node free; pitch shift; time scale

Makefile

@@ -76,7 +76,10 @@ else
   endif
 endif
 
-CPPFLAGS += -DHAVE_CEIL -DCP_USE_CGTYPES=0 -DCP_USE_DOUBLES=0 -DTINYSPLINE_FLOAT_PRECISION -DHAVE_FLOOR -DHAVE_FMOD -DHAVE_LRINT -DHAVE_LRINTF $(includeflag) -MD $(WARNING_FLAGS) -I. -DVER=\"$(VER)\" -DINFO=\"$(INFO)\"
+CPPFLAGS += -DHAVE_CEIL -DCP_USE_CGTYPES=0 -DCP_USE_DOUBLES=0 -DTINYSPLINE_FLOAT_PRECISION -DHAVE_FLOOR -DHAVE_FMOD -DHAVE_LRINT -DHAVE_LRINTF $(includeflag) -MD $(WARNING_FLAGS) -I. -DVER=\"$(VER)\" -DINFO=\"$(INFO)\" -DENABLE_SINC_MEDIUM_CONVERTER -DENABLE_SINC_FAST_CONVERTER
+
+# ENABLE_SINC_[BEST|FAST|MEDIUM]_CONVERTER
+# default, fast and medium available in game at runtime; best available in editor
 
 PKGCMD = tar --directory $(BIN) --exclude="./*.a" --exclude="./obj" -czf $(DISTDIR)/$(DIST) .
 ZIP = .tar.gz

docs/sound.md

@@ -0,0 +1,30 @@
+# SOUND
+
+Primum's sound system is well integrated and robust. It comes with flexibility in mind, to enable sounds to easily change as the result of occuring gameplay.
+
+At the highest level, audio sources feed into a series of nodes, which eventually output to the mixer. All DSP is handled as floating point.
+
+The game should specify a SAMPLERATE, BUF_FRAMES, and CHANNELS.
+
+SAMPLERATE: samples per second
+CHANNELS: number of output channels.
+BUF_FRAMES: Number of frames to keep in the buffer. Smaller is lower latency. Too small might mean choppy audio.
+
+## Audio sources
+Audio sources include wav files, midi files, mod files, etc.
+
+## Audio instances
+Audio sources can be played multiple times after loaded into the game. Each instance 
+
+## DSPs / nodes
+Nodes are how audio ends up coming out of the player's speakers. Only one node is specified at runtime: the master node, which outputs its inputs to the computers' speakers. It has 256 inputs.
+
+Each node can have any number of inputs to it, defined when the node is created.
+
+Nodes are run starting with the master node, in a pull fashion. The master node checks for inputs, and from each input pulls audio foward.
+
+Audio instances are a type of node with a single output, and can be fed directly into the master node.
+
+A node has inputs and outputs. All inputs into a node are mixed together, and then processed by the node. The node might clip them, apply a filter, or do any other number of things.
+
+## Scripting

scripts/sound.js

@@ -7,9 +7,11 @@ var dsp_node = {
   
   id: undefined,
   get volume() { return cmd(175, this.id); },
-  set volume(x) { cmd(176, this.id,x); },
+  set volume(x) { x = Math.clamp(x,0,1); cmd(176, this.id,x); },
+  get db() { return 20*Math.log10(Math.abs(this.volume)); },
+  set db(x) { x = Math.clamp(x,-100,0); this.volume = Math.pow(10, x/20); },
   get pan() { return cmd(178,this.id); },
-  set pan(x) { cmd(179,this.id,x); },
+  set pan(x) { x = Math.clamp(x,-1,1); 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); },
@@ -26,6 +28,8 @@ var dsp_source = Object.copy(dsp_node,{
   end(){},
   get loop() { return cmd(194,this.id); },
   set loop(x) { cmd(195,this.id, x);},
+  get timescale() { return cmd(201,this.id); },
+  set timescale(x) { cmd(202,this.id,x); },
   get frame() { return cmd(196,this.id); },
   set frame(x) { cmd(199, this.id, x); },
   frames() { return cmd(197,this.id); },
@@ -68,6 +72,14 @@ var DSP = {
   delay(secs,decay) {
     return dsp_node.make(cmd(185, secs, decay));
   },
+  fwd_delay(secs, decay) {
+    return dsp_node.make(cmd(207,secs,decay));
+  },
+  allpass(secs, decay) {
+    var fwd = DSP.fwd_delay(secs,-decay);
+    var fbk = DSP.delay(secs,decay);
+    fwd.plugin(fbk);
+  },
   lpf(f) {
     return dsp_node.make(cmd(186,f));
   },
@@ -77,6 +89,9 @@ var DSP = {
   mod(path) {
     return dsp_node.make(cmd(188,path));
   },
+  midi(midi,sf) {
+    return dsp_node.make(cmd(206,midi,sf));
+  },
   crush(rate, depth) {
     return dsp_node.make(cmd(189,rate,depth));
   },
@@ -89,6 +104,18 @@ var DSP = {
   noise_gate(floor) {
     return dsp_node.make(cmd(192,floor));
   },
+  pitchshift(octaves) {
+    return dsp_node.make(cmd(200,octaves));
+  },
+  noise() {
+    return dsp_node.make(cmd(203));
+  },
+  pink() {
+    return dsp_node.make(cmd(204));
+  },
+  red() {
+    return dsp_node.make(cmd(205));
+  },
 };
 
 Sound.bus.master = dsp_node.make(cmd(180));

source/engine/datastream.c

@@ -75,7 +75,7 @@ struct datastream *ds_openvideo(const char *path)
   });  
 
   ds->ring = ringnew(ds->ring, 8192);
-  plugin_node(make_node(ds, soundstream_fillbuf), masterbus);
+  plugin_node(make_node(ds, soundstream_fillbuf, NULL), masterbus);
 
   plm_set_video_decode_callback(ds->plm, render_frame, ds);
   plm_set_audio_decode_callback(ds->plm, render_audio, ds);

source/engine/jsffi.c

@@ -1218,7 +1218,7 @@ JSValue duk_cmd(JSContext *js, JSValueConst this, int argc, JSValueConst *argv)
       ret = ptr2js(masterbus);
       break;
     case 181:
-      ret = ptr2js(make_node(NULL,NULL));
+      ret = ptr2js(make_node(NULL,NULL,NULL));
       break;
     case 182:
       str = js2str(argv[1]);
@@ -1277,6 +1277,33 @@ JSValue duk_cmd(JSContext *js, JSValueConst this, int argc, JSValueConst *argv)
     case 199:
       ((sound*)((dsp_node*)js2ptr(argv[1]))->data)->frame = js2number(argv[2]);
       break;
+    case 200:
+      ret = ptr2js(dsp_pitchshift(js2number(argv[1])));
+      break;
+    case 201:
+      ret = num2js(((sound*)((dsp_node*)js2ptr(argv[1]))->data)->timescale);
+      break;
+    case 202:
+      YughWarn("%g", js2number(argv[2]));
+      ((sound*)((dsp_node*)js2ptr(argv[1]))->data)->timescale = js2number(argv[2]);
+      break;
+    case 203:
+      ret = ptr2js(dsp_whitenoise());
+      break;
+    case 204:
+      ret = ptr2js(dsp_pinknoise());
+      break;
+    case 205:
+      ret = ptr2js(dsp_rednoise());
+      break;
+    case 206:
+      str = js2str(argv[1]);
+      str2 = js2str(argv[2]);
+      ret = ptr2js(dsp_midi(str, make_soundfont(str2)));
+      break;
+    case 207:
+      ret = ptr2js(dsp_fwd_delay(js2number(argv[1]), js2number(argv[2])));
+      break;
   }
 
   if (str)

source/engine/sound/dsp.c

@@ -7,11 +7,22 @@
 #include "iir.h"
 #include "log.h"
 #include "stb_ds.h"
+#include "smbPitchShift.h"
 
 #define PI 3.14159265
 
 dsp_node *masterbus = NULL;
 
+void iir_free(struct dsp_iir *iir)
+{
+  free(iir->a);
+  free(iir->b);
+  free(iir->x);
+  free(iir->y);
+  free(iir);
+}
+
+
 void interleave(soundbyte *a, soundbyte *b, soundbyte *stereo, int frames)
 {
   for (int i = 0; i < frames; i++) {
@@ -20,6 +31,13 @@ void interleave(soundbyte *a, soundbyte *b, soundbyte *stereo, int frames)
   }
 }
 
+void deinterleave(soundbyte *stereo, soundbyte *out, int frames, int channels, int chout)
+{
+  chout--;
+  for (int i = 0; i < frames; i++)
+    out[i] = stereo[i*channels+chout];
+}
+
 void mono_to_stero(soundbyte *a, soundbyte *stereo, int frames)
 {
   interleave(a,a,stereo, frames);
@@ -37,7 +55,7 @@ void mono_expand(soundbyte *buffer, int to, int frames)
 
 dsp_node *dsp_mixer_node()
 {
-  return make_node(NULL, NULL);
+  return make_node(NULL, NULL, NULL);
 }
 
 void dsp_init()
@@ -75,7 +93,7 @@ void filter_am_mod(dsp_node *mod, soundbyte *buffer, int frames)
 
 dsp_node *dsp_am_mod(dsp_node *mod)
 {
-  return make_node(mod, filter_am_mod);
+  return make_node(mod, filter_am_mod, node_free);
 }
 
 /* Add b into a */
@@ -120,8 +138,7 @@ void dsp_node_run(dsp_node *node)
   }
 }
 
-
+dsp_node *make_node(void *data, void (*proc)(void *data, soundbyte *out, int samples), void (*fr)(void *data))
-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));
@@ -206,7 +223,7 @@ dsp_node *dsp_phasor(float amp, float freq, float (*filter)(float))
   p->freq = freq;
   p->phase = 0;
   p->filter = filter;
-  return make_node(p, filter_phasor);
+  return make_node(p, filter_phasor, NULL);
 }
 
 void filter_rectify(void *data, soundbyte *out, int n)
@@ -216,7 +233,7 @@ void filter_rectify(void *data, soundbyte *out, int n)
 
 dsp_node *dsp_rectify()
 {
-  return make_node(NULL, filter_rectify);
+  return make_node(NULL, filter_rectify, NULL);
 }
 
 soundbyte sample_whitenoise()
@@ -232,32 +249,26 @@ void gen_whitenoise(void *data, soundbyte *out, int n)
 
 dsp_node *dsp_whitenoise()
 {
-  return make_node(NULL, gen_whitenoise);
+  return make_node(NULL, gen_whitenoise, NULL);
 }
 
-void gen_pinknoise(void *data, soundbyte *out, int n)
+void gen_pinknoise(struct dsp_iir *pink, soundbyte *out, int n)
 {
-  double a[7] = {1.0, 0.0555179, 0.0750759, 0.1538520, 0.3104856, 0.5329522, 0.0168980};
+  gen_whitenoise(NULL, out, n);
-  double b[7] = {0.99886, 0.99332, 0.969, 0.8665, 0.55, -0.7616, 0.115926};  
 
-  for (int i = 0; i < n; i++) {
+  for (int i = 0; i < n*CHANNELS; i++) {
-      double pink;
+    soundbyte sum = 0;
-      double white = sample_whitenoise();
+    for (int j = 0; j < 6; j++) {
+      pink->x[j] = pink->x[j]*pink->b[j] + out[i]*pink->a[j];
+      sum += pink->x[j];
+    }
+    pink->x[6] = out[i] * 0.115926;
     
-      for (int k = 0; k < 5; k++) {
+    out[i] = sum + out[i] * 0.5362 + pink->x[6];
-        b[k] = a[k]*b[k] + white * b[k];
+    out[i] *= 0.11;
-        pink += b[k];
   }
 
-      pink += b[5] + white*0.5362;
+    /* *  https://www.firstpr.com.au/dsp/pink-noise/
-      b[5] = white*0.115926;
-
-      out[i] = pink;
-    }
-  mono_expand(out,CHANNELS,n);
-    /*
-      *  The above is a loopified version of this
-      *  https://www.firstpr.com.au/dsp/pink-noise/
     b0 = 0.99886 * b0 + white * 0.0555179;
     b1 = 0.99332 * b1 + white * 0.0750759;
     b2 = 0.96900 * b2 + white * 0.1538520;
@@ -271,7 +282,69 @@ void gen_pinknoise(void *data, soundbyte *out, int n)
 
 dsp_node *dsp_pinknoise()
 {
-  return make_node(NULL, gen_pinknoise);
+  struct dsp_iir *pink = malloc(sizeof(*pink));
+  *pink = make_iir(6);
+  float pinka[7] = {0.0555179, 0.0750759, 0.1538520, 0.3104856, 0.5329522, -0.0168980, 0.115926};  
+  float pinkb[7] = {0.99886, 0.99332, 0.969, 0.8665, 0.55, -0.7616, 0.115926};  
+  memcpy(pink->a, pinka, 7*sizeof(float));
+  memcpy(pink->b, pinkb, 7*sizeof(float));
+
+  return make_node(pink, gen_pinknoise, iir_free);
+}
+
+void filter_rednoise(soundbyte *last, soundbyte *out, int frames)
+{
+  gen_whitenoise(NULL, out, frames);
+  for (int i = 0; i < frames*CHANNELS; i++) {
+    out[i] = (*last + (0.02*out[i]))/1.02; 
+    *last = out[i];
+    out[i] *= 3.5;
+  }
+}
+
+dsp_node *dsp_rednoise()
+{
+  soundbyte *last = malloc(sizeof(soundbyte));
+  *last = 0;
+  return make_node(last,filter_rednoise, NULL);
+}
+
+void filter_pitchshift(float *octaves, soundbyte *buffer, int frames)
+{
+  soundbyte ch1[frames];
+  for (int i = 0; i < frames; i++)
+    ch1[i] = buffer[i*CHANNELS];
+  smbPitchShift(*octaves, frames, 1024, 4,  SAMPLERATE, ch1, buffer);
+  mono_expand(buffer, CHANNELS, frames);
+}
+
+dsp_node *dsp_pitchshift(float octaves)
+{
+  float *oct = malloc(sizeof(float));
+  *oct = octaves;
+  return make_node(oct, filter_pitchshift, NULL);
+}
+
+struct timescale
+{
+  float rate;
+  SRC_STATE *src;
+};
+
+static long *src_cb(struct timescale *ts, float **data)
+{
+}
+
+void filter_timescale(struct timescale *ts, soundbyte *buffer, int frames)
+{
+}
+
+dsp_node *dsp_timescale(float scale)
+{
+  struct timescale *ts = malloc(sizeof(*ts));
+  ts->rate = scale;
+  ts->src = src_callback_new(src_cb, SRC_SINC_FASTEST, scale, NULL, ts);
+  return make_node(ts, filter_timescale, NULL);
 }
 
 soundbyte iir_filter(struct dsp_iir iir, soundbyte val)
@@ -303,18 +376,19 @@ void filter_iir(struct dsp_iir *iir, soundbyte *buffer, int frames)
   }
 }
 
+
 dsp_node *dsp_lpf(float freq)
 {
   struct dsp_iir *iir = malloc(sizeof(*iir));
   *iir = bqlp_dcof(2*freq/SAMPLERATE, 5);
-  return make_node(iir, filter_iir);
+  return make_node(iir, filter_iir, iir_free);
 }
 
 dsp_node *dsp_hpf(float freq)
 {
   struct dsp_iir *iir = malloc(sizeof(*iir));
   *iir = bqhp_dcof(2*freq/SAMPLERATE,5);
-  return make_node(iir, filter_iir);
+  return make_node(iir, filter_iir, iir_free);
 }
 
 void filter_delay(delay *d, soundbyte *buf, int frames)
@@ -325,6 +399,12 @@ void filter_delay(delay *d, soundbyte *buf, int frames)
   }
 }
 
+void delay_free(delay *d)
+{
+  ringfree(d->ring);
+  free(d);
+}
+
 dsp_node *dsp_delay(double sec, double decay)
 {
   delay *d = malloc(sizeof(*d));
@@ -333,7 +413,27 @@ dsp_node *dsp_delay(double sec, double decay)
   d->ring = NULL;
   d->ring = ringnew(d->ring, sec*CHANNELS*SAMPLERATE*2);    /* Circular buffer size is enough to have the delay */
   ringheader(d->ring)->write += CHANNELS*SAMPLERATE*sec;
-  return make_node(d, filter_delay);
+  return make_node(d, filter_delay, delay_free);
+}
+
+void filter_fwd_delay(delay *d, soundbyte *buf, int frames)
+{
+  for (int i = 0; i < frames*CHANNELS; i++) {
+    ringpush(d->ring, buf[i]);  
+    buf[i] += ringshift(d->ring)*d->decay;
+  }
+}
+
+dsp_node *dsp_fwd_delay(double sec, double decay)
+{
+  delay *d = malloc(sizeof(*d));
+  d-> ms_delay = sec;
+  d->decay = decay;
+  d->ring = NULL;
+  d->ring = ringnew(d->ring, sec*CHANNELS*SAMPLERATE*2);
+  ringheader(d->ring)->write += CHANNELS*SAMPLERATE*sec;
+  YughWarn("FWD DELAY");
+  return make_node(d, filter_fwd_delay, delay_free);
 }
 
 /* Get decay constant for a given pole */
@@ -407,7 +507,7 @@ dsp_node *dsp_adsr(unsigned int atk, unsigned int dec, unsigned int sus, unsigne
     adsr->rls = rls + adsr->sus;
     adsr->rls_t = tau2pole(rls / 3000.f);
 
-    return make_node(adsr, dsp_adsr_fillbuf);
+    return make_node(adsr, dsp_adsr_fillbuf, NULL);
 }
 
 void filter_noise_gate(float *floor, soundbyte *out, int frames)
@@ -419,7 +519,7 @@ dsp_node *dsp_noise_gate(float floor)
 {
   float *v = malloc(sizeof(float));
   *v = floor;
-  return make_node(v, filter_noise_gate);
+  return make_node(v, filter_noise_gate, NULL);
 }
 
 void filter_limiter(float *ceil, soundbyte *out, int n)
@@ -431,7 +531,7 @@ dsp_node *dsp_limiter(float ceil)
 {
   float *v = malloc(sizeof(float));
   *v = ceil;
-  return make_node(v, filter_limiter);
+  return make_node(v, filter_limiter, NULL);
 }
 
 void dsp_compressor_fillbuf(struct dsp_compressor *comp, soundbyte *out, int n)
@@ -470,22 +570,21 @@ dsp_node *dsp_compressor()
 
     struct dsp_compressor *c = malloc(sizeof(*c));
     *c = new;
-    return make_node(c, dsp_compressor_fillbuf);
+    return make_node(c, dsp_compressor_fillbuf, NULL);
 }
 
 /* Assumes 2 channels in a frame */
 void pan_frames(soundbyte *out, float deg, int frames)
 {
   if (deg == 0.f) return;
-  if (deg < -100) deg = -100.f;
+  if (deg < -1) deg = -1.f;
-  else if (deg > 100) deg = 100.f;
+  else if (deg > 1) deg = 1.f;
 
   float db1, db2;
-  float pct = deg / 100.f;
 
   if (deg > 0) {
-    db1 = pct2db(1 - pct);
+    db1 = pct2db(1 - deg);
-    db2 = pct2db(pct);
+    db2 = pct2db(deg);
     for (int i = 0; i < frames; i++) {
       soundbyte L = out[i*2];
       soundbyte R = out[i*2+1];
@@ -493,8 +592,8 @@ void pan_frames(soundbyte *out, float deg, int frames)
       out[i*2+1] = (R + fgain(L, db2))/2;
     }
   } else {
-    db1 = pct2db(1 + pct);
+    db1 = pct2db(1 + deg);
-    db2 = pct2db(-1*pct);
+    db2 = pct2db(-1*deg);
     for (int i = 0; i < frames; i++) {
       soundbyte L = out[i*2];
       soundbyte R = out[i*2+1];
@@ -544,5 +643,5 @@ 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);
+  return make_node(b, filter_bitcrush, NULL);
 }

source/engine/sound/dsp.h

@@ -8,6 +8,7 @@
 #include "sound.h"
 #include "cbuf.h"
 #include "script.h"
+#include "iir.h"
 
 /* a DSP node, when processed, sums its inputs, and stores the result of proc in its cache */
 typedef struct dsp_node {
@@ -28,10 +29,11 @@ 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));
+dsp_node *make_node(void *data, void (*proc)(void *data, soundbyte *out, int samples), void (*fr)(void *data));
 void plugin_node(dsp_node *from, dsp_node *to);
 void unplug_node(dsp_node *node);
 void node_free(dsp_node *node);
+void filter_iir(struct dsp_iir *iir, soundbyte *buffer, int frames);
 
 void scale_soundbytes(soundbyte *a, float scale, int frames);
 void sum_soundbytes(soundbyte *a, soundbyte *b, int frames);
@@ -71,6 +73,8 @@ typedef struct {
 } delay;
 
 dsp_node *dsp_delay(double sec, double decay);
+dsp_node *dsp_fwd_delay(double sec, double decay);
+dsp_node *dsp_pitchshift(float octaves);
 
 struct dsp_compressor {
     double ratio;
@@ -91,6 +95,7 @@ struct phasor phasor_make(unsigned int sr, float freq);
 
 dsp_node *dsp_whitenoise();
 dsp_node *dsp_pinknoise();
+dsp_node *dsp_rednoise();
 
 float sin_phasor(float p);
 float square_phasor(float p);

source/engine/sound/iir.h

@@ -40,6 +40,7 @@ struct dsp_iir sp_hp(double fcf);
 
 double chevy_pct_to_e(double pct);
 
+struct dsp_iir make_iir(int order);
 struct dsp_iir bqlp_dcof(double fcf, float Q);
 struct dsp_iir bqhp_dcof(double fcf, float Q);
 struct dsp_iir bqbpq_dcof(double fcf, float Q);

source/engine/sound/music.c

@@ -74,6 +74,13 @@ tsf *make_soundfont(const char *path)
   return sf;
 }
 
+void dsp_midi_free(struct dsp_midi_song *ms)
+{
+  free(ms->midi);
+  tsf_close(ms->sf);
+  free(ms);
+}
+
 dsp_node *dsp_midi(const char *midi, tsf *sf)
 {
   long rawlen;
@@ -82,7 +89,8 @@ dsp_node *dsp_midi(const char *midi, tsf *sf)
   ms->time = 0.0;
   ms->midi = tml_load_memory(raw, rawlen);
   ms->sf = tsf_copy(sf);
-  return make_node(ms, dsp_midi_fillbuf);
+  free(midi);
+  return make_node(ms, dsp_midi_fillbuf, dsp_midi_free);
 }
 
 void play_song(const char *midi, const char *sf)

source/engine/sound/smbPitchShift.h

@@ -0,0 +1,317 @@
+/****************************************************************************
+*
+* NAME: smbPitchShift.c
+* VERSION: 1.2
+* HOME URL: http://blogs.zynaptiq.com/bernsee
+* KNOWN BUGS: none
+*
+* SYNOPSIS: Routine for doing pitch shifting while maintaining
+* duration using the Short Time Fourier Transform.
+*
+* DESCRIPTION: The routine takes a pitchShift factor value which is between 0.5
+* (one octave down) and 2. (one octave up). A value of exactly 1 does not change
+* the pitch. numSampsToProcess tells the routine how many samples in indata[0...
+* numSampsToProcess-1] should be pitch shifted and moved to outdata[0 ...
+* numSampsToProcess-1]. The two buffers can be identical (ie. it can process the
+* data in-place). fftFrameSize defines the FFT frame size used for the
+* processing. Typical values are 1024, 2048 and 4096. It may be any value <=
+* MAX_FRAME_LENGTH but it MUST be a power of 2. osamp is the STFT
+* oversampling factor which also determines the overlap between adjacent STFT
+* frames. It should at least be 4 for moderate scaling ratios. A value of 32 is
+* recommended for best quality. sampleRate takes the sample rate for the signal 
+* in unit Hz, ie. 44100 for 44.1 kHz audio. The data passed to the routine in 
+* indata[] should be in the range [-1.0, 1.0), which is also the output range 
+* for the data, make sure you scale the data accordingly (for 16bit signed integers
+* you would have to divide (and multiply) by 32768). 
+*
+* COPYRIGHT 1999-2015 Stephan M. Bernsee <s.bernsee [AT] zynaptiq [DOT] com>
+*
+* 						The Wide Open License (WOL)
+*
+* Permission to use, copy, modify, distribute and sell this software and its
+* documentation for any purpose is hereby granted without fee, provided that
+* the above copyright notice and this license appear in all source copies. 
+* THIS SOFTWARE IS PROVIDED "AS IS" WITHOUT EXPRESS OR IMPLIED WARRANTY OF
+* ANY KIND. See http://www.dspguru.com/wol.htm for more information.
+*
+*****************************************************************************/ 
+
+#include <string.h>
+#include <math.h>
+#include <stdio.h>
+
+#define M_PI 3.14159265358979323846
+#define MAX_FRAME_LENGTH 8192
+#define false 0
+#define true 1
+
+void smbFft(float *fftBuffer, long fftFrameSize, long sign);
+double smbAtan2(double x, double y);
+
+
+// -----------------------------------------------------------------------------------------------------------------
+
+
+void smbPitchShift(float pitchShift, long numSampsToProcess, long fftFrameSize, long osamp, float sampleRate, float *indata, float *outdata)
+/*
+	Routine smbPitchShift(). See top of file for explanation
+	Purpose: doing pitch shifting while maintaining duration using the Short
+	Time Fourier Transform.
+	Author: (c)1999-2015 Stephan M. Bernsee <s.bernsee [AT] zynaptiq [DOT] com>
+*/
+{
+
+	static float gInFIFO[MAX_FRAME_LENGTH];
+	static float gOutFIFO[MAX_FRAME_LENGTH];
+	static float gFFTworksp[2*MAX_FRAME_LENGTH];
+	static float gLastPhase[MAX_FRAME_LENGTH/2+1];
+	static float gSumPhase[MAX_FRAME_LENGTH/2+1];
+	static float gOutputAccum[2*MAX_FRAME_LENGTH];
+	static float gAnaFreq[MAX_FRAME_LENGTH];
+	static float gAnaMagn[MAX_FRAME_LENGTH];
+	static float gSynFreq[MAX_FRAME_LENGTH];
+	static float gSynMagn[MAX_FRAME_LENGTH];
+	static long gRover = false, gInit = false;
+	double magn, phase, tmp, window, real, imag;
+	double freqPerBin, expct;
+	long i,k, qpd, index, inFifoLatency, stepSize, fftFrameSize2;
+
+	/* set up some handy variables */
+	fftFrameSize2 = fftFrameSize/2;
+	stepSize = fftFrameSize/osamp;
+	freqPerBin = sampleRate/(double)fftFrameSize;
+	expct = 2.*M_PI*(double)stepSize/(double)fftFrameSize;
+	inFifoLatency = fftFrameSize-stepSize;
+	if (gRover == false) gRover = inFifoLatency;
+
+	/* initialize our static arrays */
+	if (gInit == false) {
+		memset(gInFIFO, 0, MAX_FRAME_LENGTH*sizeof(float));
+		memset(gOutFIFO, 0, MAX_FRAME_LENGTH*sizeof(float));
+		memset(gFFTworksp, 0, 2*MAX_FRAME_LENGTH*sizeof(float));
+		memset(gLastPhase, 0, (MAX_FRAME_LENGTH/2+1)*sizeof(float));
+		memset(gSumPhase, 0, (MAX_FRAME_LENGTH/2+1)*sizeof(float));
+		memset(gOutputAccum, 0, 2*MAX_FRAME_LENGTH*sizeof(float));
+		memset(gAnaFreq, 0, MAX_FRAME_LENGTH*sizeof(float));
+		memset(gAnaMagn, 0, MAX_FRAME_LENGTH*sizeof(float));
+		gInit = true;
+	}
+
+	/* main processing loop */
+	for (i = 0; i < numSampsToProcess; i++){
+
+		/* As long as we have not yet collected enough data just read in */
+		gInFIFO[gRover] = indata[i];
+		outdata[i] = gOutFIFO[gRover-inFifoLatency];
+		gRover++;
+
+		/* now we have enough data for processing */
+		if (gRover >= fftFrameSize) {
+			gRover = inFifoLatency;
+
+			/* do windowing and re,im interleave */
+			for (k = 0; k < fftFrameSize;k++) {
+				window = -.5*cos(2.*M_PI*(double)k/(double)fftFrameSize)+.5;
+				gFFTworksp[2*k] = gInFIFO[k] * window;
+				gFFTworksp[2*k+1] = 0.;
+			}
+
+
+			/* ***************** ANALYSIS ******************* */
+			/* do transform */
+			smbFft(gFFTworksp, fftFrameSize, -1);
+
+			/* this is the analysis step */
+			for (k = 0; k <= fftFrameSize2; k++) {
+
+				/* de-interlace FFT buffer */
+				real = gFFTworksp[2*k];
+				imag = gFFTworksp[2*k+1];
+
+				/* compute magnitude and phase */
+				magn = 2.*sqrt(real*real + imag*imag);
+				phase = atan2(imag,real);
+
+				/* compute phase difference */
+				tmp = phase - gLastPhase[k];
+				gLastPhase[k] = phase;
+
+				/* subtract expected phase difference */
+				tmp -= (double)k*expct;
+
+				/* map delta phase into +/- Pi interval */
+				qpd = tmp/M_PI;
+				if (qpd >= 0) qpd += qpd&1;
+				else qpd -= qpd&1;
+				tmp -= M_PI*(double)qpd;
+
+				/* get deviation from bin frequency from the +/- Pi interval */
+				tmp = osamp*tmp/(2.*M_PI);
+
+				/* compute the k-th partials' true frequency */
+				tmp = (double)k*freqPerBin + tmp*freqPerBin;
+
+				/* store magnitude and true frequency in analysis arrays */
+				gAnaMagn[k] = magn;
+				gAnaFreq[k] = tmp;
+
+			}
+
+			/* ***************** PROCESSING ******************* */
+			/* this does the actual pitch shifting */
+			memset(gSynMagn, 0, fftFrameSize*sizeof(float));
+			memset(gSynFreq, 0, fftFrameSize*sizeof(float));
+			for (k = 0; k <= fftFrameSize2; k++) { 
+				index = k*pitchShift;
+				if (index <= fftFrameSize2) { 
+					gSynMagn[index] += gAnaMagn[k]; 
+					gSynFreq[index] = gAnaFreq[k] * pitchShift; 
+				} 
+			}
+			
+			/* ***************** SYNTHESIS ******************* */
+			/* this is the synthesis step */
+			for (k = 0; k <= fftFrameSize2; k++) {
+
+				/* get magnitude and true frequency from synthesis arrays */
+				magn = gSynMagn[k];
+				tmp = gSynFreq[k];
+
+				/* subtract bin mid frequency */
+				tmp -= (double)k*freqPerBin;
+
+				/* get bin deviation from freq deviation */
+				tmp /= freqPerBin;
+
+				/* take osamp into account */
+				tmp = 2.*M_PI*tmp/osamp;
+
+				/* add the overlap phase advance back in */
+				tmp += (double)k*expct;
+
+				/* accumulate delta phase to get bin phase */
+				gSumPhase[k] += tmp;
+				phase = gSumPhase[k];
+
+				/* get real and imag part and re-interleave */
+				gFFTworksp[2*k] = magn*cos(phase);
+				gFFTworksp[2*k+1] = magn*sin(phase);
+			} 
+
+			/* zero negative frequencies */
+			for (k = fftFrameSize+2; k < 2*fftFrameSize; k++) gFFTworksp[k] = 0.;
+
+			/* do inverse transform */
+			smbFft(gFFTworksp, fftFrameSize, 1);
+
+			/* do windowing and add to output accumulator */ 
+			for(k=0; k < fftFrameSize; k++) {
+				window = -.5*cos(2.*M_PI*(double)k/(double)fftFrameSize)+.5;
+				gOutputAccum[k] += 2.*window*gFFTworksp[2*k]/(fftFrameSize2*osamp);
+			}
+			for (k = 0; k < stepSize; k++) gOutFIFO[k] = gOutputAccum[k];
+
+			/* shift accumulator */
+			memmove(gOutputAccum, gOutputAccum+stepSize, fftFrameSize*sizeof(float));
+
+			/* move input FIFO */
+			for (k = 0; k < inFifoLatency; k++) gInFIFO[k] = gInFIFO[k+stepSize];
+		}
+	}
+}
+
+// -----------------------------------------------------------------------------------------------------------------
+
+
+void smbFft(float *fftBuffer, long fftFrameSize, long sign)
+/* 
+	FFT routine, (C)1996 S.M.Bernsee. Sign = -1 is FFT, 1 is iFFT (inverse)
+	Fills fftBuffer[0...2*fftFrameSize-1] with the Fourier transform of the
+	time domain data in fftBuffer[0...2*fftFrameSize-1]. The FFT array takes
+	and returns the cosine and sine parts in an interleaved manner, ie.
+	fftBuffer[0] = cosPart[0], fftBuffer[1] = sinPart[0], asf. fftFrameSize
+	must be a power of 2. It expects a complex input signal (see footnote 2),
+	ie. when working with 'common' audio signals our input signal has to be
+	passed as {in[0],0.,in[1],0.,in[2],0.,...} asf. In that case, the transform
+	of the frequencies of interest is in fftBuffer[0...fftFrameSize].
+*/
+{
+	float wr, wi, arg, *p1, *p2, temp;
+	float tr, ti, ur, ui, *p1r, *p1i, *p2r, *p2i;
+	long i, bitm, j, le, le2, k;
+
+	for (i = 2; i < 2*fftFrameSize-2; i += 2) {
+		for (bitm = 2, j = 0; bitm < 2*fftFrameSize; bitm <<= 1) {
+			if (i & bitm) j++;
+			j <<= 1;
+		}
+		if (i < j) {
+			p1 = fftBuffer+i; p2 = fftBuffer+j;
+			temp = *p1; *(p1++) = *p2;
+			*(p2++) = temp; temp = *p1;
+			*p1 = *p2; *p2 = temp;
+		}
+	}
+	for (k = 0, le = 2; k < (long)(log(fftFrameSize)/log(2.)+.5); k++) {
+		le <<= 1;
+		le2 = le>>1;
+		ur = 1.0;
+		ui = 0.0;
+		arg = M_PI / (le2>>1);
+		wr = cos(arg);
+		wi = sign*sin(arg);
+		for (j = 0; j < le2; j += 2) {
+			p1r = fftBuffer+j; p1i = p1r+1;
+			p2r = p1r+le2; p2i = p2r+1;
+			for (i = j; i < 2*fftFrameSize; i += le) {
+				tr = *p2r * ur - *p2i * ui;
+				ti = *p2r * ui + *p2i * ur;
+				*p2r = *p1r - tr; *p2i = *p1i - ti;
+				*p1r += tr; *p1i += ti;
+				p1r += le; p1i += le;
+				p2r += le; p2i += le;
+			}
+			tr = ur*wr - ui*wi;
+			ui = ur*wi + ui*wr;
+			ur = tr;
+		}
+	}
+}
+
+
+// -----------------------------------------------------------------------------------------------------------------
+
+/*
+
+    12/12/02, smb
+    
+    PLEASE NOTE:
+    
+    There have been some reports on domain errors when the atan2() function was used
+    as in the above code. Usually, a domain error should not interrupt the program flow
+    (maybe except in Debug mode) but rather be handled "silently" and a global variable
+    should be set according to this error. However, on some occasions people ran into
+    this kind of scenario, so a replacement atan2() function is provided here.
+    
+    If you are experiencing domain errors and your program stops, simply replace all
+    instances of atan2() with calls to the smbAtan2() function below.
+    
+*/
+
+
+double smbAtan2(double x, double y)
+{
+  double signx;
+  if (x > 0.) signx = 1.;  
+  else signx = -1.;
+  
+  if (x == 0.) return 0.;
+  if (y == 0.) return signx * M_PI / 2.;
+  
+  return atan2(x, y);
+}
+
+
+// -----------------------------------------------------------------------------------------------------------------
+// -----------------------------------------------------------------------------------------------------------------
+// -----------------------------------------------------------------------------------------------------------------

source/engine/sound/sound.c

@@ -121,7 +121,7 @@ dsp_node *dsp_mod(const char *path)
   void *data = slurp_file(path, &modsize);
   pocketmod_context *mod = malloc(sizeof(*mod));
   pocketmod_init(mod, data, modsize, SAMPLERATE);
-  return make_node(mod, filter_mod);
+  return make_node(mod, filter_mod, NULL);
 }
 
 void sound_init() {
@@ -242,11 +242,17 @@ void sound_fillbuf(struct sound *s, soundbyte *buf, int n) {
   else
     end = 1;
 
+  if (s->timescale != 1) {
+    src_callback_read(s->src, s->timescale, frames, buf);
+    return;
+  }
+  
   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++;
   }
 
@@ -260,18 +266,28 @@ void sound_fillbuf(struct sound *s, soundbyte *buf, int n) {
 void free_source(struct sound *s)
 {
   JS_FreeValue(js, s->hook);
+  src_delete(s->src);
   free(s);
 }
 
+static long *src_cb(struct sound *s, float **data)
+{
+  long needed = BUF_FRAMES/s->timescale;
+  *data = s->data->data+s->frame;
+  s->frame += needed;
+  return needed;
+}
+
 struct dsp_node *dsp_source(char *path)
 {
   struct sound *self = malloc(sizeof(*self));
   self->frame = 0;
   self->data = make_sound(path);
   self->loop = false;
+  self->src = src_callback_new(src_cb, SRC_SINC_MEDIUM_QUALITY, 2, NULL, self);
+  self->timescale = 1;
   self->hook = JS_UNDEFINED;
-  dsp_node *n = make_node(self, sound_fillbuf);
+  dsp_node *n = make_node(self, sound_fillbuf, free_source);
-  n->data_free = free_source;
   return n;
 }
 

source/engine/sound/sound.h

@@ -2,6 +2,7 @@
 #define SOUND_H
 
 #include "script.h"
+#include "samplerate.h"
 
 typedef float soundbyte;
 
@@ -12,6 +13,8 @@ typedef struct sound {
     unsigned int frame; /* Pointing to the current frame on the wav */
     struct wav *data;
     int loop;
+    float timescale;
+    SRC_STATE *src;
     JSValue hook;
 } sound;