#include #include "synth.h" #include "freq_table.h" #include #include // sample rate is 8M / (3 * 64) enum { channel_count = 3, //tick_length = 400, // tick_length = 256, row_length = 4, pattern_length = 16 }; static const synth_instrument_t instruments[] = { { 1<<15, 100, 12, 0 }, { 0, 100, 12, 0 }, { 0, 200, 10, 0 }, { 1<<13, 0, 0, 2 }, { 1<<13, 0, 5, 2 }, }; static const uint8_t wave_table[][2] = { { 0, WAVE_PULSE }, { 3, WAVE_PULSE }, { 7, WAVE_PULSE }, { 12, WAVE_PULSE }, { 256 - 4, 0xff }, { 0, WAVE_PULSE }, { 2, WAVE_PULSE }, { 7, WAVE_PULSE }, { 10, WAVE_PULSE }, { 256 - 4, 0xff }, { 0, WAVE_NOISE }, { 0, WAVE_PULSE }, { 0xff, 0xff }, { 0, WAVE_PULSE }, { 0xff, 0xff }, }; static const uint8_t patterns[][pattern_length][2] PROGMEM = { {}, { { 33 - 12, 0 }, { 0, 0 }, { 0xff, 1 }, { 0, 0 }, { 33, 1 }, { 0xff, 1 }, { 33, 1 }, { 0xff, 1 }, { 33, 1 }, { 0xff, 1 }, { 33 - 12, 1 }, { 0xff, 1 }, { 33 - 12, 1 }, { 0xff, 1 }, { 33, 1 }, { 0xff, 1 }, }, { { 28 - 12, 0 }, { 0, 0 }, { 0xff, 1 }, { 0, 0 }, { 28, 1 }, { 0xff, 1 }, { 28, 1 }, { 0xff, 1 }, { 28, 1 }, { 0xff, 1 }, { 28 - 12, 1 }, { 0xff, 1 }, { 28 - 12, 1 }, { 0xff, 1 }, { 28, 1 }, { 0xff, 1 }, }, { { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 57, 3 }, }, { { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 57, 4 }, }, { { 60, 2 }, }, { { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 57, 2 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 55, 2 }, { 0, 0 }, { 57, 2 }, { 0, 0 }, }, { { 55, 2 }, }, { { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 0, 0 }, { 57, 2 }, }, { { 55-3, 2 }, }, }; static const uint8_t pattern_table[][channel_count] = { { 1, 0, 5 }, { 1, 3, 0 }, { 1, 0, 7 }, { 1, 3, 6 }, { 2, 0, 7 }, { 2, 4, 8 }, { 2, 0, 9 }, { 2, 4, 0 }, }; enum { pattern_table_length = sizeof(pattern_table) / sizeof(pattern_table[0]) }; static synth_channel_t channels[channel_count]; static int8_t sample; static int8_t tick; static int8_t row; static int8_t seq; /* PROTOTYPES */ uint8_t synth_mix(void); static uint8_t timeslots[SYNTH_BUFSIZE]; static uint8_t timeslots_write; // current write head static uint8_t timeslots_read; // current read head /*register for atomic ++ and -- */ register uint8_t timeslots_fill asm("r2"); static void enqueue_timeslot(uint8_t synthval); static uint8_t dequeue_timeslot(void); void synth_init(void) { sample = 0; tick = 0; row = 0; seq = 0; timeslots_fill = 0; } inline uint8_t synth_mix(void) { if(sample == 0) { // new tick for(int i = 1; i < channel_count; i++) { synth_channel_t* chan = &channels[i]; const synth_instrument_t* inst = &instruments[chan->inst_nr]; if(chan->level > inst->decay) chan->level -= inst->decay; else chan->level = 0; chan->pulse_width += inst->pulse_sweep; chan->pos++; if(wave_table[chan->pos][1] == 0xff) chan->pos += wave_table[chan->pos][0]; // enter new rol // w if(tick == 0) { uint8_t pattern_nr = pattern_table[seq][i]; uint8_t note = pgm_read_byte(&patterns[pattern_nr][row][0]); if(note) { // new note, maybe? if(note == 0xff) { chan->level = 0; } else { chan->level = 80; // TODO: less? chan->note = note; chan->inst_nr = pgm_read_byte(&patterns[pattern_nr][row][1]); inst = &instruments[chan->inst_nr]; chan->pos = inst->wave_table_pos; if(inst->pulse_width) chan->pulse_width = inst->pulse_width; } } } } } if(++sample == 0){ if(++tick == row_length) { tick = 0; if(++row == pattern_length) { row = 0; if(++seq == pattern_table_length) { seq = 0; } } } } uint8_t output = 0; for(int i = 0; i < channel_count; i++) { synth_channel_t* chan = &channels[i]; // const synth_instrument_t* inst = &instruments[chan->inst_nr]; chan->phase += pgm_read_word(&freq_table[(uint8_t)(chan->note + wave_table[chan->pos][0])]); uint8_t amp; switch(wave_table[chan->pos][1]) { case WAVE_PULSE: amp = -(chan->phase < chan->pulse_width); break; case WAVE_SAW: amp = (chan->phase >> 8); break; case WAVE_NOISE: // shitty noise chan->phase = (chan->phase >> 1) ^ (-(chan->phase & 1) & 0xb400); amp = (chan->phase >> 8); break; default: amp = 0; break; } output += (((amp & 0xff) * chan->level) >> 8); } return output; } /* fill all the timeslots */ inline void synth_poll(void) { /* refill timeslots queue */ // while (timeslots_fill < (SYNTH_BUFSIZE-1)) if (timeslots_fill < (SYNTH_BUFSIZE-1)) enqueue_timeslot(synth_mix()); } /* timeslot queue handling */ static inline void enqueue_timeslot(uint8_t synthval) { timeslots[timeslots_write & SYNTH_BUFMASK] = synthval; timeslots_fill++; timeslots_write++; } static inline uint8_t dequeue_timeslot() { uint8_t t = timeslots[timeslots_read & SYNTH_BUFMASK]; if(timeslots_fill){ /* buffer not underrun... move forward in readbuffer */ timeslots_fill --; timeslots_read ++; } return t; } ISR(TIMER0_COMPA_vect) { /* calculate next analog sample value in synth mixer:*/ OCR1B = dequeue_timeslot(); }