text: /* Copyright (C) 2007-2008 Jean-Marc Valin
   Copyright (C) 2008      Thorvald Natvig
  
   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://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.
   */
  
  #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 "arch.h"
  #include "speex_resampler.h"
  
  #include "stack_alloc.h"
  #include <limits.h>
  #include <math.h>
  
  #define EXPORT
  
  #ifndef M_PI
  #define M_PI 3.14159265358979323846
  #endif
  
  #define IMAX(a, b) ((a) > (b) ? (a) : (b))
  #define IMIN(a, b) ((a) < (b) ? (a) : (b))
  
  #ifndef NULL
  #define NULL 0
  #endif
  
  #ifndef UINT32_MAX
  #define UINT32_MAX 4294967296U
  #endif
  
  /* Numer of elements to allocate on the stack */
  #ifdef VAR_ARRAYS
  #define FIXED_STACK_ALLOC 8192
  #else
  #define FIXED_STACK_ALLOC 1024
  #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;
    spx_uint32_t buffer_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 const 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};
  
  struct FuncDef {
    const double *table;
    int oversample;
  };
  
  static const struct FuncDef _KAISER12 = {kaiser12_table, 64};
  #define KAISER12 (&_KAISER12)
  
  struct QualityMapping {
    int base_length;
    int oversample;
    float downsample_bandwidth;
    float upsample_bandwidth;
    const 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[1] = {
      {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, const 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];
  }
  
  /* 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,
                           const 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);
  }
  
  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];
  }
  
  /* 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) {
    const int N = st->filt_len;
    int out_sample = 0;
    int last_sample = st->last_sample[channel_index];
    spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
    const spx_word16_t *sinc_table = st->sinc_table;
    const int out_stride = st->out_stride;
    const int int_advance = st->int_advance;
    const int frac_advance = st->frac_advance;
    const spx_uint32_t den_rate = st->den_rate;
    double sum;
  
    while (!(last_sample >= (spx_int32_t)*in_len ||
             out_sample >= (spx_int32_t)*out_len)) {
      const spx_word16_t *sinct = &sinc_table[samp_frac_num * N];
      const spx_word16_t *iptr = &in[last_sample];
  
  #ifndef OVERRIDE_INNER_PRODUCT_DOUBLE
      int j;
      double accum[4] = {0, 0, 0, 0};
  
      for (j = 0; j < N; j += 4) {
        accum[0] += sinct[j] * iptr[j];
        accum[1] += sinct[j + 1] * iptr[j + 1];
        accum[2] += sinct[j + 2] * iptr[j + 2];
        accum[3] += sinct[j + 3] * iptr[j + 3];
      }
      sum = accum[0] + accum[1] + accum[2] + accum[3];
  #else
      sum = inner_product_double(sinct, iptr, N);
  #endif
  
      out[out_stride * out_sample++] = PSHR32(sum, 15);
      last_sample += int_advance;
      samp_frac_num += frac_advance;
      if (samp_frac_num >= den_rate) {
        samp_frac_num -= den_rate;
        last_sample++;
      }
    }
  
    st->last_sample[channel_index] = last_sample;
    st->samp_frac_num[channel_index] = samp_frac_num;
    return out_sample;
  }
  
  /* 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) {
    const int N = st->filt_len;
    int out_sample = 0;
    int last_sample = st->last_sample[channel_index];
    spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
    const int out_stride = st->out_stride;
    const int int_advance = st->int_advance;
    const int frac_advance = st->frac_advance;
    const spx_uint32_t den_rate = st->den_rate;
    spx_word32_t sum;
  
    while (!(last_sample >= (spx_int32_t)*in_len ||
             out_sample >= (spx_int32_t)*out_len)) {
      const spx_word16_t *iptr = &in[last_sample];
  
      const int offset = samp_frac_num * st->oversample / st->den_rate;
      const spx_word16_t frac =
          ((float)((samp_frac_num * st->oversample) % st->den_rate)) /
          st->den_rate;
      spx_word16_t interp[4];
  
  #ifndef OVERRIDE_INTERPOLATE_PRODUCT_DOUBLE
      int j;
      double accum[4] = {0, 0, 0, 0};
  
      for (j = 0; j < N; j++) {
        const double curr_in = iptr[j];
        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]);
  #else
      cubic_coef(frac, interp);
      sum = interpolate_product_double(
          iptr, st->sinc_table + st->oversample + 4 - offset - 2, N,
          st->oversample, interp);
  #endif
  
      out[out_stride * out_sample++] = PSHR32(sum, 15);
      last_sample += int_advance;
      samp_frac_num += frac_advance;
      if (samp_frac_num >= den_rate) {
        samp_frac_num -= 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 _muldiv(spx_uint32_t *result, spx_uint32_t value, spx_uint32_t mul,
                     spx_uint32_t div) {
    spx_uint32_t major = value / div;
    spx_uint32_t remainder = value % div;
    speex_assert(result);
    /* TODO: Could use 64 bits operation to check for overflow. But only
     * guaranteed in C99+ */
    if (remainder > UINT32_MAX / mul || major > UINT32_MAX / mul ||
        major * mul > UINT32_MAX - remainder * mul / div)
      return RESAMPLER_ERR_OVERFLOW;
    *result = remainder * mul / div + major * mul;
    return RESAMPLER_ERR_SUCCESS;
  }
  
  static int update_filter(SpeexResamplerState *st) {
    spx_uint32_t old_length = st->filt_len;
    spx_uint32_t old_alloc_size = st->mem_alloc_size;
    int use_direct;
    spx_uint32_t min_sinc_table_length;
    spx_uint32_t min_alloc_size;
  
    st->int_advance = st->num_rate / st->den_rate;
    st->frac_advance = st->num_rate % st->den_rate;
    st->oversample = quality_map[0].oversample;
    st->filt_len = quality_map[0].base_length;
  
    if (st->num_rate > st->den_rate) {
      /* down-sampling */
      st->cutoff =
          quality_map[0].downsample_bandwidth * st->den_rate / st->num_rate;
      if (_muldiv(&st->filt_len, st->filt_len, st->num_rate, st->den_rate) !=
          RESAMPLER_ERR_SUCCESS)
        goto fail;
      /* Round up to make sure we have a multiple of 8 for SSE */
      st->filt_len = ((st->filt_len - 1) & (~0x7)) + 8;
      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[0].upsample_bandwidth;
    }
  
    /* Choose the resampling type that requires the least amount of memory */
  #ifdef RESAMPLE_FULL_SINC_TABLE
    use_direct = 1;
    if (INT_MAX / sizeof(spx_word16_t) / st->den_rate < st->filt_len)
      goto fail;
  #else
    use_direct =
        st->filt_len * st->den_rate <= st->filt_len * st->oversample + 8 &&
        INT_MAX / sizeof(spx_word16_t) / st->den_rate >= st->filt_len;
  #endif
    if (use_direct) {
      min_sinc_table_length = st->filt_len * st->den_rate;
    } else {
      if ((INT_MAX / sizeof(spx_word16_t) - 8) / st->oversample < st->filt_len)
        goto fail;
  
      min_sinc_table_length = st->filt_len * st->oversample + 8;
    }
    if (st->sinc_table_length < min_sinc_table_length) {
      spx_word16_t *sinc_table = (spx_word16_t *)speex_realloc(
          st->sinc_table, min_sinc_table_length * sizeof(spx_word16_t));
      if (!sinc_table)
        goto fail;
  
      st->sinc_table = sinc_table;
      st->sinc_table_length = min_sinc_table_length;
    }
    if (use_direct) {
      spx_uint32_t i;
      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[0].window_func);
        }
      }
      st->resampler_ptr = resampler_basic_direct_double;
      /*fprintf (stderr, "resampler uses direct sinc table and normalised cutoff
       * %f\n", cutoff);*/
    } else {
      spx_int32_t i;
      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[0].window_func);
      st->resampler_ptr = resampler_basic_interpolate_double;
      /*fprintf (stderr, "resampler uses interpolated sinc table and normalised
       * cutoff %f\n", cutoff);*/
    }
  
    /* 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. */
  
    /* Adding buffer_size to filt_len won't overflow here because filt_len
     could be multiplied by sizeof(spx_word16_t) above. */
    min_alloc_size = st->filt_len - 1 + st->buffer_size;
    if (min_alloc_size > st->mem_alloc_size) {
      spx_word16_t *mem;
      if (INT_MAX / sizeof(spx_word16_t) / st->nb_channels < min_alloc_size)
        goto fail;
      else if (!(mem = (spx_word16_t *)speex_realloc(
                     st->mem, st->nb_channels * min_alloc_size * sizeof(*mem))))
        goto fail;
  
      st->mem = mem;
      st->mem_alloc_size = min_alloc_size;
    }
    if (!st->started) {
      spx_uint32_t i;
      for (i = 0; i < st->nb_channels * st->mem_alloc_size; i++)
        st->mem[i] = 0;
      /*speex_warning("reinit filter");*/
    } else if (st->filt_len > old_length) {
      spx_uint32_t i;
      /* Increase the filter length */
      /*speex_warning("increase filter size");*/
      for (i = st->nb_channels; i--;) {
        spx_uint32_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 - 1 + st->magic_samples[i]; 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;
      }
    }
    return RESAMPLER_ERR_SUCCESS;
  
  fail:
    st->resampler_ptr = 0;
    /* st->mem may still contain consumed input samples for the filter.
     Restore filt_len so that filt_len - 1 still points to the position after
     the last of these samples. */
    st->filt_len = old_length;
    return RESAMPLER_ERR_ALLOC_FAILED;
  }
  
  EXPORT 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);
  }
  
  EXPORT 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) {
    SpeexResamplerState *st;
    int filter_err;
  
    if (nb_channels == 0 || ratio_num == 0 || ratio_den == 0 || quality > 10 ||
        quality < 0) {
      if (err)
        *err = RESAMPLER_ERR_INVALID_ARG;
      return NULL;
    }
    st = (SpeexResamplerState *)speex_alloc(sizeof(SpeexResamplerState));
    if (!st) {
      if (err)
        *err = RESAMPLER_ERR_ALLOC_FAILED;
      return NULL;
    }
    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;
  
    st->buffer_size = 160;
  
    /* Per channel data */
    if (!(st->last_sample =
              (spx_int32_t *)speex_alloc(nb_channels * sizeof(spx_int32_t))))
      goto fail;
    if (!(st->magic_samples =
              (spx_uint32_t *)speex_alloc(nb_channels * sizeof(spx_uint32_t))))
      goto fail;
    if (!(st->samp_frac_num =
              (spx_uint32_t *)speex_alloc(nb_channels * sizeof(spx_uint32_t))))
      goto fail;
  
    speex_resampler_set_quality(st, quality);
    speex_resampler_set_rate_frac(st, ratio_num, ratio_den, in_rate, out_rate);
  
    filter_err = update_filter(st);
    if (filter_err == RESAMPLER_ERR_SUCCESS) {
      st->initialised = 1;
    } else {
      speex_resampler_destroy(st);
      st = NULL;
    }
    if (err)
      *err = filter_err;
  
    return st;
  
  fail:
    if (err)
      *err = RESAMPLER_ERR_ALLOC_FAILED;
    speex_resampler_destroy(st);
    return NULL;
  }
  
  EXPORT 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,
                                            spx_uint32_t *in_len,
                                            spx_word16_t *out,
                                            spx_uint32_t *out_len) {
    int j = 0;
    const int N = st->filt_len;
    int out_sample = 0;
    spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size;
    spx_uint32_t ilen;
  
    st->started = 1;
  
    /* Call the right resampler through the function ptr */
    out_sample = st->resampler_ptr(st, channel_index, mem, 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;
    st->last_sample[channel_index] -= *in_len;
  
    ilen = *in_len;
  
    for (j = 0; j < N - 1; ++j)
      mem[j] = mem[j + ilen];
  
    return RESAMPLER_ERR_SUCCESS;
  }
  
  static int speex_resampler_magic(SpeexResamplerState *st,
                                   spx_uint32_t channel_index, spx_word16_t **out,
                                   spx_uint32_t out_len) {
    spx_uint32_t tmp_in_len = st->magic_samples[channel_index];
    spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size;
    const int N = st->filt_len;
  
    speex_resampler_process_native(st, channel_index, &tmp_in_len, *out,
                                   &out_len);
  
    st->magic_samples[channel_index] -= tmp_in_len;
  
    /* If we couldn't process all "magic" input samples, save the rest for next
     * time */
    if (st->magic_samples[channel_index]) {
      spx_uint32_t i;
      for (i = 0; i < st->magic_samples[channel_index]; i++)
        mem[N - 1 + i] = mem[N - 1 + i + tmp_in_len];
    }
    *out += out_len * st->out_stride;
    return out_len;
  }
  
  EXPORT 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) {
    int j;
    spx_uint32_t ilen = *in_len;
    spx_uint32_t olen = *out_len;
    spx_word16_t *x = st->mem + channel_index * st->mem_alloc_size;
    const int filt_offs = st->filt_len - 1;
    const spx_uint32_t xlen = st->mem_alloc_size - filt_offs;
    const int istride = st->in_stride;
  
    if (st->magic_samples[channel_index])
      olen -= speex_resampler_magic(st, channel_index, &out, olen);
    if (!st->magic_samples[channel_index]) {
      while (ilen && olen) {
        spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen;
        spx_uint32_t ochunk = olen;
  
        if (in) {
          for (j = 0; j < ichunk; ++j)
            x[j + filt_offs] = in[j * istride];
        } else {
          for (j = 0; j < ichunk; ++j)
            x[j + filt_offs] = 0;
        }
        speex_resampler_process_native(st, channel_index, &ichunk, out, &ochunk);
        ilen -= ichunk;
        olen -= ochunk;
        out += ochunk * st->out_stride;
        if (in)
          in += ichunk * istride;
      }
    }
    *in_len -= ilen;
    *out_len -= olen;
    return st->resampler_ptr ? RESAMPLER_ERR_ALLOC_FAILED : RESAMPLER_ERR_SUCCESS;
  }
  
  static spx_uint32_t _gcd(spx_uint32_t a, spx_uint32_t b) {
    while (b != 0) {
      spx_uint32_t temp = a;
  
      a = b;
      b = temp % b;
    }
    return a;
  }
  
  EXPORT 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 (ratio_num == 0 || ratio_den == 0)
      return RESAMPLER_ERR_INVALID_ARG;
  
    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;
  
    fact = _gcd(st->num_rate, st->den_rate);
  
    st->num_rate /= fact;
    st->den_rate /= fact;
  
    if (old_den > 0) {
      for (i = 0; i < st->nb_channels; i++) {
        if (_muldiv(&st->samp_frac_num[i], st->samp_frac_num[i], st->den_rate,
                    old_den) != RESAMPLER_ERR_SUCCESS)
          return RESAMPLER_ERR_OVERFLOW;
        /* Safety net */
        if (st->samp_frac_num[i] >= st->den_rate)
          st->samp_frac_num[i] = st->den_rate - 1;
      }
    }
  
    if (st->initialised)
      return update_filter(st);
    return RESAMPLER_ERR_SUCCESS;
  }
  
  EXPORT 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)
      return update_filter(st);
    return RESAMPLER_ERR_SUCCESS;
  }
  
  EXPORT int speex_resampler_get_output_latency(SpeexResamplerState *st) {
    return ((st->filt_len / 2) * st->den_rate + (st->num_rate >> 1)) /
           st->num_rate;
  }
  
  EXPORT 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;
  }
  
  EXPORT int speex_resampler_reset_mem(SpeexResamplerState *st) {
    spx_uint32_t i;
    for (i = 0; i < st->nb_channels; i++) {
      st->last_sample[i] = 0;
      st->magic_samples[i] = 0;
      st->samp_frac_num[i] = 0;
    }
    for (i = 0; i < st->nb_channels * (st->filt_len - 1); i++)
      st->mem[i] = 0;
    return RESAMPLER_ERR_SUCCESS;
  }
  