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resample.c

/* Copyright (C) 2007 Jean-Marc Valin

   File: resample.c
   Arbitrary resampling code

   Redistribution and use in source and binary forms, with or without
   modification, are permitted provided that the following conditions are
   met:

   1. Redistributions of source code must retain the above copyright notice,
   this list of conditions and the following disclaimer.

   2. Redistributions in binary form must reproduce the above copyright
   notice, this list of conditions and the following disclaimer in the
   documentation and/or other materials provided with the distribution.

   3. The name of the author may not be used to endorse or promote products
   derived from this software without specific prior written permission.

   THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
   IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
   OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
   DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
   INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
   (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
   SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
   STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
   ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
   POSSIBILITY OF SUCH DAMAGE.
*/

/*
   The design goals of this code are:
      - Very fast algorithm
      - SIMD-friendly algorithm
      - Low memory requirement
      - Good *perceptual* quality (and not best SNR)

   Warning: This resampler is relatively new. Although I think I got rid of
   all the major bugs and I don't expect the API to change anymore, there
   may be something I've missed. So use with caution.

   This algorithm is based on this original resampling algorithm:
   Smith, Julius O. Digital Audio Resampling Home Page
   Center for Computer Research in Music and Acoustics (CCRMA),
   Stanford University, 2007.
   Web published at http://www-ccrma.stanford.edu/~jos/resample/.

   There is one main difference, though. This resampler uses cubic
   interpolation instead of linear interpolation in the above paper. This
   makes the table much smaller and makes it possible to compute that table
   on a per-stream basis. In turn, being able to tweak the table for each
   stream makes it possible to both reduce complexity on simple ratios
   (e.g. 2/3), and get rid of the rounding operations in the inner loop.
   The latter both reduces CPU time and makes the algorithm more SIMD-friendly.
*/

#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

#ifdef OUTSIDE_SPEEX
#include <stdlib.h>
static void *speex_alloc (int size) {return calloc(size,1);}
static void *speex_realloc (void *ptr, int size) {return realloc(ptr, size);}
static void speex_free (void *ptr) {free(ptr);}
#include "speex_resampler.h"
#include "arch.h"
#else /* OUTSIDE_SPEEX */

#include "speex/speex_resampler.h"
#include "arch.h"
#include "os_support.h"
#endif /* OUTSIDE_SPEEX */

#include <math.h>

#ifndef M_PI
#define M_PI 3.14159263
#endif

#ifdef FIXED_POINT
#define WORD2INT(x) ((x) < -32767 ? -32768 : ((x) > 32766 ? 32767 : (x)))
#else
#define WORD2INT(x) ((x) < -32767.5f ? -32768 : ((x) > 32766.5f ? 32767 : floor(.5+(x))))
#endif

/*#define float double*/
#define FILTER_SIZE 64
#define OVERSAMPLE 8

#define IMAX(a,b) ((a) > (b) ? (a) : (b))
#define IMIN(a,b) ((a) < (b) ? (a) : (b))

#ifndef NULL
#define NULL 0
#endif

typedef int (*resampler_basic_func)(SpeexResamplerState *, spx_uint32_t , const spx_word16_t *, spx_uint32_t *, spx_word16_t *, spx_uint32_t *);

struct SpeexResamplerState_ {
   spx_uint32_t in_rate;
   spx_uint32_t out_rate;
   spx_uint32_t num_rate;
   spx_uint32_t den_rate;

   int    quality;
   spx_uint32_t nb_channels;
   spx_uint32_t filt_len;
   spx_uint32_t mem_alloc_size;
   int          int_advance;
   int          frac_advance;
   float  cutoff;
   spx_uint32_t oversample;
   int          initialised;
   int          started;

   /* These are per-channel */
   spx_int32_t  *last_sample;
   spx_uint32_t *samp_frac_num;
   spx_uint32_t *magic_samples;

   spx_word16_t *mem;
   spx_word16_t *sinc_table;
   spx_uint32_t sinc_table_length;
   resampler_basic_func resampler_ptr;

   int    in_stride;
   int    out_stride;
} ;

static double kaiser12_table[68] = {
   0.99859849, 1.00000000, 0.99859849, 0.99440475, 0.98745105, 0.97779076,
   0.96549770, 0.95066529, 0.93340547, 0.91384741, 0.89213598, 0.86843014,
   0.84290116, 0.81573067, 0.78710866, 0.75723148, 0.72629970, 0.69451601,
   0.66208321, 0.62920216, 0.59606986, 0.56287762, 0.52980938, 0.49704014,
   0.46473455, 0.43304576, 0.40211431, 0.37206735, 0.34301800, 0.31506490,
   0.28829195, 0.26276832, 0.23854851, 0.21567274, 0.19416736, 0.17404546,
   0.15530766, 0.13794294, 0.12192957, 0.10723616, 0.09382272, 0.08164178,
   0.07063950, 0.06075685, 0.05193064, 0.04409466, 0.03718069, 0.03111947,
   0.02584161, 0.02127838, 0.01736250, 0.01402878, 0.01121463, 0.00886058,
   0.00691064, 0.00531256, 0.00401805, 0.00298291, 0.00216702, 0.00153438,
   0.00105297, 0.00069463, 0.00043489, 0.00025272, 0.00013031, 0.0000527734,
   0.00001000, 0.00000000};
/*
static double kaiser12_table[36] = {
   0.99440475, 1.00000000, 0.99440475, 0.97779076, 0.95066529, 0.91384741,
   0.86843014, 0.81573067, 0.75723148, 0.69451601, 0.62920216, 0.56287762,
   0.49704014, 0.43304576, 0.37206735, 0.31506490, 0.26276832, 0.21567274,
   0.17404546, 0.13794294, 0.10723616, 0.08164178, 0.06075685, 0.04409466,
   0.03111947, 0.02127838, 0.01402878, 0.00886058, 0.00531256, 0.00298291,
   0.00153438, 0.00069463, 0.00025272, 0.0000527734, 0.00000500, 0.00000000};
*/
static double kaiser10_table[36] = {
   0.99537781, 1.00000000, 0.99537781, 0.98162644, 0.95908712, 0.92831446,
   0.89005583, 0.84522401, 0.79486424, 0.74011713, 0.68217934, 0.62226347,
   0.56155915, 0.50119680, 0.44221549, 0.38553619, 0.33194107, 0.28205962,
   0.23636152, 0.19515633, 0.15859932, 0.12670280, 0.09935205, 0.07632451,
   0.05731132, 0.04193980, 0.02979584, 0.02044510, 0.01345224, 0.00839739,
   0.00488951, 0.00257636, 0.00115101, 0.00035515, 0.00000000, 0.00000000};

static double kaiser8_table[36] = {
   0.99635258, 1.00000000, 0.99635258, 0.98548012, 0.96759014, 0.94302200,
   0.91223751, 0.87580811, 0.83439927, 0.78875245, 0.73966538, 0.68797126,
   0.63451750, 0.58014482, 0.52566725, 0.47185369, 0.41941150, 0.36897272,
   0.32108304, 0.27619388, 0.23465776, 0.19672670, 0.16255380, 0.13219758,
   0.10562887, 0.08273982, 0.06335451, 0.04724088, 0.03412321, 0.02369490,
   0.01563093, 0.00959968, 0.00527363, 0.00233883, 0.00050000, 0.00000000};

static double kaiser6_table[36] = {
   0.99733006, 1.00000000, 0.99733006, 0.98935595, 0.97618418, 0.95799003,
   0.93501423, 0.90755855, 0.87598009, 0.84068475, 0.80211977, 0.76076565,
   0.71712752, 0.67172623, 0.62508937, 0.57774224, 0.53019925, 0.48295561,
   0.43647969, 0.39120616, 0.34752997, 0.30580127, 0.26632152, 0.22934058,
   0.19505503, 0.16360756, 0.13508755, 0.10953262, 0.08693120, 0.06722600,
   0.05031820, 0.03607231, 0.02432151, 0.01487334, 0.00752000, 0.00000000};

struct FuncDef {
   double *table;
   int oversample;
};

static struct FuncDef _KAISER12 = {kaiser12_table, 64};
#define KAISER12 (&_KAISER12)
/*static struct FuncDef _KAISER12 = {kaiser12_table, 32};
#define KAISER12 (&_KAISER12)*/
static struct FuncDef _KAISER10 = {kaiser10_table, 32};
#define KAISER10 (&_KAISER10)
static struct FuncDef _KAISER8 = {kaiser8_table, 32};
#define KAISER8 (&_KAISER8)
static struct FuncDef _KAISER6 = {kaiser6_table, 32};
#define KAISER6 (&_KAISER6)

struct QualityMapping {
   int base_length;
   int oversample;
   float downsample_bandwidth;
   float upsample_bandwidth;
   struct FuncDef *window_func;
};


/* This table maps conversion quality to internal parameters. There are two
   reasons that explain why the up-sampling bandwidth is larger than the
   down-sampling bandwidth:
   1) When up-sampling, we can assume that the spectrum is already attenuated
      close to the Nyquist rate (from an A/D or a previous resampling filter)
   2) Any aliasing that occurs very close to the Nyquist rate will be masked
      by the sinusoids/noise just below the Nyquist rate (guaranteed only for
      up-sampling).
*/
static const struct QualityMapping quality_map[11] = {
   {  8,  4, 0.830f, 0.860f, KAISER6 }, /* Q0 */
   { 16,  4, 0.850f, 0.880f, KAISER6 }, /* Q1 */
   { 32,  4, 0.882f, 0.910f, KAISER6 }, /* Q2 */  /* 82.3% cutoff ( ~60 dB stop) 6  */
   { 48,  8, 0.895f, 0.917f, KAISER8 }, /* Q3 */  /* 84.9% cutoff ( ~80 dB stop) 8  */
   { 64,  8, 0.921f, 0.940f, KAISER8 }, /* Q4 */  /* 88.7% cutoff ( ~80 dB stop) 8  */
   { 80, 16, 0.922f, 0.940f, KAISER10}, /* Q5 */  /* 89.1% cutoff (~100 dB stop) 10 */
   { 96, 16, 0.940f, 0.945f, KAISER10}, /* Q6 */  /* 91.5% cutoff (~100 dB stop) 10 */
   {128, 16, 0.950f, 0.950f, KAISER10}, /* Q7 */  /* 93.1% cutoff (~100 dB stop) 10 */
   {160, 16, 0.960f, 0.960f, KAISER10}, /* Q8 */  /* 94.5% cutoff (~100 dB stop) 10 */
   {192, 32, 0.968f, 0.968f, KAISER12}, /* Q9 */  /* 95.5% cutoff (~100 dB stop) 10 */
   {256, 32, 0.975f, 0.975f, KAISER12}, /* Q10 */ /* 96.6% cutoff (~100 dB stop) 10 */
};
/*8,24,40,56,80,104,128,160,200,256,320*/
static double compute_func(float x, struct FuncDef *func)
{
   float y, frac;
   double interp[4];
   int ind;
   y = x*func->oversample;
   ind = (int)floor(y);
   frac = (y-ind);
   /* CSE with handle the repeated powers */
   interp[3] =  -0.1666666667*frac + 0.1666666667*(frac*frac*frac);
   interp[2] = frac + 0.5*(frac*frac) - 0.5*(frac*frac*frac);
   /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac;*/
   interp[0] = -0.3333333333*frac + 0.5*(frac*frac) - 0.1666666667*(frac*frac*frac);
   /* Just to make sure we don't have rounding problems */
   interp[1] = 1.f-interp[3]-interp[2]-interp[0];

   /*sum = frac*accum[1] + (1-frac)*accum[2];*/
   return interp[0]*func->table[ind] + interp[1]*func->table[ind+1] + interp[2]*func->table[ind+2] + interp[3]*func->table[ind+3];
}

#if 0
#include <stdio.h>
int main(int argc, char **argv)
{
   int i;
   for (i=0;i<256;i++)
   {
      printf ("%f\n", compute_func(i/256., KAISER12));
   }
   return 0;
}
#endif

#ifdef FIXED_POINT
/* The slow way of computing a sinc for the table. Should improve that some day */
static spx_word16_t sinc(float cutoff, float x, int N, struct FuncDef *window_func)
{
   /*fprintf (stderr, "%f ", x);*/
   float xx = x * cutoff;
   if (fabs(x)<1e-6f)
      return WORD2INT(32768.*cutoff);
   else if (fabs(x) > .5f*N)
      return 0;
   /*FIXME: Can it really be any slower than this? */
   return WORD2INT(32768.*cutoff*sin(M_PI*xx)/(M_PI*xx) * compute_func(fabs(2.*x/N), window_func));
}
#else
/* The slow way of computing a sinc for the table. Should improve that some day */
static spx_word16_t sinc(float cutoff, float x, int N, struct FuncDef *window_func)
{
   /*fprintf (stderr, "%f ", x);*/
   float xx = x * cutoff;
   if (fabs(x)<1e-6)
      return cutoff;
   else if (fabs(x) > .5*N)
      return 0;
   /*FIXME: Can it really be any slower than this? */
   return cutoff*sin(M_PI*xx)/(M_PI*xx) * compute_func(fabs(2.*x/N), window_func);
}
#endif

#ifdef FIXED_POINT
static void cubic_coef(spx_word16_t x, spx_word16_t interp[4])
{
   /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
   but I know it's MMSE-optimal on a sinc */
   spx_word16_t x2, x3;
   x2 = MULT16_16_P15(x, x);
   x3 = MULT16_16_P15(x, x2);
   interp[0] = PSHR32(MULT16_16(QCONST16(-0.16667f, 15),x) + MULT16_16(QCONST16(0.16667f, 15),x3),15);
   interp[1] = EXTRACT16(EXTEND32(x) + SHR32(SUB32(EXTEND32(x2),EXTEND32(x3)),1));
   interp[3] = PSHR32(MULT16_16(QCONST16(-0.33333f, 15),x) + MULT16_16(QCONST16(.5f,15),x2) - MULT16_16(QCONST16(0.16667f, 15),x3),15);
   /* Just to make sure we don't have rounding problems */
   interp[2] = Q15_ONE-interp[0]-interp[1]-interp[3];
   if (interp[2]<32767)
      interp[2]+=1;
}
#else
static void cubic_coef(spx_word16_t frac, spx_word16_t interp[4])
{
   /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
   but I know it's MMSE-optimal on a sinc */
   interp[0] =  -0.16667f*frac + 0.16667f*frac*frac*frac;
   interp[1] = frac + 0.5f*frac*frac - 0.5f*frac*frac*frac;
   /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac;*/
   interp[3] = -0.33333f*frac + 0.5f*frac*frac - 0.16667f*frac*frac*frac;
   /* Just to make sure we don't have rounding problems */
   interp[2] = 1.-interp[0]-interp[1]-interp[3];
}
#endif

static int resampler_basic_direct_single(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
{
   int N = st->filt_len;
   int out_sample = 0;
   spx_word16_t *mem;
   int last_sample = st->last_sample[channel_index];
   spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
   mem = st->mem + channel_index * st->mem_alloc_size;
   while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
   {
      int j;
      spx_word32_t sum=0;

      /* We already have all the filter coefficients pre-computed in the table */
      const spx_word16_t *ptr;
      /* Do the memory part */
      for (j=0;last_sample-N+1+j < 0;j++)
      {
         sum += MULT16_16(mem[last_sample+j],st->sinc_table[samp_frac_num*st->filt_len+j]);
      }

      /* Do the new part */
      ptr = in+st->in_stride*(last_sample-N+1+j);
      for (;j<N;j++)
      {
         sum += MULT16_16(*ptr,st->sinc_table[samp_frac_num*st->filt_len+j]);
         ptr += st->in_stride;
      }

      *out = PSHR32(sum,15);
      out += st->out_stride;
      out_sample++;
      last_sample += st->int_advance;
      samp_frac_num += st->frac_advance;
      if (samp_frac_num >= st->den_rate)
      {
         samp_frac_num -= st->den_rate;
         last_sample++;
      }
   }
   st->last_sample[channel_index] = last_sample;
   st->samp_frac_num[channel_index] = samp_frac_num;
   return out_sample;
}

#ifdef FIXED_POINT
#else
/* This is the same as the previous function, except with a double-precision accumulator */
static int resampler_basic_direct_double(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
{
   int N = st->filt_len;
   int out_sample = 0;
   spx_word16_t *mem;
   int last_sample = st->last_sample[channel_index];
   spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
   mem = st->mem + channel_index * st->mem_alloc_size;
   while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
   {
      int j;
      double sum=0;

      /* We already have all the filter coefficients pre-computed in the table */
      const spx_word16_t *ptr;
      /* Do the memory part */
      for (j=0;last_sample-N+1+j < 0;j++)
      {
         sum += MULT16_16(mem[last_sample+j],(double)st->sinc_table[samp_frac_num*st->filt_len+j]);
      }

      /* Do the new part */
      ptr = in+st->in_stride*(last_sample-N+1+j);
      for (;j<N;j++)
      {
         sum += MULT16_16(*ptr,(double)st->sinc_table[samp_frac_num*st->filt_len+j]);
         ptr += st->in_stride;
      }

      *out = sum;
      out += st->out_stride;
      out_sample++;
      last_sample += st->int_advance;
      samp_frac_num += st->frac_advance;
      if (samp_frac_num >= st->den_rate)
      {
         samp_frac_num -= st->den_rate;
         last_sample++;
      }
   }
   st->last_sample[channel_index] = last_sample;
   st->samp_frac_num[channel_index] = samp_frac_num;
   return out_sample;
}
#endif

static int resampler_basic_interpolate_single(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
{
   int N = st->filt_len;
   int out_sample = 0;
   spx_word16_t *mem;
   int last_sample = st->last_sample[channel_index];
   spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
   mem = st->mem + channel_index * st->mem_alloc_size;
   while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
   {
      int j;
      spx_word32_t sum=0;

      /* We need to interpolate the sinc filter */
      spx_word32_t accum[4] = {0.f,0.f, 0.f, 0.f};
      spx_word16_t interp[4];
      const spx_word16_t *ptr;
      int offset;
      spx_word16_t frac;
      offset = samp_frac_num*st->oversample/st->den_rate;
#ifdef FIXED_POINT
      frac = PDIV32(SHL32((samp_frac_num*st->oversample) % st->den_rate,15),st->den_rate);
#else
      frac = ((float)((samp_frac_num*st->oversample) % st->den_rate))/st->den_rate;
#endif
         /* This code is written like this to make it easy to optimise with SIMD.
      For most DSPs, it would be best to split the loops in two because most DSPs
      have only two accumulators */
      for (j=0;last_sample-N+1+j < 0;j++)
      {
         spx_word16_t curr_mem = mem[last_sample+j];
         accum[0] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset-2]);
         accum[1] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset-1]);
         accum[2] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset]);
         accum[3] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset+1]);
      }
      ptr = in+st->in_stride*(last_sample-N+1+j);
      /* Do the new part */
      for (;j<N;j++)
      {
         spx_word16_t curr_in = *ptr;
         ptr += st->in_stride;
         accum[0] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-2]);
         accum[1] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-1]);
         accum[2] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset]);
         accum[3] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset+1]);
      }
      cubic_coef(frac, interp);
      sum = MULT16_32_Q15(interp[0],accum[0]) + MULT16_32_Q15(interp[1],accum[1]) + MULT16_32_Q15(interp[2],accum[2]) + MULT16_32_Q15(interp[3],accum[3]);

      *out = PSHR32(sum,15);
      out += st->out_stride;
      out_sample++;
      last_sample += st->int_advance;
      samp_frac_num += st->frac_advance;
      if (samp_frac_num >= st->den_rate)
      {
         samp_frac_num -= st->den_rate;
         last_sample++;
      }
   }
   st->last_sample[channel_index] = last_sample;
   st->samp_frac_num[channel_index] = samp_frac_num;
   return out_sample;
}

#ifdef FIXED_POINT
#else
/* This is the same as the previous function, except with a double-precision accumulator */
static int resampler_basic_interpolate_double(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
{
   int N = st->filt_len;
   int out_sample = 0;
   spx_word16_t *mem;
   int last_sample = st->last_sample[channel_index];
   spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
   mem = st->mem + channel_index * st->mem_alloc_size;
   while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
   {
      int j;
      spx_word32_t sum=0;

      /* We need to interpolate the sinc filter */
      double accum[4] = {0.f,0.f, 0.f, 0.f};
      float interp[4];
      const spx_word16_t *ptr;
      float alpha = ((float)samp_frac_num)/st->den_rate;
      int offset = samp_frac_num*st->oversample/st->den_rate;
      float frac = alpha*st->oversample - offset;
         /* This code is written like this to make it easy to optimise with SIMD.
      For most DSPs, it would be best to split the loops in two because most DSPs
      have only two accumulators */
      for (j=0;last_sample-N+1+j < 0;j++)
      {
         double curr_mem = mem[last_sample+j];
         accum[0] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset-2]);
         accum[1] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset-1]);
         accum[2] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset]);
         accum[3] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset+1]);
      }
      ptr = in+st->in_stride*(last_sample-N+1+j);
      /* Do the new part */
      for (;j<N;j++)
      {
         double curr_in = *ptr;
         ptr += st->in_stride;
         accum[0] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-2]);
         accum[1] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-1]);
         accum[2] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset]);
         accum[3] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset+1]);
      }
      cubic_coef(frac, interp);
      sum = interp[0]*accum[0] + interp[1]*accum[1] + interp[2]*accum[2] + interp[3]*accum[3];

      *out = PSHR32(sum,15);
      out += st->out_stride;
      out_sample++;
      last_sample += st->int_advance;
      samp_frac_num += st->frac_advance;
      if (samp_frac_num >= st->den_rate)
      {
         samp_frac_num -= st->den_rate;
         last_sample++;
      }
   }
   st->last_sample[channel_index] = last_sample;
   st->samp_frac_num[channel_index] = samp_frac_num;
   return out_sample;
}
#endif

static void update_filter(SpeexResamplerState *st)
{
   spx_uint32_t old_length;

   old_length = st->filt_len;
   st->oversample = quality_map[st->quality].oversample;
   st->filt_len = quality_map[st->quality].base_length;

   if (st->num_rate > st->den_rate)
   {
      /* down-sampling */
      st->cutoff = quality_map[st->quality].downsample_bandwidth * st->den_rate / st->num_rate;
      /* FIXME: divide the numerator and denominator by a certain amount if they're too large */
      st->filt_len = st->filt_len*st->num_rate / st->den_rate;
      /* Round down to make sure we have a multiple of 4 */
      st->filt_len &= (~0x3);
      if (2*st->den_rate < st->num_rate)
         st->oversample >>= 1;
      if (4*st->den_rate < st->num_rate)
         st->oversample >>= 1;
      if (8*st->den_rate < st->num_rate)
         st->oversample >>= 1;
      if (16*st->den_rate < st->num_rate)
         st->oversample >>= 1;
      if (st->oversample < 1)
         st->oversample = 1;
   } else {
      /* up-sampling */
      st->cutoff = quality_map[st->quality].upsample_bandwidth;
   }

   /* Choose the resampling type that requires the least amount of memory */
   if (st->den_rate <= st->oversample)
   {
      spx_uint32_t i;
      if (!st->sinc_table)
         st->sinc_table = (spx_word16_t *)speex_alloc(st->filt_len*st->den_rate*sizeof(spx_word16_t));
      else if (st->sinc_table_length < st->filt_len*st->den_rate)
      {
         st->sinc_table = (spx_word16_t *)speex_realloc(st->sinc_table,st->filt_len*st->den_rate*sizeof(spx_word16_t));
         st->sinc_table_length = st->filt_len*st->den_rate;
      }
      for (i=0;i<st->den_rate;i++)
      {
         spx_int32_t j;
         for (j=0;j<st->filt_len;j++)
         {
            st->sinc_table[i*st->filt_len+j] = sinc(st->cutoff,((j-(spx_int32_t)st->filt_len/2+1)-((float)i)/st->den_rate), st->filt_len, quality_map[st->quality].window_func);
         }
      }
#ifdef FIXED_POINT
      st->resampler_ptr = resampler_basic_direct_single;
#else
      if (st->quality>8)
         st->resampler_ptr = resampler_basic_direct_double;
      else
         st->resampler_ptr = resampler_basic_direct_single;
#endif
      /*fprintf (stderr, "resampler uses direct sinc table and normalised cutoff %f\n", cutoff);*/
   } else {
      spx_int32_t i;
      if (!st->sinc_table)
         st->sinc_table = (spx_word16_t *)speex_alloc((st->filt_len*st->oversample+8)*sizeof(spx_word16_t));
      else if (st->sinc_table_length < st->filt_len*st->oversample+8)
      {
         st->sinc_table = (spx_word16_t *)speex_realloc(st->sinc_table,(st->filt_len*st->oversample+8)*sizeof(spx_word16_t));
         st->sinc_table_length = st->filt_len*st->oversample+8;
      }
      for (i=-4;i<(spx_int32_t)(st->oversample*st->filt_len+4);i++)
         st->sinc_table[i+4] = sinc(st->cutoff,(i/(float)st->oversample - st->filt_len/2), st->filt_len, quality_map[st->quality].window_func);
#ifdef FIXED_POINT
      st->resampler_ptr = resampler_basic_interpolate_single;
#else
      if (st->quality>8)
         st->resampler_ptr = resampler_basic_interpolate_double;
      else
         st->resampler_ptr = resampler_basic_interpolate_single;
#endif
      /*fprintf (stderr, "resampler uses interpolated sinc table and normalised cutoff %f\n", cutoff);*/
   }
   st->int_advance = st->num_rate/st->den_rate;
   st->frac_advance = st->num_rate%st->den_rate;


   /* Here's the place where we update the filter memory to take into account
      the change in filter length. It's probably the messiest part of the code
      due to handling of lots of corner cases. */
   if (!st->mem)
   {
      spx_uint32_t i;
      st->mem = (spx_word16_t*)speex_alloc(st->nb_channels*(st->filt_len-1) * sizeof(spx_word16_t));
      for (i=0;i<st->nb_channels*(st->filt_len-1);i++)
         st->mem[i] = 0;
      st->mem_alloc_size = st->filt_len-1;
      /*speex_warning("init filter");*/
   } else if (!st->started)
   {
      spx_uint32_t i;
      st->mem = (spx_word16_t*)speex_realloc(st->mem, st->nb_channels*(st->filt_len-1) * sizeof(spx_word16_t));
      for (i=0;i<st->nb_channels*(st->filt_len-1);i++)
         st->mem[i] = 0;
      st->mem_alloc_size = st->filt_len-1;
      /*speex_warning("reinit filter");*/
   } else if (st->filt_len > old_length)
   {
      spx_int32_t i;
      /* Increase the filter length */
      /*speex_warning("increase filter size");*/
      int old_alloc_size = st->mem_alloc_size;
      if (st->filt_len-1 > st->mem_alloc_size)
      {
         st->mem = (spx_word16_t*)speex_realloc(st->mem, st->nb_channels*(st->filt_len-1) * sizeof(spx_word16_t));
         st->mem_alloc_size = st->filt_len-1;
      }
      for (i=st->nb_channels-1;i>=0;i--)
      {
         spx_int32_t j;
         spx_uint32_t olen = old_length;
         /*if (st->magic_samples[i])*/
         {
            /* Try and remove the magic samples as if nothing had happened */

            /* FIXME: This is wrong but for now we need it to avoid going over the array bounds */
            olen = old_length + 2*st->magic_samples[i];
            for (j=old_length-2+st->magic_samples[i];j>=0;j--)
               st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]] = st->mem[i*old_alloc_size+j];
            for (j=0;j<st->magic_samples[i];j++)
               st->mem[i*st->mem_alloc_size+j] = 0;
            st->magic_samples[i] = 0;
         }
         if (st->filt_len > olen)
         {
            /* If the new filter length is still bigger than the "augmented" length */
            /* Copy data going backward */
            for (j=0;j<olen-1;j++)
               st->mem[i*st->mem_alloc_size+(st->filt_len-2-j)] = st->mem[i*st->mem_alloc_size+(olen-2-j)];
            /* Then put zeros for lack of anything better */
            for (;j<st->filt_len-1;j++)
               st->mem[i*st->mem_alloc_size+(st->filt_len-2-j)] = 0;
            /* Adjust last_sample */
            st->last_sample[i] += (st->filt_len - olen)/2;
         } else {
            /* Put back some of the magic! */
            st->magic_samples[i] = (olen - st->filt_len)/2;
            for (j=0;j<st->filt_len-1+st->magic_samples[i];j++)
               st->mem[i*st->mem_alloc_size+j] = st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]];
         }
      }
   } else if (st->filt_len < old_length)
   {
      spx_uint32_t i;
      /* Reduce filter length, this a bit tricky. We need to store some of the memory as "magic"
         samples so they can be used directly as input the next time(s) */
      for (i=0;i<st->nb_channels;i++)
      {
         spx_uint32_t j;
         spx_uint32_t old_magic = st->magic_samples[i];
         st->magic_samples[i] = (old_length - st->filt_len)/2;
         /* We must copy some of the memory that's no longer used */
         /* Copy data going backward */
         for (j=0;j<st->filt_len-1+st->magic_samples[i]+old_magic;j++)
            st->mem[i*st->mem_alloc_size+j] = st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]];
         st->magic_samples[i] += old_magic;
      }
   }

}

SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, spx_uint32_t in_rate, spx_uint32_t out_rate, int quality, int *err)
{
   return speex_resampler_init_frac(nb_channels, in_rate, out_rate, in_rate, out_rate, quality, err);
}

SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels, spx_uint32_t ratio_num, spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate, int quality, int *err)
{
   spx_uint32_t i;
   SpeexResamplerState *st;
   if (quality > 10 || quality < 0)
   {
      if (err)
         *err = RESAMPLER_ERR_INVALID_ARG;
      return NULL;
   }
   st = (SpeexResamplerState *)speex_alloc(sizeof(SpeexResamplerState));
   st->initialised = 0;
   st->started = 0;
   st->in_rate = 0;
   st->out_rate = 0;
   st->num_rate = 0;
   st->den_rate = 0;
   st->quality = -1;
   st->sinc_table_length = 0;
   st->mem_alloc_size = 0;
   st->filt_len = 0;
   st->mem = 0;
   st->resampler_ptr = 0;

   st->cutoff = 1.f;
   st->nb_channels = nb_channels;
   st->in_stride = 1;
   st->out_stride = 1;

   /* Per channel data */
   st->last_sample = (spx_int32_t*)speex_alloc(nb_channels*sizeof(int));
   st->magic_samples = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(int));
   st->samp_frac_num = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(int));
   for (i=0;i<nb_channels;i++)
   {
      st->last_sample[i] = 0;
      st->magic_samples[i] = 0;
      st->samp_frac_num[i] = 0;
   }

   speex_resampler_set_quality(st, quality);
   speex_resampler_set_rate_frac(st, ratio_num, ratio_den, in_rate, out_rate);


   update_filter(st);

   st->initialised = 1;
   if (err)
      *err = RESAMPLER_ERR_SUCCESS;

   return st;
}

void speex_resampler_destroy(SpeexResamplerState *st)
{
   speex_free(st->mem);
   speex_free(st->sinc_table);
   speex_free(st->last_sample);
   speex_free(st->magic_samples);
   speex_free(st->samp_frac_num);
   speex_free(st);
}



static int speex_resampler_process_native(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
{
   int j=0;
   int N = st->filt_len;
   int out_sample = 0;
   spx_word16_t *mem;
   spx_uint32_t tmp_out_len = 0;
   mem = st->mem + channel_index * st->mem_alloc_size;
   st->started = 1;

   /* Handle the case where we have samples left from a reduction in filter length */
   if (st->magic_samples[channel_index])
   {
      int istride_save;
      spx_uint32_t tmp_in_len;
      spx_uint32_t tmp_magic;

      istride_save = st->in_stride;
      tmp_in_len = st->magic_samples[channel_index];
      tmp_out_len = *out_len;
      /* magic_samples needs to be set to zero to avoid infinite recursion */
      tmp_magic = st->magic_samples[channel_index];
      st->magic_samples[channel_index] = 0;
      st->in_stride = 1;
      speex_resampler_process_native(st, channel_index, mem+N-1, &tmp_in_len, out, &tmp_out_len);
      st->in_stride = istride_save;
      /*speex_warning_int("extra samples:", tmp_out_len);*/
      /* If we couldn't process all "magic" input samples, save the rest for next time */
      if (tmp_in_len < tmp_magic)
      {
         spx_uint32_t i;
         st->magic_samples[channel_index] = tmp_magic-tmp_in_len;
         for (i=0;i<st->magic_samples[channel_index];i++)
            mem[N-1+i]=mem[N-1+i+tmp_in_len];
      }
      out += tmp_out_len*st->out_stride;
      *out_len -= tmp_out_len;
   }

   /* Call the right resampler through the function ptr */
   out_sample = st->resampler_ptr(st, channel_index, in, in_len, out, out_len);

   if (st->last_sample[channel_index] < (spx_int32_t)*in_len)
      *in_len = st->last_sample[channel_index];
   *out_len = out_sample+tmp_out_len;
   st->last_sample[channel_index] -= *in_len;

   for (j=0;j<N-1-(spx_int32_t)*in_len;j++)
      mem[j] = mem[j+*in_len];
   for (;j<N-1;j++)
      mem[j] = in[st->in_stride*(j+*in_len-N+1)];

   return RESAMPLER_ERR_SUCCESS;
}

#define FIXED_STACK_ALLOC 1024

#ifdef FIXED_POINT
int speex_resampler_process_float(SpeexResamplerState *st, spx_uint32_t channel_index, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
{
   spx_uint32_t i;
   int istride_save, ostride_save;
#ifdef VAR_ARRAYS
   spx_word16_t x[*in_len];
   spx_word16_t y[*out_len];
   /*VARDECL(spx_word16_t *x);
   VARDECL(spx_word16_t *y);
   ALLOC(x, *in_len, spx_word16_t);
   ALLOC(y, *out_len, spx_word16_t);*/
   istride_save = st->in_stride;
   ostride_save = st->out_stride;
   for (i=0;i<*in_len;i++)
      x[i] = WORD2INT(in[i*st->in_stride]);
   st->in_stride = st->out_stride = 1;
   speex_resampler_process_native(st, channel_index, x, in_len, y, out_len);
   st->in_stride = istride_save;
   st->out_stride = ostride_save;
   for (i=0;i<*out_len;i++)
      out[i*st->out_stride] = y[i];
#else
   spx_word16_t x[FIXED_STACK_ALLOC];
   spx_word16_t y[FIXED_STACK_ALLOC];
   spx_uint32_t ilen=*in_len, olen=*out_len;
   istride_save = st->in_stride;
   ostride_save = st->out_stride;
   while (ilen && olen)
   {
      spx_uint32_t ichunk, ochunk;
      ichunk = ilen;
      ochunk = olen;
      if (ichunk>FIXED_STACK_ALLOC)
         ichunk=FIXED_STACK_ALLOC;
      if (ochunk>FIXED_STACK_ALLOC)
         ochunk=FIXED_STACK_ALLOC;
      for (i=0;i<ichunk;i++)
         x[i] = WORD2INT(in[i*st->in_stride]);
      st->in_stride = st->out_stride = 1;
      speex_resampler_process_native(st, channel_index, x, &ichunk, y, &ochunk);
      st->in_stride = istride_save;
      st->out_stride = ostride_save;
      for (i=0;i<ochunk;i++)
         out[i*st->out_stride] = y[i];
      out += ochunk;
      in += ichunk;
      ilen -= ichunk;
      olen -= ochunk;
   }
   *in_len -= ilen;
   *out_len -= olen;
#endif
   return RESAMPLER_ERR_SUCCESS;
}
int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
{
   return speex_resampler_process_native(st, channel_index, in, in_len, out, out_len);
}
#else
int speex_resampler_process_float(SpeexResamplerState *st, spx_uint32_t channel_index, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
{
   return speex_resampler_process_native(st, channel_index, in, in_len, out, out_len);
}
int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
{
   spx_uint32_t i;
   int istride_save, ostride_save;
#ifdef VAR_ARRAYS
   spx_word16_t x[*in_len];
   spx_word16_t y[*out_len];
   /*VARDECL(spx_word16_t *x);
   VARDECL(spx_word16_t *y);
   ALLOC(x, *in_len, spx_word16_t);
   ALLOC(y, *out_len, spx_word16_t);*/
   istride_save = st->in_stride;
   ostride_save = st->out_stride;
   for (i=0;i<*in_len;i++)
      x[i] = in[i*st->in_stride];
   st->in_stride = st->out_stride = 1;
   speex_resampler_process_native(st, channel_index, x, in_len, y, out_len);
   st->in_stride = istride_save;
   st->out_stride = ostride_save;
   for (i=0;i<*out_len;i++)
      out[i*st->out_stride] = WORD2INT(y[i]);
#else
   spx_word16_t x[FIXED_STACK_ALLOC];
   spx_word16_t y[FIXED_STACK_ALLOC];
   spx_uint32_t ilen=*in_len, olen=*out_len;
   istride_save = st->in_stride;
   ostride_save = st->out_stride;
   while (ilen && olen)
   {
      spx_uint32_t ichunk, ochunk;
      ichunk = ilen;
      ochunk = olen;
      if (ichunk>FIXED_STACK_ALLOC)
         ichunk=FIXED_STACK_ALLOC;
      if (ochunk>FIXED_STACK_ALLOC)
         ochunk=FIXED_STACK_ALLOC;
      for (i=0;i<ichunk;i++)
         x[i] = in[i*st->in_stride];
      st->in_stride = st->out_stride = 1;
      speex_resampler_process_native(st, channel_index, x, &ichunk, y, &ochunk);
      st->in_stride = istride_save;
      st->out_stride = ostride_save;
      for (i=0;i<ochunk;i++)
         out[i*st->out_stride] = WORD2INT(y[i]);
      out += ochunk;
      in += ichunk;
      ilen -= ichunk;
      olen -= ochunk;
   }
   *in_len -= ilen;
   *out_len -= olen;
#endif
   return RESAMPLER_ERR_SUCCESS;
}
#endif

int speex_resampler_process_interleaved_float(SpeexResamplerState *st, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
{
   spx_uint32_t i;
   int istride_save, ostride_save;
   spx_uint32_t bak_len = *out_len;
   istride_save = st->in_stride;
   ostride_save = st->out_stride;
   st->in_stride = st->out_stride = st->nb_channels;
   for (i=0;i<st->nb_channels;i++)
   {
      *out_len = bak_len;
      speex_resampler_process_float(st, i, in+i, in_len, out+i, out_len);
   }
   st->in_stride = istride_save;
   st->out_stride = ostride_save;
   return RESAMPLER_ERR_SUCCESS;
}


int speex_resampler_process_interleaved_int(SpeexResamplerState *st, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
{
   spx_uint32_t i;
   int istride_save, ostride_save;
   spx_uint32_t bak_len = *out_len;
   istride_save = st->in_stride;
   ostride_save = st->out_stride;
   st->in_stride = st->out_stride = st->nb_channels;
   for (i=0;i<st->nb_channels;i++)
   {
      *out_len = bak_len;
      speex_resampler_process_int(st, i, in+i, in_len, out+i, out_len);
   }
   st->in_stride = istride_save;
   st->out_stride = ostride_save;
   return RESAMPLER_ERR_SUCCESS;
}

int speex_resampler_set_rate(SpeexResamplerState *st, spx_uint32_t in_rate, spx_uint32_t out_rate)
{
   return speex_resampler_set_rate_frac(st, in_rate, out_rate, in_rate, out_rate);
}

void speex_resampler_get_rate(SpeexResamplerState *st, spx_uint32_t *in_rate, spx_uint32_t *out_rate)
{
   *in_rate = st->in_rate;
   *out_rate = st->out_rate;
}

int speex_resampler_set_rate_frac(SpeexResamplerState *st, spx_uint32_t ratio_num, spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate)
{
   spx_uint32_t fact;
   spx_uint32_t old_den;
   spx_uint32_t i;
   if (st->in_rate == in_rate && st->out_rate == out_rate && st->num_rate == ratio_num && st->den_rate == ratio_den)
      return RESAMPLER_ERR_SUCCESS;

   old_den = st->den_rate;
   st->in_rate = in_rate;
   st->out_rate = out_rate;
   st->num_rate = ratio_num;
   st->den_rate = ratio_den;
   /* FIXME: This is terribly inefficient, but who cares (at least for now)? */
   for (fact=2;fact<=IMIN(st->num_rate, st->den_rate);fact++)
   {
      while ((st->num_rate % fact == 0) && (st->den_rate % fact == 0))
      {
         st->num_rate /= fact;
         st->den_rate /= fact;
      }
   }

   if (old_den > 0)
   {
      for (i=0;i<st->nb_channels;i++)
      {
         st->samp_frac_num[i]=st->samp_frac_num[i]*st->den_rate/old_den;
         /* Safety net */
         if (st->samp_frac_num[i] >= st->den_rate)
            st->samp_frac_num[i] = st->den_rate-1;
      }
   }

   if (st->initialised)
      update_filter(st);
   return RESAMPLER_ERR_SUCCESS;
}

void speex_resampler_get_ratio(SpeexResamplerState *st, spx_uint32_t *ratio_num, spx_uint32_t *ratio_den)
{
   *ratio_num = st->num_rate;
   *ratio_den = st->den_rate;
}

int speex_resampler_set_quality(SpeexResamplerState *st, int quality)
{
   if (quality > 10 || quality < 0)
      return RESAMPLER_ERR_INVALID_ARG;
   if (st->quality == quality)
      return RESAMPLER_ERR_SUCCESS;
   st->quality = quality;
   if (st->initialised)
      update_filter(st);
   return RESAMPLER_ERR_SUCCESS;
}

void speex_resampler_get_quality(SpeexResamplerState *st, int *quality)
{
   *quality = st->quality;
}

void speex_resampler_set_input_stride(SpeexResamplerState *st, spx_uint32_t stride)
{
   st->in_stride = stride;
}

void speex_resampler_get_input_stride(SpeexResamplerState *st, spx_uint32_t *stride)
{
   *stride = st->in_stride;
}

void speex_resampler_set_output_stride(SpeexResamplerState *st, spx_uint32_t stride)
{
   st->out_stride = stride;
}

void speex_resampler_get_output_stride(SpeexResamplerState *st, spx_uint32_t *stride)
{
   *stride = st->out_stride;
}

int speex_resampler_skip_zeros(SpeexResamplerState *st)
{
   spx_uint32_t i;
   for (i=0;i<st->nb_channels;i++)
      st->last_sample[i] = st->filt_len/2;
   return RESAMPLER_ERR_SUCCESS;
}

int speex_resampler_reset_mem(SpeexResamplerState *st)
{
   spx_uint32_t i;
   for (i=0;i<st->nb_channels*(st->filt_len-1);i++)
      st->mem[i] = 0;
   return RESAMPLER_ERR_SUCCESS;
}

const char *speex_resampler_strerror(int err)
{
   switch (err)
   {
      case RESAMPLER_ERR_SUCCESS:
         return "Success.";
      case RESAMPLER_ERR_ALLOC_FAILED:
         return "Memory allocation failed.";
      case RESAMPLER_ERR_BAD_STATE:
         return "Bad resampler state.";
      case RESAMPLER_ERR_INVALID_ARG:
         return "Invalid argument.";
      case RESAMPLER_ERR_PTR_OVERLAP:
         return "Input and output buffers overlap.";
      default:
         return "Unknown error. Bad error code or strange version mismatch.";
   }
}

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