* Clean up speex.c a little.

* Sync to Speex SVN 12735 which includes some of our warnings fixes.
* Move decoder output to IRAM. Not much perfomance gain though.



git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12735 a1c6a512-1295-4272-9138-f99709370657

1 files changed, 370 insertions, 62 deletions

diff --git a/apps/codecs/libspeex/resample.c b/apps/codecs/libspeex/resample.c
index d6bfa3e..74d5f40 100644
--- a/apps/codecs/libspeex/resample.c
+++ b/apps/codecs/libspeex/resample.c
@@ -59,12 +59,13 @@ TODO list:
 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>
-#include "speex/speex_resampler.h"
 
 #ifndef M_PI
 #define M_PI 3.14159263
@@ -82,37 +83,6 @@ void speex_free (void *ptr) {free(ptr);}
 
 #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 *);
 
@@ -145,34 +115,190 @@ struct SpeexResamplerState_ {
          
    int    in_stride;
    int    out_stride;
-   SpeexSincType type;
 } ;
 
+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,  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)
+static spx_word16_t sinc(float cutoff, float x, int N, struct FuncDef *window_func)
 {
    /*fprintf (stderr, "%f ", x);*/
-   x *= cutoff;
+   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*x)/(M_PI*x) * (.42+.5*cos(2*x*M_PI/N)+.08*cos(4*x*M_PI/N)));
+   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)
+static spx_word16_t sinc(float cutoff, float x, int N, struct FuncDef *window_func)
 {
    /*fprintf (stderr, "%f ", x);*/
-   x *= cutoff;
+   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*x)/(M_PI*x) * (.42+.5*cos(2*x*M_PI/N)+.08*cos(4*x*M_PI/N));
+   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
 
@@ -221,6 +347,55 @@ static int resampler_basic_direct_single(SpeexResamplerState *st, int channel_in
    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;
@@ -236,6 +411,75 @@ static int resampler_basic_interpolate_single(SpeexResamplerState *st, int chann
       
       /* 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;
@@ -246,7 +490,7 @@ static int resampler_basic_interpolate_single(SpeexResamplerState *st, int chann
       have only two accumulators */
       for (j=0;last_sample-N+1+j < 0;j++)
       {
-         spx_word16_t curr_mem = mem[last_sample+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]);
@@ -256,22 +500,14 @@ static int resampler_basic_interpolate_single(SpeexResamplerState *st, int chann
       /* Do the new part */
       for (;j<N;j++)
       {
-         spx_word16_t curr_in = *ptr;
+         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]);
       }
-         /* 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];*/
+      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);
@@ -289,7 +525,7 @@ static int resampler_basic_interpolate_single(SpeexResamplerState *st, int chann
    st->samp_frac_num[channel_index] = samp_frac_num;
    return out_sample;
 }
-
+#endif
 
 static void update_filter(SpeexResamplerState *st)
 {
@@ -328,11 +564,17 @@ static void update_filter(SpeexResamplerState *st)
          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->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);
          }
       }
-      st->type = SPEEX_RESAMPLER_DIRECT_SINGLE;
+#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)
@@ -343,9 +585,15 @@ static void update_filter(SpeexResamplerState *st)
          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->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;
@@ -405,8 +653,12 @@ static void update_filter(SpeexResamplerState *st)
 
 }
 
+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(int nb_channels, int ratio_num, int ratio_den, int in_rate, int out_rate, int 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));
@@ -440,7 +692,7 @@ SpeexResamplerState *speex_resampler_init(int nb_channels, int ratio_num, int ra
    }
 
    speex_resampler_set_quality(st, quality);
-   speex_resampler_set_rate(st, ratio_num, ratio_den, in_rate, out_rate);
+   speex_resampler_set_rate_frac(st, ratio_num, ratio_den, in_rate, out_rate);
 
    
    update_filter(st);
@@ -472,16 +724,26 @@ static void speex_resampler_process_native(SpeexResamplerState *st, int channel_
    st->started = 1;
    
    /* Handle the case where we have samples left from a reduction in filter length */
-   if (st->magic_samples)
+   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 */
-      st->magic_samples = 0;
+      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;
    }
    
@@ -536,13 +798,13 @@ void speex_resampler_process_int(SpeexResamplerState *st, int channel_index, con
    istride_save = st->in_stride;
    ostride_save = st->out_stride;
    for (i=0;i<*in_len;i++)
-      x[i] = in[i+st->in_stride];
+      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]);
+      out[i*st->out_stride] = WORD2INT(y[i]);
 }
 #endif
 
@@ -561,8 +823,33 @@ void speex_resampler_process_interleaved_float(SpeexResamplerState *st, const fl
    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(SpeexResamplerState *st, int ratio_num, int ratio_den, int in_rate, int 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)
@@ -586,6 +873,12 @@ void speex_resampler_set_rate(SpeexResamplerState *st, int ratio_num, int ratio_
       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)
@@ -599,16 +892,31 @@ void speex_resampler_set_quality(SpeexResamplerState *st, int quality)
       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;