336 lines
7.6 KiB
C
336 lines
7.6 KiB
C
#include "sound.h"
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#include "limits.h"
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#include "log.h"
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#include "math.h"
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#include "music.h"
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#include "resources.h"
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#include "stb_vorbis.h"
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#include "string.h"
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#include "time.h"
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#include <stdlib.h>
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#include "samplerate.h"
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#include "stb_ds.h"
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#include "dsp.h"
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#include "mix.h"
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#include "sokol/sokol_audio.h"
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#define TSF_IMPLEMENTATION
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#include "tsf.h"
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#define TML_IMPLEMENTATION
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#include "tml.h"
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#define DR_WAV_IMPLEMENTATION
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#include "dr_wav.h"
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#define DR_FLAC_IMPLEMENTATION
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#include "dr_flac.h"
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#define DR_MP3_IMPLEMENTATION
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#include "dr_mp3.h"
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#define QOA_IMPLEMENTATION
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#include "qoa.h"
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static struct {
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char *key;
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struct wav *value;
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} *wavhash = NULL;
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static struct wav change_channels(struct wav w, int ch) {
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soundbyte *data = w.data;
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int samples = ch * w.frames;
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soundbyte *new = malloc(sizeof(soundbyte) * samples);
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if (ch > w.ch) {
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/* Sets all new channels equal to the first one */
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for (int i = 0; i < w.frames; i++) {
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for (int j = 0; j < ch; j++)
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new[i * ch + j] = data[i];
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}
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} else {
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/* Simple method; just use first N channels present in wav */
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for (int i = 0; i < w.frames; i++)
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for (int j = 0; j < ch; j++)
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new[i * ch + j] = data[i * ch + j];
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}
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free(w.data);
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w.data = new;
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return w;
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}
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static struct wav change_samplerate(struct wav w, int rate) {
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float ratio = (float)rate / w.samplerate;
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int outframes = w.frames * ratio;
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SRC_DATA ssrc;
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soundbyte *resampled = calloc(w.ch*outframes,sizeof(soundbyte));
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ssrc.data_in = w.data;
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ssrc.data_out = resampled;
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ssrc.input_frames = w.frames;
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ssrc.output_frames = outframes;
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ssrc.src_ratio = ratio;
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int err = src_simple(&ssrc, SRC_LINEAR, w.ch);
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if (err) {
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YughError("Resampling error code %d: %s", err, src_strerror(err));
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free(resampled);
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return w;
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}
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free(w.data);
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w.data = resampled;
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w.frames = outframes;
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w.samplerate = rate;
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return w;
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}
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void wav_norm_gain(struct wav *w, double lv) {
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short tarmax = db2short(lv);
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short max = 0;
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short *s = w->data;
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for (int i = 0; i < w->frames; i++) {
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for (int j = 0; j < w->ch; j++) {
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max = (abs(s[i * w->ch + j]) > max) ? abs(s[i * w->ch + j]) : max;
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}
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}
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float mult = (float)max / tarmax;
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for (int i = 0; i < w->frames; i++) {
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for (int j = 0; j < w->ch; j++) {
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s[i * w->ch + j] *= mult;
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}
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}
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}
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void push_sound(soundbyte *buffer, int frames, int chan)
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{
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bus_fill_buffers(buffer, frames*chan);
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}
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void sound_init() {
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saudio_setup(&(saudio_desc){
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.stream_cb = push_sound,
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.sample_rate = SAMPLERATE,
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.num_channels = CHANNELS,
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.buffer_frames = BUF_FRAMES,
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.logger.func = sg_logging,
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});
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mixer_init();
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}
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struct wav *make_sound(const char *wav) {
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int index = shgeti(wavhash, wav);
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if (index != -1) return wavhash[index].value;
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char *ext = strrchr(wav, '.')+1;
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if(!ext) {
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YughWarn("No extension detected for %s.", wav);
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return NULL;
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}
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struct wav mwav;
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if (!strcmp(ext, "wav")) {
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mwav.data = drwav_open_file_and_read_pcm_frames_f32(wav, &mwav.ch, &mwav.samplerate, &mwav.frames, NULL);
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} else if (!strcmp(ext, "flac")) {
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mwav.data = drflac_open_file_and_read_pcm_frames_f32(wav, &mwav.ch, &mwav.samplerate, &mwav.frames, NULL);
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} else if (!strcmp(ext, "mp3")) {
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drmp3_config cnf;
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mwav.data = drmp3_open_file_and_read_pcm_frames_f32(wav, &cnf, &mwav.frames, NULL);
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mwav.ch = cnf.channels;
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mwav.samplerate = cnf.sampleRate;
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} else if (!strcmp(ext, "qoa")) {
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unsigned char header[QOA_MIN_FILESIZE];
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FILE *f = fopen(wav, "rb");
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fread(header, QOA_MIN_FILESIZE, 1, f);
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qoa_desc qoa;
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unsigned int ff_pos = qoa_decode_header(header, QOA_MIN_FILESIZE, &qoa);
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mwav.ch = qoa.channels;
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mwav.samplerate = qoa.samplerate;
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mwav.frames = qoa.samples;
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short *qoa_data = qoa_read(wav, &qoa);
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mwav.data = malloc(sizeof(soundbyte) * mwav.frames * mwav.ch);
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src_short_to_float_array(qoa_data, mwav.data, mwav.frames*mwav.ch);
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fclose(f);
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free(qoa_data);
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} else {
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YughWarn("Cannot process file type '%s'.", ext);
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return NULL;
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}
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YughWarn("%s opened with %d ch, %d samplerate, %d frames", ext, mwav.ch, mwav.samplerate, mwav.frames);
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if (mwav.samplerate != SAMPLERATE) {
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YughWarn("Changing samplerate of %s from %d to %d.", wav, mwav.samplerate, SAMPLERATE);
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mwav = change_samplerate(mwav, SAMPLERATE);
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}
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if (mwav.ch != CHANNELS) {
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YughWarn("Changing channels of %s from %d to %d.", wav, mwav.ch, CHANNELS);
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mwav = change_channels(mwav, CHANNELS);
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}
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mwav.gain = 1.f;
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struct wav *newwav = malloc(sizeof(*newwav));
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*newwav = mwav;
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if (shlen(wavhash) == 0) sh_new_arena(wavhash);
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shput(wavhash, wav, newwav);
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YughWarn("Channels %d, sr %d", newwav->ch,newwav->samplerate);
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return newwav;
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}
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void free_sound(const char *wav) {
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struct wav *w = shget(wavhash, wav);
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if (w == NULL) return;
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free(w->data);
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free(w);
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shdel(wavhash, wav);
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}
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struct soundstream *soundstream_make() {
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struct soundstream *new = malloc(sizeof(*new));
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new->buf = circbuf_make(sizeof(short), BUF_FRAMES * CHANNELS * 2);
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return new;
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}
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void kill_oneshot(struct sound *s) {
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free(s);
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}
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void play_oneshot(struct wav *wav) {
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struct sound *new = malloc(sizeof(*new));
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new->data = wav;
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new->bus = first_free_bus(dsp_filter(new, sound_fillbuf));
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new->playing = 1;
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new->loop = 0;
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new->frame = 0;
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new->endcb = kill_oneshot;
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}
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struct sound *play_sound(struct wav *wav) {
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struct sound *new = calloc(1, sizeof(*new));
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new->data = wav;
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new->bus = first_free_bus(dsp_filter(new, sound_fillbuf));
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new->playing = 1;
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return new;
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}
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int sound_playing(const struct sound *s) {
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return s->playing;
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}
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int sound_paused(const struct sound *s) {
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return (!s->playing && s->frame < s->data->frames);
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}
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void sound_pause(struct sound *s) {
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s->playing = 0;
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bus_free(s->bus);
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}
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void sound_resume(struct sound *s) {
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s->playing = 1;
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s->bus = first_free_bus(dsp_filter(s, sound_fillbuf));
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}
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void sound_stop(struct sound *s) {
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s->playing = 0;
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s->frame = 0;
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bus_free(s->bus);
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}
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int sound_finished(const struct sound *s) {
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return !s->playing && s->frame == s->data->frames;
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}
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int sound_stopped(const struct sound *s) {
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return !s->playing && s->frame == 0;
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}
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struct mp3 make_music(const char *mp3) {
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// drmp3 new;
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// if (!drmp3_init_file(&new, mp3, NULL)) {
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// YughError("Could not open mp3 file %s.", mp3);
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// }
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struct mp3 newmp3 = {};
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return newmp3;
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}
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void close_audio_device(int device) {
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}
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int open_device(const char *adriver) {
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return 0;
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}
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void sound_fillbuf(struct sound *s, soundbyte *buf, int n) {
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float gainmult = pct2mult(s->data->gain);
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soundbyte *in = s->data->data;
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for (int i = 0; i < n; i++) {
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for (int j = 0; j < CHANNELS; j++)
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buf[i * CHANNELS + j] = in[s->frame*CHANNELS + j] * gainmult;
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s->frame++;
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if (s->frame == s->data->frames) {
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bus_free(s->bus);
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s->bus = NULL;
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s->endcb(s);
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return;
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}
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}
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}
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void mp3_fillbuf(struct sound *s, soundbyte *buf, int n) {
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}
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void soundstream_fillbuf(struct soundstream *s, soundbyte *buf, int n) {
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int max = 1;//s->buf->write - s->buf->read;
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int lim = (max < n * CHANNELS) ? max : n * CHANNELS;
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for (int i = 0; i < lim; i++) {
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// buf[i] = cbuf_shift(s->buf);
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}
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}
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float short2db(short val) {
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return 20 * log10(abs(val) / SHRT_MAX);
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}
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short db2short(float db) {
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return pow(10, db / 20.f) * SHRT_MAX;
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}
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short short_gain(short val, float db) {
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return (short)(pow(10, db / 20.f) * val);
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}
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float pct2db(float pct) {
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if (pct <= 0) return -72.f;
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return 10 * log2(pct);
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}
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float pct2mult(float pct) {
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if (pct <= 0) return 0.f;
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return pow(10, 0.5 * log2(pct));
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}
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