Add resample element based on the Speex resampling algorithm.

Original commit message from CVS:
* configure.ac:
* gst/speexresample/arch.h:
* gst/speexresample/fixed_generic.h:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_base_init), (gst_speex_resample_class_init),
(gst_speex_resample_init), (gst_speex_resample_start),
(gst_speex_resample_stop), (gst_speex_resample_get_unit_size),
(gst_speex_resample_transform_caps),
(gst_speex_resample_init_state), (gst_speex_resample_update_state),
(gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
(gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
(gst_speex_resample_event), (gst_speex_resample_check_discont),
(gst_speex_resample_process), (gst_speex_resample_transform),
(gst_speex_resample_set_property),
(gst_speex_resample_get_property), (plugin_init):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
(speex_free), (compute_func), (main), (sinc), (cubic_coef),
(resampler_basic_direct_single), (resampler_basic_direct_double),
(resampler_basic_interpolate_single),
(resampler_basic_interpolate_double), (update_filter),
(speex_resampler_init), (speex_resampler_init_frac),
(speex_resampler_destroy), (speex_resampler_process_native),
(speex_resampler_process_float), (speex_resampler_process_int),
(speex_resampler_process_interleaved_float),
(speex_resampler_process_interleaved_int),
(speex_resampler_set_rate), (speex_resampler_get_rate),
(speex_resampler_set_rate_frac), (speex_resampler_get_ratio),
(speex_resampler_set_quality), (speex_resampler_get_quality),
(speex_resampler_set_input_stride),
(speex_resampler_get_input_stride),
(speex_resampler_set_output_stride),
(speex_resampler_get_output_stride), (speex_resampler_skip_zeros),
(speex_resampler_reset_mem), (speex_resampler_strerror):
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_float.c:
* gst/speexresample/speex_resampler_int.c:
* gst/speexresample/speex_resampler_wrapper.h:
Add resample element based on the Speex resampling algorithm.

1 files changed, 1310 insertions, 0 deletions

diff --git a/gst/speexresample/resample.c b/gst/speexresample/resample.c
new file mode 100644
index 00000000..f3c97fdd
--- /dev/null
+++ b/gst/speexresample/resample.c
@@ -0,0 +1,1310 @@
+/* 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 */
+    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 */
+    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]);
+    }
+    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]);
+    }
+    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];
+  for (; j < N - 1; j++)
+    mem[j] = in[st->in_stride * (j + *in_len - N + 1)];
+
+  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;
+  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);
+  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;
+    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);
+    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;
+  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);
+  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;
+    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);
+    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;
+    speex_resampler_process_float (st, i, in + i, 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;
+    speex_resampler_process_int (st, i, in + i, 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_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.";
+  }
+}