start to use code from the original resample codebase, since it uses fixed point math instead of float; however, the code is not at all complete right now, I just commit this to get it off my HD (neither the old nor the new code in resample.cpp work anyway)

svn-id: r10089

1 files changed, 625 insertions, 473 deletions

diff --git a/sound/resample.cpp b/sound/resample.cpp
index 99deeaf3aa..5a772389fb 100644
--- a/sound/resample.cpp
+++ b/sound/resample.cpp
@@ -1,156 +1,578 @@
-/*
-* July 5, 1991
-* Copyright 1991 Lance Norskog And Sundry Contributors
-* This source code is freely redistributable and may be used for
-* any purpose.	 This copyright notice must be maintained. 
-* Lance Norskog And Sundry Contributors are not responsible for 
-* the consequences of using this software.
-*/
 
-/*
- * Sound Tools rate change effect file.
- * Spiffy rate changer using Smith & Wesson Bandwidth-Limited Interpolation.
- * The algorithm is described in "Bandlimited Interpolation -
- * Introduction and Algorithm" by Julian O. Smith III.
- * Available on ccrma-ftp.stanford.edu as
- * pub/BandlimitedInterpolation.eps.Z or similar.
+#include "stdafx.h"
+#include <math.h>
+#include "sound/resample.h"
+#include "sound/audiostream.h"
+
+
+#pragma mark -
+
+
+
+/**
+ * Calculates the filter coeffs for a Kaiser-windowed low-pass filter with a
+ * given roll-off frequency. These coeffs are stored into a array of doubles.
+ *
+ * reference: "Digital Filters, 2nd edition"
+ *            R.W. Hamming, pp. 178-179
+ *
+ * LpFilter() computes the coeffs of a Kaiser-windowed low pass filter with
+ *    the following characteristics:
  *
- * The latest stand alone version of this algorithm can be found
- * at ftp://ccrma-ftp.stanford.edu/pub/NeXT/
- * under the name of resample-version.number.tar.Z
+ *       c[]  = array in which to store computed coeffs
+ *       frq  = roll-off frequency of filter
+ *       N    = Half the window length in number of coeffs
+ *       Beta = parameter of Kaiser window
+ *       Num  = number of coeffs before 1/frq
  *
- * NOTE: There is a newer version of the resample routine then what
- * this file was originally based on.  Those adventurous might be
- * interested in reviewing its improvesments and porting it to this
- * version.
+ * Beta trades the rejection of the lowpass filter against the transition
+ *    width from passband to stopband.  Larger Beta means a slower
+ *    transition and greater stopband rejection.  See Rabiner and Gold
+ *    (Theory and Application of DSP) under Kaiser windows for more about
+ *    Beta.  The following table from Rabiner and Gold gives some feel
+ *    for the effect of Beta:
+ *
+ * All ripples in dB, width of transition band = D*N where N = window length
+ *
+ *               BETA    D       PB RIP   SB RIP
+ *               2.120   1.50  +-0.27      -30
+ *               3.384   2.23    0.0864    -40
+ *               4.538   2.93    0.0274    -50
+ *               5.658   3.62    0.00868   -60
+ *               6.764   4.32    0.00275   -70
+ *               7.865   5.0     0.000868  -80
+ *               8.960   5.7     0.000275  -90
+ *               10.056  6.4     0.000087  -100
  */
+static void LpFilter(double c[], int N, double frq, double Beta, int Num);
+
+/**
+ * Calls LpFilter() to create a filter, then scales the double coeffs into an
+ * array of half words.
+ * ERROR return codes:
+ *    0 - no error
+ *    1 - Nwing too large (Nwing is > MAXNWING)
+ *    2 - Froll is not in interval [0:1)
+ *    3 - Beta is < 1.0
+ *    4 - LpScl will not fit in 16-bits
+ */
+static int makeFilter(HWORD Imp[], HWORD ImpD[], UHWORD *LpScl, UHWORD Nwing,
+	       double Froll, double Beta);
+
+static WORD FilterUp(HWORD Imp[], HWORD ImpD[], UHWORD Nwing, bool Interp,
+	      HWORD *Xp, HWORD Inc, HWORD Ph);
+
+static WORD FilterUD(HWORD Imp[], HWORD ImpD[], UHWORD Nwing, bool Interp,
+	      HWORD *Xp, HWORD Ph, HWORD Inc, UHWORD dhb);
+
+
+
+#pragma mark -
 
-/* Fixed bug: roll off frequency was wrong, too high by 2 when upsampling,
- * too low by 2 when downsampling.
- * Andreas Wilde, 12. Feb. 1999, andreas@eakaw2.et.tu-dresden.de
-*/
 
 /*
- * October 29, 1999
- * Various changes, bugfixes(?), increased precision, by Stan Brooks.
  *
- * This source code is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ * The configuration constants below govern
+ * the number of bits in the input sample and filter coefficients, the 
+ * number of bits to the right of the binary-point for fixed-point math, etc.
  *
- */ 
-/*
- * SJB: [11/25/99]
- * TODO: another idea for improvement...
- * note that upsampling usually doesn't require interpolation,
- * therefore is faster and more accurate than downsampling.
- * Downsampling by an integer factor is also simple, since
- * it just involves decimation if the input is already 
- * lowpass-filtered to the output Nyquist freqency.
- * Get the idea? :)
  */
 
-#include "stdafx.h"
-#include <math.h>
-#include "sound/resample.h"
-#include "sound/audiostream.h"
-
-
 /* Conversion constants */
-#define Lc        7
-#define Nc       (1<<Lc)
-#define La        16
-#define Na       (1<<La)
-#define Lp       (Lc+La)
-#define Np       (1<<Lp)
-#define Amask    (Na-1)
-#define Pmask    (Np-1)
-
-#define MAXNWING  (80<<Lc)
+#define Nhc       8
+#define Na        7
+#define Np       (Nhc+Na)
+#define Npc      (1<<Nhc)
+#define Amask    ((1<<Na)-1)
+#define Pmask    ((1<<Np)-1)
+#define Nh       16
+#define Nb       16
+#define Nhxn     14
+#define Nhg      (Nh-Nhxn)
+#define NLpScl   13
+
 /* Description of constants:
  *
- * Nc - is the number of look-up values available for the lowpass filter
+ * Npc - is the number of look-up values available for the lowpass filter
  *    between the beginning of its impulse response and the "cutoff time"
  *    of the filter.  The cutoff time is defined as the reciprocal of the
  *    lowpass-filter cut off frequence in Hz.  For example, if the
- *    lowpass filter were a sinc function, Nc would be the index of the
+ *    lowpass filter were a sinc function, Npc would be the index of the
  *    impulse-response lookup-table corresponding to the first zero-
  *    crossing of the sinc function.  (The inverse first zero-crossing
  *    time of a sinc function equals its nominal cutoff frequency in Hz.)
- *    Nc must be a power of 2 due to the details of the current
- *    implementation. The default value of 128 is sufficiently high that
+ *    Npc must be a power of 2 due to the details of the current
+ *    implementation. The default value of 512 is sufficiently high that
  *    using linear interpolation to fill in between the table entries
- *    gives approximately 16-bit precision, and quadratic interpolation
- *    gives about 23-bit (float) precision in filter coefficients.
+ *    gives approximately 16-bit accuracy in filter coefficients.
  *
- * Lc - is log base 2 of Nc.
+ * Nhc - is log base 2 of Npc.
  *
- * La - is the number of bits devoted to linear interpolation of the
+ * Na - is the number of bits devoted to linear interpolation of the
  *    filter coefficients.
  *
- * Lp - is La + Lc, the number of bits to the right of the binary point
+ * Np - is Na + Nhc, the number of bits to the right of the binary point
  *    in the integer "time" variable. To the left of the point, it indexes
  *    the input array (X), and to the right, it is interpreted as a number
- *    between 0 and 1 sample of the input X.  The default value of 23 is
- *    about right.  There is a constraint that the filter window must be
- *    "addressable" in a int32_t, more precisely, if Nmult is the number
- *    of sinc zero-crossings in the right wing of the filter window, then
- *    (Nwing<<Lp) must be expressible in 31 bits.
+ *    between 0 and 1 sample of the input X.  Np must be less than 16 in
+ *    this implementation.
+ *
+ * Nh - is the number of bits in the filter coefficients. The sum of Nh and
+ *    the number of bits in the input data (typically 16) cannot exceed 32.
+ *    Thus Nh should be 16.  The largest filter coefficient should nearly
+ *    fill 16 bits (32767).
+ *
+ * Nb - is the number of bits in the input data. The sum of Nb and Nh cannot
+ *    exceed 32.
+ *
+ * Nhxn - is the number of bits to right shift after multiplying each input
+ *    sample times a filter coefficient. It can be as great as Nh and as
+ *    small as 0. Nhxn = Nh-2 gives 2 guard bits in the multiply-add
+ *    accumulation.  If Nhxn=0, the accumulation will soon overflow 32 bits.
  *
+ * Nhg - is the number of guard bits in mpy-add accumulation (equal to Nh-Nhxn)
+ *
+ * NLpScl - is the number of bits allocated to the unity-gain normalization
+ *    factor.  The output of the lowpass filter is multiplied by LpScl and
+ *    then right-shifted NLpScl bits. To avoid overflow, we must have 
+ *    Nb+Nhg+NLpScl < 32.
  */
 
 
-/* this Float MUST match that in filter.c */
-#define Float double/*float*/
+#pragma mark -
 
-/* largest factor for which exact-coefficients upsampling will be used */
-#define NQMAX 511
 
-#define BUFFSIZE 8192 /*16384*/	 /* Total I/O buffer size */
+#define IBUFFSIZE 4096                         /* Input buffer size */
+
+static inline HWORD WordToHword(WORD v, int scl)
+{
+    HWORD out;
+
+    v = (v + (1 << (NLpScl-1))) >> NLpScl;	// Round & scale
+
+    if (v>MAX_HWORD) {
+        v = MAX_HWORD;
+    } else if (v < MIN_HWORD) {
+        v = MIN_HWORD;
+    }   
+    out = (HWORD) v;
+    return out;
+}
+
+/* Sampling rate up-conversion only subroutine;
+ * Slightly faster than down-conversion;
+ */
+static int SrcUp(HWORD X[], HWORD Y[], double factor, UWORD *Time,
+                 UHWORD Nx, UHWORD Nwing, UHWORD LpScl,
+                 HWORD Imp[], HWORD ImpD[], bool Interp)
+{
+    HWORD *Xp, *Ystart;
+    WORD v;
+    
+    double dt;                  /* Step through input signal */ 
+    UWORD dtb;                  /* Fixed-point version of Dt */
+    UWORD endTime;              /* When Time reaches EndTime, return to user */
+    
+    dt = 1.0/factor;            /* Output sampling period */
+    dtb = (UWORD)(dt*(1<<Np) + 0.5);     /* Fixed-point representation */
+    
+    Ystart = Y;
+    endTime = *Time + (1<<Np)*(WORD)Nx;
+    while (*Time < endTime)
+    {
+        Xp = &X[*Time>>Np];      /* Ptr to current input sample */
+        /* Perform left-wing inner product */
+        v = FilterUp(Imp, ImpD, Nwing, Interp, Xp, (HWORD)(*Time&Pmask),-1);
+        /* Perform right-wing inner product */
+        v += FilterUp(Imp, ImpD, Nwing, Interp, Xp+1, 
+		      /* previous (triggers warning): (HWORD)((-*Time)&Pmask),1); */
+                      (HWORD)((((*Time)^Pmask)+1)&Pmask),1);
+        v >>= Nhg;              /* Make guard bits */
+        v *= LpScl;             /* Normalize for unity filter gain */
+        *Y++ = WordToHword(v,NLpScl);   /* strip guard bits, deposit output */
+        *Time += dtb;           /* Move to next sample by time increment */
+    }
+    return (Y - Ystart);        /* Return the number of output samples */
+}
+
+
+/* Sampling rate conversion subroutine */
+
+static int SrcUD(HWORD X[], HWORD Y[], double factor, UWORD *Time,
+                 UHWORD Nx, UHWORD Nwing, UHWORD LpScl,
+                 HWORD Imp[], HWORD ImpD[], bool Interp)
+{
+    HWORD *Xp, *Ystart;
+    WORD v;
+    
+    double dh;                  /* Step through filter impulse response */
+    double dt;                  /* Step through input signal */
+    UWORD endTime;              /* When Time reaches EndTime, return to user */
+    UWORD dhb, dtb;             /* Fixed-point versions of Dh,Dt */
+    
+    dt = 1.0/factor;            /* Output sampling period */
+    dtb = (UWORD)(dt*(1<<Np) + 0.5);     /* Fixed-point representation */
+    
+    dh = MIN((double)Npc, factor*Npc);  /* Filter sampling period */
+    dhb = (UWORD)(dh*(1<<Na) + 0.5);     /* Fixed-point representation */
+    
+    Ystart = Y;
+    endTime = *Time + (1<<Np)*(WORD)Nx;
+    while (*Time < endTime)
+    {
+        Xp = &X[*Time>>Np];     /* Ptr to current input sample */
+        v = FilterUD(Imp, ImpD, Nwing, Interp, Xp, (HWORD)(*Time&Pmask),
+                     -1, dhb);  /* Perform left-wing inner product */
+        v += FilterUD(Imp, ImpD, Nwing, Interp, Xp+1, 
+		      /* previous (triggers warning): (HWORD)((-*Time)&Pmask), */
+                      (HWORD)((((*Time)^Pmask)+1)&Pmask),
+                      1, dhb);  /* Perform right-wing inner product */
+        v >>= Nhg;              /* Make guard bits */
+        v *= LpScl;             /* Normalize for unity filter gain */
+        *Y++ = WordToHword(v,NLpScl);   /* strip guard bits, deposit output */
+        *Time += dtb;           /* Move to next sample by time increment */
+    }
+    return (Y - Ystart);        /* Return the number of output samples */
+}
+
+
+#pragma mark -
+
+
+#define IzeroEPSILON 1E-21               /* Max error acceptable in Izero */
+
+static double Izero(double x)
+{
+   double sum, u, halfx, temp;
+   int n;
+
+   sum = u = n = 1;
+   halfx = x/2.0;
+   do {
+      temp = halfx/(double)n;
+      n += 1;
+      temp *= temp;
+      u *= temp;
+      sum += u;
+      } while (u >= IzeroEPSILON*sum);
+   return(sum);
+}
+
+
+void LpFilter(double c[], int N, double frq, double Beta, int Num)
+{
+   double IBeta, temp, inm1;
+   int i;
+
+   /* Calculate ideal lowpass filter impulse response coefficients: */
+   c[0] = 2.0*frq;
+   for (i=1; i<N; i++) {
+       temp = M_PI*(double)i/(double)Num;
+       c[i] = sin(2.0*temp*frq)/temp; /* Analog sinc function, cutoff = frq */
+   }
+
+   /* 
+    * Calculate and Apply Kaiser window to ideal lowpass filter.
+    * Note: last window value is IBeta which is NOT zero.
+    * You're supposed to really truncate the window here, not ramp
+    * it to zero. This helps reduce the first sidelobe. 
+    */
+   IBeta = 1.0/Izero(Beta);
+   inm1 = 1.0/((double)(N-1));
+   for (i=1; i<N; i++) {
+       temp = (double)i * inm1;
+       c[i] *= Izero(Beta*sqrt(1.0-temp*temp)) * IBeta;
+   }
+}
+
+static double ImpR[MAXNWING];
+
+int makeFilter(HWORD Imp[], HWORD ImpD[], UHWORD *LpScl, UHWORD Nwing,
+	       double Froll, double Beta)
+{
+   double DCgain, Scl, Maxh;
+   HWORD Dh;
+   int i, temp;
+
+   if (Nwing > MAXNWING)                      /* Check for valid parameters */
+      return(1);
+   if ((Froll<=0) || (Froll>1))
+      return(2);
+   if (Beta < 1)
+      return(3);
+
+   /* 
+    * Design Kaiser-windowed sinc-function low-pass filter 
+    */
+   LpFilter(ImpR, (int)Nwing, 0.5*Froll, Beta, Npc); 
+
+   /* Compute the DC gain of the lowpass filter, and its maximum coefficient
+    * magnitude. Scale the coefficients so that the maximum coeffiecient just
+    * fits in Nh-bit fixed-point, and compute LpScl as the NLpScl-bit (signed)
+    * scale factor which when multiplied by the output of the lowpass filter
+    * gives unity gain. */
+   DCgain = 0;
+   Dh = Npc;                       /* Filter sampling period for factors>=1 */
+   for (i=Dh; i<Nwing; i+=Dh)
+      DCgain += ImpR[i];
+   DCgain = 2*DCgain + ImpR[0];    /* DC gain of real coefficients */
+
+   for (Maxh=i=0; i<Nwing; i++)
+      Maxh = MAX(Maxh, fabs(ImpR[i]));
+
+   Scl = ((1<<(Nh-1))-1)/Maxh;     /* Map largest coeff to 16-bit maximum */
+   temp = (int)fabs((1<<(NLpScl+Nh))/(DCgain*Scl));
+   if (temp >= 1<<16)
+      return(4);                   /* Filter scale factor overflows UHWORD */
+   *LpScl = temp;
+
+   /* Scale filter coefficients for Nh bits and convert to integer */
+   if (ImpR[0] < 0)                /* Need pos 1st value for LpScl storage */
+      Scl = -Scl;
+   for (i=0; i<Nwing; i++)         /* Scale them */
+      ImpR[i] *= Scl;
+   for (i=0; i<Nwing; i++)         /* Round them */
+      Imp[i] = (HWORD)(ImpR[i] + 0.5);
+
+   /* ImpD makes linear interpolation of the filter coefficients faster */
+   for (i=0; i<Nwing-1; i++)
+      ImpD[i] = Imp[i+1] - Imp[i];
+   ImpD[Nwing-1] = - Imp[Nwing-1];      /* Last coeff. not interpolated */
+
+   return(0);
+}
+
+
+#pragma mark -
+
+
+WORD FilterUp(HWORD Imp[], HWORD ImpD[], 
+		     UHWORD Nwing, bool Interp,
+		     HWORD *Xp, HWORD Ph, HWORD Inc)
+{
+    HWORD *Hp, *Hdp = NULL, *End;
+    HWORD a = 0;
+    WORD v, t;
+    
+    v=0;
+    Hp = &Imp[Ph>>Na];
+    End = &Imp[Nwing];
+    if (Interp) {
+	Hdp = &ImpD[Ph>>Na];
+	a = Ph & Amask;
+    }
+    if (Inc == 1)		/* If doing right wing...              */
+    {				/* ...drop extra coeff, so when Ph is  */
+	End--;			/*    0.5, we don't do too many mult's */
+	if (Ph == 0)		/* If the phase is zero...           */
+	{			/* ...then we've already skipped the */
+	    Hp += Npc;		/*    first sample, so we must also  */
+	    Hdp += Npc;		/*    skip ahead in Imp[] and ImpD[] */
+	}
+    }
+    if (Interp)
+      while (Hp < End) {
+	  t = *Hp;		/* Get filter coeff */
+	  t += (((WORD)*Hdp)*a)>>Na; /* t is now interp'd filter coeff */
+	  Hdp += Npc;		/* Filter coeff differences step */
+	  t *= *Xp;		/* Mult coeff by input sample */
+	  if (t & (1<<(Nhxn-1)))  /* Round, if needed */
+	    t += (1<<(Nhxn-1));
+	  t >>= Nhxn;		/* Leave some guard bits, but come back some */
+	  v += t;			/* The filter output */
+	  Hp += Npc;		/* Filter coeff step */
+	  Xp += Inc;		/* Input signal step. NO CHECK ON BOUNDS */
+      } 
+    else 
+      while (Hp < End) {
+	  t = *Hp;		/* Get filter coeff */
+	  t *= *Xp;		/* Mult coeff by input sample */
+	  if (t & (1<<(Nhxn-1)))  /* Round, if needed */
+	    t += (1<<(Nhxn-1));
+	  t >>= Nhxn;		/* Leave some guard bits, but come back some */
+	  v += t;			/* The filter output */
+	  Hp += Npc;		/* Filter coeff step */
+	  Xp += Inc;		/* Input signal step. NO CHECK ON BOUNDS */
+      }
+    return(v);
+}
+
+WORD FilterUD( HWORD Imp[], HWORD ImpD[],
+		     UHWORD Nwing, bool Interp,
+		     HWORD *Xp, HWORD Ph, HWORD Inc, UHWORD dhb)
+{
+    HWORD a;
+    HWORD *Hp, *Hdp, *End;
+    WORD v, t;
+    UWORD Ho;
+    
+    v=0;
+    Ho = (Ph*(UWORD)dhb)>>Np;
+    End = &Imp[Nwing];
+    if (Inc == 1)		/* If doing right wing...              */
+    {				/* ...drop extra coeff, so when Ph is  */
+	End--;			/*    0.5, we don't do too many mult's */
+	if (Ph == 0)		/* If the phase is zero...           */
+	  Ho += dhb;		/* ...then we've already skipped the */
+    }				/*    first sample, so we must also  */
+				/*    skip ahead in Imp[] and ImpD[] */
+    if (Interp)
+      while ((Hp = &Imp[Ho>>Na]) < End) {
+	  t = *Hp;		/* Get IR sample */
+	  Hdp = &ImpD[Ho>>Na];  /* get interp (lower Na) bits from diff table*/
+	  a = Ho & Amask;	/* a is logically between 0 and 1 */
+	  t += (((WORD)*Hdp)*a)>>Na; /* t is now interp'd filter coeff */
+	  t *= *Xp;		/* Mult coeff by input sample */
+	  if (t & 1<<(Nhxn-1))	/* Round, if needed */
+	    t += 1<<(Nhxn-1);
+	  t >>= Nhxn;		/* Leave some guard bits, but come back some */
+	  v += t;			/* The filter output */
+	  Ho += dhb;		/* IR step */
+	  Xp += Inc;		/* Input signal step. NO CHECK ON BOUNDS */
+      }
+    else 
+      while ((Hp = &Imp[Ho>>Na]) < End) {
+	  t = *Hp;		/* Get IR sample */
+	  t *= *Xp;		/* Mult coeff by input sample */
+	  if (t & 1<<(Nhxn-1))	/* Round, if needed */
+	    t += 1<<(Nhxn-1);
+	  t >>= Nhxn;		/* Leave some guard bits, but come back some */
+	  v += t;			/* The filter output */
+	  Ho += dhb;		/* IR step */
+	  Xp += Inc;		/* Input signal step. NO CHECK ON BOUNDS */
+      }
+    return(v);
+}
 
-/* Private data for Lerp via LCM file */
-typedef struct resamplestuff {
-	double Factor;     /* Factor = Fout/Fin sample rates */
-	double rolloff;    /* roll-off frequency */
-	double beta;	      /* passband/stopband tuning magic */
-	int quadr;	      /* non-zero to use qprodUD quadratic interpolation */
-	long Nmult;
-	long Nwing;
-	long Nq;
-	Float *Imp;	      /* impulse [Nwing+1] Filter coefficients */
-
-	double Time;	      /* Current time/pos in input sample */
-	long dhb;
-
-	long a, b;	      /* gcd-reduced input,output rates	  */
-	long t;	      /* Current time/pos for exact-coeff's method */
-
-	long Xh;	      /* number of past/future samples needed by filter	 */
-	long Xoff;	      /* Xh plus some room for creep  */
-	long Xread;	      /* X[Xread] is start-position to enter new samples */
-	long Xp;	      /* X[Xp] is position to start filter application	 */
-	long Xsize, Ysize;  /* size (Floats) of X[],Y[]	  */
-	long Yposition;		/* FIXME: offset into Y buffer */
-	Float *X, *Y;      /* I/O buffers */
-} *resample_t;
-
-static void LpFilter(double c[],
-                     long N,
-                     double frq,
-                     double Beta,
-                     long Num);
-
-/* makeFilter is used by filter.c */
-int makeFilter(Float Imp[],
-               long Nwing,
-               double Froll,
-               double Beta,
-               long Num,
-               int Normalize);
-
-static long SrcUD(resample_t r, long Nx);
-static long SrcEX(resample_t r, long Nx);
 
+#pragma mark -
+
+
+#if 0
+static int resampleWithFilter(  /* number of output samples returned */
+    double factor,              /* factor = outSampleRate/inSampleRate */
+    int infd,                   /* input and output file descriptors */
+    int outfd,
+    int inCount,                /* number of input samples to convert */
+    int outCount,               /* number of output samples to compute */
+    int nChans,                 /* number of sound channels (1 or 2) */
+    bool interpFilt,            /* TRUE means interpolate filter coeffs */
+    HWORD Imp[], HWORD ImpD[],
+    UHWORD LpScl, UHWORD Nmult, UHWORD Nwing)
+{
+    UWORD Time, Time2;          /* Current time/pos in input sample */
+    UHWORD Xp, Ncreep, Xoff, Xread;
+    int OBUFFSIZE = (int)(((double)IBUFFSIZE)*factor+2.0);
+    HWORD X1[IBUFFSIZE], Y1[OBUFFSIZE]; /* I/O buffers */
+    HWORD X2[IBUFFSIZE], Y2[OBUFFSIZE]; /* I/O buffers */
+    UHWORD Nout, Nx;
+    int i, Ycount, last;
+    
+    MUS_SAMPLE_TYPE **obufs = sndlib_allocate_buffers(nChans, OBUFFSIZE);
+    if (obufs == NULL)
+        return err_ret("Can't allocate output buffers");
+
+    /* Account for increased filter gain when using factors less than 1 */
+    if (factor < 1)
+      LpScl = LpScl*factor + 0.5;
+
+    /* Calc reach of LP filter wing & give some creeping room */
+    Xoff = ((Nmult+1)/2.0) * MAX(1.0,1.0/factor) + 10;
+
+    if (IBUFFSIZE < 2*Xoff)      /* Check input buffer size */
+      return err_ret("IBUFFSIZE (or factor) is too small");
+
+    Nx = IBUFFSIZE - 2*Xoff;     /* # of samples to process each iteration */
+    
+    last = 0;                   /* Have not read last input sample yet */
+    Ycount = 0;                 /* Current sample and length of output file */
+    Xp = Xoff;                  /* Current "now"-sample pointer for input */
+    Xread = Xoff;               /* Position in input array to read into */
+    Time = (Xoff<<Np);          /* Current-time pointer for converter */
+    
+    for (i=0; i<Xoff; X1[i++]=0); /* Need Xoff zeros at begining of sample */
+    for (i=0; i<Xoff; X2[i++]=0); /* Need Xoff zeros at begining of sample */
+        
+    do {
+        if (!last)              /* If haven't read last sample yet */
+        {
+            last = readData(infd, inCount, X1, X2, IBUFFSIZE, 
+                            nChans, (int)Xread);
+            if (last && (last-Xoff<Nx)) { /* If last sample has been read... */
+                Nx = last-Xoff; /* ...calc last sample affected by filter */
+                if (Nx <= 0)
+                  break;
+            }
+        }
+        /* Resample stuff in input buffer */
+        Time2 = Time;
+        if (factor >= 1) {      /* SrcUp() is faster if we can use it */
+            Nout=SrcUp(X1,Y1,factor,&Time,Nx,Nwing,LpScl,Imp,ImpD,interpFilt);
+            if (nChans==2)
+              Nout=SrcUp(X2,Y2,factor,&Time2,Nx,Nwing,LpScl,Imp,ImpD,
+                         interpFilt);
+        }
+        else {
+            Nout=SrcUD(X1,Y1,factor,&Time,Nx,Nwing,LpScl,Imp,ImpD,interpFilt);
+            if (nChans==2)
+              Nout=SrcUD(X2,Y2,factor,&Time2,Nx,Nwing,LpScl,Imp,ImpD,
+                         interpFilt);
+        }
+
+        Time -= (Nx<<Np);       /* Move converter Nx samples back in time */
+        Xp += Nx;               /* Advance by number of samples processed */
+        Ncreep = (Time>>Np) - Xoff; /* Calc time accumulation in Time */
+        if (Ncreep) {
+            Time -= (Ncreep<<Np);    /* Remove time accumulation */
+            Xp += Ncreep;            /* and add it to read pointer */
+        }
+        for (i=0; i<IBUFFSIZE-Xp+Xoff; i++) { /* Copy part of input signal */
+            X1[i] = X1[i+Xp-Xoff]; /* that must be re-used */
+            if (nChans==2)
+              X2[i] = X2[i+Xp-Xoff]; /* that must be re-used */
+        }
+        if (last) {             /* If near end of sample... */
+            last -= Xp;         /* ...keep track were it ends */
+            if (!last)          /* Lengthen input by 1 sample if... */
+              last++;           /* ...needed to keep flag TRUE */
+        }
+        Xread = i;              /* Pos in input buff to read new data into */
+        Xp = Xoff;
+        
+        Ycount += Nout;
+        if (Ycount>outCount) {
+            Nout -= (Ycount-outCount);
+            Ycount = outCount;
+        }
+
+        if (Nout > OBUFFSIZE) /* Check to see if output buff overflowed */
+          return err_ret("Output array overflow");
+        
+        if (nChans==1) {
+            for (i = 0; i < Nout; i++)
+              obufs[0][i] = HWORD_TO_MUS_SAMPLE_TYPE(Y1[i]);
+        } else {
+            for (i = 0; i < Nout; i++) {
+                obufs[0][i] = HWORD_TO_MUS_SAMPLE_TYPE(Y1[i]);
+                obufs[1][i] = HWORD_TO_MUS_SAMPLE_TYPE(Y2[i]);
+            }
+        }
+        /* NB: errors reported within sndlib */
+        mus_file_write(outfd, 0, Nout - 1, nChans, obufs);
+
+        printf(".");  fflush(stdout);
+
+    } while (Ycount<outCount); /* Continue until done */
+
+    return(Ycount);             /* Return # of samples in output file */
+}
+#endif
+
+
+#pragma mark -
+
+
+#if 0
 /* here for linear interp.  might be useful for other things */
 static st_rate_t st_gcd(st_rate_t a, st_rate_t b)
 {
@@ -162,63 +584,9 @@ static st_rate_t st_gcd(st_rate_t a, st_rate_t b)
 
 
 /*
- * Process options
- */
-int st_resample_getopts(eff_t effp, int n, const char **argv) {
-	resample_t r = (resample_t) effp->priv;
-
-	/* These defaults are conservative with respect to aliasing. */
-	r->rolloff = 0.80;
-	r->beta = 16; /* anything <=2 means Nutall window */
-	r->quadr = 0;
-	r->Nmult = 45;
-
-	/* This used to fail, but with sox-12.15 it works. AW */
-	if ((n >= 1)) {
-		if (!strcmp(argv[0], "-qs")) {
-			r->quadr = 1;
-			n--;
-			argv++;
-		} else if (!strcmp(argv[0], "-q")) {
-			r->rolloff = 0.875;
-			r->quadr = 1;
-			r->Nmult = 75;
-			n--;
-			argv++;
-		} else if (!strcmp(argv[0], "-ql")) {
-			r->rolloff = 0.94;
-			r->quadr = 1;
-			r->Nmult = 149;
-			n--;
-			argv++;
-		}
-	}
-
-	if ((n >= 1) && (sscanf(argv[0], "%lf", &r->rolloff) != 1)) {
-		st_fail("Usage: resample [ rolloff [ beta ] ]");
-		return (ST_EOF);
-	} else if ((r->rolloff <= 0.01) || (r->rolloff >= 1.0)) {
-		st_fail("resample: rolloff factor (%f) no good, should be 0.01<x<1.0", r->rolloff);
-		return (ST_EOF);
-	}
-
-	if ((n >= 2) && !sscanf(argv[1], "%lf", &r->beta)) {
-		st_fail("Usage: resample [ rolloff [ beta ] ]");
-		return (ST_EOF);
-	} else if (r->beta <= 2.0) {
-		r->beta = 0;
-		st_report("resample opts: Nuttall window, cutoff %f\n", r->rolloff);
-	} else {
-		st_report("resample opts: Kaiser window, cutoff %f, beta %f\n", r->rolloff, r->beta);
-	}
-	return (ST_SUCCESS);
-}
-
-/*
  * Prepare processing.
  */
-int st_resample_start(eff_t effp, st_rate_t inrate, st_rate_t outrate) {
-	resample_t r = (resample_t) effp->priv;
+int st_resample_start(resample_t r, st_rate_t inrate, st_rate_t outrate) {
 	long Xoff, gcdrate;
 	int i;
 
@@ -256,7 +624,7 @@ int st_resample_start(eff_t effp, st_rate_t inrate, st_rate_t outrate) {
 	r->Imp = (Float *)malloc(sizeof(Float) * (r->Nwing + 2)) + 1;
 	/* need Imp[-1] and Imp[Nwing] for quadratic interpolation */
 	/* returns error # <=0, or adjusted wing-len > 0 */
-	i = makeFilter(r->Imp, r->Nwing, r->rolloff, r->beta, r->Nq, 1);
+	i = makeFilter(r->Imp, r->Nwing, r->rolloff, r->beta, r->Nq);
 	if (i <= 0) {
 		st_fail("resample: Unable to make filter\n");
 		return (ST_EOF);
@@ -313,8 +681,7 @@ int st_resample_start(eff_t effp, st_rate_t inrate, st_rate_t outrate) {
  * Processed signed long samples from ibuf to obuf.
  * Return number of samples processed.
  */
-int st_resample_flow(eff_t effp, AudioInputStream &input, st_sample_t *obuf, st_size_t *osamp, st_volume_t vol) {
-	resample_t r = (resample_t) effp->priv;
+int st_resample_flow(resample_t r, AudioInputStream &input, st_sample_t *obuf, st_size_t *osamp, st_volume_t vol) {
 	long i, k, last;
 	long Nout = 0;	// The number of bytes we effectively output
 	long Nx;		// The number of bytes we will read from input
@@ -450,8 +817,7 @@ printf("osamp = %ld, Nout = %ld\n", obufSize, Nout);
 /*
  * Process tail of input samples.
  */
-int st_resample_drain(eff_t effp, st_sample_t *obuf, st_size_t *osamp, st_volume_t vol) {
-	resample_t r = (resample_t) effp->priv;
+int st_resample_drain(resample_t r, st_sample_t *obuf, st_size_t *osamp, st_volume_t vol) {
 	long osamp_res;
 	st_sample_t *Obuf;
 	int rc;
@@ -465,7 +831,7 @@ int st_resample_drain(eff_t effp, st_sample_t *obuf, st_size_t *osamp, st_volume
 	while (!zero.eos() && osamp_res > 0) {
 		st_sample_t Osamp;
 		Osamp = osamp_res;
-		rc = st_resample_flow(effp, zero, Obuf, (st_size_t *) & Osamp, vol);
+		rc = st_resample_flow(r, zero, Obuf, (st_size_t *) & Osamp, vol);
 		if (rc)
 			return rc;
 		/*fprintf(stderr,"DRAIN isamp,osamp	(%d,%d) -> (%d,%d)\n",
@@ -485,322 +851,108 @@ int st_resample_drain(eff_t effp, st_sample_t *obuf, st_size_t *osamp, st_volume
  * Do anything required when you stop reading samples.	
  * Don't close input file! 
  */
-int st_resample_stop(eff_t effp) {
-	resample_t r = (resample_t) effp->priv;
-
+int st_resample_stop(resample_t r) {
 	free(r->Imp - 1);
 	free(r->X);
 	/* free(r->Y); Y is in same block starting at X */
 	return (ST_SUCCESS);
 }
 
-/* over 90% of CPU time spent in this iprodUD() function */
-/* quadratic interpolation */
-static double qprodUD(const Float Imp[], const Float *Xp, long Inc, double T0,
-                      long dhb, long ct) {
-	const double f = 1.0 / (1 << La);
-	double v;
-	long Ho;
-
-	Ho = (long)(T0 * dhb);
-	Ho += (ct - 1) * dhb; /* so Float sum starts with smallest coef's */
-	Xp += (ct - 1) * Inc;
-	v = 0;
-	do {
-		Float coef;
-		long Hoh;
-		Hoh = Ho >> La;
-		coef = Imp[Hoh];
-		{
-			Float dm, dp, t;
-			dm = coef - Imp[Hoh - 1];
-			dp = Imp[Hoh + 1] - coef;
-			t = (Ho & Amask) * f;
-			coef += ((dp - dm) * t + (dp + dm)) * t * 0.5;
-		}
-		/* filter coef, lower La bits by quadratic interpolation */
-		v += coef * *Xp;   /* sum coeff * input sample */
-		Xp -= Inc;	   /* Input signal step. NO CHECK ON ARRAY BOUNDS */
-		Ho -= dhb;	   /* IR step */
-	} while (--ct);
-	return v;
-}
-
-/* linear interpolation */
-static double iprodUD(const Float Imp[], const Float *Xp, long Inc,
-                      double T0, long dhb, long ct) {
-	const double f = 1.0 / (1 << La);
-	double v;
-	long Ho;
-
-	Ho = (long)(T0 * dhb);
-	Ho += (ct - 1) * dhb; /* so Float sum starts with smallest coef's */
-	Xp += (ct - 1) * Inc;
-	v = 0;
-	do {
-		Float coef;
-		long Hoh;
-		Hoh = Ho >> La;
-		/* if (Hoh >= End) break; */
-		coef = Imp[Hoh] + (Imp[Hoh + 1] - Imp[Hoh]) * (Ho & Amask) * f;
-		/* filter coef, lower La bits by linear interpolation */
-		v += coef * *Xp;   /* sum coeff * input sample */
-		Xp -= Inc;	   /* Input signal step. NO CHECK ON ARRAY BOUNDS */
-		Ho -= dhb;	   /* IR step */
-	} while (--ct);
-	return v;
-}
-
-/* From resample:filters.c */
-/* Sampling rate conversion subroutine */
-
-static long SrcUD(resample_t r, long Nx) {
-	Float *Ystart, *Y;
-	double Factor;
-	double dt;		       /* Step through input signal */
-	double time;
-	double (*prodUD)(const Float Imp[], const Float *Xp, long Inc, double T0, long dhb, long ct);
-	int n;
-
-	prodUD = (r->quadr) ? qprodUD : iprodUD; /* quadratic or linear interp */
-	Factor = r->Factor;
-	time = r->Time;
-	dt = 1.0 / Factor;	   /* Output sampling period */
-	//fprintf(stderr,"Factor %f, dt %f, ",Factor,dt);
-	//fprintf(stderr,"Time %f, ",r->Time);
-	/* (Xh * dhb)>>La is max index into Imp[] */
-	/*fprintf(stderr,"ct=%d\n",ct);*/
-	//fprintf(stderr,"ct=%.2f %d\n",(double)r->Nwing*Na/r->dhb, r->Xh);
-	//fprintf(stderr,"ct=%ld, T=%.6f, dhb=%6f, dt=%.6f\n", r->Xh, time-floor(time),(double)r->dhb/Na,dt);
-	Ystart = Y = r->Y + r->Yposition;
-	n = (int)ceil((double)Nx / dt);
-	while (n--) {
-		Float *Xp;
-		double v;
-		double T;
-		T = time - floor(time);	   /* fractional part of Time */
-		Xp = r->X + (long)time;	   /* Ptr to current input sample */
-
-		/* Past  inner product: */
-		v = (*prodUD)(r->Imp, Xp, -1, T, r->dhb, r->Xh); /* needs Np*Nmult in 31 bits */
-		/* Future inner product: */
-		v += (*prodUD)(r->Imp, Xp + 1, 1, (1.0 - T), r->dhb, r->Xh); /* prefer even total */
-
-		if (Factor < 1)
-			v *= Factor;
-		*Y++ = v;		     /* Deposit output */
-		time += dt;	     /* Move to next sample by time increment */
-	}
-	r->Time = time;
-	fprintf(stderr,"Time %f\n",r->Time);
-	return (Y - Ystart);	       /* Return the number of output samples */
-}
-
-/* exact coeff's */
-static double prodEX(const Float Imp[], const Float *Xp,
-                     long Inc, long T0, long dhb, long ct) {
-	double v;
-	const Float *Cp;
-
-	Cp = Imp + (ct - 1) * dhb + T0; /* so Float sum starts with smallest coef's */
-	Xp += (ct - 1) * Inc;
-	v = 0;
-	do {
-		v += *Cp * *Xp;   /* sum coeff * input sample */
-		Cp -= dhb;	   /* IR step */
-		Xp -= Inc;	   /* Input signal step. */
-	} while (--ct);
-	return v;
-}
-
-static long SrcEX(resample_t r, long Nx) {
-	Float *Ystart, *Y;
-	double Factor;
-	long a, b;
-	long time;
-	int n;
-
-	Factor = r->Factor;
-	time = r->t;
-	a = r->a;
-	b = r->b;
-	Ystart = Y = r->Y + r->Yposition;
-	n = (Nx * b + (a - 1)) / a;
-	while (n--) {
-		Float *Xp;
-		double v;
-		long T;
-		T = time % b;		   /* fractional part of Time */
-		Xp = r->X + (time / b);	   /* Ptr to current input sample */
-
-		/* Past	 inner product: */
-		v = prodEX(r->Imp, Xp, -1, T, b, r->Xh);
-		/* Future inner product: */
-		v += prodEX(r->Imp, Xp + 1, 1, b - T, b, r->Xh);
-
-		if (Factor < 1)
-			v *= Factor;
-		*Y++ = v;	      /* Deposit output */
-		time += a;	      /* Move to next sample by time increment */
-	}
-	r->t = time;
-	return (Y - Ystart);	       /* Return the number of output samples */
-}
-
-int makeFilter(Float Imp[], long Nwing, double Froll, double Beta,
-               long Num, int Normalize) {
-	double *ImpR;
-	long Mwing, i;
-
-	if (Nwing > MAXNWING)		      /* Check for valid parameters */
-		return ( -1);
-	if ((Froll <= 0) || (Froll > 1))
-		return ( -2);
-
-	/* it does help accuracy a bit to have the window stop at
-	 * a zero-crossing of the sinc function */
-	Mwing = (long)(floor((double)Nwing / (Num / Froll)) * (Num / Froll) + 0.5);
-	if (Mwing == 0)
-		return ( -4);
-
-	ImpR = (double *) malloc(sizeof(double) * Mwing);
-
-	/* Design a Nuttall or Kaiser windowed Sinc low-pass filter */
-	LpFilter(ImpR, Mwing, Froll, Beta, Num);
-
-	if (Normalize) { /* 'correct' the DC gain of the lowpass filter */
-		long Dh;
-		double DCgain;
-		DCgain = 0;
-		Dh = Num;			 /* Filter sampling period for factors>=1 */
-		for (i = Dh; i < Mwing; i += Dh)
-			DCgain += ImpR[i];
-		DCgain = 2 * DCgain + ImpR[0];    /* DC gain of real coefficients */
-		st_report("DCgain err=%.12f",DCgain-1.0);	// FIXME
-
-		DCgain = 1.0 / DCgain;
-		for (i = 0; i < Mwing; i++)
-			Imp[i] = ImpR[i] * DCgain;
-
-	} else {
-		for (i = 0; i < Mwing; i++)
-			Imp[i] = ImpR[i];
-	}
-	free(ImpR);
-	for (i = Mwing; i <= Nwing; i++)
-		Imp[i] = 0;
-	/* Imp[Mwing] and Imp[-1] needed for quadratic interpolation */
-	Imp[ -1] = Imp[1];
-
-	return (Mwing);
-}
-
-/* LpFilter()
- *
- * reference: "Digital Filters, 2nd edition"
- *	      R.W. Hamming, pp. 178-179
- *
- * Izero() computes the 0th order modified bessel function of the first kind.
- *    (Needed to compute Kaiser window).
- *
- * LpFilter() computes the coeffs of a Kaiser-windowed low pass filter with
- *    the following characteristics:
- *
- *	 c[]  = array in which to store computed coeffs
- *	 frq  = roll-off frequency of filter
- *	 N    = Half the window length in number of coeffs
- *	 Beta = parameter of Kaiser window
- *	 Num  = number of coeffs before 1/frq
- *
- * Beta trades the rejection of the lowpass filter against the transition
- *    width from passband to stopband.	Larger Beta means a slower
- *    transition and greater stopband rejection.  See Rabiner and Gold
- *    (Theory and Application of DSP) under Kaiser windows for more about
- *    Beta.  The following table from Rabiner and Gold gives some feel
- *    for the effect of Beta:
- *
- * All ripples in dB, width of transition band = D*N where N = window length
- *
- *		 BETA	 D	 PB RIP	  SB RIP
- *		 2.120	 1.50  +-0.27	   -30
- *		 3.384	 2.23	 0.0864	   -40
- *		 4.538	 2.93	 0.0274	   -50
- *		 5.658	 3.62	 0.00868   -60
- *		 6.764	 4.32	 0.00275   -70
- *		 7.865	 5.0	 0.000868  -80
- *		 8.960	 5.7	 0.000275  -90
- *		 10.056	 6.4	 0.000087  -100
- */
-
-
-#define IzeroEPSILON 1E-21		 /* Max error acceptable in Izero */
+#endif
 
-static double Izero(double x) {
-	double sum, u, halfx, temp;
-	long n;
+#pragma mark -
 
-	sum = u = n = 1;
-	halfx = x / 2.0;
-	do {
-		temp = halfx / (double)n;
-		n += 1;
-		temp *= temp;
-		u *= temp;
-		sum += u;
-	} while (u >= IzeroEPSILON*sum);
-	return (sum);
-}
 
-static void LpFilter(double *c, long N, double frq, double Beta, long Num) {
-	long i;
+ResampleRateConverter::ResampleRateConverter(st_rate_t inrate, st_rate_t outrate, int quality) {
+	// FIXME: quality is for now a nasty hack. Valid values are 0,1,2,3
 
-	/* Calculate filter coeffs: */
-	c[0] = frq;
-	for (i = 1; i < N; i++) {
-		double x = M_PI * (double)i / (double)(Num);
-		c[i] = sin(x * frq) / x;
-	}
+	double rolloff;    /* roll-off frequency */
+	double beta;	   /* passband/stopband tuning magic */
 
-	if (Beta > 2) { /* Apply Kaiser window to filter coeffs: */
-		double IBeta = 1.0 / Izero(Beta);
-		for (i = 1; i < N; i++) {
-			double x = (double)i / (double)(N);
-			c[i] *= Izero(Beta * sqrt(1.0 - x * x)) * IBeta;
-		}
-	} else { /* Apply Nuttall window: */
-		for (i = 0; i < N; i++) {
-			double x = M_PI * i / N;
-			c[i] *= 0.36335819 + 0.4891775 * cos(x) + 0.1365995 * cos(2 * x) + 0.0106411 * cos(3 * x);
-		}
+	switch (quality) {
+	case 0:
+		/* These defaults are conservative with respect to aliasing. */
+		rolloff = 0.80;
+		beta = 16;
+		quadr = 0;
+		Nmult = 45;
+		break;
+	case 1:
+		rolloff = 0.80;
+		beta = 16;
+		quadr = 1;
+		Nmult = 45;
+		break;
+	case 2:
+		rolloff = 0.875;
+		beta = 16;
+		quadr = 1;
+		Nmult = 75;
+		break;
+	case 3:
+		rolloff = 0.94;
+		beta = 16;
+		quadr = 1;
+		Nmult = 149;
+		break;
+	default:
+		error("Illegal quality level %d\n", quality);
+		break;
 	}
-}
-
-
-#pragma mark -
 
+	makeFilter(Imp, ImpD, &LpScl, Nmult, rolloff, beta);
+
+    int OBUFFSIZE = (IBUFFSIZE * outrate / inrate + 2);
+    X1 = (HWORD *)malloc(IBUFFSIZE);
+    X2 = (HWORD *)malloc(IBUFFSIZE);
+    Y1 = (HWORD *)malloc(OBUFFSIZE);
+    Y2 = (HWORD *)malloc(OBUFFSIZE);
+    
+    // HACK this is invalid code but "fixes" a compiler warning for now
+	double factor = outrate / (double)inrate;
+	UHWORD Xp, /*Ncreep,*/ Xoff, Xread;
+	UHWORD Nout, Nx;
+	int Ycount, last;
+	
+	/* Account for increased filter gain when using factors less than 1 */
+	if (factor < 1)
+		LpScl = (UHWORD)(LpScl*factor + 0.5);
+	
+	/* Calc reach of LP filter wing & give some creeping room */
+	Xoff = (UHWORD)(((Nmult+1)/2.0) * MAX(1.0,1.0/factor) + 10);
+	
+	if (IBUFFSIZE < 2*Xoff)      /* Check input buffer size */
+		error("IBUFFSIZE (or factor) is too small");
+	
+	Nx = IBUFFSIZE - 2*Xoff;     /* # of samples to process each iteration */
+	
+	last = 0;                   /* Have not read last input sample yet */
+	Ycount = 0;                 /* Current sample and length of output file */
+	Xp = Xoff;                  /* Current "now"-sample pointer for input */
+	Xread = Xoff;               /* Position in input array to read into */
+	Time = (Xoff<<Np);          /* Current-time pointer for converter */
+	
+	Nout = SrcUp(X1, Y1, factor, &Time, Nx, Nwing, LpScl, Imp, ImpD, quadr);
+	Nout = SrcUD(X1, Y1, factor, &Time, Nx, Nwing, LpScl, Imp, ImpD, quadr);
 
-ResampleRateConverter::ResampleRateConverter(st_rate_t inrate, st_rate_t outrate, int quality) {
-	// FIXME: quality is for now a nasty hack.
-	// Valid values are 0,1,2,3 (everything else is treated like 0 for now)
-	const char *arg = 0;
-	switch (quality) {
-	case 1: arg = "-qs"; break;
-	case 2: arg = "-q"; break;
-	case 3: arg = "-ql"; break;
-	}
-	st_resample_getopts(&effp, arg ? 1 : 0, &arg);
-	st_resample_start(&effp, inrate, outrate);
+//	st_resample_start(&rstuff, inrate, outrate);
 }
 
 ResampleRateConverter::~ResampleRateConverter() {
-	st_resample_stop(&effp);
+//	st_resample_stop(&rstuff);
+	free(X1);
+	free(X2);
+	free(Y1);
+	free(Y2);
 }
 
 int ResampleRateConverter::flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
-	return st_resample_flow(&effp, input, obuf, &osamp, vol);
+//	return st_resample_flow(&rstuff, input, obuf, &osamp, vol);
+	return 0;
 }
 
 int ResampleRateConverter::drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
-	return st_resample_drain(&effp, obuf, &osamp, vol);
+//	return st_resample_drain(&rstuff, obuf, &osamp, vol);
+	return 0;
 }