kopia lustrzana https://github.com/pimoroni/pimoroni-pico
133 wiersze
5.2 KiB
C++
133 wiersze
5.2 KiB
C++
#pragma once
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#include <cstdint>
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namespace synth {
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// The duration a note is played is determined by the amount of attack,
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// decay, and release, combined with the length of the note as defined by
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// the user.
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//
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// - Attack: number of milliseconds it takes for a note to hit full volume
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// - Decay: number of milliseconds it takes for a note to settle to sustain volume
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// - Sustain: percentage of full volume that the note sustains at (duration implied by other factors)
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// - Release: number of milliseconds it takes for a note to reduce to zero volume after it has ended
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//
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// Attack (750ms) - Decay (500ms) -------- Sustain ----- Release (250ms)
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//
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// + + + +
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// | | | |
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// | | | |
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// | | | |
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// v v v v
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// 0ms 1000ms 2000ms 3000ms 4000ms
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//
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// | XXXX | | | |
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// | X X|XX | | |
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// | X | XXX | | |
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// | X | XXXXXXXXXXXXXX|XXXXXXXXXXXXXXXXXXX| |
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// | X | | |X |
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// | X | | |X |
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// | X | | | X |
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// | X | | | X |
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// | X | | | X |
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// | X | | | X |
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// | X | | | X |
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// | X | | | X |
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// | X + + + | + + + | + + + | + + + | +
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// | X | | | | | | | | | | | | | | | | |
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// |X | | | | | | | | | | | | | | | | |
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// +----+----+----+----+----+----+----+----+----+----+----+----+----+----+----+----+----+--->
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#define CHANNEL_COUNT 8
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constexpr float pi = 3.14159265358979323846f;
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const uint32_t sample_rate = 44100;
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extern uint16_t volume;
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enum Waveform {
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NOISE = 128,
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SQUARE = 64,
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SAW = 32,
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TRIANGLE = 16,
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SINE = 8,
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WAVE = 1
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};
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enum class ADSRPhase : uint8_t {
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ATTACK,
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DECAY,
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SUSTAIN,
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RELEASE,
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OFF
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};
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struct AudioChannel {
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uint8_t waveforms = 0; // bitmask for enabled waveforms (see AudioWaveform enum for values)
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uint16_t frequency = 660; // frequency of the voice (Hz)
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uint16_t volume = 0xffff; // channel volume (default 50%)
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uint16_t attack_ms = 2; // attack period
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uint16_t decay_ms = 6; // decay period
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uint16_t sustain = 0xffff; // sustain volume
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uint16_t release_ms = 1; // release period
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uint16_t pulse_width = 0x7fff; // duty cycle of square wave (default 50%)
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int16_t noise = 0; // current noise value
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uint32_t waveform_offset = 0; // voice offset (Q8)
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int32_t filter_last_sample = 0;
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bool filter_enable = false;
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uint16_t filter_cutoff_frequency = 0;
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uint32_t adsr_frame = 0; // number of frames into the current ADSR phase
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uint32_t adsr_end_frame = 0; // frame target at which the ADSR changes to the next phase
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uint32_t adsr = 0;
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int32_t adsr_step = 0;
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ADSRPhase adsr_phase = ADSRPhase::OFF;
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uint8_t wave_buf_pos = 0; //
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int16_t wave_buffer[64]; // buffer for arbitrary waveforms. small as it's filled by user callback
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void *user_data = nullptr;
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void (*wave_buffer_callback)(AudioChannel &channel);
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void trigger_attack() {
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adsr_frame = 0;
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adsr_phase = ADSRPhase::ATTACK;
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adsr_end_frame = (attack_ms * sample_rate) / 1000;
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adsr_step = (int32_t(0xffffff) - int32_t(adsr)) / int32_t(adsr_end_frame);
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}
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void trigger_decay() {
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adsr_frame = 0;
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adsr_phase = ADSRPhase::DECAY;
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adsr_end_frame = (decay_ms * sample_rate) / 1000;
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adsr_step = (int32_t(sustain << 8) - int32_t(adsr)) / int32_t(adsr_end_frame);
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}
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void trigger_sustain() {
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adsr_frame = 0;
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adsr_phase = ADSRPhase::SUSTAIN;
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adsr_end_frame = 0;
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adsr_step = 0;
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}
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void trigger_release() {
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adsr_frame = 0;
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adsr_phase = ADSRPhase::RELEASE;
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adsr_end_frame = (release_ms * sample_rate) / 1000;
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adsr_step = (int32_t(0) - int32_t(adsr)) / int32_t(adsr_end_frame);
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}
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void off() {
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adsr_frame = 0;
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adsr_phase = ADSRPhase::OFF;
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adsr_step = 0;
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}
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};
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extern AudioChannel channels[CHANNEL_COUNT];
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int16_t get_audio_frame();
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bool is_audio_playing();
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}
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