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#define SAMPLE_FREQUENCY 44100
#define NOTE_PERIOD (SAMPLE_FREQUENCY * 0x4000 / 11025)
#define ADSR_STEP (SAMPLE_FREQUENCY / 0xf)
static u16 pitch_table[] = {
12173, 11490, 10845, 10237, 9662, 9120, 8608, 8125,
7669, 7238, 6832, 6448, 6086, 5745, 5422, 5118,
4831, 4560, 4304, 4062, 3834, 3619, 3416, 3224,
3043, 2872, 2711, 2559, 2415, 2280, 2152, 2031,
1917, 1809, 1708, 1612, 1521, 1436, 1355, 1279,
1207, 1140, 1076, 1015, 958, 904, 854, 806,
760, 718, 677, 639, 603, 570, 538, 507,
479, 452, 427, 403, 380, 359, 338, 319,
301, 285, 269, 253, 239, 226, 213, 201,
190, 179, 169, 159, 150, 142, 134, 126,
119, 113, 106, 100, 95, 89, 84, 79,
75, 71, 67, 63, 59, 56, 53, 50,
47, 44, 42, 39, 37, 35, 33, 31,
29, 28, 26, 25, 23, 22, 21, 19,
18, 17, 16, 15, 14, 14, 13, 12,
2,
};
s8 square_wave[] = {
(s8)0x00 + (s8)0x80, (s8)0xFF + (s8)0x80
};
#include "text.h"
//
// REG_TM0D frequency buffer size
// | | |
// V V V
//
// Timer = 62610 = 65536 - (16777216 / 5734), buf = 96
// Timer = 63940 = 65536 - (16777216 / 10512), buf = 176
// Timer = 64282 = 65536 - (16777216 / 13379), buf = 224
// Timer = 64612 = 65536 - (16777216 / 18157), buf = 304
// Timer = 64738 = 65536 - (16777216 / 21024), buf = 352
// Timer = 64909 = 65536 - (16777216 / 26758), buf = 448
// Timer = 65004 = 65536 - (16777216 / 31536), buf = 528
// Timer = 65073 = 65536 - (16777216 / 36314), buf = 608
// Timer = 65118 = 65536 - (16777216 / 40137), buf = 672
// Timer = 65137 = 65536 - (16777216 / 42048), buf = 704
//
// Source: https://deku.gbadev.org/program/sound1.html
#define AUDIO_FREQ 18157
#define AUDIO_BUF_LEN 304
#define AUDIO_TIMER 64612
typedef struct Audio {
s8 mix_buffer[AUDIO_BUF_LEN * 2];
s8 *current_buffer;
u8 active_buffer;
} Audio;
typedef struct AudioChannel {
// Pointer to the raw data in the ROM.
s8 *data;
// Current position in the data (20.12 fixed-point).
u32 pos;
// Increment (20.12 fixed-point).
u32 inc;
// Volume (0-64, 1.6 fixed-point).
u32 vol;
// Sound length (20.12 fixed-point).
u32 length;
// Length of looped portion (20.12 fixed-point, 0 to disable looping).
u32 loop_length;
// TODO: this should be different?
u8 pitch;
} AudioChannel;
static Audio audio;
#define POLYPHONY 4
static AudioChannel channels[POLYPHONY];
void
init_sound(void) {
// Initialize audio buffers/channels.
audio = (Audio){0};
for (size_t i = 0; i < POLYPHONY; ++i) {
channels[i] = (AudioChannel){0};
}
// DEBUG: testing channel 0 with square wave
// channels[0].data = samples;
// channels[0].length = (LEN(samples) - 1) << 12;
// channels[0].pos = 0;
// channels[0].inc = (44100 << 12) / AUDIO_FREQ;
// channels[0].vol = 32;
// channels[0].loop_length = channels[0].length;
// channels[0].loop_length = 0;
// Enable the sound chip.
SOUND_STATUS = SOUND_ENABLE;
// Set max volume, left-right sound, fifo reset and use timer 0 for
// DirectSound A.
SOUND_DSOUND_MASTER = SOUND_DSOUND_RATIO_A
| SOUND_DSOUND_LEFT_A
| SOUND_DSOUND_RIGHT_A
| SOUND_DSOUND_RESET_A;
// TODO: No pitch selection yet.
TIMER_DATA_0 = AUDIO_TIMER;
TIMER_CTRL_0 = TIMER_CTRL_ENABLE;
}
void sound_vsync() {
// buffer 1 just got over
if(audio.active_buffer == 1) {
// Start playing buffer 0
dma_transfer_copy(SOUND_FIFO_A, audio.mix_buffer, 1, 1,
DMA_DST_FIXED | DMA_CHUNK_32 | DMA_REFRESH | DMA_REPEAT | DMA_ENABLE);
// Set the current buffer pointer to the start of buffer 1
audio.current_buffer = audio.mix_buffer + AUDIO_BUF_LEN;
audio.active_buffer = 0;
} else {
// buffer 0 just got over
// DMA points to buffer 1 already, so don't bother stopping and resetting it
// Set the current buffer pointer to the start of buffer 0
audio.current_buffer = audio.mix_buffer;
audio.active_buffer = 1;
}
}
void sound_mix() {
// Initialize and clear mix_buffer.
s16 mix_buffer[AUDIO_BUF_LEN];
u32 fill = 0;
dma_fill(mix_buffer, fill, sizeof(mix_buffer), 3);
// Mix channels into the temporary buffer.
for (size_t j = 0; j < POLYPHONY; ++j) {
AudioChannel *ch = &channels[j];
// Check if channel is active.
if (ch->data == NULL || ch->pitch >= 108) {
continue;
}
if (ch->pos + ch->inc * AUDIO_BUF_LEN >= ch->length) {
// Sample is going to finish, need to consider this for looping or
// stopping.
for(size_t i = 0; i < AUDIO_BUF_LEN; i++) {
// Remember we are using fixed point values.
mix_buffer[i] += (0x80 + (u8)ch->data[ch->pos >> 12]) * ch->vol;
ch->pos += ch->inc;
if (ch->pos >= ch->length) {
// If looping is not active disable the channel.
if (ch->loop_length == 0) {
ch->data = NULL;
break;
}
// Loop the sample.
while (ch->pos >= ch->length) {
ch->pos -= ch->loop_length;
}
}
}
} else {
// Sample still have room to go, no need to check for looping or
// end of sample.
for(size_t i = 0; i < AUDIO_BUF_LEN; i++) {
mix_buffer[i] += (0x80 + (u8)ch->data[ch->pos>>12]) * ch->vol;
ch->pos += ch->inc;
}
}
}
// Downsample and copy to the playing buffer.
for (size_t i = 0; i < AUDIO_BUF_LEN; ++i) {
// >> 6 to divide off the volume, >> 2 to divide by 4 channels to
// prevent overflow.
audio.current_buffer[i] = mix_buffer[i] >> 8;
}
}
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