flac, mp3, wav playing

Makefile

@@ -32,6 +32,9 @@ ifdef ED
   QFLAGS += -DED
 endif
 
+QFLAGS += -DHAVE_CEIL -DHAVE_FLOOR -DHAVE_FMOD -DHAVE_LRINT -DHAVE_LRINTF
+
+
 PTYPE != uname -m
 
 BIN = bin/$(CC)/$(INFO)/

docs/video.md

@@ -5,6 +5,6 @@ Yugine plays the open source MPEG-TS muxer, using MPEG1 video and MP2 audio.
 MPEG-1 video works best at about 1.5 Mbit/s, in SD [720x480] video.
 For HD [1920x1080] video, use 9 Mbit/s.
 
-ffmpeg -i "input.video" -vcodec mpeg1video -b:v 1.5M -s 720x480 -acodec mp2 "out.ts"
+ffmpeg -i "input.video" -c:v mpeg1video -b:v 1.5M -s 720x480 -c:a mp2 -format mpeg "out.mpg"
-ffmpeg -i "input.video" -vcodec mpeg1video -b:v 9M -s 1920x1080 -acodec mp2 "out.ts"
+ffmpeg -i "input.video" -vcodec mpeg1video -b:v 9M -s 1920x1080 -acodec mp2 -format mpeg "out.mpg"
 

source/engine/cbuf.h

@@ -0,0 +1,95 @@
+#ifndef CIRCBUF_H
+#define CIRCBUF_H
+
+#include <stdio.h>
+#include <stdint.h>
+
+struct circbuf {
+    int16_t *data;
+    uint32_t read;
+    uint32_t write;
+    unsigned int len;
+};
+
+struct circbuf *circbuf_make(size_t size, unsigned int len);
+struct circbuf circbuf_init(size_t size, unsigned int len);
+void cbuf_push(struct circbuf *buf, short data);
+short cbuf_shift(struct circbuf *buf);
+
+#endif
+
+#ifdef CBUF_IMPLEMENT
+
+#include "circbuf.h"
+
+#include "util.h"
+#include "assert.h"
+#include "stdlib.h"
+
+unsigned int powof2(unsigned int num)
+{
+    if (num != 0) {
+	num--;
+	num |= (num >> 1);
+	num |= (num >> 2);
+	num |= (num >> 4);
+	num |= (num >> 8);
+	num |= (num >> 16);
+	num++;
+    }
+
+    return num;
+}
+
+int ispow2(int num)
+{
+    return (num && !(num & (num - 1)));
+}
+
+struct circbuf *circbuf_make(size_t size, unsigned int len)
+{
+    struct circbuf *new = malloc(sizeof(*new));
+    new->len = powof2(len);
+    new->data = calloc(sizeof(short), new->len);
+    new->read = new->write = 0;
+    return new;
+}
+
+struct circbuf circbuf_init(size_t size, unsigned int len)
+{
+    struct circbuf new;
+    new.len = powof2(len);
+    new.data = calloc(sizeof(short), new.len);
+    new.read = new.write = 0;
+
+    return new;
+}
+
+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

source/engine/datastream.c

@@ -16,14 +16,15 @@
 #include "font.h"
 #include "openglrender.h"
 
+#define CBUF_IMPLEMENT
+#include "cbuf.h"
+
 #include "sokol/sokol_gfx.h"
 
 sg_shader vid_shader;
 sg_pipeline vid_pipeline;
 sg_bindings vid_bind;
 
-
-
 static void render_frame(plm_t *mpeg, plm_frame_t *frame, void *user) {
   struct datastream *ds = user;
   uint8_t rgb[frame->height*frame->width*4];
@@ -49,26 +50,25 @@ static void render_audio(plm_t *mpeg, plm_samples_t *samples, void *user) {
 }
 
 void ds_openvideo(struct datastream *ds, const char *video, const char *adriver) {
-  // ds_stop(ds);
+  ds->plm = plm_create_with_filename(video);
-  char buf[MAXPATH] = {'\0'};
-  sprintf(buf, "%s%s", "video/", video);
-  ds->plm = plm_create_with_filename(buf);
 
   if (!ds->plm) {
-    YughLog(0, 0, "Couldn't open %s", video);
+    YughError("Couldn't open %s", video);
   }
 
+  YughWarn("Opened %s - framerate: %f, samplerate: %d, audio streams: %i, duration: %f",
+          video,
+          plm_get_framerate(ds->plm),
+          plm_get_samplerate(ds->plm),
+
+          plm_get_duration(ds->plm));
+
+
   ds->img = sg_make_image(&(sg_image_desc){
     .width = plm_get_width(ds->plm),
     .height = plm_get_height(ds->plm)
   });  
 
-  YughLog(0, 0, "Opened %s - framerate: %f, samplerate: %d, audio streams: %i, duration: %f",
-          video,
-          plm_get_framerate(ds->plm),
-          plm_get_samplerate(ds->plm),
-          plm_get_num_audio_streams(ds->plm),
-          plm_get_duration(ds->plm));
 
   ds->astream = soundstream_make();
   struct dsp_filter astream_filter;

source/engine/ffi.c

@@ -641,7 +641,7 @@ JSValue duk_cmd(JSContext *js, JSValueConst this, int argc, JSValueConst *argv)
 
   case 14:
     str = JS_ToCString(js, argv[1]);
-    mini_sound(str);
+    play_oneshot(make_sound(str));
     break;
 
   case 15:
@@ -661,7 +661,7 @@ JSValue duk_cmd(JSContext *js, JSValueConst this, int argc, JSValueConst *argv)
     break;
 
   case 19:
-    mini_master(js2number(argv[1]));
+//    mini_master(js2number(argv[1]));
     break;
 
   case 20:
@@ -918,15 +918,15 @@ JSValue duk_cmd(JSContext *js, JSValueConst this, int argc, JSValueConst *argv)
 
   case 87:
     str = JS_ToCString(js, argv[1]);
-    mini_music_play(str);
+//    mini_music_play(str);
     break;
 
   case 88:
-    mini_music_pause();
+//    mini_music_pause();
     return JS_NULL;
 
   case 89:
-    mini_music_stop();
+//    mini_music_stop();
     return JS_NULL;
 
   case 90:

source/engine/sound.c

@@ -25,15 +25,24 @@
 #define TML_IMPLEMENTATION
 #include "tml.h"
 
+#define DR_WAV_IMPLEMENTATION
+#include "dr_wav.h"
+
+#define DR_FLAC_IMPLEMENTATION
+#include "dr_flac.h"
+
+#define DR_MP3_IMPLEMENTATION
+#include "dr_mp3.h"
+
 static struct {
   char *key;
   struct wav *value;
 } *wavhash = NULL;
 
 static struct wav change_channels(struct wav w, int ch) {
-  short *data = w.data;
+  soundbyte *data = w.data;
   int samples = ch * w.frames;
-  short *new = malloc(sizeof(short) * samples);
+  soundbyte *new = malloc(sizeof(soundbyte) * samples);
 
   if (ch > w.ch) {
     /* Sets all new channels equal to the first one */
@@ -57,23 +66,24 @@ 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];
+  soundbyte *resampled = calloc(w.ch*outframes,sizeof(soundbyte));
-  src_short_to_float_array(w.data, floatdata, w.frames * w.ch);
-  float resampled[w.ch * outframes];
 
-  ssrc.data_in = floatdata;
+  ssrc.data_in = w.data;
   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);
+  int err = src_simple(&ssrc, SRC_LINEAR, w.ch);
-
+  if (err) {
-  short *newdata = malloc(sizeof(short) * outframes * w.ch);
+    YughError("Resampling error code %d: %s", err, src_strerror(err));
-  src_float_to_short_array(resampled, newdata, outframes * w.ch);
+    free(resampled);
-
+    return w;
+  }
+  
   free(w.data);
-  w.data = newdata;
+  w.data = resampled;
+  w.frames = outframes;
   w.samplerate = rate;
 
   return w;
@@ -98,7 +108,7 @@ void wav_norm_gain(struct wav *w, double lv) {
   }
 }
 
-void push_sound(float *buffer, int frames, int chan)
+void push_sound(soundbyte *buffer, int frames, int chan)
 {
   bus_fill_buffers(buffer, frames*chan);
 }
@@ -115,30 +125,52 @@ void sound_init() {
 
 struct wav *make_sound(const char *wav) {
   int index = shgeti(wavhash, wav);
-  if (index != -1) return wavhash[index].value;
+  if (index != -1) {
+    YughWarn("%s was cached ...", wav);
+    return wavhash[index].value;
+  }
+  char *ext = strrchr(wav, '.')+1;
+  
+  if(!ext) {
+    YughWarn("No extension detected for %s.", wav);
+    return NULL;
+  }
 
   struct wav mwav;
-  //    mwav.data = drwav_open_file_and_read_pcm_frames_s16(wav, &mwav.ch, &mwav.samplerate, &mwav.frames, NULL);
+
+  if (!strcmp(ext, "wav")) {
+    mwav.data = drwav_open_file_and_read_pcm_frames_f32(wav, &mwav.ch, &mwav.samplerate, &mwav.frames, NULL);
+  } else if (!strcmp(ext, "flac")) {
+    mwav.data = drflac_open_file_and_read_pcm_frames_f32(wav, &mwav.ch, &mwav.samplerate, &mwav.frames, NULL);
+    YughWarn("Flac opened with %d ch, %d samplerate", mwav.ch, mwav.samplerate);
+  } else if (!strcmp(ext, "mp3")) {
+    drmp3_config cnf;
+    mwav.data = drmp3_open_file_and_read_pcm_frames_f32(wav, &cnf, &mwav.frames, NULL);
+    mwav.ch = cnf.channels;
+    mwav.samplerate = cnf.sampleRate;
+  } else {
+    YughWarn("Cannot process file type '%s'.", ext);
+    return NULL;
+  }
 
   if (mwav.samplerate != SAMPLERATE) {
-    YughInfo("Changing samplerate of %s from %d to %d.", wav, mwav.samplerate, SAMPLERATE);
+    YughWarn("Changing samplerate of %s from %d to %d.", wav, mwav.samplerate, SAMPLERATE);
-    //        mwav = change_samplerate(mwav, SAMPLERATE);
+    mwav = change_samplerate(mwav, SAMPLERATE);
   }
 
   if (mwav.ch != CHANNELS) {
-    YughInfo("Changing channels of %s from %d to %d.", wav, mwav.ch, CHANNELS);
+    YughWarn("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);
 
+  YughWarn("Channels %d, sr %d", newwav->ch,newwav->samplerate);
+  
   return newwav;
 }
 
@@ -153,37 +185,10 @@ void free_sound(const char *wav) {
 
 struct soundstream *soundstream_make() {
   struct soundstream *new = malloc(sizeof(*new));
-//  new->buf = circbuf_make(sizeof(short), BUF_FRAMES * CHANNELS * 2);
+  new->buf = circbuf_make(sizeof(short), BUF_FRAMES * CHANNELS * 2);
   return new;
 }
 
-void mini_sound(char *path) {
-  
-  //ma_engine_play_sound(engine, path, NULL);
-}
-
-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("Could not load music at path: %s", path);
-  }
-
-  YughInfo("Loading %s...", path);
-  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);
@@ -257,16 +262,16 @@ int open_device(const char *adriver) {
   return 0;
 }
 
-void sound_fillbuf(struct sound *s, short *buf, int n) {
+void sound_fillbuf(struct sound *s, soundbyte *buf, int n) {
   float gainmult = pct2mult(s->data->gain);
 
-  short *in = s->data->data;
+  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 + j] * gainmult;
+      buf[i * CHANNELS + j] = in[s->frame*CHANNELS + j] * gainmult;
     s->frame++;
+    
     if (s->frame == s->data->frames) {
-
       bus_free(s->bus);
       s->bus = NULL;
       s->endcb(s);
@@ -275,10 +280,10 @@ void sound_fillbuf(struct sound *s, short *buf, int n) {
   }
 }
 
-void mp3_fillbuf(struct sound *s, short *buf, int n) {
+void mp3_fillbuf(struct sound *s, soundbyte *buf, int n) {
 }
 
-void soundstream_fillbuf(struct soundstream *s, short *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++) {

source/engine/sound.h

@@ -1,7 +1,7 @@
 #ifndef SOUND_H
 #define SOUND_H
 
-struct circbuf;
+#include "cbuf.h"
 
 typedef float soundbyte;
 
@@ -25,7 +25,7 @@ struct soundstream {
 
 struct soundstream *soundstream_make();
 
-/* A playing sound */
+/* A bookmark into a wav, actually playing the sound */
 struct sound {
     int loop; /* How many times to loop */
     unsigned int frame; /* Pointing to the current frame on the wav */
@@ -45,7 +45,7 @@ struct wav {
     unsigned long long frames;
     float gain; /* In dB */
 
-    void *data;
+    soundbyte *data;
 };
 
 struct mp3 {
@@ -56,7 +56,7 @@ void sound_init();
 void audio_open(const char *device);
 void audio_close();
 
-void sound_fillbuf(struct sound *s, short *buf, int n);
+void sound_fillbuf(struct sound *s, soundbyte *buf, int n);
 
 void mini_sound(char *path);
 void mini_master(float v);
@@ -82,7 +82,7 @@ struct mp3 make_mp3(const char *mp3);
 
 const char *get_audio_driver();
 
-void soundstream_fillbuf(struct soundstream *stream, short *buf, int n);
+void soundstream_fillbuf(struct soundstream *stream, soundbyte *buf, int n);
 
 void close_audio_device(int device);
 int open_device(const char *adriver);

source/engine/sound/dsp.c

@@ -12,7 +12,7 @@
 
 struct dsp_filter *filters;
 
-struct dsp_filter make_dsp(void *data, void (*in)(void *data, short *out, int n)) {
+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;
@@ -24,7 +24,7 @@ struct dsp_filter make_dsp(void *data, void (*in)(void *data, short *out, int n)
     }
 }
 
-void dsp_run(struct dsp_filter filter, short *out, int n) {
+void dsp_run(struct dsp_filter filter, soundbyte *out, int n) {
     filter.dirty = 1; // Always on for testing
 
     if (!filter.dirty)
@@ -45,13 +45,13 @@ void dsp_filter_addin(struct dsp_filter filter, struct dsp_filter *in)
     filter.in[filter.inputs++] = in;
 }
 
-void am_mod(struct dsp_ammod *mod, short *c, int n)
+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++)
+//    for (int i = 0; i < n*CHANNELS; i++)
-        c[i] = (mod->abuf[i]*mod->bbuf[i])>>15;
+//        c[i] = (mod->abuf[i]*mod->bbuf[i])>>15;
 }
 
 void fm_mod(float *in1, float *in2, float *out, int n)
@@ -64,7 +64,7 @@ static struct wav make_wav(float freq, int sr, int ch) {
     new.ch = ch;
     new.samplerate = sr;
     new.frames = sr/freq;
-    new.data = calloc(new.frames*new.ch, sizeof(short));
+    new.data = calloc(new.frames*new.ch, sizeof(soundbyte));
 
     return new;
 }
@@ -75,12 +75,12 @@ struct wav gen_sine(float amp, float freq, int sr, int ch)
 
     if (amp > 1) amp = 1;
     if (amp < 0) amp = 0;
-    short samp = amp*SHRT_MAX;
+    soundbyte samp = amp*SHRT_MAX;
 
-    short *data = (short*)new.data;
+    soundbyte *data = (soundbyte*)new.data;
 
     for (int i = 0; i < new.frames; i++) {
-        short val = samp * sin(2*PI*((float)i / new.frames));
+        soundbyte val = samp * sin(2*PI*((float)i / new.frames));
 
         for (int j = 0; j < new.ch; j++) {
             data[i*new.ch+j] = val;
@@ -101,12 +101,12 @@ struct wav gen_square(float amp, float freq, int sr, int ch)
     if (amp > 1) amp = 1;
     if (amp < 0) amp = 0;
 
-    short samp = amp * SHRT_MAX;
+    soundbyte samp = amp * SHRT_MAX;
 
-    short *data = (short*)new.data;
+    soundbyte *data = (soundbyte*)new.data;
 
     for (int i = 0; i < new.frames; i++) {
-        short val = -2 * floor(2 * i / new.frames) + 1;
+        soundbyte val = -2 * floor(2 * i / new.frames) + 1;
         for (int j = 0; j < new.ch; j++) {
             data[i*new.frames+j] = val;
         }
@@ -122,10 +122,10 @@ struct wav gen_triangle(float amp, float freq, int sr, int ch)
     if (amp > 1) amp = 1;
     if (amp < 0) amp = 0;
 
-    short *data = (short*)new.data;
+    soundbyte *data = (soundbyte*)new.data;
 
     for (int i = 0; i < new.frames; i++) {
-        short val = 2 * abs( (i/new.frames) - floor( (i/new.frames) + 0.5));
+        soundbyte val = 2 * abs( (i/new.frames) - floor( (i/new.frames) + 0.5));
         for (int j = 0; j < new.ch; j++) {
             data[i+j] = val;
         }
@@ -141,12 +141,12 @@ struct wav gen_saw(float amp, float freq, int sr, int ch)
     if (amp > 1) amp = 1;
     if (amp < 0) amp = 0;
 
-    short samp = amp*SHRT_MAX;
+    soundbyte samp = amp*SHRT_MAX;
 
-    short *data = (short*)new.data;
+    soundbyte *data = (soundbyte*)new.data;
 
     for (int i = 0; i < new.frames; i++) {
-        short val = samp * 2 * i/sr - samp;
+        soundbyte val = samp * 2 * i/sr - samp;
         for (int j = 0; j < new.ch; j++) {
             data[i+j] = val;
         }
@@ -155,7 +155,7 @@ struct wav gen_saw(float amp, float freq, int sr, int ch)
     return new;
 }
 
-struct dsp_filter dsp_filter(void *data, void (*filter)(void *data, short *out, int samples))
+struct dsp_filter dsp_filter(void *data, void (*filter)(void *data, soundbyte *out, int samples))
 {
     struct dsp_filter new;
     new.data = data;
@@ -164,7 +164,7 @@ struct dsp_filter dsp_filter(void *data, void (*filter)(void *data, short *out,
     return new;
 }
 
-void dsp_rectify(short *in, short *out, int n)
+void dsp_rectify(soundbyte *in, soundbyte *out, int n)
 {
     for (int i = 0; i < n; i++)
         out[i] = abs(in[i]);
@@ -215,15 +215,15 @@ float tri_phasor(float p)
     return 4*(p * 0.5f ? p : (1-p)) - 1;
 }
 
-void osc_fillbuf(struct osc *osc, short *buf, int n)
+void osc_fillbuf(struct osc *osc, soundbyte *buf, int n)
 {
     for (int i = 0; i < n; i++) {
-        short val = SHRT_MAX * osc->f(phasor_step(&osc->p));
+        soundbyte val = SHRT_MAX * osc->f(phasor_step(&osc->p));
         buf[i*CHANNELS] = buf[i*CHANNELS+1] = val;
     }
 }
 
-void gen_whitenoise(void *data, short *out, int n)
+void gen_whitenoise(void *data, soundbyte *out, int n)
 {
     for (int i = 0; i < n; i++) {
         for (int j = 0; j < CHANNELS; j++) {
@@ -232,7 +232,7 @@ void gen_whitenoise(void *data, short *out, int n)
     }
 }
 
-void gen_pinknoise(void *data, short *out, int n)
+void gen_pinknoise(void *data, soundbyte *out, int n)
 {
     gen_whitenoise(NULL, out, n);
 
@@ -270,7 +270,7 @@ void gen_pinknoise(void *data, short *out, int n)
     */
 }
 
-short iir_filter(struct dsp_iir *miir, short val)
+soundbyte iir_filter(struct dsp_iir *miir, soundbyte val)
 {
     struct dsp_iir iir = *miir;
 
@@ -297,12 +297,12 @@ short iir_filter(struct dsp_iir *miir, short val)
     return a * SHRT_MAX;
 }
 
-void dsp_iir_fillbuf(struct dsp_iir *iir, short *out, int n)
+void dsp_iir_fillbuf(struct dsp_iir *iir, soundbyte *out, int n)
 {
     dsp_run(iir->in, out, n);
 
     for (int i = 0; i < n; i++) {
-        short v = iir_filter(iir, out[i*CHANNELS]);
+        soundbyte v = iir_filter(iir, out[i*CHANNELS]);
 
         for (int j = 0; j < CHANNELS; j++) {
            out[i*CHANNELS+j] = v;
@@ -369,7 +369,7 @@ struct dsp_filter hpf_make(int poles, float freq)
   return hpf;
 }
 
-short fir_filter(struct dsp_fir *fir, short val)
+soundbyte fir_filter(struct dsp_fir *fir, soundbyte val)
 {
     float ret = 0.f;
     fir->dx[fir->head] = (float)val/SHRT_MAX;
@@ -383,12 +383,12 @@ short fir_filter(struct dsp_fir *fir, short val)
     return ret * SHRT_MAX;
 }
 
-void dsp_fir_fillbuf(struct dsp_fir *fir, short *out, int n)
+void dsp_fir_fillbuf(struct dsp_fir *fir, soundbyte *out, int n)
 {
     dsp_run(fir->in, out, n);
 
     for (int i = 0; i < n; i++) {
-        short val = fir_filter(fir, out[i*CHANNELS]);
+        soundbyte val = fir_filter(fir, out[i*CHANNELS]);
 
         for (int j = 0; j < CHANNELS; j++)
             out[i*CHANNELS + j] = val*5;
@@ -426,7 +426,7 @@ struct dsp_delay dsp_delay_make(unsigned int ms_delay)
 
     /* Circular buffer size is enough to have the delay */
     unsigned int datasize = ms_delay * CHANNELS * (SAMPLERATE / 1000);
-//    new.buf = circbuf_init(sizeof(short), datasize);
+//    new.buf = circbuf_init(sizeof(soundbyte), datasize);
 //    new.buf.write = datasize;
 
 //    YughInfo("Buffer size is %u.", new.buf.len);
@@ -434,9 +434,9 @@ struct dsp_delay dsp_delay_make(unsigned int ms_delay)
     return new;
 }
 
-void dsp_delay_filbuf(struct dsp_delay *delay, short *buf, int n)
+void dsp_delay_filbuf(struct dsp_delay *delay, soundbyte *buf, int n)
 {
-    static short cache[BUF_FRAMES*2];
+    static soundbyte cache[BUF_FRAMES*2];
     dsp_run(delay->in, cache, n);
 
     for (int i = 0; i < n*CHANNELS; i++) {
@@ -452,9 +452,9 @@ double tau2pole(double tau)
     return exp(-1/(tau*SAMPLERATE));
 }
 
-void dsp_adsr_fillbuf(struct dsp_adsr *adsr, short *out, int n)
+void dsp_adsr_fillbuf(struct dsp_adsr *adsr, soundbyte *out, int n)
 {
-    short val;
+    soundbyte val;
 
     for (int i = 0; i < n; i++) {
         if (adsr->time > adsr->rls) {
@@ -524,7 +524,7 @@ struct dsp_filter make_reverb()
 
 }
 
-void dsp_reverb_fillbuf(struct dsp_reverb *r, short *out, int n)
+void dsp_reverb_fillbuf(struct dsp_reverb *r, soundbyte *out, int n)
 {
 
 }
@@ -551,7 +551,7 @@ struct dsp_filter dsp_make_compressor()
     return filter;
 }
 
-void dsp_compressor_fillbuf(struct dsp_compressor *comp, short *out, int n)
+void dsp_compressor_fillbuf(struct dsp_compressor *comp, soundbyte *out, int n)
 {
     float val;
     float db;
@@ -578,7 +578,7 @@ void dsp_compressor_fillbuf(struct dsp_compressor *comp, short *out, int n)
     }
 }
 
-void dsp_pan(float *deg, short *out, int n)
+void dsp_pan(float *deg, soundbyte *out, int n)
 {
     if (*deg < -100) *deg = -100.f;
     else if (*deg > 100) *deg = 100.f;
@@ -599,8 +599,8 @@ void dsp_pan(float *deg, short *out, int n)
 
     for (int i = 0; i < n; i++) {
         double pct = *deg / 100.f;
-        short L = out[i*CHANNELS];
+        soundbyte L = out[i*CHANNELS];
-        short R = out[i*CHANNELS +1];
+        soundbyte R = out[i*CHANNELS +1];
 
         if (*deg > 0) {
             out[i*CHANNELS] = short_gain(L, db1);
@@ -614,20 +614,22 @@ void dsp_pan(float *deg, short *out, int n)
     }
 }
 
-void dsp_mono(void *p, short *out, int n)
+void dsp_mono(void *p, soundbyte *out, int n)
 {
     for (int i = 0; i < n; i++) {
-        short val = (out[i*CHANNELS] + out[i*CHANNELS+1]) / 2;
+        soundbyte val = (out[i*CHANNELS] + out[i*CHANNELS+1]) / 2;
 
         for (int j = 0; j < CHANNELS; j++)
             out[i*CHANNELS+j] = val;
     }
 }
 
-void dsp_bitcrush(void *p, short *out, int n)
+void dsp_bitcrush(void *p, soundbyte *out, int n)
 {
-    for (int i = 0; i < n; i++) {
+
-        for (int j = 0; j < CHANNELS; j++)
+//    for (int i = 0; i < n; i++) {
-            out[i*CHANNELS+j] = (out[i*CHANNELS+j] | 0xFF); /* Mask out the lower 8 bits */
+//        for (int j = 0; j < CHANNELS; j++)
-    }
+//            out[i*CHANNELS+j] = (out[i*CHANNELS+j] | 0xFF); /* Mask out the lower 8 bits */
+//    }
+
 }

source/engine/sound/dsp.h

@@ -5,25 +5,27 @@
 #define BUF_FRAMES 128 /* At 48k, 128 needed for 240fps consistency */
 #define CHANNELS 2
 
+#include "sound.h"
+
 //#include "circbuf.h"
 
 struct dsp_iir;
 
 
-void dsp_rectify(short *in, short *out, int n);
+void dsp_rectify(soundbyte *in, soundbyte *out, int n);
 
 struct dsp_filter {
-    void (*filter)(void *data, short *out, int samples);
+    void (*filter)(void *data, soundbyte *out, int samples);
     void *data;
 
     int inputs;
     struct dsp_filter *in[6];
 
-    short cache[CHANNELS*BUF_FRAMES];
+    soundbyte cache[CHANNELS*BUF_FRAMES];
     int dirty;
 };
 
-struct dsp_filter dsp_filter(void *data, void (*filter)(void *data, short *out, int samples));
+struct dsp_filter dsp_filter(void *data, void (*filter)(void *data, soundbyte *out, int samples));
 
 struct dsp_fir {
     float freq;
@@ -39,7 +41,7 @@ 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, short *out, int n);
+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);
@@ -60,7 +62,7 @@ struct dsp_adsr {
     float out;
 };
 
-void dsp_adsr_fillbuf(struct dsp_adsr *adsr, short *out, int n);
+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);
 
 struct dsp_delay {
@@ -70,13 +72,13 @@ struct dsp_delay {
 };
 
 struct dsp_delay dsp_delay_make(unsigned int ms_delay);
-void dsp_delay_filbuf(struct dsp_delay *delay, short *buf, int n);
+void dsp_delay_filbuf(struct dsp_delay *delay, soundbyte *buf, int n);
 
 struct dsp_ammod {
     struct dsp_filter ina;
     struct dsp_filter inb;
-    short abuf[BUF_FRAMES*CHANNELS];
+    soundbyte abuf[BUF_FRAMES*CHANNELS];
-    short bbuf[BUF_FRAMES*CHANNELS];
+    soundbyte bbuf[BUF_FRAMES*CHANNELS];
 };
 
 struct dsp_compressor {
@@ -90,14 +92,14 @@ struct dsp_compressor {
 };
 
 struct dsp_filter dsp_make_compressor();
-void dsp_compressor_fillbuf(struct dsp_compressor *comp, short *out, int n);
+void dsp_compressor_fillbuf(struct dsp_compressor *comp, soundbyte *out, int n);
 
 struct dsp_limiter {
 
 };
 
 struct dsp_filter dsp_make_limiter();
-void dsp_limiter_fillbuf(struct dsp_limiter *lim, short *out, int n);
+void dsp_limiter_fillbuf(struct dsp_limiter *lim, soundbyte *out, int n);
 
 
 struct phasor {
@@ -116,7 +118,7 @@ struct osc {
     struct phasor p;
     unsigned int frames;
     unsigned int frame;
-    short *cache;
+    soundbyte *cache;
 };
 
 struct wav;
@@ -126,8 +128,8 @@ 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, short *out, int n);
+void gen_whitenoise(void *data, soundbyte *out, int n);
-void gen_pinknoise(void *data, short *out, int n);
+void gen_pinknoise(void *data, soundbyte *out, int n);
 
 
 float sin_phasor(float p);
@@ -135,23 +137,23 @@ float square_phasor(float p);
 float saw_phasor(float p);
 float tri_phasor(float p);
 
-void osc_fillbuf(struct osc *osc, short *buf, int n);
+void osc_fillbuf(struct osc *osc, soundbyte *buf, int n);
 
-void am_mod(struct dsp_ammod *mod, short *c, int n);
+void am_mod(struct dsp_ammod *mod, soundbyte *c, int n);
 
 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, short *out, int n);
+void dsp_reverb_fillbuf(struct dsp_reverb *r, soundbyte *out, int n);
 
-void dsp_pan(float *deg, short *out, int n);
+void dsp_pan(float *deg, soundbyte *out, int n);
 
-void dsp_mono(void *p, short *out, int n);
+void dsp_mono(void *p, soundbyte *out, int n);
 
-void dsp_bitcrush(void *p, short *out, int n);
+void dsp_bitcrush(void *p, soundbyte *out, int n);
 
-void dsp_run(struct dsp_filter filter, short *out, int n);
+void dsp_run(struct dsp_filter filter, soundbyte *out, int n);
 
 #endif

source/engine/thirdparty/libsamplerate/src/config.h

@@ -17,13 +17,13 @@
 #define CPU_IS_LITTLE_ENDIAN 1
 
 /* Enable sinc best converter. */
-#define ENABLE_SINC_BEST_CONVERTER yes
+#define ENABLE_SINC_BEST_CONVERTER 1
 
 /* Enable sinc fast converter. */
-#define ENABLE_SINC_FAST_CONVERTER yes
+#define ENABLE_SINC_FAST_CONVERTER 1
 
 /* Enable sinc balanced converter. */
-#define ENABLE_SINC_MEDIUM_CONVERTER yes
+#define ENABLE_SINC_MEDIUM_CONVERTER 1
 
 /* Define to 1 if you have the `alarm' function. */
 #define HAVE_ALARM 1

source/engine/thirdparty/libsamplerate/src/samplerate.c

@@ -6,9 +6,7 @@
 ** file at : https://github.com/libsndfile/libsamplerate/blob/master/COPYING
 */
 
-#ifdef HAVE_CONFIG_H
 #include	"config.h"
-#endif
 
 #include	<stdio.h>
 #include	<stdlib.h>

source/engine/thirdparty/libsamplerate/src/src_linear.c

@@ -6,9 +6,7 @@
 ** file at : https://github.com/libsndfile/libsamplerate/blob/master/COPYING
 */
 
-#ifdef HAVE_CONFIG_H
 #include "config.h"
-#endif
 
 #include <assert.h>
 #include <stdio.h>

source/engine/thirdparty/libsamplerate/src/src_sinc.c

@@ -6,9 +6,7 @@
 ** file at : https://github.com/libsndfile/libsamplerate/blob/master/COPYING
 */
 
-#ifdef HAVE_CONFIG_H
 #include "config.h"
-#endif
 
 #include <assert.h>
 #include <stdio.h>

source/engine/thirdparty/libsamplerate/src/src_zoh.c

@@ -6,9 +6,7 @@
 ** file at : https://github.com/libsndfile/libsamplerate/blob/master/COPYING
 */
 
-#ifdef HAVE_CONFIG_H
 #include "config.h"
-#endif
 
 #include <assert.h>
 #include <stdio.h>

source/engine/yugine.c

@@ -8,8 +8,7 @@
 #include "openglrender.h"
 #include "window.h"
 
-#define MINIAUDIO_IMPLEMENTATION
+#include "datastream.h"
-#include "miniaudio.h"
 
 #include "timer.h"
 
@@ -128,7 +127,7 @@ int main(int argc, char **args) {
   int logout = 1;
 
   script_startup();
-
+  
   logout = 0;
 
   for (int i = 1; i < argc; i++) {
@@ -224,7 +223,10 @@ int main(int argc, char **args) {
 
   input_init();
   openglInit();
-
+  
+  MakeDatastream();
+  struct datastream v;
+  ds_openvideo(&v, "bjork.mpg", NULL);
 
   while (!want_quit()) {
     double elapsed = glfwGetTime() - lastTick;

source/scripts/engine.js

@@ -725,7 +725,7 @@ var Music = {
 };
 
 var Sound = {
-  sounds: [],
+  sounds: [], /* array of loaded sound files */
   play(file) {
 //    var s = Object.create(Sound);
 //    s.path = file;