Add rate resampler plugin based on speex

Added another rate resampler plugin based on speex code.
Light weight and much better quality.

From: Jean-Marc Valin <jean-marc.valin@usherbrooke.ca>

4 files changed, 1433 insertions, 0 deletions

diff --git a/pph/Makefile.am b/pph/Makefile.am
new file mode 100644
index 0000000..d695ef6
--- /dev/null
+++ b/pph/Makefile.am
@@ -0,0 +1,19 @@
+asound_module_rate_speexrate_LTLIBRARIES = libasound_module_rate_speexrate.la
+
+asound_module_rate_speexratedir = $(libdir)/alsa-lib
+
+AM_CFLAGS = -DOUTSIDE_SPEEX -Wall -g @ALSA_CFLAGS@
+AM_LDFLAGS = -module -avoid-version -export-dynamic
+
+libasound_module_rate_speexrate_la_SOURCES = rate_speexrate.c resample.c
+libasound_module_rate_speexrate_la_LIBADD = @ALSA_LIBS@
+
+install-exec-hook:
+	rm -f $(DESTDIR)$(libdir)/alsa-lib/libasound_module_rate_speexrate_*.so
+	$(LN_S) libasound_module_rate_speexrate.so $(DESTDIR)$(libdir)/alsa-lib/libasound_module_rate_speexrate_best.so
+	$(LN_S) libasound_module_rate_speexrate.so $(DESTDIR)$(libdir)/alsa-lib/libasound_module_rate_speexrate_medium.so
+
+uninstall-hook:
+	rm -f $(DESTDIR)$(libdir)/alsa-lib/libasound_module_rate_speexrate_*.so
+
+noinst_HEADERS = speex_resampler.h
diff --git a/pph/rate_speexrate.c b/pph/rate_speexrate.c
new file mode 100644
index 0000000..07679b4
--- /dev/null
+++ b/pph/rate_speexrate.c
@@ -0,0 +1,162 @@
+/* Rate converter plugin using Public Parrot Hack
+   Copyright (C) 2007 Jean-Marc Valin
+
+   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.
+*/
+
+#include <stdio.h>
+#include <samplerate.h>
+#include <alsa/asoundlib.h>
+#include <alsa/pcm_rate.h>
+
+#include "speex_resampler.h"
+
+struct rate_src {
+	int quality;
+	unsigned int channels;
+        SpeexResamplerState *st;
+};
+
+static snd_pcm_uframes_t input_frames(void *obj, snd_pcm_uframes_t frames)
+{
+   int num, den;
+   struct rate_src *rate = obj;
+   if (frames == 0)
+      return 0;
+   speex_resampler_get_ratio(rate->st, &num, &den);
+   return (snd_pcm_uframes_t)((frames*num+(den>>1))/den);
+}
+
+static snd_pcm_uframes_t output_frames(void *obj, snd_pcm_uframes_t frames)
+{
+   int num, den;
+   struct rate_src *rate = obj;
+   if (frames == 0)
+      return 0;
+   speex_resampler_get_ratio(rate->st, &num, &den);
+   return (snd_pcm_uframes_t)((frames*den+(num>>1))/num);
+}
+
+static void pcm_src_free(void *obj)
+{
+   struct rate_src *rate = obj;
+   if (rate->st)
+   {
+      speex_resampler_destroy(rate->st);
+      rate->st = NULL;
+   }
+}
+
+static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info)
+{
+   struct rate_src *rate = obj;
+   
+   if (! rate->st || rate->channels != info->channels) {
+      if (rate->st)
+         speex_resampler_destroy(rate->st);
+      rate->channels = info->channels;
+      rate->st = speex_resampler_init_frac(rate->channels, info->in.period_size, info->out.period_size, info->in.rate, info->out.rate, rate->quality);
+      if (! rate->st)
+         return -EINVAL;
+   }
+
+   return 0;
+}
+
+static int pcm_src_adjust_pitch(void *obj, snd_pcm_rate_info_t *info)
+{
+   struct rate_src *rate = obj;
+   speex_resampler_set_rate_frac(rate->st, info->in.period_size, info->out.period_size, info->in.rate, info->out.rate);
+   return 0;
+}
+
+static void pcm_src_reset(void *obj)
+{
+   struct rate_src *rate = obj;
+   speex_resampler_reset_mem(rate->st);
+}
+
+static void pcm_src_convert_s16(void *obj, int16_t *dst, unsigned int dst_frames,
+				const int16_t *src, unsigned int src_frames)
+{
+   struct rate_src *rate = obj;
+   speex_resampler_process_interleaved_int(rate->st, src, &src_frames, dst, &dst_frames);
+}
+
+static void pcm_src_close(void *obj)
+{
+   free(obj);
+}
+
+static snd_pcm_rate_ops_t pcm_src_ops = {
+	.close = pcm_src_close,
+	.init = pcm_src_init,
+	.free = pcm_src_free,
+	.reset = pcm_src_reset,
+	.adjust_pitch = pcm_src_adjust_pitch,
+	.convert_s16 = pcm_src_convert_s16,
+	.input_frames = input_frames,
+	.output_frames = output_frames,
+};
+
+static int pcm_src_open(unsigned int version, void **objp,
+			snd_pcm_rate_ops_t *ops, int quality)
+{
+	struct rate_src *rate;
+
+	if (version != SND_PCM_RATE_PLUGIN_VERSION) {
+		fprintf(stderr, "Invalid rate plugin version %x\n", version);
+		return -EINVAL;
+	}
+
+	rate = calloc(1, sizeof(*rate));
+	if (! rate)
+		return -ENOMEM;
+	rate->quality = quality;
+
+	*objp = rate;
+	*ops = pcm_src_ops;
+	return 0;
+}
+
+int SND_PCM_RATE_PLUGIN_ENTRY(speexrate) (unsigned int version, void **objp,
+					   snd_pcm_rate_ops_t *ops)
+{
+	return pcm_src_open(version, objp, ops, 3);
+}
+
+int SND_PCM_RATE_PLUGIN_ENTRY(speexrate_best) (unsigned int version, void **objp,
+						snd_pcm_rate_ops_t *ops)
+{
+	return pcm_src_open(version, objp, ops, 10);
+}
+
+int SND_PCM_RATE_PLUGIN_ENTRY(speexrate_medium) (unsigned int version, void **objp,
+						  snd_pcm_rate_ops_t *ops)
+{
+	return pcm_src_open(version, objp, ops, 5);
+}
diff --git a/pph/resample.c b/pph/resample.c
new file mode 100644
index 0000000..8255b83
--- /dev/null
+++ b/pph/resample.c
@@ -0,0 +1,933 @@
+/* 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)
+
+   The code is working, but it's in a very early stage, so it may have
+   artifacts, noise or subliminal messages from satan. Also, the API 
+   isn't stable and I can actually promise that I *will* change the API
+   some time in the future.
+
+TODO list:
+      - Variable calculation resolution depending on quality setting
+         - Single vs double in float mode
+         - 16-bit vs 32-bit (sinc only) in fixed-point mode
+      - Make sure the filter update works even when changing params 
+             after only a few samples procesed
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#ifdef OUTSIDE_SPEEX
+#include <stdlib.h>
+void *speex_alloc (int size) {return calloc(size,1);}
+void *speex_realloc (void *ptr, int size) {return realloc(ptr, size);}
+void speex_free (void *ptr) {free(ptr);}
+#include "speex_resampler.h"
+#else
+#include "speex/speex_resampler.h"
+#include "misc.h"
+#endif
+
+#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))
+
+
+typedef int (*resampler_basic_func)(SpeexResamplerState *, int , const spx_word16_t *, int *, spx_word16_t *, int *);
+
+struct SpeexResamplerState_ {
+   int    in_rate;
+   int    out_rate;
+   int    num_rate;
+   int    den_rate;
+   
+   int    quality;
+   int    nb_channels;
+   int    filt_len;
+   int    mem_alloc_size;
+   int    int_advance;
+   int    frac_advance;
+   float  cutoff;
+   int    oversample;
+   int    initialised;
+   int    started;
+   
+   /* These are per-channel */
+   int    *last_sample;
+   int    *samp_frac_num;
+   int    *magic_samples;
+   
+   spx_word16_t *mem;
+   spx_word16_t *sinc_table;
+   int    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).
+*/
+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,  8, 0.922f, 0.940f, KAISER10}, /* Q5 */  /* 89.1% cutoff (~100 dB stop) 10 */
+   { 96,  8, 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, 16, 0.968f, 0.968f, KAISER12}, /* Q9 */  /* 95.5% cutoff (~100 dB stop) 10 */
+   {256, 16, 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, int channel_index, const spx_word16_t *in, int *in_len, spx_word16_t *out, int *out_len)
+{
+   int N = st->filt_len;
+   int out_sample = 0;
+   spx_word16_t *mem;
+   int last_sample = st->last_sample[channel_index];
+   int samp_frac_num = st->samp_frac_num[channel_index];
+   mem = st->mem + channel_index * st->mem_alloc_size;
+   while (!(last_sample >= *in_len || out_sample >= *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, int channel_index, const spx_word16_t *in, int *in_len, spx_word16_t *out, int *out_len)
+{
+   int N = st->filt_len;
+   int out_sample = 0;
+   spx_word16_t *mem;
+   int last_sample = st->last_sample[channel_index];
+   int samp_frac_num = st->samp_frac_num[channel_index];
+   mem = st->mem + channel_index * st->mem_alloc_size;
+   while (!(last_sample >= *in_len || out_sample >= *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, int channel_index, const spx_word16_t *in, int *in_len, spx_word16_t *out, int *out_len)
+{
+   int N = st->filt_len;
+   int out_sample = 0;
+   spx_word16_t *mem;
+   int last_sample = st->last_sample[channel_index];
+   int samp_frac_num = st->samp_frac_num[channel_index];
+   mem = st->mem + channel_index * st->mem_alloc_size;
+   while (!(last_sample >= *in_len || out_sample >= *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, int channel_index, const spx_word16_t *in, int *in_len, spx_word16_t *out, int *out_len)
+{
+   int N = st->filt_len;
+   int out_sample = 0;
+   spx_word16_t *mem;
+   int last_sample = st->last_sample[channel_index];
+   int samp_frac_num = st->samp_frac_num[channel_index];
+   mem = st->mem + channel_index * st->mem_alloc_size;
+   while (!(last_sample >= *in_len || out_sample >= *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)
+{
+   int i;
+   int 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);
+   } 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)
+   {
+      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++)
+      {
+         int j;
+         for (j=0;j<st->filt_len;j++)
+         {
+            st->sinc_table[i*st->filt_len+j] = sinc(st->cutoff,((j-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 {
+      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<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;
+
+   if (!st->mem)
+   {
+      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)
+   {
+      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)
+   {
+      /* 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=0;i<st->nb_channels;i++)
+      {
+         int j;
+         /* Copy data going backward */
+         for (j=0;j<old_length-1;j++)
+            st->mem[i*st->mem_alloc_size+(st->filt_len-2-j)] = st->mem[i*old_alloc_size+(old_length-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 - old_length)/2;
+      }
+   } else if (st->filt_len < old_length)
+   {
+      /* Reduce filter length, this a bit tricky */
+      /*speex_warning("decrease filter size (unimplemented)");*/
+      /* Adjust last_sample (which will likely end up negative) */
+      /*st->last_sample += (st->filt_len - old_length)/2;*/
+      for (i=0;i<st->nb_channels;i++)
+      {
+         int j;
+         st->magic_samples[i] = (old_length - st->filt_len)/2;
+         /* Copy data going backward */
+         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]];
+      }
+   }
+
+}
+
+SpeexResamplerState *speex_resampler_init(int nb_channels, int in_rate, int out_rate, int quality)
+{
+   return speex_resampler_init_frac(nb_channels, in_rate, out_rate, in_rate, out_rate, quality);
+}
+
+SpeexResamplerState *speex_resampler_init_frac(int nb_channels, int ratio_num, int ratio_den, int in_rate, int out_rate, int quality)
+{
+   int i;
+   SpeexResamplerState *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 = (int*)speex_alloc(nb_channels*sizeof(int));
+   st->magic_samples = (int*)speex_alloc(nb_channels*sizeof(int));
+   st->samp_frac_num = (int*)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;
+   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 void speex_resampler_process_native(SpeexResamplerState *st, int channel_index, const spx_word16_t *in, int *in_len, spx_word16_t *out, int *out_len)
+{
+   int j=0;
+   int N = st->filt_len;
+   int out_sample = 0;
+   spx_word16_t *mem;
+   int 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 tmp_in_len;
+      int tmp_magic;
+      tmp_in_len = st->magic_samples[channel_index];
+      tmp_out_len = *out_len;
+      /* FIXME: Need to handle the case where the out array is too small */
+      /* 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;
+      speex_resampler_process_native(st, channel_index, mem+N-1, &tmp_in_len, out, &tmp_out_len);
+      /*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)
+      {
+         int 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;
+   }
+   
+   /* 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] < *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-*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)];
+   
+}
+
+#ifdef FIXED_POINT
+void speex_resampler_process_float(SpeexResamplerState *st, int channel_index, const float *in, int *in_len, float *out, int *out_len)
+{
+   int i;
+   int istride_save, ostride_save;
+   spx_word16_t x[*in_len];
+   spx_word16_t y[*out_len];
+   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];
+}
+void speex_resampler_process_int(SpeexResamplerState *st, int channel_index, const spx_int16_t *in, int *in_len, spx_int16_t *out, int *out_len)
+{
+   speex_resampler_process_native(st, channel_index, in, in_len, out, out_len);
+}
+#else
+void speex_resampler_process_float(SpeexResamplerState *st, int channel_index, const float *in, int *in_len, float *out, int *out_len)
+{
+   speex_resampler_process_native(st, channel_index, in, in_len, out, out_len);
+}
+void speex_resampler_process_int(SpeexResamplerState *st, int channel_index, const spx_int16_t *in, int *in_len, spx_int16_t *out, int *out_len)
+{
+   int i;
+   int istride_save, ostride_save;
+   spx_word16_t x[*in_len];
+   spx_word16_t y[*out_len];
+   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]);
+}
+#endif
+
+void speex_resampler_process_interleaved_float(SpeexResamplerState *st, const float *in, int *in_len, float *out, int *out_len)
+{
+   int i;
+   int istride_save, ostride_save;
+   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++)
+   {
+      speex_resampler_process_float(st, i, in+i, in_len, out+i, out_len);
+   }
+   st->in_stride = istride_save;
+   st->out_stride = ostride_save;
+}
+
+void speex_resampler_process_interleaved_int(SpeexResamplerState *st, const spx_int16_t *in, int *in_len, spx_int16_t *out, int *out_len)
+{
+   int i;
+   int istride_save, ostride_save;
+   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++)
+   {
+      speex_resampler_process_int(st, i, in+i, in_len, out+i, out_len);
+   }
+   st->in_stride = istride_save;
+   st->out_stride = ostride_save;
+}
+
+void speex_resampler_set_rate(SpeexResamplerState *st, int in_rate, int out_rate)
+{
+   speex_resampler_set_rate_frac(st, in_rate, out_rate, in_rate, out_rate);
+}
+
+void speex_resampler_get_rate(SpeexResamplerState *st, int *in_rate, int *out_rate)
+{
+   *in_rate = st->in_rate;
+   *out_rate = st->out_rate;
+}
+
+void speex_resampler_set_rate_frac(SpeexResamplerState *st, int ratio_num, int ratio_den, int in_rate, int out_rate)
+{
+   int fact;
+   if (st->in_rate == in_rate && st->out_rate == out_rate && st->num_rate == ratio_num && st->den_rate == ratio_den)
+      return;
+   
+   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<=sqrt(IMAX(in_rate, out_rate));fact++)
+   {
+      while ((st->num_rate % fact == 0) && (st->den_rate % fact == 0))
+      {
+         st->num_rate /= fact;
+         st->den_rate /= fact;
+      }
+   }
+      
+   if (st->initialised)
+      update_filter(st);
+}
+
+void speex_resampler_get_ratio(SpeexResamplerState *st, int *ratio_num, int *ratio_den)
+{
+   *ratio_num = st->num_rate;
+   *ratio_den = st->den_rate;
+}
+
+void speex_resampler_set_quality(SpeexResamplerState *st, int quality)
+{
+   if (quality < 0)
+      quality = 0;
+   if (quality > 10)
+      quality = 10;
+   if (st->quality == quality)
+      return;
+   st->quality = quality;
+   if (st->initialised)
+      update_filter(st);
+}
+
+void speex_resampler_get_quality(SpeexResamplerState *st, int *quality)
+{
+   *quality = st->quality;
+}
+
+void speex_resampler_set_input_stride(SpeexResamplerState *st, int stride)
+{
+   st->in_stride = stride;
+}
+
+void speex_resampler_get_input_stride(SpeexResamplerState *st, int *stride)
+{
+   *stride = st->in_stride;
+}
+
+void speex_resampler_set_output_stride(SpeexResamplerState *st, int stride)
+{
+   st->out_stride = stride;
+}
+
+void speex_resampler_get_output_stride(SpeexResamplerState *st, int *stride)
+{
+   *stride = st->out_stride;
+}
+
+void speex_resampler_skip_zeros(SpeexResamplerState *st)
+{
+   int i;
+   for (i=0;i<st->nb_channels;i++)
+      st->last_sample[i] = st->filt_len/2;
+}
+
+void speex_resampler_reset_mem(SpeexResamplerState *st)
+{
+   int i;
+   for (i=0;i<st->nb_channels*(st->filt_len-1);i++)
+      st->mem[i] = 0;
+}
+
diff --git a/pph/speex_resampler.h b/pph/speex_resampler.h
new file mode 100644
index 0000000..8ba6790
--- /dev/null
+++ b/pph/speex_resampler.h
@@ -0,0 +1,319 @@
+/* Copyright (C) 2007 Jean-Marc Valin
+      
+   File: speex_resampler.h
+   Resampling code
+      
+   The design goals of this code are:
+      - Very fast algorithm
+      - Low memory requirement
+      - Good *perceptual* quality (and not best SNR)
+
+   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.
+*/
+
+
+#ifndef SPEEX_RESAMPLER_H
+#define SPEEX_RESAMPLER_H
+
+#ifdef OUTSIDE_SPEEX
+
+/********* WARNING: MENTAL SANITY ENDS HERE *************/
+
+/* If the resampler is defined outside of Speex, we change the symbol names so that 
+   there won't be any clash if linking with Speex later on. */
+
+#define RANDOM_PREFIX ALSA_PUBLIC_PARROT_HACK_PLUGIN
+#ifndef RANDOM_PREFIX
+#error "Please define RANDOM_PREFIX (above) to something specific to your project to prevent symbol name clashes"
+#endif
+
+#define CAT_PREFIX2(a,b) a ## b
+#define CAT_PREFIX(a,b) CAT_PREFIX2(a, b)
+      
+#define speex_resampler_init CAT_PREFIX(RANDOM_PREFIX,_resampler_init)
+#define speex_resampler_init_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_init_frac)
+#define speex_resampler_destroy CAT_PREFIX(RANDOM_PREFIX,_resampler_destroy)
+#define speex_resampler_process_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_float)
+#define speex_resampler_process_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_int)
+#define speex_resampler_process_interleaved_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_float)
+#define speex_resampler_process_interleaved_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_int)
+#define speex_resampler_set_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate)
+#define speex_resampler_get_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_get_rate)
+#define speex_resampler_set_rate_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate_frac)
+#define speex_resampler_get_ratio CAT_PREFIX(RANDOM_PREFIX,_resampler_get_ratio)
+#define speex_resampler_set_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_set_quality)
+#define speex_resampler_get_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_get_quality)
+#define speex_resampler_set_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_input_stride)
+#define speex_resampler_get_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_stride)
+#define speex_resample_set_output_stride CAT_PREFIX(RANDOM_PREFIX,_resample_set_output_stride)
+#define speex_resample_get_output_stride CAT_PREFIX(RANDOM_PREFIX,_resample_get_output_stride)
+#define speex_resampler_skip_zeros CAT_PREFIX(RANDOM_PREFIX,_resampler_skip_zeros)
+#define speex_resampler_reset_mem CAT_PREFIX(RANDOM_PREFIX,_resampler_reset_mem)
+
+#define spx_int16_t short
+
+#ifdef FIXED_POINT
+#define spx_word16_t short
+#define spx_word32_t int
+
+#else /* FIXED_POINT */
+
+#define spx_word16_t float
+#define spx_word32_t float
+#define MULT16_16(a,b) ((a)*(b))
+#define MULT16_32_Q15(a,b) ((a)*(b))
+#define PSHR32(a,b) (a)
+#endif /* FIXED_POINT */
+
+#else /* OUTSIDE_SPEEX */
+
+#include "speex/speex_types.h"
+
+#endif /* OUTSIDE_SPEEX */
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#define SPEEX_RESAMPLER_QUALITY_MAX 10
+#define SPEEX_RESAMPLER_QUALITY_MIN 0
+#define SPEEX_RESAMPLER_QUALITY_DEFAULT 4
+#define SPEEX_RESAMPLER_QUALITY_VOIP 3
+#define SPEEX_RESAMPLER_QUALITY_DESKTOP 5
+   
+struct SpeexResamplerState_;
+typedef struct SpeexResamplerState_ SpeexResamplerState;
+
+/** Create a new resampler with integer input and output rates.
+ * @param nb_channels Number of channels to be processed
+ * @param in_rate Input sampling rate (integer number of Hz).
+ * @param out_rate Output sampling rate (integer number of Hz).
+ * @param quality Resampling quality between 0 and 10, where 0 has poor quality
+ * and 10 has very high quality.
+ * @return Newly created resampler state
+ * @retval NULL Error: not enough memory
+ */
+SpeexResamplerState *speex_resampler_init(int nb_channels, 
+                                          int in_rate, 
+                                          int out_rate, 
+                                          int quality);
+
+/** Create a new resampler with fractional input/output rates. The sampling 
+ * rate ratio is an arbitrary rational number with both the numerator and 
+ * denominator being 32-bit integers.
+ * @param nb_channels Number of channels to be processed
+ * @param ratio_num Numerator of the sampling rate ratio
+ * @param ratio_den Denominator of the sampling rate ratio
+ * @param in_rate Input sampling rate rounded to the nearest integer (in Hz).
+ * @param out_rate Output sampling rate rounded to the nearest integer (in Hz).
+ * @param quality Resampling quality between 0 and 10, where 0 has poor quality
+ * and 10 has very high quality.
+ * @return Newly created resampler state
+ * @retval NULL Error: not enough memory
+ */
+SpeexResamplerState *speex_resampler_init_frac(int nb_channels, 
+                                               int ratio_num, 
+                                               int ratio_den, 
+                                               int in_rate, 
+                                               int out_rate, 
+                                               int quality);
+
+/** Destroy a resampler state.
+ * @param st Resampler state
+ */
+void speex_resampler_destroy(SpeexResamplerState *st);
+
+/** Resample a float array. The input and output buffers must *not* overlap.
+ * @param st Resampler state
+ * @param channel_index Index of the channel to process for the multi-channel 
+ * base (0 otherwise)
+ * @param in Input buffer
+ * @param in_len Number of input samples in the input buffer. Returns the 
+ * number of samples processed
+ * @param out Output buffer
+ * @param out_len Size of the output buffer. Returns the number of samples written
+ */
+void speex_resampler_process_float(SpeexResamplerState *st, 
+                                   int channel_index, 
+                                   const float *in, 
+                                   int *in_len, 
+                                   float *out, 
+                                   int *out_len);
+
+/** Resample an int array. The input and output buffers must *not* overlap.
+ * @param st Resampler state
+ * @param channel_index Index of the channel to process for the multi-channel 
+ * base (0 otherwise)
+ * @param in Input buffer
+ * @param in_len Number of input samples in the input buffer. Returns the number
+ * of samples processed
+ * @param out Output buffer
+ * @param out_len Size of the output buffer. Returns the number of samples written
+ */
+void speex_resampler_process_int(SpeexResamplerState *st, 
+                                 int channel_index, 
+                                 const spx_int16_t *in, 
+                                 int *in_len, 
+                                 spx_int16_t *out, 
+                                 int *out_len);
+
+/** Resample an interleaved float array. The input and output buffers must *not* overlap.
+ * @param st Resampler state
+ * @param in Input buffer
+ * @param in_len Number of input samples in the input buffer. Returns the number
+ * of samples processed. This is all per-channel.
+ * @param out Output buffer
+ * @param out_len Size of the output buffer. Returns the number of samples written.
+ * This is all per-channel.
+ */
+void speex_resampler_process_interleaved_float(SpeexResamplerState *st, 
+                                               const float *in, 
+                                               int *in_len, 
+                                               float *out, 
+                                               int *out_len);
+
+/** Resample an interleaved int array. The input and output buffers must *not* overlap.
+ * @param st Resampler state
+ * @param in Input buffer
+ * @param in_len Number of input samples in the input buffer. Returns the number
+ * of samples processed. This is all per-channel.
+ * @param out Output buffer
+ * @param out_len Size of the output buffer. Returns the number of samples written.
+ * This is all per-channel.
+ */
+void speex_resampler_process_interleaved_int(SpeexResamplerState *st, 
+                                             const spx_int16_t *in, 
+                                             int *in_len, 
+                                             spx_int16_t *out, 
+                                             int *out_len);
+
+/** Set (change) the input/output sampling rates (integer value).
+ * @param st Resampler state
+ * @param in_rate Input sampling rate (integer number of Hz).
+ * @param out_rate Output sampling rate (integer number of Hz).
+ */
+void speex_resampler_set_rate(SpeexResamplerState *st, 
+                              int in_rate, 
+                              int out_rate);
+
+/** Get the current input/output sampling rates (integer value).
+ * @param st Resampler state
+ * @param in_rate Input sampling rate (integer number of Hz) copied.
+ * @param out_rate Output sampling rate (integer number of Hz) copied.
+ */
+void speex_resampler_get_rate(SpeexResamplerState *st, 
+                              int *in_rate, 
+                              int *out_rate);
+
+/** Set (change) the input/output sampling rates and resampling ratio 
+ * (fractional values in Hz supported).
+ * @param st Resampler state
+ * @param ratio_num Numerator of the sampling rate ratio
+ * @param ratio_den Denominator of the sampling rate ratio
+ * @param in_rate Input sampling rate rounded to the nearest integer (in Hz).
+ * @param out_rate Output sampling rate rounded to the nearest integer (in Hz).
+ */
+void speex_resampler_set_rate_frac(SpeexResamplerState *st, 
+                                   int ratio_num, 
+                                   int ratio_den, 
+                                   int in_rate, 
+                                   int out_rate);
+
+/** Get the current resampling ratio. This will be reduced to the least
+ * common denominator.
+ * @param st Resampler state
+ * @param ratio_num Numerator of the sampling rate ratio copied
+ * @param ratio_den Denominator of the sampling rate ratio copied
+ */
+void speex_resampler_get_ratio(SpeexResamplerState *st, 
+                                   int *ratio_num, 
+                                   int *ratio_den);
+
+/** Set (change) the conversion quality.
+ * @param st Resampler state
+ * @param quality Resampling quality between 0 and 10, where 0 has poor 
+ * quality and 10 has very high quality.
+ */
+void speex_resampler_set_quality(SpeexResamplerState *st, 
+                                 int quality);
+
+/** Get the conversion quality.
+ * @param st Resampler state
+ * @param quality Resampling quality between 0 and 10, where 0 has poor 
+ * quality and 10 has very high quality.
+ */
+void speex_resampler_get_quality(SpeexResamplerState *st, 
+                                 int *quality);
+
+/** Set (change) the input stride.
+ * @param st Resampler state
+ * @param stride Input stride
+ */
+void speex_resampler_set_input_stride(SpeexResamplerState *st, 
+                                      int stride);
+
+/** Get the input stride.
+ * @param st Resampler state
+ * @param stride Input stride copied
+ */
+void speex_resampler_get_input_stride(SpeexResamplerState *st, 
+                                      int *stride);
+
+/** Set (change) the output stride.
+ * @param st Resampler state
+ * @param stride Output stride
+ */
+void speex_resample_set_output_stride(SpeexResamplerState *st, 
+                                      int stride);
+
+/** Get the output stride.
+ * @param st Resampler state copied
+ * @param stride Output stride
+ */
+void speex_resample_get_output_stride(SpeexResamplerState *st, 
+                                      int *stride);
+
+/** Make sure that the first samples to go out of the resamplers don't have 
+ * leading zeros. This is only useful before starting to use a newly created 
+ * resampler. It is recommended to use that when resampling an audio file, as
+ * it will generate a file with the same length. For real-time processing,
+ * it is probably easier not to use this call (so that the output duration
+ * is the same for the first frame).
+ * @param st Resampler state
+ */
+void speex_resampler_skip_zeros(SpeexResamplerState *st);
+
+/** Reset a resampler so a new (unrelated) stream can be processed.
+ * @param st Resampler state
+ */
+void speex_resampler_reset_mem(SpeexResamplerState *st);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif