Files
Pico-GB/ext/minigb_apu/minigb_apu.c
T
Vincent Mistler 29a04ad832 Output I2S audio with PIO/DMA
Requires a MAX98357A amplifier and a 2W 8ohms speaker
2022-12-02 17:24:39 +00:00

604 lines
13 KiB
C

/**
* minigb_apu is released under the terms listed within the LICENSE file.
*
* minigb_apu emulates the audio processing unit (APU) of the Game Boy. This
* project is based on MiniGBS by Alex Baines: https://github.com/baines/MiniGBS
*/
#include <stdbool.h>
#include <stdint.h>
#include <string.h>
#include "minigb_apu.h"
#define DMG_CLOCK_FREQ_U ((unsigned)DMG_CLOCK_FREQ)
#define AUDIO_NSAMPLES (AUDIO_SAMPLES * 2u)
#define AUDIO_MEM_SIZE (0xFF3F - 0xFF10 + 1)
#define AUDIO_ADDR_COMPENSATION 0xFF10
#define MAX(a, b) ( a > b ? a : b )
#define MIN(a, b) ( a <= b ? a : b )
#define VOL_INIT_MAX (INT16_MAX/8)
#define VOL_INIT_MIN (INT16_MIN/8)
/* Handles time keeping for sound generation.
* FREQ_INC_REF must be equal to, or larger than AUDIO_SAMPLE_RATE in order
* to avoid a division by zero error.
* Using a square of 2 simplifies calculations. */
#define FREQ_INC_REF (AUDIO_SAMPLE_RATE * 16)
#define MAX_CHAN_VOLUME 15
/**
* Memory holding audio registers between 0xFF10 and 0xFF3F inclusive.
*/
static uint8_t audio_mem[AUDIO_MEM_SIZE];
struct chan_len_ctr {
uint8_t load;
unsigned enabled : 1;
uint32_t counter;
uint32_t inc;
};
struct chan_vol_env {
uint8_t step;
unsigned up : 1;
uint32_t counter;
uint32_t inc;
};
struct chan_freq_sweep {
uint16_t freq;
uint8_t rate;
uint8_t shift;
unsigned up : 1;
uint32_t counter;
uint32_t inc;
};
static struct chan {
unsigned enabled : 1;
unsigned powered : 1;
unsigned on_left : 1;
unsigned on_right : 1;
unsigned muted : 1;
uint8_t volume;
uint8_t volume_init;
uint16_t freq;
uint32_t freq_counter;
uint32_t freq_inc;
int_fast16_t val;
struct chan_len_ctr len;
struct chan_vol_env env;
struct chan_freq_sweep sweep;
union {
struct {
uint8_t duty;
uint8_t duty_counter;
} square;
struct {
uint16_t lfsr_reg;
uint8_t lfsr_wide;
uint8_t lfsr_div;
} noise;
struct {
uint8_t sample;
} wave;
};
} chans[4];
static int32_t vol_l, vol_r;
static void set_note_freq(struct chan *c, const uint32_t freq)
{
/* Lowest expected value of freq is 64. */
c->freq_inc = freq * (uint32_t)(FREQ_INC_REF / AUDIO_SAMPLE_RATE);
}
static void chan_enable(const uint_fast8_t i, const bool enable)
{
uint8_t val;
chans[i].enabled = enable;
val = (audio_mem[0xFF26 - AUDIO_ADDR_COMPENSATION] & 0x80) |
(chans[3].enabled << 3) | (chans[2].enabled << 2) |
(chans[1].enabled << 1) | (chans[0].enabled << 0);
audio_mem[0xFF26 - AUDIO_ADDR_COMPENSATION] = val;
//audio_mem[0xFF26 - AUDIO_ADDR_COMPENSATION] |= 0x80 | ((uint8_t)enable) << i;
}
static void update_env(struct chan *c)
{
c->env.counter += c->env.inc;
while (c->env.counter > FREQ_INC_REF) {
if (c->env.step) {
c->volume += c->env.up ? 1 : -1;
if (c->volume == 0 || c->volume == MAX_CHAN_VOLUME) {
c->env.inc = 0;
}
c->volume = MAX(0, MIN(MAX_CHAN_VOLUME, c->volume));
}
c->env.counter -= FREQ_INC_REF;
}
}
static void update_len(struct chan *c)
{
if (!c->len.enabled)
return;
c->len.counter += c->len.inc;
if (c->len.counter > FREQ_INC_REF) {
chan_enable(c - chans, 0);
c->len.counter = 0;
}
}
static bool update_freq(struct chan *c, uint32_t *pos)
{
uint32_t inc = c->freq_inc - *pos;
c->freq_counter += inc;
if (c->freq_counter > FREQ_INC_REF) {
*pos = c->freq_inc - (c->freq_counter - FREQ_INC_REF);
c->freq_counter = 0;
return true;
} else {
*pos = c->freq_inc;
return false;
}
}
static void update_sweep(struct chan *c)
{
c->sweep.counter += c->sweep.inc;
while (c->sweep.counter > FREQ_INC_REF) {
if (c->sweep.shift) {
uint16_t inc = (c->sweep.freq >> c->sweep.shift);
if (!c->sweep.up)
inc *= -1;
c->freq += inc;
if (c->freq > 2047) {
c->enabled = 0;
} else {
set_note_freq(c,
DMG_CLOCK_FREQ_U / ((2048 - c->freq)<< 5));
c->freq_inc *= 8;
}
} else if (c->sweep.rate) {
c->enabled = 0;
}
c->sweep.counter -= FREQ_INC_REF;
}
}
static void update_square(int16_t* samples, const bool ch2)
{
uint32_t freq;
struct chan* c = chans + ch2;
if (!c->powered || !c->enabled)
return;
freq = DMG_CLOCK_FREQ_U / ((2048 - c->freq) << 5);
set_note_freq(c, freq);
c->freq_inc *= 8;
for (uint_fast16_t i = 0; i < AUDIO_NSAMPLES; i += 2) {
update_len(c);
if (!c->enabled)
continue;
update_env(c);
if (!ch2)
update_sweep(c);
uint32_t pos = 0;
uint32_t prev_pos = 0;
int32_t sample = 0;
while (update_freq(c, &pos)) {
c->square.duty_counter = (c->square.duty_counter + 1) & 7;
sample += ((pos - prev_pos) / c->freq_inc) * c->val;
c->val = (c->square.duty & (1 << c->square.duty_counter)) ?
VOL_INIT_MAX / MAX_CHAN_VOLUME :
VOL_INIT_MIN / MAX_CHAN_VOLUME;
prev_pos = pos;
}
if (c->muted)
continue;
sample += c->val;
sample *= c->volume;
sample /= 4;
samples[i + 0] += sample * c->on_left * vol_l;
samples[i + 1] += sample * c->on_right * vol_r;
}
}
static uint8_t wave_sample(const unsigned int pos, const unsigned int volume)
{
uint8_t sample;
sample = audio_mem[(0xFF30 + pos / 2) - AUDIO_ADDR_COMPENSATION];
if (pos & 1) {
sample &= 0xF;
} else {
sample >>= 4;
}
return volume ? (sample >> (volume - 1)) : 0;
}
static void update_wave(int16_t *samples)
{
uint32_t freq;
struct chan *c = chans + 2;
if (!c->powered || !c->enabled)
return;
freq = (DMG_CLOCK_FREQ_U / 64) / (2048 - c->freq);
set_note_freq(c, freq);
c->freq_inc *= 32;
for (uint_fast16_t i = 0; i < AUDIO_NSAMPLES; i += 2) {
update_len(c);
if (!c->enabled)
continue;
uint32_t pos = 0;
uint32_t prev_pos = 0;
int32_t sample = 0;
c->wave.sample = wave_sample(c->val, c->volume);
while (update_freq(c, &pos)) {
c->val = (c->val + 1) & 31;
sample += ((pos - prev_pos) / c->freq_inc) *
((int)c->wave.sample - 8) * (INT16_MAX/64);
c->wave.sample = wave_sample(c->val, c->volume);
prev_pos = pos;
}
sample += ((int)c->wave.sample - 8) * (int)(INT16_MAX/64);
if (c->volume == 0)
continue;
{
/* First element is unused. */
int16_t div[] = { INT16_MAX, 1, 2, 4 };
sample = sample / (div[c->volume]);
}
if (c->muted)
continue;
sample /= 4;
samples[i + 0] += sample * c->on_left * vol_l;
samples[i + 1] += sample * c->on_right * vol_r;
}
}
static void update_noise(int16_t *samples)
{
struct chan *c = chans + 3;
if (!c->powered)
return;
{
const uint32_t lfsr_div_lut[] = {
8, 16, 32, 48, 64, 80, 96, 112
};
uint32_t freq;
freq = DMG_CLOCK_FREQ_U / (lfsr_div_lut[c->noise.lfsr_div] << c->freq);
set_note_freq(c, freq);
}
if (c->freq >= 14)
c->enabled = 0;
for (uint_fast16_t i = 0; i < AUDIO_NSAMPLES; i += 2) {
update_len(c);
if (!c->enabled)
continue;
update_env(c);
uint32_t pos = 0;
uint32_t prev_pos = 0;
int32_t sample = 0;
while (update_freq(c, &pos)) {
c->noise.lfsr_reg = (c->noise.lfsr_reg << 1) |
(c->val >= VOL_INIT_MAX/MAX_CHAN_VOLUME);
if (c->noise.lfsr_wide) {
c->val = !(((c->noise.lfsr_reg >> 14) & 1) ^
((c->noise.lfsr_reg >> 13) & 1)) ?
VOL_INIT_MAX / MAX_CHAN_VOLUME :
VOL_INIT_MIN / MAX_CHAN_VOLUME;
} else {
c->val = !(((c->noise.lfsr_reg >> 6) & 1) ^
((c->noise.lfsr_reg >> 5) & 1)) ?
VOL_INIT_MAX / MAX_CHAN_VOLUME :
VOL_INIT_MIN / MAX_CHAN_VOLUME;
}
sample += ((pos - prev_pos) / c->freq_inc) * c->val;
prev_pos = pos;
}
if (c->muted)
continue;
sample += c->val;
sample *= c->volume;
sample /= 4;
samples[i + 0] += sample * c->on_left * vol_l;
samples[i + 1] += sample * c->on_right * vol_r;
}
}
/**
* SDL2 style audio callback function.
*/
void audio_callback(void *userdata, int16_t *stream, size_t len)
{
/* Appease unused variable warning. */
(void)userdata;
memset(stream, 0, len);
update_square(stream, 0);
update_square(stream, 1);
update_wave(stream);
update_noise(stream);
}
static void chan_trigger(uint_fast8_t i)
{
struct chan *c = chans + i;
chan_enable(i, 1);
c->volume = c->volume_init;
// volume envelope
{
uint8_t val =
audio_mem[(0xFF12 + (i * 5)) - AUDIO_ADDR_COMPENSATION];
c->env.step = val & 0x07;
c->env.up = val & 0x08 ? 1 : 0;
c->env.inc = c->env.step ?
(FREQ_INC_REF * 64ul) / ((uint32_t)c->env.step * AUDIO_SAMPLE_RATE) :
(8ul * FREQ_INC_REF) / AUDIO_SAMPLE_RATE ;
c->env.counter = 0;
}
// freq sweep
if (i == 0) {
uint8_t val = audio_mem[0xFF10 - AUDIO_ADDR_COMPENSATION];
c->sweep.freq = c->freq;
c->sweep.rate = (val >> 4) & 0x07;
c->sweep.up = !(val & 0x08);
c->sweep.shift = (val & 0x07);
c->sweep.inc = c->sweep.rate ?
((128 * FREQ_INC_REF) / (c->sweep.rate * AUDIO_SAMPLE_RATE)) : 0;
c->sweep.counter = FREQ_INC_REF;
}
int len_max = 64;
if (i == 2) { // wave
len_max = 256;
c->val = 0;
} else if (i == 3) { // noise
c->noise.lfsr_reg = 0xFFFF;
c->val = VOL_INIT_MIN / MAX_CHAN_VOLUME;
}
c->len.inc = (256 * FREQ_INC_REF) / (AUDIO_SAMPLE_RATE * (len_max - c->len.load));
c->len.counter = 0;
}
/**
* Read audio register.
* \param addr Address of audio register. Must be 0xFF10 <= addr <= 0xFF3F.
* This is not checked in this function.
* \return Byte at address.
*/
uint8_t audio_read(const uint16_t addr)
{
static const uint8_t ortab[] = {
0x80, 0x3f, 0x00, 0xff, 0xbf,
0xff, 0x3f, 0x00, 0xff, 0xbf,
0x7f, 0xff, 0x9f, 0xff, 0xbf,
0xff, 0xff, 0x00, 0x00, 0xbf,
0x00, 0x00, 0x70,
0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
};
return audio_mem[addr - AUDIO_ADDR_COMPENSATION] |
ortab[addr - AUDIO_ADDR_COMPENSATION];
}
/**
* Write audio register.
* \param addr Address of audio register. Must be 0xFF10 <= addr <= 0xFF3F.
* This is not checked in this function.
* \param val Byte to write at address.
*/
void audio_write(const uint16_t addr, const uint8_t val)
{
/* Find sound channel corresponding to register address. */
uint_fast8_t i;
if(addr == 0xFF26)
{
audio_mem[addr - AUDIO_ADDR_COMPENSATION] = val & 0x80;
/* On APU power off, clear all registers apart from wave
* RAM. */
if((val & 0x80) == 0)
{
memset(audio_mem, 0x00, 0xFF26 - AUDIO_ADDR_COMPENSATION);
chans[0].enabled = false;
chans[1].enabled = false;
chans[2].enabled = false;
chans[3].enabled = false;
}
return;
}
/* Ignore register writes if APU powered off. */
if(audio_mem[0xFF26 - AUDIO_ADDR_COMPENSATION] == 0x00)
return;
audio_mem[addr - AUDIO_ADDR_COMPENSATION] = val;
i = (addr - AUDIO_ADDR_COMPENSATION) / 5;
switch (addr) {
case 0xFF12:
case 0xFF17:
case 0xFF21: {
chans[i].volume_init = val >> 4;
chans[i].powered = (val >> 3) != 0;
// "zombie mode" stuff, needed for Prehistorik Man and probably
// others
if (chans[i].powered && chans[i].enabled) {
if ((chans[i].env.step == 0 && chans[i].env.inc != 0)) {
if (val & 0x08) {
chans[i].volume++;
} else {
chans[i].volume += 2;
}
} else {
chans[i].volume = 16 - chans[i].volume;
}
chans[i].volume &= 0x0F;
chans[i].env.step = val & 0x07;
}
} break;
case 0xFF1C:
chans[i].volume = chans[i].volume_init = (val >> 5) & 0x03;
break;
case 0xFF11:
case 0xFF16:
case 0xFF20: {
const uint8_t duty_lookup[] = { 0x10, 0x30, 0x3C, 0xCF };
chans[i].len.load = val & 0x3f;
chans[i].square.duty = duty_lookup[val >> 6];
break;
}
case 0xFF1B:
chans[i].len.load = val;
break;
case 0xFF13:
case 0xFF18:
case 0xFF1D:
chans[i].freq &= 0xFF00;
chans[i].freq |= val;
break;
case 0xFF1A:
chans[i].powered = (val & 0x80) != 0;
chan_enable(i, val & 0x80);
break;
case 0xFF14:
case 0xFF19:
case 0xFF1E:
chans[i].freq &= 0x00FF;
chans[i].freq |= ((val & 0x07) << 8);
/* Intentional fall-through. */
case 0xFF23:
chans[i].len.enabled = val & 0x40 ? 1 : 0;
if (val & 0x80)
chan_trigger(i);
break;
case 0xFF22:
chans[3].freq = val >> 4;
chans[3].noise.lfsr_wide = !(val & 0x08);
chans[3].noise.lfsr_div = val & 0x07;
break;
case 0xFF24:
{
vol_l = ((val >> 4) & 0x07);
vol_r = (val & 0x07);
break;
}
case 0xFF25:
for (uint_fast8_t j = 0; j < 4; j++) {
chans[j].on_left = (val >> (4 + j)) & 1;
chans[j].on_right = (val >> j) & 1;
}
break;
}
}
void audio_init(void)
{
/* Initialise channels and samples. */
memset(chans, 0, sizeof(chans));
chans[0].val = chans[1].val = -1;
/* Initialise IO registers. */
{
const uint8_t regs_init[] = { 0x80, 0xBF, 0xF3, 0xFF, 0x3F,
0xFF, 0x3F, 0x00, 0xFF, 0x3F,
0x7F, 0xFF, 0x9F, 0xFF, 0x3F,
0xFF, 0xFF, 0x00, 0x00, 0x3F,
0x77, 0xF3, 0xF1 };
for(uint_fast8_t i = 0; i < sizeof(regs_init); ++i)
audio_write(0xFF10 + i, regs_init[i]);
}
/* Initialise Wave Pattern RAM. */
{
const uint8_t wave_init[] = { 0xac, 0xdd, 0xda, 0x48,
0x36, 0x02, 0xcf, 0x16,
0x2c, 0x04, 0xe5, 0x2c,
0xac, 0xdd, 0xda, 0x48 };
for(uint_fast8_t i = 0; i < sizeof(wave_init); ++i)
audio_write(0xFF30 + i, wave_init[i]);
}
}