341 lines
6.8 KiB
C
341 lines
6.8 KiB
C
#include "sound.h"
|
|
#include "resources.h"
|
|
#include <stdlib.h>
|
|
#include "log.h"
|
|
#include "string.h"
|
|
#include "math.h"
|
|
#include "limits.h"
|
|
#include "time.h"
|
|
#include "music.h"
|
|
#include "stb_vorbis.h"
|
|
|
|
#include "samplerate.h"
|
|
|
|
#include "stb_ds.h"
|
|
|
|
#include "mix.h"
|
|
#include "dsp.h"
|
|
|
|
#include "miniaudio.h"
|
|
|
|
#define TSF_IMPLEMENTATION
|
|
#include "tsf.h"
|
|
|
|
#define TML_IMPLEMENTATION
|
|
#include "tml.h"
|
|
|
|
static struct {
|
|
char *key;
|
|
struct wav *value;
|
|
} *wavhash = NULL;
|
|
|
|
static struct wav change_channels(struct wav w, int ch)
|
|
{
|
|
short *data = w.data;
|
|
int samples = ch * w.frames;
|
|
short *new = malloc(sizeof(short)*samples);
|
|
|
|
if (ch > w.ch) {
|
|
/* Sets all new channels equal to the first one */
|
|
for (int i = 0; i < w.frames; i++) {
|
|
for (int j = 0; j < ch; j++)
|
|
new[i*ch+j] = data[i];
|
|
}
|
|
} else {
|
|
/* Simple method; just use first N channels present in wav */
|
|
for (int i = 0; i < w.frames; i++)
|
|
for (int j = 0; j < ch; j++)
|
|
new[i*ch+j] = data[i*ch+j];
|
|
}
|
|
|
|
free (w.data);
|
|
w.data = new;
|
|
return w;
|
|
}
|
|
|
|
static struct wav change_samplerate(struct wav w, int rate)
|
|
{
|
|
float ratio = (float)rate/w.samplerate;
|
|
int outframes = w.frames * ratio;
|
|
SRC_DATA ssrc;
|
|
float floatdata[w.frames*w.ch];
|
|
src_short_to_float_array(w.data, floatdata, w.frames*w.ch);
|
|
float resampled[w.ch*outframes];
|
|
|
|
ssrc.data_in = floatdata;
|
|
ssrc.data_out = resampled;
|
|
ssrc.input_frames = w.frames;
|
|
ssrc.output_frames = outframes;
|
|
ssrc.src_ratio = ratio;
|
|
|
|
src_simple(&ssrc, SRC_SINC_BEST_QUALITY, w.ch);
|
|
|
|
short *newdata = malloc(sizeof(short)*outframes*w.ch);
|
|
src_float_to_short_array(resampled, newdata, outframes*w.ch);
|
|
|
|
free(w.data);
|
|
w.data = newdata;
|
|
w.samplerate = rate;
|
|
|
|
return w;
|
|
}
|
|
|
|
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;
|
|
}
|
|
}
|
|
}
|
|
|
|
static ma_engine *engine;
|
|
|
|
void sound_init()
|
|
{
|
|
ma_result result;
|
|
engine = malloc(sizeof(*engine));
|
|
result = ma_engine_init(NULL, engine);
|
|
if (result != MA_SUCCESS) {
|
|
return;
|
|
}
|
|
return;
|
|
|
|
mixer_init();
|
|
}
|
|
|
|
struct wav *make_sound(const char *wav)
|
|
{
|
|
int index = shgeti(wavhash, wav);
|
|
if (index != -1) return wavhash[index].value;
|
|
|
|
struct wav mwav;
|
|
mwav.data = drwav_open_file_and_read_pcm_frames_s16(wav, &mwav.ch, &mwav.samplerate, &mwav.frames, NULL);
|
|
|
|
if (mwav.samplerate != SAMPLERATE) {
|
|
YughInfo("Changing samplerate of %s from %d to %d.", wav, mwav.samplerate, SAMPLERATE);
|
|
// mwav = change_samplerate(mwav, SAMPLERATE);
|
|
}
|
|
|
|
if (mwav.ch != CHANNELS) {
|
|
YughInfo("Changing channels of %s from %d to %d.", wav, 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);
|
|
|
|
shput(wavhash, wav, newwav);
|
|
|
|
return newwav;
|
|
}
|
|
|
|
void free_sound(const char *wav)
|
|
{
|
|
struct wav *w = shget(wavhash, wav);
|
|
if (w == NULL) return;
|
|
|
|
free(w->data);
|
|
free(w);
|
|
shdel(wavhash, wav);
|
|
}
|
|
|
|
struct soundstream *soundstream_make()
|
|
{
|
|
struct soundstream *new = malloc(sizeof(*new));
|
|
new->buf = circbuf_make(sizeof(short), BUF_FRAMES*CHANNELS*2);
|
|
return new;
|
|
}
|
|
|
|
void mini_sound(char *path)
|
|
{
|
|
ma_engine_play_sound(engine, path, NULL);
|
|
}
|
|
|
|
static ma_sound music_sound;
|
|
|
|
void mini_music_play(char *path)
|
|
{
|
|
int result = ma_sound_init_from_file(engine, path, MA_SOUND_FLAG_NO_SPATIALIZATION, NULL, NULL, &music_sound);
|
|
if (result != MA_SUCCESS) {
|
|
YughInfo("DID NOT LOAD SOUND!");
|
|
}
|
|
ma_sound_start(&music_sound);
|
|
}
|
|
|
|
void mini_music_pause()
|
|
{
|
|
ma_sound_stop(&music_sound);
|
|
}
|
|
|
|
void mini_music_stop()
|
|
{
|
|
ma_sound_stop(&music_sound);
|
|
}
|
|
|
|
void mini_master(float v)
|
|
{
|
|
ma_engine_set_volume(engine, v);
|
|
}
|
|
|
|
void kill_oneshot(struct sound *s)
|
|
{
|
|
free(s);
|
|
}
|
|
|
|
void play_oneshot(struct wav *wav) {
|
|
struct sound *new = malloc(sizeof(*new));
|
|
new->data = wav;
|
|
new->bus = first_free_bus(dsp_filter(new, sound_fillbuf));
|
|
new->playing=1;
|
|
new->loop=0;
|
|
new->frame = 0;
|
|
new->endcb = kill_oneshot;
|
|
|
|
}
|
|
|
|
struct sound *play_sound(struct wav *wav)
|
|
{
|
|
struct sound *new = calloc(1, sizeof(*new));
|
|
new->data = wav;
|
|
|
|
new->bus = first_free_bus(dsp_filter(new, sound_fillbuf));
|
|
new->playing = 1;
|
|
|
|
return new;
|
|
}
|
|
|
|
int sound_playing(const struct sound *s)
|
|
{
|
|
return s->playing;
|
|
}
|
|
|
|
int sound_paused(const struct sound *s)
|
|
{
|
|
return (!s->playing && s->frame < s->data->frames);
|
|
}
|
|
void sound_pause(struct sound *s)
|
|
{
|
|
s->playing = 0;
|
|
bus_free(s->bus);
|
|
}
|
|
|
|
void sound_resume(struct sound *s)
|
|
{
|
|
s->playing = 1;
|
|
s->bus = first_free_bus(dsp_filter(s, sound_fillbuf));
|
|
}
|
|
|
|
void sound_stop(struct sound *s)
|
|
{
|
|
s->playing = 0;
|
|
s->frame = 0;
|
|
bus_free(s->bus);
|
|
}
|
|
|
|
int sound_finished(const struct sound *s)
|
|
{
|
|
return !s->playing && s->frame == s->data->frames;
|
|
}
|
|
|
|
int sound_stopped(const struct sound *s)
|
|
{
|
|
return !s->playing && s->frame == 0;
|
|
}
|
|
|
|
struct mp3 make_music(const char *mp3)
|
|
{
|
|
// drmp3 new;
|
|
// if (!drmp3_init_file(&new, mp3, NULL)) {
|
|
// YughError("Could not open mp3 file %s.", mp3);
|
|
// }
|
|
|
|
struct mp3 newmp3 = {};
|
|
return newmp3;
|
|
}
|
|
|
|
void close_audio_device(int device)
|
|
{
|
|
}
|
|
|
|
int open_device(const char *adriver)
|
|
{
|
|
return 0;
|
|
}
|
|
|
|
void sound_fillbuf(struct sound *s, short *buf, int n)
|
|
{
|
|
float gainmult = pct2mult(s->data->gain);
|
|
|
|
short *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+j] * gainmult;
|
|
s->frame++;
|
|
if (s->frame == s->data->frames) {
|
|
|
|
bus_free(s->bus);
|
|
s->bus = NULL;
|
|
s->endcb(s);
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
void mp3_fillbuf(struct sound *s, short *buf, int n)
|
|
{
|
|
|
|
}
|
|
|
|
void soundstream_fillbuf(struct soundstream *s, short *buf, int n)
|
|
{
|
|
int max = 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)
|
|
{
|
|
return 20*log10(abs(val) / SHRT_MAX);
|
|
}
|
|
|
|
short db2short(float db)
|
|
{
|
|
return pow(10, db/20.f) * SHRT_MAX;
|
|
}
|
|
|
|
short short_gain(short val, float db)
|
|
{
|
|
return (short)(pow(10, db/20.f) * val);
|
|
}
|
|
|
|
float pct2db(float pct)
|
|
{
|
|
if (pct <= 0) return -72.f;
|
|
|
|
return 10*log2(pct);
|
|
}
|
|
|
|
float pct2mult(float pct)
|
|
{
|
|
if (pct <= 0) return 0.f;
|
|
|
|
return pow(10, 0.5*log2(pct));
|
|
}
|