mirror of
https://git.sr.ht/~rabbits/uxn
synced 2024-11-10 16:25:05 +00:00
328 lines
7.8 KiB
C
328 lines
7.8 KiB
C
#include "../uxn.h"
|
|
#include "audio.h"
|
|
#include <stdbool.h>
|
|
#include <string.h>
|
|
|
|
/*
|
|
Copyright (c) 2021-2023 Devine Lu Linvega, Andrew Alderwick, Bad Diode
|
|
|
|
Permission to use, copy, modify, and distribute this software for any
|
|
purpose with or without fee is hereby granted, provided that the above
|
|
copyright notice and this permission notice appear in all copies.
|
|
|
|
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
|
|
WITH REGARD TO THIS SOFTWARE.
|
|
*/
|
|
|
|
#define SOUND_TIMER (AUDIO_BUFSIZE / SAMPLE_FREQUENCY * 1000.0f)
|
|
#define XFADE_SAMPLES 100
|
|
#define INTERPOL_METHOD 1
|
|
|
|
typedef enum EnvStage {
|
|
ENV_ATTACK = (1 << 0),
|
|
ENV_DECAY = (1 << 1),
|
|
ENV_SUSTAIN = (1 << 2),
|
|
ENV_RELEASE = (1 << 3),
|
|
} EnvStage;
|
|
|
|
typedef struct Envelope {
|
|
float a;
|
|
float d;
|
|
float s;
|
|
float r;
|
|
float vol;
|
|
EnvStage stage;
|
|
} Envelope;
|
|
|
|
typedef struct Sample {
|
|
Uint8 *data;
|
|
float len;
|
|
float pos;
|
|
float inc;
|
|
float loop;
|
|
Uint8 pitch;
|
|
Envelope env;
|
|
} Sample;
|
|
|
|
typedef struct AudioChannel {
|
|
Sample sample;
|
|
Sample next_sample;
|
|
bool xfade;
|
|
float duration;
|
|
float vol_l;
|
|
float vol_r;
|
|
} AudioChannel;
|
|
|
|
AudioChannel channel[POLYPHONY];
|
|
|
|
/* clang-format off */
|
|
|
|
const float tuning[109] = {
|
|
0.00058853f, 0.00062352f, 0.00066060f, 0.00069988f, 0.00074150f,
|
|
0.00078559f, 0.00083230f, 0.00088179f, 0.00093423f, 0.00098978f,
|
|
0.00104863f, 0.00111099f, 0.00117705f, 0.00124704f, 0.00132120f,
|
|
0.00139976f, 0.00148299f, 0.00157118f, 0.00166460f, 0.00176359f,
|
|
0.00186845f, 0.00197956f, 0.00209727f, 0.00222198f, 0.00235410f,
|
|
0.00249409f, 0.00264239f, 0.00279952f, 0.00296599f, 0.00314235f,
|
|
0.00332921f, 0.00352717f, 0.00373691f, 0.00395912f, 0.00419454f,
|
|
0.00444396f, 0.00470821f, 0.00498817f, 0.00528479f, 0.00559904f,
|
|
0.00593197f, 0.00628471f, 0.00665841f, 0.00705434f, 0.00747382f,
|
|
0.00791823f, 0.00838908f, 0.00888792f, 0.00941642f, 0.00997635f,
|
|
0.01056957f, 0.01119807f, 0.01186395f, 0.01256941f, 0.01331683f,
|
|
0.01410869f, 0.01494763f, 0.01583647f, 0.01677815f, 0.01777583f,
|
|
0.01883284f, 0.01995270f, 0.02113915f, 0.02239615f, 0.02372789f,
|
|
0.02513882f, 0.02663366f, 0.02821738f, 0.02989527f, 0.03167293f,
|
|
0.03355631f, 0.03555167f, 0.03766568f, 0.03990540f, 0.04227830f,
|
|
0.04479229f, 0.04745578f, 0.05027765f, 0.05326731f, 0.05643475f,
|
|
0.05979054f, 0.06334587f, 0.06711261f, 0.07110333f, 0.07533136f,
|
|
0.07981079f, 0.08455659f, 0.08958459f, 0.09491156f, 0.10055530f,
|
|
0.10653463f, 0.11286951f, 0.11958108f, 0.12669174f, 0.13422522f,
|
|
0.14220667f, 0.15066272f, 0.15962159f, 0.16911318f, 0.17916918f,
|
|
0.18982313f, 0.20111060f, 0.21306926f, 0.22573902f, 0.23916216f,
|
|
0.25338348f, 0.26845044f, 0.28441334f, 0.30132544f,
|
|
};
|
|
|
|
/* clang-format on */
|
|
|
|
void
|
|
env_on(Envelope *env)
|
|
{
|
|
env->stage = ENV_ATTACK;
|
|
env->vol = 0.0f;
|
|
if(env->a > 0) {
|
|
env->a = (SOUND_TIMER / AUDIO_BUFSIZE) / env->a;
|
|
} else if(env->stage == ENV_ATTACK) {
|
|
env->stage = ENV_DECAY;
|
|
env->vol = 1.0f;
|
|
}
|
|
if(env->d < 10.0f) {
|
|
env->d = 10.0f;
|
|
}
|
|
env->d = (SOUND_TIMER / AUDIO_BUFSIZE) / env->d;
|
|
if(env->r < 10.0f) {
|
|
env->r = 10.0f;
|
|
}
|
|
env->r = (SOUND_TIMER / AUDIO_BUFSIZE) / env->r;
|
|
}
|
|
|
|
void
|
|
env_off(Envelope *env)
|
|
{
|
|
env->stage = ENV_RELEASE;
|
|
}
|
|
|
|
void
|
|
note_on(AudioChannel *channel, Uint16 duration, Uint8 *data, Uint16 len, Uint8 vol, Uint8 attack, Uint8 decay, Uint8 sustain, Uint8 release, Uint8 pitch, bool loop)
|
|
{
|
|
channel->duration = duration > 0 ? duration : len / 44.1f;
|
|
channel->vol_l = (vol >> 4) / 15.0f;
|
|
channel->vol_r = (vol & 0xf) / 15.0f;
|
|
|
|
Sample sample = {0};
|
|
sample.data = data;
|
|
sample.len = len;
|
|
sample.pos = 0;
|
|
sample.env.a = attack * 64.0f;
|
|
sample.env.d = decay * 64.0f;
|
|
sample.env.s = sustain / 16.0f;
|
|
sample.env.r = release * 64.0f;
|
|
if(loop) {
|
|
sample.loop = len;
|
|
} else {
|
|
sample.loop = 0;
|
|
}
|
|
env_on(&sample.env);
|
|
if(pitch < 20) {
|
|
pitch = 20;
|
|
}
|
|
float sample_rate = 44100 / 261.60;
|
|
if(len <= 256) {
|
|
sample_rate = len;
|
|
}
|
|
const float *inc = &tuning[pitch - 20];
|
|
sample.inc = *(inc)*sample_rate;
|
|
|
|
channel->next_sample = sample;
|
|
channel->xfade = true;
|
|
}
|
|
|
|
void
|
|
note_off(AudioChannel *channel, Uint16 duration)
|
|
{
|
|
channel->duration = duration;
|
|
env_off(&channel->sample.env);
|
|
}
|
|
|
|
void
|
|
env_advance(Envelope *env)
|
|
{
|
|
switch(env->stage) {
|
|
case ENV_ATTACK: {
|
|
env->vol += env->a;
|
|
if(env->vol >= 1.0f) {
|
|
env->stage = ENV_DECAY;
|
|
env->vol = 1.0f;
|
|
}
|
|
} break;
|
|
case ENV_DECAY: {
|
|
env->vol -= env->d;
|
|
if(env->vol <= env->s || env->d <= 0) {
|
|
env->stage = ENV_SUSTAIN;
|
|
env->vol = env->s;
|
|
}
|
|
} break;
|
|
case ENV_SUSTAIN: {
|
|
env->vol = env->s;
|
|
} break;
|
|
case ENV_RELEASE: {
|
|
if(env->vol <= 0 || env->r <= 0) {
|
|
env->vol = 0;
|
|
} else {
|
|
env->vol -= env->r;
|
|
}
|
|
} break;
|
|
}
|
|
}
|
|
|
|
float
|
|
interpolate_sample(Uint8 *data, Uint16 len, float pos)
|
|
{
|
|
#if INTERPOL_METHOD == 0
|
|
return data[(int)pos];
|
|
|
|
#elif INTERPOL_METHOD == 1
|
|
float x = pos;
|
|
int x0 = (int)x;
|
|
int x1 = (x0 + 1);
|
|
float y0 = data[x0];
|
|
float y1 = data[x1 % len];
|
|
x = x - x0;
|
|
float y = y0 + x * (y1 - y0);
|
|
return y;
|
|
|
|
#elif INTERPOL_METHOD == 2
|
|
float x = pos;
|
|
int x0 = x - 1;
|
|
int x1 = x;
|
|
int x2 = x + 1;
|
|
int x3 = x + 2;
|
|
float y0 = data[x0 % len];
|
|
float y1 = data[x1];
|
|
float y2 = data[x2 % len];
|
|
float y3 = data[x3 % len];
|
|
x = x - x1;
|
|
float c0 = y1;
|
|
float c1 = 0.5f * (y2 - y0);
|
|
float c2 = y0 - 2.5f * y1 + 2.f * y2 - 0.5f * y3;
|
|
float c3 = 1.5f * (y1 - y2) + 0.5f * (y3 - y0);
|
|
return ((c3 * x + c2) * x + c1) * x + c0;
|
|
#endif
|
|
}
|
|
|
|
Sint16
|
|
next_sample(Sample *sample)
|
|
{
|
|
if(sample->pos >= sample->len) {
|
|
if(sample->loop == 0) {
|
|
sample->data = 0;
|
|
return 0;
|
|
}
|
|
while(sample->pos >= sample->len) {
|
|
sample->pos -= sample->loop;
|
|
}
|
|
}
|
|
|
|
float val = interpolate_sample(sample->data, sample->len, sample->pos);
|
|
val *= sample->env.vol;
|
|
Sint8 next = (Sint8)0x80 ^ (Uint8)val;
|
|
|
|
sample->pos += sample->inc;
|
|
env_advance(&sample->env);
|
|
return next;
|
|
}
|
|
|
|
void
|
|
audio_handler(void *ctx, Uint8 *out_stream, int len)
|
|
{
|
|
Sint16 *stream = (Sint16 *)out_stream;
|
|
memset(stream, 0x00, len);
|
|
|
|
int n;
|
|
for(n = 0; n < POLYPHONY; n++) {
|
|
Uint8 device = (3 + n) << 4;
|
|
Uxn *u = (Uxn *)ctx;
|
|
Uint8 *addr = &u->dev[device];
|
|
if(channel[n].duration <= 0 && PEEK2(addr)) {
|
|
uxn_eval(u, PEEK2(addr));
|
|
}
|
|
channel[n].duration -= SOUND_TIMER;
|
|
|
|
int x = 0;
|
|
if(channel[n].xfade) {
|
|
float delta = 1.0f / (XFADE_SAMPLES * 2);
|
|
while(x < XFADE_SAMPLES * 2) {
|
|
float alpha = x * delta;
|
|
float beta = 1.0f - alpha;
|
|
Sint16 next_a = next_sample(&channel[n].next_sample);
|
|
Sint16 next_b = 0;
|
|
if(channel[n].sample.data != 0) {
|
|
next_b = next_sample(&channel[n].sample);
|
|
}
|
|
Sint16 next = alpha * next_a + beta * next_b;
|
|
stream[x++] += next * channel[n].vol_l;
|
|
stream[x++] += next * channel[n].vol_r;
|
|
}
|
|
channel[n].sample = channel[n].next_sample;
|
|
channel[n].xfade = false;
|
|
}
|
|
Sample *sample = &channel[n].sample;
|
|
while(x < len / 2) {
|
|
if(sample->data == 0) {
|
|
break;
|
|
}
|
|
Sint16 next = next_sample(sample);
|
|
stream[x++] += next * channel[n].vol_l;
|
|
stream[x++] += next * channel[n].vol_r;
|
|
}
|
|
}
|
|
int i;
|
|
for(i = 0; i < len / 2; i++) {
|
|
stream[i] <<= 6;
|
|
}
|
|
}
|
|
|
|
void
|
|
audio_start(int idx, Uint8 *d, Uxn *u)
|
|
{
|
|
Uint16 duration = PEEK2(d + 0x5);
|
|
Uint8 off = d[0xf] == 0x00;
|
|
|
|
if(!off) {
|
|
Uint16 addr = PEEK2(d + 0xc);
|
|
Uint8 *data = &u->ram[addr];
|
|
Uint16 len = PEEK2(d + 0xa);
|
|
Uint8 volume = d[0xe];
|
|
bool loop = !(d[0xf] & 0x80);
|
|
Uint8 pitch = d[0xf] & 0x7f;
|
|
Uint16 adsr = PEEK2(d + 0x8);
|
|
Uint8 attack = (adsr >> 12) & 0xF;
|
|
Uint8 decay = (adsr >> 8) & 0xF;
|
|
Uint8 sustain = (adsr >> 4) & 0xF;
|
|
Uint8 release = (adsr >> 0) & 0xF;
|
|
note_on(&channel[idx], duration, data, len, volume, attack, decay, sustain, release, pitch, loop);
|
|
} else {
|
|
note_off(&channel[idx], duration);
|
|
}
|
|
}
|
|
|
|
Uint8
|
|
audio_get_vu(int instance)
|
|
{
|
|
return channel[instance].sample.env.vol * 255.0f;
|
|
}
|
|
|
|
Uint16
|
|
audio_get_position(int instance)
|
|
{
|
|
return channel[instance].sample.pos;
|
|
}
|