* Sync Speex codec with Speex SVN revision 12449 (roughly Speex 1.2beta1).

* Redo the changes required to make Speex compile in Rockbox. Should be a bit easier to keep in sync with Speex SVN now.
* Fix name of Speex library in codecs Makefile.



git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12254 a1c6a512-1295-4272-9138-f99709370657

1 files changed, 625 insertions, 0 deletions

diff --git a/apps/codecs/libspeex/resample.c b/apps/codecs/libspeex/resample.c
new file mode 100644
index 0000000..d6bfa3e
--- /dev/null
+++ b/apps/codecs/libspeex/resample.c
@@ -0,0 +1,625 @@
+/* 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);}
+#else
+#include "misc.h"
+#endif
+
+#include <math.h>
+#include "speex/speex_resampler.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))
+
+struct QualityMapping {
+   int base_length;
+   int oversample;
+   float downsample_bandwidth;
+   float upsample_bandwidth;
+};
+
+/* 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, 0.70f, 0.80f}, /* 0 */
+   { 16,  4, 0.74f, 0.83f}, /* 1 */
+   { 32,  4, 0.77f, 0.87f}, /* 2 */
+   { 48,  8, 0.84f, 0.90f}, /* 3 */
+   { 64,  8, 0.88f, 0.92f}, /* 4 */
+   { 80,  8, 0.90f, 0.94f}, /* 5 */
+   { 96,  8, 0.91f, 0.94f}, /* 6 */
+   {128, 16, 0.93f, 0.95f}, /* 7 */
+   {160, 16, 0.94f, 0.96f}, /* 8 */
+   {192, 16, 0.95f, 0.96f}, /* 9 */
+   {256, 16, 0.96f, 0.97f}, /* 10 */
+};
+
+typedef enum {SPEEX_RESAMPLER_DIRECT_SINGLE=0, SPEEX_RESAMPLER_INTERPOLATE_SINGLE=1} SpeexSincType;
+
+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;
+   SpeexSincType type;
+} ;
+
+#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)
+{
+   /*fprintf (stderr, "%f ", x);*/
+   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*x)/(M_PI*x) * (.42+.5*cos(2*x*M_PI/N)+.08*cos(4*x*M_PI/N)));
+}
+#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)
+{
+   /*fprintf (stderr, "%f ", x);*/
+   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*x)/(M_PI*x) * (.42+.5*cos(2*x*M_PI/N)+.08*cos(4*x*M_PI/N));
+}
+#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;
+}
+
+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};
+      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++)
+      {
+         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]);
+      }
+         /* 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.f-interp[0]-interp[1]-interp[3];
+      /*sum = frac*accum[1] + (1-frac)*accum[2];*/
+      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;
+}
+
+
+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);
+         }
+      }
+      st->type = SPEEX_RESAMPLER_DIRECT_SINGLE;
+      st->resampler_ptr = resampler_basic_direct_single;
+      /*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);
+      st->type = SPEEX_RESAMPLER_INTERPOLATE_SINGLE;
+      st->resampler_ptr = resampler_basic_interpolate_single;
+      /*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 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(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)
+   {
+      int tmp_in_len;
+      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 */
+      st->magic_samples = 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);*/
+      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_set_rate(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_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_set_input_stride(SpeexResamplerState *st, int stride)
+{
+   st->in_stride = stride;
+}
+
+void speex_resampler_set_output_stride(SpeexResamplerState *st, int stride)
+{
+   st->out_stride = 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;
+}
+