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audio reworked

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This commit is contained in:
John Alanbrook 2023-11-27 20:29:55 +00:00
parent d047ceb07b
commit 44febe1c6d
27 changed files with 1858 additions and 1246 deletions
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4
docs/scripting.md
View file

@ -17,6 +17,6 @@ All objects in the Yugine can have an associated script. This script can perform
|update(dt)|called once per game frame tick| |update(dt)|called once per game frame tick|
|physupdate(dt)|called once per physics tick| |physupdate(dt)|called once per physics tick|
|stop|called when the object is killed| |stop|called when the object is killed|
|debug|use draw functions with the object's world position| |debug|use draw functions with the object's world position, during debug pass|
|gui|draw functions in screen space| |gui|draw functions in screen space, during gameplay gui pass|
|draw|draw functions in world space| |draw|draw functions in world space|

51
scripts/ai.js Normal file
View file

@ -0,0 +1,51 @@
var AI = {
tick() {
},
sequence(list) {
},
race(list) {
return function() {
var good = false;
list.forEach(x => if (x()) good = true);
return good;
};
},
do(times, list) {
},
sync(list) {
return function() {
var good = true;
list.forEach(x => if (!x()) good = false);
return good;
};
},
moveto(pos) {
return function() {
var dir = pos.sub(this.pos);
if (Vector.length(dir) < 10) return true;
this.velocity = Vector.normalize(pos.sub(this.pos)).scale(20);
return false;
}
},
wait(secs) {
secs ??= 1;
var accum = 0;
return function() {
accum += Game.dt;
if (accum >= secs)
return true;
return false;
};
},
};

5
scripts/base.js
View file

@ -522,6 +522,11 @@ Object.defineProperty(String.prototype, 'endswith', {
} }
}); });
Object.defineProperty(String.prototype, 'pct', {
value: function(val) {
}
});
Object.defineProperty(String.prototype, 'uc', { value: function() { return this.toUpperCase(); } }); Object.defineProperty(String.prototype, 'uc', { value: function() { return this.toUpperCase(); } });
Object.defineProperty(String.prototype, 'lc', {value:function() { return this.toLowerCase(); }}); Object.defineProperty(String.prototype, 'lc', {value:function() { return this.toLowerCase(); }});

3
scripts/engine.js
View file

@ -314,6 +314,7 @@ var timer = {
remain: 1, remain: 1,
loop: false, loop: false,
on: false, on: false,
apptime: false, /* If true, based on app's time instead of game world time */
start() { start() {
this.on = true; this.on = true;
}, },
@ -337,7 +338,7 @@ var timer = {
this.restart(); this.restart();
}, },
pct() { return this.remain / this.time; }, pct() { return 1 - (this.remain / this.time); },
kill() { kill() {
Register.unregister_obj(this); Register.unregister_obj(this);

17
scripts/entity.js
View file

@ -17,6 +17,8 @@ actor.kill = function(actor){};
actor.delay = function(fn, seconds) {}; actor.delay = function(fn, seconds) {};
actor.clock = function(fn){}; actor.clock = function(fn){};
var Empyrean = Object.create(actor);
var gameobject = { var gameobject = {
impl: { impl: {
full_path() { full_path() {
@ -58,16 +60,10 @@ var gameobject = {
}, },
cry(file) { cry(file) {
Sound.play(file); var p = Sound.play(file, Sound.bus.sfx);
return; var killfn = p.kill.bind(p);
this.timers.push(killfn);
if (this.curcry && !Sound.finished(this.curcry)) return; return killfn;
this.curcry = Sound.play(file);
var r = this.curcry;
Log.warn(r);
var fn = function() { Log.warn(r); if (r) Sound.stop(r); };
this.timers.push(fn);
return fn;
}, },
set max_velocity(x) { cmd(151, this.body, x); }, set max_velocity(x) { cmd(151, this.body, x); },
@ -324,7 +320,6 @@ var gameobject = {
if (!k.startswith("on_")) continue; if (!k.startswith("on_")) continue;
var signal = k.fromfirst("on_"); var signal = k.fromfirst("on_");
Event.observe(signal, obj, obj[k]); Event.observe(signal, obj, obj[k]);
Log.warn("REGISTERED " + signal);
}; };
obj.components.forEach(function(x) { obj.components.forEach(function(x) {

91
scripts/sound.js
View file

@ -1,39 +1,90 @@
var Audio = { var dsp_node = {
make(id) {
var n = Object.create(this);
n.id = id;
return n;
},
id: undefined,
get volume() { return cmd(175, this.id); },
set volume(x) { cmd(176, this.id,x); },
get pan() { return cmd(178,this.id); },
set pan(x) { cmd(179,this.id,x); },
off(t) { cmd(183, this.id, t); }, /* True to turn off */
bypass(t) { cmd(184, this.id, t); }, /* True to bypass filter effect */
unplug() { cmd(164, this.id); },
plugin(to) { cmd(177, this.id, to.id); },
kill() {
if (this._dead) return;
this._dead = true;
cmd(193, this.id); },
}; };
var dsp_source = Object.copy(dsp_node,{
end(){},
get loop() { return cmd(194,this.id); },
set loop(x) { cmd(195,this.id, x);},
});
var Sound = { var Sound = {
bus: {},
sounds: [], /* array of loaded sound files */ sounds: [], /* array of loaded sound files */
play(file) { play(file, bus) {
if (!IO.exists(file)) { if (!IO.exists(file)) {
Log.error(`Cannot play sound ${file}: does not exist.`); Log.error(`Cannot play sound ${file}: does not exist.`);
return; return;
} }
var p = cmd(14,file); var src = DSP.source(file);
return p; bus ??= Sound.bus.master;
}, src.plugin(bus);
return src;
finished(sound) {
return cmd(165, sound);
},
stop(sound) {
cmd(164, sound);
}, },
music(midi, sf) { music(midi, sf) {
cmd(13, midi, sf); cmd(13, midi, sf);
}, },
musicstop() {
cmd(15);
},
/* Between 0 and 100 */
set volume(x) { cmd(19, x); },
get volume() { 0; },
}; };
var DSP = {
mix(to) {
var n = dsp_node.make(cmd(181));
if (to) n.plugin(to);
return n;
},
source(path) {
var src = Object.create(dsp_source);
src.id = cmd(182,path,src);
return src;
},
delay(secs,decay) {
return dsp_node.make(cmd(185, secs, decay));
},
lpf(f) {
return dsp_node.make(cmd(186,f));
},
hpf(f) {
return dsp_node.make(cmd(187,f));
},
mod(path) {
return dsp_node.make(cmd(188,path));
},
crush(rate, depth) {
return dsp_node.make(cmd(189,rate,depth));
},
compressor() {
return dsp_node.make(cmd(190));
},
limiter(ceil) {
return dsp_node.make(cmd(191,ceil));
},
noise_gate(floor) {
return dsp_node.make(cmd(192,floor));
},
};
Sound.bus.master = dsp_node.make(cmd(180));
Sound.master = Sound.bus.master;
Sound.play.doc = "Play the given file once."; Sound.play.doc = "Play the given file once.";
Sound.doc = {}; Sound.doc = {};
Sound.doc.volume = "Set the master volume. 0 is no sound and 100 is loudest."; Sound.doc.volume = "Set the master volume. 0 is no sound and 100 is loudest.";

124
source/engine/cbuf.h
View file

@ -3,90 +3,50 @@
#include <stdio.h> #include <stdio.h>
#include <stdint.h> #include <stdint.h>
#include <stdlib.h>
struct circbuf { static inline unsigned int powof2(unsigned int x)
int16_t *data; {
uint32_t read; x = x-1;
uint32_t write; x |= (x >> 1);
unsigned int len; x |= (x >> 2);
x |= (x >> 4);
x |= (x >> 8);
x |= (x >> 16);
return x+1;
}
struct rheader
{
unsigned int read;
unsigned int write;
int len;
}; };
struct circbuf *circbuf_make(size_t size, unsigned int len); #define ringheader(r) ((struct rheader *)r-1)
struct circbuf circbuf_init(size_t size, unsigned int len);
void cbuf_push(struct circbuf *buf, short data); static inline void *ringmake(void *ring, size_t elemsize, unsigned int n)
short cbuf_shift(struct circbuf *buf); {
n = powof2(n);
#endif if (ring) {
struct rheader *h = ringheader(ring);
#ifdef CBUF_IMPLEMENT if (n <= h->len) return h+1;
h = realloc(h, elemsize*n+sizeof(struct rheader));
#include "assert.h" return h+1;
#include "stdlib.h" }
unsigned int powof2(unsigned int num) struct rheader *h = malloc(elemsize*n+sizeof(struct rheader));
{ h->len = n; h->read = 0; h->write = 0;
if (num != 0) { return h+1;
num--; }
num |= (num >> 1);
num |= (num >> 2); #define ringnew(r,n) (r = ringmake(r, sizeof(*r),n))
num |= (num >> 4); #define ringfree(r) ((r) ? free(ringheader(r)) : 0)
num |= (num >> 8); #define ringmask(r,v) (v & (ringheader(r)->len-1))
num |= (num >> 16); #define ringpush(r,v) (r[ringmask(r,ringheader(r)->write++)] = v)
num++; #define ringshift(r) (r[ringmask(r,ringheader(r)->read++)])
} #define ringsize(r) ((r) ? ringheader(r)->write - ringheader(r)->read : 0)
#define ringfull(r) ((r) ? ringsize(r) == ringheader(r)->len : 0)
return num; #define ringempty(r) ((r) ? ringheader(r)->read == ringheader(r)->write : 0)
}
int ispow2(int num)
{
return (num && !(num & (num - 1)));
}
struct circbuf *circbuf_make(size_t size, unsigned int len)
{
struct circbuf *self = malloc(sizeof(*self));
self->len = powof2(len);
self->data = calloc(sizeof(short), self->len);
self->read = self->write = 0;
return self;
}
struct circbuf circbuf_init(size_t size, unsigned int len)
{
struct circbuf self;
self.len = powof2(len);
self.data = calloc(sizeof(short), self.len);
self.read = self.write = 0;
return self;
}
int cbuf_size(struct circbuf *buf) {
return buf->write - buf->read;
}
int cbuf_empty(struct circbuf *buf) {
return buf->read == buf->write;
}
int cbuf_full(struct circbuf *buf) {
return cbuf_size(buf) >= buf->len;
}
uint32_t cbuf_mask(struct circbuf *buf, uint32_t n) {
return n & (buf->len-1);
}
void cbuf_push(struct circbuf *buf, short data) {
//assert(!cbuf_full(buf));
buf->data[cbuf_mask(buf,buf->write++)] = data;
}
short cbuf_shift(struct circbuf *buf) {
//assert(!cbuf_empty(buf));
return buf->data[cbuf_mask(buf, buf->read++)];
}
#endif #endif

3
source/engine/config.c
View file

@ -13,6 +13,9 @@
#define MSF_GIF_IMPL #define MSF_GIF_IMPL
#include "msf_gif.h" #include "msf_gif.h"
#define STB_HEXWAVE_IMPLEMENTATION
#include "stb_hexwave.h"
#define STB_DS_IMPLEMENTATION #define STB_DS_IMPLEMENTATION
#include <stb_ds.h> #include <stb_ds.h>

73
source/engine/datastream.c
View file

@ -5,7 +5,6 @@
#include "iir.h" #include "iir.h"
#include "limits.h" #include "limits.h"
#include "log.h" #include "log.h"
#include "mix.h"
#include "resources.h" #include "resources.h"
#include "shader.h" #include "shader.h"
#include "sound.h" #include "sound.h"
@ -17,7 +16,6 @@
#include "mpeg2.sglsl.h" #include "mpeg2.sglsl.h"
#define CBUF_IMPLEMENT
#include "cbuf.h" #include "cbuf.h"
#include "sokol/sokol_gfx.h" #include "sokol/sokol_gfx.h"
@ -26,8 +24,13 @@ sg_shader vid_shader;
sg_pipeline vid_pipeline; sg_pipeline vid_pipeline;
sg_bindings vid_bind; sg_bindings vid_bind;
static void render_frame(plm_t *mpeg, plm_frame_t *frame, void *user) { void soundstream_fillbuf(struct datastream *ds, soundbyte *buf, int frames) {
struct datastream *ds = user; for (int i = 0; i < frames*CHANNELS; i++)
buf[i] = ringshift(ds->ring);
}
static void render_frame(plm_t *mpeg, plm_frame_t *frame, struct datastream *ds) {
return;
uint8_t rgb[frame->height*frame->width*4]; uint8_t rgb[frame->height*frame->width*4];
plm_frame_to_rgba(frame, rgb, frame->width*4); plm_frame_to_rgba(frame, rgb, frame->width*4);
sg_image_data imgd; sg_image_data imgd;
@ -40,32 +43,29 @@ static void render_frame(plm_t *mpeg, plm_frame_t *frame, void *user) {
sg_update_image(ds->img, &imgd); sg_update_image(ds->img, &imgd);
} }
static void render_audio(plm_t *mpeg, plm_samples_t *samples, void *user) { static void render_audio(plm_t *mpeg, plm_samples_t *samples, struct datastream *ds) {
struct datastream *ds = user; for (int i = 0; i < samples->count * CHANNELS; i++)
short t; ringpush(ds->ring, samples->interleaved[i]);
for (int i = 0; i < samples->count * CHANNELS; i++) {
t = (short)(samples->interleaved[i] * SHRT_MAX);
// cbuf_push(ds->astream->buf, t * 5);
}
} }
void ds_openvideo(struct datastream *ds, const char *video, const char *adriver) { struct datastream *ds_openvideo(const char *path)
{
struct datastream *ds = malloc(sizeof(*ds));
size_t rawlen; size_t rawlen;
void *raw; void *raw;
raw = slurp_file(video, &rawlen); raw = slurp_file(path, &rawlen);
ds->plm = plm_create_with_memory(raw, rawlen, 0); ds->plm = plm_create_with_memory(raw, rawlen, 0);
free(raw); free(raw);
if (!ds->plm) { if (!ds->plm) {
YughError("Couldn't open %s", video); YughError("Couldn't open %s", path);
} }
YughWarn("Opened %s - framerate: %f, samplerate: %d, audio streams: %i, duration: %f", YughWarn("Opened %s - framerate: %f, samplerate: %d,audio streams: %d, duration: %g",
video, path,
plm_get_framerate(ds->plm), plm_get_framerate(ds->plm),
plm_get_samplerate(ds->plm), plm_get_samplerate(ds->plm),
plm_get_num_audio_streams(ds->plm),
plm_get_duration(ds->plm)); plm_get_duration(ds->plm));
@ -74,28 +74,8 @@ void ds_openvideo(struct datastream *ds, const char *video, const char *adriver)
.height = plm_get_height(ds->plm) .height = plm_get_height(ds->plm)
}); });
ds->ring = ringnew(ds->ring, 8192);
ds->astream = soundstream_make(); plugin_node(make_node(ds, soundstream_fillbuf), masterbus);
struct dsp_filter astream_filter;
astream_filter.data = &ds->astream;
astream_filter.filter = soundstream_fillbuf;
// struct dsp_filter lpf = lpf_make(8, 10000);
struct dsp_filter lpf = lpf_make(1, 200);
struct dsp_iir *iir = lpf.data;
iir->in = astream_filter;
struct dsp_filter hpf = hpf_make(1, 2000);
struct dsp_iir *hiir = hpf.data;
hiir->in = astream_filter;
/*
struct dsp_filter llpf = lp_fir_make(20);
struct dsp_fir *fir = llpf.data;
fir->in = astream_filter;
*/
// first_free_bus(astream_filter);
plm_set_video_decode_callback(ds->plm, render_frame, ds); plm_set_video_decode_callback(ds->plm, render_frame, ds);
plm_set_audio_decode_callback(ds->plm, render_audio, ds); plm_set_audio_decode_callback(ds->plm, render_audio, ds);
@ -108,19 +88,19 @@ void ds_openvideo(struct datastream *ds, const char *video, const char *adriver)
plm_set_audio_lead_time(ds->plm, BUF_FRAMES / SAMPLERATE); plm_set_audio_lead_time(ds->plm, BUF_FRAMES / SAMPLERATE);
ds->playing = true; ds->playing = true;
return ds;
} }
void MakeDatastream() { void MakeDatastream() {
vid_shader = sg_make_shader(mpeg2_shader_desc(sg_query_backend()));} vid_shader = sg_make_shader(mpeg2_shader_desc(sg_query_backend()));
}
void ds_advance(struct datastream *ds, double s) { void ds_advance(struct datastream *ds, double s) {
if (ds->playing) { if (ds->playing) plm_decode(ds->plm, s);
plm_decode(ds->plm, s);
}
} }
void ds_seek(struct datastream *ds, double time) { void ds_seek(struct datastream *ds, double time) {
// clear_raw(ds->audio_device);
plm_seek(ds->plm, time, false); plm_seek(ds->plm, time, false);
} }
@ -140,8 +120,7 @@ void ds_stop(struct datastream *ds) {
plm_destroy(ds->plm); plm_destroy(ds->plm);
ds->plm = NULL; ds->plm = NULL;
} }
if (ds->audio_device)
close_audio_device(ds->audio_device);
ds->playing = false; ds->playing = false;
} }

7
source/engine/datastream.h
View file

@ -3,6 +3,8 @@
#include <pl_mpeg.h> #include <pl_mpeg.h>
#include <stdint.h> #include <stdint.h>
#include "dsp.h"
#include "utringbuffer.h"
#include "sokol/sokol_gfx.h" #include "sokol/sokol_gfx.h"
@ -12,17 +14,16 @@ struct datastream {
plm_t *plm; plm_t *plm;
double last_time; double last_time;
int playing; int playing;
int audio_device;
sg_image img; sg_image img;
int width; int width;
int height; int height;
struct soundstream *astream; soundbyte *ring;
}; };
struct Texture; struct Texture;
void MakeDatastream(); void MakeDatastream();
void ds_openvideo(struct datastream *ds, const char *path, const char *adriver); struct datastream *ds_openvideo(const char *path);
struct Texture *ds_maketexture(struct datastream *); struct Texture *ds_maketexture(struct datastream *);
void ds_advance(struct datastream *ds, double); void ds_advance(struct datastream *ds, double);
void ds_seek(struct datastream *ds, double); void ds_seek(struct datastream *ds, double);

94
source/engine/jsffi.c
View file

@ -10,10 +10,10 @@
#include "input.h" #include "input.h"
#include "level.h" #include "level.h"
#include "log.h" #include "log.h"
#include "mix.h" #include "dsp.h"
#include "music.h" #include "music.h"
#include "2dphysics.h" #include "2dphysics.h"
#include "datastream.h"
#include "sound.h" #include "sound.h"
#include "sprite.h" #include "sprite.h"
#include "stb_ds.h" #include "stb_ds.h"
@ -546,12 +546,12 @@ JSValue duk_cmd(JSContext *js, JSValueConst this, int argc, JSValueConst *argv)
case 14: case 14:
str = JS_ToCString(js, argv[1]); str = JS_ToCString(js, argv[1]);
//ret = ptr2js(play_sound(make_sound(str))); ret = ptr2js(dsp_source(str));
play_oneshot(make_sound(str)); ((sound*)((dsp_node*)js2ptr(ret))->data)->hook = argv[2];
break; break;
case 15: case 15:
music_stop(); // music_stop();
break; break;
case 16: case 16:
@ -1162,10 +1162,10 @@ JSValue duk_cmd(JSContext *js, JSValueConst this, int argc, JSValueConst *argv)
ret = int2js(rename(str, str2)); ret = int2js(rename(str, str2));
break; break;
case 164: case 164:
sound_stop(js2ptr(argv[1])); unplug_node(js2ptr(argv[1]));
break; break;
case 165: case 165:
ret = bool2js(sound_paused(js2ptr(argv[1]))); // ret = bool2js(sound_paused(js2ptr(argv[1])));
break; break;
case 166: case 166:
str = js2str(argv[1]); str = js2str(argv[1]);
@ -1194,6 +1194,77 @@ JSValue duk_cmd(JSContext *js, JSValueConst this, int argc, JSValueConst *argv)
str = js2str(argv[1]); str = js2str(argv[1]);
capture_screen(js2number(argv[2]), js2number(argv[3]), js2number(argv[4]), js2number(argv[5]), str); capture_screen(js2number(argv[2]), js2number(argv[3]), js2number(argv[4]), js2number(argv[5]), str);
break; break;
case 174:
str = js2str(argv[1]);
ds_openvideo(str);
break;
case 175:
ret = num2js(((dsp_node*)js2ptr(argv[1]))->gain);
break;
case 176:
((dsp_node*)js2ptr(argv[1]))->gain = js2number(argv[2]);
break;
case 177:
plugin_node(js2ptr(argv[1]), js2ptr(argv[2]));
break;
case 178:
ret = num2js(((dsp_node*)js2ptr(argv[1]))->pan);
break;
case 179:
((dsp_node*)js2ptr(argv[1]))->pan=js2number(argv[2]);
break;
case 180:
ret = ptr2js(masterbus);
break;
case 181:
ret = ptr2js(make_node(NULL,NULL));
break;
case 182:
str = js2str(argv[1]);
ret = ptr2js(dsp_source(str));
((sound*)((dsp_node*)js2ptr(ret))->data)->hook = JS_DupValue(js,argv[2]);
break;
case 183:
((dsp_node*)js2ptr(argv[1]))->off = js2bool(argv[2]);
break;
case 184:
((dsp_node*)js2ptr(argv[1]))->pass = js2bool(argv[2]);
break;
case 185:
ret = ptr2js(dsp_delay(js2number(argv[1]), js2number(argv[2])));
break;
case 186:
ret = ptr2js(dsp_lpf(js2number(argv[1])));
break;
case 187:
ret = ptr2js(dsp_hpf(js2number(argv[1])));
break;
case 188:
str = js2str(argv[1]);
ret = ptr2js(dsp_mod(str));
break;
case 189:
ret = ptr2js(dsp_bitcrush(js2number(argv[1]), js2number(argv[2])));
break;
case 190:
ret = ptr2js(dsp_compressor());
break;
case 191:
ret = ptr2js(dsp_limiter(js2number(argv[1])));
break;
case 192:
ret = ptr2js(dsp_noise_gate(js2number(argv[1])));
break;
case 193:
node_free(js2ptr(argv[1]));
break;
case 194:
ret = bool2js(((sound*)((dsp_node*)js2ptr(argv[1]))->data)->loop);
break;
case 195:
((sound*)((dsp_node*)js2ptr(argv[1]))->data)->loop = js2bool(argv[2]);
break;
} }
if (str) if (str)
@ -1701,10 +1772,11 @@ JSValue duk_anim(JSContext *js, JSValueConst this, int argc, JSValueConst *argv)
} }
JSValue duk_make_timer(JSContext *js, JSValueConst this, int argc, JSValueConst *argv) { JSValue duk_make_timer(JSContext *js, JSValueConst this, int argc, JSValueConst *argv) {
double secs = js2number(argv[1]); // double secs = js2number(argv[1]);
struct callee *c = make_callee(argv[0], argv[3]); // struct callee *c = make_callee(argv[0], argv[3]);
int id = timer_make(secs, call_callee, c, 1, js2bool(argv[2])); // int id = timer_make(secs, call_callee, c, 1, js2bool(argv[2]));
return JS_NewInt64(js, id); // return JS_NewInt64(js, id);
return JS_NULL;
} }
JSValue duk_cmd_points(JSContext *js, JSValueConst this, int argc, JSValueConst *argv) JSValue duk_cmd_points(JSContext *js, JSValueConst this, int argc, JSValueConst *argv)

871
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/* See end of file for license */
#ifndef POCKETMOD_H_INCLUDED
#define POCKETMOD_H_INCLUDED
#ifdef __cplusplus
extern "C" {
#endif
typedef struct pocketmod_context pocketmod_context;
int pocketmod_init(pocketmod_context *c, const void *data, int size, int rate);
int pocketmod_render(pocketmod_context *c, void *buffer, int size);
int pocketmod_loop_count(pocketmod_context *c);
#ifndef POCKETMOD_MAX_CHANNELS
#define POCKETMOD_MAX_CHANNELS 32
#endif
#ifndef POCKETMOD_MAX_SAMPLES
#define POCKETMOD_MAX_SAMPLES 31
#endif
typedef struct {
signed char *data; /* Sample data buffer */
unsigned int length; /* Data length (in bytes) */
} _pocketmod_sample;
typedef struct {
unsigned char dirty; /* Pitch/volume dirty flags */
unsigned char sample; /* Sample number (0..31) */
unsigned char volume; /* Base volume without tremolo (0..64) */
unsigned char balance; /* Stereo balance (0..255) */
unsigned short period; /* Note period (113..856) */
unsigned short delayed; /* Delayed note period (113..856) */
unsigned short target; /* Target period (for tone portamento) */
unsigned char finetune; /* Note finetune (0..15) */
unsigned char loop_count; /* E6x loop counter */
unsigned char loop_line; /* E6x target line */
unsigned char lfo_step; /* Vibrato/tremolo LFO step counter */
unsigned char lfo_type[2]; /* LFO type for vibrato/tremolo */
unsigned char effect; /* Current effect (0x0..0xf or 0xe0..0xef) */
unsigned char param; /* Raw effect parameter value */
unsigned char param3; /* Parameter memory for 3xx */
unsigned char param4; /* Parameter memory for 4xy */
unsigned char param7; /* Parameter memory for 7xy */
unsigned char param9; /* Parameter memory for 9xx */
unsigned char paramE1; /* Parameter memory for E1x */
unsigned char paramE2; /* Parameter memory for E2x */
unsigned char paramEA; /* Parameter memory for EAx */
unsigned char paramEB; /* Parameter memory for EBx */
unsigned char real_volume; /* Volume (with tremolo adjustment) */
float position; /* Position in sample data buffer */
float increment; /* Position increment per output sample */
} _pocketmod_chan;
struct pocketmod_context
{
/* Read-only song data */
_pocketmod_sample samples[POCKETMOD_MAX_SAMPLES];
unsigned char *source; /* Pointer to source MOD data */
unsigned char *order; /* Pattern order table */
unsigned char *patterns; /* Start of pattern data */
unsigned char length; /* Patterns in the order (1..128) */
unsigned char reset; /* Pattern to loop back to (0..127) */
unsigned char num_patterns; /* Patterns in the file (1..128) */
unsigned char num_samples; /* Sample count (15 or 31) */
unsigned char num_channels; /* Channel count (1..32) */
/* Timing variables */
int samples_per_second; /* Sample rate (set by user) */
int ticks_per_line; /* A.K.A. song speed (initially 6) */
float samples_per_tick; /* Depends on sample rate and BPM */
/* Loop detection state */
unsigned char visited[16]; /* Bit mask of previously visited patterns */
int loop_count; /* How many times the song has looped */
/* Render state */
_pocketmod_chan channels[POCKETMOD_MAX_CHANNELS];
unsigned char pattern_delay;/* EEx pattern delay counter */
unsigned int lfo_rng; /* RNG used for the random LFO waveform */
/* Position in song (from least to most granular) */
signed char pattern; /* Current pattern in order */
signed char line; /* Current line in pattern */
short tick; /* Current tick in line */
float sample; /* Current sample in tick */
};
#ifdef POCKETMOD_IMPLEMENTATION
/* Memorize a parameter unless the new value is zero */
#define POCKETMOD_MEM(dst, src) do { \
(dst) = (src) ? (src) : (dst); \
} while (0)
/* Same thing, but memorize each nibble separately */
#define POCKETMOD_MEM2(dst, src) do { \
(dst) = (((src) & 0x0f) ? ((src) & 0x0f) : ((dst) & 0x0f)) \
| (((src) & 0xf0) ? ((src) & 0xf0) : ((dst) & 0xf0)); \
} while (0)
/* Shortcut to sample metadata (sample must be nonzero) */
#define POCKETMOD_SAMPLE(c, sample) ((c)->source + 12 + 30 * (sample))
/* Channel dirty flags */
#define POCKETMOD_PITCH 0x01
#define POCKETMOD_VOLUME 0x02
/* The size of one sample in bytes */
#define POCKETMOD_SAMPLE_SIZE sizeof(float[2])
/* Finetune adjustment table. Three octaves for each finetune setting. */
static const signed char _pocketmod_finetune[16][36] = {
{ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
{ -6, -6, -5, -5, -4, -3, -3, -3, -3, -3, -3, -3, -3, -3, -2, -3, -2, -2, -2, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, 0},
{-12,-12,-10,-11, -8, -8, -7, -7, -6, -6, -6, -6, -6, -6, -5, -5, -4, -4, -4, -3, -3, -3, -3, -2, -3, -3, -2, -3, -3, -2, -2, -2, -2, -2, -2, -1},
{-18,-17,-16,-16,-13,-12,-12,-11,-10,-10,-10, -9, -9, -9, -8, -8, -7, -6, -6, -5, -5, -5, -5, -4, -5, -4, -3, -4, -4, -3, -3, -3, -3, -2, -2, -2},
{-24,-23,-21,-21,-18,-17,-16,-15,-14,-13,-13,-12,-12,-12,-11,-10, -9, -8, -8, -7, -7, -7, -7, -6, -6, -6, -5, -5, -5, -4, -4, -4, -4, -3, -3, -3},
{-30,-29,-26,-26,-23,-21,-20,-19,-18,-17,-17,-16,-15,-14,-13,-13,-11,-11,-10, -9, -9, -9, -8, -7, -8, -7, -6, -6, -6, -5, -5, -5, -5, -4, -4, -4},
{-36,-34,-32,-31,-27,-26,-24,-23,-22,-21,-20,-19,-18,-17,-16,-15,-14,-13,-12,-11,-11,-10,-10, -9, -9, -9, -7, -8, -7, -6, -6, -6, -6, -5, -5, -4},
{-42,-40,-37,-36,-32,-30,-29,-27,-25,-24,-23,-22,-21,-20,-18,-18,-16,-15,-14,-13,-13,-12,-12,-10,-10,-10, -9, -9, -9, -8, -7, -7, -7, -6, -6, -5},
{ 51, 48, 46, 42, 42, 38, 36, 34, 32, 30, 24, 27, 25, 24, 23, 21, 21, 19, 18, 17, 16, 15, 14, 14, 12, 12, 12, 10, 10, 10, 9, 8, 8, 8, 7, 7},
{ 44, 42, 40, 37, 37, 35, 32, 31, 29, 27, 25, 24, 22, 21, 20, 19, 18, 17, 16, 15, 15, 14, 13, 12, 11, 10, 10, 9, 9, 9, 8, 7, 7, 7, 6, 6},
{ 38, 36, 34, 32, 31, 30, 28, 27, 25, 24, 22, 21, 19, 18, 17, 16, 16, 15, 14, 13, 13, 12, 11, 11, 9, 9, 9, 8, 7, 7, 7, 6, 6, 6, 5, 5},
{ 31, 30, 29, 26, 26, 25, 24, 22, 21, 20, 18, 17, 16, 15, 14, 13, 13, 12, 12, 11, 11, 10, 9, 9, 8, 7, 8, 7, 6, 6, 6, 5, 5, 5, 5, 5},
{ 25, 24, 23, 21, 21, 20, 19, 18, 17, 16, 14, 14, 13, 12, 11, 10, 11, 10, 10, 9, 9, 8, 7, 7, 6, 6, 6, 5, 5, 5, 5, 4, 4, 4, 3, 4},
{ 19, 18, 17, 16, 16, 15, 15, 14, 13, 12, 11, 10, 9, 9, 9, 8, 8, 18, 7, 7, 7, 6, 5, 6, 5, 4, 5, 4, 4, 4, 4, 3, 3, 3, 3, 3},
{ 12, 12, 12, 10, 11, 11, 10, 10, 9, 8, 7, 7, 6, 6, 6, 5, 6, 5, 5, 5, 5, 4, 4, 4, 3, 3, 3, 3, 2, 3, 3, 2, 2, 2, 2, 2},
{ 6, 6, 6, 5, 6, 6, 6, 5, 5, 5, 4, 4, 3, 3, 3, 3, 3, 3, 3, 3, 3, 2, 2, 2, 2, 1, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1}
};
/* Min/max helper functions */
static int _pocketmod_min(int x, int y) { return x < y ? x : y; }
static int _pocketmod_max(int x, int y) { return x > y ? x : y; }
/* Clamp a volume value to the 0..64 range */
static int _pocketmod_clamp_volume(int x)
{
x = _pocketmod_max(x, 0x00);
x = _pocketmod_min(x, 0x40);
return x;
}
/* Zero out a block of memory */
static void _pocketmod_zero(void *data, int size)
{
char *byte = data, *end = byte + size;
while (byte != end) { *byte++ = 0; }
}
/* Convert a period (at finetune = 0) to a note index in 0..35 */
static int _pocketmod_period_to_note(int period)
{
switch (period) {
case 856: return 0; case 808: return 1; case 762: return 2;
case 720: return 3; case 678: return 4; case 640: return 5;
case 604: return 6; case 570: return 7; case 538: return 8;
case 508: return 9; case 480: return 10; case 453: return 11;
case 428: return 12; case 404: return 13; case 381: return 14;
case 360: return 15; case 339: return 16; case 320: return 17;
case 302: return 18; case 285: return 19; case 269: return 20;
case 254: return 21; case 240: return 22; case 226: return 23;
case 214: return 24; case 202: return 25; case 190: return 26;
case 180: return 27; case 170: return 28; case 160: return 29;
case 151: return 30; case 143: return 31; case 135: return 32;
case 127: return 33; case 120: return 34; case 113: return 35;
default: return 0;
}
}
/* Table-based sine wave oscillator */
static int _pocketmod_sin(int step)
{
/* round(sin(x * pi / 32) * 255) for x in 0..15 */
static const unsigned char sin[16] = {
0x00, 0x19, 0x32, 0x4a, 0x62, 0x78, 0x8e, 0xa2,
0xb4, 0xc5, 0xd4, 0xe0, 0xec, 0xf4, 0xfa, 0xfe
};
int x = sin[step & 0x0f];
x = (step & 0x1f) < 0x10 ? x : 0xff - x;
return step < 0x20 ? x : -x;
}
/* Oscillators for vibrato/tremolo effects */
static int _pocketmod_lfo(pocketmod_context *c, _pocketmod_chan *ch, int step)
{
switch (ch->lfo_type[ch->effect == 7] & 3) {
case 0: return _pocketmod_sin(step & 0x3f); /* Sine */
case 1: return 0xff - ((step & 0x3f) << 3); /* Saw */
case 2: return (step & 0x3f) < 0x20 ? 0xff : -0xff; /* Square */
case 3: return (c->lfo_rng & 0x1ff) - 0xff; /* Random */
default: return 0; /* Hush little compiler */
}
}
static void _pocketmod_update_pitch(pocketmod_context *c, _pocketmod_chan *ch)
{
/* Don't do anything if the period is zero */
ch->increment = 0.0f;
if (ch->period) {
float period = ch->period;
/* Apply vibrato (if active) */
if (ch->effect == 0x4 || ch->effect == 0x6) {
int step = (ch->param4 >> 4) * ch->lfo_step;
int rate = ch->param4 & 0x0f;
period += _pocketmod_lfo(c, ch, step) * rate / 128.0f;
/* Apply arpeggio (if active) */
} else if (ch->effect == 0x0 && ch->param) {
static const float arpeggio[16] = { /* 2^(X/12) for X in 0..15 */
1.000000f, 1.059463f, 1.122462f, 1.189207f,
1.259921f, 1.334840f, 1.414214f, 1.498307f,
1.587401f, 1.681793f, 1.781797f, 1.887749f,
2.000000f, 2.118926f, 2.244924f, 2.378414f
};
int step = (ch->param >> ((2 - c->tick % 3) << 2)) & 0x0f;
period /= arpeggio[step];
}
/* Calculate sample buffer position increment */
ch->increment = 3546894.6f / (period * c->samples_per_second);
}
/* Clear the pitch dirty flag */
ch->dirty &= ~POCKETMOD_PITCH;
}
static void _pocketmod_update_volume(pocketmod_context *c, _pocketmod_chan *ch)
{
int volume = ch->volume;
if (ch->effect == 0x7) {
int step = ch->lfo_step * (ch->param7 >> 4);
volume += _pocketmod_lfo(c, ch, step) * (ch->param7 & 0x0f) >> 6;
}
ch->real_volume = _pocketmod_clamp_volume(volume);
ch->dirty &= ~POCKETMOD_VOLUME;
}
static void _pocketmod_pitch_slide(_pocketmod_chan *ch, int amount)
{
int max = 856 + _pocketmod_finetune[ch->finetune][ 0];
int min = 113 + _pocketmod_finetune[ch->finetune][35];
ch->period += amount;
ch->period = _pocketmod_max(ch->period, min);
ch->period = _pocketmod_min(ch->period, max);
ch->dirty |= POCKETMOD_PITCH;
}
static void _pocketmod_volume_slide(_pocketmod_chan *ch, int param)
{
/* Undocumented quirk: If both x and y are nonzero, then the value of x */
/* takes precedence. (Yes, there are songs that rely on this behavior.) */
int change = (param & 0xf0) ? (param >> 4) : -(param & 0x0f);
ch->volume = _pocketmod_clamp_volume(ch->volume + change);
ch->dirty |= POCKETMOD_VOLUME;
}
static void _pocketmod_next_line(pocketmod_context *c)
{
unsigned char (*data)[4];
int i, pos, pattern_break = -1;
/* When entering a new pattern order index, mark it as "visited" */
if (c->line == 0) {
c->visited[c->pattern >> 3] |= 1 << (c->pattern & 7);
}
/* Move to the next pattern if this was the last line */
if (++c->line == 64) {
if (++c->pattern == c->length) {
c->pattern = c->reset;
}
c->line = 0;
}
/* Find the pattern data for the current line */
pos = (c->order[c->pattern] * 64 + c->line) * c->num_channels * 4;
data = (unsigned char(*)[4]) (c->patterns + pos);
for (i = 0; i < c->num_channels; i++) {
/* Decode columns */
int sample = (data[i][0] & 0xf0) | (data[i][2] >> 4);
int period = ((data[i][0] & 0x0f) << 8) | data[i][1];
int effect = ((data[i][2] & 0x0f) << 8) | data[i][3];
/* Memorize effect parameter values */
_pocketmod_chan *ch = &c->channels[i];
ch->effect = (effect >> 8) != 0xe ? (effect >> 8) : (effect >> 4);
ch->param = (effect >> 8) != 0xe ? (effect & 0xff) : (effect & 0x0f);
/* Set sample */
if (sample) {
if (sample <= POCKETMOD_MAX_SAMPLES) {
unsigned char *sample_data = POCKETMOD_SAMPLE(c, sample);
ch->sample = sample;
ch->finetune = sample_data[2] & 0x0f;
ch->volume = _pocketmod_min(sample_data[3], 0x40);
if (ch->effect != 0xED) {
ch->dirty |= POCKETMOD_VOLUME;
}
} else {
ch->sample = 0;
}
}
/* Set note */
if (period) {
int note = _pocketmod_period_to_note(period);
period += _pocketmod_finetune[ch->finetune][note];
if (ch->effect != 0x3) {
if (ch->effect != 0xED) {
ch->period = period;
ch->dirty |= POCKETMOD_PITCH;
ch->position = 0.0f;
ch->lfo_step = 0;
} else {
ch->delayed = period;
}
}
}
/* Handle pattern effects */
switch (ch->effect) {
/* Memorize parameters */
case 0x3: POCKETMOD_MEM(ch->param3, ch->param); /* Fall through */
case 0x5: POCKETMOD_MEM(ch->target, period); break;
case 0x4: POCKETMOD_MEM2(ch->param4, ch->param); break;
case 0x7: POCKETMOD_MEM2(ch->param7, ch->param); break;
case 0xE1: POCKETMOD_MEM(ch->paramE1, ch->param); break;
case 0xE2: POCKETMOD_MEM(ch->paramE2, ch->param); break;
case 0xEA: POCKETMOD_MEM(ch->paramEA, ch->param); break;
case 0xEB: POCKETMOD_MEM(ch->paramEB, ch->param); break;
/* 8xx: Set stereo balance (nonstandard) */
case 0x8: {
ch->balance = ch->param;
} break;
/* 9xx: Set sample offset */
case 0x9: {
if (period != 0 || sample != 0) {
ch->param9 = ch->param ? ch->param : ch->param9;
ch->position = ch->param9 << 8;
}
} break;
/* Bxx: Jump to pattern */
case 0xB: {
c->pattern = ch->param < c->length ? ch->param : 0;
c->line = -1;
} break;
/* Cxx: Set volume */
case 0xC: {
ch->volume = _pocketmod_clamp_volume(ch->param);
ch->dirty |= POCKETMOD_VOLUME;
} break;
/* Dxy: Pattern break */
case 0xD: {
pattern_break = (ch->param >> 4) * 10 + (ch->param & 15);
} break;
/* E4x: Set vibrato waveform */
case 0xE4: {
ch->lfo_type[0] = ch->param;
} break;
/* E5x: Set sample finetune */
case 0xE5: {
ch->finetune = ch->param;
ch->dirty |= POCKETMOD_PITCH;
} break;
/* E6x: Pattern loop */
case 0xE6: {
if (ch->param) {
if (!ch->loop_count) {
ch->loop_count = ch->param;
c->line = ch->loop_line;
} else if (--ch->loop_count) {
c->line = ch->loop_line;
}
} else {
ch->loop_line = c->line - 1;
}
} break;
/* E7x: Set tremolo waveform */
case 0xE7: {
ch->lfo_type[1] = ch->param;
} break;
/* E8x: Set stereo balance (nonstandard) */
case 0xE8: {
ch->balance = ch->param << 4;
} break;
/* EEx: Pattern delay */
case 0xEE: {
c->pattern_delay = ch->param;
} break;
/* Fxx: Set speed */
case 0xF: {
if (ch->param != 0) {
if (ch->param < 0x20) {
c->ticks_per_line = ch->param;
} else {
float rate = c->samples_per_second;
c->samples_per_tick = rate / (0.4f * ch->param);
}
}
} break;
default: break;
}
}
/* Pattern breaks are handled here, so that only one jump happens even */
/* when multiple Dxy commands appear on the same line. (You guessed it: */
/* There are songs that rely on this behavior!) */
if (pattern_break != -1) {
c->line = (pattern_break < 64 ? pattern_break : 0) - 1;
if (++c->pattern == c->length) {
c->pattern = c->reset;
}
}
}
static void _pocketmod_next_tick(pocketmod_context *c)
{
int i;
/* Move to the next line if this was the last tick */
if (++c->tick == c->ticks_per_line) {
if (c->pattern_delay > 0) {
c->pattern_delay--;
} else {
_pocketmod_next_line(c);
}
c->tick = 0;
}
/* Make per-tick adjustments for all channels */
for (i = 0; i < c->num_channels; i++) {
_pocketmod_chan *ch = &c->channels[i];
int param = ch->param;
/* Advance the LFO random number generator */
c->lfo_rng = 0x0019660d * c->lfo_rng + 0x3c6ef35f;
/* Handle effects that may happen on any tick of a line */
switch (ch->effect) {
/* 0xy: Arpeggio */
case 0x0: {
ch->dirty |= POCKETMOD_PITCH;
} break;
/* E9x: Retrigger note every x ticks */
case 0xE9: {
if (!(param && c->tick % param)) {
ch->position = 0.0f;
ch->lfo_step = 0;
}
} break;
/* ECx: Cut note after x ticks */
case 0xEC: {
if (c->tick == param) {
ch->volume = 0;
ch->dirty |= POCKETMOD_VOLUME;
}
} break;
/* EDx: Delay note for x ticks */
case 0xED: {
if (c->tick == param && ch->sample) {
ch->dirty |= POCKETMOD_VOLUME | POCKETMOD_PITCH;
ch->period = ch->delayed;
ch->position = 0.0f;
ch->lfo_step = 0;
}
} break;
default: break;
}
/* Handle effects that only happen on the first tick of a line */
if (c->tick == 0) {
switch (ch->effect) {
case 0xE1: _pocketmod_pitch_slide(ch, -ch->paramE1); break;
case 0xE2: _pocketmod_pitch_slide(ch, +ch->paramE2); break;
case 0xEA: _pocketmod_volume_slide(ch, ch->paramEA << 4); break;
case 0xEB: _pocketmod_volume_slide(ch, ch->paramEB & 15); break;
default: break;
}
/* Handle effects that are not applied on the first tick of a line */
} else {
switch (ch->effect) {
/* 1xx: Portamento up */
case 0x1: {
_pocketmod_pitch_slide(ch, -param);
} break;
/* 2xx: Portamento down */
case 0x2: {
_pocketmod_pitch_slide(ch, +param);
} break;
/* 5xy: Volume slide + tone portamento */
case 0x5: {
_pocketmod_volume_slide(ch, param);
} /* Fall through */
/* 3xx: Tone portamento */
case 0x3: {
int rate = ch->param3;
int order = ch->period < ch->target;
int closer = ch->period + (order ? rate : -rate);
int new_order = closer < ch->target;
ch->period = new_order == order ? closer : ch->target;
ch->dirty |= POCKETMOD_PITCH;
} break;
/* 6xy: Volume slide + vibrato */
case 0x6: {
_pocketmod_volume_slide(ch, param);
} /* Fall through */
/* 4xy: Vibrato */
case 0x4: {
ch->lfo_step++;
ch->dirty |= POCKETMOD_PITCH;
} break;
/* 7xy: Tremolo */
case 0x7: {
ch->lfo_step++;
ch->dirty |= POCKETMOD_VOLUME;
} break;
/* Axy: Volume slide */
case 0xA: {
_pocketmod_volume_slide(ch, param);
} break;
default: break;
}
}
/* Update channel volume/pitch if either is out of date */
if (ch->dirty & POCKETMOD_VOLUME) { _pocketmod_update_volume(c, ch); }
if (ch->dirty & POCKETMOD_PITCH) { _pocketmod_update_pitch(c, ch); }
}
}
static void _pocketmod_render_channel(pocketmod_context *c,
_pocketmod_chan *chan,
float *output,
int samples_to_write)
{
/* Gather some loop data */
_pocketmod_sample *sample = &c->samples[chan->sample - 1];
unsigned char *data = POCKETMOD_SAMPLE(c, chan->sample);
const int loop_start = ((data[4] << 8) | data[5]) << 1;
const int loop_length = ((data[6] << 8) | data[7]) << 1;
const int loop_end = loop_length > 2 ? loop_start + loop_length : 0xffffff;
const float sample_end = 1 + _pocketmod_min(loop_end, sample->length);
/* Calculate left/right levels */
const float volume = chan->real_volume / (float) (128 * 64 * 4);
const float level_l = volume * (1.0f - chan->balance / 255.0f);
const float level_r = volume * (0.0f + chan->balance / 255.0f);
/* Write samples */
int i, num;
do {
/* Calculate how many samples we can write in one go */
num = (sample_end - chan->position) / chan->increment;
num = _pocketmod_min(num, samples_to_write);
/* Resample and write 'num' samples */
for (i = 0; i < num; i++) {
int x0 = chan->position;
#ifdef POCKETMOD_NO_INTERPOLATION
float s = sample->data[x0];
#else
int x1 = x0 + 1 - loop_length * (x0 + 1 >= loop_end);
float t = chan->position - x0;
float s = (1.0f - t) * sample->data[x0] + t * sample->data[x1];
#endif
chan->position += chan->increment;
*output++ += level_l * s;
*output++ += level_r * s;
}
/* Rewind the sample when reaching the loop point */
if (chan->position >= loop_end) {
chan->position -= loop_length;
/* Cut the sample if the end is reached */
} else if (chan->position >= sample->length) {
chan->position = -1.0f;
break;
}
samples_to_write -= num;
} while (num > 0);
}
static int _pocketmod_ident(pocketmod_context *c, unsigned char *data, int size)
{
int i, j;
/* 31-instrument files are at least 1084 bytes long */
if (size >= 1084) {
/* The format tag is located at offset 1080 */
unsigned char *tag = data + 1080;
/* List of recognized format tags (possibly incomplete) */
static const struct {
char name[5];
char channels;
} tags[] = {
/* TODO: FLT8 intentionally omitted because I haven't been able */
/* to find a specimen to test its funky pattern pairing format */
{"M.K.", 4}, {"M!K!", 4}, {"FLT4", 4}, {"4CHN", 4},
{"OKTA", 8}, {"OCTA", 8}, {"CD81", 8}, {"FA08", 8},
{"1CHN", 1}, {"2CHN", 2}, {"3CHN", 3}, {"4CHN", 4},
{"5CHN", 5}, {"6CHN", 6}, {"7CHN", 7}, {"8CHN", 8},
{"9CHN", 9}, {"10CH", 10}, {"11CH", 11}, {"12CH", 12},
{"13CH", 13}, {"14CH", 14}, {"15CH", 15}, {"16CH", 16},
{"17CH", 17}, {"18CH", 18}, {"19CH", 19}, {"20CH", 20},
{"21CH", 21}, {"22CH", 22}, {"23CH", 23}, {"24CH", 24},
{"25CH", 25}, {"26CH", 26}, {"27CH", 27}, {"28CH", 28},
{"29CH", 29}, {"30CH", 30}, {"31CH", 31}, {"32CH", 32}
};
/* Check the format tag to determine if this is a 31-sample MOD */
for (i = 0; i < (int) (sizeof(tags) / sizeof(*tags)); i++) {
if (tags[i].name[0] == tag[0] && tags[i].name[1] == tag[1]
&& tags[i].name[2] == tag[2] && tags[i].name[3] == tag[3]) {
c->num_channels = tags[i].channels;
c->length = data[950];
c->reset = data[951];
c->order = &data[952];
c->patterns = &data[1084];
c->num_samples = 31;
return 1;
}
}
}
/* A 15-instrument MOD has to be at least 600 bytes long */
if (size < 600) {
return 0;
}
/* Check that the song title only contains ASCII bytes (or null) */
for (i = 0; i < 20; i++) {
if (data[i] != '\0' && (data[i] < ' ' || data[i] > '~')) {
return 0;
}
}
/* Check that sample names only contain ASCII bytes (or null) */
for (i = 0; i < 15; i++) {
for (j = 0; j < 22; j++) {
char chr = data[20 + i * 30 + j];
if (chr != '\0' && (chr < ' ' || chr > '~')) {
return 0;
}
}
}
/* It looks like we have an older 15-instrument MOD */
c->length = data[470];
c->reset = data[471];
c->order = &data[472];
c->patterns = &data[600];
c->num_samples = 15;
c->num_channels = 4;
return 1;
}
int pocketmod_init(pocketmod_context *c, const void *data, int size, int rate)
{
int i, remaining, header_bytes, pattern_bytes;
unsigned char *byte = (unsigned char*) c;
signed char *sample_data;
/* Check that arguments look more or less sane */
if (!c || !data || rate <= 0 || size <= 0) {
return 0;
}
/* Zero out the whole context and identify the MOD type */
_pocketmod_zero(c, sizeof(pocketmod_context));
c->source = (unsigned char*) data;
if (!_pocketmod_ident(c, c->source, size)) {
return 0;
}
/* Check that we are compiled with support for enough channels */
if (c->num_channels > POCKETMOD_MAX_CHANNELS) {
return 0;
}
/* Check that we have enough sample slots for this file */
if (POCKETMOD_MAX_SAMPLES < 31) {
byte = (unsigned char*) data + 20;
for (i = 0; i < c->num_samples; i++) {
unsigned int length = 2 * ((byte[22] << 8) | byte[23]);
if (i >= POCKETMOD_MAX_SAMPLES && length > 2) {
return 0; /* Can't fit this sample */
}
byte += 30;
}
}
/* Check that the song length is in valid range (1..128) */
if (c->length == 0 || c->length > 128) {
return 0;
}
/* Make sure that the reset pattern doesn't take us out of bounds */
if (c->reset >= c->length) {
c->reset = 0;
}
/* Count how many patterns there are in the file */
c->num_patterns = 0;
for (i = 0; i < 128 && c->order[i] < 128; i++) {
c->num_patterns = _pocketmod_max(c->num_patterns, c->order[i]);
}
pattern_bytes = 256 * c->num_channels * ++c->num_patterns;
header_bytes = (int) ((char*) c->patterns - (char*) data);
/* Check that each pattern in the order is within file bounds */
for (i = 0; i < c->length; i++) {
if (header_bytes + 256 * c->num_channels * c->order[i] > size) {
return 0; /* Reading this pattern would be a buffer over-read! */
}
}
/* Check that the pattern data doesn't extend past the end of the file */
if (header_bytes + pattern_bytes > size) {
return 0;
}
/* Load sample payload data, truncating ones that extend outside the file */
remaining = size - header_bytes - pattern_bytes;
sample_data = (signed char*) data + header_bytes + pattern_bytes;
for (i = 0; i < c->num_samples; i++) {
unsigned char *data = POCKETMOD_SAMPLE(c, i + 1);
unsigned int length = ((data[0] << 8) | data[1]) << 1;
_pocketmod_sample *sample = &c->samples[i];
sample->data = sample_data;
sample->length = _pocketmod_min(length > 2 ? length : 0, remaining);
sample_data += sample->length;
remaining -= sample->length;
}
/* Set up ProTracker default panning for all channels */
for (i = 0; i < c->num_channels; i++) {
c->channels[i].balance = 0x80 + ((((i + 1) >> 1) & 1) ? 0x20 : -0x20);
}
/* Prepare to render from the start */
c->ticks_per_line = 6;
c->samples_per_second = rate;
c->samples_per_tick = rate / 50.0f;
c->lfo_rng = 0xbadc0de;
c->line = -1;
c->tick = c->ticks_per_line - 1;
_pocketmod_next_tick(c);
return 1;
}
int pocketmod_render(pocketmod_context *c, void *buffer, int buffer_size)
{
int i, samples_rendered = 0;
int samples_remaining = buffer_size / POCKETMOD_SAMPLE_SIZE;
if (c && buffer) {
float (*output)[2] = (float(*)[2]) buffer;
while (samples_remaining > 0) {
/* Calculate the number of samples left in this tick */
int num = (int) (c->samples_per_tick - c->sample);
num = _pocketmod_min(num + !num, samples_remaining);
/* Render and mix 'num' samples from each channel */
_pocketmod_zero(output, num * POCKETMOD_SAMPLE_SIZE);
for (i = 0; i < c->num_channels; i++) {
_pocketmod_chan *chan = &c->channels[i];
if (chan->sample != 0 && chan->position >= 0.0f) {
_pocketmod_render_channel(c, chan, *output, num);
}
}
samples_remaining -= num;
samples_rendered += num;
output += num;
/* Advance song position by 'num' samples */
if ((c->sample += num) >= c->samples_per_tick) {
c->sample -= c->samples_per_tick;
_pocketmod_next_tick(c);
/* Stop if a new pattern was reached */
if (c->line == 0 && c->tick == 0) {
/* Increment loop counter as needed */
if (c->visited[c->pattern >> 3] & (1 << (c->pattern & 7))) {
_pocketmod_zero(c->visited, sizeof(c->visited));
c->loop_count++;
}
break;
}
}
}
}
return samples_rendered * POCKETMOD_SAMPLE_SIZE;
}
int pocketmod_loop_count(pocketmod_context *c)
{
return c->loop_count;
}
#endif /* #ifdef POCKETMOD_IMPLEMENTATION */
#ifdef __cplusplus
}
#endif
#endif /* #ifndef POCKETMOD_H_INCLUDED */
/*******************************************************************************
MIT License
Copyright (c) 2018 rombankzero
Permission is hereby granted, free of charge, to any person obtaining a copy
of this software and associated documentation files (the "Software"), to deal
in the Software without restriction, including without limitation the rights
to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
copies of the Software, and to permit persons to whom the Software is
furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in all
copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
SOFTWARE.
*******************************************************************************/

28
source/engine/script.c
View file

@ -211,6 +211,15 @@ void script_eval_w_env(const char *s, JSValue env, const char *file) {
JS_FreeValue(js, v); JS_FreeValue(js, v);
} }
void call_env(JSValue env, const char *eval)
{
if (!JS_IsObject(env)) return;
JSValue fn = JS_GetPropertyStr(js, env, eval);
JSValue v = JS_EvalThis(js, env, eval, strlen(eval), "script", JS_EVAL_FLAGS);
js_print_exception(v);
JS_FreeValue(js,v);
}
void file_eval_env(const char *file, JSValue env) void file_eval_env(const char *file, JSValue env)
{ {
size_t len; size_t len;
@ -222,6 +231,7 @@ void file_eval_env(const char *file, JSValue env)
} }
void script_call_sym(JSValue sym) { void script_call_sym(JSValue sym) {
if (!JS_IsFunction(js, sym)) return;
struct callee c; struct callee c;
c.fn = sym; c.fn = sym;
c.obj = JS_GetGlobalObject(js); c.obj = JS_GetGlobalObject(js);
@ -239,6 +249,9 @@ void out_memusage(const char *file)
JSValue js_callee_exec(struct callee *c, int argc, JSValue *argv) JSValue js_callee_exec(struct callee *c, int argc, JSValue *argv)
{ {
if (JS_IsUndefined(c->fn)) return JS_NULL;
if (JS_IsUndefined(c->obj)) return JS_NULL;
JSValue ret = JS_Call(js, c->fn, c->obj, argc, argv); JSValue ret = JS_Call(js, c->fn, c->obj, argc, argv);
js_print_exception(ret); js_print_exception(ret);
JS_FreeValue(js, ret); JS_FreeValue(js, ret);
@ -271,19 +284,18 @@ void script_callee(struct callee c, int argc, JSValue *argv) {
js_callee_exec(&c, argc, argv); js_callee_exec(&c, argc, argv);
} }
struct callee *make_callee(JSValue fn, JSValue obj) struct callee make_callee(JSValue fn, JSValue obj)
{ {
struct callee *c = malloc(sizeof(*c)); struct callee c;
c->fn = JS_DupValue(js, fn); c.fn = JS_DupValue(js, fn);
c->obj = JS_DupValue(js, obj); c.obj = JS_DupValue(js, obj);
return c; return c;
} }
void free_callee(struct callee *c) void free_callee(struct callee c)
{ {
JS_FreeValue(js,c->fn); JS_FreeValue(js,c.fn);
JS_FreeValue(js,c->obj); JS_FreeValue(js,c.obj);
free(c);
} }
void send_signal(const char *signal, int argc, JSValue *argv) void send_signal(const char *signal, int argc, JSValue *argv)

5
source/engine/script.h
View file

@ -27,8 +27,8 @@ time_t jso_file(const char *file);
JSValue script_runfile(const char *file); JSValue script_runfile(const char *file);
void script_update(double dt); void script_update(double dt);
void script_draw(); void script_draw();
struct callee *make_callee(JSValue fn, JSValue obj); struct callee make_callee(JSValue fn, JSValue obj);
void free_callee(struct callee *c); void free_callee(struct callee c);
void duk_run_err(); void duk_run_err();
void js_dump_stack(); void js_dump_stack();
@ -42,6 +42,7 @@ void call_callee(struct callee *c);
void script_callee(struct callee c, int argc, JSValue *argv); void script_callee(struct callee c, int argc, JSValue *argv);
int script_has_sym(void *sym); int script_has_sym(void *sym);
void script_eval_w_env(const char *s, JSValue env, const char *file); void script_eval_w_env(const char *s, JSValue env, const char *file);
void call_env(JSValue env, const char *eval);
void file_eval_env(const char *file, JSValue env); void file_eval_env(const char *file, JSValue env);
time_t file_mod_secs(const char *file); time_t file_mod_secs(const char *file);

744
source/engine/sound/dsp.c
View file

@ -10,191 +10,164 @@
#define PI 3.14159265 #define PI 3.14159265
struct dsp_filter *filters; dsp_node *masterbus = NULL;
struct dsp_filter make_dsp(void *data, void (*in)(void *data, soundbyte *out, int n)) {
struct dsp_filter new;
new.data = data;
new.filter = in;
return new;
if (arrlen(filters) == 0) {
void interleave(soundbyte *a, soundbyte *b, soundbyte *stereo, int frames)
{
for (int i = 0; i < frames; i++) {
stereo[i*2] = a[i];
stereo[i*2+1] = b[i];
} }
} }
void dsp_run(struct dsp_filter filter, soundbyte *out, int n) { void mono_to_stero(soundbyte *a, soundbyte *stereo, int frames)
filter.dirty = 1; // Always on for testing {
interleave(a,a,stereo, frames);
}
if (!filter.dirty) void mono_expand(soundbyte *buffer, int to, int frames)
{
soundbyte hold[frames];
memcpy(hold, buffer, sizeof(soundbyte)*frames);
for (int i = 0; i < frames; i++)
for (int j = 0; j < to; j++)
buffer[i*to+j] = hold[i];
}
dsp_node *dsp_mixer_node()
{
return make_node(NULL, NULL);
}
void dsp_init()
{
masterbus = dsp_limiter(1.0);
}
soundbyte *dsp_node_out(dsp_node *node)
{
zero_soundbytes(node->cache, BUF_FRAMES*CHANNELS);
if (node->off) return node->cache;
/* Sum all inputs */
for (int i = 0; i < arrlen(node->ins); i++) {
soundbyte *out = dsp_node_out(node->ins[i]);
sum_soundbytes(node->cache, out, BUF_FRAMES*CHANNELS);
}
/* If there's a filter, run it */
if (!node->pass && node->proc)
node->proc(node->data, node->cache, BUF_FRAMES);
scale_soundbytes(node->cache, node->gain, BUF_FRAMES*CHANNELS);
pan_frames(node->cache, node->pan, BUF_FRAMES);
return node->cache;
}
void filter_am_mod(dsp_node *mod, soundbyte *buffer, int frames)
{
soundbyte *m = dsp_node_out(mod);
for (int i = 0; i < frames*CHANNELS; i++) buffer[i] *= m[i];
}
dsp_node *dsp_am_mod(dsp_node *mod)
{
return make_node(mod, filter_am_mod);
}
/* Add b into a */
void sum_soundbytes(soundbyte *a, soundbyte *b, int samples)
{
for (int i = 0; i < samples; i++) a[i] += b[i];
}
void norm_soundbytes(soundbyte *a, float lvl, int samples)
{
float tar = lvl;
float max = 0 ;
for (int i = 0; i < samples; i++) max = (fabsf(a[i] > max) ? fabsf(a[i]) : max);
float mult = max/tar;
scale_soundbytes(a, mult, samples);
}
void scale_soundbytes(soundbyte *a, float scale, int samples)
{
if (scale == 1) return;
for (int i = 0; i < samples; i++) a[i] *= scale;
}
void zero_soundbytes(soundbyte *a, int samples)
{
memset(a, 0, sizeof(soundbyte)*samples);
}
void set_soundbytes(soundbyte *a, soundbyte *b, int samples)
{
zero_soundbytes(a, samples);
sum_soundbytes(a,b,samples);
}
void dsp_node_run(dsp_node *node)
{
zero_soundbytes(node->cache, BUF_FRAMES*CHANNELS);
for (int i = 0; i < arrlen(node->ins); i++) {
soundbyte *out = dsp_node_out(node->ins[i]);
sum_soundbytes(node->cache, out, BUF_FRAMES);
}
}
dsp_node *make_node(void *data, void (*proc)(void *in, soundbyte *out, int samples))
{
dsp_node *self = malloc(sizeof(dsp_node));
memset(self, 0, sizeof(*self));
self->data = data;
self->proc = proc;
self->pass = 0;
self->gain = 1;
return self;
}
void node_free(dsp_node *node)
{
unplug_node(node);
if (node->data)
if (node->data_free) node->data_free(node->data);
else free(node->data);
free(node);
}
void plugin_node(dsp_node *from, dsp_node *to)
{
if (from->out) return;
arrput(to->ins, from);
from->out = to;
}
/* Unplug the given node from its output */
void unplug_node(dsp_node *node)
{
if (!node->out) return;
for (int i = 0; arrlen(node->out->ins); i++)
if (node == node->out->ins[i]) {
arrdelswap(node->out->ins, i);
node->out = NULL;
return; return;
for (int i = 0; i < filter.inputs; i++)
dsp_run(*(filter.in[i]), out, n);
filter.filter(filter.data, out, n);
}
void dsp_filter_addin(struct dsp_filter filter, struct dsp_filter *in)
{
if (filter.inputs > 5) {
YughError("Too many inputs in filter.", 0);
}
filter.in[filter.inputs++] = in;
}
void am_mod(struct dsp_ammod *mod, soundbyte *c, int n)
{
dsp_run(mod->ina, mod->abuf, n);
dsp_run(mod->inb, mod->bbuf, n);
// for (int i = 0; i < n*CHANNELS; i++)
// c[i] = (mod->abuf[i]*mod->bbuf[i])>>15;
}
void fm_mod(float *in1, float *in2, float *out, int n)
{
}
static struct wav make_wav(float freq, int sr, int ch) {
struct wav new;
new.ch = ch;
new.samplerate = sr;
new.frames = sr/freq;
new.data = calloc(new.frames*new.ch, sizeof(soundbyte));
return new;
}
struct wav gen_sine(float amp, float freq, int sr, int ch)
{
struct wav new = make_wav(freq, sr, ch);
if (amp > 1) amp = 1;
if (amp < 0) amp = 0;
soundbyte samp = amp*SHRT_MAX;
soundbyte *data = (soundbyte*)new.data;
for (int i = 0; i < new.frames; i++) {
soundbyte val = samp * sin(2*PI*((float)i / new.frames));
for (int j = 0; j < new.ch; j++) {
data[i*new.ch+j] = val;
} }
} }
YughInfo("Made sine with %i frames.", new.frames); typedef struct {
float amp;
return new; float freq;
} float phase; /* from 0 to 1, marking where we are */
float (*filter)(float phase);
struct wav gen_square(float amp, float freq, int sr, int ch) } phasor;
{
struct wav new = make_wav(freq, sr, ch);
int crossover = new.frames/2;
if (amp > 1) amp = 1;
if (amp < 0) amp = 0;
soundbyte samp = amp * SHRT_MAX;
soundbyte *data = (soundbyte*)new.data;
for (int i = 0; i < new.frames; i++) {
soundbyte val = -2 * floor(2 * i / new.frames) + 1;
for (int j = 0; j < new.ch; j++) {
data[i*new.frames+j] = val;
}
}
return new;
}
struct wav gen_triangle(float amp, float freq, int sr, int ch)
{
struct wav new = make_wav(freq, sr, ch);
if (amp > 1) amp = 1;
if (amp < 0) amp = 0;
soundbyte *data = (soundbyte*)new.data;
for (int i = 0; i < new.frames; i++) {
soundbyte val = 2 * fabsf( (i/new.frames) - floor( (i/new.frames) + 0.5));
for (int j = 0; j < new.ch; j++) {
data[i+j] = val;
}
}
return new;
}
struct wav gen_saw(float amp, float freq, int sr, int ch)
{
struct wav new = make_wav(freq, sr, ch);
if (amp > 1) amp = 1;
if (amp < 0) amp = 0;
soundbyte samp = amp*SHRT_MAX;
soundbyte *data = (soundbyte*)new.data;
for (int i = 0; i < new.frames; i++) {
soundbyte val = samp * 2 * i/sr - samp;
for (int j = 0; j < new.ch; j++) {
data[i+j] = val;
}
}
return new;
}
struct dsp_filter dsp_filter(void *data, void (*filter)(void *data, soundbyte *out, int samples))
{
struct dsp_filter new;
new.data = data;
new.filter = filter;
new.inputs = 0;
new.bus = NULL;
return new;
}
void dsp_rectify(soundbyte *in, soundbyte *out, int n)
{
for (int i = 0; i < n; i++)
out[i] = abs(in[i]);
}
struct phasor phasor_make(unsigned int sr, float freq)
{
struct phasor new;
new.sr = sr;
new.cur = 0.f;
new.freq = freq;
new.cstep = 0;
new.clen = new.sr / new.freq;
new.cache = malloc(new.clen * sizeof(float));
for (int i = 0; i < new.clen; i++) {
new.cache[i] = (float)i / new.clen;
}
return new;
}
float phasor_step(struct phasor *p)
{
p->cur += p->freq/p->sr;
if (p->cur >= 1.f) p->cur = 0.f;
return p->cur;
}
float sin_phasor(float p) float sin_phasor(float p)
{ {
@ -216,47 +189,72 @@ float tri_phasor(float p)
return 4*(p * 0.5f ? p : (1-p)) - 1; return 4*(p * 0.5f ? p : (1-p)) - 1;
} }
void osc_fillbuf(struct osc *osc, soundbyte *buf, int n) void filter_phasor(phasor *p, soundbyte *buffer, int frames)
{ {
for (int i = 0; i < n; i++) { for (int i = 0; i < frames; i++) {
soundbyte val = SHRT_MAX * osc->f(phasor_step(&osc->p)); buffer[i] = p->filter(p->phase) * p->amp;
buf[i*CHANNELS] = buf[i*CHANNELS+1] = val; p->phase += p->freq/SAMPLERATE;
} }
p->phase = p->phase - (int)p->phase;
mono_expand(buffer, CHANNELS, frames);
}
dsp_node *dsp_phasor(float amp, float freq, float (*filter)(float))
{
phasor *p = malloc(sizeof(*p));
p->amp = amp;
p->freq = freq;
p->phase = 0;
p->filter = filter;
return make_node(p, filter_phasor);
}
void filter_rectify(void *data, soundbyte *out, int n)
{
for (int i = 0; i < n; i++) out[i] = abs(out[i]);
}
dsp_node *dsp_rectify()
{
return make_node(NULL, filter_rectify);
}
soundbyte sample_whitenoise()
{
return ((float)rand()/(float)(RAND_MAX/2))-1;
} }
void gen_whitenoise(void *data, soundbyte *out, int n) void gen_whitenoise(void *data, soundbyte *out, int n)
{ {
for (int i = 0; i < n; i++) { for (int i = 0; i < n; i++) out[i] = sample_whitenoise();
for (int j = 0; j < CHANNELS; j++) { mono_expand(out, CHANNELS, n);
out[i*CHANNELS+j] = (rand()>>15) - USHRT_MAX;
}
} }
dsp_node *dsp_whitenoise()
{
return make_node(NULL, gen_whitenoise);
} }
void gen_pinknoise(void *data, soundbyte *out, int n) void gen_pinknoise(void *data, soundbyte *out, int n)
{ {
gen_whitenoise(NULL, out, n); double a[7] = {1.0, 0.0555179, 0.0750759, 0.1538520, 0.3104856, 0.5329522, 0.0168980};
double b[7] = {0.99886, 0.99332, 0.969, 0.8665, 0.55, -0.7616, 0.115926};
double b[2][7] = {0};
double ccof[6] = {0.99886, 0.99332, 0.96900, 0.8550, 0.55000, -0.76160};
double dcof[6] = {0.0555179, 0.0750759, 0.1538520, 0.3104856, 0.5329522, 0.0168960};
for (int i = 0; i < n; i++) { for (int i = 0; i < n; i++) {
for (int j = 0; j < CHANNELS; j++) {
double pink; double pink;
double white = (double)out[i*CHANNELS+j]/SHRT_MAX; double white = sample_whitenoise();
for (int k = 0; k < 5; k++) { for (int k = 0; k < 5; k++) {
b[j][k] = ccof[k]*b[j][k] + white * dcof[k]; b[k] = a[k]*b[k] + white * b[k];
pink += b[j][k]; pink += b[k];
} }
pink += b[j][5] + white*0.5362; pink += b[5] + white*0.5362;
b[j][5] = white*0.115926; b[5] = white*0.115926;
out[i*CHANNELS+j] = pink * SHRT_MAX; out[i] = pink;
}
} }
mono_expand(out,CHANNELS,n);
/* /*
* The above is a loopified version of this * The above is a loopified version of this
* https://www.firstpr.com.au/dsp/pink-noise/ * https://www.firstpr.com.au/dsp/pink-noise/
@ -271,179 +269,71 @@ void gen_pinknoise(void *data, soundbyte *out, int n)
*/ */
} }
soundbyte iir_filter(struct dsp_iir *miir, soundbyte val) dsp_node *dsp_pinknoise()
{ {
struct dsp_iir iir = *miir; return make_node(NULL, gen_pinknoise);
}
float a = 0.f; soundbyte iir_filter(struct dsp_iir iir, soundbyte val)
{
iir.dx[0] = (float)val/SHRT_MAX; iir.y[0] = 0.0;
for (int i = 0; i < iir.n; i++)
a += iir.ccof[i] * iir.dx[i];
for (int i = iir.n-1; i > 0; i--)
iir.dx[i] = iir.dx[i-1];
iir.x[0] = val;
for (int i = 0; i < iir.n; i++) for (int i = 0; i < iir.n; i++)
a -= iir.dcof[i] * iir.dy[i]; iir.y[0] += iir.a[i] * iir.x[i];
iir.dy[0] = a; for (int i = 1; i < iir.n; i++)
iir.y[0] -= iir.b[i] * iir.y[i];
for (int i = iir.n-1; i > 0; i--) /* Shift values in */
iir.dy[i] = iir.dy[i-1]; for (int i = iir.n-1; i > 0; i--) {
iir.x[i] = iir.x[i-1];
iir.y[i] = iir.y[i-1];
return a * SHRT_MAX;
} }
void dsp_iir_fillbuf(struct dsp_iir *iir, soundbyte *out, int n) return iir.y[0];
}
void filter_iir(struct dsp_iir *iir, soundbyte *buffer, int frames)
{ {
dsp_run(iir->in, out, n); for (int i = 0; i < frames; i++) {
soundbyte v = iir_filter(*iir, buffer[i*CHANNELS]);
for (int i = 0; i < n; i++) { for (int j = 0; j < CHANNELS; j++) buffer[i*CHANNELS+j] = v;
soundbyte v = iir_filter(iir, out[i*CHANNELS]);
for (int j = 0; j < CHANNELS; j++) {
out[i*CHANNELS+j] = v;
}
} }
} }
struct dsp_filter lpf_make(int poles, float freq) dsp_node *dsp_lpf(float freq)
{ {
struct dsp_iir *new = malloc(sizeof(*new)); struct dsp_iir *iir = malloc(sizeof(*iir));
(*new) = make_iir(3, 1); *iir = bqlp_dcof(2*freq/SAMPLERATE, 5);
return make_node(iir, filter_iir);
double fcf = new->freq*2/SAMPLERATE;
double sf = sf_bwlp(poles, fcf);
YughInfo("Making LPF filter, fcf: %f, coeffs: %i, scale %1.15lf", fcf, new->n, sf);
int *ccof = ccof_bwlp(new->n);
new->dcof = dcof_bwlp(new->n, fcf);
for (int i = 0; i < new->n; i++)
new->ccof[i] = (float)ccof[i] * sf;
new->dcof[0] = 0.f;
free(ccof);
YughInfo("LPF coefficients are:", 0);
for (int i = 0; i < new->n; i++)
YughInfo("%f, %f", new->ccof[i], new->dcof[i]);
struct dsp_filter lpf;
lpf.data = new;
lpf.filter = dsp_iir_fillbuf;
return lpf;
} }
struct dsp_filter hpf_make(int poles, float freq) dsp_node *dsp_hpf(float freq)
{ {
struct dsp_iir *new = malloc(sizeof(*new)); struct dsp_iir *iir = malloc(sizeof(*iir));
*new = make_iir(3, 1); *iir = bqhp_dcof(2*freq/SAMPLERATE,5);
return make_node(iir, filter_iir);
double fcf = new->freq*2/SAMPLERATE;
double sf = sf_bwhp(new->n, fcf);
int *ccof = ccof_bwhp(new->n);
new->dcof = dcof_bwhp(new->n, fcf);
for (int i = 0; i < new->n; i++)
new->ccof[i] = ccof[i] * sf;
for (int i = 0; i < new->n; i++)
YughInfo("%f, %f", new->ccof[i], new->dcof[i]);
free(ccof);
struct dsp_filter hpf;
hpf.data = new;
hpf.filter = dsp_iir_fillbuf;
return hpf;
} }
soundbyte fir_filter(struct dsp_fir *fir, soundbyte val) void filter_delay(delay *d, soundbyte *buf, int frames)
{ {
float ret = 0.f; for (int i = 0; i < frames*CHANNELS; i++) {
fir->dx[fir->head] = (float)val/SHRT_MAX; buf[i] += ringshift(d->ring)*d->decay;
ringpush(d->ring, buf[i]);
for (int i = 0; i < fir->n; i++) { }
ret += fir->cof[i] * fir->dx[fir->head--];
if (fir->head < 0) fir->head = fir->n-1;
} }
return ret * SHRT_MAX; dsp_node *dsp_delay(double sec, double decay)
}
void dsp_fir_fillbuf(struct dsp_fir *fir, soundbyte *out, int n)
{ {
dsp_run(fir->in, out, n); delay *d = malloc(sizeof(*d));
d->ms_delay = sec;
for (int i = 0; i < n; i++) { d->decay = decay;
soundbyte val = fir_filter(fir, out[i*CHANNELS]); d->ring = NULL;
d->ring = ringnew(d->ring, sec*CHANNELS*SAMPLERATE*2); /* Circular buffer size is enough to have the delay */
for (int j = 0; j < CHANNELS; j++) ringheader(d->ring)->write += CHANNELS*SAMPLERATE*sec;
out[i*CHANNELS + j] = val*5; return make_node(d, filter_delay);
}
}
struct dsp_filter lp_fir_make(float freq)
{
struct dsp_fir fir;
fir.freq = freq;
fir.n = 9;
fir.head = 0;
double fcf = freq * 2 / SAMPLERATE;
fir.dx = calloc(sizeof(float), fir.n);
fir.cof = fir_lp(fir.n, fcf);
struct dsp_filter new;
new.data = malloc(sizeof(fir));
*(struct dsp_fir*)(new.data) = fir;
new.filter = dsp_fir_fillbuf;
for (int i = 0; i < fir.n; i++) {
printf("%f\n", fir.cof[i]);
}
return new;
}
struct dsp_delay dsp_delay_make(unsigned int ms_delay)
{
struct dsp_delay new;
new.ms_delay = ms_delay;
/* Circular buffer size is enough to have the delay */
unsigned int datasize = ms_delay * CHANNELS * (SAMPLERATE / 1000);
// new.buf = circbuf_init(sizeof(soundbyte), datasize);
// new.buf.write = datasize;
// YughInfo("Buffer size is %u.", new.buf.len);
return new;
}
void dsp_delay_filbuf(struct dsp_delay *delay, soundbyte *buf, int n)
{
soundbyte cache[BUF_FRAMES*2];
dsp_run(delay->in, cache, n);
for (int i = 0; i < n*CHANNELS; i++) {
// cbuf_push(&delay->buf, cache[i] / 2);
// buf[i] = cache[i] + cbuf_shift(&delay->buf);
}
} }
/* Get decay constant for a given pole */ /* Get decay constant for a given pole */
@ -501,9 +391,9 @@ void dsp_adsr_fillbuf(struct dsp_adsr *adsr, soundbyte *out, int n)
struct dsp_filter make_adsr(unsigned int atk, unsigned int dec, unsigned int sus, unsigned int rls) dsp_node *dsp_adsr(unsigned int atk, unsigned int dec, unsigned int sus, unsigned int rls)
{ {
struct dsp_adsr *adsr = calloc(sizeof(*adsr), 1); struct dsp_adsr *adsr = malloc(sizeof(*adsr));
adsr->atk = atk; adsr->atk = atk;
/* decay to 3 tau */ /* decay to 3 tau */
adsr->atk_t = tau2pole(atk / 3000.f); adsr->atk_t = tau2pole(atk / 3000.f);
@ -517,22 +407,57 @@ struct dsp_filter make_adsr(unsigned int atk, unsigned int dec, unsigned int sus
adsr->rls = rls + adsr->sus; adsr->rls = rls + adsr->sus;
adsr->rls_t = tau2pole(rls / 3000.f); adsr->rls_t = tau2pole(rls / 3000.f);
return make_dsp(adsr, dsp_adsr_fillbuf); return make_node(adsr, dsp_adsr_fillbuf);
} }
struct dsp_filter make_reverb() void filter_noise_gate(float *floor, soundbyte *out, int frames)
{ {
for (int i = 0; i < frames*CHANNELS; i++) out[i] = fabsf(out[i]) < *floor ? 0.0 : out[i];
} }
void dsp_reverb_fillbuf(struct dsp_reverb *r, soundbyte *out, int n) dsp_node *dsp_noise_gate(float floor)
{ {
float *v = malloc(sizeof(float));
*v = floor;
return make_node(v, filter_noise_gate);
} }
struct dsp_filter dsp_make_compressor() void filter_limiter(float *ceil, soundbyte *out, int n)
{
for (int i = 0; i < n*CHANNELS; i++) out[i] = fabsf(out[i]) > *ceil ? *ceil : out[i];
}
dsp_node *dsp_limiter(float ceil)
{
float *v = malloc(sizeof(float));
*v = ceil;
return make_node(v, filter_limiter);
}
void dsp_compressor_fillbuf(struct dsp_compressor *comp, soundbyte *out, int n)
{
float val;
float db;
db = comp->target * (val - comp->threshold) / comp->ratio;
for (int i = 0; i < n; i++) {
val = float2db(out[i*CHANNELS]);
if (val < comp->threshold) {
comp->target = comp->rls_tau * comp->target;
val += db;
} else {
comp->target = (1 - comp->atk_tau) + comp->atk_tau * comp->target; // TODO: Bake in the 1 - atk_tau
val -= db;
}
// Apply same compression to both channels
out[i*CHANNELS] = out[i*CHANNELS+1] = db2float(val) * ( out[i*CHANNELS] > 0 ? 1 : -1);
}
}
dsp_node *dsp_compressor()
{ {
struct dsp_filter filter;
struct dsp_compressor new; struct dsp_compressor new;
new.ratio = 4000; new.ratio = 4000;
@ -545,73 +470,37 @@ struct dsp_filter dsp_make_compressor()
struct dsp_compressor *c = malloc(sizeof(*c)); struct dsp_compressor *c = malloc(sizeof(*c));
*c = new; *c = new;
return make_node(c, dsp_compressor_fillbuf);
filter.data = c;
filter.filter = dsp_compressor_fillbuf;
return filter;
} }
void dsp_compressor_fillbuf(struct dsp_compressor *comp, soundbyte *out, int n) /* Assumes 2 channels in a frame */
void pan_frames(soundbyte *out, float deg, int frames)
{ {
float val; if (deg == 0.f) return;
float db; if (deg < -100) deg = -100.f;
db = comp->target * (val - comp->threshold) / comp->ratio; else if (deg > 100) deg = 100.f;
for (int i = 0; i < n; i++) {
val = short2db(out[i*CHANNELS]);
if (val < comp->threshold) {
comp->target = comp->rls_tau * comp->target;
val += db;
} else {
comp->target = (1 - comp->atk_tau) + comp->atk_tau * comp->target; // TODO: Bake in the 1 - atk_tau
val -= db;
}
// Apply same compression to both channels
out[i*CHANNELS] = out[i*CHANNELS+1] = db2short(val) * ( out[i*CHANNELS] > 0 ? 1 : -1);
}
}
void dsp_pan(float *deg, soundbyte *out, int n)
{
if (*deg < -100) *deg = -100.f;
else if (*deg > 100) *deg = 100.f;
if (*deg == 0.f) return;
float db1, db2; float db1, db2;
float pct = *deg / 100.f; float pct = deg / 100.f;
if (*deg > 0) { if (deg > 0) {
db1 = pct2db(1 - pct); db1 = pct2db(1 - pct);
db2 = pct2db(pct); db2 = pct2db(pct);
for (int i = 0; i < frames; i++) {
soundbyte L = out[i*2];
soundbyte R = out[i*2+1];
out[i*2] = fgain(L, db1);
out[i*2+1] = (R + fgain(L, db2))/2;
}
} else { } else {
db1 = pct2db(1 + pct); db1 = pct2db(1 + pct);
db2 = pct2db(-1*pct); db2 = pct2db(-1*pct);
for (int i = 0; i < frames; i++) {
soundbyte L = out[i*2];
soundbyte R = out[i*2+1];
out[i*2+1] = fgain(R,db1);
out[i*2] = fgain(L, db1) + fgain(R, db2);
} }
for (int i = 0; i < n; i++) {
double pct = *deg / 100.f;
soundbyte L = out[i*CHANNELS];
soundbyte R = out[i*CHANNELS +1];
if (*deg > 0) {
out[i*CHANNELS] = short_gain(L, db1);
out[i*CHANNELS+1] = (R + short_gain(L, db2)) / 2;
continue;
}
out[i*CHANNELS+1] = short_gain(R, db1);
out[i*CHANNELS] = short_gain(L, db1) + short_gain(R, db2);
} }
} }
@ -625,12 +514,35 @@ void dsp_mono(void *p, soundbyte *out, int n)
} }
} }
void dsp_bitcrush(void *p, soundbyte *out, int n) struct bitcrush {
float sr;
float depth;
};
#define ROUND(f) ((float)((f>0.0)?floor(f+0.5):ceil(f-0.5)))
void filter_bitcrush(struct bitcrush *b, soundbyte *out, int frames)
{ {
int max = pow(2,b->depth) - 1;
int step = SAMPLERATE/b->sr;
// for (int i = 0; i < n; i++) { int i = 0;
// for (int j = 0; j < CHANNELS; j++) while (i < frames) {
// out[i*CHANNELS+j] = (out[i*CHANNELS+j] | 0xFF); /* Mask out the lower 8 bits */ float left = ROUND((out[0]+1.0)*max)/(max-1.0);
// } float right = ROUND((out[1]+1.0)*max)/(max-1.0);
for (int j = 0; j < step && i < frames; j++) {
out[0] = left;
out[1] = right;
out += CHANNELS;
i++;
}
}
}
dsp_node *dsp_bitcrush(float sr, float res)
{
struct bitcrush *b = malloc(sizeof(*b));
b->sr = sr;
b->depth = res;
return make_node(b, filter_bitcrush);
} }

151
source/engine/sound/dsp.h
View file

@ -6,47 +6,46 @@
#define CHANNELS 2 #define CHANNELS 2
#include "sound.h" #include "sound.h"
#include "cbuf.h"
#include "script.h"
//#include "circbuf.h" /* a DSP node, when processed, sums its inputs, and stores the result of proc in its cache */
typedef struct dsp_node {
void (*proc)(void *dsp, soundbyte *buf, int samples); /* processor */
void *data; /* Node specific data to use in the proc function, passed in as dsp */
void (*data_free)(void *data);
soundbyte cache[BUF_FRAMES*CHANNELS]; /* Cached process */
struct dsp_node **ins; /* Array of in nodes */
struct dsp_node *out; /* node this one is connected to */
int pass; /* True if the filter should be bypassed */
int off; /* True if the filter shouldn't output */
float gain; /* Between 0 and 1, to attenuate this output */
float pan; /* Between -100 and +100, panning left to right in the speakers */
} dsp_node;
struct dsp_iir; void dsp_init();
/* Get the output of a node */
soundbyte *dsp_node_out(dsp_node *node);
void dsp_node_run(dsp_node *node);
dsp_node *make_node(void *data, void (*proc)(void *data, soundbyte *out, int samples));
void plugin_node(dsp_node *from, dsp_node *to);
void unplug_node(dsp_node *node);
void node_free(dsp_node *node);
void dsp_rectify(soundbyte *in, soundbyte *out, int n); void scale_soundbytes(soundbyte *a, float scale, int frames);
void sum_soundbytes(soundbyte *a, soundbyte *b, int frames);
void zero_soundbytes(soundbyte *a, int frames);
void set_soundbytes(soundbyte *a, soundbyte *b, int frames);
struct dsp_filter { dsp_node *dsp_mixer_node();
void (*filter)(void *data, soundbyte *out, int samples); dsp_node *dsp_am_mod(dsp_node *mod);
void *data; dsp_node *dsp_rectify();
int inputs; extern dsp_node *masterbus;
struct dsp_filter *in[6];
struct bus *bus;
soundbyte cache[CHANNELS*BUF_FRAMES]; dsp_node *dsp_hpf(float freq);
int dirty; dsp_node *dsp_lpf(float freq);
};
struct dsp_filter dsp_filter(void *data, void (*filter)(void *data, soundbyte *out, int samples));
struct dsp_fir {
float freq;
int n;
int head;
float *cof;
float *dx;
struct dsp_filter in;
};
void dsp_filter_addin(struct dsp_filter filter, struct dsp_filter *in);
struct dsp_filter lp_fir_make(float freq);
void dsp_iir_fillbuf(struct dsp_iir *iir, soundbyte *out, int n);
struct dsp_filter hpf_make(int poles, float freq);
struct dsp_filter lpf_make(int poles, float freq);
struct dsp_filter bpf_make(int poles, float freq1, float freq2);
struct dsp_filter npf_make(int poles, float freq1, float freq2);
/* atk, dec, sus, rls specify the time, in miliseconds, the phase begins */ /* atk, dec, sus, rls specify the time, in miliseconds, the phase begins */
struct dsp_adsr { struct dsp_adsr {
@ -63,24 +62,15 @@ struct dsp_adsr {
float out; float out;
}; };
void dsp_adsr_fillbuf(struct dsp_adsr *adsr, soundbyte *out, int n); dsp_node *dsp_adsr(unsigned int atk, unsigned int dec, unsigned int sus, unsigned int rls);
struct dsp_filter make_adsr(unsigned int atk, unsigned int dec, unsigned int sus, unsigned int rls);
struct dsp_delay { typedef struct {
unsigned int ms_delay; unsigned int ms_delay;
// struct circbuf buf; float decay; /* Each echo should be multiplied by this number */
struct dsp_filter in; soundbyte *ring;
}; } delay;
struct dsp_delay dsp_delay_make(unsigned int ms_delay); dsp_node *dsp_delay(double sec, double decay);
void dsp_delay_filbuf(struct dsp_delay *delay, soundbyte *buf, int n);
struct dsp_ammod {
struct dsp_filter ina;
struct dsp_filter inb;
soundbyte abuf[BUF_FRAMES*CHANNELS];
soundbyte bbuf[BUF_FRAMES*CHANNELS];
};
struct dsp_compressor { struct dsp_compressor {
double ratio; double ratio;
@ -92,69 +82,26 @@ struct dsp_compressor {
double rls_tau; double rls_tau;
}; };
struct dsp_filter dsp_make_compressor(); dsp_node *dsp_compressor();
void dsp_compressor_fillbuf(struct dsp_compressor *comp, soundbyte *out, int n);
struct dsp_limiter { dsp_node *dsp_limiter(float ceil);
dsp_node *dsp_noise_gate(float floor);
};
struct dsp_filter dsp_make_limiter();
void dsp_limiter_fillbuf(struct dsp_limiter *lim, soundbyte *out, int n);
struct phasor {
unsigned int sr;
float cur;
float freq;
unsigned int clen;
unsigned int cstep;
float *cache;
};
struct phasor phasor_make(unsigned int sr, float freq); struct phasor phasor_make(unsigned int sr, float freq);
struct osc { dsp_node *dsp_whitenoise();
float (*f)(float p); dsp_node *dsp_pinknoise();
struct phasor p;
unsigned int frames;
unsigned int frame;
soundbyte *cache;
};
struct wav;
struct wav gen_sine(float amp, float freq, int sr, int ch);
struct wav gen_square(float amp, float freq, int sr, int ch);
struct wav gen_triangle(float amp, float freq, int sr, int ch);
struct wav gen_saw(float amp, float freq, int sr, int ch);
void gen_whitenoise(void *data, soundbyte *out, int n);
void gen_pinknoise(void *data, soundbyte *out, int n);
float sin_phasor(float p); float sin_phasor(float p);
float square_phasor(float p); float square_phasor(float p);
float saw_phasor(float p); float saw_phasor(float p);
float tri_phasor(float p); float tri_phasor(float p);
void osc_fillbuf(struct osc *osc, soundbyte *buf, int n); dsp_node *dsp_reverb();
dsp_node *dsp_sinewave(float amp, float freq);
void am_mod(struct dsp_ammod *mod, soundbyte *c, int n); dsp_node *dsp_square(float amp, float freq, int sr, int ch);
dsp_node *dsp_bitcrush(float sr, float res);
struct dsp_reverb {
unsigned int time; /* Time in miliseconds for the sound to decay out */
};
struct dsp_filter make_reverb();
void dsp_reverb_fillbuf(struct dsp_reverb *r, soundbyte *out, int n);
void dsp_pan(float *deg, soundbyte *out, int n);
void dsp_mono(void *p, soundbyte *out, int n); void dsp_mono(void *p, soundbyte *out, int n);
void pan_frames(soundbyte *out, float deg, int frames);
void dsp_bitcrush(void *p, soundbyte *out, int n);
void dsp_run(struct dsp_filter filter, soundbyte *out, int n);
#endif #endif

175
source/engine/sound/iir.c
View file

@ -550,7 +550,6 @@ double sf_bwbp( int n, double f1f, double f2f )
sf_bwbs - calculates the scaling factor for a butterworth bandstop filter. sf_bwbs - calculates the scaling factor for a butterworth bandstop filter.
The scaling factor is what the c coefficients must be multiplied by so The scaling factor is what the c coefficients must be multiplied by so
that the filter response has a maximum value of 1. that the filter response has a maximum value of 1.
*/ */
double sf_bwbs( int n, double f1f, double f2f ) double sf_bwbs( int n, double f1f, double f2f )
@ -582,9 +581,6 @@ double sf_bwbs( int n, double f1f, double f2f )
return( 1.0 / sfr ); return( 1.0 / sfr );
} }
float *fir_lp(int n, double fcf) float *fir_lp(int n, double fcf)
{ {
float *ret = malloc(n * sizeof(*ret)); float *ret = malloc(n * sizeof(*ret));
@ -640,43 +636,32 @@ float *fir_bpf(int n, double fcf1, double fcf2)
return ret; return ret;
} }
/* Biquad filters */ /* Biquad filters */
struct dsp_iir make_iir(int order)
struct dsp_iir make_iir(int cofs, int order)
{ {
struct dsp_iir new; struct dsp_iir new;
new.n = cofs; new.n = order+1;
new.order = order; new.a = calloc(sizeof(float), new.n);
new.b = calloc(sizeof(float), new.n);
new.dcof = calloc(sizeof(float), cofs *order); new.x = calloc(sizeof(float), new.n);
new.ccof = calloc(sizeof(float), cofs *order); new.y = calloc(sizeof(float), new.n);
new.dx = calloc(sizeof(float), cofs *order);
new.dy = calloc(sizeof(float), cofs *order);
return new; return new;
} }
struct dsp_iir biquad_iir() struct dsp_iir biquad_iir()
{ {
return make_iir(3, 1); return make_iir(2);
} }
void biquad_iir_fill(struct dsp_iir bq, double *a, double *b) void biquad_iir_fill(struct dsp_iir bq, double *a, double *b)
{ {
bq.ccof[0] = (b[0] / a[0]); bq.a[0] = (b[0] / a[0]);
bq.ccof[1] = (b[1] / a[0]); bq.a[1] = (b[1] / a[0]);
bq.ccof[2] = (b[2] / a[0]); bq.a[2] = (b[2] / a[0]);
bq.dcof[0] = 0.f; bq.b[0] = 0.f;
bq.dcof[1] = (a[1] / a[0]); bq.b[1] = (a[1] / a[0]);
bq.dcof[2] = (a[2] / a[0]); bq.b[2] = (a[2] / a[0]);
} }
struct dsp_iir bqlp_dcof(double fcf, float Q) struct dsp_iir bqlp_dcof(double fcf, float Q)
@ -687,7 +672,7 @@ struct dsp_iir bqlp_dcof(double fcf, float Q)
double b[3]; double b[3];
double az = sin(w0) / (2 * Q); double az = sin(w0) / (2 * Q);
b[0] = (1 - cos(w0)) / 2; b[0] = (1 - cos(w0)) / 2.0;
b[1] = 1 - cos(w0); b[1] = 1 - cos(w0);
b[2] = b[0]; b[2] = b[0];
@ -841,10 +826,23 @@ struct dsp_iir bqhs_dcof(double fcf, float Q, float dbgain)
return new; return new;
} }
/* Bipole Butterworth, Critically damped, and Bessel */ /* Bipole Butterworth, Critically damped, and Bessel */
/* https://unicorn.us.com/trading/allpolefilters.html */ /* https://unicorn.us.com/trading/allpolefilters.html */
/*
struct p2_iir {
int order;
int n;
float *a;
float *b;
float *x;
float *y;
}
soundbyte p2_calc(struct p2_iir iir, soundbyte val)
{
for (int i = 0; i < iir.order; i++) {
}
}
void p2_ccalc(double fcf, double p, double g, double *a, double *b) void p2_ccalc(double fcf, double p, double g, double *a, double *b)
{ {
@ -867,7 +865,7 @@ struct dsp_iir p2_bwlp(double fcf)
double g = 1; double g = 1;
struct dsp_iir new = biquad_iir(); struct dsp_iir new = biquad_iir();
p2_ccalc(fcf, p, g, new.ccof, new.dcof); p2_ccalc(fcf, p, g, new.a, new.b);
return new; return new;
} }
@ -875,8 +873,8 @@ struct dsp_iir p2_bwlp(double fcf)
struct dsp_iir p2_bwhp(double fcf) struct dsp_iir p2_bwhp(double fcf)
{ {
struct dsp_iir new = p2_bwlp(fcf); struct dsp_iir new = p2_bwlp(fcf);
new.ccof[1] *= -1; new.a[1] *= -1;
new.dcof[1] *= -1; new.b[1] *= -1;
return new; return new;
} }
@ -887,7 +885,7 @@ struct dsp_iir p2_cdlp(double fcf)
double p = 2; double p = 2;
struct dsp_iir new = biquad_iir(); struct dsp_iir new = biquad_iir();
p2_ccalc(fcf, p, g, new.ccof, new.dcof); p2_ccalc(fcf, p, g, new.a, new.b);
return new; return new;
} }
@ -895,8 +893,8 @@ struct dsp_iir p2_cdlp(double fcf)
struct dsp_iir p2_cdhp(double fcf) struct dsp_iir p2_cdhp(double fcf)
{ {
struct dsp_iir new = p2_cdlp(fcf); struct dsp_iir new = p2_cdlp(fcf);
new.ccof[1] *= -1; new.a[1] *= -1;
new.dcof[1] *= -1; new.b[1] *= -1;
return new; return new;
} }
@ -907,7 +905,7 @@ struct dsp_iir p2_beslp(double fcf)
double p = 3; double p = 3;
struct dsp_iir new = biquad_iir(); struct dsp_iir new = biquad_iir();
p2_ccalc(fcf, p, g, new.ccof, new.dcof); p2_ccalc(fcf, p, g, new.a, new.b);
return new; return new;
} }
@ -915,56 +913,12 @@ struct dsp_iir p2_beslp(double fcf)
struct dsp_iir p2_beshp(double fcf) struct dsp_iir p2_beshp(double fcf)
{ {
struct dsp_iir new = p2_beslp(fcf); struct dsp_iir new = p2_beslp(fcf);
new.ccof[1] *= -1; new.a[1] *= -1;
new.dcof[1] *= -1; new.b[1] *= -1;
return new; return new;
} }
struct dsp_iir p2_iir_order(int order)
{
struct dsp_iir new;
new.n = 3;
new.order = order;
new.ccof = calloc(sizeof(float), 3 * order);
new.dcof = calloc(sizeof(float), 3 * order);
new.dx = calloc(sizeof(float), 3 * order);
new.dy = calloc(sizeof(float), 3 * order);
return new;
}
short p2_filter(struct dsp_iir iir, short val)
{
float a = (float)val/SHRT_MAX;
for (int i = 0; i < iir.order; i++) {
int indx = i * iir.n;
iir.dx[indx] = a;
a = 0.f;
for (int j = 0; j < iir.n; j++)
a += iir.ccof[indx + j] * iir.dx[indx];
for (int j = iir.n-1; j > 0; j--)
iir.dx[indx] = iir.dx[indx-1];
for (int j = 0; j < iir.n; j++)
a -= iir.dcof[indx+j] * iir.dy[indx];
iir.dy[indx] = a;
for (int j = iir.n-1; j > 0; j--)
iir.dy[indx] = iir.dy[indx-1];
}
return a * SHRT_MAX;
}
struct dsp_iir che_lp(int order, double fcf, double e) struct dsp_iir che_lp(int order, double fcf, double e)
{ {
struct dsp_iir new = p2_iir_order(order); struct dsp_iir new = p2_iir_order(order);
@ -986,13 +940,13 @@ struct dsp_iir che_lp(int order, double fcf, double e)
s = a2*c + 2.f*a*b + 1.f; s = a2*c + 2.f*a*b + 1.f;
double A = a2/(4.f); double A = a2/(4.f);
new.ccof[0*i] = ep * 1.f/A; new.a[0*i] = ep * 1.f/A;
new.ccof[1*i] = ep * -2.f/A; new.a[1*i] = ep * -2.f/A;
new.ccof[2*i] = ep * 1.f/A; new.a[2*i] = ep * 1.f/A;
new.dcof[0*i] = ep * 0.f; new.b[0*i] = ep * 0.f;
new.dcof[1*i] = ep * 2.f*(1-a2*c); new.b[1*i] = ep * 2.f*(1-a2*c);
new.dcof[2*i] = ep * -(a2*c - 2.f*a*b + 1.f); new.b[2*i] = ep * -(a2*c - 2.f*a*b + 1.f);
} }
return new; return new;
@ -1018,9 +972,9 @@ struct dsp_iir che_hp(int order, double fcf, double e)
s = a2*c + 2.f*a*b + 1.f; s = a2*c + 2.f*a*b + 1.f;
double A = 1.f/(4.f); double A = 1.f/(4.f);
new.ccof[0*i] = ep * 1.f/A; new.a[0*i] = ep * 1.f/A;
new.ccof[1*i] = ep * -2.f/A; new.a[1*i] = ep * -2.f/A;
new.ccof[2*i] = ep * 1.f/A; new.a[2*i] = ep * 1.f/A;
} }
@ -1037,7 +991,7 @@ struct dsp_iir che_bp(int order, double s, double fcf1, double fcf2, double e)
double ep = 2.f/e; double ep = 2.f/e;
int n = order / 4; int n = order / 4;
struct dsp_iir new = p2_iir_order(order); struct dsp_iir new = biquad_iir();
double a = cos(M_PI*(fcf1+fcf2)/2) / cos(M_PI*(fcf2-fcf1)/s); double a = cos(M_PI*(fcf1+fcf2)/2) / cos(M_PI*(fcf2-fcf1)/s);
double a2 = pow(a, 2); double a2 = pow(a, 2);
@ -1055,15 +1009,15 @@ struct dsp_iir che_bp(int order, double s, double fcf1, double fcf2, double e)
c = pow(r, 2) + pow(c, 2); c = pow(r, 2) + pow(c, 2);
s = b2*c + 2.f*b*r + 1.f; s = b2*c + 2.f*b*r + 1.f;
new.ccof[0*i] = ep * 1.f/A; new.a[0*i] = ep * 1.f/A;
new.ccof[1*i] = ep * -2.f/A; new.a[1*i] = ep * -2.f/A;
new.ccof[2*i] = ep * 1.f/A; new.a[2*i] = ep * 1.f/A;
new.dcof[0*i] = 0.f; new.b[0*i] = 0.f;
new.dcof[1*i] = ep * 4.f*a*(1.f+b*r)/s; new.b[1*i] = ep * 4.f*a*(1.f+b*r)/s;
new.dcof[2*i] = ep * 2.f*(b2*c-2.f*a2-1.f)/s; new.b[2*i] = ep * 2.f*(b2*c-2.f*a2-1.f)/s;
new.dcof[3*i] = ep * 4.f*a*(1.f-b*r)/s; new.b[3*i] = ep * 4.f*a*(1.f-b*r)/s;
new.dcof[4*i] = ep * -(b2*c - 2.f*b*r + 1.f) / s; new.b[4*i] = ep * -(b2*c - 2.f*b*r + 1.f) / s;
} }
return new; return new;
@ -1096,16 +1050,17 @@ double ep = 2.f/e;
c = pow(r, 2) + pow(c, 2); c = pow(r, 2) + pow(c, 2);
s = b2*c + 2.f*b*r + 1.f; s = b2*c + 2.f*b*r + 1.f;
new.ccof[0*i] = ep * 1.f/A; new.a[0*i] = ep * 1.f/A;
new.ccof[1*i] = ep * -2.f/A; new.a[1*i] = ep * -2.f/A;
new.ccof[2*i] = ep * 1.f/A; new.a[2*i] = ep * 1.f/A;
new.dcof[0*i] = 0.f; new.b[0*i] = 0.f;
new.dcof[1*i] = ep * 4.f*a*(c+b*r)/s; new.b[1*i] = ep * 4.f*a*(c+b*r)/s;
new.dcof[2*i] = ep * 2.f*(b2-2.f*a2*c-c)/s; new.b[2*i] = ep * 2.f*(b2-2.f*a2*c-c)/s;
new.dcof[3*i] = ep * 4.f*a*(c-b*r)/s; new.b[3*i] = ep * 4.f*a*(c-b*r)/s;
new.dcof[4*i] = ep * -(b2 - 2.f*b*r + c) / s; new.b[4*i] = ep * -(b2 - 2.f*b*r + c) / s;
} }
return new; return new;
} }
*/

16
source/engine/sound/iir.h
View file

@ -1,21 +1,17 @@
#ifndef IIR_H #ifndef IIR_H
#define IIR_H #define IIR_H
#include "dsp.h" #include "sound.h"
struct dsp_iir { struct dsp_iir {
float freq;
int n; // Amount of constants int n; // Amount of constants
int order; // How many times it's applied float *a;
float *ccof; float *b;
float *dcof; float *x;
float *dx; float *y;
float *dy;
struct dsp_filter in;
}; };
struct dsp_iir make_iir(int cofs, int order); struct dsp_iir make_iir(int order);
double *binomial_mult( int n, double *p ); double *binomial_mult( int n, double *p );
double *trinomial_mult( int n, double *b, double *c ); double *trinomial_mult( int n, double *b, double *c );

117
source/engine/sound/mix.c
View file

@ -1,117 +0,0 @@
#include "mix.h"
#include "stddef.h"
#include "time.h"
#include "sound.h"
#include "dsp.h"
#include <string.h>
#include "log.h"
#include "stdlib.h"
#include <assert.h>
#define BUS_N 256
static struct bus *bus;
static int first = 0; /* First bus available */
static int first_on = -1; /* First bus to fill buffer with */
soundbyte *mastermix = NULL;
static float master_volume = 1.f;
void mix_master_vol(float v) {
if (v < 0.f) v = 0.f;
if (v > 100.f) v = 100.f;
master_volume = v / 100.f;
}
void mixer_init() {
bus = malloc(sizeof(struct bus)*BUS_N);
mastermix = malloc(BUF_FRAMES*CHANNELS);
for (int i = 0; i < BUS_N; i++) {
bus[i].next = i+1;
bus[i].on = 0;
bus[i].id = i;
}
bus[BUS_N-1].next = -1;
}
void filter_to_bus(struct dsp_filter *f)
{
struct bus *b = first_free_bus(*f);
if (b)
f->bus = b;
}
void unplug_filter(struct dsp_filter *f)
{
if (!f->bus) return;
bus_free(f->bus);
f->bus = NULL;
}
struct bus *first_free_bus(struct dsp_filter in) {
for (int i = 0; i < 255; i++)
if (!bus[i].on) {
bus[i].on = 1;
bus[i].in = in;
return &bus[i];
}
return NULL;
if (first == -1) return NULL;
int ret = first;
first = bus[ret].next;
bus[ret].on = 1;
bus[ret].in = in;
if (first_on != -1) bus[first_on].prev = ret;
bus[ret].next = first_on;
bus[ret].prev = -1;
first_on = ret;
return &bus[ret];
}
void bus_free(struct bus *b)
{
if (!b) return;
b->on = 0;
return;
if (first_on == b->id) first_on = b->next;
if (b->next != -1) bus[b->next].prev = b->prev;
if (b->prev != -1) bus[b->prev].next = b->next;
b->next = first;
first = b->id;
b->on = 0;
}
void bus_fill_buffers(soundbyte *master, int n) {
int curbus = first_on;
// if (curbus == -1) return;
memset(master, 0, BUF_FRAMES*CHANNELS*sizeof(soundbyte));
for (int i = 0; i < 255; i++) {
if (!bus[i].on) continue;
dsp_run(bus[i].in, bus[i].buf, BUF_FRAMES);
for (int j = 0; j < BUF_FRAMES*CHANNELS; j++)
master[j] += bus[i].buf[j] * master_volume;
}
return;
while (curbus != -1) {
int nextbus = bus[curbus].next; /* Save this in case busses get changed during fill */
dsp_run(bus[curbus].in, bus[curbus].buf, BUF_FRAMES);
for (int i = 0; i < BUF_FRAMES*CHANNELS; i++)
master[i] += bus[curbus].buf[i] * master_volume;
curbus = nextbus;
}
}

31
source/engine/sound/mix.h
View file

@ -1,31 +0,0 @@
#ifndef MIX_H
#define MIX_H
#include "dsp.h"
#include "sound.h"
struct bus {
struct dsp_filter in;
soundbyte buf[BUF_FRAMES*CHANNELS];
float gain;
int on;
int next; /* Next available bus */
int prev;
int id;
};
extern soundbyte *mastermix;
void mixer_init();
struct bus *first_free_bus(struct dsp_filter in);
void filter_to_bus(struct dsp_filter *f);
void unplug_filter(struct dsp_filter *f);
void bus_fill_buffers(soundbyte *master, int n);
/* Set volume between 0 and 100% */
void mix_master_vol(float v);
void bus_free(struct bus *bus);
#endif

71
source/engine/sound/music.c
View file

@ -3,22 +3,22 @@
#include "dsp.h" #include "dsp.h"
#include "tsf.h" #include "tsf.h"
#include "tml.h" #include "tml.h"
#include "mix.h"
#include "sound.h" #include "sound.h"
#include "log.h" #include "log.h"
#include "resources.h" #include "resources.h"
#include <stdlib.h> #include <stdlib.h>
#include "stb_ds.h"
#define TSF_BLOCK 32 #define TSF_BLOCK 32
struct dsp_midi_song gsong; static struct {
struct dsp_filter songfil; char *key;
tsf **value;
float music_pan = 0.f; } *sf_hash = NULL;
void dsp_midi_fillbuf(struct dsp_midi_song *song, void *out, int n) void dsp_midi_fillbuf(struct dsp_midi_song *song, void *out, int n)
{ {
soundbyte *o = (soundbyte*)out; soundbyte *o = out;
tml_message *midi = song->midi; tml_message *midi = song->midi;
for (int i = 0; i < n; i += TSF_BLOCK) { for (int i = 0; i < n; i += TSF_BLOCK) {
@ -55,44 +55,37 @@ void dsp_midi_fillbuf(struct dsp_midi_song *song, void *out, int n)
song->midi = midi; song->midi = midi;
} }
struct bus *musicbus; tsf *make_soundfont(const char *path)
{
int idx = shgeti(sf_hash, path);
if (idx != -1) return sf_hash[idx].value;
void play_song(const char *midi, const char *sf) long rawlen;
void *raw = slurp_file(path, &rawlen);
tsf *sf = tsf_load_memory(raw,rawlen);
free(raw);
if (!sf) { YughWarn("Soundfont %s not found.", sf); return NULL; }
tsf_set_output(sf, TSF_STEREO_INTERLEAVED, SAMPLERATE, 0.f);
// Preset on 10th MIDI channel to use percussion sound bank if possible
tsf_channel_set_bank_preset(sf, 0, 128, 0);
shput(sf_hash, path, sf);
return sf;
}
dsp_node *dsp_midi(const char *midi, tsf *sf)
{ {
long rawlen; long rawlen;
void *raw = slurp_file(midi, &rawlen); void *raw = slurp_file(midi, &rawlen);
struct dsp_midi_song *ms = malloc(sizeof(*ms));
gsong.midi = tml_load_memory(raw, rawlen); ms->time = 0.0;
if (gsong.midi == NULL) { ms->midi = tml_load_memory(raw, rawlen);
YughWarn("Midi %s not found.", midi); ms->sf = tsf_copy(sf);
free(raw); return make_node(ms, dsp_midi_fillbuf);
return;
}
free(raw);
raw = slurp_file(sf, &rawlen);
gsong.sf = tsf_load_memory(raw, rawlen);
if (gsong.sf == NULL) {
YughWarn("SF2 %s not found.", sf);
free(raw);
return;
}
free(raw);
gsong.time = 0.f;
tsf_set_output(gsong.sf, TSF_STEREO_INTERLEAVED, SAMPLERATE, 0.f);
// Preset on 10th MIDI channel to use percussion sound bank if possible
tsf_channel_set_bank_preset(gsong.sf, 9, 128, 0);
songfil.data = &gsong;
songfil.filter = dsp_midi_fillbuf;
filter_to_bus(&songfil);
} }
void music_stop() void play_song(const char *midi, const char *sf)
{ {
unplug_filter(&songfil); plugin_node(dsp_midi(midi, make_soundfont(sf)), masterbus);
} }

6
source/engine/sound/music.h
View file

@ -3,16 +3,18 @@
#include "tsf.h" #include "tsf.h"
#include "tml.h" #include "tml.h"
#include "dsp.h"
struct dsp_midi_song { struct dsp_midi_song {
float bpm; float bpm;
double time; double time;
tsf *sf;
tml_message *midi; tml_message *midi;
tsf *sf;
}; };
dsp_node *dsp_midi(const char *midi, tsf *sf);
tsf *make_soundfont(const char *sf);
void play_song(const char *midi, const char *sf); void play_song(const char *midi, const char *sf);
void dsp_midi_fillbuf(struct dsp_midi_song *song, void *out, int n); void dsp_midi_fillbuf(struct dsp_midi_song *song, void *out, int n);
void music_stop();
#endif #endif

277
source/engine/sound/sound.c
View file

@ -14,7 +14,9 @@
#include "stb_ds.h" #include "stb_ds.h"
#include "dsp.h" #include "dsp.h"
#include "mix.h"
#define POCKETMOD_IMPLEMENTATION
#include "pocketmod.h"
#include "sokol/sokol_audio.h" #include "sokol/sokol_audio.h"
@ -53,82 +55,77 @@ static struct {
struct wav *value; struct wav *value;
} *wavhash = NULL; } *wavhash = NULL;
static struct wav change_channels(struct wav w, int ch) { void change_channels(struct wav *w, int ch) {
soundbyte *data = w.data; if (w->ch == ch) return;
int samples = ch * w.frames; soundbyte *data = w->data;
int samples = ch * w->frames;
soundbyte *new = malloc(sizeof(soundbyte) * samples); soundbyte *new = malloc(sizeof(soundbyte) * samples);
if (ch > w.ch) { if (ch > w->ch) {
/* Sets all new channels equal to the first one */ /* Sets all new channels equal to the first one */
for (int i = 0; i < w.frames; i++) { for (int i = 0; i < w->frames; i++) {
for (int j = 0; j < ch; j++) for (int j = 0; j < ch; j++)
new[i * ch + j] = data[i]; new[i * ch + j] = data[i];
} }
} else { } else {
/* Simple method; just use first N channels present in wav */ /* Simple method; just use first N channels present in wav */
for (int i = 0; i < w.frames; i++) for (int i = 0; i < w->frames; i++)
for (int j = 0; j < ch; j++) for (int j = 0; j < ch; j++)
new[i * ch + j] = data[i * ch + j]; new[i * ch + j] = data[i * ch + j];
} }
free(w.data); free(w->data);
w.data = new; w->data = new;
return w;
} }
static struct wav change_samplerate(struct wav w, int rate) { void resample(soundbyte *in, soundbyte *out, int in_frames, int out_frames, int channels)
float ratio = (float)rate / w.samplerate; {
int outframes = w.frames * ratio; float ratio = (float)in_frames/out_frames;
SRC_DATA ssrc; SRC_DATA ssrc;
soundbyte *resampled = calloc(w.ch*outframes,sizeof(soundbyte)); ssrc.data_in = in;
ssrc.data_out = out;
ssrc.data_in = w.data; ssrc.input_frames = in_frames;
ssrc.data_out = resampled; ssrc.output_frames = out_frames;
ssrc.input_frames = w.frames;
ssrc.output_frames = outframes;
ssrc.src_ratio = ratio; ssrc.src_ratio = ratio;
int err = src_simple(&ssrc, SRC_LINEAR, channels);
int err = src_simple(&ssrc, SRC_LINEAR, w.ch); if (err)
if (err) {
YughError("Resampling error code %d: %s", err, src_strerror(err)); YughError("Resampling error code %d: %s", err, src_strerror(err));
free(resampled);
return w;
} }
free(w.data); void change_samplerate(struct wav *w, int rate) {
w.data = resampled; if (rate == w->samplerate) return;
w.frames = outframes; float ratio = (float)rate / w->samplerate;
w.samplerate = rate; int outframes = w->frames * ratio;
soundbyte *resampled = malloc(w->ch*outframes*sizeof(soundbyte));
resample(w->data, resampled, w->frames, outframes, w->ch);
free(w->data);
return w; w->data = resampled;
} w->frames = outframes;
w->samplerate = rate;
void wav_norm_gain(struct wav *w, double lv) {
short tarmax = db2short(lv);
short max = 0;
short *s = w->data;
for (int i = 0; i < w->frames; i++) {
for (int j = 0; j < w->ch; j++) {
max = (abs(s[i * w->ch + j]) > max) ? abs(s[i * w->ch + j]) : max;
}
}
float mult = (float)max / tarmax;
for (int i = 0; i < w->frames; i++) {
for (int j = 0; j < w->ch; j++) {
s[i * w->ch + j] *= mult;
}
}
} }
void push_sound(soundbyte *buffer, int frames, int chan) void push_sound(soundbyte *buffer, int frames, int chan)
{ {
bus_fill_buffers(buffer, frames*chan); set_soundbytes(buffer, dsp_node_out(masterbus), frames*chan);
}
void filter_mod(pocketmod_context *mod, soundbyte *buffer, int frames)
{
pocketmod_render(mod, buffer, frames*CHANNELS*sizeof(soundbyte));
}
dsp_node *dsp_mod(const char *path)
{
long modsize;
void *data = slurp_file(path, &modsize);
pocketmod_context *mod = malloc(sizeof(*mod));
pocketmod_init(mod, data, modsize, SAMPLERATE);
return make_node(mod, filter_mod);
} }
void sound_init() { void sound_init() {
mixer_init(); dsp_init();
saudio_setup(&(saudio_desc){ saudio_setup(&(saudio_desc){
.stream_cb = push_sound, .stream_cb = push_sound,
.sample_rate = SAMPLERATE, .sample_rate = SAMPLERATE,
@ -138,6 +135,25 @@ void sound_init() {
}); });
} }
typedef struct {
int channels;
int samplerate;
void *f;
} stream;
void mp3_filter(stream *mp3, soundbyte *buffer, int frames)
{
if (mp3->samplerate == SAMPLERATE) {
drmp3_read_pcm_frames_f32(mp3->f, frames, buffer);
return;
}
int in_frames = (float)mp3->samplerate/SAMPLERATE;
soundbyte *decode = malloc(sizeof(*decode)*in_frames*mp3->channels);
drmp3_read_pcm_frames_f32(mp3->f, in_frames, decode);
resample(decode, buffer, in_frames, frames, CHANNELS);
}
struct wav *make_sound(const char *wav) { struct wav *make_sound(const char *wav) {
int index = shgeti(wavhash, wav); int index = shgeti(wavhash, wav);
if (index != -1) if (index != -1)
@ -150,7 +166,7 @@ struct wav *make_sound(const char *wav) {
return NULL; return NULL;
} }
struct wav mwav; struct wav *mwav = malloc(sizeof(*mwav));
long rawlen; long rawlen;
void *raw = slurp_file(wav, &rawlen); void *raw = slurp_file(wav, &rawlen);
if (!raw) { if (!raw) {
@ -159,11 +175,11 @@ struct wav *make_sound(const char *wav) {
} }
if (!strcmp(ext, "wav")) if (!strcmp(ext, "wav"))
mwav.data = drwav_open_memory_and_read_pcm_frames_f32(raw, rawlen, &mwav.ch, &mwav.samplerate, &mwav.frames, NULL); mwav->data = drwav_open_memory_and_read_pcm_frames_f32(raw, rawlen, &mwav->ch, &mwav->samplerate, &mwav->frames, NULL);
else if (!strcmp(ext, "flac")) { else if (!strcmp(ext, "flac")) {
#ifndef NFLAC #ifndef NFLAC
mwav.data = drflac_open_memory_and_read_pcm_frames_f32(raw, rawlen, &mwav.ch, &mwav.samplerate, &mwav.frames, NULL); mwav->data = drflac_open_memory_and_read_pcm_frames_f32(raw, rawlen, &mwav->ch, &mwav->samplerate, &mwav->frames, NULL);
#else #else
YughWarn("Could not load %s because Primum was built without FLAC support.", wav); YughWarn("Could not load %s because Primum was built without FLAC support.", wav);
#endif #endif
@ -171,9 +187,9 @@ struct wav *make_sound(const char *wav) {
else if (!strcmp(ext, "mp3")) { else if (!strcmp(ext, "mp3")) {
#ifndef NMP3 #ifndef NMP3
drmp3_config cnf; drmp3_config cnf;
mwav.data = drmp3_open_memory_and_read_pcm_frames_f32(raw, rawlen, &cnf, &mwav.frames, NULL); mwav->data = drmp3_open_memory_and_read_pcm_frames_f32(raw, rawlen, &cnf, &mwav->frames, NULL);
mwav.ch = cnf.channels; mwav->ch = cnf.channels;
mwav.samplerate = cnf.sampleRate; mwav->samplerate = cnf.sampleRate;
#else #else
YughWarn("Could not load %s because Primum was built without MP3 support.", wav); YughWarn("Could not load %s because Primum was built without MP3 support.", wav);
#endif #endif
@ -182,11 +198,11 @@ struct wav *make_sound(const char *wav) {
#ifndef NQOA #ifndef NQOA
qoa_desc qoa; qoa_desc qoa;
short *qoa_data = qoa_decode(raw, rawlen, &qoa); short *qoa_data = qoa_decode(raw, rawlen, &qoa);
mwav.ch = qoa.channels; mwav->ch = qoa.channels;
mwav.samplerate = qoa.samplerate; mwav->samplerate = qoa.samplerate;
mwav.frames = qoa.samples; mwav->frames = qoa.samples;
mwav.data = malloc(sizeof(soundbyte) * mwav.frames * mwav.ch); mwav->data = malloc(sizeof(soundbyte) * mwav->frames * mwav->ch);
src_short_to_float_array(qoa_data, mwav.data, mwav.frames*mwav.ch); src_short_to_float_array(qoa_data, mwav->data, mwav->frames*mwav->ch);
free(qoa_data); free(qoa_data);
#else #else
YughWarn("Could not load %s because Primum was built without QOA support.", wav); YughWarn("Could not load %s because Primum was built without QOA support.", wav);
@ -194,23 +210,18 @@ struct wav *make_sound(const char *wav) {
} else { } else {
YughWarn("File with unknown type '%s'.", wav); YughWarn("File with unknown type '%s'.", wav);
free (raw); free (raw);
free(mwav);
return NULL; return NULL;
} }
free(raw); free(raw);
if (mwav.samplerate != SAMPLERATE) change_samplerate(mwav, SAMPLERATE);
mwav = change_samplerate(mwav, SAMPLERATE); change_channels(mwav, CHANNELS);
if (mwav.ch != CHANNELS)
mwav = change_channels(mwav, CHANNELS);
mwav.gain = 1.f;
struct wav *newwav = malloc(sizeof(*newwav));
*newwav = mwav;
if (shlen(wavhash) == 0) sh_new_arena(wavhash); if (shlen(wavhash) == 0) sh_new_arena(wavhash);
shput(wavhash, wav, newwav); shput(wavhash, wav, mwav);
return newwav; return mwav;
} }
void free_sound(const char *wav) { void free_sound(const char *wav) {
@ -222,68 +233,52 @@ void free_sound(const char *wav) {
shdel(wavhash, wav); shdel(wavhash, wav);
} }
struct soundstream *soundstream_make() { void sound_fillbuf(struct sound *s, soundbyte *buf, int n) {
struct soundstream *new = malloc(sizeof(*new)); int frames = s->data->frames - s->frame;
new->buf = circbuf_make(sizeof(short), BUF_FRAMES * CHANNELS * 2); if (frames == 0) return;
return new; int end = 0;
if (frames > n)
frames = n;
else
end = 1;
soundbyte *in = s->data->data;
for (int i = 0; i < frames; i++) {
for (int j = 0; j < CHANNELS; j++)
buf[i * CHANNELS + j] = in[s->frame*CHANNELS + j];
s->frame++;
} }
void kill_oneshot(struct sound *s) { if(end) {
if (s->loop)
s->frame = 0;
call_env(s->hook, "this.end();");
}
}
void free_source(struct sound *s)
{
JS_FreeValue(js, s->hook);
free(s); free(s);
} }
void play_oneshot(struct wav *wav) { struct dsp_node *dsp_source(char *path)
{
struct sound *self = malloc(sizeof(*self)); struct sound *self = malloc(sizeof(*self));
self->data = wav;
self->bus = first_free_bus(dsp_filter(self, sound_fillbuf));
self->playing = 1;
self->loop = 0;
self->frame = 0; self->frame = 0;
self->endcb = kill_oneshot; self->data = make_sound(path);
} self->loop = false;
self->hook = JS_UNDEFINED;
struct sound *play_sound(struct wav *wav) { dsp_node *n = make_node(self, sound_fillbuf);
struct sound *self = calloc(1, sizeof(*self)); n->data_free = free_source;
self->data = wav; return n;
self->bus = first_free_bus(dsp_filter(self, sound_fillbuf));
self->playing = 1;
self->loop = 0;
self->frame = 0;
self->endcb = kill_oneshot;
return self;
}
int sound_playing(const struct sound *s) {
return !sound_paused(s);
}
int sound_paused(const struct sound *s) {
return s->bus == NULL;
}
void sound_pause(struct sound *s) {
if (s->bus == NULL) return;
bus_free(s->bus);
s->bus = NULL;
}
void sound_resume(struct sound *s) {
if (s->bus != NULL) return;
s->bus = first_free_bus(dsp_filter(s, sound_fillbuf));
}
void sound_stop(struct sound *s) {
sound_pause(s);
s->frame = 0;
} }
int sound_finished(const struct sound *s) { int sound_finished(const struct sound *s) {
return s->frame == s->data->frames; return s->frame == s->data->frames;
} }
int sound_stopped(const struct sound *s) {
return s->bus == NULL;
}
struct mp3 make_music(const char *mp3) { struct mp3 make_music(const char *mp3) {
// drmp3 new; // drmp3 new;
// if (!drmp3_init_file(&new, mp3, NULL)) { // if (!drmp3_init_file(&new, mp3, NULL)) {
@ -294,41 +289,6 @@ struct mp3 make_music(const char *mp3) {
return newmp3; return newmp3;
} }
void close_audio_device(int device) {
}
int open_device(const char *adriver) {
return 0;
}
void sound_fillbuf(struct sound *s, soundbyte *buf, int n) {
float gainmult = pct2mult(s->data->gain);
soundbyte *in = s->data->data;
for (int i = 0; i < n; i++) {
for (int j = 0; j < CHANNELS; j++)
buf[i * CHANNELS + j] = in[s->frame*CHANNELS + j] * gainmult;
s->frame++;
if (s->frame == s->data->frames) {
sound_stop(s);
s->endcb(s);
return;
}
}
}
void mp3_fillbuf(struct sound *s, soundbyte *buf, int n) {
}
void soundstream_fillbuf(struct soundstream *s, soundbyte *buf, int n) {
int max = 1;//s->buf->write - s->buf->read;
int lim = (max < n * CHANNELS) ? max : n * CHANNELS;
for (int i = 0; i < lim; i++) {
// buf[i] = cbuf_shift(s->buf);
}
}
float short2db(short val) { float short2db(short val) {
return 20 * log10(abs(val) / SHRT_MAX); return 20 * log10(abs(val) / SHRT_MAX);
} }
@ -341,6 +301,13 @@ short short_gain(short val, float db) {
return (short)(pow(10, db / 20.f) * val); return (short)(pow(10, db / 20.f) * val);
} }
float float2db(float val) { return 20 * log10(fabsf(val)); }
float db2float(float db) { return pow(10, db/20); }
float fgain(float val, float db) {
return pow(10,db/20.f)*val;
}
float pct2db(float pct) { float pct2db(float pct) {
if (pct <= 0) return -72.f; if (pct <= 0) return -72.f;

60
source/engine/sound/sound.h
View file

@ -1,81 +1,59 @@
#ifndef SOUND_H #ifndef SOUND_H
#define SOUND_H #define SOUND_H
#include "cbuf.h" #include "script.h"
typedef float soundbyte; typedef float soundbyte;
struct soundstream { struct dsp_node;
struct circbuf *buf;
};
struct soundstream *soundstream_make();
/* A bookmark into a wav, actually playing the sound */ /* A bookmark into a wav, actually playing the sound */
struct sound { typedef struct sound {
int loop; /* How many times to loop */
unsigned int frame; /* Pointing to the current frame on the wav */ unsigned int frame; /* Pointing to the current frame on the wav */
int playing;
float gain;
struct wav *data; struct wav *data;
struct bus *bus; int loop;
JSValue hook;
void (*endcb)(struct sound*); } sound;
};
/* Represents a sound file source, fulled loaded*/ /* Represents a sound file source, fulled loaded*/
struct wav { typedef struct wav {
unsigned int ch; unsigned int ch;
unsigned int samplerate; unsigned int samplerate;
unsigned long long frames; unsigned long long frames;
float gain; /* In dB */
soundbyte *data; soundbyte *data;
}; } wav;
/* Represents a sound file stream */ /* Represents a sound file stream */
struct mp3 { typedef struct mp3 {
}; } mp3;
typedef struct qoa {
// struct qoa_desc desc;
} qoa;
void sound_init(); void sound_init();
void audio_open(const char *device); void audio_open(const char *device);
void audio_close(); void audio_close();
void sound_fillbuf(struct sound *s, soundbyte *buf, int n);
struct wav *make_sound(const char *wav); struct wav *make_sound(const char *wav);
void free_sound(const char *wav); void free_sound(const char *wav);
void wav_norm_gain(struct wav *w, double lv); void wav_norm_gain(struct wav *w, double lv);
struct sound *play_sound(struct wav *wav); struct dsp_node *dsp_source(char *path);
void play_oneshot(struct wav *wav); struct dsp_node *dsp_mod(const char *path);
int sound_playing(const struct sound *s);
int sound_paused(const struct sound *s);
int sound_stopped(const struct sound *s);
int sound_finished(const struct sound *s); int sound_finished(const struct sound *s);
void sound_pause(struct sound *s);
void sound_resume(struct sound *s);
void sound_stop(struct sound *s);
struct mp3 make_mp3(const char *mp3); struct mp3 make_mp3(const char *mp3);
const char *get_audio_driver();
void soundstream_fillbuf(struct soundstream *stream, soundbyte *buf, int n);
void close_audio_device(int device);
int open_device(const char *adriver);
float short2db(short val); float short2db(short val);
short db2short(float db); short db2short(float db);
short short_gain(short val, float db); short short_gain(short val, float db);
float fgain(float val, float db);
float float2db(float val);
float db2float(float db);
float pct2db(float pct); float pct2db(float pct);
float pct2mult(float pct); float pct2mult(float pct);
void audio_init();
#endif #endif

6
source/engine/thirdparty/sokol/sokol_app.h vendored
View file

@ -3503,9 +3503,15 @@ NSMenu* menu_bar = [[NSMenu alloc] init];
action:@selector(unhideAllApplications:) action:@selector(unhideAllApplications:)
keyEquivalent:@""]; keyEquivalent:@""];
NSMenuItem *fullscreen_item = [[NSMenuItem alloc]
initWithTitle:@"Enter Full Screen"
action:@selector(toggleFullScreen:)
keyEquivalent:@"f"];
[app_menu addItem:hide_menu_item]; [app_menu addItem:hide_menu_item];
[app_menu addItem:hide_others_item]; [app_menu addItem:hide_others_item];
[app_menu addItem:show_all_item]; [app_menu addItem:show_all_item];
[app_menu addItem:fullscreen_item];
[app_menu addItem:[NSMenuItem separatorItem]]; [app_menu addItem:[NSMenuItem separatorItem]];
[app_menu addItem:quit_menu_item]; [app_menu addItem:quit_menu_item];
app_menu_item.submenu = app_menu; app_menu_item.submenu = app_menu;

8
source/engine/timer.c
View file

@ -77,7 +77,7 @@ void timer_start(struct timer *t) {
void timer_remove(int id) { void timer_remove(int id) {
struct timer *t = id2timer(id); struct timer *t = id2timer(id);
free_callee(t->data); // free_callee(t->data);
t->next = first; t->next = first;
t->on = 0; t->on = 0;
first = id; first = id;
@ -94,7 +94,7 @@ struct timer *id2timer(int id) {
void timers_free() void timers_free()
{ {
for (int i = 0; i < arrlen(timers); i++) // for (int i = 0; i < arrlen(timers); i++)
if (timers[i].on) // if (timers[i].on)
free_callee(timers[i].data); // free_callee(timers[i].data);
} }

10
source/engine/yugine.c
View file

@ -7,7 +7,6 @@
#include "window.h" #include "window.h"
#include "sound.h" #include "sound.h"
#include "resources.h" #include "resources.h"
#include <stdio.h> #include <stdio.h>
#include "datastream.h" #include "datastream.h"
@ -142,8 +141,10 @@ const char *engine_info()
static int argc; static int argc;
static char **args; static char **args;
struct datastream *bjork;
void c_init() { void c_init() {
sound_init();
input_init(); input_init();
script_evalf("world_start();"); script_evalf("world_start();");
@ -151,6 +152,7 @@ void c_init() {
window_set_icon("icons/moon.gif"); window_set_icon("icons/moon.gif");
window_resize(sapp_width(), sapp_height()); window_resize(sapp_width(), sapp_height());
script_evalf("Game.init();"); script_evalf("Game.init();");
// bjork = ds_openvideo("bjork.mpg");
} }
int frame_fps() { int frame_fps() {
@ -160,6 +162,7 @@ int frame_fps() {
static void process_frame() static void process_frame()
{ {
double elapsed = stm_sec(stm_laptime(&frame_t)); double elapsed = stm_sec(stm_laptime(&frame_t));
// ds_advance(bjork, elapsed);
input_poll(0); input_poll(0);
/* Timers all update every frame - once per monitor refresh */ /* Timers all update every frame - once per monitor refresh */
timer_update(elapsed, timescale); timer_update(elapsed, timescale);
@ -374,7 +377,7 @@ dam->update_activity(dam, &da, NULL, NULL);
stm_setup(); /* time */ stm_setup(); /* time */
start_t = frame_t = stm_now(); start_t = frame_t = stm_now();
physlast = updatelast = start_t; physlast = updatelast = start_t;
sound_init();
resources_init(); resources_init();
phys2d_init(); phys2d_init();
script_startup(); script_startup();
@ -398,7 +401,6 @@ dam->update_activity(dam, &da, NULL, NULL);
start_desc.width = mainwin.width; start_desc.width = mainwin.width;
start_desc.height = mainwin.height; start_desc.height = mainwin.height;
start_desc.fullscreen = 0; start_desc.fullscreen = 0;
sapp_run(&start_desc); sapp_run(&start_desc);
return 0; return 0;