|
||||
csm_keyboard_split_ = value;
|
||||
// b6: "Controls the split point of the keyboard. When 0, the keyboard split is the
|
||||
// second bit from the bit 8 of the F-Number. When 1, the MSB of the F-Number is used."
|
||||
break;
|
||||
case 0xbd: depth_rhythm_control_ = value; break;
|
||||
|
||||
default: break;
|
||||
}
|
||||
|
||||
|
||||
//
|
||||
// Operator modifications.
|
||||
//
|
||||
|
||||
if((address >= 0x20 && address < 0xa0) || address >= 0xe0) {
|
||||
// The 18 operators are spreat out across 22 addresses; each group of
|
||||
// six is framed within an eight-byte area thusly:
|
||||
constexpr int operator_by_address[] = {
|
||||
0, 1, 2, 3, 4, 5, -1, -1,
|
||||
6, 7, 8, 9, 10, 11, -1, -1,
|
||||
12, 13, 14, 15, 16, 17, -1, -1,
|
||||
-1, -1, -1, -1, -1, -1, -1, -1,
|
||||
};
|
||||
|
||||
const auto index = operator_by_address[address & 0x1f];
|
||||
if(index == -1) return;
|
||||
|
||||
switch(address & 0xe0) {
|
||||
case 0x20: operators_[index].set_am_vibrato_hold_sustain_ksr_multiple(value); break;
|
||||
case 0x40: operators_[index].set_scaling_output(value); break;
|
||||
case 0x60: operators_[index].set_attack_decay(value); break;
|
||||
case 0x80: operators_[index].set_sustain_release(value); break;
|
||||
case 0xe0: operators_[index].set_waveform(value); break;
|
||||
|
||||
default: break;
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
//
|
||||
// Channel modifications.
|
||||
//
|
||||
|
||||
const auto index = address & 0xf;
|
||||
if(index > 8) return;
|
||||
|
||||
switch(address & 0xf0) {
|
||||
case 0xa0: channels_[index].set_frequency_low(value); break;
|
||||
case 0xb0: channels_[index].set_10bit_frequency_octave_key_on(value); break;
|
||||
case 0xc0: channels_[index].set_feedback_mode(value); break;
|
||||
|
||||
default: break;
|
||||
}
|
||||
});
|
||||
}
|
||||
|
||||
uint8_t OPL2::read(uint16_t address) {
|
||||
// TODO. There's a status register where:
|
||||
// b7 = IRQ status (set if interrupt request ongoing)
|
||||
// b6 = timer 1 flag (set if timer 1 expired)
|
||||
// b5 = timer 2 flag
|
||||
return 0xff;
|
||||
}
|
||||
|
||||
*/
|
@ -1,123 +0,0 @@
|
||||
//
|
||||
// OPL2.hpp
|
||||
// Clock Signal
|
||||
//
|
||||
// Created by Thomas Harte on 02/04/2020.
|
||||
// Copyright © 2020 Thomas Harte. All rights reserved.
|
||||
//
|
||||
|
||||
#ifndef OPL2_hpp
|
||||
#define OPL2_hpp
|
||||
|
||||
#include "../../Outputs/Speaker/Implementation/SampleSource.hpp"
|
||||
#include "../../Concurrency/AsyncTaskQueue.hpp"
|
||||
|
||||
#include "Implementation/Channel.hpp"
|
||||
#include "Implementation/Operator.hpp"
|
||||
|
||||
#include <atomic>
|
||||
|
||||
namespace Yamaha {
|
||||
namespace OPL {
|
||||
|
||||
/*
|
||||
struct OPL2: public OPLBase<OPL2> {
|
||||
public:
|
||||
// Creates a new OPL2.
|
||||
OPL2(Concurrency::DeferringAsyncTaskQueue &task_queue);
|
||||
|
||||
/// As per ::SampleSource; provides a broadphase test for silence.
|
||||
bool is_zero_level();
|
||||
|
||||
/// As per ::SampleSource; provides audio output.
|
||||
void get_samples(std::size_t number_of_samples, std::int16_t *target);
|
||||
void set_sample_volume_range(std::int16_t range);
|
||||
|
||||
/// Reads from the OPL.
|
||||
uint8_t read(uint16_t address);
|
||||
|
||||
private:
|
||||
friend OPLBase<OPL2>;
|
||||
|
||||
Operator operators_[18];
|
||||
Channel channels_[9];
|
||||
|
||||
// Synchronous properties, valid only on the emulation thread.
|
||||
uint8_t timers_[2] = {0, 0};
|
||||
uint8_t timer_control_ = 0;
|
||||
|
||||
void write_register(uint8_t address, uint8_t value);
|
||||
};
|
||||
|
||||
struct OPLL: public OPLBase<OPLL> {
|
||||
public:
|
||||
// Creates a new OPLL or VRC7.
|
||||
OPLL(Concurrency::DeferringAsyncTaskQueue &task_queue, int audio_divider = 1, bool is_vrc7 = false);
|
||||
|
||||
/// As per ::SampleSource; provides a broadphase test for silence.
|
||||
bool is_zero_level();
|
||||
|
||||
/// As per ::SampleSource; provides audio output.
|
||||
void get_samples(std::size_t number_of_samples, std::int16_t *target);
|
||||
void set_sample_volume_range(std::int16_t range);
|
||||
|
||||
/// Reads from the OPL.
|
||||
uint8_t read(uint16_t address);
|
||||
|
||||
private:
|
||||
friend OPLBase<OPLL>;
|
||||
|
||||
Operator operators_[38]; // There's an extra level of indirection with the OPLL; these 38
|
||||
// operators are to describe 19 hypothetical channels, being
|
||||
// one user-configurable channel, 15 hard-coded channels, and
|
||||
// three channels configured for rhythm generation.
|
||||
|
||||
|
||||
struct Channel: public ::Yamaha::OPL::Channel {
|
||||
void update(const LowFrequencyOscillator &oscillator) {
|
||||
Yamaha::OPL::Channel::update(true, oscillator, modulator[0]);
|
||||
Yamaha::OPL::Channel::update(false, oscillator, modulator[1], false, &overrides);
|
||||
}
|
||||
|
||||
void update(const LowFrequencyOscillator &oscillator, Operator *mod, bool key_on) {
|
||||
Yamaha::OPL::Channel::update(true, oscillator, mod[0], key_on);
|
||||
Yamaha::OPL::Channel::update(false, oscillator, mod[1], key_on, &overrides);
|
||||
}
|
||||
|
||||
using ::Yamaha::OPL::Channel::update;
|
||||
|
||||
int melodic_output() {
|
||||
return Yamaha::OPL::Channel::melodic_output(modulator[0], modulator[1], &overrides);
|
||||
}
|
||||
|
||||
int melodic_output(const OperatorOverrides *overrides) {
|
||||
return Yamaha::OPL::Channel::melodic_output(modulator[0], modulator[1], overrides);
|
||||
}
|
||||
|
||||
bool is_audible() {
|
||||
return Yamaha::OPL::Channel::is_audible(modulator + 1, &overrides);
|
||||
}
|
||||
|
||||
Operator *modulator; // Implicitly, the carrier is modulator+1.
|
||||
OperatorOverrides overrides;
|
||||
};
|
||||
void update_all_chanels();
|
||||
Channel channels_[9];
|
||||
int output_levels_[18];
|
||||
OperatorOverrides rhythm_overrides_[6];
|
||||
|
||||
void setup_fixed_instrument(int number, const uint8_t *data);
|
||||
uint8_t custom_instrument_[8];
|
||||
|
||||
void write_register(uint8_t address, uint8_t value);
|
||||
|
||||
const int audio_divider_ = 1;
|
||||
int audio_offset_ = 0;
|
||||
|
||||
std::atomic<int> total_volume_;
|
||||
};*/
|
||||
|
||||
}
|
||||
}
|
||||
|
||||
#endif /* OPL2_hpp */
|
@ -781,12 +781,6 @@
|
||||
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||||
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||||
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||||
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||||
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||||
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||||
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||||
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||||
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||||
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||||
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||||
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||||
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||||
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||||
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||||
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||||
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||||
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||||
@ -1684,8 +1674,6 @@
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||||
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@ -3529,10 +3517,6 @@
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||||
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||||
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||||
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||||
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||||
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||||
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||||
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@ -4387,7 +4369,6 @@
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||||
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||||
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||||
4B055AA71FAE85EF0060FFFF /* SegmentParser.cpp in Sources */,
|
||||
4BC0CB392447EC9A00A79DBB /* Channel.cpp in Sources */,
|
||||
4BB0A65E204500A900FB3688 /* StaticAnalyser.cpp in Sources */,
|
||||
4B055AC11FAE98DC0060FFFF /* MachineForTarget.cpp in Sources */,
|
||||
4B65086122F4CFE0009C1100 /* Keyboard.cpp in Sources */,
|
||||
@ -4490,7 +4471,6 @@
|
||||
4B0ACC2923775819008902D0 /* DMAController.cpp in Sources */,
|
||||
4B055A951FAE85BB0060FFFF /* BitReverse.cpp in Sources */,
|
||||
4B055ACE1FAE9B030060FFFF /* Plus3.cpp in Sources */,
|
||||
4BC0CB342447EC7D00A79DBB /* Operator.cpp in Sources */,
|
||||
4B055A8D1FAE85920060FFFF /* AsyncTaskQueue.cpp in Sources */,
|
||||
4BAD13441FF709C700FD114A /* MSX.cpp in Sources */,
|
||||
4B055AC41FAE9AE80060FFFF /* Keyboard.cpp in Sources */,
|
||||
@ -4567,7 +4547,6 @@
|
||||
4B89451E201967B4007DE474 /* Tape.cpp in Sources */,
|
||||
4BAF2B4E2004580C00480230 /* DMK.cpp in Sources */,
|
||||
4BB697CE1D4BA44400248BDF /* CommodoreGCR.cpp in Sources */,
|
||||
4BC0CB322447EC7D00A79DBB /* Operator.cpp in Sources */,
|
||||
4B0ACC3023775819008902D0 /* TIASound.cpp in Sources */,
|
||||
4B7136861F78724F008B8ED9 /* Encoder.cpp in Sources */,
|
||||
4B0E04EA1FC9E5DA00F43484 /* CAS.cpp in Sources */,
|
||||
@ -4726,7 +4705,6 @@
|
||||
4BEBFB4D2002C4BF000708CC /* MSXDSK.cpp in Sources */,
|
||||
4BBFBB6C1EE8401E00C01E7A /* ZX8081.cpp in Sources */,
|
||||
4B83348A1F5DB94B0097E338 /* IRQDelegatePortHandler.cpp in Sources */,
|
||||
4BC0CB372447EC9A00A79DBB /* Channel.cpp in Sources */,
|
||||
4B894524201967B4007DE474 /* Tape.cpp in Sources */,
|
||||
4B7136891F78725F008B8ED9 /* Shifter.cpp in Sources */,
|
||||
4BDB61EB2032806E0048AF91 /* CSAtari2600.mm in Sources */,
|
||||
@ -4795,7 +4773,6 @@
|
||||
4B778F4A23A5F1FB0000D260 /* StaticAnalyser.cpp in Sources */,
|
||||
4BD91D772401C2B8007BDC91 /* PatrikRakTests.swift in Sources */,
|
||||
4B680CE223A5553100451D43 /* 68000ComparativeTests.mm in Sources */,
|
||||
4BC0CB332447EC7D00A79DBB /* Operator.cpp in Sources */,
|
||||
4B778F3723A5F11C0000D260 /* Parser.cpp in Sources */,
|
||||
4B778F4523A5F1CD0000D260 /* SegmentParser.cpp in Sources */,
|
||||
4B90467422C6FADD000E2074 /* 68000BitwiseTests.mm in Sources */,
|
||||
@ -4883,7 +4860,6 @@
|
||||
4B778EEF23A5D6680000D260 /* AsyncTaskQueue.cpp in Sources */,
|
||||
4B778F1223A5EC720000D260 /* CRT.cpp in Sources */,
|
||||
4B778EF423A5DB3A0000D260 /* C1540.cpp in Sources */,
|
||||
4BC0CB382447EC9A00A79DBB /* Channel.cpp in Sources */,
|
||||
4B778F3C23A5F16F0000D260 /* FIRFilter.cpp in Sources */,
|
||||
4B778F5423A5F2600000D260 /* UnformattedTrack.cpp in Sources */,
|
||||
4B778EF823A5EB6E0000D260 /* NIB.cpp in Sources */,
|
||||
|
Loading…
Reference in New Issue
Block a user