update

17 files changed, 6590 insertions, 140 deletions

diff --git a/src/3rdparty/cmixer/cmixer.c b/src/3rdparty/cmixer/cmixer.c
new file mode 100644
index 0000000..b3f8a44
--- /dev/null
+++ b/src/3rdparty/cmixer/cmixer.c
@@ -0,0 +1,738 @@
+/*
+** Copyright (c) 2017 rxi
+**
+** 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.
+**/
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "cmixer.h"
+
+#define UNUSED(x)         ((void) (x))
+#define CLAMP(x, a, b)    ((x) < (a) ? (a) : (x) > (b) ? (b) : (x))
+#define MIN(a, b)         ((a) < (b) ? (a) : (b))
+#define MAX(a, b)         ((a) > (b) ? (a) : (b))
+
+#define FX_BITS           (12)
+#define FX_UNIT           (1 << FX_BITS)
+#define FX_MASK           (FX_UNIT - 1)
+#define FX_FROM_FLOAT(f)  ((f) * FX_UNIT)
+#define FX_LERP(a, b, p)  ((a) + ((((b) - (a)) * (p)) >> FX_BITS))
+
+#define BUFFER_SIZE       (512)
+#define BUFFER_MASK       (BUFFER_SIZE - 1)
+
+#define CM_USE_STB_VORBIS
+
+struct cm_Source {
+  cm_Source *next;              /* Next source in list */
+  cm_Int16 buffer[BUFFER_SIZE]; /* Internal buffer with raw stereo PCM */
+  cm_EventHandler handler;      /* Event handler */
+  void *udata;          /* Stream's udata (from cm_SourceInfo) */
+  int samplerate;       /* Stream's native samplerate */
+  int length;           /* Stream's length in frames */
+  int end;              /* End index for the current play-through */
+  int state;            /* Current state (playing|paused|stopped) */
+  cm_Int64 position;    /* Current playhead position (fixed point) */
+  int lgain, rgain;     /* Left and right gain (fixed point) */
+  int rate;             /* Playback rate (fixed point) */
+  int nextfill;         /* Next frame idx where the buffer needs to be filled */
+  int loop;             /* Whether the source will loop when `end` is reached */
+  int rewind;           /* Whether the source will rewind before playing */
+  int active;           /* Whether the source is part of `sources` list */
+  double gain;          /* Gain set by `cm_set_gain()` */
+  double pan;           /* Pan set by `cm_set_pan()` */
+};
+
+
+static struct {
+  const char *lasterror;        /* Last error message */
+  cm_EventHandler lock;         /* Event handler for lock/unlock events */
+  cm_Source *sources;           /* Linked list of active (playing) sources */
+  cm_Int32 buffer[BUFFER_SIZE]; /* Internal master buffer */
+  int samplerate;               /* Master samplerate */
+  int gain;                     /* Master gain (fixed point) */
+} cmixer;
+
+
+static void dummy_handler(cm_Event *e) {
+  UNUSED(e);
+}
+
+
+static void lock(void) {
+  cm_Event e;
+  e.type = CM_EVENT_LOCK;
+  cmixer.lock(&e);
+}
+
+
+static void unlock(void) {
+  cm_Event e;
+  e.type = CM_EVENT_UNLOCK;
+  cmixer.lock(&e);
+}
+
+
+const char* cm_get_error(void) {
+  const char *res = cmixer.lasterror;
+  cmixer.lasterror = NULL;
+  return res;
+}
+
+
+static const char* error(const char *msg) {
+  cmixer.lasterror = msg;
+  return msg;
+}
+
+
+void cm_init(int samplerate) {
+  cmixer.samplerate = samplerate;
+  cmixer.lock = dummy_handler;
+  cmixer.sources = NULL;
+  cmixer.gain = FX_UNIT;
+}
+
+
+void cm_set_lock(cm_EventHandler lock) {
+  cmixer.lock = lock;
+}
+
+
+void cm_set_master_gain(double gain) {
+  cmixer.gain = FX_FROM_FLOAT(gain);
+}
+
+
+static void rewind_source(cm_Source *src) {
+  cm_Event e;
+  e.type = CM_EVENT_REWIND;
+  e.udata = src->udata;
+  src->handler(&e);
+  src->position = 0;
+  src->rewind = 0;
+  src->end = src->length;
+  src->nextfill = 0;
+}
+
+
+static void fill_source_buffer(cm_Source *src, int offset, int length) {
+  cm_Event e;
+  e.type = CM_EVENT_SAMPLES;
+  e.udata = src->udata;
+  e.buffer = src->buffer + offset;
+  e.length = length;
+  src->handler(&e);
+}
+
+
+static void process_source(cm_Source *src, int len) {
+  int i, n, a, b, p;
+  int frame, count;
+  cm_Int32 *dst = cmixer.buffer;
+
+  /* Do rewind if flag is set */
+  if (src->rewind) {
+    rewind_source(src);
+  }
+
+  /* Don't process if not playing */
+  if (src->state != CM_STATE_PLAYING) {
+    return;
+  }
+
+  /* Process audio */
+  while (len > 0) {
+    /* Get current position frame */
+    frame = src->position >> FX_BITS;
+
+    /* Fill buffer if required */
+    if (frame + 3 >= src->nextfill) {
+      fill_source_buffer(src, (src->nextfill*2) & BUFFER_MASK, BUFFER_SIZE/2);
+      src->nextfill += BUFFER_SIZE / 4;
+    }
+
+    /* Handle reaching the end of the playthrough */
+    if (frame >= src->end) {
+      /* As streams continiously fill the raw buffer in a loop we simply
+      ** increment the end idx by one length and continue reading from it for
+      ** another play-through */
+      src->end = frame + src->length;
+      /* Set state and stop processing if we're not set to loop */
+      if (!src->loop) {
+        src->state = CM_STATE_STOPPED;
+        break;
+      }
+    }
+
+    /* Work out how many frames we should process in the loop */
+    n = MIN(src->nextfill - 2, src->end) - frame;
+    count = (n << FX_BITS) / src->rate;
+    count = MAX(count, 1);
+    count = MIN(count, len / 2);
+    len -= count * 2;
+
+    /* Add audio to master buffer */
+    if (src->rate == FX_UNIT) {
+      /* Add audio to buffer -- basic */
+      n = frame * 2;
+      for (i = 0; i < count; i++) {
+        dst[0] += (src->buffer[(n    ) & BUFFER_MASK] * src->lgain) >> FX_BITS;
+        dst[1] += (src->buffer[(n + 1) & BUFFER_MASK] * src->rgain) >> FX_BITS;
+        n += 2;
+        dst += 2;
+      }
+      src->position += count * FX_UNIT;
+
+    } else {
+      /* Add audio to buffer -- interpolated */
+      for (i = 0; i < count; i++) {
+        n = (src->position >> FX_BITS) * 2;
+        p = src->position & FX_MASK;
+        a = src->buffer[(n    ) & BUFFER_MASK];
+        b = src->buffer[(n + 2) & BUFFER_MASK];
+        dst[0] += (FX_LERP(a, b, p) * src->lgain) >> FX_BITS;
+        n++;
+        a = src->buffer[(n    ) & BUFFER_MASK];
+        b = src->buffer[(n + 2) & BUFFER_MASK];
+        dst[1] += (FX_LERP(a, b, p) * src->rgain) >> FX_BITS;
+        src->position += src->rate;
+        dst += 2;
+      }
+    }
+
+  }
+}
+
+
+void cm_process(cm_Int16 *dst, int len) {
+  int i;
+  cm_Source **s;
+
+  /* Process in chunks of BUFFER_SIZE if `len` is larger than BUFFER_SIZE */
+  while (len > BUFFER_SIZE) {
+    cm_process(dst, BUFFER_SIZE);
+    dst += BUFFER_SIZE;
+    len -= BUFFER_SIZE;
+  }
+
+  /* Zeroset internal buffer */
+  memset(cmixer.buffer, 0, len * sizeof(cmixer.buffer[0]));
+
+  /* Process active sources */
+  lock();
+  s = &cmixer.sources;
+  while (*s) {
+    process_source(*s, len);
+    /* Remove source from list if it is no longer playing */
+    if ((*s)->state != CM_STATE_PLAYING) {
+      (*s)->active = 0;
+      *s = (*s)->next;
+    } else {
+      s = &(*s)->next;
+    }
+  }
+  unlock();
+
+  /* Copy internal buffer to destination and clip */
+  for (i = 0; i < len; i++) {
+    int x = (cmixer.buffer[i] * cmixer.gain) >> FX_BITS;
+    dst[i] = CLAMP(x, -32768, 32767);
+  }
+}
+
+
+cm_Source* cm_new_source(const cm_SourceInfo *info) {
+  cm_Source *src = (cm_Source *)calloc(1, sizeof(*src));
+  if (!src) {
+    error("allocation failed");
+    return NULL;
+  }
+  src->handler = info->handler;
+  src->length = info->length;
+  src->samplerate = info->samplerate;
+  src->udata = info->udata;
+  cm_set_gain(src, 1);
+  cm_set_pan(src, 0);
+  cm_set_pitch(src, 1);
+  cm_set_loop(src, 0);
+  cm_stop(src);
+  return src;
+}
+
+
+static const char* wav_init(cm_SourceInfo *info, void *data, int len, int ownsdata);
+
+#ifdef CM_USE_STB_VORBIS
+static const char* ogg_init(cm_SourceInfo *info, void *data, int len, int ownsdata);
+#endif
+
+
+static int check_header(void *data, int size, char *str, int offset) {
+  int len = strlen(str);
+  return (size >= offset + len) && !memcmp((char*) data + offset, str, len);
+}
+
+
+static cm_Source* new_source_from_mem(void *data, int size, int ownsdata) {
+  const char *err;
+  cm_SourceInfo info;
+
+  if (check_header(data, size, "WAVE", 8)) {
+    err = wav_init(&info, data, size, ownsdata);
+    if (err) {
+      return NULL;
+    }
+    return cm_new_source(&info);
+  }
+
+#ifdef CM_USE_STB_VORBIS
+  if (check_header(data, size, "OggS", 0)) {
+    err = ogg_init(&info, data, size, ownsdata);
+    if (err) {
+      return NULL;
+    }
+    return cm_new_source(&info);
+  }
+#endif
+
+  error("unknown format or invalid data");
+  return NULL;
+}
+
+
+static void* load_file(const char *filename, int *size) {
+  FILE *fp;
+  void *data;
+  int n;
+
+  fp = fopen(filename, "rb");
+  if (!fp) {
+    return NULL;
+  }
+
+  /* Get size */
+  fseek(fp, 0, SEEK_END);
+  *size = ftell(fp);
+  rewind(fp);
+
+  /* Malloc, read and return data */
+  data = malloc(*size);
+  if (!data) {
+    fclose(fp);
+    return NULL;
+  }
+  n = fread(data, 1, *size, fp);
+  fclose(fp);
+  if (n != *size) {
+    free(data);
+    return NULL;
+  }
+
+  return data;
+}
+
+
+cm_Source* cm_new_source_from_file(const char *filename) {
+  int size;
+  cm_Source *src;
+  void *data;
+
+  /* Load file into memory */
+  data = load_file(filename, &size);
+  if (!data) {
+    error("could not load file");
+    return NULL;
+  }
+
+  /* Try to load and return */
+  src = new_source_from_mem(data, size, 1);
+  if (!src) {
+    free(data);
+    return NULL;
+  }
+
+  return src;
+}
+
+
+cm_Source* cm_new_source_from_mem(void *data, int size) {
+  return new_source_from_mem(data, size, 0);
+}
+
+
+void cm_destroy_source(cm_Source *src) {
+  cm_Event e;
+  lock();
+  if (src->active) {
+    cm_Source **s = &cmixer.sources;
+    while (*s) {
+      if (*s == src) {
+        *s = src->next;
+        break;
+      }
+    }
+  }
+  unlock();
+  e.type = CM_EVENT_DESTROY;
+  e.udata = src->udata;
+  src->handler(&e);
+  free(src);
+}
+
+
+double cm_get_length(cm_Source *src) {
+  return src->length / (double) src->samplerate;
+}
+
+
+double cm_get_position(cm_Source *src) {
+  return ((src->position >> FX_BITS) % src->length) / (double) src->samplerate;
+}
+
+
+int cm_get_state(cm_Source *src) {
+  return src->state;
+}
+
+
+static void recalc_source_gains(cm_Source *src) {
+  double l, r;
+  double pan = src->pan;
+  l = src->gain * (pan <= 0. ? 1. : 1. - pan);
+  r = src->gain * (pan >= 0. ? 1. : 1. + pan);
+  src->lgain = FX_FROM_FLOAT(l);
+  src->rgain = FX_FROM_FLOAT(r);
+}
+
+
+void cm_set_gain(cm_Source *src, double gain) {
+  src->gain = gain;
+  recalc_source_gains(src);
+}
+
+
+void cm_set_pan(cm_Source *src, double pan) {
+  src->pan = CLAMP(pan, -1.0, 1.0);
+  recalc_source_gains(src);
+}
+
+
+void cm_set_pitch(cm_Source *src, double pitch) {
+  double rate;
+  if (pitch > 0.) {
+    rate = src->samplerate / (double) cmixer.samplerate * pitch;
+  } else {
+    rate = 0.001;
+  }
+  src->rate = FX_FROM_FLOAT(rate);
+}
+
+
+void cm_set_loop(cm_Source *src, int loop) {
+  src->loop = loop;
+}
+
+
+void cm_play(cm_Source *src) {
+  lock();
+  src->state = CM_STATE_PLAYING;
+  if (!src->active) {
+    src->active = 1;
+    src->next = cmixer.sources;
+    cmixer.sources = src;
+  }
+  unlock();
+}
+
+
+void cm_pause(cm_Source *src) {
+  src->state = CM_STATE_PAUSED;
+}
+
+
+void cm_stop(cm_Source *src) {
+  src->state = CM_STATE_STOPPED;
+  src->rewind = 1;
+}
+
+
+/*============================================================================
+** Wav stream
+**============================================================================*/
+
+typedef struct {
+  void *data;
+  int bitdepth;
+  int samplerate;
+  int channels;
+  int length;
+} Wav;
+
+typedef struct {
+  Wav wav;
+  void *data;
+  int idx;
+} WavStream;
+
+
+static char* find_subchunk(char *data, int len, char *id, int *size) {
+  /* TODO : Error handling on malformed wav file */
+  int idlen = strlen(id);
+  char *p = data + 12;
+next:
+  *size = *((cm_UInt32*) (p + 4));
+  if (memcmp(p, id, idlen)) {
+    p += 8 + *size;
+    if (p > data + len) return NULL;
+    goto next;
+  }
+  return p + 8;
+}
+
+
+static const char* read_wav(Wav *w, void *data, int len) {
+  int bitdepth, channels, samplerate, format;
+  int sz;
+  char *p = (char*)data;
+  memset(w, 0, sizeof(*w));
+
+  /* Check header */
+  if (memcmp(p, "RIFF", 4) || memcmp(p + 8, "WAVE", 4)) {
+    return error("bad wav header");
+  }
+  /* Find fmt subchunk */
+  p = find_subchunk((char*)data, len, "fmt", &sz);
+  if (!p) {
+    return error("no fmt subchunk");
+  }
+
+  /* Load fmt info */
+  format      = *((cm_UInt16*) (p));
+  channels    = *((cm_UInt16*) (p + 2));
+  samplerate  = *((cm_UInt32*) (p + 4));
+  bitdepth    = *((cm_UInt16*) (p + 14));
+  if (format != 1) {
+    return error("unsupported format");
+  }
+  if (channels == 0 || samplerate == 0 || bitdepth == 0) {
+    return error("bad format");
+  }
+
+  /* Find data subchunk */
+  p = find_subchunk((char*)data, len, "data", &sz);
+  if (!p) {
+    return error("no data subchunk");
+  }
+
+  /* Init struct */
+  w->data = (void*) p;
+  w->samplerate = samplerate;
+  w->channels = channels;
+  w->length = (sz / (bitdepth / 8)) / channels;
+  w->bitdepth = bitdepth;
+  /* Done */
+  return NULL;
+}
+
+
+#define WAV_PROCESS_LOOP(X) \
+  while (n--) {             \
+    X                       \
+    dst += 2;               \
+    s->idx++;               \
+  }
+
+static void wav_handler(cm_Event *e) {
+  int x, n;
+  cm_Int16 *dst;
+  WavStream *s = (WavStream *)e->udata;
+  int len;
+
+  switch (e->type) {
+
+    case CM_EVENT_DESTROY:
+      free(s->data);
+      free(s);
+      break;
+
+    case CM_EVENT_SAMPLES:
+      dst = e->buffer;
+      len = e->length / 2;
+fill:
+      n = MIN(len, s->wav.length - s->idx);
+      len -= n;
+      if (s->wav.bitdepth == 16 && s->wav.channels == 1) {
+        WAV_PROCESS_LOOP({
+          dst[0] = dst[1] = ((cm_Int16*) s->wav.data)[s->idx];
+        });
+      } else if (s->wav.bitdepth == 16 && s->wav.channels == 2) {
+        WAV_PROCESS_LOOP({
+          x = s->idx * 2;
+          dst[0] = ((cm_Int16*) s->wav.data)[x    ];
+          dst[1] = ((cm_Int16*) s->wav.data)[x + 1];
+        });
+      } else if (s->wav.bitdepth == 8 && s->wav.channels == 1) {
+        WAV_PROCESS_LOOP({
+          dst[0] = dst[1] = (((cm_UInt8*) s->wav.data)[s->idx] - 128) << 8;
+        });
+      } else if (s->wav.bitdepth == 8 && s->wav.channels == 2) {
+        WAV_PROCESS_LOOP({
+          x = s->idx * 2;
+          dst[0] = (((cm_UInt8*) s->wav.data)[x    ] - 128) << 8;
+          dst[1] = (((cm_UInt8*) s->wav.data)[x + 1] - 128) << 8;
+        });
+      }
+      /* Loop back and continue filling buffer if we didn't fill the buffer */
+      if (len > 0) {
+        s->idx = 0;
+        goto fill;
+      }
+      break;
+
+    case CM_EVENT_REWIND:
+      s->idx = 0;
+      break;
+  }
+}
+
+
+static const char* wav_init(cm_SourceInfo *info, void *data, int len, int ownsdata) {
+  WavStream *stream;
+  Wav wav;
+
+  const char *err = read_wav(&wav, data, len);
+  if (err != NULL) {
+    return err;
+  }
+
+  if (wav.channels > 2 || (wav.bitdepth != 16 && wav.bitdepth != 8)) {
+    return error("unsupported wav format");
+  }
+
+  stream = (WavStream *)calloc(1, sizeof(*stream));
+  if (!stream) {
+    return error("allocation failed");
+  }
+  stream->wav = wav;
+
+  if (ownsdata) {
+    stream->data = data;
+  }
+  stream->idx = 0;
+
+  info->udata = stream;
+  info->handler = wav_handler;
+  info->samplerate = wav.samplerate;
+  info->length = wav.length;
+
+  /* Return NULL (no error) for success */
+  return NULL;
+}
+
+
+/*============================================================================
+** Ogg stream
+**============================================================================*/
+
+#ifdef CM_USE_STB_VORBIS
+
+#define STB_VORBIS_HEADER_ONLY
+#include "cmixer_stb_vorbis.h"
+
+typedef struct {
+  stb_vorbis *ogg;
+  void *data;
+} OggStream;
+
+
+static void ogg_handler(cm_Event *e) {
+  int n, len;
+  OggStream *s = (OggStream *)e->udata;
+  cm_Int16 *buf;
+
+  switch (e->type) {
+
+    case CM_EVENT_DESTROY:
+      stb_vorbis_close(s->ogg);
+      free(s->data);
+      free(s);
+      break;
+
+    case CM_EVENT_SAMPLES:
+      len = e->length;
+      buf = e->buffer;
+fill:
+      n = stb_vorbis_get_samples_short_interleaved(s->ogg, 2, buf, len);
+      n *= 2;
+      /* rewind and fill remaining buffer if we reached the end of the ogg
+      ** before filling it */
+      if (len != n) {
+        stb_vorbis_seek_start(s->ogg);
+        buf += n;
+        len -= n;
+        goto fill;
+      }
+      break;
+
+    case CM_EVENT_REWIND:
+      stb_vorbis_seek_start(s->ogg);
+      break;
+  }
+}
+
+
+static const char* ogg_init(cm_SourceInfo *info, void *data, int len, int ownsdata) {
+  OggStream *stream;
+  stb_vorbis *ogg;
+  stb_vorbis_info ogginfo;
+  int err;
+
+  ogg = stb_vorbis_open_memory((unsigned char *)data, len, &err, NULL);
+  if (!ogg) {
+    return error("invalid ogg data");
+  }
+
+  stream = (OggStream *)calloc(1, sizeof(*stream));
+  if (!stream) {
+    stb_vorbis_close(ogg);
+    return error("allocation failed");
+  }
+
+  stream->ogg = ogg;
+  if (ownsdata) {
+    stream->data = data;
+  }
+
+  ogginfo = stb_vorbis_get_info(ogg);
+
+  info->udata = stream;
+  info->handler = ogg_handler;
+  info->samplerate = ogginfo.sample_rate;
+  info->length = stb_vorbis_stream_length_in_samples(ogg);
+
+  /* Return NULL (no error) for success */
+  return NULL;
+}
+
+
+#endif
diff --git a/src/3rdparty/cmixer/cmixer.h b/src/3rdparty/cmixer/cmixer.h
new file mode 100644
index 0000000..b926df7
--- /dev/null
+++ b/src/3rdparty/cmixer/cmixer.h
@@ -0,0 +1,78 @@
+/*
+** Copyright (c) 2017 rxi
+**
+** This library is free software; you can redistribute it and/or modify it
+** under the terms of the MIT license. See `cmixer.c` for details.
+**/
+
+#ifndef CMIXER_H
+#define CMIXER_H
+
+#define CM_VERSION "0.1.1"
+
+typedef short           cm_Int16;
+typedef int             cm_Int32;
+typedef long long       cm_Int64;
+typedef unsigned char   cm_UInt8;
+typedef unsigned short  cm_UInt16;
+typedef unsigned        cm_UInt32;
+
+typedef struct cm_Source cm_Source;
+
+typedef struct {
+  int type;
+  void *udata;
+  const char *msg;
+  cm_Int16 *buffer;
+  int length;
+} cm_Event;
+
+typedef void (*cm_EventHandler)(cm_Event *e);
+
+typedef struct {
+  cm_EventHandler handler;
+  void *udata;
+  int samplerate;
+  int length;
+} cm_SourceInfo;
+
+
+enum {
+  CM_STATE_STOPPED,
+  CM_STATE_PLAYING,
+  CM_STATE_PAUSED
+};
+
+enum {
+  CM_EVENT_LOCK,
+  CM_EVENT_UNLOCK,
+  CM_EVENT_DESTROY,
+  CM_EVENT_SAMPLES,
+  CM_EVENT_REWIND
+};
+
+
+const char* cm_get_error(void);
+void cm_init(int samplerate);
+void cm_set_lock(cm_EventHandler lock);
+void cm_set_master_gain(double gain);
+void cm_process(cm_Int16 *dst, int len);
+
+cm_Source* cm_new_source(const cm_SourceInfo *info);
+cm_Source* cm_new_source_from_file(const char *filename);
+cm_Source* cm_new_source_from_mem(void *data, int size);
+
+void cm_destroy_source(cm_Source *src);
+double cm_get_length(cm_Source *src);
+double cm_get_position(cm_Source *src);
+int cm_get_state(cm_Source *src);
+
+void cm_set_gain(cm_Source *src, double gain);
+void cm_set_pan(cm_Source *src, double pan);
+void cm_set_pitch(cm_Source *src, double pitch);
+void cm_set_loop(cm_Source *src, int loop);
+void cm_play(cm_Source *src);
+void cm_pause(cm_Source *src);
+void cm_stop(cm_Source *src);
+
+#endif
diff --git a/src/3rdparty/cmixer/cmixer_stb_vorbis.h b/src/3rdparty/cmixer/cmixer_stb_vorbis.h
new file mode 100644
index 0000000..c43db84
--- /dev/null
+++ b/src/3rdparty/cmixer/cmixer_stb_vorbis.h
@@ -0,0 +1,5519 @@
+// Ogg Vorbis audio decoder - v1.14 - public domain
+// http://nothings.org/stb_vorbis/
+//
+// Original version written by Sean Barrett in 2007.
+//
+// Originally sponsored by RAD Game Tools. Seeking implementation
+// sponsored by Phillip Bennefall, Marc Andersen, Aaron Baker,
+// Elias Software, Aras Pranckevicius, and Sean Barrett.
+//
+// LICENSE
+//
+//   See end of file for license information.
+//
+// Limitations:
+//
+//   - floor 0 not supported (used in old ogg vorbis files pre-2004)
+//   - lossless sample-truncation at beginning ignored
+//   - cannot concatenate multiple vorbis streams
+//   - sample positions are 32-bit, limiting seekable 192Khz
+//       files to around 6 hours (Ogg supports 64-bit)
+//
+// Feature contributors:
+//    Dougall Johnson (sample-exact seeking)
+//
+// Bugfix/warning contributors:
+//    Terje Mathisen     Niklas Frykholm     Andy Hill
+//    Casey Muratori     John Bolton         Gargaj
+//    Laurent Gomila     Marc LeBlanc        Ronny Chevalier
+//    Bernhard Wodo      Evan Balster        alxprd@github
+//    Tom Beaumont       Ingo Leitgeb        Nicolas Guillemot
+//    Phillip Bennefall  Rohit               Thiago Goulart
+//    manxorist@github   saga musix          github:infatum
+//    Timur Gagiev
+//
+// Partial history:
+//    1.14    - 2018-02-11 - delete bogus dealloca usage
+//    1.13    - 2018-01-29 - fix truncation of last frame (hopefully)
+//    1.12    - 2017-11-21 - limit residue begin/end to blocksize/2 to avoid large temp allocs in bad/corrupt files
+//    1.11    - 2017-07-23 - fix MinGW compilation 
+//    1.10    - 2017-03-03 - more robust seeking; fix negative ilog(); clear error in open_memory
+//    1.09    - 2016-04-04 - back out 'truncation of last frame' fix from previous version
+//    1.08    - 2016-04-02 - warnings; setup memory leaks; truncation of last frame
+//    1.07    - 2015-01-16 - fixes for crashes on invalid files; warning fixes; const
+//    1.06    - 2015-08-31 - full, correct support for seeking API (Dougall Johnson)
+//                           some crash fixes when out of memory or with corrupt files
+//                           fix some inappropriately signed shifts
+//    1.05    - 2015-04-19 - don't define __forceinline if it's redundant
+//    1.04    - 2014-08-27 - fix missing const-correct case in API
+//    1.03    - 2014-08-07 - warning fixes
+//    1.02    - 2014-07-09 - declare qsort comparison as explicitly _cdecl in Windows
+//    1.01    - 2014-06-18 - fix stb_vorbis_get_samples_float (interleaved was correct)
+//    1.0     - 2014-05-26 - fix memory leaks; fix warnings; fix bugs in >2-channel;
+//                           (API change) report sample rate for decode-full-file funcs
+//
+// See end of file for full version history.
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+//  HEADER BEGINS HERE
+//
+
+#ifndef STB_VORBIS_INCLUDE_STB_VORBIS_H
+#define STB_VORBIS_INCLUDE_STB_VORBIS_H
+
+#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO)
+#define STB_VORBIS_NO_STDIO 1
+#endif
+
+#ifndef STB_VORBIS_NO_STDIO
+#include <stdio.h>
+#endif
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+    ///////////   THREAD SAFETY
+
+    // Individual stb_vorbis* handles are not thread-safe; you cannot decode from
+    // them from multiple threads at the same time. However, you can have multiple
+    // stb_vorbis* handles and decode from them independently in multiple thrads.
+
+
+    ///////////   MEMORY ALLOCATION
+
+    // normally stb_vorbis uses malloc() to allocate memory at startup,
+    // and alloca() to allocate temporary memory during a frame on the
+    // stack. (Memory consumption will depend on the amount of setup
+    // data in the file and how you set the compile flags for speed
+    // vs. size. In my test files the maximal-size usage is ~150KB.)
+    //
+    // You can modify the wrapper functions in the source (setup_malloc,
+    // setup_temp_malloc, temp_malloc) to change this behavior, or you
+    // can use a simpler allocation model: you pass in a buffer from
+    // which stb_vorbis will allocate _all_ its memory (including the
+    // temp memory). "open" may fail with a VORBIS_outofmem if you
+    // do not pass in enough data; there is no way to determine how
+    // much you do need except to succeed (at which point you can
+    // query get_info to find the exact amount required. yes I know
+    // this is lame).
+    //
+    // If you pass in a non-NULL buffer of the type below, allocation
+    // will occur from it as described above. Otherwise just pass NULL
+    // to use malloc()/alloca()
+
+    typedef struct
+    {
+        char *alloc_buffer;
+        int   alloc_buffer_length_in_bytes;
+    } stb_vorbis_alloc;
+
+
+    ///////////   FUNCTIONS USEABLE WITH ALL INPUT MODES
+
+    typedef struct stb_vorbis stb_vorbis;
+
+    typedef struct
+    {
+        unsigned int sample_rate;
+        int channels;
+
+        unsigned int setup_memory_required;
+        unsigned int setup_temp_memory_required;
+        unsigned int temp_memory_required;
+
+        int max_frame_size;
+    } stb_vorbis_info;
+
+    // get general information about the file
+    extern stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f);
+
+    // get the last error detected (clears it, too)
+    extern int stb_vorbis_get_error(stb_vorbis *f);
+
+    // close an ogg vorbis file and free all memory in use
+    extern void stb_vorbis_close(stb_vorbis *f);
+
+    // this function returns the offset (in samples) from the beginning of the
+    // file that will be returned by the next decode, if it is known, or -1
+    // otherwise. after a flush_pushdata() call, this may take a while before
+    // it becomes valid again.
+    // NOT WORKING YET after a seek with PULLDATA API
+    extern int stb_vorbis_get_sample_offset(stb_vorbis *f);
+
+    // returns the current seek point within the file, or offset from the beginning
+    // of the memory buffer. In pushdata mode it returns 0.
+    extern unsigned int stb_vorbis_get_file_offset(stb_vorbis *f);
+
+    ///////////   PUSHDATA API
+
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+
+    // this API allows you to get blocks of data from any source and hand
+    // them to stb_vorbis. you have to buffer them; stb_vorbis will tell
+    // you how much it used, and you have to give it the rest next time;
+    // and stb_vorbis may not have enough data to work with and you will
+    // need to give it the same data again PLUS more. Note that the Vorbis
+    // specification does not bound the size of an individual frame.
+
+    extern stb_vorbis *stb_vorbis_open_pushdata(
+        const unsigned char * datablock, int datablock_length_in_bytes,
+        int *datablock_memory_consumed_in_bytes,
+        int *error,
+        const stb_vorbis_alloc *alloc_buffer);
+    // create a vorbis decoder by passing in the initial data block containing
+    //    the ogg&vorbis headers (you don't need to do parse them, just provide
+    //    the first N bytes of the file--you're told if it's not enough, see below)
+    // on success, returns an stb_vorbis *, does not set error, returns the amount of
+    //    data parsed/consumed on this call in *datablock_memory_consumed_in_bytes;
+    // on failure, returns NULL on error and sets *error, does not change *datablock_memory_consumed
+    // if returns NULL and *error is VORBIS_need_more_data, then the input block was
+    //       incomplete and you need to pass in a larger block from the start of the file
+
+    extern int stb_vorbis_decode_frame_pushdata(
+        stb_vorbis *f,
+        const unsigned char *datablock, int datablock_length_in_bytes,
+        int *channels,             // place to write number of float * buffers
+        float ***output,           // place to write float ** array of float * buffers
+        int *samples               // place to write number of output samples
+    );
+    // decode a frame of audio sample data if possible from the passed-in data block
+    //
+    // return value: number of bytes we used from datablock
+    //
+    // possible cases:
+    //     0 bytes used, 0 samples output (need more data)
+    //     N bytes used, 0 samples output (resynching the stream, keep going)
+    //     N bytes used, M samples output (one frame of data)
+    // note that after opening a file, you will ALWAYS get one N-bytes,0-sample
+    // frame, because Vorbis always "discards" the first frame.
+    //
+    // Note that on resynch, stb_vorbis will rarely consume all of the buffer,
+    // instead only datablock_length_in_bytes-3 or less. This is because it wants
+    // to avoid missing parts of a page header if they cross a datablock boundary,
+    // without writing state-machiney code to record a partial detection.
+    //
+    // The number of channels returned are stored in *channels (which can be
+    // NULL--it is always the same as the number of channels reported by
+    // get_info). *output will contain an array of float* buffers, one per
+    // channel. In other words, (*output)[0][0] contains the first sample from
+    // the first channel, and (*output)[1][0] contains the first sample from
+    // the second channel.
+
+    extern void stb_vorbis_flush_pushdata(stb_vorbis *f);
+    // inform stb_vorbis that your next datablock will not be contiguous with
+    // previous ones (e.g. you've seeked in the data); future attempts to decode
+    // frames will cause stb_vorbis to resynchronize (as noted above), and
+    // once it sees a valid Ogg page (typically 4-8KB, as large as 64KB), it
+    // will begin decoding the _next_ frame.
+    //
+    // if you want to seek using pushdata, you need to seek in your file, then
+    // call stb_vorbis_flush_pushdata(), then start calling decoding, then once
+    // decoding is returning you data, call stb_vorbis_get_sample_offset, and
+    // if you don't like the result, seek your file again and repeat.
+#endif
+
+
+    //////////   PULLING INPUT API
+
+#ifndef STB_VORBIS_NO_PULLDATA_API
+    // This API assumes stb_vorbis is allowed to pull data from a source--
+    // either a block of memory containing the _entire_ vorbis stream, or a
+    // FILE * that you or it create, or possibly some other reading mechanism
+    // if you go modify the source to replace the FILE * case with some kind
+    // of callback to your code. (But if you don't support seeking, you may
+    // just want to go ahead and use pushdata.)
+
+#if !defined(STB_VORBIS_NO_STDIO) && !defined(STB_VORBIS_NO_INTEGER_CONVERSION)
+    extern int stb_vorbis_decode_filename(const char *filename, int *channels, int *sample_rate, short **output);
+#endif
+#if !defined(STB_VORBIS_NO_INTEGER_CONVERSION)
+    extern int stb_vorbis_decode_memory(const unsigned char *mem, int len, int *channels, int *sample_rate, short **output);
+#endif
+    // decode an entire file and output the data interleaved into a malloc()ed
+    // buffer stored in *output. The return value is the number of samples
+    // decoded, or -1 if the file could not be opened or was not an ogg vorbis file.
+    // When you're done with it, just free() the pointer returned in *output.
+
+    extern stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len,
+        int *error, const stb_vorbis_alloc *alloc_buffer);
+    // create an ogg vorbis decoder from an ogg vorbis stream in memory (note
+    // this must be the entire stream!). on failure, returns NULL and sets *error
+
+#ifndef STB_VORBIS_NO_STDIO
+    extern stb_vorbis * stb_vorbis_open_filename(const char *filename,
+        int *error, const stb_vorbis_alloc *alloc_buffer);
+    // create an ogg vorbis decoder from a filename via fopen(). on failure,
+    // returns NULL and sets *error (possibly to VORBIS_file_open_failure).
+
+    extern stb_vorbis * stb_vorbis_open_file(FILE *f, int close_handle_on_close,
+        int *error, const stb_vorbis_alloc *alloc_buffer);
+    // create an ogg vorbis decoder from an open FILE *, looking for a stream at
+    // the _current_ seek point (ftell). on failure, returns NULL and sets *error.
+    // note that stb_vorbis must "own" this stream; if you seek it in between
+    // calls to stb_vorbis, it will become confused. Morever, if you attempt to
+    // perform stb_vorbis_seek_*() operations on this file, it will assume it
+    // owns the _entire_ rest of the file after the start point. Use the next
+    // function, stb_vorbis_open_file_section(), to limit it.
+
+    extern stb_vorbis * stb_vorbis_open_file_section(FILE *f, int close_handle_on_close,
+        int *error, const stb_vorbis_alloc *alloc_buffer, unsigned int len);
+    // create an ogg vorbis decoder from an open FILE *, looking for a stream at
+    // the _current_ seek point (ftell); the stream will be of length 'len' bytes.
+    // on failure, returns NULL and sets *error. note that stb_vorbis must "own"
+    // this stream; if you seek it in between calls to stb_vorbis, it will become
+    // confused.
+#endif
+
+    extern int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number);
+    extern int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number);
+    // these functions seek in the Vorbis file to (approximately) 'sample_number'.
+    // after calling seek_frame(), the next call to get_frame_*() will include
+    // the specified sample. after calling stb_vorbis_seek(), the next call to
+    // stb_vorbis_get_samples_* will start with the specified sample. If you
+    // do not need to seek to EXACTLY the target sample when using get_samples_*,
+    // you can also use seek_frame().
+
+    extern int stb_vorbis_seek_start(stb_vorbis *f);
+    // this function is equivalent to stb_vorbis_seek(f,0)
+
+    extern unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f);
+    extern float        stb_vorbis_stream_length_in_seconds(stb_vorbis *f);
+    // these functions return the total length of the vorbis stream
+
+    extern int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output);
+    // decode the next frame and return the number of samples. the number of
+    // channels returned are stored in *channels (which can be NULL--it is always
+    // the same as the number of channels reported by get_info). *output will
+    // contain an array of float* buffers, one per channel. These outputs will
+    // be overwritten on the next call to stb_vorbis_get_frame_*.
+    //
+    // You generally should not intermix calls to stb_vorbis_get_frame_*()
+    // and stb_vorbis_get_samples_*(), since the latter calls the former.
+
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+    extern int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts);
+    extern int stb_vorbis_get_frame_short(stb_vorbis *f, int num_c, short **buffer, int num_samples);
+#endif
+    // decode the next frame and return the number of *samples* per channel.
+    // Note that for interleaved data, you pass in the number of shorts (the
+    // size of your array), but the return value is the number of samples per
+    // channel, not the total number of samples.
+    //
+    // The data is coerced to the number of channels you request according to the
+    // channel coercion rules (see below). You must pass in the size of your
+    // buffer(s) so that stb_vorbis will not overwrite the end of the buffer.
+    // The maximum buffer size needed can be gotten from get_info(); however,
+    // the Vorbis I specification implies an absolute maximum of 4096 samples
+    // per channel.
+
+    // Channel coercion rules:
+    //    Let M be the number of channels requested, and N the number of channels present,
+    //    and Cn be the nth channel; let stereo L be the sum of all L and center channels,
+    //    and stereo R be the sum of all R and center channels (channel assignment from the
+    //    vorbis spec).
+    //        M    N       output
+    //        1    k      sum(Ck) for all k
+    //        2    *      stereo L, stereo R
+    //        k    l      k > l, the first l channels, then 0s
+    //        k    l      k <= l, the first k channels
+    //    Note that this is not _good_ surround etc. mixing at all! It's just so
+    //    you get something useful.
+
+    extern int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats);
+    extern int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples);
+    // gets num_samples samples, not necessarily on a frame boundary--this requires
+    // buffering so you have to supply the buffers. DOES NOT APPLY THE COERCION RULES.
+    // Returns the number of samples stored per channel; it may be less than requested
+    // at the end of the file. If there are no more samples in the file, returns 0.
+
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+    extern int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts);
+    extern int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int num_samples);
+#endif
+    // gets num_samples samples, not necessarily on a frame boundary--this requires
+    // buffering so you have to supply the buffers. Applies the coercion rules above
+    // to produce 'channels' channels. Returns the number of samples stored per channel;
+    // it may be less than requested at the end of the file. If there are no more
+    // samples in the file, returns 0.
+
+#endif
+
+    ////////   ERROR CODES
+
+    enum STBVorbisError
+    {
+        VORBIS__no_error,
+
+        VORBIS_need_more_data = 1,             // not a real error
+
+        VORBIS_invalid_api_mixing,           // can't mix API modes
+        VORBIS_outofmem,                     // not enough memory
+        VORBIS_feature_not_supported,        // uses floor 0
+        VORBIS_too_many_channels,            // STB_VORBIS_MAX_CHANNELS is too small
+        VORBIS_file_open_failure,            // fopen() failed
+        VORBIS_seek_without_length,          // can't seek in unknown-length file
+
+        VORBIS_unexpected_eof = 10,            // file is truncated?
+        VORBIS_seek_invalid,                 // seek past EOF
+
+                                             // decoding errors (corrupt/invalid stream) -- you probably
+                                             // don't care about the exact details of these
+
+                                             // vorbis errors:
+                                             VORBIS_invalid_setup = 20,
+                                             VORBIS_invalid_stream,
+
+                                             // ogg errors:
+                                             VORBIS_missing_capture_pattern = 30,
+                                             VORBIS_invalid_stream_structure_version,
+                                             VORBIS_continued_packet_flag_invalid,
+                                             VORBIS_incorrect_stream_serial_number,
+                                             VORBIS_invalid_first_page,
+                                             VORBIS_bad_packet_type,
+                                             VORBIS_cant_find_last_page,
+                                             VORBIS_seek_failed
+    };
+
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif // STB_VORBIS_INCLUDE_STB_VORBIS_H
+//
+//  HEADER ENDS HERE
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#ifndef STB_VORBIS_HEADER_ONLY
+
+// global configuration settings (e.g. set these in the project/makefile),
+// or just set them in this file at the top (although ideally the first few
+// should be visible when the header file is compiled too, although it's not
+// crucial)
+
+// STB_VORBIS_NO_PUSHDATA_API
+//     does not compile the code for the various stb_vorbis_*_pushdata()
+//     functions
+// #define STB_VORBIS_NO_PUSHDATA_API
+
+// STB_VORBIS_NO_PULLDATA_API
+//     does not compile the code for the non-pushdata APIs
+// #define STB_VORBIS_NO_PULLDATA_API
+
+// STB_VORBIS_NO_STDIO
+//     does not compile the code for the APIs that use FILE *s internally
+//     or externally (implied by STB_VORBIS_NO_PULLDATA_API)
+// #define STB_VORBIS_NO_STDIO
+
+// STB_VORBIS_NO_INTEGER_CONVERSION
+//     does not compile the code for converting audio sample data from
+//     float to integer (implied by STB_VORBIS_NO_PULLDATA_API)
+// #define STB_VORBIS_NO_INTEGER_CONVERSION
+
+// STB_VORBIS_NO_FAST_SCALED_FLOAT
+//      does not use a fast float-to-int trick to accelerate float-to-int on
+//      most platforms which requires endianness be defined correctly.
+//#define STB_VORBIS_NO_FAST_SCALED_FLOAT
+
+
+// STB_VORBIS_MAX_CHANNELS [number]
+//     globally define this to the maximum number of channels you need.
+//     The spec does not put a restriction on channels except that
+//     the count is stored in a byte, so 255 is the hard limit.
+//     Reducing this saves about 16 bytes per value, so using 16 saves
+//     (255-16)*16 or around 4KB. Plus anything other memory usage
+//     I forgot to account for. Can probably go as low as 8 (7.1 audio),
+//     6 (5.1 audio), or 2 (stereo only).
+#ifndef STB_VORBIS_MAX_CHANNELS
+#define STB_VORBIS_MAX_CHANNELS    16  // enough for anyone?
+#endif
+
+// STB_VORBIS_PUSHDATA_CRC_COUNT [number]
+//     after a flush_pushdata(), stb_vorbis begins scanning for the
+//     next valid page, without backtracking. when it finds something
+//     that looks like a page, it streams through it and verifies its
+//     CRC32. Should that validation fail, it keeps scanning. But it's
+//     possible that _while_ streaming through to check the CRC32 of
+//     one candidate page, it sees another candidate page. This #define
+//     determines how many "overlapping" candidate pages it can search
+//     at once. Note that "real" pages are typically ~4KB to ~8KB, whereas
+//     garbage pages could be as big as 64KB, but probably average ~16KB.
+//     So don't hose ourselves by scanning an apparent 64KB page and
+//     missing a ton of real ones in the interim; so minimum of 2
+#ifndef STB_VORBIS_PUSHDATA_CRC_COUNT
+#define STB_VORBIS_PUSHDATA_CRC_COUNT  4
+#endif
+
+// STB_VORBIS_FAST_HUFFMAN_LENGTH [number]
+//     sets the log size of the huffman-acceleration table.  Maximum
+//     supported value is 24. with larger numbers, more decodings are O(1),
+//     but the table size is larger so worse cache missing, so you'll have
+//     to probe (and try multiple ogg vorbis files) to find the sweet spot.
+#ifndef STB_VORBIS_FAST_HUFFMAN_LENGTH
+#define STB_VORBIS_FAST_HUFFMAN_LENGTH   10
+#endif
+
+// STB_VORBIS_FAST_BINARY_LENGTH [number]
+//     sets the log size of the binary-search acceleration table. this
+//     is used in similar fashion to the fast-huffman size to set initial
+//     parameters for the binary search
+
+// STB_VORBIS_FAST_HUFFMAN_INT
+//     The fast huffman tables are much more efficient if they can be
+//     stored as 16-bit results instead of 32-bit results. This restricts
+//     the codebooks to having only 65535 possible outcomes, though.
+//     (At least, accelerated by the huffman table.)
+#ifndef STB_VORBIS_FAST_HUFFMAN_INT
+#define STB_VORBIS_FAST_HUFFMAN_SHORT
+#endif
+
+// STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH
+//     If the 'fast huffman' search doesn't succeed, then stb_vorbis falls
+//     back on binary searching for the correct one. This requires storing
+//     extra tables with the huffman codes in sorted order. Defining this
+//     symbol trades off space for speed by forcing a linear search in the
+//     non-fast case, except for "sparse" codebooks.
+// #define STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH
+
+// STB_VORBIS_DIVIDES_IN_RESIDUE
+//     stb_vorbis precomputes the result of the scalar residue decoding
+//     that would otherwise require a divide per chunk. you can trade off
+//     space for time by defining this symbol.
+// #define STB_VORBIS_DIVIDES_IN_RESIDUE
+
+// STB_VORBIS_DIVIDES_IN_CODEBOOK
+//     vorbis VQ codebooks can be encoded two ways: with every case explicitly
+//     stored, or with all elements being chosen from a small range of values,
+//     and all values possible in all elements. By default, stb_vorbis expands
+//     this latter kind out to look like the former kind for ease of decoding,
+//     because otherwise an integer divide-per-vector-element is required to
+//     unpack the index. If you define STB_VORBIS_DIVIDES_IN_CODEBOOK, you can
+//     trade off storage for speed.
+//#define STB_VORBIS_DIVIDES_IN_CODEBOOK
+
+#ifdef STB_VORBIS_CODEBOOK_SHORTS
+#error "STB_VORBIS_CODEBOOK_SHORTS is no longer supported as it produced incorrect results for some input formats"
+#endif
+
+// STB_VORBIS_DIVIDE_TABLE
+//     this replaces small integer divides in the floor decode loop with
+//     table lookups. made less than 1% difference, so disabled by default.
+
+// STB_VORBIS_NO_INLINE_DECODE
+//     disables the inlining of the scalar codebook fast-huffman decode.
+//     might save a little codespace; useful for debugging
+// #define STB_VORBIS_NO_INLINE_DECODE
+
+// STB_VORBIS_NO_DEFER_FLOOR
+//     Normally we only decode the floor without synthesizing the actual
+//     full curve. We can instead synthesize the curve immediately. This
+//     requires more memory and is very likely slower, so I don't think
+//     you'd ever want to do it except for debugging.
+// #define STB_VORBIS_NO_DEFER_FLOOR
+
+
+
+
+//////////////////////////////////////////////////////////////////////////////
+
+#ifdef STB_VORBIS_NO_PULLDATA_API
+#define STB_VORBIS_NO_INTEGER_CONVERSION
+#define STB_VORBIS_NO_STDIO
+#endif
+
+#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO)
+#define STB_VORBIS_NO_STDIO 1
+#endif
+
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT
+
+// only need endianness for fast-float-to-int, which we don't
+// use for pushdata
+
+#ifndef STB_VORBIS_BIG_ENDIAN
+#define STB_VORBIS_ENDIAN  0
+#else
+#define STB_VORBIS_ENDIAN  1
+#endif
+
+#endif
+#endif
+
+
+#ifndef STB_VORBIS_NO_STDIO
+#include <stdio.h>
+#endif
+
+#ifndef STB_VORBIS_NO_CRT
+#include <stdlib.h>
+#include <string.h>
+#include <assert.h>
+#include <math.h>
+
+// find definition of alloca if it's not in stdlib.h:
+#if defined(_MSC_VER) || defined(__MINGW32__)
+#include <malloc.h>
+#endif
+#if defined(__linux__) || defined(__linux) || defined(__EMSCRIPTEN__)
+#include <alloca.h>
+#endif
+#else // STB_VORBIS_NO_CRT
+#define NULL 0
+#define malloc(s)   0
+#define free(s)     ((void) 0)
+#define realloc(s)  0
+#endif // STB_VORBIS_NO_CRT
+
+#include <limits.h>
+
+#ifdef __MINGW32__
+// eff you mingw:
+//     "fixed":
+//         http://sourceforge.net/p/mingw-w64/mailman/message/32882927/
+//     "no that broke the build, reverted, who cares about C":
+//         http://sourceforge.net/p/mingw-w64/mailman/message/32890381/
+#ifdef __forceinline
+#undef __forceinline
+#endif
+#define __forceinline
+#define alloca __builtin_alloca
+#elif !defined(_MSC_VER)
+#if __GNUC__
+#define __forceinline inline
+#else
+#define __forceinline
+#endif
+#endif
+
+#if STB_VORBIS_MAX_CHANNELS > 256
+#error "Value of STB_VORBIS_MAX_CHANNELS outside of allowed range"
+#endif
+
+#if STB_VORBIS_FAST_HUFFMAN_LENGTH > 24
+#error "Value of STB_VORBIS_FAST_HUFFMAN_LENGTH outside of allowed range"
+#endif
+
+
+#if 0
+#include <crtdbg.h>
+#define CHECK(f)   _CrtIsValidHeapPointer(f->channel_buffers[1])
+#else
+#define CHECK(f)   ((void) 0)
+#endif
+
+#define MAX_BLOCKSIZE_LOG  13   // from specification
+#define MAX_BLOCKSIZE      (1 << MAX_BLOCKSIZE_LOG)
+
+
+typedef unsigned char  uint8;
+typedef   signed char   int8;
+typedef unsigned short uint16;
+typedef   signed short  int16;
+typedef unsigned int   uint32;
+typedef   signed int    int32;
+
+#ifndef TRUE
+#define TRUE 1
+#define FALSE 0
+#endif
+
+typedef float codetype;
+
+// @NOTE
+//
+// Some arrays below are tagged "//varies", which means it's actually
+// a variable-sized piece of data, but rather than malloc I assume it's
+// small enough it's better to just allocate it all together with the
+// main thing
+//
+// Most of the variables are specified with the smallest size I could pack
+// them into. It might give better performance to make them all full-sized
+// integers. It should be safe to freely rearrange the structures or change
+// the sizes larger--nothing relies on silently truncating etc., nor the
+// order of variables.
+
+#define FAST_HUFFMAN_TABLE_SIZE   (1 << STB_VORBIS_FAST_HUFFMAN_LENGTH)
+#define FAST_HUFFMAN_TABLE_MASK   (FAST_HUFFMAN_TABLE_SIZE - 1)
+
+typedef struct
+{
+    int dimensions, entries;
+    uint8 *codeword_lengths;
+    float  minimum_value;
+    float  delta_value;
+    uint8  value_bits;
+    uint8  lookup_type;
+    uint8  sequence_p;
+    uint8  sparse;
+    uint32 lookup_values;
+    codetype *multiplicands;
+    uint32 *codewords;
+#ifdef STB_VORBIS_FAST_HUFFMAN_SHORT
+    int16  fast_huffman[FAST_HUFFMAN_TABLE_SIZE];
+#else
+    int32  fast_huffman[FAST_HUFFMAN_TABLE_SIZE];
+#endif
+    uint32 *sorted_codewords;
+    int    *sorted_values;
+    int     sorted_entries;
+} Codebook;
+
+typedef struct
+{
+    uint8 order;
+    uint16 rate;
+    uint16 bark_map_size;
+    uint8 amplitude_bits;
+    uint8 amplitude_offset;
+    uint8 number_of_books;
+    uint8 book_list[16]; // varies
+} Floor0;
+
+typedef struct
+{
+    uint8 partitions;
+    uint8 partition_class_list[32]; // varies
+    uint8 class_dimensions[16]; // varies
+    uint8 class_subclasses[16]; // varies
+    uint8 class_masterbooks[16]; // varies
+    int16 subclass_books[16][8]; // varies
+    uint16 Xlist[31 * 8 + 2]; // varies
+    uint8 sorted_order[31 * 8 + 2];
+    uint8 neighbors[31 * 8 + 2][2];
+    uint8 floor1_multiplier;
+    uint8 rangebits;
+    int values;
+} Floor1;
+
+typedef union
+{
+    Floor0 floor0;
+    Floor1 floor1;
+} Floor;
+
+typedef struct
+{
+    uint32 begin, end;
+    uint32 part_size;
+    uint8 classifications;
+    uint8 classbook;
+    uint8 **classdata;
+    int16(*residue_books)[8];
+} Residue;
+
+typedef struct
+{
+    uint8 magnitude;
+    uint8 angle;
+    uint8 mux;
+} MappingChannel;
+
+typedef struct
+{
+    uint16 coupling_steps;
+    MappingChannel *chan;
+    uint8  submaps;
+    uint8  submap_floor[15]; // varies
+    uint8  submap_residue[15]; // varies
+} Mapping;
+
+typedef struct
+{
+    uint8 blockflag;
+    uint8 mapping;
+    uint16 windowtype;
+    uint16 transformtype;
+} Mode;
+
+typedef struct
+{
+    uint32  goal_crc;    // expected crc if match
+    int     bytes_left;  // bytes left in packet
+    uint32  crc_so_far;  // running crc
+    int     bytes_done;  // bytes processed in _current_ chunk
+    uint32  sample_loc;  // granule pos encoded in page
+} CRCscan;
+
+typedef struct
+{
+    uint32 page_start, page_end;
+    uint32 last_decoded_sample;
+} ProbedPage;
+
+struct stb_vorbis
+{
+    // user-accessible info
+    unsigned int sample_rate;
+    int channels;
+
+    unsigned int setup_memory_required;
+    unsigned int temp_memory_required;
+    unsigned int setup_temp_memory_required;
+
+    // input config
+#ifndef STB_VORBIS_NO_STDIO
+    FILE *f;
+    uint32 f_start;
+    int close_on_free;
+#endif
+
+    uint8 *stream;
+    uint8 *stream_start;
+    uint8 *stream_end;
+
+    uint32 stream_len;
+
+    uint8  push_mode;
+
+    uint32 first_audio_page_offset;
+
+    ProbedPage p_first, p_last;
+
+    // memory management
+    stb_vorbis_alloc alloc;
+    int setup_offset;
+    int temp_offset;
+
+    // run-time results
+    int eof;
+    enum STBVorbisError error;
+
+    // user-useful data
+
+    // header info
+    int blocksize[2];
+    int blocksize_0, blocksize_1;
+    int codebook_count;
+    Codebook *codebooks;
+    int floor_count;
+    uint16 floor_types[64]; // varies
+    Floor *floor_config;
+    int residue_count;
+    uint16 residue_types[64]; // varies
+    Residue *residue_config;
+    int mapping_count;
+    Mapping *mapping;
+    int mode_count;
+    Mode mode_config[64];  // varies
+
+    uint32 total_samples;
+
+    // decode buffer
+    float *channel_buffers[STB_VORBIS_MAX_CHANNELS];
+    float *outputs[STB_VORBIS_MAX_CHANNELS];
+
+    float *previous_window[STB_VORBIS_MAX_CHANNELS];
+    int previous_length;
+
+#ifndef STB_VORBIS_NO_DEFER_FLOOR
+    int16 *finalY[STB_VORBIS_MAX_CHANNELS];
+#else
+    float *floor_buffers[STB_VORBIS_MAX_CHANNELS];
+#endif
+
+    uint32 current_loc; // sample location of next frame to decode
+    int    current_loc_valid;
+
+    // per-blocksize precomputed data
+
+    // twiddle factors
+    float *A[2], *B[2], *C[2];
+    float *window[2];
+    uint16 *bit_reverse[2];
+
+    // current page/packet/segment streaming info
+    uint32 serial; // stream serial number for verification
+    int last_page;
+    int segment_count;
+    uint8 segments[255];
+    uint8 page_flag;
+    uint8 bytes_in_seg;
+    uint8 first_decode;
+    int next_seg;
+    int last_seg;  // flag that we're on the last segment
+    int last_seg_which; // what was the segment number of the last seg?
+    uint32 acc;
+    int valid_bits;
+    int packet_bytes;
+    int end_seg_with_known_loc;
+    uint32 known_loc_for_packet;
+    int discard_samples_deferred;
+    uint32 samples_output;
+
+    // push mode scanning
+    int page_crc_tests; // only in push_mode: number of tests active; -1 if not searching
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+    CRCscan scan[STB_VORBIS_PUSHDATA_CRC_COUNT];
+#endif
+
+    // sample-access
+    int channel_buffer_start;
+    int channel_buffer_end;
+};
+
+#if defined(STB_VORBIS_NO_PUSHDATA_API)
+#define IS_PUSH_MODE(f)   FALSE
+#elif defined(STB_VORBIS_NO_PULLDATA_API)
+#define IS_PUSH_MODE(f)   TRUE
+#else
+#define IS_PUSH_MODE(f)   ((f)->push_mode)
+#endif
+
+typedef struct stb_vorbis vorb;
+
+static int error(vorb *f, enum STBVorbisError e)
+{
+    f->error = e;
+    if (!f->eof && e != VORBIS_need_more_data) {
+        f->error = e; // breakpoint for debugging
+    }
+    return 0;
+}
+
+
+// these functions are used for allocating temporary memory
+// while decoding. if you can afford the stack space, use
+// alloca(); otherwise, provide a temp buffer and it will
+// allocate out of those.
+
+#define array_size_required(count,size)  (count*(sizeof(void *)+(size)))
+
+#define temp_alloc(f,size)              (f->alloc.alloc_buffer ? setup_temp_malloc(f,size) : alloca(size))
+#define temp_free(f,p)                  0
+#define temp_alloc_save(f)              ((f)->temp_offset)
+#define temp_alloc_restore(f,p)         ((f)->temp_offset = (p))
+
+#define temp_block_array(f,count,size)  make_block_array(temp_alloc(f,array_size_required(count,size)), count, size)
+
+// given a sufficiently large block of memory, make an array of pointers to subblocks of it
+static void *make_block_array(void *mem, int count, int size)
+{
+    int i;
+    void ** p = (void **)mem;
+    char *q = (char *)(p + count);
+    for (i = 0; i < count; ++i) {
+        p[i] = q;
+        q += size;
+    }
+    return p;
+}
+
+static void *setup_malloc(vorb *f, int sz)
+{
+    sz = (sz + 3) & ~3;
+    f->setup_memory_required += sz;
+    if (f->alloc.alloc_buffer) {
+        void *p = (char *)f->alloc.alloc_buffer + f->setup_offset;
+        if (f->setup_offset + sz > f->temp_offset) return NULL;
+        f->setup_offset += sz;
+        return p;
+    }
+    return sz ? malloc(sz) : NULL;
+}
+
+static void setup_free(vorb *f, void *p)
+{
+    if (f->alloc.alloc_buffer) return; // do nothing; setup mem is a stack
+    free(p);
+}
+
+static void *setup_temp_malloc(vorb *f, int sz)
+{
+    sz = (sz + 3) & ~3;
+    if (f->alloc.alloc_buffer) {
+        if (f->temp_offset - sz < f->setup_offset) return NULL;
+        f->temp_offset -= sz;
+        return (char *)f->alloc.alloc_buffer + f->temp_offset;
+    }
+    return malloc(sz);
+}
+
+static void setup_temp_free(vorb *f, void *p, int sz)
+{
+    if (f->alloc.alloc_buffer) {
+        f->temp_offset += (sz + 3)&~3;
+        return;
+    }
+    free(p);
+}
+
+#define CRC32_POLY    0x04c11db7   // from spec
+
+static uint32 crc_table[256];
+static void crc32_init(void)
+{
+    int i, j;
+    uint32 s;
+    for (i = 0; i < 256; i++) {
+        for (s = (uint32)i << 24, j = 0; j < 8; ++j)
+            s = (s << 1) ^ (s >= (1U << 31) ? CRC32_POLY : 0);
+        crc_table[i] = s;
+    }
+}
+
+static __forceinline uint32 crc32_update(uint32 crc, uint8 byte)
+{
+    return (crc << 8) ^ crc_table[byte ^ (crc >> 24)];
+}
+
+
+// used in setup, and for huffman that doesn't go fast path
+static unsigned int bit_reverse(unsigned int n)
+{
+    n = ((n & 0xAAAAAAAA) >> 1) | ((n & 0x55555555) << 1);
+    n = ((n & 0xCCCCCCCC) >> 2) | ((n & 0x33333333) << 2);
+    n = ((n & 0xF0F0F0F0) >> 4) | ((n & 0x0F0F0F0F) << 4);
+    n = ((n & 0xFF00FF00) >> 8) | ((n & 0x00FF00FF) << 8);
+    return (n >> 16) | (n << 16);
+}
+
+static float square(float x)
+{
+    return x*x;
+}
+
+// this is a weird definition of log2() for which log2(1) = 1, log2(2) = 2, log2(4) = 3
+// as required by the specification. fast(?) implementation from stb.h
+// @OPTIMIZE: called multiple times per-packet with "constants"; move to setup
+static int ilog(int32 n)
+{
+    static signed char log2_4[16] = { 0,1,2,2,3,3,3,3,4,4,4,4,4,4,4,4 };
+
+    if (n < 0) return 0; // signed n returns 0
+
+                         // 2 compares if n < 16, 3 compares otherwise (4 if signed or n > 1<<29)
+    if (n < (1 << 14))
+        if (n < (1 << 4))            return  0 + log2_4[n];
+        else if (n < (1 << 9))       return  5 + log2_4[n >> 5];
+        else                     return 10 + log2_4[n >> 10];
+    else if (n < (1 << 24))
+        if (n < (1 << 19))       return 15 + log2_4[n >> 15];
+        else                     return 20 + log2_4[n >> 20];
+    else if (n < (1 << 29))       return 25 + log2_4[n >> 25];
+    else                     return 30 + log2_4[n >> 30];
+}
+
+#ifndef M_PI
+#define M_PI  3.14159265358979323846264f  // from CRC
+#endif
+
+// code length assigned to a value with no huffman encoding
+#define NO_CODE   255
+
+/////////////////////// LEAF SETUP FUNCTIONS //////////////////////////
+//
+// these functions are only called at setup, and only a few times
+// per file
+
+static float float32_unpack(uint32 x)
+{
+    // from the specification
+    uint32 mantissa = x & 0x1fffff;
+    uint32 sign = x & 0x80000000;
+    uint32 exp = (x & 0x7fe00000) >> 21;
+    double res = sign ? -(double)mantissa : (double)mantissa;
+    return (float)ldexp((float)res, exp - 788);
+}
+
+
+// zlib & jpeg huffman tables assume that the output symbols
+// can either be arbitrarily arranged, or have monotonically
+// increasing frequencies--they rely on the lengths being sorted;
+// this makes for a very simple generation algorithm.
+// vorbis allows a huffman table with non-sorted lengths. This
+// requires a more sophisticated construction, since symbols in
+// order do not map to huffman codes "in order".
+static void add_entry(Codebook *c, uint32 huff_code, int symbol, int count, int len, uint32 *values)
+{
+    if (!c->sparse) {
+        c->codewords[symbol] = huff_code;
+    }
+    else {
+        c->codewords[count] = huff_code;
+        c->codeword_lengths[count] = len;
+        values[count] = symbol;
+    }
+}
+
+static int compute_codewords(Codebook *c, uint8 *len, int n, uint32 *values)
+{
+    int i, k, m = 0;
+    uint32 available[32];
+
+    memset(available, 0, sizeof(available));
+    // find the first entry
+    for (k = 0; k < n; ++k) if (len[k] < NO_CODE) break;
+    if (k == n) { assert(c->sorted_entries == 0); return TRUE; }
+    // add to the list
+    add_entry(c, 0, k, m++, len[k], values);
+    // add all available leaves
+    for (i = 1; i <= len[k]; ++i)
+        available[i] = 1U << (32 - i);
+    // note that the above code treats the first case specially,
+    // but it's really the same as the following code, so they
+    // could probably be combined (except the initial code is 0,
+    // and I use 0 in available[] to mean 'empty')
+    for (i = k + 1; i < n; ++i) {
+        uint32 res;
+        int z = len[i], y;
+        if (z == NO_CODE) continue;
+        // find lowest available leaf (should always be earliest,
+        // which is what the specification calls for)
+        // note that this property, and the fact we can never have
+        // more than one free leaf at a given level, isn't totally
+        // trivial to prove, but it seems true and the assert never
+        // fires, so!
+        while (z > 0 && !available[z]) --z;
+        if (z == 0) { return FALSE; }
+        res = available[z];
+        assert(z >= 0 && z < 32);
+        available[z] = 0;
+        add_entry(c, bit_reverse(res), i, m++, len[i], values);
+        // propogate availability up the tree
+        if (z != len[i]) {
+            assert(len[i] >= 0 && len[i] < 32);
+            for (y = len[i]; y > z; --y) {
+                assert(available[y] == 0);
+                available[y] = res + (1 << (32 - y));
+            }
+        }
+    }
+    return TRUE;
+}
+
+// accelerated huffman table allows fast O(1) match of all symbols
+// of length <= STB_VORBIS_FAST_HUFFMAN_LENGTH
+static void compute_accelerated_huffman(Codebook *c)
+{
+    int i, len;
+    for (i = 0; i < FAST_HUFFMAN_TABLE_SIZE; ++i)
+        c->fast_huffman[i] = -1;
+
+    len = c->sparse ? c->sorted_entries : c->entries;
+#ifdef STB_VORBIS_FAST_HUFFMAN_SHORT
+    if (len > 32767) len = 32767; // largest possible value we can encode!
+#endif
+    for (i = 0; i < len; ++i) {
+        if (c->codeword_lengths[i] <= STB_VORBIS_FAST_HUFFMAN_LENGTH) {
+            uint32 z = c->sparse ? bit_reverse(c->sorted_codewords[i]) : c->codewords[i];
+            // set table entries for all bit combinations in the higher bits
+            while (z < FAST_HUFFMAN_TABLE_SIZE) {
+                c->fast_huffman[z] = i;
+                z += 1 << c->codeword_lengths[i];
+            }
+        }
+    }
+}
+
+#ifdef _MSC_VER
+#define STBV_CDECL __cdecl
+#else
+#define STBV_CDECL
+#endif
+
+static int STBV_CDECL uint32_compare(const void *p, const void *q)
+{
+    uint32 x = *(uint32 *)p;
+    uint32 y = *(uint32 *)q;
+    return x < y ? -1 : x > y;
+}
+
+static int include_in_sort(Codebook *c, uint8 len)
+{
+    if (c->sparse) { assert(len != NO_CODE); return TRUE; }
+    if (len == NO_CODE) return FALSE;
+    if (len > STB_VORBIS_FAST_HUFFMAN_LENGTH) return TRUE;
+    return FALSE;
+}
+
+// if the fast table above doesn't work, we want to binary
+// search them... need to reverse the bits
+static void compute_sorted_huffman(Codebook *c, uint8 *lengths, uint32 *values)
+{
+    int i, len;
+    // build a list of all the entries
+    // OPTIMIZATION: don't include the short ones, since they'll be caught by FAST_HUFFMAN.
+    // this is kind of a frivolous optimization--I don't see any performance improvement,
+    // but it's like 4 extra lines of code, so.
+    if (!c->sparse) {
+        int k = 0;
+        for (i = 0; i < c->entries; ++i)
+            if (include_in_sort(c, lengths[i]))
+                c->sorted_codewords[k++] = bit_reverse(c->codewords[i]);
+        assert(k == c->sorted_entries);
+    }
+    else {
+        for (i = 0; i < c->sorted_entries; ++i)
+            c->sorted_codewords[i] = bit_reverse(c->codewords[i]);
+    }
+
+    qsort(c->sorted_codewords, c->sorted_entries, sizeof(c->sorted_codewords[0]), uint32_compare);
+    c->sorted_codewords[c->sorted_entries] = 0xffffffff;
+
+    len = c->sparse ? c->sorted_entries : c->entries;
+    // now we need to indicate how they correspond; we could either
+    //   #1: sort a different data structure that says who they correspond to
+    //   #2: for each sorted entry, search the original list to find who corresponds
+    //   #3: for each original entry, find the sorted entry
+    // #1 requires extra storage, #2 is slow, #3 can use binary search!
+    for (i = 0; i < len; ++i) {
+        int huff_len = c->sparse ? lengths[values[i]] : lengths[i];
+        if (include_in_sort(c, huff_len)) {
+            uint32 code = bit_reverse(c->codewords[i]);
+            int x = 0, n = c->sorted_entries;
+            while (n > 1) {
+                // invariant: sc[x] <= code < sc[x+n]
+                int m = x + (n >> 1);
+                if (c->sorted_codewords[m] <= code) {
+                    x = m;
+                    n -= (n >> 1);
+                }
+                else {
+                    n >>= 1;
+                }
+            }
+            assert(c->sorted_codewords[x] == code);
+            if (c->sparse) {
+                c->sorted_values[x] = values[i];
+                c->codeword_lengths[x] = huff_len;
+            }
+            else {
+                c->sorted_values[x] = i;
+            }
+        }
+    }
+}
+
+// only run while parsing the header (3 times)
+static int vorbis_validate(uint8 *data)
+{
+    static uint8 vorbis[6] = { 'v', 'o', 'r', 'b', 'i', 's' };
+    return memcmp(data, vorbis, 6) == 0;
+}
+
+// called from setup only, once per code book
+// (formula implied by specification)
+static int lookup1_values(int entries, int dim)
+{
+    int r = (int)floor(exp((float)log((float)entries) / dim));
+    if ((int)floor(pow((float)r + 1, dim)) <= entries)   // (int) cast for MinGW warning;
+        ++r;                                              // floor() to avoid _ftol() when non-CRT
+    assert(pow((float)r + 1, dim) > entries);
+    assert((int)floor(pow((float)r, dim)) <= entries); // (int),floor() as above
+    return r;
+}
+
+// called twice per file
+static void compute_twiddle_factors(int n, float *A, float *B, float *C)
+{
+    int n4 = n >> 2, n8 = n >> 3;
+    int k, k2;
+
+    for (k = k2 = 0; k < n4; ++k, k2 += 2) {
+        A[k2] = (float)cos(4 * k*M_PI / n);
+        A[k2 + 1] = (float)-sin(4 * k*M_PI / n);
+        B[k2] = (float)cos((k2 + 1)*M_PI / n / 2) * 0.5f;
+        B[k2 + 1] = (float)sin((k2 + 1)*M_PI / n / 2) * 0.5f;
+    }
+    for (k = k2 = 0; k < n8; ++k, k2 += 2) {
+        C[k2] = (float)cos(2 * (k2 + 1)*M_PI / n);
+        C[k2 + 1] = (float)-sin(2 * (k2 + 1)*M_PI / n);
+    }
+}
+
+static void compute_window(int n, float *window)
+{
+    int n2 = n >> 1, i;
+    for (i = 0; i < n2; ++i)
+        window[i] = (float)sin(0.5 * M_PI * square((float)sin((i - 0 + 0.5) / n2 * 0.5 * M_PI)));
+}
+
+static void compute_bitreverse(int n, uint16 *rev)
+{
+    int ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
+    int i, n8 = n >> 3;
+    for (i = 0; i < n8; ++i)
+        rev[i] = (bit_reverse(i) >> (32 - ld + 3)) << 2;
+}
+
+static int init_blocksize(vorb *f, int b, int n)
+{
+    int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3;
+    f->A[b] = (float *)setup_malloc(f, sizeof(float) * n2);
+    f->B[b] = (float *)setup_malloc(f, sizeof(float) * n2);
+    f->C[b] = (float *)setup_malloc(f, sizeof(float) * n4);
+    if (!f->A[b] || !f->B[b] || !f->C[b]) return error(f, VORBIS_outofmem);
+    compute_twiddle_factors(n, f->A[b], f->B[b], f->C[b]);
+    f->window[b] = (float *)setup_malloc(f, sizeof(float) * n2);
+    if (!f->window[b]) return error(f, VORBIS_outofmem);
+    compute_window(n, f->window[b]);
+    f->bit_reverse[b] = (uint16 *)setup_malloc(f, sizeof(uint16) * n8);
+    if (!f->bit_reverse[b]) return error(f, VORBIS_outofmem);
+    compute_bitreverse(n, f->bit_reverse[b]);
+    return TRUE;
+}
+
+static void neighbors(uint16 *x, int n, int *plow, int *phigh)
+{
+    int low = -1;
+    int high = 65536;
+    int i;
+    for (i = 0; i < n; ++i) {
+        if (x[i] > low  && x[i] < x[n]) { *plow = i; low = x[i]; }
+        if (x[i] < high && x[i] > x[n]) { *phigh = i; high = x[i]; }
+    }
+}
+
+// this has been repurposed so y is now the original index instead of y
+typedef struct
+{
+    uint16 x, id;
+} stbv__floor_ordering;
+
+static int STBV_CDECL point_compare(const void *p, const void *q)
+{
+    stbv__floor_ordering *a = (stbv__floor_ordering *)p;
+    stbv__floor_ordering *b = (stbv__floor_ordering *)q;
+    return a->x < b->x ? -1 : a->x > b->x;
+}
+
+//
+/////////////////////// END LEAF SETUP FUNCTIONS //////////////////////////
+
+
+#if defined(STB_VORBIS_NO_STDIO)
+#define USE_MEMORY(z)    TRUE
+#else
+#define USE_MEMORY(z)    ((z)->stream)
+#endif
+
+static uint8 get8(vorb *z)
+{
+    if (USE_MEMORY(z)) {
+        if (z->stream >= z->stream_end) { z->eof = TRUE; return 0; }
+        return *z->stream++;
+    }
+
+#ifndef STB_VORBIS_NO_STDIO
+    {
+        int c = fgetc(z->f);
+        if (c == EOF) { z->eof = TRUE; return 0; }
+        return c;
+    }
+#endif
+}
+
+static uint32 get32(vorb *f)
+{
+    uint32 x;
+    x = get8(f);
+    x += get8(f) << 8;
+    x += get8(f) << 16;
+    x += (uint32)get8(f) << 24;
+    return x;
+}
+
+static int getn(vorb *z, uint8 *data, int n)
+{
+    if (USE_MEMORY(z)) {
+        if (z->stream + n > z->stream_end) { z->eof = 1; return 0; }
+        memcpy(data, z->stream, n);
+        z->stream += n;
+        return 1;
+    }
+
+#ifndef STB_VORBIS_NO_STDIO   
+    if (fread(data, n, 1, z->f) == 1)
+        return 1;
+    else {
+        z->eof = 1;
+        return 0;
+    }
+#endif
+}
+
+static void skip(vorb *z, int n)
+{
+    if (USE_MEMORY(z)) {
+        z->stream += n;
+        if (z->stream >= z->stream_end) z->eof = 1;
+        return;
+    }
+#ifndef STB_VORBIS_NO_STDIO
+    {
+        long x = ftell(z->f);
+        fseek(z->f, x + n, SEEK_SET);
+    }
+#endif
+}
+
+static int set_file_offset(stb_vorbis *f, unsigned int loc)
+{
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+    if (f->push_mode) return 0;
+#endif
+    f->eof = 0;
+    if (USE_MEMORY(f)) {
+        if (f->stream_start + loc >= f->stream_end || f->stream_start + loc < f->stream_start) {
+            f->stream = f->stream_end;
+            f->eof = 1;
+            return 0;
+        }
+        else {
+            f->stream = f->stream_start + loc;
+            return 1;
+        }
+    }
+#ifndef STB_VORBIS_NO_STDIO
+    if (loc + f->f_start < loc || loc >= 0x80000000) {
+        loc = 0x7fffffff;
+        f->eof = 1;
+    }
+    else {
+        loc += f->f_start;
+    }
+    if (!fseek(f->f, loc, SEEK_SET))
+        return 1;
+    f->eof = 1;
+    fseek(f->f, f->f_start, SEEK_END);
+    return 0;
+#endif
+}
+
+
+static uint8 ogg_page_header[4] = { 0x4f, 0x67, 0x67, 0x53 };
+
+static int capture_pattern(vorb *f)
+{
+    if (0x4f != get8(f)) return FALSE;
+    if (0x67 != get8(f)) return FALSE;
+    if (0x67 != get8(f)) return FALSE;
+    if (0x53 != get8(f)) return FALSE;
+    return TRUE;
+}
+
+#define PAGEFLAG_continued_packet   1
+#define PAGEFLAG_first_page         2
+#define PAGEFLAG_last_page          4
+
+static int start_page_no_capturepattern(vorb *f)
+{
+    uint32 loc0, loc1, n;
+    // stream structure version
+    if (0 != get8(f)) return error(f, VORBIS_invalid_stream_structure_version);
+    // header flag
+    f->page_flag = get8(f);
+    // absolute granule position
+    loc0 = get32(f);
+    loc1 = get32(f);
+    // @TODO: validate loc0,loc1 as valid positions?
+    // stream serial number -- vorbis doesn't interleave, so discard
+    get32(f);
+    //if (f->serial != get32(f)) return error(f, VORBIS_incorrect_stream_serial_number);
+    // page sequence number
+    n = get32(f);
+    f->last_page = n;
+    // CRC32
+    get32(f);
+    // page_segments
+    f->segment_count = get8(f);
+    if (!getn(f, f->segments, f->segment_count))
+        return error(f, VORBIS_unexpected_eof);
+    // assume we _don't_ know any the sample position of any segments
+    f->end_seg_with_known_loc = -2;
+    if (loc0 != ~0U || loc1 != ~0U) {
+        int i;
+        // determine which packet is the last one that will complete
+        for (i = f->segment_count - 1; i >= 0; --i)
+            if (f->segments[i] < 255)
+                break;
+        // 'i' is now the index of the _last_ segment of a packet that ends
+        if (i >= 0) {
+            f->end_seg_with_known_loc = i;
+            f->known_loc_for_packet = loc0;
+        }
+    }
+    if (f->first_decode) {
+        int i, len;
+        ProbedPage p;
+        len = 0;
+        for (i = 0; i < f->segment_count; ++i)
+            len += f->segments[i];
+        len += 27 + f->segment_count;
+        p.page_start = f->first_audio_page_offset;
+        p.page_end = p.page_start + len;
+        p.last_decoded_sample = loc0;
+        f->p_first = p;
+    }
+    f->next_seg = 0;
+    return TRUE;
+}
+
+static int start_page(vorb *f)
+{
+    if (!capture_pattern(f)) return error(f, VORBIS_missing_capture_pattern);
+    return start_page_no_capturepattern(f);
+}
+
+static int start_packet(vorb *f)
+{
+    while (f->next_seg == -1) {
+        if (!start_page(f)) return FALSE;
+        if (f->page_flag & PAGEFLAG_continued_packet)
+            return error(f, VORBIS_continued_packet_flag_invalid);
+    }
+    f->last_seg = FALSE;
+    f->valid_bits = 0;
+    f->packet_bytes = 0;
+    f->bytes_in_seg = 0;
+    // f->next_seg is now valid
+    return TRUE;
+}
+
+static int maybe_start_packet(vorb *f)
+{
+    if (f->next_seg == -1) {
+        int x = get8(f);
+        if (f->eof) return FALSE; // EOF at page boundary is not an error!
+        if (0x4f != x) return error(f, VORBIS_missing_capture_pattern);
+        if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
+        if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
+        if (0x53 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
+        if (!start_page_no_capturepattern(f)) return FALSE;
+        if (f->page_flag & PAGEFLAG_continued_packet) {
+            // set up enough state that we can read this packet if we want,
+            // e.g. during recovery
+            f->last_seg = FALSE;
+            f->bytes_in_seg = 0;
+            return error(f, VORBIS_continued_packet_flag_invalid);
+        }
+    }
+    return start_packet(f);
+}
+
+static int next_segment(vorb *f)
+{
+    int len;
+    if (f->last_seg) return 0;
+    if (f->next_seg == -1) {
+        f->last_seg_which = f->segment_count - 1; // in case start_page fails
+        if (!start_page(f)) { f->last_seg = 1; return 0; }
+        if (!(f->page_flag & PAGEFLAG_continued_packet)) return error(f, VORBIS_continued_packet_flag_invalid);
+    }
+    len = f->segments[f->next_seg++];
+    if (len < 255) {
+        f->last_seg = TRUE;
+        f->last_seg_which = f->next_seg - 1;
+    }
+    if (f->next_seg >= f->segment_count)
+        f->next_seg = -1;
+    assert(f->bytes_in_seg == 0);
+    f->bytes_in_seg = len;
+    return len;
+}
+
+#define EOP    (-1)
+#define INVALID_BITS  (-1)
+
+static int get8_packet_raw(vorb *f)
+{
+    if (!f->bytes_in_seg) {  // CLANG!
+        if (f->last_seg) return EOP;
+        else if (!next_segment(f)) return EOP;
+    }
+    assert(f->bytes_in_seg > 0);
+    --f->bytes_in_seg;
+    ++f->packet_bytes;
+    return get8(f);
+}
+
+static int get8_packet(vorb *f)
+{
+    int x = get8_packet_raw(f);
+    f->valid_bits = 0;
+    return x;
+}
+
+static void flush_packet(vorb *f)
+{
+    while (get8_packet_raw(f) != EOP);
+}
+
+// @OPTIMIZE: this is the secondary bit decoder, so it's probably not as important
+// as the huffman decoder?
+static uint32 get_bits(vorb *f, int n)
+{
+    uint32 z;
+
+    if (f->valid_bits < 0) return 0;
+    if (f->valid_bits < n) {
+        if (n > 24) {
+            // the accumulator technique below would not work correctly in this case
+            z = get_bits(f, 24);
+            z += get_bits(f, n - 24) << 24;
+            return z;
+        }
+        if (f->valid_bits == 0) f->acc = 0;
+        while (f->valid_bits < n) {
+            int z = get8_packet_raw(f);
+            if (z == EOP) {
+                f->valid_bits = INVALID_BITS;
+                return 0;
+            }
+            f->acc += z << f->valid_bits;
+            f->valid_bits += 8;
+        }
+    }
+    if (f->valid_bits < 0) return 0;
+    z = f->acc & ((1 << n) - 1);
+    f->acc >>= n;
+    f->valid_bits -= n;
+    return z;
+}
+
+// @OPTIMIZE: primary accumulator for huffman
+// expand the buffer to as many bits as possible without reading off end of packet
+// it might be nice to allow f->valid_bits and f->acc to be stored in registers,
+// e.g. cache them locally and decode locally
+static __forceinline void prep_huffman(vorb *f)
+{
+    if (f->valid_bits <= 24) {
+        if (f->valid_bits == 0) f->acc = 0;
+        do {
+            int z;
+            if (f->last_seg && !f->bytes_in_seg) return;
+            z = get8_packet_raw(f);
+            if (z == EOP) return;
+            f->acc += (unsigned)z << f->valid_bits;
+            f->valid_bits += 8;
+        } while (f->valid_bits <= 24);
+    }
+}
+
+enum
+{
+    VORBIS_packet_id = 1,
+    VORBIS_packet_comment = 3,
+    VORBIS_packet_setup = 5
+};
+
+static int codebook_decode_scalar_raw(vorb *f, Codebook *c)
+{
+    int i;
+    prep_huffman(f);
+
+    if (c->codewords == NULL && c->sorted_codewords == NULL)
+        return -1;
+
+    // cases to use binary search: sorted_codewords && !c->codewords
+    //                             sorted_codewords && c->entries > 8
+    if (c->entries > 8 ? c->sorted_codewords != NULL : !c->codewords) {
+        // binary search
+        uint32 code = bit_reverse(f->acc);
+        int x = 0, n = c->sorted_entries, len;
+
+        while (n > 1) {
+            // invariant: sc[x] <= code < sc[x+n]
+            int m = x + (n >> 1);
+            if (c->sorted_codewords[m] <= code) {
+                x = m;
+                n -= (n >> 1);
+            }
+            else {
+                n >>= 1;
+            }
+        }
+        // x is now the sorted index
+        if (!c->sparse) x = c->sorted_values[x];
+        // x is now sorted index if sparse, or symbol otherwise
+        len = c->codeword_lengths[x];
+        if (f->valid_bits >= len) {
+            f->acc >>= len;
+            f->valid_bits -= len;
+            return x;
+        }
+
+        f->valid_bits = 0;
+        return -1;
+    }
+
+    // if small, linear search
+    assert(!c->sparse);
+    for (i = 0; i < c->entries; ++i) {
+        if (c->codeword_lengths[i] == NO_CODE) continue;
+        if (c->codewords[i] == (f->acc & ((1 << c->codeword_lengths[i]) - 1))) {
+            if (f->valid_bits >= c->codeword_lengths[i]) {
+                f->acc >>= c->codeword_lengths[i];
+                f->valid_bits -= c->codeword_lengths[i];
+                return i;
+            }
+            f->valid_bits = 0;
+            return -1;
+        }
+    }
+
+    error(f, VORBIS_invalid_stream);
+    f->valid_bits = 0;
+    return -1;
+}
+
+#ifndef STB_VORBIS_NO_INLINE_DECODE
+
+#define DECODE_RAW(var, f,c)                                  \
+   if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH)        \
+      prep_huffman(f);                                        \
+   var = f->acc & FAST_HUFFMAN_TABLE_MASK;                    \
+   var = c->fast_huffman[var];                                \
+   if (var >= 0) {                                            \
+      int n = c->codeword_lengths[var];                       \
+      f->acc >>= n;                                           \
+      f->valid_bits -= n;                                     \
+      if (f->valid_bits < 0) { f->valid_bits = 0; var = -1; } \
+   } else {                                                   \
+      var = codebook_decode_scalar_raw(f,c);                  \
+   }
+
+#else
+
+static int codebook_decode_scalar(vorb *f, Codebook *c)
+{
+    int i;
+    if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH)
+        prep_huffman(f);
+    // fast huffman table lookup
+    i = f->acc & FAST_HUFFMAN_TABLE_MASK;
+    i = c->fast_huffman[i];
+    if (i >= 0) {
+        f->acc >>= c->codeword_lengths[i];
+        f->valid_bits -= c->codeword_lengths[i];
+        if (f->valid_bits < 0) { f->valid_bits = 0; return -1; }
+        return i;
+    }
+    return codebook_decode_scalar_raw(f, c);
+}
+
+#define DECODE_RAW(var,f,c)    var = codebook_decode_scalar(f,c);
+
+#endif
+
+#define DECODE(var,f,c)                                       \
+   DECODE_RAW(var,f,c)                                        \
+   if (c->sparse) var = c->sorted_values[var];
+
+#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
+#define DECODE_VQ(var,f,c)   DECODE_RAW(var,f,c)
+#else
+#define DECODE_VQ(var,f,c)   DECODE(var,f,c)
+#endif
+
+
+
+
+
+
+// CODEBOOK_ELEMENT_FAST is an optimization for the CODEBOOK_FLOATS case
+// where we avoid one addition
+#define CODEBOOK_ELEMENT(c,off)          (c->multiplicands[off])
+#define CODEBOOK_ELEMENT_FAST(c,off)     (c->multiplicands[off])
+#define CODEBOOK_ELEMENT_BASE(c)         (0)
+
+static int codebook_decode_start(vorb *f, Codebook *c)
+{
+    int z = -1;
+
+    // type 0 is only legal in a scalar context
+    if (c->lookup_type == 0)
+        error(f, VORBIS_invalid_stream);
+    else {
+        DECODE_VQ(z, f, c);
+        if (c->sparse) assert(z < c->sorted_entries);
+        if (z < 0) {  // check for EOP
+            if (!f->bytes_in_seg)
+                if (f->last_seg)
+                    return z;
+            error(f, VORBIS_invalid_stream);
+        }
+    }
+    return z;
+}
+
+static int codebook_decode(vorb *f, Codebook *c, float *output, int len)
+{
+    int i, z = codebook_decode_start(f, c);
+    if (z < 0) return FALSE;
+    if (len > c->dimensions) len = c->dimensions;
+
+#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
+    if (c->lookup_type == 1) {
+        float last = CODEBOOK_ELEMENT_BASE(c);
+        int div = 1;
+        for (i = 0; i < len; ++i) {
+            int off = (z / div) % c->lookup_values;
+            float val = CODEBOOK_ELEMENT_FAST(c, off) + last;
+            output[i] += val;
+            if (c->sequence_p) last = val + c->minimum_value;
+            div *= c->lookup_values;
+        }
+        return TRUE;
+    }
+#endif
+
+    z *= c->dimensions;
+    if (c->sequence_p) {
+        float last = CODEBOOK_ELEMENT_BASE(c);
+        for (i = 0; i < len; ++i) {
+            float val = CODEBOOK_ELEMENT_FAST(c, z + i) + last;
+            output[i] += val;
+            last = val + c->minimum_value;
+        }
+    }
+    else {
+        float last = CODEBOOK_ELEMENT_BASE(c);
+        for (i = 0; i < len; ++i) {
+            output[i] += CODEBOOK_ELEMENT_FAST(c, z + i) + last;
+        }
+    }
+
+    return TRUE;
+}
+
+static int codebook_decode_step(vorb *f, Codebook *c, float *output, int len, int step)
+{
+    int i, z = codebook_decode_start(f, c);
+    float last = CODEBOOK_ELEMENT_BASE(c);
+    if (z < 0) return FALSE;
+    if (len > c->dimensions) len = c->dimensions;
+
+#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
+    if (c->lookup_type == 1) {
+        int div = 1;
+        for (i = 0; i < len; ++i) {
+            int off = (z / div) % c->lookup_values;
+            float val = CODEBOOK_ELEMENT_FAST(c, off) + last;
+            output[i*step] += val;
+            if (c->sequence_p) last = val;
+            div *= c->lookup_values;
+        }
+        return TRUE;
+    }
+#endif
+
+    z *= c->dimensions;
+    for (i = 0; i < len; ++i) {
+        float val = CODEBOOK_ELEMENT_FAST(c, z + i) + last;
+        output[i*step] += val;
+        if (c->sequence_p) last = val;
+    }
+
+    return TRUE;
+}
+
+static int codebook_decode_deinterleave_repeat(vorb *f, Codebook *c, float **outputs, int ch, int *c_inter_p, int *p_inter_p, int len, int total_decode)
+{
+    int c_inter = *c_inter_p;
+    int p_inter = *p_inter_p;
+    int i, z, effective = c->dimensions;
+
+    // type 0 is only legal in a scalar context
+    if (c->lookup_type == 0)   return error(f, VORBIS_invalid_stream);
+
+    while (total_decode > 0) {
+        float last = CODEBOOK_ELEMENT_BASE(c);
+        DECODE_VQ(z, f, c);
+#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
+        assert(!c->sparse || z < c->sorted_entries);
+#endif
+        if (z < 0) {
+            if (!f->bytes_in_seg)
+                if (f->last_seg) return FALSE;
+            return error(f, VORBIS_invalid_stream);
+        }
+
+        // if this will take us off the end of the buffers, stop short!
+        // we check by computing the length of the virtual interleaved
+        // buffer (len*ch), our current offset within it (p_inter*ch)+(c_inter),
+        // and the length we'll be using (effective)
+        if (c_inter + p_inter*ch + effective > len * ch) {
+            effective = len*ch - (p_inter*ch - c_inter);
+        }
+
+#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
+        if (c->lookup_type == 1) {
+            int div = 1;
+            for (i = 0; i < effective; ++i) {
+                int off = (z / div) % c->lookup_values;
+                float val = CODEBOOK_ELEMENT_FAST(c, off) + last;
+                if (outputs[c_inter])
+                    outputs[c_inter][p_inter] += val;
+                if (++c_inter == ch) { c_inter = 0; ++p_inter; }
+                if (c->sequence_p) last = val;
+                div *= c->lookup_values;
+            }
+        }
+        else
+#endif
+        {
+            z *= c->dimensions;
+            if (c->sequence_p) {
+                for (i = 0; i < effective; ++i) {
+                    float val = CODEBOOK_ELEMENT_FAST(c, z + i) + last;
+                    if (outputs[c_inter])
+                        outputs[c_inter][p_inter] += val;
+                    if (++c_inter == ch) { c_inter = 0; ++p_inter; }
+                    last = val;
+                }
+            }
+            else {
+                for (i = 0; i < effective; ++i) {
+                    float val = CODEBOOK_ELEMENT_FAST(c, z + i) + last;
+                    if (outputs[c_inter])
+                        outputs[c_inter][p_inter] += val;
+                    if (++c_inter == ch) { c_inter = 0; ++p_inter; }
+                }
+            }
+        }
+
+        total_decode -= effective;
+    }
+    *c_inter_p = c_inter;
+    *p_inter_p = p_inter;
+    return TRUE;
+}
+
+static int predict_point(int x, int x0, int x1, int y0, int y1)
+{
+    int dy = y1 - y0;
+    int adx = x1 - x0;
+    // @OPTIMIZE: force int division to round in the right direction... is this necessary on x86?
+    int err = abs(dy) * (x - x0);
+    int off = err / adx;
+    return dy < 0 ? y0 - off : y0 + off;
+}
+
+// the following table is block-copied from the specification
+static float inverse_db_table[256] =
+{
+    1.0649863e-07f, 1.1341951e-07f, 1.2079015e-07f, 1.2863978e-07f,
+    1.3699951e-07f, 1.4590251e-07f, 1.5538408e-07f, 1.6548181e-07f,
+    1.7623575e-07f, 1.8768855e-07f, 1.9988561e-07f, 2.1287530e-07f,
+    2.2670913e-07f, 2.4144197e-07f, 2.5713223e-07f, 2.7384213e-07f,
+    2.9163793e-07f, 3.1059021e-07f, 3.3077411e-07f, 3.5226968e-07f,
+    3.7516214e-07f, 3.9954229e-07f, 4.2550680e-07f, 4.5315863e-07f,
+    4.8260743e-07f, 5.1396998e-07f, 5.4737065e-07f, 5.8294187e-07f,
+    6.2082472e-07f, 6.6116941e-07f, 7.0413592e-07f, 7.4989464e-07f,
+    7.9862701e-07f, 8.5052630e-07f, 9.0579828e-07f, 9.6466216e-07f,
+    1.0273513e-06f, 1.0941144e-06f, 1.1652161e-06f, 1.2409384e-06f,
+    1.3215816e-06f, 1.4074654e-06f, 1.4989305e-06f, 1.5963394e-06f,
+    1.7000785e-06f, 1.8105592e-06f, 1.9282195e-06f, 2.0535261e-06f,
+    2.1869758e-06f, 2.3290978e-06f, 2.4804557e-06f, 2.6416497e-06f,
+    2.8133190e-06f, 2.9961443e-06f, 3.1908506e-06f, 3.3982101e-06f,
+    3.6190449e-06f, 3.8542308e-06f, 4.1047004e-06f, 4.3714470e-06f,
+    4.6555282e-06f, 4.9580707e-06f, 5.2802740e-06f, 5.6234160e-06f,
+    5.9888572e-06f, 6.3780469e-06f, 6.7925283e-06f, 7.2339451e-06f,
+    7.7040476e-06f, 8.2047000e-06f, 8.7378876e-06f, 9.3057248e-06f,
+    9.9104632e-06f, 1.0554501e-05f, 1.1240392e-05f, 1.1970856e-05f,
+    1.2748789e-05f, 1.3577278e-05f, 1.4459606e-05f, 1.5399272e-05f,
+    1.6400004e-05f, 1.7465768e-05f, 1.8600792e-05f, 1.9809576e-05f,
+    2.1096914e-05f, 2.2467911e-05f, 2.3928002e-05f, 2.5482978e-05f,
+    2.7139006e-05f, 2.8902651e-05f, 3.0780908e-05f, 3.2781225e-05f,
+    3.4911534e-05f, 3.7180282e-05f, 3.9596466e-05f, 4.2169667e-05f,
+    4.4910090e-05f, 4.7828601e-05f, 5.0936773e-05f, 5.4246931e-05f,
+    5.7772202e-05f, 6.1526565e-05f, 6.5524908e-05f, 6.9783085e-05f,
+    7.4317983e-05f, 7.9147585e-05f, 8.4291040e-05f, 8.9768747e-05f,
+    9.5602426e-05f, 0.00010181521f, 0.00010843174f, 0.00011547824f,
+    0.00012298267f, 0.00013097477f, 0.00013948625f, 0.00014855085f,
+    0.00015820453f, 0.00016848555f, 0.00017943469f, 0.00019109536f,
+    0.00020351382f, 0.00021673929f, 0.00023082423f, 0.00024582449f,
+    0.00026179955f, 0.00027881276f, 0.00029693158f, 0.00031622787f,
+    0.00033677814f, 0.00035866388f, 0.00038197188f, 0.00040679456f,
+    0.00043323036f, 0.00046138411f, 0.00049136745f, 0.00052329927f,
+    0.00055730621f, 0.00059352311f, 0.00063209358f, 0.00067317058f,
+    0.00071691700f, 0.00076350630f, 0.00081312324f, 0.00086596457f,
+    0.00092223983f, 0.00098217216f, 0.0010459992f,  0.0011139742f,
+    0.0011863665f,  0.0012634633f,  0.0013455702f,  0.0014330129f,
+    0.0015261382f,  0.0016253153f,  0.0017309374f,  0.0018434235f,
+    0.0019632195f,  0.0020908006f,  0.0022266726f,  0.0023713743f,
+    0.0025254795f,  0.0026895994f,  0.0028643847f,  0.0030505286f,
+    0.0032487691f,  0.0034598925f,  0.0036847358f,  0.0039241906f,
+    0.0041792066f,  0.0044507950f,  0.0047400328f,  0.0050480668f,
+    0.0053761186f,  0.0057254891f,  0.0060975636f,  0.0064938176f,
+    0.0069158225f,  0.0073652516f,  0.0078438871f,  0.0083536271f,
+    0.0088964928f,  0.009474637f,   0.010090352f,   0.010746080f,
+    0.011444421f,   0.012188144f,   0.012980198f,   0.013823725f,
+    0.014722068f,   0.015678791f,   0.016697687f,   0.017782797f,
+    0.018938423f,   0.020169149f,   0.021479854f,   0.022875735f,
+    0.024362330f,   0.025945531f,   0.027631618f,   0.029427276f,
+    0.031339626f,   0.033376252f,   0.035545228f,   0.037855157f,
+    0.040315199f,   0.042935108f,   0.045725273f,   0.048696758f,
+    0.051861348f,   0.055231591f,   0.058820850f,   0.062643361f,
+    0.066714279f,   0.071049749f,   0.075666962f,   0.080584227f,
+    0.085821044f,   0.091398179f,   0.097337747f,   0.10366330f,
+    0.11039993f,    0.11757434f,    0.12521498f,    0.13335215f,
+    0.14201813f,    0.15124727f,    0.16107617f,    0.17154380f,
+    0.18269168f,    0.19456402f,    0.20720788f,    0.22067342f,
+    0.23501402f,    0.25028656f,    0.26655159f,    0.28387361f,
+    0.30232132f,    0.32196786f,    0.34289114f,    0.36517414f,
+    0.38890521f,    0.41417847f,    0.44109412f,    0.46975890f,
+    0.50028648f,    0.53279791f,    0.56742212f,    0.60429640f,
+    0.64356699f,    0.68538959f,    0.72993007f,    0.77736504f,
+    0.82788260f,    0.88168307f,    0.9389798f,     1.0f
+};
+
+
+// @OPTIMIZE: if you want to replace this bresenham line-drawing routine,
+// note that you must produce bit-identical output to decode correctly;
+// this specific sequence of operations is specified in the spec (it's
+// drawing integer-quantized frequency-space lines that the encoder
+// expects to be exactly the same)
+//     ... also, isn't the whole point of Bresenham's algorithm to NOT
+// have to divide in the setup? sigh.
+#ifndef STB_VORBIS_NO_DEFER_FLOOR
+#define LINE_OP(a,b)   a *= b
+#else
+#define LINE_OP(a,b)   a = b
+#endif
+
+#ifdef STB_VORBIS_DIVIDE_TABLE
+#define DIVTAB_NUMER   32
+#define DIVTAB_DENOM   64
+int8 integer_divide_table[DIVTAB_NUMER][DIVTAB_DENOM]; // 2KB
+#endif
+
+static __forceinline void draw_line(float *output, int x0, int y0, int x1, int y1, int n)
+{
+    int dy = y1 - y0;
+    int adx = x1 - x0;
+    int ady = abs(dy);
+    int base;
+    int x = x0, y = y0;
+    int err = 0;
+    int sy;
+
+#ifdef STB_VORBIS_DIVIDE_TABLE
+    if (adx < DIVTAB_DENOM && ady < DIVTAB_NUMER) {
+        if (dy < 0) {
+            base = -integer_divide_table[ady][adx];
+            sy = base - 1;
+        }
+        else {
+            base = integer_divide_table[ady][adx];
+            sy = base + 1;
+        }
+    }
+    else {
+        base = dy / adx;
+        if (dy < 0)
+            sy = base - 1;
+        else
+            sy = base + 1;
+    }
+#else
+    base = dy / adx;
+    if (dy < 0)
+        sy = base - 1;
+    else
+        sy = base + 1;
+#endif
+    ady -= abs(base) * adx;
+    if (x1 > n) x1 = n;
+    if (x < x1) {
+        LINE_OP(output[x], inverse_db_table[y]);
+        for (++x; x < x1; ++x) {
+            err += ady;
+            if (err >= adx) {
+                err -= adx;
+                y += sy;
+            }
+            else
+                y += base;
+            LINE_OP(output[x], inverse_db_table[y]);
+        }
+    }
+}
+
+static int residue_decode(vorb *f, Codebook *book, float *target, int offset, int n, int rtype)
+{
+    int k;
+    if (rtype == 0) {
+        int step = n / book->dimensions;
+        for (k = 0; k < step; ++k)
+            if (!codebook_decode_step(f, book, target + offset + k, n - offset - k, step))
+                return FALSE;
+    }
+    else {
+        for (k = 0; k < n; ) {
+            if (!codebook_decode(f, book, target + offset, n - k))
+                return FALSE;
+            k += book->dimensions;
+            offset += book->dimensions;
+        }
+    }
+    return TRUE;
+}
+
+// n is 1/2 of the blocksize --
+// specification: "Correct per-vector decode length is [n]/2"
+static void decode_residue(vorb *f, float *residue_buffers[], int ch, int n, int rn, uint8 *do_not_decode)
+{
+    int i, j, pass;
+    Residue *r = f->residue_config + rn;
+    int rtype = f->residue_types[rn];
+    int c = r->classbook;
+    int classwords = f->codebooks[c].dimensions;
+    unsigned int actual_size = rtype == 2 ? n * 2 : n;
+    unsigned int limit_r_begin = (r->begin < actual_size ? r->begin : actual_size);
+    unsigned int limit_r_end = (r->end < actual_size ? r->end : actual_size);
+    int n_read = limit_r_end - limit_r_begin;
+    int part_read = n_read / r->part_size;
+    int temp_alloc_point = temp_alloc_save(f);
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+    uint8 ***part_classdata = (uint8 ***)temp_block_array(f, f->channels, part_read * sizeof(**part_classdata));
+#else
+    int **classifications = (int **)temp_block_array(f, f->channels, part_read * sizeof(**classifications));
+#endif
+
+    CHECK(f);
+
+    for (i = 0; i < ch; ++i)
+        if (!do_not_decode[i])
+            memset(residue_buffers[i], 0, sizeof(float) * n);
+
+    if (rtype == 2 && ch != 1) {
+        for (j = 0; j < ch; ++j)
+            if (!do_not_decode[j])
+                break;
+        if (j == ch)
+            goto done;
+
+        for (pass = 0; pass < 8; ++pass) {
+            int pcount = 0, class_set = 0;
+            if (ch == 2) {
+                while (pcount < part_read) {
+                    int z = r->begin + pcount*r->part_size;
+                    int c_inter = (z & 1), p_inter = z >> 1;
+                    if (pass == 0) {
+                        Codebook *c = f->codebooks + r->classbook;
+                        int q;
+                        DECODE(q, f, c);
+                        if (q == EOP) goto done;
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                        part_classdata[0][class_set] = r->classdata[q];
+#else
+                        for (i = classwords - 1; i >= 0; --i) {
+                            classifications[0][i + pcount] = q % r->classifications;
+                            q /= r->classifications;
+                        }
+#endif
+                    }
+                    for (i = 0; i < classwords && pcount < part_read; ++i, ++pcount) {
+                        int z = r->begin + pcount*r->part_size;
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                        int c = part_classdata[0][class_set][i];
+#else
+                        int c = classifications[0][pcount];
+#endif
+                        int b = r->residue_books[c][pass];
+                        if (b >= 0) {
+                            Codebook *book = f->codebooks + b;
+#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
+                            if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
+                                goto done;
+#else
+                            // saves 1%
+                            if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
+                                goto done;
+#endif
+                        }
+                        else {
+                            z += r->part_size;
+                            c_inter = z & 1;
+                            p_inter = z >> 1;
+                        }
+                    }
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                    ++class_set;
+#endif
+                }
+            }
+            else if (ch == 1) {
+                while (pcount < part_read) {
+                    int z = r->begin + pcount*r->part_size;
+                    int c_inter = 0, p_inter = z;
+                    if (pass == 0) {
+                        Codebook *c = f->codebooks + r->classbook;
+                        int q;
+                        DECODE(q, f, c);
+                        if (q == EOP) goto done;
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                        part_classdata[0][class_set] = r->classdata[q];
+#else
+                        for (i = classwords - 1; i >= 0; --i) {
+                            classifications[0][i + pcount] = q % r->classifications;
+                            q /= r->classifications;
+                        }
+#endif
+                    }
+                    for (i = 0; i < classwords && pcount < part_read; ++i, ++pcount) {
+                        int z = r->begin + pcount*r->part_size;
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                        int c = part_classdata[0][class_set][i];
+#else
+                        int c = classifications[0][pcount];
+#endif
+                        int b = r->residue_books[c][pass];
+                        if (b >= 0) {
+                            Codebook *book = f->codebooks + b;
+                            if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
+                                goto done;
+                        }
+                        else {
+                            z += r->part_size;
+                            c_inter = 0;
+                            p_inter = z;
+                        }
+                    }
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                    ++class_set;
+#endif
+                }
+            }
+            else {
+                while (pcount < part_read) {
+                    int z = r->begin + pcount*r->part_size;
+                    int c_inter = z % ch, p_inter = z / ch;
+                    if (pass == 0) {
+                        Codebook *c = f->codebooks + r->classbook;
+                        int q;
+                        DECODE(q, f, c);
+                        if (q == EOP) goto done;
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                        part_classdata[0][class_set] = r->classdata[q];
+#else
+                        for (i = classwords - 1; i >= 0; --i) {
+                            classifications[0][i + pcount] = q % r->classifications;
+                            q /= r->classifications;
+                        }
+#endif
+                    }
+                    for (i = 0; i < classwords && pcount < part_read; ++i, ++pcount) {
+                        int z = r->begin + pcount*r->part_size;
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                        int c = part_classdata[0][class_set][i];
+#else
+                        int c = classifications[0][pcount];
+#endif
+                        int b = r->residue_books[c][pass];
+                        if (b >= 0) {
+                            Codebook *book = f->codebooks + b;
+                            if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
+                                goto done;
+                        }
+                        else {
+                            z += r->part_size;
+                            c_inter = z % ch;
+                            p_inter = z / ch;
+                        }
+                    }
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                    ++class_set;
+#endif
+                }
+            }
+        }
+        goto done;
+    }
+    CHECK(f);
+
+    for (pass = 0; pass < 8; ++pass) {
+        int pcount = 0, class_set = 0;
+        while (pcount < part_read) {
+            if (pass == 0) {
+                for (j = 0; j < ch; ++j) {
+                    if (!do_not_decode[j]) {
+                        Codebook *c = f->codebooks + r->classbook;
+                        int temp;
+                        DECODE(temp, f, c);
+                        if (temp == EOP) goto done;
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                        part_classdata[j][class_set] = r->classdata[temp];
+#else
+                        for (i = classwords - 1; i >= 0; --i) {
+                            classifications[j][i + pcount] = temp % r->classifications;
+                            temp /= r->classifications;
+                        }
+#endif
+                    }
+                }
+            }
+            for (i = 0; i < classwords && pcount < part_read; ++i, ++pcount) {
+                for (j = 0; j < ch; ++j) {
+                    if (!do_not_decode[j]) {
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                        int c = part_classdata[j][class_set][i];
+#else
+                        int c = classifications[j][pcount];
+#endif
+                        int b = r->residue_books[c][pass];
+                        if (b >= 0) {
+                            float *target = residue_buffers[j];
+                            int offset = r->begin + pcount * r->part_size;
+                            int n = r->part_size;
+                            Codebook *book = f->codebooks + b;
+                            if (!residue_decode(f, book, target, offset, n, rtype))
+                                goto done;
+                        }
+                    }
+                }
+            }
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+            ++class_set;
+#endif
+        }
+    }
+done:
+    CHECK(f);
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+    temp_free(f, part_classdata);
+#else
+    temp_free(f, classifications);
+#endif
+    temp_alloc_restore(f, temp_alloc_point);
+}
+
+
+#if 0
+// slow way for debugging
+void inverse_mdct_slow(float *buffer, int n)
+{
+    int i, j;
+    int n2 = n >> 1;
+    float *x = (float *)malloc(sizeof(*x) * n2);
+    memcpy(x, buffer, sizeof(*x) * n2);
+    for (i = 0; i < n; ++i) {
+        float acc = 0;
+        for (j = 0; j < n2; ++j)
+            // formula from paper:
+            //acc += n/4.0f * x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1));
+            // formula from wikipedia
+            //acc += 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5));
+            // these are equivalent, except the formula from the paper inverts the multiplier!
+            // however, what actually works is NO MULTIPLIER!?!
+            //acc += 64 * 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5));
+            acc += x[j] * (float)cos(M_PI / 2 / n * (2 * i + 1 + n / 2.0)*(2 * j + 1));
+        buffer[i] = acc;
+    }
+    free(x);
+}
+#elif 0
+// same as above, but just barely able to run in real time on modern machines
+void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype)
+{
+    float mcos[16384];
+    int i, j;
+    int n2 = n >> 1, nmask = (n << 2) - 1;
+    float *x = (float *)malloc(sizeof(*x) * n2);
+    memcpy(x, buffer, sizeof(*x) * n2);
+    for (i = 0; i < 4 * n; ++i)
+        mcos[i] = (float)cos(M_PI / 2 * i / n);
+
+    for (i = 0; i < n; ++i) {
+        float acc = 0;
+        for (j = 0; j < n2; ++j)
+            acc += x[j] * mcos[(2 * i + 1 + n2)*(2 * j + 1) & nmask];
+        buffer[i] = acc;
+    }
+    free(x);
+}
+#elif 0
+// transform to use a slow dct-iv; this is STILL basically trivial,
+// but only requires half as many ops
+void dct_iv_slow(float *buffer, int n)
+{
+    float mcos[16384];
+    float x[2048];
+    int i, j;
+    int n2 = n >> 1, nmask = (n << 3) - 1;
+    memcpy(x, buffer, sizeof(*x) * n);
+    for (i = 0; i < 8 * n; ++i)
+        mcos[i] = (float)cos(M_PI / 4 * i / n);
+    for (i = 0; i < n; ++i) {
+        float acc = 0;
+        for (j = 0; j < n; ++j)
+            acc += x[j] * mcos[((2 * i + 1)*(2 * j + 1)) & nmask];
+        buffer[i] = acc;
+    }
+}
+
+void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype)
+{
+    int i, n4 = n >> 2, n2 = n >> 1, n3_4 = n - n4;
+    float temp[4096];
+
+    memcpy(temp, buffer, n2 * sizeof(float));
+    dct_iv_slow(temp, n2);  // returns -c'-d, a-b'
+
+    for (i = 0; i < n4; ++i) buffer[i] = temp[i + n4];            // a-b'
+    for (; i < n3_4; ++i) buffer[i] = -temp[n3_4 - i - 1];   // b-a', c+d'
+    for (; i < n; ++i) buffer[i] = -temp[i - n3_4];       // c'+d
+}
+#endif
+
+#ifndef LIBVORBIS_MDCT
+#define LIBVORBIS_MDCT 0
+#endif
+
+#if LIBVORBIS_MDCT
+// directly call the vorbis MDCT using an interface documented
+// by Jeff Roberts... useful for performance comparison
+typedef struct
+{
+    int n;
+    int log2n;
+
+    float *trig;
+    int   *bitrev;
+
+    float scale;
+} mdct_lookup;
+
+extern void mdct_init(mdct_lookup *lookup, int n);
+extern void mdct_clear(mdct_lookup *l);
+extern void mdct_backward(mdct_lookup *init, float *in, float *out);
+
+mdct_lookup M1, M2;
+
+void inverse_mdct(float *buffer, int n, vorb *f, int blocktype)
+{
+    mdct_lookup *M;
+    if (M1.n == n) M = &M1;
+    else if (M2.n == n) M = &M2;
+    else if (M1.n == 0) { mdct_init(&M1, n); M = &M1; }
+    else {
+        if (M2.n) __asm int 3;
+        mdct_init(&M2, n);
+        M = &M2;
+    }
+
+    mdct_backward(M, buffer, buffer);
+}
+#endif
+
+
+// the following were split out into separate functions while optimizing;
+// they could be pushed back up but eh. __forceinline showed no change;
+// they're probably already being inlined.
+static void imdct_step3_iter0_loop(int n, float *e, int i_off, int k_off, float *A)
+{
+    float *ee0 = e + i_off;
+    float *ee2 = ee0 + k_off;
+    int i;
+
+    assert((n & 3) == 0);
+    for (i = (n >> 2); i > 0; --i) {
+        float k00_20, k01_21;
+        k00_20 = ee0[0] - ee2[0];
+        k01_21 = ee0[-1] - ee2[-1];
+        ee0[0] += ee2[0];//ee0[ 0] = ee0[ 0] + ee2[ 0];
+        ee0[-1] += ee2[-1];//ee0[-1] = ee0[-1] + ee2[-1];
+        ee2[0] = k00_20 * A[0] - k01_21 * A[1];
+        ee2[-1] = k01_21 * A[0] + k00_20 * A[1];
+        A += 8;
+
+        k00_20 = ee0[-2] - ee2[-2];
+        k01_21 = ee0[-3] - ee2[-3];
+        ee0[-2] += ee2[-2];//ee0[-2] = ee0[-2] + ee2[-2];
+        ee0[-3] += ee2[-3];//ee0[-3] = ee0[-3] + ee2[-3];
+        ee2[-2] = k00_20 * A[0] - k01_21 * A[1];
+        ee2[-3] = k01_21 * A[0] + k00_20 * A[1];
+        A += 8;
+
+        k00_20 = ee0[-4] - ee2[-4];
+        k01_21 = ee0[-5] - ee2[-5];
+        ee0[-4] += ee2[-4];//ee0[-4] = ee0[-4] + ee2[-4];
+        ee0[-5] += ee2[-5];//ee0[-5] = ee0[-5] + ee2[-5];
+        ee2[-4] = k00_20 * A[0] - k01_21 * A[1];
+        ee2[-5] = k01_21 * A[0] + k00_20 * A[1];
+        A += 8;
+
+        k00_20 = ee0[-6] - ee2[-6];
+        k01_21 = ee0[-7] - ee2[-7];
+        ee0[-6] += ee2[-6];//ee0[-6] = ee0[-6] + ee2[-6];
+        ee0[-7] += ee2[-7];//ee0[-7] = ee0[-7] + ee2[-7];
+        ee2[-6] = k00_20 * A[0] - k01_21 * A[1];
+        ee2[-7] = k01_21 * A[0] + k00_20 * A[1];
+        A += 8;
+        ee0 -= 8;
+        ee2 -= 8;
+    }
+}
+
+static void imdct_step3_inner_r_loop(int lim, float *e, int d0, int k_off, float *A, int k1)
+{
+    int i;
+    float k00_20, k01_21;
+
+    float *e0 = e + d0;
+    float *e2 = e0 + k_off;
+
+    for (i = lim >> 2; i > 0; --i) {
+        k00_20 = e0[-0] - e2[-0];
+        k01_21 = e0[-1] - e2[-1];
+        e0[-0] += e2[-0];//e0[-0] = e0[-0] + e2[-0];
+        e0[-1] += e2[-1];//e0[-1] = e0[-1] + e2[-1];
+        e2[-0] = (k00_20)*A[0] - (k01_21)* A[1];
+        e2[-1] = (k01_21)*A[0] + (k00_20)* A[1];
+
+        A += k1;
+
+        k00_20 = e0[-2] - e2[-2];
+        k01_21 = e0[-3] - e2[-3];
+        e0[-2] += e2[-2];//e0[-2] = e0[-2] + e2[-2];
+        e0[-3] += e2[-3];//e0[-3] = e0[-3] + e2[-3];
+        e2[-2] = (k00_20)*A[0] - (k01_21)* A[1];
+        e2[-3] = (k01_21)*A[0] + (k00_20)* A[1];
+
+        A += k1;
+
+        k00_20 = e0[-4] - e2[-4];
+        k01_21 = e0[-5] - e2[-5];
+        e0[-4] += e2[-4];//e0[-4] = e0[-4] + e2[-4];
+        e0[-5] += e2[-5];//e0[-5] = e0[-5] + e2[-5];
+        e2[-4] = (k00_20)*A[0] - (k01_21)* A[1];
+        e2[-5] = (k01_21)*A[0] + (k00_20)* A[1];
+
+        A += k1;
+
+        k00_20 = e0[-6] - e2[-6];
+        k01_21 = e0[-7] - e2[-7];
+        e0[-6] += e2[-6];//e0[-6] = e0[-6] + e2[-6];
+        e0[-7] += e2[-7];//e0[-7] = e0[-7] + e2[-7];
+        e2[-6] = (k00_20)*A[0] - (k01_21)* A[1];
+        e2[-7] = (k01_21)*A[0] + (k00_20)* A[1];
+
+        e0 -= 8;
+        e2 -= 8;
+
+        A += k1;
+    }
+}
+
+static void imdct_step3_inner_s_loop(int n, float *e, int i_off, int k_off, float *A, int a_off, int k0)
+{
+    int i;
+    float A0 = A[0];
+    float A1 = A[0 + 1];
+    float A2 = A[0 + a_off];
+    float A3 = A[0 + a_off + 1];
+    float A4 = A[0 + a_off * 2 + 0];
+    float A5 = A[0 + a_off * 2 + 1];
+    float A6 = A[0 + a_off * 3 + 0];
+    float A7 = A[0 + a_off * 3 + 1];
+
+    float k00, k11;
+
+    float *ee0 = e + i_off;
+    float *ee2 = ee0 + k_off;
+
+    for (i = n; i > 0; --i) {
+        k00 = ee0[0] - ee2[0];
+        k11 = ee0[-1] - ee2[-1];
+        ee0[0] = ee0[0] + ee2[0];
+        ee0[-1] = ee0[-1] + ee2[-1];
+        ee2[0] = (k00)* A0 - (k11)* A1;
+        ee2[-1] = (k11)* A0 + (k00)* A1;
+
+        k00 = ee0[-2] - ee2[-2];
+        k11 = ee0[-3] - ee2[-3];
+        ee0[-2] = ee0[-2] + ee2[-2];
+        ee0[-3] = ee0[-3] + ee2[-3];
+        ee2[-2] = (k00)* A2 - (k11)* A3;
+        ee2[-3] = (k11)* A2 + (k00)* A3;
+
+        k00 = ee0[-4] - ee2[-4];
+        k11 = ee0[-5] - ee2[-5];
+        ee0[-4] = ee0[-4] + ee2[-4];
+        ee0[-5] = ee0[-5] + ee2[-5];
+        ee2[-4] = (k00)* A4 - (k11)* A5;
+        ee2[-5] = (k11)* A4 + (k00)* A5;
+
+        k00 = ee0[-6] - ee2[-6];
+        k11 = ee0[-7] - ee2[-7];
+        ee0[-6] = ee0[-6] + ee2[-6];
+        ee0[-7] = ee0[-7] + ee2[-7];
+        ee2[-6] = (k00)* A6 - (k11)* A7;
+        ee2[-7] = (k11)* A6 + (k00)* A7;
+
+        ee0 -= k0;
+        ee2 -= k0;
+    }
+}
+
+static __forceinline void iter_54(float *z)
+{
+    float k00, k11, k22, k33;
+    float y0, y1, y2, y3;
+
+    k00 = z[0] - z[-4];
+    y0 = z[0] + z[-4];
+    y2 = z[-2] + z[-6];
+    k22 = z[-2] - z[-6];
+
+    z[-0] = y0 + y2;      // z0 + z4 + z2 + z6
+    z[-2] = y0 - y2;      // z0 + z4 - z2 - z6
+
+                          // done with y0,y2
+
+    k33 = z[-3] - z[-7];
+
+    z[-4] = k00 + k33;    // z0 - z4 + z3 - z7
+    z[-6] = k00 - k33;    // z0 - z4 - z3 + z7
+
+                          // done with k33
+
+    k11 = z[-1] - z[-5];
+    y1 = z[-1] + z[-5];
+    y3 = z[-3] + z[-7];
+
+    z[-1] = y1 + y3;      // z1 + z5 + z3 + z7
+    z[-3] = y1 - y3;      // z1 + z5 - z3 - z7
+    z[-5] = k11 - k22;    // z1 - z5 + z2 - z6
+    z[-7] = k11 + k22;    // z1 - z5 - z2 + z6
+}
+
+static void imdct_step3_inner_s_loop_ld654(int n, float *e, int i_off, float *A, int base_n)
+{
+    int a_off = base_n >> 3;
+    float A2 = A[0 + a_off];
+    float *z = e + i_off;
+    float *base = z - 16 * n;
+
+    while (z > base) {
+        float k00, k11;
+
+        k00 = z[-0] - z[-8];
+        k11 = z[-1] - z[-9];
+        z[-0] = z[-0] + z[-8];
+        z[-1] = z[-1] + z[-9];
+        z[-8] = k00;
+        z[-9] = k11;
+
+        k00 = z[-2] - z[-10];
+        k11 = z[-3] - z[-11];
+        z[-2] = z[-2] + z[-10];
+        z[-3] = z[-3] + z[-11];
+        z[-10] = (k00 + k11) * A2;
+        z[-11] = (k11 - k00) * A2;
+
+        k00 = z[-12] - z[-4];  // reverse to avoid a unary negation
+        k11 = z[-5] - z[-13];
+        z[-4] = z[-4] + z[-12];
+        z[-5] = z[-5] + z[-13];
+        z[-12] = k11;
+        z[-13] = k00;
+
+        k00 = z[-14] - z[-6];  // reverse to avoid a unary negation
+        k11 = z[-7] - z[-15];
+        z[-6] = z[-6] + z[-14];
+        z[-7] = z[-7] + z[-15];
+        z[-14] = (k00 + k11) * A2;
+        z[-15] = (k00 - k11) * A2;
+
+        iter_54(z);
+        iter_54(z - 8);
+        z -= 16;
+    }
+}
+
+static void inverse_mdct(float *buffer, int n, vorb *f, int blocktype)
+{
+    int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l;
+    int ld;
+    // @OPTIMIZE: reduce register pressure by using fewer variables?
+    int save_point = temp_alloc_save(f);
+    float *buf2 = (float *)temp_alloc(f, n2 * sizeof(*buf2));
+    float *u = NULL, *v = NULL;
+    // twiddle factors
+    float *A = f->A[blocktype];
+
+    // IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio"
+    // See notes about bugs in that paper in less-optimal implementation 'inverse_mdct_old' after this function.
+
+    // kernel from paper
+
+
+    // merged:
+    //   copy and reflect spectral data
+    //   step 0
+
+    // note that it turns out that the items added together during
+    // this step are, in fact, being added to themselves (as reflected
+    // by step 0). inexplicable inefficiency! this became obvious
+    // once I combined the passes.
+
+    // so there's a missing 'times 2' here (for adding X to itself).
+    // this propogates through linearly to the end, where the numbers
+    // are 1/2 too small, and need to be compensated for.
+
+    {
+        float *d, *e, *AA, *e_stop;
+        d = &buf2[n2 - 2];
+        AA = A;
+        e = &buffer[0];
+        e_stop = &buffer[n2];
+        while (e != e_stop) {
+            d[1] = (e[0] * AA[0] - e[2] * AA[1]);
+            d[0] = (e[0] * AA[1] + e[2] * AA[0]);
+            d -= 2;
+            AA += 2;
+            e += 4;
+        }
+
+        e = &buffer[n2 - 3];
+        while (d >= buf2) {
+            d[1] = (-e[2] * AA[0] - -e[0] * AA[1]);
+            d[0] = (-e[2] * AA[1] + -e[0] * AA[0]);
+            d -= 2;
+            AA += 2;
+            e -= 4;
+        }
+    }
+
+    // now we use symbolic names for these, so that we can
+    // possibly swap their meaning as we change which operations
+    // are in place
+
+    u = buffer;
+    v = buf2;
+
+    // step 2    (paper output is w, now u)
+    // this could be in place, but the data ends up in the wrong
+    // place... _somebody_'s got to swap it, so this is nominated
+    {
+        float *AA = &A[n2 - 8];
+        float *d0, *d1, *e0, *e1;
+
+        e0 = &v[n4];
+        e1 = &v[0];
+
+        d0 = &u[n4];
+        d1 = &u[0];
+
+        while (AA >= A) {
+            float v40_20, v41_21;
+
+            v41_21 = e0[1] - e1[1];
+            v40_20 = e0[0] - e1[0];
+            d0[1] = e0[1] + e1[1];
+            d0[0] = e0[0] + e1[0];
+            d1[1] = v41_21*AA[4] - v40_20*AA[5];
+            d1[0] = v40_20*AA[4] + v41_21*AA[5];
+
+            v41_21 = e0[3] - e1[3];
+            v40_20 = e0[2] - e1[2];
+            d0[3] = e0[3] + e1[3];
+            d0[2] = e0[2] + e1[2];
+            d1[3] = v41_21*AA[0] - v40_20*AA[1];
+            d1[2] = v40_20*AA[0] + v41_21*AA[1];
+
+            AA -= 8;
+
+            d0 += 4;
+            d1 += 4;
+            e0 += 4;
+            e1 += 4;
+        }
+    }
+
+    // step 3
+    ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
+
+                      // optimized step 3:
+
+                      // the original step3 loop can be nested r inside s or s inside r;
+                      // it's written originally as s inside r, but this is dumb when r
+                      // iterates many times, and s few. So I have two copies of it and
+                      // switch between them halfway.
+
+                      // this is iteration 0 of step 3
+    imdct_step3_iter0_loop(n >> 4, u, n2 - 1 - n4 * 0, -(n >> 3), A);
+    imdct_step3_iter0_loop(n >> 4, u, n2 - 1 - n4 * 1, -(n >> 3), A);
+
+    // this is iteration 1 of step 3
+    imdct_step3_inner_r_loop(n >> 5, u, n2 - 1 - n8 * 0, -(n >> 4), A, 16);
+    imdct_step3_inner_r_loop(n >> 5, u, n2 - 1 - n8 * 1, -(n >> 4), A, 16);
+    imdct_step3_inner_r_loop(n >> 5, u, n2 - 1 - n8 * 2, -(n >> 4), A, 16);
+    imdct_step3_inner_r_loop(n >> 5, u, n2 - 1 - n8 * 3, -(n >> 4), A, 16);
+
+    l = 2;
+    for (; l < (ld - 3) >> 1; ++l) {
+        int k0 = n >> (l + 2), k0_2 = k0 >> 1;
+        int lim = 1 << (l + 1);
+        int i;
+        for (i = 0; i < lim; ++i)
+            imdct_step3_inner_r_loop(n >> (l + 4), u, n2 - 1 - k0*i, -k0_2, A, 1 << (l + 3));
+    }
+
+    for (; l < ld - 6; ++l) {
+        int k0 = n >> (l + 2), k1 = 1 << (l + 3), k0_2 = k0 >> 1;
+        int rlim = n >> (l + 6), r;
+        int lim = 1 << (l + 1);
+        int i_off;
+        float *A0 = A;
+        i_off = n2 - 1;
+        for (r = rlim; r > 0; --r) {
+            imdct_step3_inner_s_loop(lim, u, i_off, -k0_2, A0, k1, k0);
+            A0 += k1 * 4;
+            i_off -= 8;
+        }
+    }
+
+    // iterations with count:
+    //   ld-6,-5,-4 all interleaved together
+    //       the big win comes from getting rid of needless flops
+    //         due to the constants on pass 5 & 4 being all 1 and 0;
+    //       combining them to be simultaneous to improve cache made little difference
+    imdct_step3_inner_s_loop_ld654(n >> 5, u, n2 - 1, A, n);
+
+    // output is u
+
+    // step 4, 5, and 6
+    // cannot be in-place because of step 5
+    {
+        uint16 *bitrev = f->bit_reverse[blocktype];
+        // weirdly, I'd have thought reading sequentially and writing
+        // erratically would have been better than vice-versa, but in
+        // fact that's not what my testing showed. (That is, with
+        // j = bitreverse(i), do you read i and write j, or read j and write i.)
+
+        float *d0 = &v[n4 - 4];
+        float *d1 = &v[n2 - 4];
+        while (d0 >= v) {
+            int k4;
+
+            k4 = bitrev[0];
+            d1[3] = u[k4 + 0];
+            d1[2] = u[k4 + 1];
+            d0[3] = u[k4 + 2];
+            d0[2] = u[k4 + 3];
+
+            k4 = bitrev[1];
+            d1[1] = u[k4 + 0];
+            d1[0] = u[k4 + 1];
+            d0[1] = u[k4 + 2];
+            d0[0] = u[k4 + 3];
+
+            d0 -= 4;
+            d1 -= 4;
+            bitrev += 2;
+        }
+    }
+    // (paper output is u, now v)
+
+
+    // data must be in buf2
+    assert(v == buf2);
+
+    // step 7   (paper output is v, now v)
+    // this is now in place
+    {
+        float *C = f->C[blocktype];
+        float *d, *e;
+
+        d = v;
+        e = v + n2 - 4;
+
+        while (d < e) {
+            float a02, a11, b0, b1, b2, b3;
+
+            a02 = d[0] - e[2];
+            a11 = d[1] + e[3];
+
+            b0 = C[1] * a02 + C[0] * a11;
+            b1 = C[1] * a11 - C[0] * a02;
+
+            b2 = d[0] + e[2];
+            b3 = d[1] - e[3];
+
+            d[0] = b2 + b0;
+            d[1] = b3 + b1;
+            e[2] = b2 - b0;
+            e[3] = b1 - b3;
+
+            a02 = d[2] - e[0];
+            a11 = d[3] + e[1];
+
+            b0 = C[3] * a02 + C[2] * a11;
+            b1 = C[3] * a11 - C[2] * a02;
+
+            b2 = d[2] + e[0];
+            b3 = d[3] - e[1];
+
+            d[2] = b2 + b0;
+            d[3] = b3 + b1;
+            e[0] = b2 - b0;
+            e[1] = b1 - b3;
+
+            C += 4;
+            d += 4;
+            e -= 4;
+        }
+    }
+
+    // data must be in buf2
+
+
+    // step 8+decode   (paper output is X, now buffer)
+    // this generates pairs of data a la 8 and pushes them directly through
+    // the decode kernel (pushing rather than pulling) to avoid having
+    // to make another pass later
+
+    // this cannot POSSIBLY be in place, so we refer to the buffers directly
+
+    {
+        float *d0, *d1, *d2, *d3;
+
+        float *B = f->B[blocktype] + n2 - 8;
+        float *e = buf2 + n2 - 8;
+        d0 = &buffer[0];
+        d1 = &buffer[n2 - 4];
+        d2 = &buffer[n2];
+        d3 = &buffer[n - 4];
+        while (e >= v) {
+            float p0, p1, p2, p3;
+
+            p3 = e[6] * B[7] - e[7] * B[6];
+            p2 = -e[6] * B[6] - e[7] * B[7];
+
+            d0[0] = p3;
+            d1[3] = -p3;
+            d2[0] = p2;
+            d3[3] = p2;
+
+            p1 = e[4] * B[5] - e[5] * B[4];
+            p0 = -e[4] * B[4] - e[5] * B[5];
+
+            d0[1] = p1;
+            d1[2] = -p1;
+            d2[1] = p0;
+            d3[2] = p0;
+
+            p3 = e[2] * B[3] - e[3] * B[2];
+            p2 = -e[2] * B[2] - e[3] * B[3];
+
+            d0[2] = p3;
+            d1[1] = -p3;
+            d2[2] = p2;
+            d3[1] = p2;
+
+            p1 = e[0] * B[1] - e[1] * B[0];
+            p0 = -e[0] * B[0] - e[1] * B[1];
+
+            d0[3] = p1;
+            d1[0] = -p1;
+            d2[3] = p0;
+            d3[0] = p0;
+
+            B -= 8;
+            e -= 8;
+            d0 += 4;
+            d2 += 4;
+            d1 -= 4;
+            d3 -= 4;
+        }
+    }
+
+    temp_free(f, buf2);
+    temp_alloc_restore(f, save_point);
+}
+
+#if 0
+// this is the original version of the above code, if you want to optimize it from scratch
+void inverse_mdct_naive(float *buffer, int n)
+{
+    float s;
+    float A[1 << 12], B[1 << 12], C[1 << 11];
+    int i, k, k2, k4, n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l;
+    int n3_4 = n - n4, ld;
+    // how can they claim this only uses N words?!
+    // oh, because they're only used sparsely, whoops
+    float u[1 << 13], X[1 << 13], v[1 << 13], w[1 << 13];
+    // set up twiddle factors
+
+    for (k = k2 = 0; k < n4; ++k, k2 += 2) {
+        A[k2] = (float)cos(4 * k*M_PI / n);
+        A[k2 + 1] = (float)-sin(4 * k*M_PI / n);
+        B[k2] = (float)cos((k2 + 1)*M_PI / n / 2);
+        B[k2 + 1] = (float)sin((k2 + 1)*M_PI / n / 2);
+    }
+    for (k = k2 = 0; k < n8; ++k, k2 += 2) {
+        C[k2] = (float)cos(2 * (k2 + 1)*M_PI / n);
+        C[k2 + 1] = (float)-sin(2 * (k2 + 1)*M_PI / n);
+    }
+
+    // IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio"
+    // Note there are bugs in that pseudocode, presumably due to them attempting
+    // to rename the arrays nicely rather than representing the way their actual
+    // implementation bounces buffers back and forth. As a result, even in the
+    // "some formulars corrected" version, a direct implementation fails. These
+    // are noted below as "paper bug".
+
+    // copy and reflect spectral data
+    for (k = 0; k < n2; ++k) u[k] = buffer[k];
+    for (; k < n; ++k) u[k] = -buffer[n - k - 1];
+    // kernel from paper
+    // step 1
+    for (k = k2 = k4 = 0; k < n4; k += 1, k2 += 2, k4 += 4) {
+        v[n - k4 - 1] = (u[k4] - u[n - k4 - 1]) * A[k2] - (u[k4 + 2] - u[n - k4 - 3])*A[k2 + 1];
+        v[n - k4 - 3] = (u[k4] - u[n - k4 - 1]) * A[k2 + 1] + (u[k4 + 2] - u[n - k4 - 3])*A[k2];
+    }
+    // step 2
+    for (k = k4 = 0; k < n8; k += 1, k4 += 4) {
+        w[n2 + 3 + k4] = v[n2 + 3 + k4] + v[k4 + 3];
+        w[n2 + 1 + k4] = v[n2 + 1 + k4] + v[k4 + 1];
+        w[k4 + 3] = (v[n2 + 3 + k4] - v[k4 + 3])*A[n2 - 4 - k4] - (v[n2 + 1 + k4] - v[k4 + 1])*A[n2 - 3 - k4];
+        w[k4 + 1] = (v[n2 + 1 + k4] - v[k4 + 1])*A[n2 - 4 - k4] + (v[n2 + 3 + k4] - v[k4 + 3])*A[n2 - 3 - k4];
+    }
+    // step 3
+    ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
+    for (l = 0; l < ld - 3; ++l) {
+        int k0 = n >> (l + 2), k1 = 1 << (l + 3);
+        int rlim = n >> (l + 4), r4, r;
+        int s2lim = 1 << (l + 2), s2;
+        for (r = r4 = 0; r < rlim; r4 += 4, ++r) {
+            for (s2 = 0; s2 < s2lim; s2 += 2) {
+                u[n - 1 - k0*s2 - r4] = w[n - 1 - k0*s2 - r4] + w[n - 1 - k0*(s2 + 1) - r4];
+                u[n - 3 - k0*s2 - r4] = w[n - 3 - k0*s2 - r4] + w[n - 3 - k0*(s2 + 1) - r4];
+                u[n - 1 - k0*(s2 + 1) - r4] = (w[n - 1 - k0*s2 - r4] - w[n - 1 - k0*(s2 + 1) - r4]) * A[r*k1]
+                    - (w[n - 3 - k0*s2 - r4] - w[n - 3 - k0*(s2 + 1) - r4]) * A[r*k1 + 1];
+                u[n - 3 - k0*(s2 + 1) - r4] = (w[n - 3 - k0*s2 - r4] - w[n - 3 - k0*(s2 + 1) - r4]) * A[r*k1]
+                    + (w[n - 1 - k0*s2 - r4] - w[n - 1 - k0*(s2 + 1) - r4]) * A[r*k1 + 1];
+            }
+        }
+        if (l + 1 < ld - 3) {
+            // paper bug: ping-ponging of u&w here is omitted
+            memcpy(w, u, sizeof(u));
+        }
+    }
+
+    // step 4
+    for (i = 0; i < n8; ++i) {
+        int j = bit_reverse(i) >> (32 - ld + 3);
+        assert(j < n8);
+        if (i == j) {
+            // paper bug: original code probably swapped in place; if copying,
+            //            need to directly copy in this case
+            int i8 = i << 3;
+            v[i8 + 1] = u[i8 + 1];
+            v[i8 + 3] = u[i8 + 3];
+            v[i8 + 5] = u[i8 + 5];
+            v[i8 + 7] = u[i8 + 7];
+        }
+        else if (i < j) {
+            int i8 = i << 3, j8 = j << 3;
+            v[j8 + 1] = u[i8 + 1], v[i8 + 1] = u[j8 + 1];
+            v[j8 + 3] = u[i8 + 3], v[i8 + 3] = u[j8 + 3];
+            v[j8 + 5] = u[i8 + 5], v[i8 + 5] = u[j8 + 5];
+            v[j8 + 7] = u[i8 + 7], v[i8 + 7] = u[j8 + 7];
+        }
+    }
+    // step 5
+    for (k = 0; k < n2; ++k) {
+        w[k] = v[k * 2 + 1];
+    }
+    // step 6
+    for (k = k2 = k4 = 0; k < n8; ++k, k2 += 2, k4 += 4) {
+        u[n - 1 - k2] = w[k4];
+        u[n - 2 - k2] = w[k4 + 1];
+        u[n3_4 - 1 - k2] = w[k4 + 2];
+        u[n3_4 - 2 - k2] = w[k4 + 3];
+    }
+    // step 7
+    for (k = k2 = 0; k < n8; ++k, k2 += 2) {
+        v[n2 + k2] = (u[n2 + k2] + u[n - 2 - k2] + C[k2 + 1] * (u[n2 + k2] - u[n - 2 - k2]) + C[k2] * (u[n2 + k2 + 1] + u[n - 2 - k2 + 1])) / 2;
+        v[n - 2 - k2] = (u[n2 + k2] + u[n - 2 - k2] - C[k2 + 1] * (u[n2 + k2] - u[n - 2 - k2]) - C[k2] * (u[n2 + k2 + 1] + u[n - 2 - k2 + 1])) / 2;
+        v[n2 + 1 + k2] = (u[n2 + 1 + k2] - u[n - 1 - k2] + C[k2 + 1] * (u[n2 + 1 + k2] + u[n - 1 - k2]) - C[k2] * (u[n2 + k2] - u[n - 2 - k2])) / 2;
+        v[n - 1 - k2] = (-u[n2 + 1 + k2] + u[n - 1 - k2] + C[k2 + 1] * (u[n2 + 1 + k2] + u[n - 1 - k2]) - C[k2] * (u[n2 + k2] - u[n - 2 - k2])) / 2;
+    }
+    // step 8
+    for (k = k2 = 0; k < n4; ++k, k2 += 2) {
+        X[k] = v[k2 + n2] * B[k2] + v[k2 + 1 + n2] * B[k2 + 1];
+        X[n2 - 1 - k] = v[k2 + n2] * B[k2 + 1] - v[k2 + 1 + n2] * B[k2];
+    }
+
+    // decode kernel to output
+    // determined the following value experimentally
+    // (by first figuring out what made inverse_mdct_slow work); then matching that here
+    // (probably vorbis encoder premultiplies by n or n/2, to save it on the decoder?)
+    s = 0.5; // theoretically would be n4
+
+             // [[[ note! the s value of 0.5 is compensated for by the B[] in the current code,
+             //     so it needs to use the "old" B values to behave correctly, or else
+             //     set s to 1.0 ]]]
+    for (i = 0; i < n4; ++i) buffer[i] = s * X[i + n4];
+    for (; i < n3_4; ++i) buffer[i] = -s * X[n3_4 - i - 1];
+    for (; i < n; ++i) buffer[i] = -s * X[i - n3_4];
+}
+#endif
+
+static float *get_window(vorb *f, int len)
+{
+    len <<= 1;
+    if (len == f->blocksize_0) return f->window[0];
+    if (len == f->blocksize_1) return f->window[1];
+    assert(0);
+    return NULL;
+}
+
+#ifndef STB_VORBIS_NO_DEFER_FLOOR
+typedef int16 YTYPE;
+#else
+typedef int YTYPE;
+#endif
+static int do_floor(vorb *f, Mapping *map, int i, int n, float *target, YTYPE *finalY, uint8 *step2_flag)
+{
+    int n2 = n >> 1;
+    int s = map->chan[i].mux, floor;
+    floor = map->submap_floor[s];
+    if (f->floor_types[floor] == 0) {
+        return error(f, VORBIS_invalid_stream);
+    }
+    else {
+        Floor1 *g = &f->floor_config[floor].floor1;
+        int j, q;
+        int lx = 0, ly = finalY[0] * g->floor1_multiplier;
+        for (q = 1; q < g->values; ++q) {
+            j = g->sorted_order[q];
+#ifndef STB_VORBIS_NO_DEFER_FLOOR
+            if (finalY[j] >= 0)
+#else
+            if (step2_flag[j])
+#endif
+            {
+                int hy = finalY[j] * g->floor1_multiplier;
+                int hx = g->Xlist[j];
+                if (lx != hx)
+                    draw_line(target, lx, ly, hx, hy, n2);
+                CHECK(f);
+                lx = hx, ly = hy;
+            }
+        }
+        if (lx < n2) {
+            // optimization of: draw_line(target, lx,ly, n,ly, n2);
+            for (j = lx; j < n2; ++j)
+                LINE_OP(target[j], inverse_db_table[ly]);
+            CHECK(f);
+        }
+    }
+    return TRUE;
+}
+
+// The meaning of "left" and "right"
+//
+// For a given frame:
+//     we compute samples from 0..n
+//     window_center is n/2
+//     we'll window and mix the samples from left_start to left_end with data from the previous frame
+//     all of the samples from left_end to right_start can be output without mixing; however,
+//        this interval is 0-length except when transitioning between short and long frames
+//     all of the samples from right_start to right_end need to be mixed with the next frame,
+//        which we don't have, so those get saved in a buffer
+//     frame N's right_end-right_start, the number of samples to mix with the next frame,
+//        has to be the same as frame N+1's left_end-left_start (which they are by
+//        construction)
+
+static int vorbis_decode_initial(vorb *f, int *p_left_start, int *p_left_end, int *p_right_start, int *p_right_end, int *mode)
+{
+    Mode *m;
+    int i, n, prev, next, window_center;
+    f->channel_buffer_start = f->channel_buffer_end = 0;
+
+retry:
+    if (f->eof) return FALSE;
+    if (!maybe_start_packet(f))
+        return FALSE;
+    // check packet type
+    if (get_bits(f, 1) != 0) {
+        if (IS_PUSH_MODE(f))
+            return error(f, VORBIS_bad_packet_type);
+        while (EOP != get8_packet(f));
+        goto retry;
+    }
+
+    if (f->alloc.alloc_buffer)
+        assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
+
+    i = get_bits(f, ilog(f->mode_count - 1));
+    if (i == EOP) return FALSE;
+    if (i >= f->mode_count) return FALSE;
+    *mode = i;
+    m = f->mode_config + i;
+    if (m->blockflag) {
+        n = f->blocksize_1;
+        prev = get_bits(f, 1);
+        next = get_bits(f, 1);
+    }
+    else {
+        prev = next = 0;
+        n = f->blocksize_0;
+    }
+
+    // WINDOWING
+
+    window_center = n >> 1;
+    if (m->blockflag && !prev) {
+        *p_left_start = (n - f->blocksize_0) >> 2;
+        *p_left_end = (n + f->blocksize_0) >> 2;
+    }
+    else {
+        *p_left_start = 0;
+        *p_left_end = window_center;
+    }
+    if (m->blockflag && !next) {
+        *p_right_start = (n * 3 - f->blocksize_0) >> 2;
+        *p_right_end = (n * 3 + f->blocksize_0) >> 2;
+    }
+    else {
+        *p_right_start = window_center;
+        *p_right_end = n;
+    }
+
+    return TRUE;
+}
+
+static int vorbis_decode_packet_rest(vorb *f, int *len, Mode *m, int left_start, int left_end, int right_start, int right_end, int *p_left)
+{
+    Mapping *map;
+    int i, j, k, n, n2;
+    int zero_channel[256];
+    int really_zero_channel[256];
+
+    // WINDOWING
+
+    n = f->blocksize[m->blockflag];
+    map = &f->mapping[m->mapping];
+
+    // FLOORS
+    n2 = n >> 1;
+
+    CHECK(f);
+
+    for (i = 0; i < f->channels; ++i) {
+        int s = map->chan[i].mux, floor;
+        zero_channel[i] = FALSE;
+        floor = map->submap_floor[s];
+        if (f->floor_types[floor] == 0) {
+            return error(f, VORBIS_invalid_stream);
+        }
+        else {
+            Floor1 *g = &f->floor_config[floor].floor1;
+            if (get_bits(f, 1)) {
+                short *finalY;
+                uint8 step2_flag[256];
+                static int range_list[4] = { 256, 128, 86, 64 };
+                int range = range_list[g->floor1_multiplier - 1];
+                int offset = 2;
+                finalY = f->finalY[i];
+                finalY[0] = get_bits(f, ilog(range) - 1);
+                finalY[1] = get_bits(f, ilog(range) - 1);
+                for (j = 0; j < g->partitions; ++j) {
+                    int pclass = g->partition_class_list[j];
+                    int cdim = g->class_dimensions[pclass];
+                    int cbits = g->class_subclasses[pclass];
+                    int csub = (1 << cbits) - 1;
+                    int cval = 0;
+                    if (cbits) {
+                        Codebook *c = f->codebooks + g->class_masterbooks[pclass];
+                        DECODE(cval, f, c);
+                    }
+                    for (k = 0; k < cdim; ++k) {
+                        int book = g->subclass_books[pclass][cval & csub];
+                        cval = cval >> cbits;
+                        if (book >= 0) {
+                            int temp;
+                            Codebook *c = f->codebooks + book;
+                            DECODE(temp, f, c);
+                            finalY[offset++] = temp;
+                        }
+                        else
+                            finalY[offset++] = 0;
+                    }
+                }
+                if (f->valid_bits == INVALID_BITS) goto error; // behavior according to spec
+                step2_flag[0] = step2_flag[1] = 1;
+                for (j = 2; j < g->values; ++j) {
+                    int low, high, pred, highroom, lowroom, room, val;
+                    low = g->neighbors[j][0];
+                    high = g->neighbors[j][1];
+                    //neighbors(g->Xlist, j, &low, &high);
+                    pred = predict_point(g->Xlist[j], g->Xlist[low], g->Xlist[high], finalY[low], finalY[high]);
+                    val = finalY[j];
+                    highroom = range - pred;
+                    lowroom = pred;
+                    if (highroom < lowroom)
+                        room = highroom * 2;
+                    else
+                        room = lowroom * 2;
+                    if (val) {
+                        step2_flag[low] = step2_flag[high] = 1;
+                        step2_flag[j] = 1;
+                        if (val >= room)
+                            if (highroom > lowroom)
+                                finalY[j] = val - lowroom + pred;
+                            else
+                                finalY[j] = pred - val + highroom - 1;
+                        else
+                            if (val & 1)
+                                finalY[j] = pred - ((val + 1) >> 1);
+                            else
+                                finalY[j] = pred + (val >> 1);
+                    }
+                    else {
+                        step2_flag[j] = 0;
+                        finalY[j] = pred;
+                    }
+                }
+
+#ifdef STB_VORBIS_NO_DEFER_FLOOR
+                do_floor(f, map, i, n, f->floor_buffers[i], finalY, step2_flag);
+#else
+                // defer final floor computation until _after_ residue
+                for (j = 0; j < g->values; ++j) {
+                    if (!step2_flag[j])
+                        finalY[j] = -1;
+                }
+#endif
+            }
+            else {
+            error:
+                zero_channel[i] = TRUE;
+            }
+            // So we just defer everything else to later
+
+            // at this point we've decoded the floor into buffer
+        }
+    }
+    CHECK(f);
+    // at this point we've decoded all floors
+
+    if (f->alloc.alloc_buffer)
+        assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
+
+    // re-enable coupled channels if necessary
+    memcpy(really_zero_channel, zero_channel, sizeof(really_zero_channel[0]) * f->channels);
+    for (i = 0; i < map->coupling_steps; ++i)
+        if (!zero_channel[map->chan[i].magnitude] || !zero_channel[map->chan[i].angle]) {
+            zero_channel[map->chan[i].magnitude] = zero_channel[map->chan[i].angle] = FALSE;
+        }
+
+    CHECK(f);
+    // RESIDUE DECODE
+    for (i = 0; i < map->submaps; ++i) {
+        float *residue_buffers[STB_VORBIS_MAX_CHANNELS];
+        int r;
+        uint8 do_not_decode[256];
+        int ch = 0;
+        for (j = 0; j < f->channels; ++j) {
+            if (map->chan[j].mux == i) {
+                if (zero_channel[j]) {
+                    do_not_decode[ch] = TRUE;
+                    residue_buffers[ch] = NULL;
+                }
+                else {
+                    do_not_decode[ch] = FALSE;
+                    residue_buffers[ch] = f->channel_buffers[j];
+                }
+                ++ch;
+            }
+        }
+        r = map->submap_residue[i];
+        decode_residue(f, residue_buffers, ch, n2, r, do_not_decode);
+    }
+
+    if (f->alloc.alloc_buffer)
+        assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
+    CHECK(f);
+
+    // INVERSE COUPLING
+    for (i = map->coupling_steps - 1; i >= 0; --i) {
+        int n2 = n >> 1;
+        float *m = f->channel_buffers[map->chan[i].magnitude];
+        float *a = f->channel_buffers[map->chan[i].angle];
+        for (j = 0; j < n2; ++j) {
+            float a2, m2;
+            if (m[j] > 0)
+                if (a[j] > 0)
+                    m2 = m[j], a2 = m[j] - a[j];
+                else
+                    a2 = m[j], m2 = m[j] + a[j];
+            else
+                if (a[j] > 0)
+                    m2 = m[j], a2 = m[j] + a[j];
+                else
+                    a2 = m[j], m2 = m[j] - a[j];
+            m[j] = m2;
+            a[j] = a2;
+        }
+    }
+    CHECK(f);
+
+    // finish decoding the floors
+#ifndef STB_VORBIS_NO_DEFER_FLOOR
+    for (i = 0; i < f->channels; ++i) {
+        if (really_zero_channel[i]) {
+            memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2);
+        }
+        else {
+            do_floor(f, map, i, n, f->channel_buffers[i], f->finalY[i], NULL);
+        }
+    }
+#else
+    for (i = 0; i < f->channels; ++i) {
+        if (really_zero_channel[i]) {
+            memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2);
+        }
+        else {
+            for (j = 0; j < n2; ++j)
+                f->channel_buffers[i][j] *= f->floor_buffers[i][j];
+        }
+    }
+#endif
+
+    // INVERSE MDCT
+    CHECK(f);
+    for (i = 0; i < f->channels; ++i)
+        inverse_mdct(f->channel_buffers[i], n, f, m->blockflag);
+    CHECK(f);
+
+    // this shouldn't be necessary, unless we exited on an error
+    // and want to flush to get to the next packet
+    flush_packet(f);
+
+    if (f->first_decode) {
+        // assume we start so first non-discarded sample is sample 0
+        // this isn't to spec, but spec would require us to read ahead
+        // and decode the size of all current frames--could be done,
+        // but presumably it's not a commonly used feature
+        f->current_loc = -n2; // start of first frame is positioned for discard
+                              // we might have to discard samples "from" the next frame too,
+                              // if we're lapping a large block then a small at the start?
+        f->discard_samples_deferred = n - right_end;
+        f->current_loc_valid = TRUE;
+        f->first_decode = FALSE;
+    }
+    else if (f->discard_samples_deferred) {
+        if (f->discard_samples_deferred >= right_start - left_start) {
+            f->discard_samples_deferred -= (right_start - left_start);
+            left_start = right_start;
+            *p_left = left_start;
+        }
+        else {
+            left_start += f->discard_samples_deferred;
+            *p_left = left_start;
+            f->discard_samples_deferred = 0;
+        }
+    }
+    else if (f->previous_length == 0 && f->current_loc_valid) {
+        // we're recovering from a seek... that means we're going to discard
+        // the samples from this packet even though we know our position from
+        // the last page header, so we need to update the position based on
+        // the discarded samples here
+        // but wait, the code below is going to add this in itself even
+        // on a discard, so we don't need to do it here...
+    }
+
+    // check if we have ogg information about the sample # for this packet
+    if (f->last_seg_which == f->end_seg_with_known_loc) {
+        // if we have a valid current loc, and this is final:
+        if (f->current_loc_valid && (f->page_flag & PAGEFLAG_last_page)) {
+            uint32 current_end = f->known_loc_for_packet;
+            // then let's infer the size of the (probably) short final frame
+            if (current_end < f->current_loc + (right_end - left_start)) {
+                if (current_end < f->current_loc) {
+                    // negative truncation, that's impossible!
+                    *len = 0;
+                }
+                else {
+                    *len = current_end - f->current_loc;
+                }
+                *len += left_start; // this doesn't seem right, but has no ill effect on my test files
+                if (*len > right_end) *len = right_end; // this should never happen
+                f->current_loc += *len;
+                return TRUE;
+            }
+        }
+        // otherwise, just set our sample loc
+        // guess that the ogg granule pos refers to the _middle_ of the
+        // last frame?
+        // set f->current_loc to the position of left_start
+        f->current_loc = f->known_loc_for_packet - (n2 - left_start);
+        f->current_loc_valid = TRUE;
+    }
+    if (f->current_loc_valid)
+        f->current_loc += (right_start - left_start);
+
+    if (f->alloc.alloc_buffer)
+        assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
+    *len = right_end;  // ignore samples after the window goes to 0
+    CHECK(f);
+
+    return TRUE;
+}
+
+static int vorbis_decode_packet(vorb *f, int *len, int *p_left, int *p_right)
+{
+    int mode, left_end, right_end;
+    if (!vorbis_decode_initial(f, p_left, &left_end, p_right, &right_end, &mode)) return 0;
+    return vorbis_decode_packet_rest(f, len, f->mode_config + mode, *p_left, left_end, *p_right, right_end, p_left);
+}
+
+static int vorbis_finish_frame(stb_vorbis *f, int len, int left, int right)
+{
+    int prev, i, j;
+    // we use right&left (the start of the right- and left-window sin()-regions)
+    // to determine how much to return, rather than inferring from the rules
+    // (same result, clearer code); 'left' indicates where our sin() window
+    // starts, therefore where the previous window's right edge starts, and
+    // therefore where to start mixing from the previous buffer. 'right'
+    // indicates where our sin() ending-window starts, therefore that's where
+    // we start saving, and where our returned-data ends.
+
+    // mixin from previous window
+    if (f->previous_length) {
+        int i, j, n = f->previous_length;
+        float *w = get_window(f, n);
+        for (i = 0; i < f->channels; ++i) {
+            for (j = 0; j < n; ++j)
+                f->channel_buffers[i][left + j] =
+                f->channel_buffers[i][left + j] * w[j] +
+                f->previous_window[i][j] * w[n - 1 - j];
+        }
+    }
+
+    prev = f->previous_length;
+
+    // last half of this data becomes previous window
+    f->previous_length = len - right;
+
+    // @OPTIMIZE: could avoid this copy by double-buffering the
+    // output (flipping previous_window with channel_buffers), but
+    // then previous_window would have to be 2x as large, and
+    // channel_buffers couldn't be temp mem (although they're NOT
+    // currently temp mem, they could be (unless we want to level
+    // performance by spreading out the computation))
+    for (i = 0; i < f->channels; ++i)
+        for (j = 0; right + j < len; ++j)
+            f->previous_window[i][j] = f->channel_buffers[i][right + j];
+
+    if (!prev)
+        // there was no previous packet, so this data isn't valid...
+        // this isn't entirely true, only the would-have-overlapped data
+        // isn't valid, but this seems to be what the spec requires
+        return 0;
+
+    // truncate a short frame
+    if (len < right) right = len;
+
+    f->samples_output += right - left;
+
+    return right - left;
+}
+
+static int vorbis_pump_first_frame(stb_vorbis *f)
+{
+    int len, right, left, res;
+    res = vorbis_decode_packet(f, &len, &left, &right);
+    if (res)
+        vorbis_finish_frame(f, len, left, right);
+    return res;
+}
+
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+static int is_whole_packet_present(stb_vorbis *f, int end_page)
+{
+    // make sure that we have the packet available before continuing...
+    // this requires a full ogg parse, but we know we can fetch from f->stream
+
+    // instead of coding this out explicitly, we could save the current read state,
+    // read the next packet with get8() until end-of-packet, check f->eof, then
+    // reset the state? but that would be slower, esp. since we'd have over 256 bytes
+    // of state to restore (primarily the page segment table)
+
+    int s = f->next_seg, first = TRUE;
+    uint8 *p = f->stream;
+
+    if (s != -1) { // if we're not starting the packet with a 'continue on next page' flag
+        for (; s < f->segment_count; ++s) {
+            p += f->segments[s];
+            if (f->segments[s] < 255)               // stop at first short segment
+                break;
+        }
+        // either this continues, or it ends it...
+        if (end_page)
+            if (s < f->segment_count - 1)             return error(f, VORBIS_invalid_stream);
+        if (s == f->segment_count)
+            s = -1; // set 'crosses page' flag
+        if (p > f->stream_end)                     return error(f, VORBIS_need_more_data);
+        first = FALSE;
+    }
+    for (; s == -1;) {
+        uint8 *q;
+        int n;
+
+        // check that we have the page header ready
+        if (p + 26 >= f->stream_end)               return error(f, VORBIS_need_more_data);
+        // validate the page
+        if (memcmp(p, ogg_page_header, 4))         return error(f, VORBIS_invalid_stream);
+        if (p[4] != 0)                             return error(f, VORBIS_invalid_stream);
+        if (first) { // the first segment must NOT have 'continued_packet', later ones MUST
+            if (f->previous_length)
+                if ((p[5] & PAGEFLAG_continued_packet))  return error(f, VORBIS_invalid_stream);
+            // if no previous length, we're resynching, so we can come in on a continued-packet,
+            // which we'll just drop
+        }
+        else {
+            if (!(p[5] & PAGEFLAG_continued_packet)) return error(f, VORBIS_invalid_stream);
+        }
+        n = p[26]; // segment counts
+        q = p + 27;  // q points to segment table
+        p = q + n; // advance past header
+                   // make sure we've read the segment table
+        if (p > f->stream_end)                     return error(f, VORBIS_need_more_data);
+        for (s = 0; s < n; ++s) {
+            p += q[s];
+            if (q[s] < 255)
+                break;
+        }
+        if (end_page)
+            if (s < n - 1)                            return error(f, VORBIS_invalid_stream);
+        if (s == n)
+            s = -1; // set 'crosses page' flag
+        if (p > f->stream_end)                     return error(f, VORBIS_need_more_data);
+        first = FALSE;
+    }
+    return TRUE;
+}
+#endif // !STB_VORBIS_NO_PUSHDATA_API
+
+static int start_decoder(vorb *f)
+{
+    uint8 header[6], x, y;
+    int len, i, j, k, max_submaps = 0;
+    int longest_floorlist = 0;
+
+    // first page, first packet
+
+    if (!start_page(f))                              return FALSE;
+    // validate page flag
+    if (!(f->page_flag & PAGEFLAG_first_page))       return error(f, VORBIS_invalid_first_page);
+    if (f->page_flag & PAGEFLAG_last_page)           return error(f, VORBIS_invalid_first_page);
+    if (f->page_flag & PAGEFLAG_continued_packet)    return error(f, VORBIS_invalid_first_page);
+    // check for expected packet length
+    if (f->segment_count != 1)                       return error(f, VORBIS_invalid_first_page);
+    if (f->segments[0] != 30)                        return error(f, VORBIS_invalid_first_page);
+    // read packet
+    // check packet header
+    if (get8(f) != VORBIS_packet_id)                 return error(f, VORBIS_invalid_first_page);
+    if (!getn(f, header, 6))                         return error(f, VORBIS_unexpected_eof);
+    if (!vorbis_validate(header))                    return error(f, VORBIS_invalid_first_page);
+    // vorbis_version
+    if (get32(f) != 0)                               return error(f, VORBIS_invalid_first_page);
+    f->channels = get8(f); if (!f->channels)         return error(f, VORBIS_invalid_first_page);
+    if (f->channels > STB_VORBIS_MAX_CHANNELS)       return error(f, VORBIS_too_many_channels);
+    f->sample_rate = get32(f); if (!f->sample_rate)  return error(f, VORBIS_invalid_first_page);
+    get32(f); // bitrate_maximum
+    get32(f); // bitrate_nominal
+    get32(f); // bitrate_minimum
+    x = get8(f);
+    {
+        int log0, log1;
+        log0 = x & 15;
+        log1 = x >> 4;
+        f->blocksize_0 = 1 << log0;
+        f->blocksize_1 = 1 << log1;
+        if (log0 < 6 || log0 > 13)                       return error(f, VORBIS_invalid_setup);
+        if (log1 < 6 || log1 > 13)                       return error(f, VORBIS_invalid_setup);
+        if (log0 > log1)                                 return error(f, VORBIS_invalid_setup);
+    }
+
+    // framing_flag
+    x = get8(f);
+    if (!(x & 1))                                    return error(f, VORBIS_invalid_first_page);
+
+    // second packet!
+    if (!start_page(f))                              return FALSE;
+
+    if (!start_packet(f))                            return FALSE;
+    do {
+        len = next_segment(f);
+        skip(f, len);
+        f->bytes_in_seg = 0;
+    } while (len);
+
+    // third packet!
+    if (!start_packet(f))                            return FALSE;
+
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+    if (IS_PUSH_MODE(f)) {
+        if (!is_whole_packet_present(f, TRUE)) {
+            // convert error in ogg header to write type
+            if (f->error == VORBIS_invalid_stream)
+                f->error = VORBIS_invalid_setup;
+            return FALSE;
+        }
+    }
+#endif
+
+    crc32_init(); // always init it, to avoid multithread race conditions
+
+    if (get8_packet(f) != VORBIS_packet_setup)       return error(f, VORBIS_invalid_setup);
+    for (i = 0; i < 6; ++i) header[i] = get8_packet(f);
+    if (!vorbis_validate(header))                    return error(f, VORBIS_invalid_setup);
+
+    // codebooks
+
+    f->codebook_count = get_bits(f, 8) + 1;
+    f->codebooks = (Codebook *)setup_malloc(f, sizeof(*f->codebooks) * f->codebook_count);
+    if (f->codebooks == NULL)                        return error(f, VORBIS_outofmem);
+    memset(f->codebooks, 0, sizeof(*f->codebooks) * f->codebook_count);
+    for (i = 0; i < f->codebook_count; ++i) {
+        uint32 *values;
+        int ordered, sorted_count;
+        int total = 0;
+        uint8 *lengths;
+        Codebook *c = f->codebooks + i;
+        CHECK(f);
+        x = get_bits(f, 8); if (x != 0x42)            return error(f, VORBIS_invalid_setup);
+        x = get_bits(f, 8); if (x != 0x43)            return error(f, VORBIS_invalid_setup);
+        x = get_bits(f, 8); if (x != 0x56)            return error(f, VORBIS_invalid_setup);
+        x = get_bits(f, 8);
+        c->dimensions = (get_bits(f, 8) << 8) + x;
+        x = get_bits(f, 8);
+        y = get_bits(f, 8);
+        c->entries = (get_bits(f, 8) << 16) + (y << 8) + x;
+        ordered = get_bits(f, 1);
+        c->sparse = ordered ? 0 : get_bits(f, 1);
+
+        if (c->dimensions == 0 && c->entries != 0)    return error(f, VORBIS_invalid_setup);
+
+        if (c->sparse)
+            lengths = (uint8 *)setup_temp_malloc(f, c->entries);
+        else
+            lengths = c->codeword_lengths = (uint8 *)setup_malloc(f, c->entries);
+
+        if (!lengths) return error(f, VORBIS_outofmem);
+
+        if (ordered) {
+            int current_entry = 0;
+            int current_length = get_bits(f, 5) + 1;
+            while (current_entry < c->entries) {
+                int limit = c->entries - current_entry;
+                int n = get_bits(f, ilog(limit));
+                if (current_entry + n > (int) c->entries) { return error(f, VORBIS_invalid_setup); }
+                memset(lengths + current_entry, current_length, n);
+                current_entry += n;
+                ++current_length;
+            }
+        }
+        else {
+            for (j = 0; j < c->entries; ++j) {
+                int present = c->sparse ? get_bits(f, 1) : 1;
+                if (present) {
+                    lengths[j] = get_bits(f, 5) + 1;
+                    ++total;
+                    if (lengths[j] == 32)
+                        return error(f, VORBIS_invalid_setup);
+                }
+                else {
+                    lengths[j] = NO_CODE;
+                }
+            }
+        }
+
+        if (c->sparse && total >= c->entries >> 2) {
+            // convert sparse items to non-sparse!
+            if (c->entries > (int)f->setup_temp_memory_required)
+                f->setup_temp_memory_required = c->entries;
+
+            c->codeword_lengths = (uint8 *)setup_malloc(f, c->entries);
+            if (c->codeword_lengths == NULL) return error(f, VORBIS_outofmem);
+            memcpy(c->codeword_lengths, lengths, c->entries);
+            setup_temp_free(f, lengths, c->entries); // note this is only safe if there have been no intervening temp mallocs!
+            lengths = c->codeword_lengths;
+            c->sparse = 0;
+        }
+
+        // compute the size of the sorted tables
+        if (c->sparse) {
+            sorted_count = total;
+        }
+        else {
+            sorted_count = 0;
+#ifndef STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH
+            for (j = 0; j < c->entries; ++j)
+                if (lengths[j] > STB_VORBIS_FAST_HUFFMAN_LENGTH && lengths[j] != NO_CODE)
+                    ++sorted_count;
+#endif
+        }
+
+        c->sorted_entries = sorted_count;
+        values = NULL;
+
+        CHECK(f);
+        if (!c->sparse) {
+            c->codewords = (uint32 *)setup_malloc(f, sizeof(c->codewords[0]) * c->entries);
+            if (!c->codewords)                  return error(f, VORBIS_outofmem);
+        }
+        else {
+            unsigned int size;
+            if (c->sorted_entries) {
+                c->codeword_lengths = (uint8 *)setup_malloc(f, c->sorted_entries);
+                if (!c->codeword_lengths)           return error(f, VORBIS_outofmem);
+                c->codewords = (uint32 *)setup_temp_malloc(f, sizeof(*c->codewords) * c->sorted_entries);
+                if (!c->codewords)                  return error(f, VORBIS_outofmem);
+                values = (uint32 *)setup_temp_malloc(f, sizeof(*values) * c->sorted_entries);
+                if (!values)                        return error(f, VORBIS_outofmem);
+            }
+            size = c->entries + (sizeof(*c->codewords) + sizeof(*values)) * c->sorted_entries;
+            if (size > f->setup_temp_memory_required)
+                f->setup_temp_memory_required = size;
+        }
+
+        if (!compute_codewords(c, lengths, c->entries, values)) {
+            if (c->sparse) setup_temp_free(f, values, 0);
+            return error(f, VORBIS_invalid_setup);
+        }
+
+        if (c->sorted_entries) {
+            // allocate an extra slot for sentinels
+            c->sorted_codewords = (uint32 *)setup_malloc(f, sizeof(*c->sorted_codewords) * (c->sorted_entries + 1));
+            if (c->sorted_codewords == NULL) return error(f, VORBIS_outofmem);
+            // allocate an extra slot at the front so that c->sorted_values[-1] is defined
+            // so that we can catch that case without an extra if
+            c->sorted_values = (int   *)setup_malloc(f, sizeof(*c->sorted_values) * (c->sorted_entries + 1));
+            if (c->sorted_values == NULL) return error(f, VORBIS_outofmem);
+            ++c->sorted_values;
+            c->sorted_values[-1] = -1;
+            compute_sorted_huffman(c, lengths, values);
+        }
+
+        if (c->sparse) {
+            setup_temp_free(f, values, sizeof(*values)*c->sorted_entries);
+            setup_temp_free(f, c->codewords, sizeof(*c->codewords)*c->sorted_entries);
+            setup_temp_free(f, lengths, c->entries);
+            c->codewords = NULL;
+        }
+
+        compute_accelerated_huffman(c);
+
+        CHECK(f);
+        c->lookup_type = get_bits(f, 4);
+        if (c->lookup_type > 2) return error(f, VORBIS_invalid_setup);
+        if (c->lookup_type > 0) {
+            uint16 *mults;
+            c->minimum_value = float32_unpack(get_bits(f, 32));
+            c->delta_value = float32_unpack(get_bits(f, 32));
+            c->value_bits = get_bits(f, 4) + 1;
+            c->sequence_p = get_bits(f, 1);
+            if (c->lookup_type == 1) {
+                c->lookup_values = lookup1_values(c->entries, c->dimensions);
+            }
+            else {
+                c->lookup_values = c->entries * c->dimensions;
+            }
+            if (c->lookup_values == 0) return error(f, VORBIS_invalid_setup);
+            mults = (uint16 *)setup_temp_malloc(f, sizeof(mults[0]) * c->lookup_values);
+            if (mults == NULL) return error(f, VORBIS_outofmem);
+            for (j = 0; j < (int)c->lookup_values; ++j) {
+                int q = get_bits(f, c->value_bits);
+                if (q == EOP) { setup_temp_free(f, mults, sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_invalid_setup); }
+                mults[j] = q;
+            }
+
+#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
+            if (c->lookup_type == 1) {
+                int len, sparse = c->sparse;
+                float last = 0;
+                // pre-expand the lookup1-style multiplicands, to avoid a divide in the inner loop
+                if (sparse) {
+                    if (c->sorted_entries == 0) goto skip;
+                    c->multiplicands = (codetype *)setup_malloc(f, sizeof(c->multiplicands[0]) * c->sorted_entries * c->dimensions);
+                }
+                else
+                    c->multiplicands = (codetype *)setup_malloc(f, sizeof(c->multiplicands[0]) * c->entries        * c->dimensions);
+                if (c->multiplicands == NULL) { setup_temp_free(f, mults, sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_outofmem); }
+                len = sparse ? c->sorted_entries : c->entries;
+                for (j = 0; j < len; ++j) {
+                    unsigned int z = sparse ? c->sorted_values[j] : j;
+                    unsigned int div = 1;
+                    for (k = 0; k < c->dimensions; ++k) {
+                        int off = (z / div) % c->lookup_values;
+                        float val = mults[off];
+                        val = mults[off] * c->delta_value + c->minimum_value + last;
+                        c->multiplicands[j*c->dimensions + k] = val;
+                        if (c->sequence_p)
+                            last = val;
+                        if (k + 1 < c->dimensions) {
+                            if (div > UINT_MAX / (unsigned int)c->lookup_values) {
+                                setup_temp_free(f, mults, sizeof(mults[0])*c->lookup_values);
+                                return error(f, VORBIS_invalid_setup);
+                            }
+                            div *= c->lookup_values;
+                        }
+                    }
+                }
+                c->lookup_type = 2;
+            }
+            else
+#endif
+            {
+                float last = 0;
+                CHECK(f);
+                c->multiplicands = (codetype *)setup_malloc(f, sizeof(c->multiplicands[0]) * c->lookup_values);
+                if (c->multiplicands == NULL) { setup_temp_free(f, mults, sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_outofmem); }
+                for (j = 0; j < (int)c->lookup_values; ++j) {
+                    float val = mults[j] * c->delta_value + c->minimum_value + last;
+                    c->multiplicands[j] = val;
+                    if (c->sequence_p)
+                        last = val;
+                }
+            }
+#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
+            skip : ;
+#endif
+                   setup_temp_free(f, mults, sizeof(mults[0])*c->lookup_values);
+
+                   CHECK(f);
+        }
+        CHECK(f);
+    }
+
+    // time domain transfers (notused)
+
+    x = get_bits(f, 6) + 1;
+    for (i = 0; i < x; ++i) {
+        uint32 z = get_bits(f, 16);
+        if (z != 0) return error(f, VORBIS_invalid_setup);
+    }
+
+    // Floors
+    f->floor_count = get_bits(f, 6) + 1;
+    f->floor_config = (Floor *)setup_malloc(f, f->floor_count * sizeof(*f->floor_config));
+    if (f->floor_config == NULL) return error(f, VORBIS_outofmem);
+    for (i = 0; i < f->floor_count; ++i) {
+        f->floor_types[i] = get_bits(f, 16);
+        if (f->floor_types[i] > 1) return error(f, VORBIS_invalid_setup);
+        if (f->floor_types[i] == 0) {
+            Floor0 *g = &f->floor_config[i].floor0;
+            g->order = get_bits(f, 8);
+            g->rate = get_bits(f, 16);
+            g->bark_map_size = get_bits(f, 16);
+            g->amplitude_bits = get_bits(f, 6);
+            g->amplitude_offset = get_bits(f, 8);
+            g->number_of_books = get_bits(f, 4) + 1;
+            for (j = 0; j < g->number_of_books; ++j)
+                g->book_list[j] = get_bits(f, 8);
+            return error(f, VORBIS_feature_not_supported);
+        }
+        else {
+            stbv__floor_ordering p[31 * 8 + 2];
+            Floor1 *g = &f->floor_config[i].floor1;
+            int max_class = -1;
+            g->partitions = get_bits(f, 5);
+            for (j = 0; j < g->partitions; ++j) {
+                g->partition_class_list[j] = get_bits(f, 4);
+                if (g->partition_class_list[j] > max_class)
+                    max_class = g->partition_class_list[j];
+            }
+            for (j = 0; j <= max_class; ++j) {
+                g->class_dimensions[j] = get_bits(f, 3) + 1;
+                g->class_subclasses[j] = get_bits(f, 2);
+                if (g->class_subclasses[j]) {
+                    g->class_masterbooks[j] = get_bits(f, 8);
+                    if (g->class_masterbooks[j] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
+                }
+                for (k = 0; k < 1 << g->class_subclasses[j]; ++k) {
+                    g->subclass_books[j][k] = get_bits(f, 8) - 1;
+                    if (g->subclass_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
+                }
+            }
+            g->floor1_multiplier = get_bits(f, 2) + 1;
+            g->rangebits = get_bits(f, 4);
+            g->Xlist[0] = 0;
+            g->Xlist[1] = 1 << g->rangebits;
+            g->values = 2;
+            for (j = 0; j < g->partitions; ++j) {
+                int c = g->partition_class_list[j];
+                for (k = 0; k < g->class_dimensions[c]; ++k) {
+                    g->Xlist[g->values] = get_bits(f, g->rangebits);
+                    ++g->values;
+                }
+            }
+            // precompute the sorting
+            for (j = 0; j < g->values; ++j) {
+                p[j].x = g->Xlist[j];
+                p[j].id = j;
+            }
+            qsort(p, g->values, sizeof(p[0]), point_compare);
+            for (j = 0; j < g->values; ++j)
+                g->sorted_order[j] = (uint8)p[j].id;
+            // precompute the neighbors
+            for (j = 2; j < g->values; ++j) {
+                int low, hi;
+                neighbors(g->Xlist, j, &low, &hi);
+                g->neighbors[j][0] = low;
+                g->neighbors[j][1] = hi;
+            }
+
+            if (g->values > longest_floorlist)
+                longest_floorlist = g->values;
+        }
+    }
+
+    // Residue
+    f->residue_count = get_bits(f, 6) + 1;
+    f->residue_config = (Residue *)setup_malloc(f, f->residue_count * sizeof(f->residue_config[0]));
+    if (f->residue_config == NULL) return error(f, VORBIS_outofmem);
+    memset(f->residue_config, 0, f->residue_count * sizeof(f->residue_config[0]));
+    for (i = 0; i < f->residue_count; ++i) {
+        uint8 residue_cascade[64];
+        Residue *r = f->residue_config + i;
+        f->residue_types[i] = get_bits(f, 16);
+        if (f->residue_types[i] > 2) return error(f, VORBIS_invalid_setup);
+        r->begin = get_bits(f, 24);
+        r->end = get_bits(f, 24);
+        if (r->end < r->begin) return error(f, VORBIS_invalid_setup);
+        r->part_size = get_bits(f, 24) + 1;
+        r->classifications = get_bits(f, 6) + 1;
+        r->classbook = get_bits(f, 8);
+        if (r->classbook >= f->codebook_count) return error(f, VORBIS_invalid_setup);
+        for (j = 0; j < r->classifications; ++j) {
+            uint8 high_bits = 0;
+            uint8 low_bits = get_bits(f, 3);
+            if (get_bits(f, 1))
+                high_bits = get_bits(f, 5);
+            residue_cascade[j] = high_bits * 8 + low_bits;
+        }
+        r->residue_books = (short(*)[8]) setup_malloc(f, sizeof(r->residue_books[0]) * r->classifications);
+        if (r->residue_books == NULL) return error(f, VORBIS_outofmem);
+        for (j = 0; j < r->classifications; ++j) {
+            for (k = 0; k < 8; ++k) {
+                if (residue_cascade[j] & (1 << k)) {
+                    r->residue_books[j][k] = get_bits(f, 8);
+                    if (r->residue_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
+                }
+                else {
+                    r->residue_books[j][k] = -1;
+                }
+            }
+        }
+        // precompute the classifications[] array to avoid inner-loop mod/divide
+        // call it 'classdata' since we already have r->classifications
+        r->classdata = (uint8 **)setup_malloc(f, sizeof(*r->classdata) * f->codebooks[r->classbook].entries);
+        if (!r->classdata) return error(f, VORBIS_outofmem);
+        memset(r->classdata, 0, sizeof(*r->classdata) * f->codebooks[r->classbook].entries);
+        for (j = 0; j < f->codebooks[r->classbook].entries; ++j) {
+            int classwords = f->codebooks[r->classbook].dimensions;
+            int temp = j;
+            r->classdata[j] = (uint8 *)setup_malloc(f, sizeof(r->classdata[j][0]) * classwords);
+            if (r->classdata[j] == NULL) return error(f, VORBIS_outofmem);
+            for (k = classwords - 1; k >= 0; --k) {
+                r->classdata[j][k] = temp % r->classifications;
+                temp /= r->classifications;
+            }
+        }
+    }
+
+    f->mapping_count = get_bits(f, 6) + 1;
+    f->mapping = (Mapping *)setup_malloc(f, f->mapping_count * sizeof(*f->mapping));
+    if (f->mapping == NULL) return error(f, VORBIS_outofmem);
+    memset(f->mapping, 0, f->mapping_count * sizeof(*f->mapping));
+    for (i = 0; i < f->mapping_count; ++i) {
+        Mapping *m = f->mapping + i;
+        int mapping_type = get_bits(f, 16);
+        if (mapping_type != 0) return error(f, VORBIS_invalid_setup);
+        m->chan = (MappingChannel *)setup_malloc(f, f->channels * sizeof(*m->chan));
+        if (m->chan == NULL) return error(f, VORBIS_outofmem);
+        if (get_bits(f, 1))
+            m->submaps = get_bits(f, 4) + 1;
+        else
+            m->submaps = 1;
+        if (m->submaps > max_submaps)
+            max_submaps = m->submaps;
+        if (get_bits(f, 1)) {
+            m->coupling_steps = get_bits(f, 8) + 1;
+            for (k = 0; k < m->coupling_steps; ++k) {
+                m->chan[k].magnitude = get_bits(f, ilog(f->channels - 1));
+                m->chan[k].angle = get_bits(f, ilog(f->channels - 1));
+                if (m->chan[k].magnitude >= f->channels)        return error(f, VORBIS_invalid_setup);
+                if (m->chan[k].angle >= f->channels)        return error(f, VORBIS_invalid_setup);
+                if (m->chan[k].magnitude == m->chan[k].angle)   return error(f, VORBIS_invalid_setup);
+            }
+        }
+        else
+            m->coupling_steps = 0;
+
+        // reserved field
+        if (get_bits(f, 2)) return error(f, VORBIS_invalid_setup);
+        if (m->submaps > 1) {
+            for (j = 0; j < f->channels; ++j) {
+                m->chan[j].mux = get_bits(f, 4);
+                if (m->chan[j].mux >= m->submaps)                return error(f, VORBIS_invalid_setup);
+            }
+        }
+        else
+            // @SPECIFICATION: this case is missing from the spec
+            for (j = 0; j < f->channels; ++j)
+                m->chan[j].mux = 0;
+
+        for (j = 0; j < m->submaps; ++j) {
+            get_bits(f, 8); // discard
+            m->submap_floor[j] = get_bits(f, 8);
+            m->submap_residue[j] = get_bits(f, 8);
+            if (m->submap_floor[j] >= f->floor_count)      return error(f, VORBIS_invalid_setup);
+            if (m->submap_residue[j] >= f->residue_count)  return error(f, VORBIS_invalid_setup);
+        }
+    }
+
+    // Modes
+    f->mode_count = get_bits(f, 6) + 1;
+    for (i = 0; i < f->mode_count; ++i) {
+        Mode *m = f->mode_config + i;
+        m->blockflag = get_bits(f, 1);
+        m->windowtype = get_bits(f, 16);
+        m->transformtype = get_bits(f, 16);
+        m->mapping = get_bits(f, 8);
+        if (m->windowtype != 0)                 return error(f, VORBIS_invalid_setup);
+        if (m->transformtype != 0)              return error(f, VORBIS_invalid_setup);
+        if (m->mapping >= f->mapping_count)     return error(f, VORBIS_invalid_setup);
+    }
+
+    flush_packet(f);
+
+    f->previous_length = 0;
+
+    for (i = 0; i < f->channels; ++i) {
+        f->channel_buffers[i] = (float *)setup_malloc(f, sizeof(float) * f->blocksize_1);
+        f->previous_window[i] = (float *)setup_malloc(f, sizeof(float) * f->blocksize_1 / 2);
+        f->finalY[i] = (int16 *)setup_malloc(f, sizeof(int16) * longest_floorlist);
+        if (f->channel_buffers[i] == NULL || f->previous_window[i] == NULL || f->finalY[i] == NULL) return error(f, VORBIS_outofmem);
+        memset(f->channel_buffers[i], 0, sizeof(float) * f->blocksize_1);
+#ifdef STB_VORBIS_NO_DEFER_FLOOR
+        f->floor_buffers[i] = (float *)setup_malloc(f, sizeof(float) * f->blocksize_1 / 2);
+        if (f->floor_buffers[i] == NULL) return error(f, VORBIS_outofmem);
+#endif
+    }
+
+    if (!init_blocksize(f, 0, f->blocksize_0)) return FALSE;
+    if (!init_blocksize(f, 1, f->blocksize_1)) return FALSE;
+    f->blocksize[0] = f->blocksize_0;
+    f->blocksize[1] = f->blocksize_1;
+
+#ifdef STB_VORBIS_DIVIDE_TABLE
+    if (integer_divide_table[1][1] == 0)
+        for (i = 0; i < DIVTAB_NUMER; ++i)
+            for (j = 1; j < DIVTAB_DENOM; ++j)
+                integer_divide_table[i][j] = i / j;
+#endif
+
+    // compute how much temporary memory is needed
+
+    // 1.
+    {
+        uint32 imdct_mem = (f->blocksize_1 * sizeof(float) >> 1);
+        uint32 classify_mem;
+        int i, max_part_read = 0;
+        for (i = 0; i < f->residue_count; ++i) {
+            Residue *r = f->residue_config + i;
+            unsigned int actual_size = f->blocksize_1 / 2;
+            unsigned int limit_r_begin = r->begin < actual_size ? r->begin : actual_size;
+            unsigned int limit_r_end = r->end < actual_size ? r->end : actual_size;
+            int n_read = limit_r_end - limit_r_begin;
+            int part_read = n_read / r->part_size;
+            if (part_read > max_part_read)
+                max_part_read = part_read;
+        }
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+        classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(uint8 *));
+#else
+        classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(int *));
+#endif
+
+        // maximum reasonable partition size is f->blocksize_1
+
+        f->temp_memory_required = classify_mem;
+        if (imdct_mem > f->temp_memory_required)
+            f->temp_memory_required = imdct_mem;
+    }
+
+    f->first_decode = TRUE;
+
+    if (f->alloc.alloc_buffer) {
+        assert(f->temp_offset == f->alloc.alloc_buffer_length_in_bytes);
+        // check if there's enough temp memory so we don't error later
+        if (f->setup_offset + sizeof(*f) + f->temp_memory_required > (unsigned)f->temp_offset)
+            return error(f, VORBIS_outofmem);
+    }
+
+    f->first_audio_page_offset = stb_vorbis_get_file_offset(f);
+
+    return TRUE;
+}
+
+static void vorbis_deinit(stb_vorbis *p)
+{
+    int i, j;
+    if (p->residue_config) {
+        for (i = 0; i < p->residue_count; ++i) {
+            Residue *r = p->residue_config + i;
+            if (r->classdata) {
+                for (j = 0; j < p->codebooks[r->classbook].entries; ++j)
+                    setup_free(p, r->classdata[j]);
+                setup_free(p, r->classdata);
+            }
+            setup_free(p, r->residue_books);
+        }
+    }
+
+    if (p->codebooks) {
+        CHECK(p);
+        for (i = 0; i < p->codebook_count; ++i) {
+            Codebook *c = p->codebooks + i;
+            setup_free(p, c->codeword_lengths);
+            setup_free(p, c->multiplicands);
+            setup_free(p, c->codewords);
+            setup_free(p, c->sorted_codewords);
+            // c->sorted_values[-1] is the first entry in the array
+            setup_free(p, c->sorted_values ? c->sorted_values - 1 : NULL);
+        }
+        setup_free(p, p->codebooks);
+    }
+    setup_free(p, p->floor_config);
+    setup_free(p, p->residue_config);
+    if (p->mapping) {
+        for (i = 0; i < p->mapping_count; ++i)
+            setup_free(p, p->mapping[i].chan);
+        setup_free(p, p->mapping);
+    }
+    CHECK(p);
+    for (i = 0; i < p->channels && i < STB_VORBIS_MAX_CHANNELS; ++i) {
+        setup_free(p, p->channel_buffers[i]);
+        setup_free(p, p->previous_window[i]);
+#ifdef STB_VORBIS_NO_DEFER_FLOOR
+        setup_free(p, p->floor_buffers[i]);
+#endif
+        setup_free(p, p->finalY[i]);
+    }
+    for (i = 0; i < 2; ++i) {
+        setup_free(p, p->A[i]);
+        setup_free(p, p->B[i]);
+        setup_free(p, p->C[i]);
+        setup_free(p, p->window[i]);
+        setup_free(p, p->bit_reverse[i]);
+    }
+#ifndef STB_VORBIS_NO_STDIO
+    if (p->close_on_free) fclose(p->f);
+#endif
+}
+
+void stb_vorbis_close(stb_vorbis *p)
+{
+    if (p == NULL) return;
+    vorbis_deinit(p);
+    setup_free(p, p);
+}
+
+static void vorbis_init(stb_vorbis *p, const stb_vorbis_alloc *z)
+{
+    memset(p, 0, sizeof(*p)); // NULL out all malloc'd pointers to start
+    if (z) {
+        p->alloc = *z;
+        p->alloc.alloc_buffer_length_in_bytes = (p->alloc.alloc_buffer_length_in_bytes + 3) & ~3;
+        p->temp_offset = p->alloc.alloc_buffer_length_in_bytes;
+    }
+    p->eof = 0;
+    p->error = VORBIS__no_error;
+    p->stream = NULL;
+    p->codebooks = NULL;
+    p->page_crc_tests = -1;
+#ifndef STB_VORBIS_NO_STDIO
+    p->close_on_free = FALSE;
+    p->f = NULL;
+#endif
+}
+
+int stb_vorbis_get_sample_offset(stb_vorbis *f)
+{
+    if (f->current_loc_valid)
+        return f->current_loc;
+    else
+        return -1;
+}
+
+stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f)
+{
+    stb_vorbis_info d;
+    d.channels = f->channels;
+    d.sample_rate = f->sample_rate;
+    d.setup_memory_required = f->setup_memory_required;
+    d.setup_temp_memory_required = f->setup_temp_memory_required;
+    d.temp_memory_required = f->temp_memory_required;
+    d.max_frame_size = f->blocksize_1 >> 1;
+    return d;
+}
+
+int stb_vorbis_get_error(stb_vorbis *f)
+{
+    int e = f->error;
+    f->error = VORBIS__no_error;
+    return e;
+}
+
+static stb_vorbis * vorbis_alloc(stb_vorbis *f)
+{
+    stb_vorbis *p = (stb_vorbis *)setup_malloc(f, sizeof(*p));
+    return p;
+}
+
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+
+void stb_vorbis_flush_pushdata(stb_vorbis *f)
+{
+    f->previous_length = 0;
+    f->page_crc_tests = 0;
+    f->discard_samples_deferred = 0;
+    f->current_loc_valid = FALSE;
+    f->first_decode = FALSE;
+    f->samples_output = 0;
+    f->channel_buffer_start = 0;
+    f->channel_buffer_end = 0;
+}
+
+static int vorbis_search_for_page_pushdata(vorb *f, uint8 *data, int data_len)
+{
+    int i, n;
+    for (i = 0; i < f->page_crc_tests; ++i)
+        f->scan[i].bytes_done = 0;
+
+    // if we have room for more scans, search for them first, because
+    // they may cause us to stop early if their header is incomplete
+    if (f->page_crc_tests < STB_VORBIS_PUSHDATA_CRC_COUNT) {
+        if (data_len < 4) return 0;
+        data_len -= 3; // need to look for 4-byte sequence, so don't miss
+                       // one that straddles a boundary
+        for (i = 0; i < data_len; ++i) {
+            if (data[i] == 0x4f) {
+                if (0 == memcmp(data + i, ogg_page_header, 4)) {
+                    int j, len;
+                    uint32 crc;
+                    // make sure we have the whole page header
+                    if (i + 26 >= data_len || i + 27 + data[i + 26] >= data_len) {
+                        // only read up to this page start, so hopefully we'll
+                        // have the whole page header start next time
+                        data_len = i;
+                        break;
+                    }
+                    // ok, we have it all; compute the length of the page
+                    len = 27 + data[i + 26];
+                    for (j = 0; j < data[i + 26]; ++j)
+                        len += data[i + 27 + j];
+                    // scan everything up to the embedded crc (which we must 0)
+                    crc = 0;
+                    for (j = 0; j < 22; ++j)
+                        crc = crc32_update(crc, data[i + j]);
+                    // now process 4 0-bytes
+                    for (; j < 26; ++j)
+                        crc = crc32_update(crc, 0);
+                    // len is the total number of bytes we need to scan
+                    n = f->page_crc_tests++;
+                    f->scan[n].bytes_left = len - j;
+                    f->scan[n].crc_so_far = crc;
+                    f->scan[n].goal_crc = data[i + 22] + (data[i + 23] << 8) + (data[i + 24] << 16) + (data[i + 25] << 24);
+                    // if the last frame on a page is continued to the next, then
+                    // we can't recover the sample_loc immediately
+                    if (data[i + 27 + data[i + 26] - 1] == 255)
+                        f->scan[n].sample_loc = ~0;
+                    else
+                        f->scan[n].sample_loc = data[i + 6] + (data[i + 7] << 8) + (data[i + 8] << 16) + (data[i + 9] << 24);
+                    f->scan[n].bytes_done = i + j;
+                    if (f->page_crc_tests == STB_VORBIS_PUSHDATA_CRC_COUNT)
+                        break;
+                    // keep going if we still have room for more
+                }
+            }
+        }
+    }
+
+    for (i = 0; i < f->page_crc_tests;) {
+        uint32 crc;
+        int j;
+        int n = f->scan[i].bytes_done;
+        int m = f->scan[i].bytes_left;
+        if (m > data_len - n) m = data_len - n;
+        // m is the bytes to scan in the current chunk
+        crc = f->scan[i].crc_so_far;
+        for (j = 0; j < m; ++j)
+            crc = crc32_update(crc, data[n + j]);
+        f->scan[i].bytes_left -= m;
+        f->scan[i].crc_so_far = crc;
+        if (f->scan[i].bytes_left == 0) {
+            // does it match?
+            if (f->scan[i].crc_so_far == f->scan[i].goal_crc) {
+                // Houston, we have page
+                data_len = n + m; // consumption amount is wherever that scan ended
+                f->page_crc_tests = -1; // drop out of page scan mode
+                f->previous_length = 0; // decode-but-don't-output one frame
+                f->next_seg = -1;       // start a new page
+                f->current_loc = f->scan[i].sample_loc; // set the current sample location
+                                                        // to the amount we'd have decoded had we decoded this page
+                f->current_loc_valid = f->current_loc != ~0U;
+                return data_len;
+            }
+            // delete entry
+            f->scan[i] = f->scan[--f->page_crc_tests];
+        }
+        else {
+            ++i;
+        }
+    }
+
+    return data_len;
+}
+
+// return value: number of bytes we used
+int stb_vorbis_decode_frame_pushdata(
+    stb_vorbis *f,                   // the file we're decoding
+    const uint8 *data, int data_len, // the memory available for decoding
+    int *channels,                   // place to write number of float * buffers
+    float ***output,                 // place to write float ** array of float * buffers
+    int *samples                     // place to write number of output samples
+)
+{
+    int i;
+    int len, right, left;
+
+    if (!IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
+
+    if (f->page_crc_tests >= 0) {
+        *samples = 0;
+        return vorbis_search_for_page_pushdata(f, (uint8 *)data, data_len);
+    }
+
+    f->stream = (uint8 *)data;
+    f->stream_end = (uint8 *)data + data_len;
+    f->error = VORBIS__no_error;
+
+    // check that we have the entire packet in memory
+    if (!is_whole_packet_present(f, FALSE)) {
+        *samples = 0;
+        return 0;
+    }
+
+    if (!vorbis_decode_packet(f, &len, &left, &right)) {
+        // save the actual error we encountered
+        enum STBVorbisError error = f->error;
+        if (error == VORBIS_bad_packet_type) {
+            // flush and resynch
+            f->error = VORBIS__no_error;
+            while (get8_packet(f) != EOP)
+                if (f->eof) break;
+            *samples = 0;
+            return (int)(f->stream - data);
+        }
+        if (error == VORBIS_continued_packet_flag_invalid) {
+            if (f->previous_length == 0) {
+                // we may be resynching, in which case it's ok to hit one
+                // of these; just discard the packet
+                f->error = VORBIS__no_error;
+                while (get8_packet(f) != EOP)
+                    if (f->eof) break;
+                *samples = 0;
+                return (int)(f->stream - data);
+            }
+        }
+        // if we get an error while parsing, what to do?
+        // well, it DEFINITELY won't work to continue from where we are!
+        stb_vorbis_flush_pushdata(f);
+        // restore the error that actually made us bail
+        f->error = error;
+        *samples = 0;
+        return 1;
+    }
+
+    // success!
+    len = vorbis_finish_frame(f, len, left, right);
+    for (i = 0; i < f->channels; ++i)
+        f->outputs[i] = f->channel_buffers[i] + left;
+
+    if (channels) *channels = f->channels;
+    *samples = len;
+    *output = f->outputs;
+    return (int)(f->stream - data);
+}
+
+stb_vorbis *stb_vorbis_open_pushdata(
+    const unsigned char *data, int data_len, // the memory available for decoding
+    int *data_used,              // only defined if result is not NULL
+    int *error, const stb_vorbis_alloc *alloc)
+{
+    stb_vorbis *f, p;
+    vorbis_init(&p, alloc);
+    p.stream = (uint8 *)data;
+    p.stream_end = (uint8 *)data + data_len;
+    p.push_mode = TRUE;
+    if (!start_decoder(&p)) {
+        if (p.eof)
+            *error = VORBIS_need_more_data;
+        else
+            *error = p.error;
+        return NULL;
+    }
+    f = vorbis_alloc(&p);
+    if (f) {
+        *f = p;
+        *data_used = (int)(f->stream - data);
+        *error = 0;
+        return f;
+    }
+    else {
+        vorbis_deinit(&p);
+        return NULL;
+    }
+}
+#endif // STB_VORBIS_NO_PUSHDATA_API
+
+unsigned int stb_vorbis_get_file_offset(stb_vorbis *f)
+{
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+    if (f->push_mode) return 0;
+#endif
+    if (USE_MEMORY(f)) return (unsigned int)(f->stream - f->stream_start);
+#ifndef STB_VORBIS_NO_STDIO
+    return (unsigned int)(ftell(f->f) - f->f_start);
+#endif
+}
+
+#ifndef STB_VORBIS_NO_PULLDATA_API
+//
+// DATA-PULLING API
+//
+
+static uint32 vorbis_find_page(stb_vorbis *f, uint32 *end, uint32 *last)
+{
+    for (;;) {
+        int n;
+        if (f->eof) return 0;
+        n = get8(f);
+        if (n == 0x4f) { // page header candidate
+            unsigned int retry_loc = stb_vorbis_get_file_offset(f);
+            int i;
+            // check if we're off the end of a file_section stream
+            if (retry_loc - 25 > f->stream_len)
+                return 0;
+            // check the rest of the header
+            for (i = 1; i < 4; ++i)
+                if (get8(f) != ogg_page_header[i])
+                    break;
+            if (f->eof) return 0;
+            if (i == 4) {
+                uint8 header[27];
+                uint32 i, crc, goal, len;
+                for (i = 0; i < 4; ++i)
+                    header[i] = ogg_page_header[i];
+                for (; i < 27; ++i)
+                    header[i] = get8(f);
+                if (f->eof) return 0;
+                if (header[4] != 0) goto invalid;
+                goal = header[22] + (header[23] << 8) + (header[24] << 16) + (header[25] << 24);
+                for (i = 22; i < 26; ++i)
+                    header[i] = 0;
+                crc = 0;
+                for (i = 0; i < 27; ++i)
+                    crc = crc32_update(crc, header[i]);
+                len = 0;
+                for (i = 0; i < header[26]; ++i) {
+                    int s = get8(f);
+                    crc = crc32_update(crc, s);
+                    len += s;
+                }
+                if (len && f->eof) return 0;
+                for (i = 0; i < len; ++i)
+                    crc = crc32_update(crc, get8(f));
+                // finished parsing probable page
+                if (crc == goal) {
+                    // we could now check that it's either got the last
+                    // page flag set, OR it's followed by the capture
+                    // pattern, but I guess TECHNICALLY you could have
+                    // a file with garbage between each ogg page and recover
+                    // from it automatically? So even though that paranoia
+                    // might decrease the chance of an invalid decode by
+                    // another 2^32, not worth it since it would hose those
+                    // invalid-but-useful files?
+                    if (end)
+                        *end = stb_vorbis_get_file_offset(f);
+                    if (last) {
+                        if (header[5] & 0x04)
+                            *last = 1;
+                        else
+                            *last = 0;
+                    }
+                    set_file_offset(f, retry_loc - 1);
+                    return 1;
+                }
+            }
+        invalid:
+            // not a valid page, so rewind and look for next one
+            set_file_offset(f, retry_loc);
+        }
+    }
+}
+
+
+#define SAMPLE_unknown  0xffffffff
+
+// seeking is implemented with a binary search, which narrows down the range to
+// 64K, before using a linear search (because finding the synchronization
+// pattern can be expensive, and the chance we'd find the end page again is
+// relatively high for small ranges)
+//
+// two initial interpolation-style probes are used at the start of the search
+// to try to bound either side of the binary search sensibly, while still
+// working in O(log n) time if they fail.
+
+static int get_seek_page_info(stb_vorbis *f, ProbedPage *z)
+{
+    uint8 header[27], lacing[255];
+    int i, len;
+
+    // record where the page starts
+    z->page_start = stb_vorbis_get_file_offset(f);
+
+    // parse the header
+    getn(f, header, 27);
+    if (header[0] != 'O' || header[1] != 'g' || header[2] != 'g' || header[3] != 'S')
+        return 0;
+    getn(f, lacing, header[26]);
+
+    // determine the length of the payload
+    len = 0;
+    for (i = 0; i < header[26]; ++i)
+        len += lacing[i];
+
+    // this implies where the page ends
+    z->page_end = z->page_start + 27 + header[26] + len;
+
+    // read the last-decoded sample out of the data
+    z->last_decoded_sample = header[6] + (header[7] << 8) + (header[8] << 16) + (header[9] << 24);
+
+    // restore file state to where we were
+    set_file_offset(f, z->page_start);
+    return 1;
+}
+
+// rarely used function to seek back to the preceeding page while finding the
+// start of a packet
+static int go_to_page_before(stb_vorbis *f, unsigned int limit_offset)
+{
+    unsigned int previous_safe, end;
+
+    // now we want to seek back 64K from the limit
+    if (limit_offset >= 65536 && limit_offset - 65536 >= f->first_audio_page_offset)
+        previous_safe = limit_offset - 65536;
+    else
+        previous_safe = f->first_audio_page_offset;
+
+    set_file_offset(f, previous_safe);
+
+    while (vorbis_find_page(f, &end, NULL)) {
+        if (end >= limit_offset && stb_vorbis_get_file_offset(f) < limit_offset)
+            return 1;
+        set_file_offset(f, end);
+    }
+
+    return 0;
+}
+
+// implements the search logic for finding a page and starting decoding. if
+// the function succeeds, current_loc_valid will be true and current_loc will
+// be less than or equal to the provided sample number (the closer the
+// better).
+static int seek_to_sample_coarse(stb_vorbis *f, uint32 sample_number)
+{
+    ProbedPage left, right, mid;
+    int i, start_seg_with_known_loc, end_pos, page_start;
+    uint32 delta, stream_length, padding;
+    double offset, bytes_per_sample;
+    int probe = 0;
+
+    // find the last page and validate the target sample
+    stream_length = stb_vorbis_stream_length_in_samples(f);
+    if (stream_length == 0)            return error(f, VORBIS_seek_without_length);
+    if (sample_number > stream_length) return error(f, VORBIS_seek_invalid);
+
+    // this is the maximum difference between the window-center (which is the
+    // actual granule position value), and the right-start (which the spec
+    // indicates should be the granule position (give or take one)).
+    padding = ((f->blocksize_1 - f->blocksize_0) >> 2);
+    if (sample_number < padding)
+        sample_number = 0;
+    else
+        sample_number -= padding;
+
+    left = f->p_first;
+    while (left.last_decoded_sample == ~0U) {
+        // (untested) the first page does not have a 'last_decoded_sample'
+        set_file_offset(f, left.page_end);
+        if (!get_seek_page_info(f, &left)) goto error;
+    }
+
+    right = f->p_last;
+    assert(right.last_decoded_sample != ~0U);
+
+    // starting from the start is handled differently
+    if (sample_number <= left.last_decoded_sample) {
+        if (stb_vorbis_seek_start(f))
+            return 1;
+        return 0;
+    }
+
+    while (left.page_end != right.page_start) {
+        assert(left.page_end < right.page_start);
+        // search range in bytes
+        delta = right.page_start - left.page_end;
+        if (delta <= 65536) {
+            // there's only 64K left to search - handle it linearly
+            set_file_offset(f, left.page_end);
+        }
+        else {
+            if (probe < 2) {
+                if (probe == 0) {
+                    // first probe (interpolate)
+                    double data_bytes = right.page_end - left.page_start;
+                    bytes_per_sample = data_bytes / right.last_decoded_sample;
+                    offset = left.page_start + bytes_per_sample * (sample_number - left.last_decoded_sample);
+                }
+                else {
+                    // second probe (try to bound the other side)
+                    double error = ((double)sample_number - mid.last_decoded_sample) * bytes_per_sample;
+                    if (error >= 0 && error < 8000) error = 8000;
+                    if (error <  0 && error > -8000) error = -8000;
+                    offset += error * 2;
+                }
+
+                // ensure the offset is valid
+                if (offset < left.page_end)
+                    offset = left.page_end;
+                if (offset > right.page_start - 65536)
+                    offset = right.page_start - 65536;
+
+                set_file_offset(f, (unsigned int)offset);
+            }
+            else {
+                // binary search for large ranges (offset by 32K to ensure
+                // we don't hit the right page)
+                set_file_offset(f, left.page_end + (delta / 2) - 32768);
+            }
+
+            if (!vorbis_find_page(f, NULL, NULL)) goto error;
+        }
+
+        for (;;) {
+            if (!get_seek_page_info(f, &mid)) goto error;
+            if (mid.last_decoded_sample != ~0U) break;
+            // (untested) no frames end on this page
+            set_file_offset(f, mid.page_end);
+            assert(mid.page_start < right.page_start);
+        }
+
+        // if we've just found the last page again then we're in a tricky file,
+        // and we're close enough.
+        if (mid.page_start == right.page_start)
+            break;
+
+        if (sample_number < mid.last_decoded_sample)
+            right = mid;
+        else
+            left = mid;
+
+        ++probe;
+    }
+
+    // seek back to start of the last packet
+    page_start = left.page_start;
+    set_file_offset(f, page_start);
+    if (!start_page(f)) return error(f, VORBIS_seek_failed);
+    end_pos = f->end_seg_with_known_loc;
+    assert(end_pos >= 0);
+
+    for (;;) {
+        for (i = end_pos; i > 0; --i)
+            if (f->segments[i - 1] != 255)
+                break;
+
+        start_seg_with_known_loc = i;
+
+        if (start_seg_with_known_loc > 0 || !(f->page_flag & PAGEFLAG_continued_packet))
+            break;
+
+        // (untested) the final packet begins on an earlier page
+        if (!go_to_page_before(f, page_start))
+            goto error;
+
+        page_start = stb_vorbis_get_file_offset(f);
+        if (!start_page(f)) goto error;
+        end_pos = f->segment_count - 1;
+    }
+
+    // prepare to start decoding
+    f->current_loc_valid = FALSE;
+    f->last_seg = FALSE;
+    f->valid_bits = 0;
+    f->packet_bytes = 0;
+    f->bytes_in_seg = 0;
+    f->previous_length = 0;
+    f->next_seg = start_seg_with_known_loc;
+
+    for (i = 0; i < start_seg_with_known_loc; i++)
+        skip(f, f->segments[i]);
+
+    // start decoding (optimizable - this frame is generally discarded)
+    if (!vorbis_pump_first_frame(f))
+        return 0;
+    if (f->current_loc > sample_number)
+        return error(f, VORBIS_seek_failed);
+    return 1;
+
+error:
+    // try to restore the file to a valid state
+    stb_vorbis_seek_start(f);
+    return error(f, VORBIS_seek_failed);
+}
+
+// the same as vorbis_decode_initial, but without advancing
+static int peek_decode_initial(vorb *f, int *p_left_start, int *p_left_end, int *p_right_start, int *p_right_end, int *mode)
+{
+    int bits_read, bytes_read;
+
+    if (!vorbis_decode_initial(f, p_left_start, p_left_end, p_right_start, p_right_end, mode))
+        return 0;
+
+    // either 1 or 2 bytes were read, figure out which so we can rewind
+    bits_read = 1 + ilog(f->mode_count - 1);
+    if (f->mode_config[*mode].blockflag)
+        bits_read += 2;
+    bytes_read = (bits_read + 7) / 8;
+
+    f->bytes_in_seg += bytes_read;
+    f->packet_bytes -= bytes_read;
+    skip(f, -bytes_read);
+    if (f->next_seg == -1)
+        f->next_seg = f->segment_count - 1;
+    else
+        f->next_seg--;
+    f->valid_bits = 0;
+
+    return 1;
+}
+
+int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number)
+{
+    uint32 max_frame_samples;
+
+    if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
+
+    // fast page-level search
+    if (!seek_to_sample_coarse(f, sample_number))
+        return 0;
+
+    assert(f->current_loc_valid);
+    assert(f->current_loc <= sample_number);
+
+    // linear search for the relevant packet
+    max_frame_samples = (f->blocksize_1 * 3 - f->blocksize_0) >> 2;
+    while (f->current_loc < sample_number) {
+        int left_start, left_end, right_start, right_end, mode, frame_samples;
+        if (!peek_decode_initial(f, &left_start, &left_end, &right_start, &right_end, &mode))
+            return error(f, VORBIS_seek_failed);
+        // calculate the number of samples returned by the next frame
+        frame_samples = right_start - left_start;
+        if (f->current_loc + frame_samples > sample_number) {
+            return 1; // the next frame will contain the sample
+        }
+        else if (f->current_loc + frame_samples + max_frame_samples > sample_number) {
+            // there's a chance the frame after this could contain the sample
+            vorbis_pump_first_frame(f);
+        }
+        else {
+            // this frame is too early to be relevant
+            f->current_loc += frame_samples;
+            f->previous_length = 0;
+            maybe_start_packet(f);
+            flush_packet(f);
+        }
+    }
+    // the next frame will start with the sample
+    assert(f->current_loc == sample_number);
+    return 1;
+}
+
+int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number)
+{
+    if (!stb_vorbis_seek_frame(f, sample_number))
+        return 0;
+
+    if (sample_number != f->current_loc) {
+        int n;
+        uint32 frame_start = f->current_loc;
+        stb_vorbis_get_frame_float(f, &n, NULL);
+        assert(sample_number > frame_start);
+        assert(f->channel_buffer_start + (int)(sample_number - frame_start) <= f->channel_buffer_end);
+        f->channel_buffer_start += (sample_number - frame_start);
+    }
+
+    return 1;
+}
+
+int stb_vorbis_seek_start(stb_vorbis *f)
+{
+    if (IS_PUSH_MODE(f)) { return error(f, VORBIS_invalid_api_mixing); }
+    set_file_offset(f, f->first_audio_page_offset);
+    f->previous_length = 0;
+    f->first_decode = TRUE;
+    f->next_seg = -1;
+    return vorbis_pump_first_frame(f);
+}
+
+unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f)
+{
+    unsigned int restore_offset, previous_safe;
+    unsigned int end, last_page_loc;
+
+    if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
+    if (!f->total_samples) {
+        unsigned int last;
+        uint32 lo, hi;
+        char header[6];
+
+        // first, store the current decode position so we can restore it
+        restore_offset = stb_vorbis_get_file_offset(f);
+
+        // now we want to seek back 64K from the end (the last page must
+        // be at most a little less than 64K, but let's allow a little slop)
+        if (f->stream_len >= 65536 && f->stream_len - 65536 >= f->first_audio_page_offset)
+            previous_safe = f->stream_len - 65536;
+        else
+            previous_safe = f->first_audio_page_offset;
+
+        set_file_offset(f, previous_safe);
+        // previous_safe is now our candidate 'earliest known place that seeking
+        // to will lead to the final page'
+
+        if (!vorbis_find_page(f, &end, &last)) {
+            // if we can't find a page, we're hosed!
+            f->error = VORBIS_cant_find_last_page;
+            f->total_samples = 0xffffffff;
+            goto done;
+        }
+
+        // check if there are more pages
+        last_page_loc = stb_vorbis_get_file_offset(f);
+
+        // stop when the last_page flag is set, not when we reach eof;
+        // this allows us to stop short of a 'file_section' end without
+        // explicitly checking the length of the section
+        while (!last) {
+            set_file_offset(f, end);
+            if (!vorbis_find_page(f, &end, &last)) {
+                // the last page we found didn't have the 'last page' flag
+                // set. whoops!
+                break;
+            }
+            previous_safe = last_page_loc + 1;
+            last_page_loc = stb_vorbis_get_file_offset(f);
+        }
+
+        set_file_offset(f, last_page_loc);
+
+        // parse the header
+        getn(f, (unsigned char *)header, 6);
+        // extract the absolute granule position
+        lo = get32(f);
+        hi = get32(f);
+        if (lo == 0xffffffff && hi == 0xffffffff) {
+            f->error = VORBIS_cant_find_last_page;
+            f->total_samples = SAMPLE_unknown;
+            goto done;
+        }
+        if (hi)
+            lo = 0xfffffffe; // saturate
+        f->total_samples = lo;
+
+        f->p_last.page_start = last_page_loc;
+        f->p_last.page_end = end;
+        f->p_last.last_decoded_sample = lo;
+
+    done:
+        set_file_offset(f, restore_offset);
+    }
+    return f->total_samples == SAMPLE_unknown ? 0 : f->total_samples;
+}
+
+float stb_vorbis_stream_length_in_seconds(stb_vorbis *f)
+{
+    return stb_vorbis_stream_length_in_samples(f) / (float)f->sample_rate;
+}
+
+
+
+int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output)
+{
+    int len, right, left, i;
+    if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
+
+    if (!vorbis_decode_packet(f, &len, &left, &right)) {
+        f->channel_buffer_start = f->channel_buffer_end = 0;
+        return 0;
+    }
+
+    len = vorbis_finish_frame(f, len, left, right);
+    for (i = 0; i < f->channels; ++i)
+        f->outputs[i] = f->channel_buffers[i] + left;
+
+    f->channel_buffer_start = left;
+    f->channel_buffer_end = left + len;
+
+    if (channels) *channels = f->channels;
+    if (output)   *output = f->outputs;
+    return len;
+}
+
+#ifndef STB_VORBIS_NO_STDIO
+
+stb_vorbis * stb_vorbis_open_file_section(FILE *file, int close_on_free, int *error, const stb_vorbis_alloc *alloc, unsigned int length)
+{
+    stb_vorbis *f, p;
+    vorbis_init(&p, alloc);
+    p.f = file;
+    p.f_start = (uint32)ftell(file);
+    p.stream_len = length;
+    p.close_on_free = close_on_free;
+    if (start_decoder(&p)) {
+        f = vorbis_alloc(&p);
+        if (f) {
+            *f = p;
+            vorbis_pump_first_frame(f);
+            return f;
+        }
+    }
+    if (error) *error = p.error;
+    vorbis_deinit(&p);
+    return NULL;
+}
+
+stb_vorbis * stb_vorbis_open_file(FILE *file, int close_on_free, int *error, const stb_vorbis_alloc *alloc)
+{
+    unsigned int len, start;
+    start = (unsigned int)ftell(file);
+    fseek(file, 0, SEEK_END);
+    len = (unsigned int)(ftell(file) - start);
+    fseek(file, start, SEEK_SET);
+    return stb_vorbis_open_file_section(file, close_on_free, error, alloc, len);
+}
+
+stb_vorbis * stb_vorbis_open_filename(const char *filename, int *error, const stb_vorbis_alloc *alloc)
+{
+    FILE *f = fopen(filename, "rb");
+    if (f)
+        return stb_vorbis_open_file(f, TRUE, error, alloc);
+    if (error) *error = VORBIS_file_open_failure;
+    return NULL;
+}
+#endif // STB_VORBIS_NO_STDIO
+
+stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len, int *error, const stb_vorbis_alloc *alloc)
+{
+    stb_vorbis *f, p;
+    if (data == NULL) return NULL;
+    vorbis_init(&p, alloc);
+    p.stream = (uint8 *)data;
+    p.stream_end = (uint8 *)data + len;
+    p.stream_start = (uint8 *)p.stream;
+    p.stream_len = len;
+    p.push_mode = FALSE;
+    if (start_decoder(&p)) {
+        f = vorbis_alloc(&p);
+        if (f) {
+            *f = p;
+            vorbis_pump_first_frame(f);
+            if (error) *error = VORBIS__no_error;
+            return f;
+        }
+    }
+    if (error) *error = p.error;
+    vorbis_deinit(&p);
+    return NULL;
+}
+
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+#define PLAYBACK_MONO     1
+#define PLAYBACK_LEFT     2
+#define PLAYBACK_RIGHT    4
+
+#define L  (PLAYBACK_LEFT  | PLAYBACK_MONO)
+#define C  (PLAYBACK_LEFT  | PLAYBACK_RIGHT | PLAYBACK_MONO)
+#define R  (PLAYBACK_RIGHT | PLAYBACK_MONO)
+
+static int8 channel_position[7][6] =
+{
+    { 0 },
+    { C },
+    { L, R },
+    { L, C, R },
+    { L, R, L, R },
+    { L, C, R, L, R },
+    { L, C, R, L, R, C },
+};
+
+
+#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT
+typedef union {
+    float f;
+    int i;
+} float_conv;
+typedef char stb_vorbis_float_size_test[sizeof(float) == 4 && sizeof(int) == 4];
+#define FASTDEF(x) float_conv x
+// add (1<<23) to convert to int, then divide by 2^SHIFT, then add 0.5/2^SHIFT to round
+#define MAGIC(SHIFT) (1.5f * (1 << (23-SHIFT)) + 0.5f/(1 << SHIFT))
+#define ADDEND(SHIFT) (((150-SHIFT) << 23) + (1 << 22))
+#define FAST_SCALED_FLOAT_TO_INT(temp,x,s) (temp.f = (x) + MAGIC(s), temp.i - ADDEND(s))
+#define check_endianness()  
+#else
+#define FAST_SCALED_FLOAT_TO_INT(temp,x,s) ((int) ((x) * (1 << (s))))
+#define check_endianness()
+#define FASTDEF(x)
+#endif
+
+static void copy_samples(short *dest, float *src, int len)
+{
+    int i;
+    check_endianness();
+    for (i = 0; i < len; ++i) {
+        FASTDEF(temp);
+        int v = FAST_SCALED_FLOAT_TO_INT(temp, src[i], 15);
+        if ((unsigned int)(v + 32768) > 65535)
+            v = v < 0 ? -32768 : 32767;
+        dest[i] = v;
+    }
+}
+
+static void compute_samples(int mask, short *output, int num_c, float **data, int d_offset, int len)
+{
+#define BUFFER_SIZE  32
+    float buffer[BUFFER_SIZE];
+    int i, j, o, n = BUFFER_SIZE;
+    check_endianness();
+    for (o = 0; o < len; o += BUFFER_SIZE) {
+        memset(buffer, 0, sizeof(buffer));
+        if (o + n > len) n = len - o;
+        for (j = 0; j < num_c; ++j) {
+            if (channel_position[num_c][j] & mask) {
+                for (i = 0; i < n; ++i)
+                    buffer[i] += data[j][d_offset + o + i];
+            }
+        }
+        for (i = 0; i < n; ++i) {
+            FASTDEF(temp);
+            int v = FAST_SCALED_FLOAT_TO_INT(temp, buffer[i], 15);
+            if ((unsigned int)(v + 32768) > 65535)
+                v = v < 0 ? -32768 : 32767;
+            output[o + i] = v;
+        }
+    }
+}
+
+static void compute_stereo_samples(short *output, int num_c, float **data, int d_offset, int len)
+{
+#define BUFFER_SIZE  32
+    float buffer[BUFFER_SIZE];
+    int i, j, o, n = BUFFER_SIZE >> 1;
+    // o is the offset in the source data
+    check_endianness();
+    for (o = 0; o < len; o += BUFFER_SIZE >> 1) {
+        // o2 is the offset in the output data
+        int o2 = o << 1;
+        memset(buffer, 0, sizeof(buffer));
+        if (o + n > len) n = len - o;
+        for (j = 0; j < num_c; ++j) {
+            int m = channel_position[num_c][j] & (PLAYBACK_LEFT | PLAYBACK_RIGHT);
+            if (m == (PLAYBACK_LEFT | PLAYBACK_RIGHT)) {
+                for (i = 0; i < n; ++i) {
+                    buffer[i * 2 + 0] += data[j][d_offset + o + i];
+                    buffer[i * 2 + 1] += data[j][d_offset + o + i];
+                }
+            }
+            else if (m == PLAYBACK_LEFT) {
+                for (i = 0; i < n; ++i) {
+                    buffer[i * 2 + 0] += data[j][d_offset + o + i];
+                }
+            }
+            else if (m == PLAYBACK_RIGHT) {
+                for (i = 0; i < n; ++i) {
+                    buffer[i * 2 + 1] += data[j][d_offset + o + i];
+                }
+            }
+        }
+        for (i = 0; i < (n << 1); ++i) {
+            FASTDEF(temp);
+            int v = FAST_SCALED_FLOAT_TO_INT(temp, buffer[i], 15);
+            if ((unsigned int)(v + 32768) > 65535)
+                v = v < 0 ? -32768 : 32767;
+            output[o2 + i] = v;
+        }
+    }
+}
+
+static void convert_samples_short(int buf_c, short **buffer, int b_offset, int data_c, float **data, int d_offset, int samples)
+{
+    int i;
+    if (buf_c != data_c && buf_c <= 2 && data_c <= 6) {
+        static int channel_selector[3][2] = { { 0 },{ PLAYBACK_MONO },{ PLAYBACK_LEFT, PLAYBACK_RIGHT } };
+        for (i = 0; i < buf_c; ++i)
+            compute_samples(channel_selector[buf_c][i], buffer[i] + b_offset, data_c, data, d_offset, samples);
+    }
+    else {
+        int limit = buf_c < data_c ? buf_c : data_c;
+        for (i = 0; i < limit; ++i)
+            copy_samples(buffer[i] + b_offset, data[i] + d_offset, samples);
+        for (; i < buf_c; ++i)
+            memset(buffer[i] + b_offset, 0, sizeof(short) * samples);
+    }
+}
+
+int stb_vorbis_get_frame_short(stb_vorbis *f, int num_c, short **buffer, int num_samples)
+{
+    float **output;
+    int len = stb_vorbis_get_frame_float(f, NULL, &output);
+    if (len > num_samples) len = num_samples;
+    if (len)
+        convert_samples_short(num_c, buffer, 0, f->channels, output, 0, len);
+    return len;
+}
+
+static void convert_channels_short_interleaved(int buf_c, short *buffer, int data_c, float **data, int d_offset, int len)
+{
+    int i;
+    check_endianness();
+    if (buf_c != data_c && buf_c <= 2 && data_c <= 6) {
+        assert(buf_c == 2);
+        for (i = 0; i < buf_c; ++i)
+            compute_stereo_samples(buffer, data_c, data, d_offset, len);
+    }
+    else {
+        int limit = buf_c < data_c ? buf_c : data_c;
+        int j;
+        for (j = 0; j < len; ++j) {
+            for (i = 0; i < limit; ++i) {
+                FASTDEF(temp);
+                float f = data[i][d_offset + j];
+                int v = FAST_SCALED_FLOAT_TO_INT(temp, f, 15);//data[i][d_offset+j],15);
+                if ((unsigned int)(v + 32768) > 65535)
+                    v = v < 0 ? -32768 : 32767;
+                *buffer++ = v;
+            }
+            for (; i < buf_c; ++i)
+                *buffer++ = 0;
+        }
+    }
+}
+
+int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts)
+{
+    float **output;
+    int len;
+    if (num_c == 1) return stb_vorbis_get_frame_short(f, num_c, &buffer, num_shorts);
+    len = stb_vorbis_get_frame_float(f, NULL, &output);
+    if (len) {
+        if (len*num_c > num_shorts) len = num_shorts / num_c;
+        convert_channels_short_interleaved(num_c, buffer, f->channels, output, 0, len);
+    }
+    return len;
+}
+
+int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts)
+{
+    float **outputs;
+    int len = num_shorts / channels;
+    int n = 0;
+    int z = f->channels;
+    if (z > channels) z = channels;
+    while (n < len) {
+        int k = f->channel_buffer_end - f->channel_buffer_start;
+        if (n + k >= len) k = len - n;
+        if (k)
+            convert_channels_short_interleaved(channels, buffer, f->channels, f->channel_buffers, f->channel_buffer_start, k);
+        buffer += k*channels;
+        n += k;
+        f->channel_buffer_start += k;
+        if (n == len) break;
+        if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break;
+    }
+    return n;
+}
+
+int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int len)
+{
+    float **outputs;
+    int n = 0;
+    int z = f->channels;
+    if (z > channels) z = channels;
+    while (n < len) {
+        int k = f->channel_buffer_end - f->channel_buffer_start;
+        if (n + k >= len) k = len - n;
+        if (k)
+            convert_samples_short(channels, buffer, n, f->channels, f->channel_buffers, f->channel_buffer_start, k);
+        n += k;
+        f->channel_buffer_start += k;
+        if (n == len) break;
+        if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break;
+    }
+    return n;
+}
+
+#ifndef STB_VORBIS_NO_STDIO
+int stb_vorbis_decode_filename(const char *filename, int *channels, int *sample_rate, short **output)
+{
+    int data_len, offset, total, limit, error;
+    short *data;
+    stb_vorbis *v = stb_vorbis_open_filename(filename, &error, NULL);
+    if (v == NULL) return -1;
+    limit = v->channels * 4096;
+    *channels = v->channels;
+    if (sample_rate)
+        *sample_rate = v->sample_rate;
+    offset = data_len = 0;
+    total = limit;
+    data = (short *)malloc(total * sizeof(*data));
+    if (data == NULL) {
+        stb_vorbis_close(v);
+        return -2;
+    }
+    for (;;) {
+        int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data + offset, total - offset);
+        if (n == 0) break;
+        data_len += n;
+        offset += n * v->channels;
+        if (offset + limit > total) {
+            short *data2;
+            total *= 2;
+            data2 = (short *)realloc(data, total * sizeof(*data));
+            if (data2 == NULL) {
+                free(data);
+                stb_vorbis_close(v);
+                return -2;
+            }
+            data = data2;
+        }
+    }
+    *output = data;
+    stb_vorbis_close(v);
+    return data_len;
+}
+#endif // NO_STDIO
+
+int stb_vorbis_decode_memory(const uint8 *mem, int len, int *channels, int *sample_rate, short **output)
+{
+    int data_len, offset, total, limit, error;
+    short *data;
+    stb_vorbis *v = stb_vorbis_open_memory(mem, len, &error, NULL);
+    if (v == NULL) return -1;
+    limit = v->channels * 4096;
+    *channels = v->channels;
+    if (sample_rate)
+        *sample_rate = v->sample_rate;
+    offset = data_len = 0;
+    total = limit;
+    data = (short *)malloc(total * sizeof(*data));
+    if (data == NULL) {
+        stb_vorbis_close(v);
+        return -2;
+    }
+    for (;;) {
+        int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data + offset, total - offset);
+        if (n == 0) break;
+        data_len += n;
+        offset += n * v->channels;
+        if (offset + limit > total) {
+            short *data2;
+            total *= 2;
+            data2 = (short *)realloc(data, total * sizeof(*data));
+            if (data2 == NULL) {
+                free(data);
+                stb_vorbis_close(v);
+                return -2;
+            }
+            data = data2;
+        }
+    }
+    *output = data;
+    stb_vorbis_close(v);
+    return data_len;
+}
+#endif // STB_VORBIS_NO_INTEGER_CONVERSION
+
+int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats)
+{
+    float **outputs;
+    int len = num_floats / channels;
+    int n = 0;
+    int z = f->channels;
+    if (z > channels) z = channels;
+    while (n < len) {
+        int i, j;
+        int k = f->channel_buffer_end - f->channel_buffer_start;
+        if (n + k >= len) k = len - n;
+        for (j = 0; j < k; ++j) {
+            for (i = 0; i < z; ++i)
+                *buffer++ = f->channel_buffers[i][f->channel_buffer_start + j];
+            for (; i < channels; ++i)
+                *buffer++ = 0;
+        }
+        n += k;
+        f->channel_buffer_start += k;
+        if (n == len)
+            break;
+        if (!stb_vorbis_get_frame_float(f, NULL, &outputs))
+            break;
+    }
+    return n;
+}
+
+int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples)
+{
+    float **outputs;
+    int n = 0;
+    int z = f->channels;
+    if (z > channels) z = channels;
+    while (n < num_samples) {
+        int i;
+        int k = f->channel_buffer_end - f->channel_buffer_start;
+        if (n + k >= num_samples) k = num_samples - n;
+        if (k) {
+            for (i = 0; i < z; ++i)
+                memcpy(buffer[i] + n, f->channel_buffers[i] + f->channel_buffer_start, sizeof(float)*k);
+            for (; i < channels; ++i)
+                memset(buffer[i] + n, 0, sizeof(float) * k);
+        }
+        n += k;
+        f->channel_buffer_start += k;
+        if (n == num_samples)
+            break;
+        if (!stb_vorbis_get_frame_float(f, NULL, &outputs))
+            break;
+    }
+    return n;
+}
+#endif // STB_VORBIS_NO_PULLDATA_API
+
+/* Version history
+1.12    - 2017-11-21 - limit residue begin/end to blocksize/2 to avoid large temp allocs in bad/corrupt files
+1.11    - 2017-07-23 - fix MinGW compilation
+1.10    - 2017-03-03 - more robust seeking; fix negative ilog(); clear error in open_memory
+1.09    - 2016-04-04 - back out 'avoid discarding last frame' fix from previous version
+1.08    - 2016-04-02 - fixed multiple warnings; fix setup memory leaks;
+avoid discarding last frame of audio data
+1.07    - 2015-01-16 - fixed some warnings, fix mingw, const-correct API
+some more crash fixes when out of memory or with corrupt files
+1.06    - 2015-08-31 - full, correct support for seeking API (Dougall Johnson)
+some crash fixes when out of memory or with corrupt files
+1.05    - 2015-04-19 - don't define __forceinline if it's redundant
+1.04    - 2014-08-27 - fix missing const-correct case in API
+1.03    - 2014-08-07 - Warning fixes
+1.02    - 2014-07-09 - Declare qsort compare function _cdecl on windows
+1.01    - 2014-06-18 - fix stb_vorbis_get_samples_float
+1.0     - 2014-05-26 - fix memory leaks; fix warnings; fix bugs in multichannel
+(API change) report sample rate for decode-full-file funcs
+0.99996 - bracket #include <malloc.h> for macintosh compilation by Laurent Gomila
+0.99995 - use union instead of pointer-cast for fast-float-to-int to avoid alias-optimization problem
+0.99994 - change fast-float-to-int to work in single-precision FPU mode, remove endian-dependence
+0.99993 - remove assert that fired on legal files with empty tables
+0.99992 - rewind-to-start
+0.99991 - bugfix to stb_vorbis_get_samples_short by Bernhard Wodo
+0.9999 - (should have been 0.99990) fix no-CRT support, compiling as C++
+0.9998 - add a full-decode function with a memory source
+0.9997 - fix a bug in the read-from-FILE case in 0.9996 addition
+0.9996 - query length of vorbis stream in samples/seconds
+0.9995 - bugfix to another optimization that only happened in certain files
+0.9994 - bugfix to one of the optimizations that caused significant (but inaudible?) errors
+0.9993 - performance improvements; runs in 99% to 104% of time of reference implementation
+0.9992 - performance improvement of IMDCT; now performs close to reference implementation
+0.9991 - performance improvement of IMDCT
+0.999 - (should have been 0.9990) performance improvement of IMDCT
+0.998 - no-CRT support from Casey Muratori
+0.997 - bugfixes for bugs found by Terje Mathisen
+0.996 - bugfix: fast-huffman decode initialized incorrectly for sparse codebooks; fixing gives 10% speedup - found by Terje Mathisen
+0.995 - bugfix: fix to 'effective' overrun detection - found by Terje Mathisen
+0.994 - bugfix: garbage decode on final VQ symbol of a non-multiple - found by Terje Mathisen
+0.993 - bugfix: pushdata API required 1 extra byte for empty page (failed to consume final page if empty) - found by Terje Mathisen
+0.992 - fixes for MinGW warning
+0.991 - turn fast-float-conversion on by default
+0.990 - fix push-mode seek recovery if you seek into the headers
+0.98b - fix to bad release of 0.98
+0.98 - fix push-mode seek recovery; robustify float-to-int and support non-fast mode
+0.97 - builds under c++ (typecasting, don't use 'class' keyword)
+0.96 - somehow MY 0.95 was right, but the web one was wrong, so here's my 0.95 rereleased as 0.96, fixes a typo in the clamping code
+0.95 - clamping code for 16-bit functions
+0.94 - not publically released
+0.93 - fixed all-zero-floor case (was decoding garbage)
+0.92 - fixed a memory leak
+0.91 - conditional compiles to omit parts of the API and the infrastructure to support them: STB_VORBIS_NO_PULLDATA_API, STB_VORBIS_NO_PUSHDATA_API, STB_VORBIS_NO_STDIO, STB_VORBIS_NO_INTEGER_CONVERSION
+0.90 - first public release
+*/
+
+#endif // STB_VORBIS_HEADER_ONLY
+
+
+/*
+------------------------------------------------------------------------------
+This software is available under 2 licenses -- choose whichever you prefer.
+------------------------------------------------------------------------------
+ALTERNATIVE A - MIT License
+Copyright (c) 2017 Sean Barrett
+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.
+------------------------------------------------------------------------------
+ALTERNATIVE B - Public Domain (www.unlicense.org)
+This is free and unencumbered software released into the public domain.
+Anyone is free to copy, modify, publish, use, compile, sell, or distribute this
+software, either in source code form or as a compiled binary, for any purpose,
+commercial or non-commercial, and by any means.
+In jurisdictions that recognize copyright laws, the author or authors of this
+software dedicate any and all copyright interest in the software to the public
+domain. We make this dedication for the benefit of the public at large and to
+the detriment of our heirs and successors. We intend this dedication to be an
+overt act of relinquishment in perpetuity of all present and future rights to
+this software under copyright law.
+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 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.
+------------------------------------------------------------------------------
+*/
diff --git a/src/3rdparty/lls/lls.c b/src/3rdparty/lls/lls.c
deleted file mode 100644
index e69de29..0000000
--- a/src/3rdparty/lls/lls.c
+++ /dev/null
diff --git a/src/3rdparty/lls/lls.h b/src/3rdparty/lls/lls.h
deleted file mode 100644
index e69de29..0000000
--- a/src/3rdparty/lls/lls.h
+++ /dev/null
diff --git a/src/3rdparty/stb/stb_vorbis.c b/src/3rdparty/stb/stb_vorbis.c
index 873eda6..a863042 100644
--- a/src/3rdparty/stb/stb_vorbis.c
+++ b/src/3rdparty/stb/stb_vorbis.c
@@ -5331,7 +5331,7 @@ int stb_vorbis_decode_memory(const uint8 *mem, int len, int *channels, int *samp
     stb_vorbis *v = stb_vorbis_open_memory(mem, len, &error, NULL);
     if (v == NULL) return -1;
     limit = v->channels * 4096;
-    *channels = v->channels;
+    *channels = v->channels;    
     if (sample_rate)
         *sample_rate = v->sample_rate;
     offset = data_len = 0;
diff --git a/src/libjin/audio/sdl/audio.cpp b/src/libjin/audio/sdl/audio.cpp
index be4caa2..79ca3f2 100644
--- a/src/libjin/audio/sdl/audio.cpp
+++ b/src/libjin/audio/sdl/audio.cpp
@@ -1,6 +1,7 @@
 #include <iostream>
 #include "audio.h"
 #include "source.h"
+#include "../../math/math.h"
 
 namespace jin
 {
@@ -8,9 +9,9 @@ namespace audio
 {
 
     /* ע��callback������Ƶ�̵߳��� */
-    void defaultCallback(void *userdata, Uint8 *stream, int size)
+    static void defaultCallback(void *userdata, Uint8 *stream, int size)
     {
-        SDLAudio* audio = static_cast<SDLAudio*>(userdata);
+        static SDLAudio* audio = static_cast<SDLAudio*>(userdata);
         audio->lock();
         audio->processCommands();
         audio->processSources(stream, size);
@@ -22,14 +23,21 @@ namespace audio
         if (SDL_Init(SDL_INIT_AUDIO) < 0)
             return false;
         SDL_AudioSpec spec;
-        spec.freq     = 44100;        // 44100 Hz
-        spec.format   = AUDIO_S16SYS; // signed 16bits
-        spec.channels = 2;            // ������
-        spec.samples  = 1 << 15;      // Uin16��������2����
+        Setting* setting = (Setting*)s;
+        if (setting == nullptr)
+            return false;
+
+        unsigned int samplerate = setting->samplerate;
+        unsigned int samples = clamp(setting->samples, 1, setting->samplerate);
+
+        spec.freq     = samplerate;        // ÿ�������sample��,���õ��� 11025, 22050, 44100 and 48000 Hz. 
+        spec.format   = AUDIO_S16SYS;      // signed 16-bit samples in native byte order
+        spec.channels = SDLAUDIO_CHANNELS; // ������
+        spec.samples  = samples;           // ÿ�β���ʱ��һ�������ã����=setting->samplerate��ÿ��ֻ����1��
         spec.userdata = this;
         spec.callback = defaultCallback;
 
-        audioDevice = SDL_OpenAudioDevice(NULL, 0, &spec, NULL, 0); 
+        audioDevice = SDL_OpenAudioDevice(NULL, 0, &spec, NULL, 0);
         if (audioDevice == 0)
             return false;
         /* start audio */
diff --git a/src/libjin/audio/sdl/audio.h b/src/libjin/audio/sdl/audio.h
index 528fa7d..6e42690 100644
--- a/src/libjin/audio/sdl/audio.h
+++ b/src/libjin/audio/sdl/audio.h
@@ -9,11 +9,22 @@ namespace jin
 namespace audio
 {
 
+#define SDLAUDIO_BITDEPTH 16
+#define SDLAUDIO_BYTEDEPTH (SDLAUDIO_BITDEPTH  >> 3)
+#define SDLAUDIO_CHANNELS  2
+
     class SDLAudio : public Audio
     {
 
     public:
 
+        struct Setting : SettingBase
+        {
+        public:
+            int samplerate; // ����Ƶ��
+            int samples;    // ��������sample����<=samplerate
+        };
+
         static inline Audio* get()
         {
             return audio != NULL ? audio : (audio = new SDLAudio());
diff --git a/src/libjin/audio/sdl/source.cpp b/src/libjin/audio/sdl/source.cpp
index 1a5c5d1..51b67e4 100644
--- a/src/libjin/audio/sdl/source.cpp
+++ b/src/libjin/audio/sdl/source.cpp
@@ -8,11 +8,15 @@
 #define STB_VORBIS_HEADER_ONLY
 #include "3rdparty/stb/stb_vorbis.c"
 
+#include "audio.h"
+
 namespace jin
 {
 namespace audio
 {
 
+#define BITS 8
+
     typedef struct SDLSourceCommand
     {
         typedef enum Action
@@ -75,6 +79,7 @@ namespace audio
         char* buffer = (char*)malloc(size);
         memset(buffer, 0, size);
         fs.read(buffer, size); 
+        fs.close();
         SDLSource* source = createSource(buffer, size);
         free(buffer);
         return source;
@@ -85,17 +90,15 @@ namespace audio
         if (mem == nullptr)
             return nullptr; 
         SDLSource* source = new SDLSource();
-#define parse(FMT) case FMT : source->load##FMT(mem, size); break
         try
         {
             SourceType format = getType(mem, size);
             switch (format)
             {
-                parse(OGG);
-                parse(WAV);
+            case OGG: source->decode_ogg(mem, size); break;
+            case WAV: source->decode_wav(mem, size); break;
             }
         }
-#undef parse
         catch (SourceException& exp)
         {
             delete source;
@@ -117,37 +120,44 @@ namespace audio
         raw.data = 0;
     }
 
-    void SDLSource::loadWAV(void* mem, int size)
+    void SDLSource::decode_wav(void* mem, int size)
     {
         wav_t wav; 
         if (wav_read(&wav, mem, size) == 0)
         {
-            raw.data     = wav.data;
-            raw.size     = wav.length * wav.bitdepth / 8;
-            raw.end      = (char*)raw.data + raw.size;
-            raw.rate     = wav.samplerate;
-            raw.bitdepth = wav.bitdepth;
-            raw.samples  = raw.size / (wav.bitdepth / 8.f);
-            raw.channel  = clamp(wav.channels, CHANNEL::MONO, CHANNEL::STEREO);
-            raw.silence  = 0;
+            raw.data       = wav.data;
+            raw.length     = wav.length * wav.bitdepth / 8;
+            raw.end        = (char*)raw.data + raw.length;
+            raw.samplerate = wav.samplerate;
+            raw.bitdepth   = wav.bitdepth;
+            raw.samples    = raw.length / (wav.bitdepth / 8.f) / wav.channels;
+            raw.channels   = clamp(wav.channels, CHANNEL::MONO, CHANNEL::STEREO);
         }
         else
             throw SourceException();
     }
 
-    void SDLSource::loadOGG(void* mem, int size)
+    void SDLSource::decode_ogg(void* _mem, int size)
     {
-        raw.samples  = stb_vorbis_decode_memory((unsigned char*)mem, size, (int*)&raw.channel, &raw.rate, (short**)&raw.data) << 1;
-        raw.channel  = clamp(raw.channel, CHANNEL::MONO, CHANNEL::STEREO);
-        raw.size     = raw.samples << 1; // 2 bytes each sample
-        raw.bitdepth = 16; 
-        raw.end      = (char*)raw.data + raw.size; 
-        raw.silence  = 0;
-
-        if (raw.samples < 0)
-        {
-            throw SourceException();
-        }
+        unsigned char* mem = (unsigned char*)_mem;
+        int channels;
+        int samplerate; 
+        short* data = (short*)raw.data;
+        int samples = stb_vorbis_decode_memory(
+            mem,
+            size,
+            &channels,
+            &samplerate,
+            &data
+        );
+        const int bitdepth = sizeof(short) * BITS;
+        raw.channels   = channels;
+        raw.samplerate = samplerate;
+        raw.data       = data; 
+        raw.samples    = samples;
+        raw.length     = samples * channels * sizeof(short);
+        raw.bitdepth   = bitdepth;
+        raw.end        = (char*)data + raw.length; 
     }
 
 #define ActionNone(T)\
@@ -227,6 +237,85 @@ Manager::get()->pushCommand(cmd);           \
         ActionFloat(SetRate, rate);
     }
 
+    inline void SDLSource::handle(
+        SDLSourceManager* manager, 
+        SDLSourceCommand* cmd
+    )
+    {
+        switch (cmd->action)
+        {
+        case Command::Action::Play:
+            manager->removeSource(this);
+            manager->pushSource(this);
+            status.state = PLAYING;
+            status.pos = 0;  // rewind
+            break;
+        case Command::Action::Stop:
+            manager->removeSource(this);
+            status.state = STOPPED;
+            status.pos = 0; // rewind
+            break;
+        case Command::Action::Pause:
+            manager->removeSource(this);
+            status.state = PAUSED;
+            break;
+        case Command::Action::Resume:
+            manager->removeSource(this);
+            manager->pushSource(this);
+            status.state = PLAYING;
+            break;
+        case Command::Action::Rewind:
+            status.state = PLAYING;
+            status.pos = 0;
+            break;
+        case Command::Action::SetVolume:
+            //float cmd->parameter._float;
+            break;
+        case Command::Action::SetLoop:
+            status.loop = cmd->parameter._boolean;
+            break;
+        }
+    }
+    
+    inline void SDLSource::process(void* buf, size_t size)
+    {
+        short* buffer = (short*)buf; // AUDIO_S16SYS
+        unsigned int samples = size / SDLAUDIO_BYTEDEPTH;
+        short* sample;
+        short  origin;
+
+        const char bitdepth = raw.bitdepth;
+        const char channles = raw.channels;
+
+        int pos     = status.pos; 
+        int pitch   = status.pitch; 
+        int state   = status.state;
+        bool loop   = status.loop;
+        int volume  = status.volume;
+        short* clip16 = nullptr;
+        char* clip8 = nullptr;
+        int clip    = 0;
+        
+        if (bitdepth == 8)
+            clip8 = (char*)raw.data;
+        else if (bitdepth == 16)
+            clip16 = (short*)raw.data;
+
+        for (int i = 0; i < samples; i+=2 /*˫����*/)
+        {
+            /* ��Ƶ�ļ���sample���ᱻ���������� */
+            sample = buffer + i * SDLAUDIO_BYTEDEPTH;
+            origin = *sample;
+            if (bitdepth == 8)
+            {
+                clip = *clip8;
+            }
+            else if (bitdepth == 16)
+                clip = *clip16;
+            
+        }
+    }
+
     Manager* Manager::get()
     {
         return (manager == nullptr ? manager = new Manager() : manager);
@@ -243,43 +332,7 @@ Manager::get()->pushCommand(cmd);           \
             {
                 source = cmd->source;
                 if (source != nullptr)
-                {
-                    switch (cmd->action)
-                    {
-                    case Command::Action::Play:
-                        removeSource(source);
-                        pushSource(source);
-                        source->status.state = PLAYING;
-                        source->status.pos = 0;  // rewind
-                        break;
-                    case Command::Action::Stop:
-                        manager->removeSource(source);
-                        source->status.state = STOPPED;
-                        source->status.pos = 0; // rewind
-                        break;
-                    case Command::Action::Pause:
-                        manager->removeSource(source);
-                        source->status.state = PAUSED;
-                        break;
-                    case Command::Action::Resume:
-                        manager->removeSource(source);
-                        manager->pushSource(source);
-                        source->status.state = PLAYING;
-                        break;
-                    case Command::Action::Rewind:
-                        source->status.state = PLAYING;
-                        source->status.pos = 0;
-                        break;
-                    case Command::Action::SetVolume:
-                        //float cmd->parameter._float;
-                        break;
-                    case Command::Action::SetLoop:
-                        source->status.loop = cmd->parameter._boolean;
-                        break;
-                        /*case Command::Action::SetRate:
-                        */
-                    }
-                }
+                    source->handle(manager, cmd);
             }
             commands.pop();
         }
@@ -288,38 +341,15 @@ Manager::get()->pushCommand(cmd);           \
     /* AUDIO_S16SYS[size>>1] buffer */
     shared void Manager::processSources(void* buf, size_t size)
     {
-        Sint16* buffer = (Sint16*)buf;
-        unsigned int samples = size >> 1;
-        memset(buffer, 0, size);
+        /* clear render buffer */
+        memset(buf, 0, size);
         SDLSource* src = nullptr; 
         std::vector<SDLSource*>::iterator it = sources.begin();
         for (; it != sources.end();)
         {
             src = *it;
-            if (src == nullptr)
-                goto next;
-            src->status.pos;
-            Sint16* source = (Sint16*)((char*)src->raw.data + src->status.pos);
-            unsigned int remain = src->raw.samples - (src->status.pos >> 1);
-            for (int i = 0; i < min(remain, samples); ++i)
-            {
-                buffer[i] += (src->raw.channel == CHANNEL::STEREO ? source[i] : source[(i % 2) * 2]);
-            }
-            if (remain < samples)
-            {
-                if (!src->status.loop)
-                {
-                    it = sources.erase(it);
-                    continue;
-                }
-                int j = 0;
-                for (int i = samples - remain; i < samples; ++i, ++j)
-                    buffer[i] += (src->raw.channel == CHANNEL::STEREO ? source[j] : source[j % 2 * 2]);
-                src->status.pos = (j << 1);
-                continue;
-            }
-            src->status.pos += (samples << 1);
-        next:
+            if (src != nullptr)
+                src->process(buf, size);
             ++it;
         }
     }
@@ -360,13 +390,5 @@ Manager::get()->pushCommand(cmd);           \
         return new Command();
     }
 
-    shared void Manager::collectCommand(Command* cmd)
-    {
-        if (cmd != nullptr)
-        {
-            commandsPool.push(cmd);
-        }
-    }
-
 }
 }
\ No newline at end of file
diff --git a/src/libjin/audio/sdl/source.h b/src/libjin/audio/sdl/source.h
index 365d6ff..ff311b6 100644
--- a/src/libjin/audio/sdl/source.h
+++ b/src/libjin/audio/sdl/source.h
@@ -14,6 +14,7 @@ namespace audio
 {
 
     typedef struct SDLSourceCommand;
+    class SDLSourceManager;
 
     class SDLSource : public Source
     {
@@ -39,33 +40,33 @@ namespace audio
         bool setLoop(bool loop) override;
         void setRate(float rate) override;
 
+        inline void handle(SDLSourceManager* manager, SDLSourceCommand* cmd);
+        inline void process(void* buffer, size_t size);
+
     private:
 
         SDLSource();
 
-        friend class SDLSourceManager;
-
-        void loadWAV(void* mem, int size);
-        void loadOGG(void* mem, int size);
+        void decode_wav(void* mem, int size);
+        void decode_ogg(void* mem, int size);
 
         inline bool is(int state) const { return (status.state & state) == state; }
 
         struct
         {
             const void* data;       // ��Ƶ����
-            int size;               // data�ֽڳ���
+            int length;             // data�ֽڳ���
             const void* end;        // data��β = (unsigned char*)data + size
-            int rate;               // ����Ƶ��
+            int samplerate;         // ����Ƶ��
             unsigned char bitdepth; // ÿ��sample�ı��س���
             int samples;            // sample�� = size / (bitdepth / 8)
-            unsigned char channel;  // channel��1(mono)��2(stereo)
-            char silence;           // 0
+            unsigned char channels; // channel��1(mono)��2(stereo)
         } raw;
 
         /* Procedure controller variable */
         struct
         {
-            int pos;                // ��ǰ���ŵ�λ��
+            int pos;                // ��ǰ���ŵ�sample
             int pitch;              // pitch
             int state;              // ��ǰ״̬
             bool loop;              // loop or not
@@ -85,10 +86,6 @@ namespace audio
         static void processCommands();
         static void processSources(void* buffer, size_t size);
 
-    private:
-
-        friend class SDLSource;
-
         static void removeSource(SDLSource* source);
         static void pushSource(SDLSource* source);
         static SDLSourceCommand* getCommand();
diff --git a/src/libjin/audio/source.cpp b/src/libjin/audio/source.cpp
index ec80a0f..a09bbab 100644
--- a/src/libjin/audio/source.cpp
+++ b/src/libjin/audio/source.cpp
@@ -6,14 +6,17 @@ namespace jin
 namespace audio
 {
 
+    static int check_header(const void *data, int size, char *str, int offset) {
+        int len = strlen(str);
+        return (size >= offset + len) && !memcmp((char*)data + offset, str, len);
+    }
+
     SourceType Source::getType(const void* mem, int size)
     {
-        const char* p = (const char* )mem;
-        if (memcmp(p, "RIFF", 4) == 0 && memcmp(p + 8, "WAVE", 4) == 0)
+        if(check_header(mem, size, "WAVE", 8))
             return SourceType::WAV;
-        if (memcmp(p, "OggS", 4) == 0)
+        if(check_header(mem, size, "OggS", 0))
             return SourceType::OGG;
-
         return SourceType::INVALID;
     }
 
diff --git a/src/libjin/utils/unittest.cpp b/src/libjin/utils/unittest.cpp
index ce72173..951b996 100644
--- a/src/libjin/utils/unittest.cpp
+++ b/src/libjin/utils/unittest.cpp
@@ -29,4 +29,80 @@ int main(int argc, char* argv[])
     return 0;
 }
 
+/*
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <SDL2/SDL.h>
+
+#include <3rdparty/cmixer/cmixer.h>
+
+static SDL_mutex* audio_mutex;
+
+static void lock_handler(cm_Event *e) {
+    if (e->type == CM_EVENT_LOCK) {
+        SDL_LockMutex(audio_mutex);
+    }
+    if (e->type == CM_EVENT_UNLOCK) {
+        SDL_UnlockMutex(audio_mutex);
+    }
+}
+
+
+static void audio_callback(void *udata, Uint8 *stream, int size) {
+    cm_process((cm_Int16*)stream, size / 2);
+}
+
+
+int main(int argc, char **argv) {
+    SDL_AudioDeviceID dev;
+    SDL_AudioSpec fmt, got;
+    cm_Source *src;
+    cm_Source* src2;
+
+
+    SDL_Init(SDL_INIT_AUDIO);
+    audio_mutex = SDL_CreateMutex();
+
+    memset(&fmt, 0, sizeof(fmt));
+    fmt.freq = 44100;
+    fmt.format = AUDIO_S16;
+    fmt.channels = 2;
+    fmt.samples = 1024;
+    fmt.callback = audio_callback;
+
+    dev = SDL_OpenAudioDevice(NULL, 0, &fmt, &got, SDL_AUDIO_ALLOW_FREQUENCY_CHANGE);
+    if (dev == 0) {
+        fprintf(stderr, "Error: failed to open audio device '%s'\n", SDL_GetError());
+        exit(EXIT_FAILURE);
+    }
+
+    cm_init(got.freq);
+    cm_set_lock(lock_handler);
+    cm_set_master_gain(0.5);
+
+    SDL_PauseAudioDevice(dev, 0);
+
+    src = cm_new_source_from_file("a.ogg");
+    src2 = cm_new_source_from_file("loop.wav");
+    if (!src) {
+        fprintf(stderr, "Error: failed to create source '%s'\n", cm_get_error());
+        exit(EXIT_FAILURE);
+    }
+    cm_set_loop(src2, 1);
+
+    cm_play(src);
+    cm_play(src2);
+
+    printf("Press [return] to exit\n");
+    getchar();
+
+    cm_destroy_source(src);
+    SDL_CloseAudioDevice(dev);
+    SDL_Quit();
+
+    return EXIT_SUCCESS;
+}
+*/
+
 #endif
\ No newline at end of file
diff --git a/src/lls/llsbind_jin.h b/src/lls/llsbind_jin.h
deleted file mode 100644
index 6445b4d..0000000
--- a/src/lls/llsbind_jin.h
+++ /dev/null
@@ -1,2 +0,0 @@
-
-void llsbind_jin();
\ No newline at end of file
diff --git a/src/lua/graphics/luaopen_graphics.cpp b/src/lua/graphics/luaopen_graphics.cpp
index 7db43c5..362ee93 100644
--- a/src/lua/graphics/luaopen_graphics.cpp
+++ b/src/lua/graphics/luaopen_graphics.cpp
@@ -491,7 +491,7 @@ namespace lua
         {"line",     l_drawLine}, 
         {"rect",     l_drawRect},
         {"circle",   l_drawCircle},
-        {"triangle", l_drawTriangle}, 
+        {"triangle", l_drawTriangle},
         {"polygon",  l_drawPolygon},
         // 
         {"destroy",  l_destroy},
diff --git a/src/lua/luaopen_jin.cpp b/src/lua/luaopen_jin.cpp
index 93e0422..c50db14 100644
--- a/src/lua/luaopen_jin.cpp
+++ b/src/lua/luaopen_jin.cpp
@@ -8,7 +8,7 @@ namespace jin
 {
 namespace lua
 {
-
+    
     extern int luaopen_core(lua_State* L);
     extern int luaopen_graphics(lua_State* L);
     extern int luaopen_audio(lua_State* L);
@@ -53,16 +53,16 @@ namespace lua
     
     // submodules 
     static const luaL_Reg mods[] = {
-        {"core",        luaopen_core},
-        {"event",       luaopen_event},
-        {"graphics",    luaopen_graphics},
-        {"time",        luaopen_time},
-        {"mouse",       luaopen_mouse},
-        {"keyboard",    luaopen_keyboard},
-        {"filesystem",  luaopen_filesystem},
-        {"net",         luaopen_net},
-        {"audio",       luaopen_audio},
-        {"joypad",      luaopen_joypad},
+        {"core",       luaopen_core},
+        {"event",      luaopen_event},
+        {"graphics",   luaopen_graphics},
+        {"time",       luaopen_time},
+        {"mouse",      luaopen_mouse},
+        {"keyboard",   luaopen_keyboard},
+        {"filesystem", luaopen_filesystem},
+        {"net",        luaopen_net},
+        {"audio",      luaopen_audio},
+        {"joypad",     luaopen_joypad},
         {0, 0}
     };
 
diff --git a/src/lua/luaopen_types.h b/src/lua/luaopen_types.h
index ed0d16d..8c1756f 100644
--- a/src/lua/luaopen_types.h
+++ b/src/lua/luaopen_types.h
@@ -18,12 +18,12 @@ namespace lua
 	class Proxy
 	{
 	public:
-		inline void bind(void* obj)
+		inline void bind(const void* obj)
 		{
 			object = obj;
 		}
 
-		void* object;
+		const void* object;
 	};
 
 }
diff --git a/src/lua/time/luaopen_time.cpp b/src/lua/time/luaopen_time.cpp
index f7310dc..370d868 100644
--- a/src/lua/time/luaopen_time.cpp
+++ b/src/lua/time/luaopen_time.cpp
@@ -19,8 +19,8 @@ namespace lua
     }
 
     static const luaL_Reg f[] = {
-        {"second",  l_sec},
-        {"sleep",   l_sleep},
+        {"second", l_sec},
+        {"sleep",  l_sleep},
         {0, 0},
     };