gosu 0.7.45 → 0.7.46

data/dependencies/libvorbis/lib/barkmel.c

@@ -0,0 +1,64 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: bark scale utility
+ last mod: $Id: barkmel.c 16037 2009-05-26 21:10:58Z xiphmont $
+
+ ********************************************************************/
+
+#include <stdio.h>
+#include "scales.h"
+int main(){
+  int i;
+  double rate;
+  for(i=64;i<32000;i*=2){
+    rate=48000.f;
+    fprintf(stderr,"rate=%gHz, block=%d, f(1)=%.2gHz bark(1)=%.2g (of %.2g)\n",
+            rate,i,rate/2 / (i/2),toBARK(rate/2 /(i/2)),toBARK(rate/2));
+
+    rate=44100.f;
+    fprintf(stderr,"rate=%gHz, block=%d, f(1)=%.2gHz bark(1)=%.2g (of %.2g)\n",
+            rate,i,rate/2 / (i/2),toBARK(rate/2 /(i/2)),toBARK(rate/2));
+
+    rate=32000.f;
+    fprintf(stderr,"rate=%gHz, block=%d, f(1)=%.2gHz bark(1)=%.2g (of %.2g)\n",
+            rate,i,rate/2 / (i/2),toBARK(rate/2 /(i/2)),toBARK(rate/2));
+
+    rate=22050.f;
+    fprintf(stderr,"rate=%gHz, block=%d, f(1)=%.2gHz bark(1)=%.2g (of %.2g)\n",
+            rate,i,rate/2 / (i/2),toBARK(rate/2 /(i/2)),toBARK(rate/2));
+
+    rate=16000.f;
+    fprintf(stderr,"rate=%gHz, block=%d, f(1)=%.2gHz bark(1)=%.2g (of %.2g)\n",
+            rate,i,rate/2 / (i/2),toBARK(rate/2 /(i/2)),toBARK(rate/2));
+
+    rate=11025.f;
+    fprintf(stderr,"rate=%gHz, block=%d, f(1)=%.2gHz bark(1)=%.2g (of %.2g)\n",
+            rate,i,rate/2 / (i/2),toBARK(rate/2 /(i/2)),toBARK(rate/2));
+
+    rate=8000.f;
+    fprintf(stderr,"rate=%gHz, block=%d, f(1)=%.2gHz bark(1)=%.2g (of %.2g)\n\n",
+            rate,i,rate/2 / (i/2),toBARK(rate/2 /(i/2)),toBARK(rate/2));
+
+
+  }
+  {
+    float i;
+    int j;
+    for(i=0.,j=0;i<28;i+=1,j++){
+      fprintf(stderr,"(%d) bark=%f %gHz (%d of 128)\n",
+              j,i,fromBARK(i),(int)(fromBARK(i)/22050.*128.));
+    }
+  }
+  return(0);
+}
+

data/dependencies/libvorbis/lib/bitrate.c

@@ -0,0 +1,253 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: bitrate tracking and management
+ last mod: $Id: bitrate.c 16227 2009-07-08 06:58:46Z xiphmont $
+
+ ********************************************************************/
+
+#include <stdlib.h>
+#include <string.h>
+#include <math.h>
+#include <ogg/ogg.h>
+#include "vorbis/codec.h"
+#include "codec_internal.h"
+#include "os.h"
+#include "misc.h"
+#include "bitrate.h"
+
+/* compute bitrate tracking setup  */
+void vorbis_bitrate_init(vorbis_info *vi,bitrate_manager_state *bm){
+  codec_setup_info *ci=vi->codec_setup;
+  bitrate_manager_info *bi=&ci->bi;
+
+  memset(bm,0,sizeof(*bm));
+
+  if(bi && (bi->reservoir_bits>0)){
+    long ratesamples=vi->rate;
+    int  halfsamples=ci->blocksizes[0]>>1;
+
+    bm->short_per_long=ci->blocksizes[1]/ci->blocksizes[0];
+    bm->managed=1;
+
+    bm->avg_bitsper= rint(1.*bi->avg_rate*halfsamples/ratesamples);
+    bm->min_bitsper= rint(1.*bi->min_rate*halfsamples/ratesamples);
+    bm->max_bitsper= rint(1.*bi->max_rate*halfsamples/ratesamples);
+
+    bm->avgfloat=PACKETBLOBS/2;
+
+    /* not a necessary fix, but one that leads to a more balanced
+       typical initialization */
+    {
+      long desired_fill=bi->reservoir_bits*bi->reservoir_bias;
+      bm->minmax_reservoir=desired_fill;
+      bm->avg_reservoir=desired_fill;
+    }
+
+  }
+}
+
+void vorbis_bitrate_clear(bitrate_manager_state *bm){
+  memset(bm,0,sizeof(*bm));
+  return;
+}
+
+int vorbis_bitrate_managed(vorbis_block *vb){
+  vorbis_dsp_state      *vd=vb->vd;
+  private_state         *b=vd->backend_state;
+  bitrate_manager_state *bm=&b->bms;
+
+  if(bm && bm->managed)return(1);
+  return(0);
+}
+
+/* finish taking in the block we just processed */
+int vorbis_bitrate_addblock(vorbis_block *vb){
+  vorbis_block_internal *vbi=vb->internal;
+  vorbis_dsp_state      *vd=vb->vd;
+  private_state         *b=vd->backend_state;
+  bitrate_manager_state *bm=&b->bms;
+  vorbis_info           *vi=vd->vi;
+  codec_setup_info      *ci=vi->codec_setup;
+  bitrate_manager_info  *bi=&ci->bi;
+
+  int  choice=rint(bm->avgfloat);
+  long this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
+  long min_target_bits=(vb->W?bm->min_bitsper*bm->short_per_long:bm->min_bitsper);
+  long max_target_bits=(vb->W?bm->max_bitsper*bm->short_per_long:bm->max_bitsper);
+  int  samples=ci->blocksizes[vb->W]>>1;
+  long desired_fill=bi->reservoir_bits*bi->reservoir_bias;
+  if(!bm->managed){
+    /* not a bitrate managed stream, but for API simplicity, we'll
+       buffer the packet to keep the code path clean */
+
+    if(bm->vb)return(-1); /* one has been submitted without
+                             being claimed */
+    bm->vb=vb;
+    return(0);
+  }
+
+  bm->vb=vb;
+
+  /* look ahead for avg floater */
+  if(bm->avg_bitsper>0){
+    double slew=0.;
+    long avg_target_bits=(vb->W?bm->avg_bitsper*bm->short_per_long:bm->avg_bitsper);
+    double slewlimit= 15./bi->slew_damp;
+
+    /* choosing a new floater:
+       if we're over target, we slew down
+       if we're under target, we slew up
+
+       choose slew as follows: look through packetblobs of this frame
+       and set slew as the first in the appropriate direction that
+       gives us the slew we want.  This may mean no slew if delta is
+       already favorable.
+
+       Then limit slew to slew max */
+
+    if(bm->avg_reservoir+(this_bits-avg_target_bits)>desired_fill){
+      while(choice>0 && this_bits>avg_target_bits &&
+            bm->avg_reservoir+(this_bits-avg_target_bits)>desired_fill){
+        choice--;
+        this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
+      }
+    }else if(bm->avg_reservoir+(this_bits-avg_target_bits)<desired_fill){
+      while(choice+1<PACKETBLOBS && this_bits<avg_target_bits &&
+            bm->avg_reservoir+(this_bits-avg_target_bits)<desired_fill){
+        choice++;
+        this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
+      }
+    }
+
+    slew=rint(choice-bm->avgfloat)/samples*vi->rate;
+    if(slew<-slewlimit)slew=-slewlimit;
+    if(slew>slewlimit)slew=slewlimit;
+    choice=rint(bm->avgfloat+= slew/vi->rate*samples);
+    this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
+  }
+
+
+
+  /* enforce min(if used) on the current floater (if used) */
+  if(bm->min_bitsper>0){
+    /* do we need to force the bitrate up? */
+    if(this_bits<min_target_bits){
+      while(bm->minmax_reservoir-(min_target_bits-this_bits)<0){
+        choice++;
+        if(choice>=PACKETBLOBS)break;
+        this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
+      }
+    }
+  }
+
+  /* enforce max (if used) on the current floater (if used) */
+  if(bm->max_bitsper>0){
+    /* do we need to force the bitrate down? */
+    if(this_bits>max_target_bits){
+      while(bm->minmax_reservoir+(this_bits-max_target_bits)>bi->reservoir_bits){
+        choice--;
+        if(choice<0)break;
+        this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
+      }
+    }
+  }
+
+  /* Choice of packetblobs now made based on floater, and min/max
+     requirements. Now boundary check extreme choices */
+
+  if(choice<0){
+    /* choosing a smaller packetblob is insufficient to trim bitrate.
+       frame will need to be truncated */
+    long maxsize=(max_target_bits+(bi->reservoir_bits-bm->minmax_reservoir))/8;
+    bm->choice=choice=0;
+
+    if(oggpack_bytes(vbi->packetblob[choice])>maxsize){
+
+      oggpack_writetrunc(vbi->packetblob[choice],maxsize*8);
+      this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
+    }
+  }else{
+    long minsize=(min_target_bits-bm->minmax_reservoir+7)/8;
+    if(choice>=PACKETBLOBS)
+      choice=PACKETBLOBS-1;
+
+    bm->choice=choice;
+
+    /* prop up bitrate according to demand. pad this frame out with zeroes */
+    minsize-=oggpack_bytes(vbi->packetblob[choice]);
+    while(minsize-->0)oggpack_write(vbi->packetblob[choice],0,8);
+    this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
+
+  }
+
+  /* now we have the final packet and the final packet size.  Update statistics */
+  /* min and max reservoir */
+  if(bm->min_bitsper>0 || bm->max_bitsper>0){
+
+    if(max_target_bits>0 && this_bits>max_target_bits){
+      bm->minmax_reservoir+=(this_bits-max_target_bits);
+    }else if(min_target_bits>0 && this_bits<min_target_bits){
+      bm->minmax_reservoir+=(this_bits-min_target_bits);
+    }else{
+      /* inbetween; we want to take reservoir toward but not past desired_fill */
+      if(bm->minmax_reservoir>desired_fill){
+        if(max_target_bits>0){ /* logical bulletproofing against initialization state */
+          bm->minmax_reservoir+=(this_bits-max_target_bits);
+          if(bm->minmax_reservoir<desired_fill)bm->minmax_reservoir=desired_fill;
+        }else{
+          bm->minmax_reservoir=desired_fill;
+        }
+      }else{
+        if(min_target_bits>0){ /* logical bulletproofing against initialization state */
+          bm->minmax_reservoir+=(this_bits-min_target_bits);
+          if(bm->minmax_reservoir>desired_fill)bm->minmax_reservoir=desired_fill;
+        }else{
+          bm->minmax_reservoir=desired_fill;
+        }
+      }
+    }
+  }
+
+  /* avg reservoir */
+  if(bm->avg_bitsper>0){
+    long avg_target_bits=(vb->W?bm->avg_bitsper*bm->short_per_long:bm->avg_bitsper);
+    bm->avg_reservoir+=this_bits-avg_target_bits;
+  }
+
+  return(0);
+}
+
+int vorbis_bitrate_flushpacket(vorbis_dsp_state *vd,ogg_packet *op){
+  private_state         *b=vd->backend_state;
+  bitrate_manager_state *bm=&b->bms;
+  vorbis_block          *vb=bm->vb;
+  int                    choice=PACKETBLOBS/2;
+  if(!vb)return 0;
+
+  if(op){
+    vorbis_block_internal *vbi=vb->internal;
+
+    if(vorbis_bitrate_managed(vb))
+      choice=bm->choice;
+
+    op->packet=oggpack_get_buffer(vbi->packetblob[choice]);
+    op->bytes=oggpack_bytes(vbi->packetblob[choice]);
+    op->b_o_s=0;
+    op->e_o_s=vb->eofflag;
+    op->granulepos=vb->granulepos;
+    op->packetno=vb->sequence; /* for sake of completeness */
+  }
+
+  bm->vb=0;
+  return(1);
+}

data/dependencies/libvorbis/lib/bitrate.h

@@ -0,0 +1,59 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: bitrate tracking and management
+ last mod: $Id: bitrate.h 13293 2007-07-24 00:09:47Z xiphmont $
+
+ ********************************************************************/
+
+#ifndef _V_BITRATE_H_
+#define _V_BITRATE_H_
+
+#include "vorbis/codec.h"
+#include "codec_internal.h"
+#include "os.h"
+
+/* encode side bitrate tracking */
+typedef struct bitrate_manager_state {
+  int            managed;
+
+  long           avg_reservoir;
+  long           minmax_reservoir;
+  long           avg_bitsper;
+  long           min_bitsper;
+  long           max_bitsper;
+
+  long           short_per_long;
+  double         avgfloat;
+
+  vorbis_block  *vb;
+  int            choice;
+} bitrate_manager_state;
+
+typedef struct bitrate_manager_info{
+  long           avg_rate;
+  long           min_rate;
+  long           max_rate;
+  long           reservoir_bits;
+  double         reservoir_bias;
+
+  double         slew_damp;
+
+} bitrate_manager_info;
+
+extern void vorbis_bitrate_init(vorbis_info *vi,bitrate_manager_state *bs);
+extern void vorbis_bitrate_clear(bitrate_manager_state *bs);
+extern int vorbis_bitrate_managed(vorbis_block *vb);
+extern int vorbis_bitrate_addblock(vorbis_block *vb);
+extern int vorbis_bitrate_flushpacket(vorbis_dsp_state *vd, ogg_packet *op);
+
+#endif

data/dependencies/libvorbis/lib/block.c

@@ -0,0 +1,1046 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: PCM data vector blocking, windowing and dis/reassembly
+ last mod: $Id: block.c 17561 2010-10-23 10:34:24Z xiphmont $
+
+ Handle windowing, overlap-add, etc of the PCM vectors.  This is made
+ more amusing by Vorbis' current two allowed block sizes.
+
+ ********************************************************************/
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <ogg/ogg.h>
+#include "vorbis/codec.h"
+#include "codec_internal.h"
+
+#include "window.h"
+#include "mdct.h"
+#include "lpc.h"
+#include "registry.h"
+#include "misc.h"
+
+static int ilog2(unsigned int v){
+  int ret=0;
+  if(v)--v;
+  while(v){
+    ret++;
+    v>>=1;
+  }
+  return(ret);
+}
+
+/* pcm accumulator examples (not exhaustive):
+
+ <-------------- lW ---------------->
+                   <--------------- W ---------------->
+:            .....|.....       _______________         |
+:        .'''     |     '''_---      |       |\        |
+:.....'''         |_____--- '''......|       | \_______|
+:.................|__________________|_______|__|______|
+                  |<------ Sl ------>|      > Sr <     |endW
+                  |beginSl           |endSl  |  |endSr
+                  |beginW            |endlW  |beginSr
+
+
+                      |< lW >|
+                   <--------------- W ---------------->
+                  |   |  ..  ______________            |
+                  |   | '  `/        |     ---_        |
+                  |___.'___/`.       |         ---_____|
+                  |_______|__|_______|_________________|
+                  |      >|Sl|<      |<------ Sr ----->|endW
+                  |       |  |endSl  |beginSr          |endSr
+                  |beginW |  |endlW
+                  mult[0] |beginSl                     mult[n]
+
+ <-------------- lW ----------------->
+                          |<--W-->|
+:            ..............  ___  |   |
+:        .'''             |`/   \ |   |
+:.....'''                 |/`....\|...|
+:.........................|___|___|___|
+                          |Sl |Sr |endW
+                          |   |   |endSr
+                          |   |beginSr
+                          |   |endSl
+                          |beginSl
+                          |beginW
+*/
+
+/* block abstraction setup *********************************************/
+
+#ifndef WORD_ALIGN
+#define WORD_ALIGN 8
+#endif
+
+int vorbis_block_init(vorbis_dsp_state *v, vorbis_block *vb){
+  int i;
+  memset(vb,0,sizeof(*vb));
+  vb->vd=v;
+  vb->localalloc=0;
+  vb->localstore=NULL;
+  if(v->analysisp){
+    vorbis_block_internal *vbi=
+      vb->internal=_ogg_calloc(1,sizeof(vorbis_block_internal));
+    vbi->ampmax=-9999;
+
+    for(i=0;i<PACKETBLOBS;i++){
+      if(i==PACKETBLOBS/2){
+        vbi->packetblob[i]=&vb->opb;
+      }else{
+        vbi->packetblob[i]=
+          _ogg_calloc(1,sizeof(oggpack_buffer));
+      }
+      oggpack_writeinit(vbi->packetblob[i]);
+    }
+  }
+
+  return(0);
+}
+
+void *_vorbis_block_alloc(vorbis_block *vb,long bytes){
+  bytes=(bytes+(WORD_ALIGN-1)) & ~(WORD_ALIGN-1);
+  if(bytes+vb->localtop>vb->localalloc){
+    /* can't just _ogg_realloc... there are outstanding pointers */
+    if(vb->localstore){
+      struct alloc_chain *link=_ogg_malloc(sizeof(*link));
+      vb->totaluse+=vb->localtop;
+      link->next=vb->reap;
+      link->ptr=vb->localstore;
+      vb->reap=link;
+    }
+    /* highly conservative */
+    vb->localalloc=bytes;
+    vb->localstore=_ogg_malloc(vb->localalloc);
+    vb->localtop=0;
+  }
+  {
+    void *ret=(void *)(((char *)vb->localstore)+vb->localtop);
+    vb->localtop+=bytes;
+    return ret;
+  }
+}
+
+/* reap the chain, pull the ripcord */
+void _vorbis_block_ripcord(vorbis_block *vb){
+  /* reap the chain */
+  struct alloc_chain *reap=vb->reap;
+  while(reap){
+    struct alloc_chain *next=reap->next;
+    _ogg_free(reap->ptr);
+    memset(reap,0,sizeof(*reap));
+    _ogg_free(reap);
+    reap=next;
+  }
+  /* consolidate storage */
+  if(vb->totaluse){
+    vb->localstore=_ogg_realloc(vb->localstore,vb->totaluse+vb->localalloc);
+    vb->localalloc+=vb->totaluse;
+    vb->totaluse=0;
+  }
+
+  /* pull the ripcord */
+  vb->localtop=0;
+  vb->reap=NULL;
+}
+
+int vorbis_block_clear(vorbis_block *vb){
+  int i;
+  vorbis_block_internal *vbi=vb->internal;
+
+  _vorbis_block_ripcord(vb);
+  if(vb->localstore)_ogg_free(vb->localstore);
+
+  if(vbi){
+    for(i=0;i<PACKETBLOBS;i++){
+      oggpack_writeclear(vbi->packetblob[i]);
+      if(i!=PACKETBLOBS/2)_ogg_free(vbi->packetblob[i]);
+    }
+    _ogg_free(vbi);
+  }
+  memset(vb,0,sizeof(*vb));
+  return(0);
+}
+
+/* Analysis side code, but directly related to blocking.  Thus it's
+   here and not in analysis.c (which is for analysis transforms only).
+   The init is here because some of it is shared */
+
+static int _vds_shared_init(vorbis_dsp_state *v,vorbis_info *vi,int encp){
+  int i;
+  codec_setup_info *ci=vi->codec_setup;
+  private_state *b=NULL;
+  int hs;
+
+  if(ci==NULL) return 1;
+  hs=ci->halfrate_flag;
+
+  memset(v,0,sizeof(*v));
+  b=v->backend_state=_ogg_calloc(1,sizeof(*b));
+
+  v->vi=vi;
+  b->modebits=ilog2(ci->modes);
+
+  b->transform[0]=_ogg_calloc(VI_TRANSFORMB,sizeof(*b->transform[0]));
+  b->transform[1]=_ogg_calloc(VI_TRANSFORMB,sizeof(*b->transform[1]));
+
+  /* MDCT is tranform 0 */
+
+  b->transform[0][0]=_ogg_calloc(1,sizeof(mdct_lookup));
+  b->transform[1][0]=_ogg_calloc(1,sizeof(mdct_lookup));
+  mdct_init(b->transform[0][0],ci->blocksizes[0]>>hs);
+  mdct_init(b->transform[1][0],ci->blocksizes[1]>>hs);
+
+  /* Vorbis I uses only window type 0 */
+  b->window[0]=ilog2(ci->blocksizes[0])-6;
+  b->window[1]=ilog2(ci->blocksizes[1])-6;
+
+  if(encp){ /* encode/decode differ here */
+
+    /* analysis always needs an fft */
+    drft_init(&b->fft_look[0],ci->blocksizes[0]);
+    drft_init(&b->fft_look[1],ci->blocksizes[1]);
+
+    /* finish the codebooks */
+    if(!ci->fullbooks){
+      ci->fullbooks=_ogg_calloc(ci->books,sizeof(*ci->fullbooks));
+      for(i=0;i<ci->books;i++)
+        vorbis_book_init_encode(ci->fullbooks+i,ci->book_param[i]);
+    }
+
+    b->psy=_ogg_calloc(ci->psys,sizeof(*b->psy));
+    for(i=0;i<ci->psys;i++){
+      _vp_psy_init(b->psy+i,
+                   ci->psy_param[i],
+                   &ci->psy_g_param,
+                   ci->blocksizes[ci->psy_param[i]->blockflag]/2,
+                   vi->rate);
+    }
+
+    v->analysisp=1;
+  }else{
+    /* finish the codebooks */
+    if(!ci->fullbooks){
+      ci->fullbooks=_ogg_calloc(ci->books,sizeof(*ci->fullbooks));
+      for(i=0;i<ci->books;i++){
+        if(ci->book_param[i]==NULL)
+          goto abort_books;
+        if(vorbis_book_init_decode(ci->fullbooks+i,ci->book_param[i]))
+          goto abort_books;
+        /* decode codebooks are now standalone after init */
+        vorbis_staticbook_destroy(ci->book_param[i]);
+        ci->book_param[i]=NULL;
+      }
+    }
+  }
+
+  /* initialize the storage vectors. blocksize[1] is small for encode,
+     but the correct size for decode */
+  v->pcm_storage=ci->blocksizes[1];
+  v->pcm=_ogg_malloc(vi->channels*sizeof(*v->pcm));
+  v->pcmret=_ogg_malloc(vi->channels*sizeof(*v->pcmret));
+  {
+    int i;
+    for(i=0;i<vi->channels;i++)
+      v->pcm[i]=_ogg_calloc(v->pcm_storage,sizeof(*v->pcm[i]));
+  }
+
+  /* all 1 (large block) or 0 (small block) */
+  /* explicitly set for the sake of clarity */
+  v->lW=0; /* previous window size */
+  v->W=0;  /* current window size */
+
+  /* all vector indexes */
+  v->centerW=ci->blocksizes[1]/2;
+
+  v->pcm_current=v->centerW;
+
+  /* initialize all the backend lookups */
+  b->flr=_ogg_calloc(ci->floors,sizeof(*b->flr));
+  b->residue=_ogg_calloc(ci->residues,sizeof(*b->residue));
+
+  for(i=0;i<ci->floors;i++)
+    b->flr[i]=_floor_P[ci->floor_type[i]]->
+      look(v,ci->floor_param[i]);
+
+  for(i=0;i<ci->residues;i++)
+    b->residue[i]=_residue_P[ci->residue_type[i]]->
+      look(v,ci->residue_param[i]);
+
+  return 0;
+ abort_books:
+  for(i=0;i<ci->books;i++){
+    if(ci->book_param[i]!=NULL){
+      vorbis_staticbook_destroy(ci->book_param[i]);
+      ci->book_param[i]=NULL;
+    }
+  }
+  vorbis_dsp_clear(v);
+  return -1;
+}
+
+/* arbitrary settings and spec-mandated numbers get filled in here */
+int vorbis_analysis_init(vorbis_dsp_state *v,vorbis_info *vi){
+  private_state *b=NULL;
+
+  if(_vds_shared_init(v,vi,1))return 1;
+  b=v->backend_state;
+  b->psy_g_look=_vp_global_look(vi);
+
+  /* Initialize the envelope state storage */
+  b->ve=_ogg_calloc(1,sizeof(*b->ve));
+  _ve_envelope_init(b->ve,vi);
+
+  vorbis_bitrate_init(vi,&b->bms);
+
+  /* compressed audio packets start after the headers
+     with sequence number 3 */
+  v->sequence=3;
+
+  return(0);
+}
+
+void vorbis_dsp_clear(vorbis_dsp_state *v){
+  int i;
+  if(v){
+    vorbis_info *vi=v->vi;
+    codec_setup_info *ci=(vi?vi->codec_setup:NULL);
+    private_state *b=v->backend_state;
+
+    if(b){
+
+      if(b->ve){
+        _ve_envelope_clear(b->ve);
+        _ogg_free(b->ve);
+      }
+
+      if(b->transform[0]){
+        mdct_clear(b->transform[0][0]);
+        _ogg_free(b->transform[0][0]);
+        _ogg_free(b->transform[0]);
+      }
+      if(b->transform[1]){
+        mdct_clear(b->transform[1][0]);
+        _ogg_free(b->transform[1][0]);
+        _ogg_free(b->transform[1]);
+      }
+
+      if(b->flr){
+        if(ci)
+          for(i=0;i<ci->floors;i++)
+            _floor_P[ci->floor_type[i]]->
+              free_look(b->flr[i]);
+        _ogg_free(b->flr);
+      }
+      if(b->residue){
+        if(ci)
+          for(i=0;i<ci->residues;i++)
+            _residue_P[ci->residue_type[i]]->
+              free_look(b->residue[i]);
+        _ogg_free(b->residue);
+      }
+      if(b->psy){
+        if(ci)
+          for(i=0;i<ci->psys;i++)
+            _vp_psy_clear(b->psy+i);
+        _ogg_free(b->psy);
+      }
+
+      if(b->psy_g_look)_vp_global_free(b->psy_g_look);
+      vorbis_bitrate_clear(&b->bms);
+
+      drft_clear(&b->fft_look[0]);
+      drft_clear(&b->fft_look[1]);
+
+    }
+
+    if(v->pcm){
+      if(vi)
+        for(i=0;i<vi->channels;i++)
+          if(v->pcm[i])_ogg_free(v->pcm[i]);
+      _ogg_free(v->pcm);
+      if(v->pcmret)_ogg_free(v->pcmret);
+    }
+
+    if(b){
+      /* free header, header1, header2 */
+      if(b->header)_ogg_free(b->header);
+      if(b->header1)_ogg_free(b->header1);
+      if(b->header2)_ogg_free(b->header2);
+      _ogg_free(b);
+    }
+
+    memset(v,0,sizeof(*v));
+  }
+}
+
+float **vorbis_analysis_buffer(vorbis_dsp_state *v, int vals){
+  int i;
+  vorbis_info *vi=v->vi;
+  private_state *b=v->backend_state;
+
+  /* free header, header1, header2 */
+  if(b->header)_ogg_free(b->header);b->header=NULL;
+  if(b->header1)_ogg_free(b->header1);b->header1=NULL;
+  if(b->header2)_ogg_free(b->header2);b->header2=NULL;
+
+  /* Do we have enough storage space for the requested buffer? If not,
+     expand the PCM (and envelope) storage */
+
+  if(v->pcm_current+vals>=v->pcm_storage){
+    v->pcm_storage=v->pcm_current+vals*2;
+
+    for(i=0;i<vi->channels;i++){
+      v->pcm[i]=_ogg_realloc(v->pcm[i],v->pcm_storage*sizeof(*v->pcm[i]));
+    }
+  }
+
+  for(i=0;i<vi->channels;i++)
+    v->pcmret[i]=v->pcm[i]+v->pcm_current;
+
+  return(v->pcmret);
+}
+
+static void _preextrapolate_helper(vorbis_dsp_state *v){
+  int i;
+  int order=16;
+  float *lpc=alloca(order*sizeof(*lpc));
+  float *work=alloca(v->pcm_current*sizeof(*work));
+  long j;
+  v->preextrapolate=1;
+
+  if(v->pcm_current-v->centerW>order*2){ /* safety */
+    for(i=0;i<v->vi->channels;i++){
+      /* need to run the extrapolation in reverse! */
+      for(j=0;j<v->pcm_current;j++)
+        work[j]=v->pcm[i][v->pcm_current-j-1];
+
+      /* prime as above */
+      vorbis_lpc_from_data(work,lpc,v->pcm_current-v->centerW,order);
+
+#if 0
+      if(v->vi->channels==2){
+        if(i==0)
+          _analysis_output("predataL",0,work,v->pcm_current-v->centerW,0,0,0);
+        else
+          _analysis_output("predataR",0,work,v->pcm_current-v->centerW,0,0,0);
+      }else{
+        _analysis_output("predata",0,work,v->pcm_current-v->centerW,0,0,0);
+      }
+#endif
+
+      /* run the predictor filter */
+      vorbis_lpc_predict(lpc,work+v->pcm_current-v->centerW-order,
+                         order,
+                         work+v->pcm_current-v->centerW,
+                         v->centerW);
+
+      for(j=0;j<v->pcm_current;j++)
+        v->pcm[i][v->pcm_current-j-1]=work[j];
+
+    }
+  }
+}
+
+
+/* call with val<=0 to set eof */
+
+int vorbis_analysis_wrote(vorbis_dsp_state *v, int vals){
+  vorbis_info *vi=v->vi;
+  codec_setup_info *ci=vi->codec_setup;
+
+  if(vals<=0){
+    int order=32;
+    int i;
+    float *lpc=alloca(order*sizeof(*lpc));
+
+    /* if it wasn't done earlier (very short sample) */
+    if(!v->preextrapolate)
+      _preextrapolate_helper(v);
+
+    /* We're encoding the end of the stream.  Just make sure we have
+       [at least] a few full blocks of zeroes at the end. */
+    /* actually, we don't want zeroes; that could drop a large
+       amplitude off a cliff, creating spread spectrum noise that will
+       suck to encode.  Extrapolate for the sake of cleanliness. */
+
+    vorbis_analysis_buffer(v,ci->blocksizes[1]*3);
+    v->eofflag=v->pcm_current;
+    v->pcm_current+=ci->blocksizes[1]*3;
+
+    for(i=0;i<vi->channels;i++){
+      if(v->eofflag>order*2){
+        /* extrapolate with LPC to fill in */
+        long n;
+
+        /* make a predictor filter */
+        n=v->eofflag;
+        if(n>ci->blocksizes[1])n=ci->blocksizes[1];
+        vorbis_lpc_from_data(v->pcm[i]+v->eofflag-n,lpc,n,order);
+
+        /* run the predictor filter */
+        vorbis_lpc_predict(lpc,v->pcm[i]+v->eofflag-order,order,
+                           v->pcm[i]+v->eofflag,v->pcm_current-v->eofflag);
+      }else{
+        /* not enough data to extrapolate (unlikely to happen due to
+           guarding the overlap, but bulletproof in case that
+           assumtion goes away). zeroes will do. */
+        memset(v->pcm[i]+v->eofflag,0,
+               (v->pcm_current-v->eofflag)*sizeof(*v->pcm[i]));
+
+      }
+    }
+  }else{
+
+    if(v->pcm_current+vals>v->pcm_storage)
+      return(OV_EINVAL);
+
+    v->pcm_current+=vals;
+
+    /* we may want to reverse extrapolate the beginning of a stream
+       too... in case we're beginning on a cliff! */
+    /* clumsy, but simple.  It only runs once, so simple is good. */
+    if(!v->preextrapolate && v->pcm_current-v->centerW>ci->blocksizes[1])
+      _preextrapolate_helper(v);
+
+  }
+  return(0);
+}
+
+/* do the deltas, envelope shaping, pre-echo and determine the size of
+   the next block on which to continue analysis */
+int vorbis_analysis_blockout(vorbis_dsp_state *v,vorbis_block *vb){
+  int i;
+  vorbis_info *vi=v->vi;
+  codec_setup_info *ci=vi->codec_setup;
+  private_state *b=v->backend_state;
+  vorbis_look_psy_global *g=b->psy_g_look;
+  long beginW=v->centerW-ci->blocksizes[v->W]/2,centerNext;
+  vorbis_block_internal *vbi=(vorbis_block_internal *)vb->internal;
+
+  /* check to see if we're started... */
+  if(!v->preextrapolate)return(0);
+
+  /* check to see if we're done... */
+  if(v->eofflag==-1)return(0);
+
+  /* By our invariant, we have lW, W and centerW set.  Search for
+     the next boundary so we can determine nW (the next window size)
+     which lets us compute the shape of the current block's window */
+
+  /* we do an envelope search even on a single blocksize; we may still
+     be throwing more bits at impulses, and envelope search handles
+     marking impulses too. */
+  {
+    long bp=_ve_envelope_search(v);
+    if(bp==-1){
+
+      if(v->eofflag==0)return(0); /* not enough data currently to search for a
+                                     full long block */
+      v->nW=0;
+    }else{
+
+      if(ci->blocksizes[0]==ci->blocksizes[1])
+        v->nW=0;
+      else
+        v->nW=bp;
+    }
+  }
+
+  centerNext=v->centerW+ci->blocksizes[v->W]/4+ci->blocksizes[v->nW]/4;
+
+  {
+    /* center of next block + next block maximum right side. */
+
+    long blockbound=centerNext+ci->blocksizes[v->nW]/2;
+    if(v->pcm_current<blockbound)return(0); /* not enough data yet;
+                                               although this check is
+                                               less strict that the
+                                               _ve_envelope_search,
+                                               the search is not run
+                                               if we only use one
+                                               block size */
+
+
+  }
+
+  /* fill in the block.  Note that for a short window, lW and nW are *short*
+     regardless of actual settings in the stream */
+
+  _vorbis_block_ripcord(vb);
+  vb->lW=v->lW;
+  vb->W=v->W;
+  vb->nW=v->nW;
+
+  if(v->W){
+    if(!v->lW || !v->nW){
+      vbi->blocktype=BLOCKTYPE_TRANSITION;
+      /*fprintf(stderr,"-");*/
+    }else{
+      vbi->blocktype=BLOCKTYPE_LONG;
+      /*fprintf(stderr,"_");*/
+    }
+  }else{
+    if(_ve_envelope_mark(v)){
+      vbi->blocktype=BLOCKTYPE_IMPULSE;
+      /*fprintf(stderr,"|");*/
+
+    }else{
+      vbi->blocktype=BLOCKTYPE_PADDING;
+      /*fprintf(stderr,".");*/
+
+    }
+  }
+
+  vb->vd=v;
+  vb->sequence=v->sequence++;
+  vb->granulepos=v->granulepos;
+  vb->pcmend=ci->blocksizes[v->W];
+
+  /* copy the vectors; this uses the local storage in vb */
+
+  /* this tracks 'strongest peak' for later psychoacoustics */
+  /* moved to the global psy state; clean this mess up */
+  if(vbi->ampmax>g->ampmax)g->ampmax=vbi->ampmax;
+  g->ampmax=_vp_ampmax_decay(g->ampmax,v);
+  vbi->ampmax=g->ampmax;
+
+  vb->pcm=_vorbis_block_alloc(vb,sizeof(*vb->pcm)*vi->channels);
+  vbi->pcmdelay=_vorbis_block_alloc(vb,sizeof(*vbi->pcmdelay)*vi->channels);
+  for(i=0;i<vi->channels;i++){
+    vbi->pcmdelay[i]=
+      _vorbis_block_alloc(vb,(vb->pcmend+beginW)*sizeof(*vbi->pcmdelay[i]));
+    memcpy(vbi->pcmdelay[i],v->pcm[i],(vb->pcmend+beginW)*sizeof(*vbi->pcmdelay[i]));
+    vb->pcm[i]=vbi->pcmdelay[i]+beginW;
+
+    /* before we added the delay
+       vb->pcm[i]=_vorbis_block_alloc(vb,vb->pcmend*sizeof(*vb->pcm[i]));
+       memcpy(vb->pcm[i],v->pcm[i]+beginW,ci->blocksizes[v->W]*sizeof(*vb->pcm[i]));
+    */
+
+  }
+
+  /* handle eof detection: eof==0 means that we've not yet received EOF
+                           eof>0  marks the last 'real' sample in pcm[]
+                           eof<0  'no more to do'; doesn't get here */
+
+  if(v->eofflag){
+    if(v->centerW>=v->eofflag){
+      v->eofflag=-1;
+      vb->eofflag=1;
+      return(1);
+    }
+  }
+
+  /* advance storage vectors and clean up */
+  {
+    int new_centerNext=ci->blocksizes[1]/2;
+    int movementW=centerNext-new_centerNext;
+
+    if(movementW>0){
+
+      _ve_envelope_shift(b->ve,movementW);
+      v->pcm_current-=movementW;
+
+      for(i=0;i<vi->channels;i++)
+        memmove(v->pcm[i],v->pcm[i]+movementW,
+                v->pcm_current*sizeof(*v->pcm[i]));
+
+
+      v->lW=v->W;
+      v->W=v->nW;
+      v->centerW=new_centerNext;
+
+      if(v->eofflag){
+        v->eofflag-=movementW;
+        if(v->eofflag<=0)v->eofflag=-1;
+        /* do not add padding to end of stream! */
+        if(v->centerW>=v->eofflag){
+          v->granulepos+=movementW-(v->centerW-v->eofflag);
+        }else{
+          v->granulepos+=movementW;
+        }
+      }else{
+        v->granulepos+=movementW;
+      }
+    }
+  }
+
+  /* done */
+  return(1);
+}
+
+int vorbis_synthesis_restart(vorbis_dsp_state *v){
+  vorbis_info *vi=v->vi;
+  codec_setup_info *ci;
+  int hs;
+
+  if(!v->backend_state)return -1;
+  if(!vi)return -1;
+  ci=vi->codec_setup;
+  if(!ci)return -1;
+  hs=ci->halfrate_flag;
+
+  v->centerW=ci->blocksizes[1]>>(hs+1);
+  v->pcm_current=v->centerW>>hs;
+
+  v->pcm_returned=-1;
+  v->granulepos=-1;
+  v->sequence=-1;
+  v->eofflag=0;
+  ((private_state *)(v->backend_state))->sample_count=-1;
+
+  return(0);
+}
+
+int vorbis_synthesis_init(vorbis_dsp_state *v,vorbis_info *vi){
+  if(_vds_shared_init(v,vi,0)){
+    vorbis_dsp_clear(v);
+    return 1;
+  }
+  vorbis_synthesis_restart(v);
+  return 0;
+}
+
+/* Unlike in analysis, the window is only partially applied for each
+   block.  The time domain envelope is not yet handled at the point of
+   calling (as it relies on the previous block). */
+
+int vorbis_synthesis_blockin(vorbis_dsp_state *v,vorbis_block *vb){
+  vorbis_info *vi=v->vi;
+  codec_setup_info *ci=vi->codec_setup;
+  private_state *b=v->backend_state;
+  int hs=ci->halfrate_flag;
+  int i,j;
+
+  if(!vb)return(OV_EINVAL);
+  if(v->pcm_current>v->pcm_returned  && v->pcm_returned!=-1)return(OV_EINVAL);
+
+  v->lW=v->W;
+  v->W=vb->W;
+  v->nW=-1;
+
+  if((v->sequence==-1)||
+     (v->sequence+1 != vb->sequence)){
+    v->granulepos=-1; /* out of sequence; lose count */
+    b->sample_count=-1;
+  }
+
+  v->sequence=vb->sequence;
+
+  if(vb->pcm){  /* no pcm to process if vorbis_synthesis_trackonly
+                   was called on block */
+    int n=ci->blocksizes[v->W]>>(hs+1);
+    int n0=ci->blocksizes[0]>>(hs+1);
+    int n1=ci->blocksizes[1]>>(hs+1);
+
+    int thisCenter;
+    int prevCenter;
+
+    v->glue_bits+=vb->glue_bits;
+    v->time_bits+=vb->time_bits;
+    v->floor_bits+=vb->floor_bits;
+    v->res_bits+=vb->res_bits;
+
+    if(v->centerW){
+      thisCenter=n1;
+      prevCenter=0;
+    }else{
+      thisCenter=0;
+      prevCenter=n1;
+    }
+
+    /* v->pcm is now used like a two-stage double buffer.  We don't want
+       to have to constantly shift *or* adjust memory usage.  Don't
+       accept a new block until the old is shifted out */
+
+    for(j=0;j<vi->channels;j++){
+      /* the overlap/add section */
+      if(v->lW){
+        if(v->W){
+          /* large/large */
+          float *w=_vorbis_window_get(b->window[1]-hs);
+          float *pcm=v->pcm[j]+prevCenter;
+          float *p=vb->pcm[j];
+          for(i=0;i<n1;i++)
+            pcm[i]=pcm[i]*w[n1-i-1] + p[i]*w[i];
+        }else{
+          /* large/small */
+          float *w=_vorbis_window_get(b->window[0]-hs);
+          float *pcm=v->pcm[j]+prevCenter+n1/2-n0/2;
+          float *p=vb->pcm[j];
+          for(i=0;i<n0;i++)
+            pcm[i]=pcm[i]*w[n0-i-1] +p[i]*w[i];
+        }
+      }else{
+        if(v->W){
+          /* small/large */
+          float *w=_vorbis_window_get(b->window[0]-hs);
+          float *pcm=v->pcm[j]+prevCenter;
+          float *p=vb->pcm[j]+n1/2-n0/2;
+          for(i=0;i<n0;i++)
+            pcm[i]=pcm[i]*w[n0-i-1] +p[i]*w[i];
+          for(;i<n1/2+n0/2;i++)
+            pcm[i]=p[i];
+        }else{
+          /* small/small */
+          float *w=_vorbis_window_get(b->window[0]-hs);
+          float *pcm=v->pcm[j]+prevCenter;
+          float *p=vb->pcm[j];
+          for(i=0;i<n0;i++)
+            pcm[i]=pcm[i]*w[n0-i-1] +p[i]*w[i];
+        }
+      }
+
+      /* the copy section */
+      {
+        float *pcm=v->pcm[j]+thisCenter;
+        float *p=vb->pcm[j]+n;
+        for(i=0;i<n;i++)
+          pcm[i]=p[i];
+      }
+    }
+
+    if(v->centerW)
+      v->centerW=0;
+    else
+      v->centerW=n1;
+
+    /* deal with initial packet state; we do this using the explicit
+       pcm_returned==-1 flag otherwise we're sensitive to first block
+       being short or long */
+
+    if(v->pcm_returned==-1){
+      v->pcm_returned=thisCenter;
+      v->pcm_current=thisCenter;
+    }else{
+      v->pcm_returned=prevCenter;
+      v->pcm_current=prevCenter+
+        ((ci->blocksizes[v->lW]/4+
+        ci->blocksizes[v->W]/4)>>hs);
+    }
+
+  }
+
+  /* track the frame number... This is for convenience, but also
+     making sure our last packet doesn't end with added padding.  If
+     the last packet is partial, the number of samples we'll have to
+     return will be past the vb->granulepos.
+
+     This is not foolproof!  It will be confused if we begin
+     decoding at the last page after a seek or hole.  In that case,
+     we don't have a starting point to judge where the last frame
+     is.  For this reason, vorbisfile will always try to make sure
+     it reads the last two marked pages in proper sequence */
+
+  if(b->sample_count==-1){
+    b->sample_count=0;
+  }else{
+    b->sample_count+=ci->blocksizes[v->lW]/4+ci->blocksizes[v->W]/4;
+  }
+
+  if(v->granulepos==-1){
+    if(vb->granulepos!=-1){ /* only set if we have a position to set to */
+
+      v->granulepos=vb->granulepos;
+
+      /* is this a short page? */
+      if(b->sample_count>v->granulepos){
+        /* corner case; if this is both the first and last audio page,
+           then spec says the end is cut, not beginning */
+       long extra=b->sample_count-vb->granulepos;
+
+        /* we use ogg_int64_t for granule positions because a
+           uint64 isn't universally available.  Unfortunately,
+           that means granposes can be 'negative' and result in
+           extra being negative */
+        if(extra<0)
+          extra=0;
+
+        if(vb->eofflag){
+          /* trim the end */
+          /* no preceding granulepos; assume we started at zero (we'd
+             have to in a short single-page stream) */
+          /* granulepos could be -1 due to a seek, but that would result
+             in a long count, not short count */
+
+          /* Guard against corrupt/malicious frames that set EOP and
+             a backdated granpos; don't rewind more samples than we
+             actually have */
+          if(extra > (v->pcm_current - v->pcm_returned)<<hs)
+            extra = (v->pcm_current - v->pcm_returned)<<hs;
+
+          v->pcm_current-=extra>>hs;
+        }else{
+          /* trim the beginning */
+          v->pcm_returned+=extra>>hs;
+          if(v->pcm_returned>v->pcm_current)
+            v->pcm_returned=v->pcm_current;
+        }
+
+      }
+
+    }
+  }else{
+    v->granulepos+=ci->blocksizes[v->lW]/4+ci->blocksizes[v->W]/4;
+    if(vb->granulepos!=-1 && v->granulepos!=vb->granulepos){
+
+      if(v->granulepos>vb->granulepos){
+        long extra=v->granulepos-vb->granulepos;
+
+        if(extra)
+          if(vb->eofflag){
+            /* partial last frame.  Strip the extra samples off */
+
+            /* Guard against corrupt/malicious frames that set EOP and
+               a backdated granpos; don't rewind more samples than we
+               actually have */
+            if(extra > (v->pcm_current - v->pcm_returned)<<hs)
+              extra = (v->pcm_current - v->pcm_returned)<<hs;
+
+            /* we use ogg_int64_t for granule positions because a
+               uint64 isn't universally available.  Unfortunately,
+               that means granposes can be 'negative' and result in
+               extra being negative */
+            if(extra<0)
+              extra=0;
+
+            v->pcm_current-=extra>>hs;
+          } /* else {Shouldn't happen *unless* the bitstream is out of
+               spec.  Either way, believe the bitstream } */
+      } /* else {Shouldn't happen *unless* the bitstream is out of
+           spec.  Either way, believe the bitstream } */
+      v->granulepos=vb->granulepos;
+    }
+  }
+
+  /* Update, cleanup */
+
+  if(vb->eofflag)v->eofflag=1;
+  return(0);
+
+}
+
+/* pcm==NULL indicates we just want the pending samples, no more */
+int vorbis_synthesis_pcmout(vorbis_dsp_state *v,float ***pcm){
+  vorbis_info *vi=v->vi;
+
+  if(v->pcm_returned>-1 && v->pcm_returned<v->pcm_current){
+    if(pcm){
+      int i;
+      for(i=0;i<vi->channels;i++)
+        v->pcmret[i]=v->pcm[i]+v->pcm_returned;
+      *pcm=v->pcmret;
+    }
+    return(v->pcm_current-v->pcm_returned);
+  }
+  return(0);
+}
+
+int vorbis_synthesis_read(vorbis_dsp_state *v,int n){
+  if(n && v->pcm_returned+n>v->pcm_current)return(OV_EINVAL);
+  v->pcm_returned+=n;
+  return(0);
+}
+
+/* intended for use with a specific vorbisfile feature; we want access
+   to the [usually synthetic/postextrapolated] buffer and lapping at
+   the end of a decode cycle, specifically, a half-short-block worth.
+   This funtion works like pcmout above, except it will also expose
+   this implicit buffer data not normally decoded. */
+int vorbis_synthesis_lapout(vorbis_dsp_state *v,float ***pcm){
+  vorbis_info *vi=v->vi;
+  codec_setup_info *ci=vi->codec_setup;
+  int hs=ci->halfrate_flag;
+
+  int n=ci->blocksizes[v->W]>>(hs+1);
+  int n0=ci->blocksizes[0]>>(hs+1);
+  int n1=ci->blocksizes[1]>>(hs+1);
+  int i,j;
+
+  if(v->pcm_returned<0)return 0;
+
+  /* our returned data ends at pcm_returned; because the synthesis pcm
+     buffer is a two-fragment ring, that means our data block may be
+     fragmented by buffering, wrapping or a short block not filling
+     out a buffer.  To simplify things, we unfragment if it's at all
+     possibly needed. Otherwise, we'd need to call lapout more than
+     once as well as hold additional dsp state.  Opt for
+     simplicity. */
+
+  /* centerW was advanced by blockin; it would be the center of the
+     *next* block */
+  if(v->centerW==n1){
+    /* the data buffer wraps; swap the halves */
+    /* slow, sure, small */
+    for(j=0;j<vi->channels;j++){
+      float *p=v->pcm[j];
+      for(i=0;i<n1;i++){
+        float temp=p[i];
+        p[i]=p[i+n1];
+        p[i+n1]=temp;
+      }
+    }
+
+    v->pcm_current-=n1;
+    v->pcm_returned-=n1;
+    v->centerW=0;
+  }
+
+  /* solidify buffer into contiguous space */
+  if((v->lW^v->W)==1){
+    /* long/short or short/long */
+    for(j=0;j<vi->channels;j++){
+      float *s=v->pcm[j];
+      float *d=v->pcm[j]+(n1-n0)/2;
+      for(i=(n1+n0)/2-1;i>=0;--i)
+        d[i]=s[i];
+    }
+    v->pcm_returned+=(n1-n0)/2;
+    v->pcm_current+=(n1-n0)/2;
+  }else{
+    if(v->lW==0){
+      /* short/short */
+      for(j=0;j<vi->channels;j++){
+        float *s=v->pcm[j];
+        float *d=v->pcm[j]+n1-n0;
+        for(i=n0-1;i>=0;--i)
+          d[i]=s[i];
+      }
+      v->pcm_returned+=n1-n0;
+      v->pcm_current+=n1-n0;
+    }
+  }
+
+  if(pcm){
+    int i;
+    for(i=0;i<vi->channels;i++)
+      v->pcmret[i]=v->pcm[i]+v->pcm_returned;
+    *pcm=v->pcmret;
+  }
+
+  return(n1+n-v->pcm_returned);
+
+}
+
+float *vorbis_window(vorbis_dsp_state *v,int W){
+  vorbis_info *vi=v->vi;
+  codec_setup_info *ci=vi->codec_setup;
+  int hs=ci->halfrate_flag;
+  private_state *b=v->backend_state;
+
+  if(b->window[W]-1<0)return NULL;
+  return _vorbis_window_get(b->window[W]-hs);
+}