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diff --git a/src/libspeex/resample.c b/src/libspeex/resample.c
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+/* Copyright (C) 2007 Jean-Marc Valin
+      
+   File: resample.c
+   Arbitrary resampling code
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions are
+   met:
+
+   1. Redistributions of source code must retain the above copyright notice,
+   this list of conditions and the following disclaimer.
+
+   2. Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   3. The name of the author may not be used to endorse or promote products
+   derived from this software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
+   IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
+   OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+   DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
+   INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+   (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+   SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
+   STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
+   ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/*
+   The design goals of this code are:
+      - Very fast algorithm
+      - SIMD-friendly algorithm
+      - Low memory requirement
+      - Good *perceptual* quality (and not best SNR)
+
+   Warning: This resampler is relatively new. Although I think I got rid of 
+   all the major bugs and I don't expect the API to change anymore, there
+   may be something I've missed. So use with caution.
+
+   This algorithm is based on this original resampling algorithm:
+   Smith, Julius O. Digital Audio Resampling Home Page
+   Center for Computer Research in Music and Acoustics (CCRMA), 
+   Stanford University, 2007.
+   Web published at http://www-ccrma.stanford.edu/~jos/resample/.
+
+   There is one main difference, though. This resampler uses cubic 
+   interpolation instead of linear interpolation in the above paper. This
+   makes the table much smaller and makes it possible to compute that table
+   on a per-stream basis. In turn, being able to tweak the table for each 
+   stream makes it possible to both reduce complexity on simple ratios 
+   (e.g. 2/3), and get rid of the rounding operations in the inner loop. 
+   The latter both reduces CPU time and makes the algorithm more SIMD-friendly.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#ifdef OUTSIDE_SPEEX
+#include <stdlib.h>
+static void *speex_alloc (int size) {return calloc(size,1);}
+static void *speex_realloc (void *ptr, int size) {return realloc(ptr, size);}
+static void speex_free (void *ptr) {free(ptr);}
+#include "speex_resampler.h"
+#include "arch.h"
+#else /* OUTSIDE_SPEEX */
+               
+#include "speex/speex_resampler.h"
+#include "arch.h"
+#include "os_support.h"
+#endif /* OUTSIDE_SPEEX */
+
+#include <math.h>
+
+#ifndef M_PI
+#define M_PI 3.14159263
+#endif
+
+#ifdef FIXED_POINT
+#define WORD2INT(x) ((x) < -32767 ? -32768 : ((x) > 32766 ? 32767 : (x)))  
+#else
+#define WORD2INT(x) ((x) < -32767.5f ? -32768 : ((x) > 32766.5f ? 32767 : floor(.5+(x))))  
+#endif
+               
+/*#define float double*/
+#define FILTER_SIZE 64
+#define OVERSAMPLE 8
+
+#define IMAX(a,b) ((a) > (b) ? (a) : (b))
+#define IMIN(a,b) ((a) < (b) ? (a) : (b))
+
+#ifndef NULL
+#define NULL 0
+#endif
+
+typedef int (*resampler_basic_func)(SpeexResamplerState *, spx_uint32_t , const spx_word16_t *, spx_uint32_t *, spx_word16_t *, spx_uint32_t *);
+
+struct SpeexResamplerState_ {
+   spx_uint32_t in_rate;
+   spx_uint32_t out_rate;
+   spx_uint32_t num_rate;
+   spx_uint32_t den_rate;
+   
+   int    quality;
+   spx_uint32_t nb_channels;
+   spx_uint32_t filt_len;
+   spx_uint32_t mem_alloc_size;
+   int          int_advance;
+   int          frac_advance;
+   float  cutoff;
+   spx_uint32_t oversample;
+   int          initialised;
+   int          started;
+   
+   /* These are per-channel */
+   spx_int32_t  *last_sample;
+   spx_uint32_t *samp_frac_num;
+   spx_uint32_t *magic_samples;
+   
+   spx_word16_t *mem;
+   spx_word16_t *sinc_table;
+   spx_uint32_t sinc_table_length;
+   resampler_basic_func resampler_ptr;
+         
+   int    in_stride;
+   int    out_stride;
+} ;
+
+static double kaiser12_table[68] = {
+   0.99859849, 1.00000000, 0.99859849, 0.99440475, 0.98745105, 0.97779076,
+   0.96549770, 0.95066529, 0.93340547, 0.91384741, 0.89213598, 0.86843014,
+   0.84290116, 0.81573067, 0.78710866, 0.75723148, 0.72629970, 0.69451601,
+   0.66208321, 0.62920216, 0.59606986, 0.56287762, 0.52980938, 0.49704014,
+   0.46473455, 0.43304576, 0.40211431, 0.37206735, 0.34301800, 0.31506490,
+   0.28829195, 0.26276832, 0.23854851, 0.21567274, 0.19416736, 0.17404546,
+   0.15530766, 0.13794294, 0.12192957, 0.10723616, 0.09382272, 0.08164178,
+   0.07063950, 0.06075685, 0.05193064, 0.04409466, 0.03718069, 0.03111947,
+   0.02584161, 0.02127838, 0.01736250, 0.01402878, 0.01121463, 0.00886058,
+   0.00691064, 0.00531256, 0.00401805, 0.00298291, 0.00216702, 0.00153438,
+   0.00105297, 0.00069463, 0.00043489, 0.00025272, 0.00013031, 0.0000527734,
+   0.00001000, 0.00000000};
+/*
+static double kaiser12_table[36] = {
+   0.99440475, 1.00000000, 0.99440475, 0.97779076, 0.95066529, 0.91384741,
+   0.86843014, 0.81573067, 0.75723148, 0.69451601, 0.62920216, 0.56287762,
+   0.49704014, 0.43304576, 0.37206735, 0.31506490, 0.26276832, 0.21567274,
+   0.17404546, 0.13794294, 0.10723616, 0.08164178, 0.06075685, 0.04409466,
+   0.03111947, 0.02127838, 0.01402878, 0.00886058, 0.00531256, 0.00298291,
+   0.00153438, 0.00069463, 0.00025272, 0.0000527734, 0.00000500, 0.00000000};
+*/
+static double kaiser10_table[36] = {
+   0.99537781, 1.00000000, 0.99537781, 0.98162644, 0.95908712, 0.92831446,
+   0.89005583, 0.84522401, 0.79486424, 0.74011713, 0.68217934, 0.62226347,
+   0.56155915, 0.50119680, 0.44221549, 0.38553619, 0.33194107, 0.28205962,
+   0.23636152, 0.19515633, 0.15859932, 0.12670280, 0.09935205, 0.07632451,
+   0.05731132, 0.04193980, 0.02979584, 0.02044510, 0.01345224, 0.00839739,
+   0.00488951, 0.00257636, 0.00115101, 0.00035515, 0.00000000, 0.00000000};
+
+static double kaiser8_table[36] = {
+   0.99635258, 1.00000000, 0.99635258, 0.98548012, 0.96759014, 0.94302200,
+   0.91223751, 0.87580811, 0.83439927, 0.78875245, 0.73966538, 0.68797126,
+   0.63451750, 0.58014482, 0.52566725, 0.47185369, 0.41941150, 0.36897272,
+   0.32108304, 0.27619388, 0.23465776, 0.19672670, 0.16255380, 0.13219758,
+   0.10562887, 0.08273982, 0.06335451, 0.04724088, 0.03412321, 0.02369490,
+   0.01563093, 0.00959968, 0.00527363, 0.00233883, 0.00050000, 0.00000000};
+   
+static double kaiser6_table[36] = {
+   0.99733006, 1.00000000, 0.99733006, 0.98935595, 0.97618418, 0.95799003,
+   0.93501423, 0.90755855, 0.87598009, 0.84068475, 0.80211977, 0.76076565,
+   0.71712752, 0.67172623, 0.62508937, 0.57774224, 0.53019925, 0.48295561,
+   0.43647969, 0.39120616, 0.34752997, 0.30580127, 0.26632152, 0.22934058,
+   0.19505503, 0.16360756, 0.13508755, 0.10953262, 0.08693120, 0.06722600,
+   0.05031820, 0.03607231, 0.02432151, 0.01487334, 0.00752000, 0.00000000};
+
+struct FuncDef {
+   double *table;
+   int oversample;
+};
+      
+static struct FuncDef _KAISER12 = {kaiser12_table, 64};
+#define KAISER12 (&_KAISER12)
+/*static struct FuncDef _KAISER12 = {kaiser12_table, 32};
+#define KAISER12 (&_KAISER12)*/
+static struct FuncDef _KAISER10 = {kaiser10_table, 32};
+#define KAISER10 (&_KAISER10)
+static struct FuncDef _KAISER8 = {kaiser8_table, 32};
+#define KAISER8 (&_KAISER8)
+static struct FuncDef _KAISER6 = {kaiser6_table, 32};
+#define KAISER6 (&_KAISER6)
+
+struct QualityMapping {
+   int base_length;
+   int oversample;
+   float downsample_bandwidth;
+   float upsample_bandwidth;
+   struct FuncDef *window_func;
+};
+
+
+/* This table maps conversion quality to internal parameters. There are two
+   reasons that explain why the up-sampling bandwidth is larger than the 
+   down-sampling bandwidth:
+   1) When up-sampling, we can assume that the spectrum is already attenuated
+      close to the Nyquist rate (from an A/D or a previous resampling filter)
+   2) Any aliasing that occurs very close to the Nyquist rate will be masked
+      by the sinusoids/noise just below the Nyquist rate (guaranteed only for
+      up-sampling).
+*/
+static const struct QualityMapping quality_map[11] = {
+   {  8,  4, 0.830f, 0.860f, KAISER6 }, /* Q0 */
+   { 16,  4, 0.850f, 0.880f, KAISER6 }, /* Q1 */
+   { 32,  4, 0.882f, 0.910f, KAISER6 }, /* Q2 */  /* 82.3% cutoff ( ~60 dB stop) 6  */
+   { 48,  8, 0.895f, 0.917f, KAISER8 }, /* Q3 */  /* 84.9% cutoff ( ~80 dB stop) 8  */
+   { 64,  8, 0.921f, 0.940f, KAISER8 }, /* Q4 */  /* 88.7% cutoff ( ~80 dB stop) 8  */
+   { 80, 16, 0.922f, 0.940f, KAISER10}, /* Q5 */  /* 89.1% cutoff (~100 dB stop) 10 */
+   { 96, 16, 0.940f, 0.945f, KAISER10}, /* Q6 */  /* 91.5% cutoff (~100 dB stop) 10 */
+   {128, 16, 0.950f, 0.950f, KAISER10}, /* Q7 */  /* 93.1% cutoff (~100 dB stop) 10 */
+   {160, 16, 0.960f, 0.960f, KAISER10}, /* Q8 */  /* 94.5% cutoff (~100 dB stop) 10 */
+   {192, 32, 0.968f, 0.968f, KAISER12}, /* Q9 */  /* 95.5% cutoff (~100 dB stop) 10 */
+   {256, 32, 0.975f, 0.975f, KAISER12}, /* Q10 */ /* 96.6% cutoff (~100 dB stop) 10 */
+};
+/*8,24,40,56,80,104,128,160,200,256,320*/
+static double compute_func(float x, struct FuncDef *func)
+{
+   float y, frac;
+   double interp[4];
+   int ind; 
+   y = x*func->oversample;
+   ind = (int)floor(y);
+   frac = (y-ind);
+   /* CSE with handle the repeated powers */
+   interp[3] =  -0.1666666667*frac + 0.1666666667*(frac*frac*frac);
+   interp[2] = frac + 0.5*(frac*frac) - 0.5*(frac*frac*frac);
+   /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac;*/
+   interp[0] = -0.3333333333*frac + 0.5*(frac*frac) - 0.1666666667*(frac*frac*frac);
+   /* Just to make sure we don't have rounding problems */
+   interp[1] = 1.f-interp[3]-interp[2]-interp[0];
+   
+   /*sum = frac*accum[1] + (1-frac)*accum[2];*/
+   return interp[0]*func->table[ind] + interp[1]*func->table[ind+1] + interp[2]*func->table[ind+2] + interp[3]*func->table[ind+3];
+}
+
+#if 0
+#include <stdio.h>
+int main(int argc, char **argv)
+{
+   int i;
+   for (i=0;i<256;i++)
+   {
+      printf ("%f\n", compute_func(i/256., KAISER12));
+   }
+   return 0;
+}
+#endif
+
+#ifdef FIXED_POINT
+/* The slow way of computing a sinc for the table. Should improve that some day */
+static spx_word16_t sinc(float cutoff, float x, int N, struct FuncDef *window_func)
+{
+   /*fprintf (stderr, "%f ", x);*/
+   float xx = x * cutoff;
+   if (fabs(x)<1e-6f)
+      return WORD2INT(32768.*cutoff);
+   else if (fabs(x) > .5f*N)
+      return 0;
+   /*FIXME: Can it really be any slower than this? */
+   return WORD2INT(32768.*cutoff*sin(M_PI*xx)/(M_PI*xx) * compute_func(fabs(2.*x/N), window_func));
+}
+#else
+/* The slow way of computing a sinc for the table. Should improve that some day */
+static spx_word16_t sinc(float cutoff, float x, int N, struct FuncDef *window_func)
+{
+   /*fprintf (stderr, "%f ", x);*/
+   float xx = x * cutoff;
+   if (fabs(x)<1e-6)
+      return cutoff;
+   else if (fabs(x) > .5*N)
+      return 0;
+   /*FIXME: Can it really be any slower than this? */
+   return cutoff*sin(M_PI*xx)/(M_PI*xx) * compute_func(fabs(2.*x/N), window_func);
+}
+#endif
+
+#ifdef FIXED_POINT
+static void cubic_coef(spx_word16_t x, spx_word16_t interp[4])
+{
+   /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
+   but I know it's MMSE-optimal on a sinc */
+   spx_word16_t x2, x3;
+   x2 = MULT16_16_P15(x, x);
+   x3 = MULT16_16_P15(x, x2);
+   interp[0] = PSHR32(MULT16_16(QCONST16(-0.16667f, 15),x) + MULT16_16(QCONST16(0.16667f, 15),x3),15);
+   interp[1] = EXTRACT16(EXTEND32(x) + SHR32(SUB32(EXTEND32(x2),EXTEND32(x3)),1));
+   interp[3] = PSHR32(MULT16_16(QCONST16(-0.33333f, 15),x) + MULT16_16(QCONST16(.5f,15),x2) - MULT16_16(QCONST16(0.16667f, 15),x3),15);
+   /* Just to make sure we don't have rounding problems */
+   interp[2] = Q15_ONE-interp[0]-interp[1]-interp[3];
+   if (interp[2]<32767)
+      interp[2]+=1;
+}
+#else
+static void cubic_coef(spx_word16_t frac, spx_word16_t interp[4])
+{
+   /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
+   but I know it's MMSE-optimal on a sinc */
+   interp[0] =  -0.16667f*frac + 0.16667f*frac*frac*frac;
+   interp[1] = frac + 0.5f*frac*frac - 0.5f*frac*frac*frac;
+   /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac;*/
+   interp[3] = -0.33333f*frac + 0.5f*frac*frac - 0.16667f*frac*frac*frac;
+   /* Just to make sure we don't have rounding problems */
+   interp[2] = 1.-interp[0]-interp[1]-interp[3];
+}
+#endif
+
+static int resampler_basic_direct_single(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
+{
+   int N = st->filt_len;
+   int out_sample = 0;
+   spx_word16_t *mem;
+   int last_sample = st->last_sample[channel_index];
+   spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+   mem = st->mem + channel_index * st->mem_alloc_size;
+   while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
+   {
+      int j;
+      spx_word32_t sum=0;
+      
+      /* We already have all the filter coefficients pre-computed in the table */
+      const spx_word16_t *ptr;
+      /* Do the memory part */
+      for (j=0;last_sample-N+1+j < 0;j++)
+      {
+         sum += MULT16_16(mem[last_sample+j],st->sinc_table[samp_frac_num*st->filt_len+j]);
+      }
+      
+      /* Do the new part */
+      if (in != NULL)
+      {
+         ptr = in+st->in_stride*(last_sample-N+1+j);
+         for (;j<N;j++)
+         {
+            sum += MULT16_16(*ptr,st->sinc_table[samp_frac_num*st->filt_len+j]);
+            ptr += st->in_stride;
+         }
+      }
+      
+      *out = PSHR32(sum,15);
+      out += st->out_stride;
+      out_sample++;
+      last_sample += st->int_advance;
+      samp_frac_num += st->frac_advance;
+      if (samp_frac_num >= st->den_rate)
+      {
+         samp_frac_num -= st->den_rate;
+         last_sample++;
+      }
+   }
+   st->last_sample[channel_index] = last_sample;
+   st->samp_frac_num[channel_index] = samp_frac_num;
+   return out_sample;
+}
+
+#ifdef FIXED_POINT
+#else
+/* This is the same as the previous function, except with a double-precision accumulator */
+static int resampler_basic_direct_double(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
+{
+   int N = st->filt_len;
+   int out_sample = 0;
+   spx_word16_t *mem;
+   int last_sample = st->last_sample[channel_index];
+   spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+   mem = st->mem + channel_index * st->mem_alloc_size;
+   while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
+   {
+      int j;
+      double sum=0;
+      
+      /* We already have all the filter coefficients pre-computed in the table */
+      const spx_word16_t *ptr;
+      /* Do the memory part */
+      for (j=0;last_sample-N+1+j < 0;j++)
+      {
+         sum += MULT16_16(mem[last_sample+j],(double)st->sinc_table[samp_frac_num*st->filt_len+j]);
+      }
+      
+      /* Do the new part */
+      if (in != NULL)
+      {
+         ptr = in+st->in_stride*(last_sample-N+1+j);
+         for (;j<N;j++)
+         {
+            sum += MULT16_16(*ptr,(double)st->sinc_table[samp_frac_num*st->filt_len+j]);
+            ptr += st->in_stride;
+         }
+      }
+      
+      *out = sum;
+      out += st->out_stride;
+      out_sample++;
+      last_sample += st->int_advance;
+      samp_frac_num += st->frac_advance;
+      if (samp_frac_num >= st->den_rate)
+      {
+         samp_frac_num -= st->den_rate;
+         last_sample++;
+      }
+   }
+   st->last_sample[channel_index] = last_sample;
+   st->samp_frac_num[channel_index] = samp_frac_num;
+   return out_sample;
+}
+#endif
+
+static int resampler_basic_interpolate_single(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
+{
+   int N = st->filt_len;
+   int out_sample = 0;
+   spx_word16_t *mem;
+   int last_sample = st->last_sample[channel_index];
+   spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+   mem = st->mem + channel_index * st->mem_alloc_size;
+   while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
+   {
+      int j;
+      spx_word32_t sum=0;
+      
+      /* We need to interpolate the sinc filter */
+      spx_word32_t accum[4] = {0.f,0.f, 0.f, 0.f};
+      spx_word16_t interp[4];
+      const spx_word16_t *ptr;
+      int offset;
+      spx_word16_t frac;
+      offset = samp_frac_num*st->oversample/st->den_rate;
+#ifdef FIXED_POINT
+      frac = PDIV32(SHL32((samp_frac_num*st->oversample) % st->den_rate,15),st->den_rate);
+#else
+      frac = ((float)((samp_frac_num*st->oversample) % st->den_rate))/st->den_rate;
+#endif
+         /* This code is written like this to make it easy to optimise with SIMD.
+      For most DSPs, it would be best to split the loops in two because most DSPs 
+      have only two accumulators */
+      for (j=0;last_sample-N+1+j < 0;j++)
+      {
+         spx_word16_t curr_mem = mem[last_sample+j];
+         accum[0] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset-2]);
+         accum[1] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset-1]);
+         accum[2] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset]);
+         accum[3] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset+1]);
+      }
+      
+      if (in != NULL)
+      {
+         ptr = in+st->in_stride*(last_sample-N+1+j);
+         /* Do the new part */
+         for (;j<N;j++)
+         {
+            spx_word16_t curr_in = *ptr;
+            ptr += st->in_stride;
+            accum[0] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-2]);
+            accum[1] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-1]);
+            accum[2] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset]);
+            accum[3] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset+1]);
+         }
+      }
+      cubic_coef(frac, interp);
+      sum = MULT16_32_Q15(interp[0],accum[0]) + MULT16_32_Q15(interp[1],accum[1]) + MULT16_32_Q15(interp[2],accum[2]) + MULT16_32_Q15(interp[3],accum[3]);
+   
+      *out = PSHR32(sum,15);
+      out += st->out_stride;
+      out_sample++;
+      last_sample += st->int_advance;
+      samp_frac_num += st->frac_advance;
+      if (samp_frac_num >= st->den_rate)
+      {
+         samp_frac_num -= st->den_rate;
+         last_sample++;
+      }
+   }
+   st->last_sample[channel_index] = last_sample;
+   st->samp_frac_num[channel_index] = samp_frac_num;
+   return out_sample;
+}
+
+#ifdef FIXED_POINT
+#else
+/* This is the same as the previous function, except with a double-precision accumulator */
+static int resampler_basic_interpolate_double(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
+{
+   int N = st->filt_len;
+   int out_sample = 0;
+   spx_word16_t *mem;
+   int last_sample = st->last_sample[channel_index];
+   spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+   mem = st->mem + channel_index * st->mem_alloc_size;
+   while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
+   {
+      int j;
+      spx_word32_t sum=0;
+      
+      /* We need to interpolate the sinc filter */
+      double accum[4] = {0.f,0.f, 0.f, 0.f};
+      float interp[4];
+      const spx_word16_t *ptr;
+      float alpha = ((float)samp_frac_num)/st->den_rate;
+      int offset = samp_frac_num*st->oversample/st->den_rate;
+      float frac = alpha*st->oversample - offset;
+         /* This code is written like this to make it easy to optimise with SIMD.
+      For most DSPs, it would be best to split the loops in two because most DSPs 
+      have only two accumulators */
+      for (j=0;last_sample-N+1+j < 0;j++)
+      {
+         double curr_mem = mem[last_sample+j];
+         accum[0] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset-2]);
+         accum[1] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset-1]);
+         accum[2] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset]);
+         accum[3] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset+1]);
+      }
+      if (in != NULL)
+      {
+         ptr = in+st->in_stride*(last_sample-N+1+j);
+         /* Do the new part */
+         for (;j<N;j++)
+         {
+            double curr_in = *ptr;
+            ptr += st->in_stride;
+            accum[0] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-2]);
+            accum[1] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-1]);
+            accum[2] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset]);
+            accum[3] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset+1]);
+         }
+      }
+      cubic_coef(frac, interp);
+      sum = interp[0]*accum[0] + interp[1]*accum[1] + interp[2]*accum[2] + interp[3]*accum[3];
+   
+      *out = PSHR32(sum,15);
+      out += st->out_stride;
+      out_sample++;
+      last_sample += st->int_advance;
+      samp_frac_num += st->frac_advance;
+      if (samp_frac_num >= st->den_rate)
+      {
+         samp_frac_num -= st->den_rate;
+         last_sample++;
+      }
+   }
+   st->last_sample[channel_index] = last_sample;
+   st->samp_frac_num[channel_index] = samp_frac_num;
+   return out_sample;
+}
+#endif
+
+static void update_filter(SpeexResamplerState *st)
+{
+   spx_uint32_t old_length;
+   
+   old_length = st->filt_len;
+   st->oversample = quality_map[st->quality].oversample;
+   st->filt_len = quality_map[st->quality].base_length;
+   
+   if (st->num_rate > st->den_rate)
+   {
+      /* down-sampling */
+      st->cutoff = quality_map[st->quality].downsample_bandwidth * st->den_rate / st->num_rate;
+      /* FIXME: divide the numerator and denominator by a certain amount if they're too large */
+      st->filt_len = st->filt_len*st->num_rate / st->den_rate;
+      /* Round down to make sure we have a multiple of 4 */
+      st->filt_len &= (~0x3);
+      if (2*st->den_rate < st->num_rate)
+         st->oversample >>= 1;
+      if (4*st->den_rate < st->num_rate)
+         st->oversample >>= 1;
+      if (8*st->den_rate < st->num_rate)
+         st->oversample >>= 1;
+      if (16*st->den_rate < st->num_rate)
+         st->oversample >>= 1;
+      if (st->oversample < 1)
+         st->oversample = 1;
+   } else {
+      /* up-sampling */
+      st->cutoff = quality_map[st->quality].upsample_bandwidth;
+   }
+
+   /* Choose the resampling type that requires the least amount of memory */
+   if (st->den_rate <= st->oversample)
+   {
+      spx_uint32_t i;
+      if (!st->sinc_table)
+         st->sinc_table = (spx_word16_t *)speex_alloc(st->filt_len*st->den_rate*sizeof(spx_word16_t));
+      else if (st->sinc_table_length < st->filt_len*st->den_rate)
+      {
+         st->sinc_table = (spx_word16_t *)speex_realloc(st->sinc_table,st->filt_len*st->den_rate*sizeof(spx_word16_t));
+         st->sinc_table_length = st->filt_len*st->den_rate;
+      }
+      for (i=0;i<st->den_rate;i++)
+      {
+         spx_int32_t j;
+         for (j=0;j<st->filt_len;j++)
+         {
+            st->sinc_table[i*st->filt_len+j] = sinc(st->cutoff,((j-(spx_int32_t)st->filt_len/2+1)-((float)i)/st->den_rate), st->filt_len, quality_map[st->quality].window_func);
+         }
+      }
+#ifdef FIXED_POINT
+      st->resampler_ptr = resampler_basic_direct_single;
+#else
+      if (st->quality>8)
+         st->resampler_ptr = resampler_basic_direct_double;
+      else
+         st->resampler_ptr = resampler_basic_direct_single;
+#endif
+      /*fprintf (stderr, "resampler uses direct sinc table and normalised cutoff %f\n", cutoff);*/
+   } else {
+      spx_int32_t i;
+      if (!st->sinc_table)
+         st->sinc_table = (spx_word16_t *)speex_alloc((st->filt_len*st->oversample+8)*sizeof(spx_word16_t));
+      else if (st->sinc_table_length < st->filt_len*st->oversample+8)
+      {
+         st->sinc_table = (spx_word16_t *)speex_realloc(st->sinc_table,(st->filt_len*st->oversample+8)*sizeof(spx_word16_t));
+         st->sinc_table_length = st->filt_len*st->oversample+8;
+      }
+      for (i=-4;i<(spx_int32_t)(st->oversample*st->filt_len+4);i++)
+         st->sinc_table[i+4] = sinc(st->cutoff,(i/(float)st->oversample - st->filt_len/2), st->filt_len, quality_map[st->quality].window_func);
+#ifdef FIXED_POINT
+      st->resampler_ptr = resampler_basic_interpolate_single;
+#else
+      if (st->quality>8)
+         st->resampler_ptr = resampler_basic_interpolate_double;
+      else
+         st->resampler_ptr = resampler_basic_interpolate_single;
+#endif
+      /*fprintf (stderr, "resampler uses interpolated sinc table and normalised cutoff %f\n", cutoff);*/
+   }
+   st->int_advance = st->num_rate/st->den_rate;
+   st->frac_advance = st->num_rate%st->den_rate;
+
+   
+   /* Here's the place where we update the filter memory to take into account
+      the change in filter length. It's probably the messiest part of the code
+      due to handling of lots of corner cases. */
+   if (!st->mem)
+   {
+      spx_uint32_t i;
+      st->mem = (spx_word16_t*)speex_alloc(st->nb_channels*(st->filt_len-1) * sizeof(spx_word16_t));
+      for (i=0;i<st->nb_channels*(st->filt_len-1);i++)
+         st->mem[i] = 0;
+      st->mem_alloc_size = st->filt_len-1;
+      /*speex_warning("init filter");*/
+   } else if (!st->started)
+   {
+      spx_uint32_t i;
+      st->mem = (spx_word16_t*)speex_realloc(st->mem, st->nb_channels*(st->filt_len-1) * sizeof(spx_word16_t));
+      for (i=0;i<st->nb_channels*(st->filt_len-1);i++)
+         st->mem[i] = 0;
+      st->mem_alloc_size = st->filt_len-1;
+      /*speex_warning("reinit filter");*/
+   } else if (st->filt_len > old_length)
+   {
+      spx_int32_t i;
+      /* Increase the filter length */
+      /*speex_warning("increase filter size");*/
+      int old_alloc_size = st->mem_alloc_size;
+      if (st->filt_len-1 > st->mem_alloc_size)
+      {
+         st->mem = (spx_word16_t*)speex_realloc(st->mem, st->nb_channels*(st->filt_len-1) * sizeof(spx_word16_t));
+         st->mem_alloc_size = st->filt_len-1;
+      }
+      for (i=st->nb_channels-1;i>=0;i--)
+      {
+         spx_int32_t j;
+         spx_uint32_t olen = old_length;
+         /*if (st->magic_samples[i])*/
+         {
+            /* Try and remove the magic samples as if nothing had happened */
+            
+            /* FIXME: This is wrong but for now we need it to avoid going over the array bounds */
+            olen = old_length + 2*st->magic_samples[i];
+            for (j=old_length-2+st->magic_samples[i];j>=0;j--)
+               st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]] = st->mem[i*old_alloc_size+j];
+            for (j=0;j<st->magic_samples[i];j++)
+               st->mem[i*st->mem_alloc_size+j] = 0;
+            st->magic_samples[i] = 0;
+         }
+         if (st->filt_len > olen)
+         {
+            /* If the new filter length is still bigger than the "augmented" length */
+            /* Copy data going backward */
+            for (j=0;j<olen-1;j++)
+               st->mem[i*st->mem_alloc_size+(st->filt_len-2-j)] = st->mem[i*st->mem_alloc_size+(olen-2-j)];
+            /* Then put zeros for lack of anything better */
+            for (;j<st->filt_len-1;j++)
+               st->mem[i*st->mem_alloc_size+(st->filt_len-2-j)] = 0;
+            /* Adjust last_sample */
+            st->last_sample[i] += (st->filt_len - olen)/2;
+         } else {
+            /* Put back some of the magic! */
+            st->magic_samples[i] = (olen - st->filt_len)/2;
+            for (j=0;j<st->filt_len-1+st->magic_samples[i];j++)
+               st->mem[i*st->mem_alloc_size+j] = st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]];
+         }
+      }
+   } else if (st->filt_len < old_length)
+   {
+      spx_uint32_t i;
+      /* Reduce filter length, this a bit tricky. We need to store some of the memory as "magic"
+         samples so they can be used directly as input the next time(s) */
+      for (i=0;i<st->nb_channels;i++)
+      {
+         spx_uint32_t j;
+         spx_uint32_t old_magic = st->magic_samples[i];
+         st->magic_samples[i] = (old_length - st->filt_len)/2;
+         /* We must copy some of the memory that's no longer used */
+         /* Copy data going backward */
+         for (j=0;j<st->filt_len-1+st->magic_samples[i]+old_magic;j++)
+            st->mem[i*st->mem_alloc_size+j] = st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]];
+         st->magic_samples[i] += old_magic;
+      }
+   }
+
+}
+
+SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, spx_uint32_t in_rate, spx_uint32_t out_rate, int quality, int *err)
+{
+   return speex_resampler_init_frac(nb_channels, in_rate, out_rate, in_rate, out_rate, quality, err);
+}
+
+SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels, spx_uint32_t ratio_num, spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate, int quality, int *err)
+{
+   spx_uint32_t i;
+   SpeexResamplerState *st;
+   if (quality > 10 || quality < 0)
+   {
+      if (err)
+         *err = RESAMPLER_ERR_INVALID_ARG;
+      return NULL;
+   }
+   st = (SpeexResamplerState *)speex_alloc(sizeof(SpeexResamplerState));
+   st->initialised = 0;
+   st->started = 0;
+   st->in_rate = 0;
+   st->out_rate = 0;
+   st->num_rate = 0;
+   st->den_rate = 0;
+   st->quality = -1;
+   st->sinc_table_length = 0;
+   st->mem_alloc_size = 0;
+   st->filt_len = 0;
+   st->mem = 0;
+   st->resampler_ptr = 0;
+         
+   st->cutoff = 1.f;
+   st->nb_channels = nb_channels;
+   st->in_stride = 1;
+   st->out_stride = 1;
+   
+   /* Per channel data */
+   st->last_sample = (spx_int32_t*)speex_alloc(nb_channels*sizeof(int));
+   st->magic_samples = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(int));
+   st->samp_frac_num = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(int));
+   for (i=0;i<nb_channels;i++)
+   {
+      st->last_sample[i] = 0;
+      st->magic_samples[i] = 0;
+      st->samp_frac_num[i] = 0;
+   }
+
+   speex_resampler_set_quality(st, quality);
+   speex_resampler_set_rate_frac(st, ratio_num, ratio_den, in_rate, out_rate);
+
+   
+   update_filter(st);
+   
+   st->initialised = 1;
+   if (err)
+      *err = RESAMPLER_ERR_SUCCESS;
+
+   return st;
+}
+
+void speex_resampler_destroy(SpeexResamplerState *st)
+{
+   speex_free(st->mem);
+   speex_free(st->sinc_table);
+   speex_free(st->last_sample);
+   speex_free(st->magic_samples);
+   speex_free(st->samp_frac_num);
+   speex_free(st);
+}
+
+
+
+static int speex_resampler_process_native(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
+{
+   int j=0;
+   int N = st->filt_len;
+   int out_sample = 0;
+   spx_word16_t *mem;
+   spx_uint32_t tmp_out_len = 0;
+   mem = st->mem + channel_index * st->mem_alloc_size;
+   st->started = 1;
+   
+   /* Handle the case where we have samples left from a reduction in filter length */
+   if (st->magic_samples[channel_index])
+   {
+      int istride_save;
+      spx_uint32_t tmp_in_len;
+      spx_uint32_t tmp_magic;
+      
+      istride_save = st->in_stride;
+      tmp_in_len = st->magic_samples[channel_index];
+      tmp_out_len = *out_len;
+      /* magic_samples needs to be set to zero to avoid infinite recursion */
+      tmp_magic = st->magic_samples[channel_index];
+      st->magic_samples[channel_index] = 0;
+      st->in_stride = 1;
+      speex_resampler_process_native(st, channel_index, mem+N-1, &tmp_in_len, out, &tmp_out_len);
+      st->in_stride = istride_save;
+      /*speex_warning_int("extra samples:", tmp_out_len);*/
+      /* If we couldn't process all "magic" input samples, save the rest for next time */
+      if (tmp_in_len < tmp_magic)
+      {
+         spx_uint32_t i;
+         st->magic_samples[channel_index] = tmp_magic-tmp_in_len;
+         for (i=0;i<st->magic_samples[channel_index];i++)
+            mem[N-1+i]=mem[N-1+i+tmp_in_len];
+      }
+      out += tmp_out_len*st->out_stride;
+      *out_len -= tmp_out_len;
+   }
+   
+   /* Call the right resampler through the function ptr */
+   out_sample = st->resampler_ptr(st, channel_index, in, in_len, out, out_len);
+   
+   if (st->last_sample[channel_index] < (spx_int32_t)*in_len)
+      *in_len = st->last_sample[channel_index];
+   *out_len = out_sample+tmp_out_len;
+   st->last_sample[channel_index] -= *in_len;
+   
+   for (j=0;j<N-1-(spx_int32_t)*in_len;j++)
+      mem[j] = mem[j+*in_len];
+   if (in != NULL)
+   {
+      for (;j<N-1;j++)
+         mem[j] = in[st->in_stride*(j+*in_len-N+1)];
+   } else {
+      for (;j<N-1;j++)
+         mem[j] = 0;      
+   }
+   return RESAMPLER_ERR_SUCCESS;
+}
+
+#define FIXED_STACK_ALLOC 1024
+
+#ifdef FIXED_POINT
+int speex_resampler_process_float(SpeexResamplerState *st, spx_uint32_t channel_index, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
+{
+   spx_uint32_t i;
+   int istride_save, ostride_save;
+#ifdef VAR_ARRAYS
+   spx_word16_t x[*in_len];
+   spx_word16_t y[*out_len];
+   /*VARDECL(spx_word16_t *x);
+   VARDECL(spx_word16_t *y);
+   ALLOC(x, *in_len, spx_word16_t);
+   ALLOC(y, *out_len, spx_word16_t);*/
+   istride_save = st->in_stride;
+   ostride_save = st->out_stride;
+   if (in != NULL)
+   {
+      for (i=0;i<*in_len;i++)
+         x[i] = WORD2INT(in[i*st->in_stride]);
+      st->in_stride = st->out_stride = 1;
+      speex_resampler_process_native(st, channel_index, x, in_len, y, out_len);
+   } else {
+      st->in_stride = st->out_stride = 1;
+      speex_resampler_process_native(st, channel_index, NULL, in_len, y, out_len);
+   }
+   st->in_stride = istride_save;
+   st->out_stride = ostride_save;
+   for (i=0;i<*out_len;i++)
+      out[i*st->out_stride] = y[i];
+#else
+   spx_word16_t x[FIXED_STACK_ALLOC];
+   spx_word16_t y[FIXED_STACK_ALLOC];
+   spx_uint32_t ilen=*in_len, olen=*out_len;
+   istride_save = st->in_stride;
+   ostride_save = st->out_stride;
+   while (ilen && olen)
+   {
+      spx_uint32_t ichunk, ochunk;
+      ichunk = ilen;
+      ochunk = olen;
+      if (ichunk>FIXED_STACK_ALLOC)
+         ichunk=FIXED_STACK_ALLOC;
+      if (ochunk>FIXED_STACK_ALLOC)
+         ochunk=FIXED_STACK_ALLOC;
+      if (in != NULL)
+      {
+         for (i=0;i<ichunk;i++)
+            x[i] = WORD2INT(in[i*st->in_stride]);
+         st->in_stride = st->out_stride = 1;
+         speex_resampler_process_native(st, channel_index, x, &ichunk, y, &ochunk);
+      } else {
+         st->in_stride = st->out_stride = 1;
+         speex_resampler_process_native(st, channel_index, NULL, &ichunk, y, &ochunk);
+      }
+      st->in_stride = istride_save;
+      st->out_stride = ostride_save;
+      for (i=0;i<ochunk;i++)
+         out[i*st->out_stride] = y[i];
+      out += ochunk;
+      in += ichunk;
+      ilen -= ichunk;
+      olen -= ochunk;
+   }
+   *in_len -= ilen;
+   *out_len -= olen;   
+#endif
+   return RESAMPLER_ERR_SUCCESS;
+}
+int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
+{
+   return speex_resampler_process_native(st, channel_index, in, in_len, out, out_len);
+}
+#else
+int speex_resampler_process_float(SpeexResamplerState *st, spx_uint32_t channel_index, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
+{
+   return speex_resampler_process_native(st, channel_index, in, in_len, out, out_len);
+}
+int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
+{
+   spx_uint32_t i;
+   int istride_save, ostride_save;
+#ifdef VAR_ARRAYS
+   spx_word16_t x[*in_len];
+   spx_word16_t y[*out_len];
+   /*VARDECL(spx_word16_t *x);
+   VARDECL(spx_word16_t *y);
+   ALLOC(x, *in_len, spx_word16_t);
+   ALLOC(y, *out_len, spx_word16_t);*/
+   istride_save = st->in_stride;
+   ostride_save = st->out_stride;
+   if (in != NULL)
+   {
+      for (i=0;i<*in_len;i++)
+         x[i] = in[i*st->in_stride];
+      st->in_stride = st->out_stride = 1;
+      speex_resampler_process_native(st, channel_index, x, in_len, y, out_len);
+   } else {
+      st->in_stride = st->out_stride = 1;
+      speex_resampler_process_native(st, channel_index, NULL, in_len, y, out_len);
+   }
+   st->in_stride = istride_save;
+   st->out_stride = ostride_save;
+   for (i=0;i<*out_len;i++)
+      out[i*st->out_stride] = WORD2INT(y[i]);
+#else
+   spx_word16_t x[FIXED_STACK_ALLOC];
+   spx_word16_t y[FIXED_STACK_ALLOC];
+   spx_uint32_t ilen=*in_len, olen=*out_len;
+   istride_save = st->in_stride;
+   ostride_save = st->out_stride;
+   while (ilen && olen)
+   {
+      spx_uint32_t ichunk, ochunk;
+      ichunk = ilen;
+      ochunk = olen;
+      if (ichunk>FIXED_STACK_ALLOC)
+         ichunk=FIXED_STACK_ALLOC;
+      if (ochunk>FIXED_STACK_ALLOC)
+         ochunk=FIXED_STACK_ALLOC;
+      if (in != NULL)
+      {
+         for (i=0;i<ichunk;i++)
+            x[i] = in[i*st->in_stride];
+         st->in_stride = st->out_stride = 1;
+         speex_resampler_process_native(st, channel_index, x, &ichunk, y, &ochunk);
+      } else {
+         st->in_stride = st->out_stride = 1;
+         speex_resampler_process_native(st, channel_index, NULL, &ichunk, y, &ochunk);
+      }
+      st->in_stride = istride_save;
+      st->out_stride = ostride_save;
+      for (i=0;i<ochunk;i++)
+         out[i*st->out_stride] = WORD2INT(y[i]);
+      out += ochunk;
+      in += ichunk;
+      ilen -= ichunk;
+      olen -= ochunk;
+   }
+   *in_len -= ilen;
+   *out_len -= olen;   
+#endif
+   return RESAMPLER_ERR_SUCCESS;
+}
+#endif
+
+int speex_resampler_process_interleaved_float(SpeexResamplerState *st, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
+{
+   spx_uint32_t i;
+   int istride_save, ostride_save;
+   spx_uint32_t bak_len = *out_len;
+   istride_save = st->in_stride;
+   ostride_save = st->out_stride;
+   st->in_stride = st->out_stride = st->nb_channels;
+   for (i=0;i<st->nb_channels;i++)
+   {
+      *out_len = bak_len;
+      if (in != NULL)
+         speex_resampler_process_float(st, i, in+i, in_len, out+i, out_len);
+      else
+         speex_resampler_process_float(st, i, NULL, in_len, out+i, out_len);
+   }
+   st->in_stride = istride_save;
+   st->out_stride = ostride_save;
+   return RESAMPLER_ERR_SUCCESS;
+}
+
+               
+int speex_resampler_process_interleaved_int(SpeexResamplerState *st, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
+{
+   spx_uint32_t i;
+   int istride_save, ostride_save;
+   spx_uint32_t bak_len = *out_len;
+   istride_save = st->in_stride;
+   ostride_save = st->out_stride;
+   st->in_stride = st->out_stride = st->nb_channels;
+   for (i=0;i<st->nb_channels;i++)
+   {
+      *out_len = bak_len;
+      if (in != NULL)
+         speex_resampler_process_int(st, i, in+i, in_len, out+i, out_len);
+      else
+         speex_resampler_process_int(st, i, NULL, in_len, out+i, out_len);
+   }
+   st->in_stride = istride_save;
+   st->out_stride = ostride_save;
+   return RESAMPLER_ERR_SUCCESS;
+}
+
+int speex_resampler_set_rate(SpeexResamplerState *st, spx_uint32_t in_rate, spx_uint32_t out_rate)
+{
+   return speex_resampler_set_rate_frac(st, in_rate, out_rate, in_rate, out_rate);
+}
+
+void speex_resampler_get_rate(SpeexResamplerState *st, spx_uint32_t *in_rate, spx_uint32_t *out_rate)
+{
+   *in_rate = st->in_rate;
+   *out_rate = st->out_rate;
+}
+
+int speex_resampler_set_rate_frac(SpeexResamplerState *st, spx_uint32_t ratio_num, spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate)
+{
+   spx_uint32_t fact;
+   spx_uint32_t old_den;
+   spx_uint32_t i;
+   if (st->in_rate == in_rate && st->out_rate == out_rate && st->num_rate == ratio_num && st->den_rate == ratio_den)
+      return RESAMPLER_ERR_SUCCESS;
+   
+   old_den = st->den_rate;
+   st->in_rate = in_rate;
+   st->out_rate = out_rate;
+   st->num_rate = ratio_num;
+   st->den_rate = ratio_den;
+   /* FIXME: This is terribly inefficient, but who cares (at least for now)? */
+   for (fact=2;fact<=IMIN(st->num_rate, st->den_rate);fact++)
+   {
+      while ((st->num_rate % fact == 0) && (st->den_rate % fact == 0))
+      {
+         st->num_rate /= fact;
+         st->den_rate /= fact;
+      }
+   }
+      
+   if (old_den > 0)
+   {
+      for (i=0;i<st->nb_channels;i++)
+      {
+         st->samp_frac_num[i]=st->samp_frac_num[i]*st->den_rate/old_den;
+         /* Safety net */
+         if (st->samp_frac_num[i] >= st->den_rate)
+            st->samp_frac_num[i] = st->den_rate-1;
+      }
+   }
+   
+   if (st->initialised)
+      update_filter(st);
+   return RESAMPLER_ERR_SUCCESS;
+}
+
+void speex_resampler_get_ratio(SpeexResamplerState *st, spx_uint32_t *ratio_num, spx_uint32_t *ratio_den)
+{
+   *ratio_num = st->num_rate;
+   *ratio_den = st->den_rate;
+}
+
+int speex_resampler_set_quality(SpeexResamplerState *st, int quality)
+{
+   if (quality > 10 || quality < 0)
+      return RESAMPLER_ERR_INVALID_ARG;
+   if (st->quality == quality)
+      return RESAMPLER_ERR_SUCCESS;
+   st->quality = quality;
+   if (st->initialised)
+      update_filter(st);
+   return RESAMPLER_ERR_SUCCESS;
+}
+
+void speex_resampler_get_quality(SpeexResamplerState *st, int *quality)
+{
+   *quality = st->quality;
+}
+
+void speex_resampler_set_input_stride(SpeexResamplerState *st, spx_uint32_t stride)
+{
+   st->in_stride = stride;
+}
+
+void speex_resampler_get_input_stride(SpeexResamplerState *st, spx_uint32_t *stride)
+{
+   *stride = st->in_stride;
+}
+
+void speex_resampler_set_output_stride(SpeexResamplerState *st, spx_uint32_t stride)
+{
+   st->out_stride = stride;
+}
+
+void speex_resampler_get_output_stride(SpeexResamplerState *st, spx_uint32_t *stride)
+{
+   *stride = st->out_stride;
+}
+
+int speex_resampler_get_input_latency(SpeexResamplerState *st)
+{
+  return st->filt_len / 2;
+}
+
+int speex_resampler_get_output_latency(SpeexResamplerState *st)
+{
+  return ((st->filt_len / 2) * st->den_rate + (st->num_rate >> 1)) / st->num_rate;
+}
+
+int speex_resampler_skip_zeros(SpeexResamplerState *st)
+{
+   spx_uint32_t i;
+   for (i=0;i<st->nb_channels;i++)
+      st->last_sample[i] = st->filt_len/2;
+   return RESAMPLER_ERR_SUCCESS;
+}
+
+int speex_resampler_reset_mem(SpeexResamplerState *st)
+{
+   spx_uint32_t i;
+   for (i=0;i<st->nb_channels*(st->filt_len-1);i++)
+      st->mem[i] = 0;
+   return RESAMPLER_ERR_SUCCESS;
+}
+
+const char *speex_resampler_strerror(int err)
+{
+   switch (err)
+   {
+      case RESAMPLER_ERR_SUCCESS:
+         return "Success.";
+      case RESAMPLER_ERR_ALLOC_FAILED:
+         return "Memory allocation failed.";
+      case RESAMPLER_ERR_BAD_STATE:
+         return "Bad resampler state.";
+      case RESAMPLER_ERR_INVALID_ARG:
+         return "Invalid argument.";
+      case RESAMPLER_ERR_PTR_OVERLAP:
+         return "Input and output buffers overlap.";
+      default:
+         return "Unknown error. Bad error code or strange version mismatch.";
+   }
+}