pimoroni-pico/examples/pico_audio/synth.hpp

133 lines
5.2 KiB
C++

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