/* Copyright (C) 2003 Epic Games (written by Jean-Marc Valin)
   Copyright (C) 2004-2006 Epic Games 
   
   File: preprocess.c
   Preprocessor with denoising based on the algorithm by Ephraim and Malah

   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.
*/


/*
   Recommended papers:
   
   Y. Ephraim and D. Malah, "Speech enhancement using minimum mean-square error
   short-time spectral amplitude estimator". IEEE Transactions on Acoustics, 
   Speech and Signal Processing, vol. ASSP-32, no. 6, pp. 1109-1121, 1984.
   
   Y. Ephraim and D. Malah, "Speech enhancement using minimum mean-square error
   log-spectral amplitude estimator". IEEE Transactions on Acoustics, Speech and 
   Signal Processing, vol. ASSP-33, no. 2, pp. 443-445, 1985.
   
   I. Cohen and B. Berdugo, "Speech enhancement for non-stationary noise environments".
   Signal Processing, vol. 81, no. 2, pp. 2403-2418, 2001.

   Stefan Gustafsson, Rainer Martin, Peter Jax, and Peter Vary. "A psychoacoustic 
   approach to combined acoustic echo cancellation and noise reduction". IEEE 
   Transactions on Speech and Audio Processing, 2002.
   
   J.-M. Valin, J. Rouat, and F. Michaud, "Microphone array post-filter for separation
   of simultaneous non-stationary sources". In Proceedings IEEE International 
   Conference on Acoustics, Speech, and Signal Processing, 2004.
*/

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

#include <math.h>
#include "speex/speex_preprocess.h"
#include "speex/speex_echo.h"
#include "arch.h"
#include "fftwrap.h"
#include "filterbank.h"
#include "math_approx.h"
#include "os_support.h"

#ifndef M_PI
#define M_PI 3.14159263
#endif

#define LOUDNESS_EXP 5.f
#define AMP_SCALE .001f
#define AMP_SCALE_1 1000.f
      
#define NB_BANDS 24

#define SPEECH_PROB_START_DEFAULT       QCONST16(0.35f,15)
#define SPEECH_PROB_CONTINUE_DEFAULT    QCONST16(0.20f,15)
#define NOISE_SUPPRESS_DEFAULT       -15
#define ECHO_SUPPRESS_DEFAULT        -40
#define ECHO_SUPPRESS_ACTIVE_DEFAULT -15

#ifndef NULL
#define NULL 0
#endif

#define SQR(x) ((x)*(x))
#define SQR16(x) (MULT16_16((x),(x)))
#define SQR16_Q15(x) (MULT16_16_Q15((x),(x)))

#ifdef FIXED_POINT
static inline spx_word16_t DIV32_16_Q8(spx_word32_t a, spx_word32_t b)
{
   if (SHR32(a,7) >= b)
   {
      return 32767;
   } else {
      if (b>=QCONST32(1,23))
      {
         a = SHR32(a,8);
         b = SHR32(b,8);
      }
      if (b>=QCONST32(1,19))
      {
         a = SHR32(a,4);
         b = SHR32(b,4);
      }
      if (b>=QCONST32(1,15))
      {
         a = SHR32(a,4);
         b = SHR32(b,4);
      }
      a = SHL32(a,8);
      return PDIV32_16(a,b);
   }
   
}
static inline spx_word16_t DIV32_16_Q15(spx_word32_t a, spx_word32_t b)
{
   if (SHR32(a,15) >= b)
   {
      return 32767;
   } else {
      if (b>=QCONST32(1,23))
      {
         a = SHR32(a,8);
         b = SHR32(b,8);
      }
      if (b>=QCONST32(1,19))
      {
         a = SHR32(a,4);
         b = SHR32(b,4);
      }
      if (b>=QCONST32(1,15))
      {
         a = SHR32(a,4);
         b = SHR32(b,4);
      }
      a = SHL32(a,15)-a;
      return DIV32_16(a,b);
   }
}
#define SNR_SCALING 256.f
#define SNR_SCALING_1 0.0039062f
#define SNR_SHIFT 8

#define FRAC_SCALING 32767.f
#define FRAC_SCALING_1 3.0518e-05
#define FRAC_SHIFT 1

#define EXPIN_SCALING 2048.f
#define EXPIN_SCALING_1 0.00048828f
#define EXPIN_SHIFT 11
#define EXPOUT_SCALING_1 1.5259e-05

#define NOISE_SHIFT 7

#else

#define DIV32_16_Q8(a,b) ((a)/(b))
#define DIV32_16_Q15(a,b) ((a)/(b))
#define SNR_SCALING 1.f
#define SNR_SCALING_1 1.f
#define SNR_SHIFT 0
#define FRAC_SCALING 1.f
#define FRAC_SCALING_1 1.f
#define FRAC_SHIFT 0
#define NOISE_SHIFT 0

#define EXPIN_SCALING 1.f
#define EXPIN_SCALING_1 1.f
#define EXPOUT_SCALING_1 1.f

#endif

/** Speex pre-processor state. */
struct SpeexPreprocessState_ {
   /* Basic info */
   int    frame_size;        /**< Number of samples processed each time */
   int    ps_size;           /**< Number of points in the power spectrum */
   int    sampling_rate;     /**< Sampling rate of the input/output */
   int    nbands;
   FilterBank *bank;
   
   /* Parameters */
   int    denoise_enabled;
   int    vad_enabled;
   int    dereverb_enabled;
   spx_word16_t  reverb_decay;
   spx_word16_t  reverb_level;
   spx_word16_t speech_prob_start;
   spx_word16_t speech_prob_continue;
   int    noise_suppress;
   int    echo_suppress;
   int    echo_suppress_active;
   SpeexEchoState *echo_state;
   
   spx_word16_t	speech_prob;  /**< Probability last frame was speech */

   /* DSP-related arrays */
   spx_word16_t *frame;      /**< Processing frame (2*ps_size) */
   spx_word16_t *ft;         /**< Processing frame in freq domain (2*ps_size) */
   spx_word32_t *ps;         /**< Current power spectrum */
   spx_word16_t *gain2;      /**< Adjusted gains */
   spx_word16_t *gain_floor; /**< Minimum gain allowed */
   spx_word16_t *window;     /**< Analysis/Synthesis window */
   spx_word32_t *noise;      /**< Noise estimate */
   spx_word32_t *reverb_estimate; /**< Estimate of reverb energy */
   spx_word32_t *old_ps;     /**< Power spectrum for last frame */
   spx_word16_t *gain;       /**< Ephraim Malah gain */
   spx_word16_t *prior;      /**< A-priori SNR */
   spx_word16_t *post;       /**< A-posteriori SNR */

   spx_word32_t *S;          /**< Smoothed power spectrum */
   spx_word32_t *Smin;       /**< See Cohen paper */
   spx_word32_t *Stmp;       /**< See Cohen paper */
   int *update_prob;         /**< Probability of speech presence for noise update */

   spx_word16_t *zeta;       /**< Smoothed a priori SNR */
   spx_word32_t *echo_noise;
   spx_word32_t *residual_echo;

   /* Misc */
   spx_word16_t *inbuf;      /**< Input buffer (overlapped analysis) */
   spx_word16_t *outbuf;     /**< Output buffer (for overlap and add) */

   /* AGC stuff, only for floating point for now */
#ifndef FIXED_POINT
   int    agc_enabled;
   float  agc_level;
   float  loudness_accum;
   float *loudness_weight;   /**< Perceptual loudness curve */
   float  loudness;          /**< Loudness estimate */
   float  agc_gain;          /**< Current AGC gain */
   float  max_gain;          /**< Maximum gain allowed */
   float  max_increase_step; /**< Maximum increase in gain from one frame to another */
   float  max_decrease_step; /**< Maximum decrease in gain from one frame to another */
   float  prev_loudness;     /**< Loudness of previous frame */
   float  init_max;          /**< Current gain limit during initialisation */
#endif
   int    nb_adapt;          /**< Number of frames used for adaptation so far */
   int    was_speech;
   int    min_count;         /**< Number of frames processed so far */
   void  *fft_lookup;        /**< Lookup table for the FFT */
#ifdef FIXED_POINT
   int    frame_shift;
#endif
};


static void conj_window(spx_word16_t *w, int len)
{
   int i;
   for (i=0;i<len;i++)
   {
      spx_word16_t tmp;
#ifdef FIXED_POINT
      spx_word16_t x = DIV32_16(MULT16_16(32767,i),len);
#else      
      spx_word16_t x = DIV32_16(MULT16_16(QCONST16(4.f,13),i),len);
#endif
      int inv=0;
      if (x<QCONST16(1.f,13))
      {
      } else if (x<QCONST16(2.f,13))
      {
         x=QCONST16(2.f,13)-x;
         inv=1;
      } else if (x<QCONST16(3.f,13))
      {
         x=x-QCONST16(2.f,13);
         inv=1;
      } else {
         x=QCONST16(2.f,13)-x+QCONST16(2.f,13); /* 4 - x */
      }
      x = MULT16_16_Q14(QCONST16(1.271903f,14), x);
      tmp = SQR16_Q15(QCONST16(.5f,15)-MULT16_16_P15(QCONST16(.5f,15),spx_cos_norm(SHL32(EXTEND32(x),2))));
      if (inv)
         tmp=SUB16(Q15_ONE,tmp);
      w[i]=spx_sqrt(SHL32(EXTEND32(tmp),15));
   }
}

      
#ifdef FIXED_POINT
/* This function approximates the gain function 
   y = gamma(1.25)^2 * M(-.25;1;-x) / sqrt(x)  
   which multiplied by xi/(1+xi) is the optimal gain
   in the loudness domain ( sqrt[amplitude] )
   Input in Q11 format, output in Q15
*/
static inline spx_word32_t hypergeom_gain(spx_word32_t xx)
{
   int ind;
   spx_word16_t frac;
   /* Q13 table */
   static const spx_word16_t table[21] = {
       6730,  8357,  9868, 11267, 12563, 13770, 14898,
      15959, 16961, 17911, 18816, 19682, 20512, 21311,
      22082, 22827, 23549, 24250, 24931, 25594, 26241};
      ind = SHR32(xx,10);
      if (ind<0)
         return Q15_ONE;
      if (ind>19)
         return ADD32(EXTEND32(Q15_ONE),EXTEND32(DIV32_16(QCONST32(.1296,23), SHR32(xx,EXPIN_SHIFT-SNR_SHIFT))));
      frac = SHL32(xx-SHL32(ind,10),5);
      return SHL32(DIV32_16(PSHR32(MULT16_16(Q15_ONE-frac,table[ind]) + MULT16_16(frac,table[ind+1]),7),(spx_sqrt(SHL32(xx,15)+6711))),7);
}

static inline spx_word16_t qcurve(spx_word16_t x)
{
   x = MAX16(x, 1);
   return DIV32_16(SHL32(EXTEND32(32767),9),ADD16(512,MULT16_16_Q15(QCONST16(.60f,15),DIV32_16(32767,x))));
}

/* Compute the gain floor based on different floors for the background noise and residual echo */
static void compute_gain_floor(int noise_suppress, int effective_echo_suppress, spx_word32_t *noise, spx_word32_t *echo, spx_word16_t *gain_floor, int len)
{
   int i;
   
   if (noise_suppress > effective_echo_suppress)
   {
      spx_word16_t noise_gain, gain_ratio;
      noise_gain = EXTRACT16(MIN32(Q15_ONE,SHR32(spx_exp(MULT16_16(QCONST16(0.11513,11),noise_suppress)),1)));
      gain_ratio = EXTRACT16(MIN32(Q15_ONE,SHR32(spx_exp(MULT16_16(QCONST16(.2302585f,11),effective_echo_suppress-noise_suppress)),1)));

      /* gain_floor = sqrt [ (noise*noise_floor + echo*echo_floor) / (noise+echo) ] */
      for (i=0;i<len;i++)
         gain_floor[i] = MULT16_16_Q15(noise_gain,
                                       spx_sqrt(SHL32(EXTEND32(DIV32_16_Q15(PSHR32(noise[i],NOISE_SHIFT) + MULT16_32_Q15(gain_ratio,echo[i]),
                                             (1+PSHR32(noise[i],NOISE_SHIFT) + echo[i]) )),15)));
   } else {
      spx_word16_t echo_gain, gain_ratio;
      echo_gain = EXTRACT16(MIN32(Q15_ONE,SHR32(spx_exp(MULT16_16(QCONST16(0.11513,11),effective_echo_suppress)),1)));
      gain_ratio = EXTRACT16(MIN32(Q15_ONE,SHR32(spx_exp(MULT16_16(QCONST16(.2302585f,11),noise_suppress-effective_echo_suppress)),1)));

      /* gain_floor = sqrt [ (noise*noise_floor + echo*echo_floor) / (noise+echo) ] */
      for (i=0;i<len;i++)
         gain_floor[i] = MULT16_16_Q15(echo_gain,
                                       spx_sqrt(SHL32(EXTEND32(DIV32_16_Q15(MULT16_32_Q15(gain_ratio,PSHR32(noise[i],NOISE_SHIFT)) + echo[i],
                                             (1+PSHR32(noise[i],NOISE_SHIFT) + echo[i]) )),15)));
   }
}

#else
/* This function approximates the gain function 
   y = gamma(1.25)^2 * M(-.25;1;-x) / sqrt(x)  
   which multiplied by xi/(1+xi) is the optimal gain
   in the loudness domain ( sqrt[amplitude] )
*/
static inline spx_word32_t hypergeom_gain(spx_word32_t xx)
{
   int ind;
   float integer, frac;
   float x;
   static const float table[21] = {
      0.82157f, 1.02017f, 1.20461f, 1.37534f, 1.53363f, 1.68092f, 1.81865f,
      1.94811f, 2.07038f, 2.18638f, 2.29688f, 2.40255f, 2.50391f, 2.60144f,
      2.69551f, 2.78647f, 2.87458f, 2.96015f, 3.04333f, 3.12431f, 3.20326f};
      x = EXPIN_SCALING_1*xx;
      integer = floor(2*x);
      ind = (int)integer;
      if (ind<0)
         return FRAC_SCALING;
      if (ind>19)
         return FRAC_SCALING*(1+.1296/x);
      frac = 2*x-integer;
      return FRAC_SCALING*((1-frac)*table[ind] + frac*table[ind+1])/sqrt(x+.0001f);
}

static inline spx_word16_t qcurve(spx_word16_t x)
{
   return 1.f/(1.f+.15f/(SNR_SCALING_1*x));
}

static void compute_gain_floor(int noise_suppress, int effective_echo_suppress, spx_word32_t *noise, spx_word32_t *echo, spx_word16_t *gain_floor, int len)
{
   int i;
   float echo_floor;
   float noise_floor;

   noise_floor = exp(.2302585f*noise_suppress);
   echo_floor = exp(.2302585f*effective_echo_suppress);

   /* Compute the gain floor based on different floors for the background noise and residual echo */
   for (i=0;i<len;i++)
      gain_floor[i] = FRAC_SCALING*sqrt(noise_floor*PSHR32(noise[i],NOISE_SHIFT) + echo_floor*echo[i])/sqrt(1+PSHR32(noise[i],NOISE_SHIFT) + echo[i]);
}

#endif
EXPORT SpeexPreprocessState *speex_preprocess_state_init(int frame_size, int sampling_rate)
{
   int i;
   int N, N3, N4, M;

   SpeexPreprocessState *st = (SpeexPreprocessState *)speex_alloc(sizeof(SpeexPreprocessState));
   st->frame_size = frame_size;

   /* Round ps_size down to the nearest power of two */
#if 0
   i=1;
   st->ps_size = st->frame_size;
   while(1)
   {
      if (st->ps_size & ~i)
      {
         st->ps_size &= ~i;
         i<<=1;
      } else {
         break;
      }
   }
   
   
   if (st->ps_size < 3*st->frame_size/4)
      st->ps_size = st->ps_size * 3 / 2;
#else
   st->ps_size = st->frame_size;
#endif

   N = st->ps_size;
   N3 = 2*N - st->frame_size;
   N4 = st->frame_size - N3;
   
   st->sampling_rate = sampling_rate;
   st->denoise_enabled = 1;
   st->vad_enabled = 0;
   st->dereverb_enabled = 0;
   st->reverb_decay = 0;
   st->reverb_level = 0;
   st->noise_suppress = NOISE_SUPPRESS_DEFAULT;
   st->echo_suppress = ECHO_SUPPRESS_DEFAULT;
   st->echo_suppress_active = ECHO_SUPPRESS_ACTIVE_DEFAULT;

   st->speech_prob_start = SPEECH_PROB_START_DEFAULT;
   st->speech_prob_continue = SPEECH_PROB_CONTINUE_DEFAULT;

   st->echo_state = NULL;
   
   st->nbands = NB_BANDS;
   M = st->nbands;
   st->bank = filterbank_new(M, sampling_rate, N, 1);
   
   st->frame = (spx_word16_t*)speex_alloc(2*N*sizeof(spx_word16_t));
   st->window = (spx_word16_t*)speex_alloc(2*N*sizeof(spx_word16_t));
   st->ft = (spx_word16_t*)speex_alloc(2*N*sizeof(spx_word16_t));
   
   st->ps = (spx_word32_t*)speex_alloc((N+M)*sizeof(spx_word32_t));
   st->noise = (spx_word32_t*)speex_alloc((N+M)*sizeof(spx_word32_t));
   st->echo_noise = (spx_word32_t*)speex_alloc((N+M)*sizeof(spx_word32_t));
   st->residual_echo = (spx_word32_t*)speex_alloc((N+M)*sizeof(spx_word32_t));
   st->reverb_estimate = (spx_word32_t*)speex_alloc((N+M)*sizeof(spx_word32_t));
   st->old_ps = (spx_word32_t*)speex_alloc((N+M)*sizeof(spx_word32_t));
   st->prior = (spx_word16_t*)speex_alloc((N+M)*sizeof(spx_word16_t));
   st->post = (spx_word16_t*)speex_alloc((N+M)*sizeof(spx_word16_t));
   st->gain = (spx_word16_t*)speex_alloc((N+M)*sizeof(spx_word16_t));
   st->gain2 = (spx_word16_t*)speex_alloc((N+M)*sizeof(spx_word16_t));
   st->gain_floor = (spx_word16_t*)speex_alloc((N+M)*sizeof(spx_word16_t));
   st->zeta = (spx_word16_t*)speex_alloc((N+M)*sizeof(spx_word16_t));
   
   st->S = (spx_word32_t*)speex_alloc(N*sizeof(spx_word32_t));
   st->Smin = (spx_word32_t*)speex_alloc(N*sizeof(spx_word32_t));
   st->Stmp = (spx_word32_t*)speex_alloc(N*sizeof(spx_word32_t));
   st->update_prob = (int*)speex_alloc(N*sizeof(int));
   
   st->inbuf = (spx_word16_t*)speex_alloc(N3*sizeof(spx_word16_t));
   st->outbuf = (spx_word16_t*)speex_alloc(N3*sizeof(spx_word16_t));

   conj_window(st->window, 2*N3);
   for (i=2*N3;i<2*st->ps_size;i++)
      st->window[i]=Q15_ONE;
   
   if (N4>0)
   {
      for (i=N3-1;i>=0;i--)
      {
         st->window[i+N3+N4]=st->window[i+N3];
         st->window[i+N3]=1;
      }
   }
   for (i=0;i<N+M;i++)
   {
      st->noise[i]=QCONST32(1.f,NOISE_SHIFT);
      st->reverb_estimate[i]=0;
      st->old_ps[i]=1;
      st->gain[i]=Q15_ONE;
      st->post[i]=SHL16(1, SNR_SHIFT);
      st->prior[i]=SHL16(1, SNR_SHIFT);
   }

   for (i=0;i<N;i++)
      st->update_prob[i] = 1;
   for (i=0;i<N3;i++)
   {
      st->inbuf[i]=0;
      st->outbuf[i]=0;
   }
#ifndef FIXED_POINT
   st->agc_enabled = 0;
   st->agc_level = 8000;
   st->loudness_weight = (float*)speex_alloc(N*sizeof(float));
   for (i=0;i<N;i++)
   {
      float ff=((float)i)*.5*sampling_rate/((float)N);
      /*st->loudness_weight[i] = .5f*(1.f/(1.f+ff/8000.f))+1.f*exp(-.5f*(ff-3800.f)*(ff-3800.f)/9e5f);*/
      st->loudness_weight[i] = .35f-.35f*ff/16000.f+.73f*exp(-.5f*(ff-3800)*(ff-3800)/9e5f);
      if (st->loudness_weight[i]<.01f)
         st->loudness_weight[i]=.01f;
      st->loudness_weight[i] *= st->loudness_weight[i];
   }
   /*st->loudness = pow(AMP_SCALE*st->agc_level,LOUDNESS_EXP);*/
   st->loudness = 1e-15;
   st->agc_gain = 1;
   st->max_gain = 30;
   st->max_increase_step = exp(0.11513f * 12.*st->frame_size / st->sampling_rate);
   st->max_decrease_step = exp(-0.11513f * 40.*st->frame_size / st->sampling_rate);
   st->prev_loudness = 1;
   st->init_max = 1;
#endif
   st->was_speech = 0;

   st->fft_lookup = spx_fft_init(2*N);

   st->nb_adapt=0;
   st->min_count=0;
   return st;
}

EXPORT void speex_preprocess_state_destroy(SpeexPreprocessState *st)
{
   speex_free(st->frame);
   speex_free(st->ft);
   speex_free(st->ps);
   speex_free(st->gain2);
   speex_free(st->gain_floor);
   speex_free(st->window);
   speex_free(st->noise);
   speex_free(st->reverb_estimate);
   speex_free(st->old_ps);
   speex_free(st->gain);
   speex_free(st->prior);
   speex_free(st->post);
#ifndef FIXED_POINT
   speex_free(st->loudness_weight);
#endif
   speex_free(st->echo_noise);
   speex_free(st->residual_echo);

   speex_free(st->S);
   speex_free(st->Smin);
   speex_free(st->Stmp);
   speex_free(st->update_prob);
   speex_free(st->zeta);

   speex_free(st->inbuf);
   speex_free(st->outbuf);

   spx_fft_destroy(st->fft_lookup);
   filterbank_destroy(st->bank);
   speex_free(st);
}

/* FIXME: The AGC doesn't work yet with fixed-point*/
#ifndef FIXED_POINT
static void speex_compute_agc(SpeexPreprocessState *st, spx_word16_t Pframe, spx_word16_t *ft)
{
   int i;
   int N = st->ps_size;
   float target_gain;
   float loudness=1.f;
   float rate;
   
   for (i=2;i<N;i++)
   {
      loudness += 2.f*N*st->ps[i]* st->loudness_weight[i];
   }
   loudness=sqrt(loudness);
      /*if (loudness < 2*pow(st->loudness, 1.0/LOUDNESS_EXP) &&
   loudness*2 > pow(st->loudness, 1.0/LOUDNESS_EXP))*/
   if (Pframe>.3f)
   {
      /*rate=2.0f*Pframe*Pframe/(1+st->nb_loudness_adapt);*/
      rate = .03*Pframe*Pframe;
      st->loudness = (1-rate)*st->loudness + (rate)*pow(AMP_SCALE*loudness, LOUDNESS_EXP);
      st->loudness_accum = (1-rate)*st->loudness_accum + rate;
      if (st->init_max < st->max_gain && st->nb_adapt > 20)
         st->init_max *= 1.f + .1f*Pframe*Pframe;
   }
   /*printf ("%f %f %f %f\n", Pframe, loudness, pow(st->loudness, 1.0f/LOUDNESS_EXP), st->loudness2);*/
   
   target_gain = AMP_SCALE*st->agc_level*pow(st->loudness/(1e-4+st->loudness_accum), -1.0f/LOUDNESS_EXP);

   if ((Pframe>.5  && st->nb_adapt > 20) || target_gain < st->agc_gain)
   {
      if (target_gain > st->max_increase_step*st->agc_gain)
         target_gain = st->max_increase_step*st->agc_gain;
      if (target_gain < st->max_decrease_step*st->agc_gain && loudness < 10*st->prev_loudness)
         target_gain = st->max_decrease_step*st->agc_gain;
      if (target_gain > st->max_gain)
         target_gain = st->max_gain;
      if (target_gain > st->init_max)
         target_gain = st->init_max;
   
      st->agc_gain = target_gain;
   }
   /*fprintf (stderr, "%f %f %f\n", loudness, (float)AMP_SCALE_1*pow(st->loudness, 1.0f/LOUDNESS_EXP), st->agc_gain);*/
      
   for (i=0;i<2*N;i++)
      ft[i] *= st->agc_gain;
   st->prev_loudness = loudness;
}
#endif

static void preprocess_analysis(SpeexPreprocessState *st, spx_int16_t *x)
{
   int i;
   int N = st->ps_size;
   int N3 = 2*N - st->frame_size;
   int N4 = st->frame_size - N3;
   spx_word32_t *ps=st->ps;

   /* 'Build' input frame */
   for (i=0;i<N3;i++)
      st->frame[i]=st->inbuf[i];
   for (i=0;i<st->frame_size;i++)
      st->frame[N3+i]=x[i];
   
   /* Update inbuf */
   for (i=0;i<N3;i++)
      st->inbuf[i]=x[N4+i];

   /* Windowing */
   for (i=0;i<2*N;i++)
      st->frame[i] = MULT16_16_Q15(st->frame[i], st->window[i]);

#ifdef FIXED_POINT
   {
      spx_word16_t max_val=0;
      for (i=0;i<2*N;i++)
         max_val = MAX16(max_val, ABS16(st->frame[i]));
      st->frame_shift = 14-spx_ilog2(EXTEND32(max_val));
      for (i=0;i<2*N;i++)
         st->frame[i] = SHL16(st->frame[i], st->frame_shift);
   }
#endif
   
   /* Perform FFT */
   spx_fft(st->fft_lookup, st->frame, st->ft);
         
   /* Power spectrum */
   ps[0]=MULT16_16(st->ft[0],st->ft[0]);
   for (i=1;i<N;i++)
      ps[i]=MULT16_16(st->ft[2*i-1],st->ft[2*i-1]) + MULT16_16(st->ft[2*i],st->ft[2*i]);
   for (i=0;i<N;i++)
      st->ps[i] = PSHR32(st->ps[i], 2*st->frame_shift);

   filterbank_compute_bank32(st->bank, ps, ps+N);
}

static void update_noise_prob(SpeexPreprocessState *st)
{
   int i;
   int min_range;
   int N = st->ps_size;

   for (i=1;i<N-1;i++)
      st->S[i] =  MULT16_32_Q15(QCONST16(.8f,15),st->S[i]) + MULT16_32_Q15(QCONST16(.05f,15),st->ps[i-1]) 
                      + MULT16_32_Q15(QCONST16(.1f,15),st->ps[i]) + MULT16_32_Q15(QCONST16(.05f,15),st->ps[i+1]);
   st->S[0] =  MULT16_32_Q15(QCONST16(.8f,15),st->S[0]) + MULT16_32_Q15(QCONST16(.2f,15),st->ps[0]);
   st->S[N-1] =  MULT16_32_Q15(QCONST16(.8f,15),st->S[N-1]) + MULT16_32_Q15(QCONST16(.2f,15),st->ps[N-1]);
   
   if (st->nb_adapt==1)
   {
      for (i=0;i<N;i++)
         st->Smin[i] = st->Stmp[i] = 0;
   }

   if (st->nb_adapt < 100)
      min_range = 15;
   else if (st->nb_adapt < 1000)
      min_range = 50;
   else if (st->nb_adapt < 10000)
      min_range = 150;
   else
      min_range = 300;
   if (st->min_count > min_range)
   {
      st->min_count = 0;
      for (i=0;i<N;i++)
      {
         st->Smin[i] = MIN32(st->Stmp[i], st->S[i]);
         st->Stmp[i] = st->S[i];
      }
   } else {
      for (i=0;i<N;i++)
      {
         st->Smin[i] = MIN32(st->Smin[i], st->S[i]);
         st->Stmp[i] = MIN32(st->Stmp[i], st->S[i]);      
      }
   }
   for (i=0;i<N;i++)
   {
      if (MULT16_32_Q15(QCONST16(.4f,15),st->S[i]) > st->Smin[i])
         st->update_prob[i] = 1;
      else
         st->update_prob[i] = 0;
      /*fprintf (stderr, "%f ", st->S[i]/st->Smin[i]);*/
      /*fprintf (stderr, "%f ", st->update_prob[i]);*/
   }

}

#define NOISE_OVERCOMPENS 1.

void speex_echo_get_residual(SpeexEchoState *st, spx_word32_t *Yout, int len);

EXPORT int speex_preprocess(SpeexPreprocessState *st, spx_int16_t *x, spx_int32_t *echo)
{
   return speex_preprocess_run(st, x);
}

EXPORT int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x)
{
   int i;
   int M;
   int N = st->ps_size;
   int N3 = 2*N - st->frame_size;
   int N4 = st->frame_size - N3;
   spx_word32_t *ps=st->ps;
   spx_word32_t Zframe;
   spx_word16_t Pframe;
   spx_word16_t beta, beta_1;
   spx_word16_t effective_echo_suppress;
   
   st->nb_adapt++;
   if (st->nb_adapt>20000)
      st->nb_adapt = 20000;
   st->min_count++;
   
   beta = MAX16(QCONST16(.03,15),DIV32_16(Q15_ONE,st->nb_adapt));
   beta_1 = Q15_ONE-beta;
   M = st->nbands;
   /* Deal with residual echo if provided */
   if (st->echo_state)
   {
      speex_echo_get_residual(st->echo_state, st->residual_echo, N);
#ifndef FIXED_POINT
      /* If there are NaNs or ridiculous values, it'll show up in the DC and we just reset everything to zero */
      if (!(st->residual_echo[0] >=0 && st->residual_echo[0]<N*1e9f))
      {
         for (i=0;i<N;i++)
            st->residual_echo[i] = 0;
      }
#endif
      for (i=0;i<N;i++)
         st->echo_noise[i] = MAX32(MULT16_32_Q15(QCONST16(.6f,15),st->echo_noise[i]), st->residual_echo[i]);
      filterbank_compute_bank32(st->bank, st->echo_noise, st->echo_noise+N);
   } else {
      for (i=0;i<N+M;i++)
         st->echo_noise[i] = 0;
   }
   preprocess_analysis(st, x);

   update_noise_prob(st);

   /* Noise estimation always updated for the 10 first frames */
   /*if (st->nb_adapt<10)
   {
      for (i=1;i<N-1;i++)
         st->update_prob[i] = 0;
   }
   */
   
   /* Update the noise estimate for the frequencies where it can be */
   for (i=0;i<N;i++)
   {
      if (!st->update_prob[i] || st->ps[i] < PSHR32(st->noise[i], NOISE_SHIFT))
         st->noise[i] = MAX32(EXTEND32(0),MULT16_32_Q15(beta_1,st->noise[i]) + MULT16_32_Q15(beta,SHL32(st->ps[i],NOISE_SHIFT)));
   }
   filterbank_compute_bank32(st->bank, st->noise, st->noise+N);

   /* Special case for first frame */
   if (st->nb_adapt==1)
      for (i=0;i<N+M;i++)
         st->old_ps[i] = ps[i];

   /* Compute a posteriori SNR */
   for (i=0;i<N+M;i++)
   {
      spx_word16_t gamma;
      
      /* Total noise estimate including residual echo and reverberation */
      spx_word32_t tot_noise = ADD32(ADD32(ADD32(EXTEND32(1), PSHR32(st->noise[i],NOISE_SHIFT)) , st->echo_noise[i]) , st->reverb_estimate[i]);
      
      /* A posteriori SNR = ps/noise - 1*/
      st->post[i] = SUB16(DIV32_16_Q8(ps[i],tot_noise), QCONST16(1.f,SNR_SHIFT));
      st->post[i]=MIN16(st->post[i], QCONST16(100.f,SNR_SHIFT));
      
      /* Computing update gamma = .1 + .9*(old/(old+noise))^2 */
      gamma = QCONST16(.1f,15)+MULT16_16_Q15(QCONST16(.89f,15),SQR16_Q15(DIV32_16_Q15(st->old_ps[i],ADD32(st->old_ps[i],tot_noise))));
      
      /* A priori SNR update = gamma*max(0,post) + (1-gamma)*old/noise */
      st->prior[i] = EXTRACT16(PSHR32(ADD32(MULT16_16(gamma,MAX16(0,st->post[i])), MULT16_16(Q15_ONE-gamma,DIV32_16_Q8(st->old_ps[i],tot_noise))), 15));
      st->prior[i]=MIN16(st->prior[i], QCONST16(100.f,SNR_SHIFT));
   }

   /*print_vec(st->post, N+M, "");*/

   /* Recursive average of the a priori SNR. A bit smoothed for the psd components */
   st->zeta[0] = PSHR32(ADD32(MULT16_16(QCONST16(.7f,15),st->zeta[0]), MULT16_16(QCONST16(.3f,15),st->prior[0])),15);
   for (i=1;i<N-1;i++)
      st->zeta[i] = PSHR32(ADD32(ADD32(ADD32(MULT16_16(QCONST16(.7f,15),st->zeta[i]), MULT16_16(QCONST16(.15f,15),st->prior[i])),
                           MULT16_16(QCONST16(.075f,15),st->prior[i-1])), MULT16_16(QCONST16(.075f,15),st->prior[i+1])),15);
   for (i=N-1;i<N+M;i++)
      st->zeta[i] = PSHR32(ADD32(MULT16_16(QCONST16(.7f,15),st->zeta[i]), MULT16_16(QCONST16(.3f,15),st->prior[i])),15);

   /* Speech probability of presence for the entire frame is based on the average filterbank a priori SNR */
   Zframe = 0;
   for (i=N;i<N+M;i++)
      Zframe = ADD32(Zframe, EXTEND32(st->zeta[i]));
   Pframe = QCONST16(.1f,15)+MULT16_16_Q15(QCONST16(.899f,15),qcurve(DIV32_16(Zframe,st->nbands)));
   
   effective_echo_suppress = EXTRACT16(PSHR32(ADD32(MULT16_16(SUB16(Q15_ONE,Pframe), st->echo_suppress), MULT16_16(Pframe, st->echo_suppress_active)),15));
   
   compute_gain_floor(st->noise_suppress, effective_echo_suppress, st->noise+N, st->echo_noise+N, st->gain_floor+N, M);
         
   /* Compute Ephraim & Malah gain speech probability of presence for each critical band (Bark scale) 
      Technically this is actually wrong because the EM gaim assumes a slightly different probability 
      distribution */
   for (i=N;i<N+M;i++)
   {
      /* See EM and Cohen papers*/
      spx_word32_t theta;
      /* Gain from hypergeometric function */
      spx_word32_t MM;
      /* Weiner filter gain */
      spx_word16_t prior_ratio;
      /* a priority probability of speech presence based on Bark sub-band alone */
      spx_word16_t P1;
      /* Speech absence a priori probability (considering sub-band and frame) */
      spx_word16_t q;
#ifdef FIXED_POINT
      spx_word16_t tmp;
#endif
      
      prior_ratio = PDIV32_16(SHL32(EXTEND32(st->prior[i]), 15), ADD16(st->prior[i], SHL32(1,SNR_SHIFT)));
      theta = MULT16_32_P15(prior_ratio, QCONST32(1.f,EXPIN_SHIFT)+SHL32(EXTEND32(st->post[i]),EXPIN_SHIFT-SNR_SHIFT));

      MM = hypergeom_gain(theta);
      /* Gain with bound */
      st->gain[i] = EXTRACT16(MIN32(Q15_ONE, MULT16_32_Q15(prior_ratio, MM)));
      /* Save old Bark power spectrum */
      st->old_ps[i] = MULT16_32_P15(QCONST16(.2f,15),st->old_ps[i]) + MULT16_32_P15(MULT16_16_P15(QCONST16(.8f,15),SQR16_Q15(st->gain[i])),ps[i]);

      P1 = QCONST16(.199f,15)+MULT16_16_Q15(QCONST16(.8f,15),qcurve (st->zeta[i]));
      q = Q15_ONE-MULT16_16_Q15(Pframe,P1);
#ifdef FIXED_POINT
      theta = MIN32(theta, EXTEND32(32767));
/*Q8*/tmp = MULT16_16_Q15((SHL32(1,SNR_SHIFT)+st->prior[i]),EXTRACT16(MIN32(Q15ONE,SHR32(spx_exp(-EXTRACT16(theta)),1))));
      tmp = MIN16(QCONST16(3.,SNR_SHIFT), tmp); /* Prevent overflows in the next line*/
/*Q8*/tmp = EXTRACT16(PSHR32(MULT16_16(PDIV32_16(SHL32(EXTEND32(q),8),(Q15_ONE-q)),tmp),8));
      st->gain2[i]=DIV32_16(SHL32(EXTEND32(32767),SNR_SHIFT), ADD16(256,tmp));
#else
      st->gain2[i]=1/(1.f + (q/(1.f-q))*(1+st->prior[i])*exp(-theta));
#endif
   }
   /* Convert the EM gains and speech prob to linear frequency */
   filterbank_compute_psd16(st->bank,st->gain2+N, st->gain2);
   filterbank_compute_psd16(st->bank,st->gain+N, st->gain);
   
   /* Use 1 for linear gain resolution (best) or 0 for Bark gain resolution (faster) */
   if (1)
   {
      filterbank_compute_psd16(st->bank,st->gain_floor+N, st->gain_floor);
   
      /* Compute gain according to the Ephraim-Malah algorithm -- linear frequency */
      for (i=0;i<N;i++)
      {
         spx_word32_t MM;
         spx_word32_t theta;
         spx_word16_t prior_ratio;
         spx_word16_t tmp;
         spx_word16_t p;
         spx_word16_t g;
         
         /* Wiener filter gain */
         prior_ratio = PDIV32_16(SHL32(EXTEND32(st->prior[i]), 15), ADD16(st->prior[i], SHL32(1,SNR_SHIFT)));
         theta = MULT16_32_P15(prior_ratio, QCONST32(1.f,EXPIN_SHIFT)+SHL32(EXTEND32(st->post[i]),EXPIN_SHIFT-SNR_SHIFT));

         /* Optimal estimator for loudness domain */
         MM = hypergeom_gain(theta);
         /* EM gain with bound */
         g = EXTRACT16(MIN32(Q15_ONE, MULT16_32_Q15(prior_ratio, MM)));
         /* Interpolated speech probability of presence */
         p = st->gain2[i];
                  
         /* Constrain the gain to be close to the Bark scale gain */
         if (MULT16_16_Q15(QCONST16(.333f,15),g) > st->gain[i])
            g = MULT16_16(3,st->gain[i]);
         st->gain[i] = g;
         
         /* Save old power spectrum */
         st->old_ps[i] = MULT16_32_P15(QCONST16(.2f,15),st->old_ps[i]) + MULT16_32_P15(MULT16_16_P15(QCONST16(.8f,15),SQR16_Q15(st->gain[i])),ps[i]);
         
         /* Apply gain floor */
         if (st->gain[i] < st->gain_floor[i])
            st->gain[i] = st->gain_floor[i];

         /* Exponential decay model for reverberation (unused) */
         /*st->reverb_estimate[i] = st->reverb_decay*st->reverb_estimate[i] + st->reverb_decay*st->reverb_level*st->gain[i]*st->gain[i]*st->ps[i];*/
         
         /* Take into account speech probability of presence (loudness domain MMSE estimator) */
         /* gain2 = [p*sqrt(gain)+(1-p)*sqrt(gain _floor) ]^2 */
         tmp = MULT16_16_P15(p,spx_sqrt(SHL32(EXTEND32(st->gain[i]),15))) + MULT16_16_P15(SUB16(Q15_ONE,p),spx_sqrt(SHL32(EXTEND32(st->gain_floor[i]),15)));
         st->gain2[i]=SQR16_Q15(tmp);

         /* Use this if you want a log-domain MMSE estimator instead */
         /*st->gain2[i] = pow(st->gain[i], p) * pow(st->gain_floor[i],1.f-p);*/
      }
   } else {
      for (i=N;i<N+M;i++)
      {
         spx_word16_t tmp;
         spx_word16_t p = st->gain2[i];
         st->gain[i] = MAX16(st->gain[i], st->gain_floor[i]);         
         tmp = MULT16_16_P15(p,spx_sqrt(SHL32(EXTEND32(st->gain[i]),15))) + MULT16_16_P15(SUB16(Q15_ONE,p),spx_sqrt(SHL32(EXTEND32(st->gain_floor[i]),15)));
         st->gain2[i]=SQR16_Q15(tmp);
      }
      filterbank_compute_psd16(st->bank,st->gain2+N, st->gain2);
   }
   
   /* If noise suppression is off, don't apply the gain (but then why call this in the first place!) */
   if (!st->denoise_enabled)
   {
      for (i=0;i<N+M;i++)
         st->gain2[i]=Q15_ONE;
   }
      
   /* Apply computed gain */
   for (i=1;i<N;i++)
   {
      st->ft[2*i-1] = MULT16_16_P15(st->gain2[i],st->ft[2*i-1]);
      st->ft[2*i] = MULT16_16_P15(st->gain2[i],st->ft[2*i]);
   }
   st->ft[0] = MULT16_16_P15(st->gain2[0],st->ft[0]);
   st->ft[2*N-1] = MULT16_16_P15(st->gain2[N-1],st->ft[2*N-1]);
   
   /*FIXME: This *will* not work for fixed-point */
#ifndef FIXED_POINT
   if (st->agc_enabled)
      speex_compute_agc(st, Pframe, st->ft);
#endif

   /* Inverse FFT with 1/N scaling */
   spx_ifft(st->fft_lookup, st->ft, st->frame);
   /* Scale back to original (lower) amplitude */
   for (i=0;i<2*N;i++)
      st->frame[i] = PSHR16(st->frame[i], st->frame_shift);

   /*FIXME: This *will* not work for fixed-point */
#ifndef FIXED_POINT
   if (st->agc_enabled)
   {
      float max_sample=0;
      for (i=0;i<2*N;i++)
         if (fabs(st->frame[i])>max_sample)
            max_sample = fabs(st->frame[i]);
      if (max_sample>28000.f)
      {
         float damp = 28000.f/max_sample;
         for (i=0;i<2*N;i++)
            st->frame[i] *= damp;
      }
   }
#endif
   
   /* Synthesis window (for WOLA) */
   for (i=0;i<2*N;i++)
      st->frame[i] = MULT16_16_Q15(st->frame[i], st->window[i]);

   /* Perform overlap and add */
   for (i=0;i<N3;i++)
      x[i] = st->outbuf[i] + st->frame[i];
   for (i=0;i<N4;i++)
      x[N3+i] = st->frame[N3+i];
   
   /* Update outbuf */
   for (i=0;i<N3;i++)
      st->outbuf[i] = st->frame[st->frame_size+i];

   /* FIXME: This VAD is a kludge */
   st->speech_prob = Pframe;
   if (st->vad_enabled)
   {
      if (st->speech_prob > st->speech_prob_start || (st->was_speech && st->speech_prob > st->speech_prob_continue))
      {
         st->was_speech=1;
         return 1;
      } else
      {
         st->was_speech=0;
         return 0;
      }
   } else {
      return 1;
   }
}

EXPORT void speex_preprocess_estimate_update(SpeexPreprocessState *st, spx_int16_t *x)
{
   int i;
   int N = st->ps_size;
   int N3 = 2*N - st->frame_size;
   int M;
   spx_word32_t *ps=st->ps;

   M = st->nbands;
   st->min_count++;
   
   preprocess_analysis(st, x);

   update_noise_prob(st);
   
   for (i=1;i<N-1;i++)
   {
      if (!st->update_prob[i] || st->ps[i] < PSHR32(st->noise[i],NOISE_SHIFT))
      {
         st->noise[i] = MULT16_32_Q15(QCONST16(.95f,15),st->noise[i]) + MULT16_32_Q15(QCONST16(.05f,15),SHL32(st->ps[i],NOISE_SHIFT));
      }
   }

   for (i=0;i<N3;i++)
      st->outbuf[i] = MULT16_16_Q15(x[st->frame_size-N3+i],st->window[st->frame_size+i]);

   /* Save old power spectrum */
   for (i=0;i<N+M;i++)
      st->old_ps[i] = ps[i];

   for (i=0;i<N;i++)
      st->reverb_estimate[i] = MULT16_32_Q15(st->reverb_decay, st->reverb_estimate[i]);
}


EXPORT int speex_preprocess_ctl(SpeexPreprocessState *state, int request, void *ptr)
{
   int i;
   SpeexPreprocessState *st;
   st=(SpeexPreprocessState*)state;
   switch(request)
   {
   case SPEEX_PREPROCESS_SET_DENOISE:
      st->denoise_enabled = (*(spx_int32_t*)ptr);
      break;
   case SPEEX_PREPROCESS_GET_DENOISE:
      (*(spx_int32_t*)ptr) = st->denoise_enabled;
      break;
#ifndef FIXED_POINT
   case SPEEX_PREPROCESS_SET_AGC:
      st->agc_enabled = (*(spx_int32_t*)ptr);
      break;
   case SPEEX_PREPROCESS_GET_AGC:
      (*(spx_int32_t*)ptr) = st->agc_enabled;
      break;
#ifndef DISABLE_FLOAT_API
   case SPEEX_PREPROCESS_SET_AGC_LEVEL:
      st->agc_level = (*(float*)ptr);
      if (st->agc_level<1)
         st->agc_level=1;
      if (st->agc_level>32768)
         st->agc_level=32768;
      break;
   case SPEEX_PREPROCESS_GET_AGC_LEVEL:
      (*(float*)ptr) = st->agc_level;
      break;
#endif /* #ifndef DISABLE_FLOAT_API */
   case SPEEX_PREPROCESS_SET_AGC_INCREMENT:
      st->max_increase_step = exp(0.11513f * (*(spx_int32_t*)ptr)*st->frame_size / st->sampling_rate);
      break;
   case SPEEX_PREPROCESS_GET_AGC_INCREMENT:
      (*(spx_int32_t*)ptr) = floor(.5+8.6858*log(st->max_increase_step)*st->sampling_rate/st->frame_size);
      break;
   case SPEEX_PREPROCESS_SET_AGC_DECREMENT:
      st->max_decrease_step = exp(0.11513f * (*(spx_int32_t*)ptr)*st->frame_size / st->sampling_rate);
      break;
   case SPEEX_PREPROCESS_GET_AGC_DECREMENT:
      (*(spx_int32_t*)ptr) = floor(.5+8.6858*log(st->max_decrease_step)*st->sampling_rate/st->frame_size);
      break;
   case SPEEX_PREPROCESS_SET_AGC_MAX_GAIN:
      st->max_gain = exp(0.11513f * (*(spx_int32_t*)ptr));
      break;
   case SPEEX_PREPROCESS_GET_AGC_MAX_GAIN:
      (*(spx_int32_t*)ptr) = floor(.5+8.6858*log(st->max_gain));
      break;
#endif
   case SPEEX_PREPROCESS_SET_VAD:
      // Disabled by mlebeau, we don't want this in the launched Google Mobile App for iPhone.
      //speex_warning("The VAD has been replaced by a hack pending a complete rewrite");
      st->vad_enabled = (*(spx_int32_t*)ptr);
      break;
   case SPEEX_PREPROCESS_GET_VAD:
      (*(spx_int32_t*)ptr) = st->vad_enabled;
      break;
   
   case SPEEX_PREPROCESS_SET_DEREVERB:
      st->dereverb_enabled = (*(spx_int32_t*)ptr);
      for (i=0;i<st->ps_size;i++)
         st->reverb_estimate[i]=0;
      break;
   case SPEEX_PREPROCESS_GET_DEREVERB:
      (*(spx_int32_t*)ptr) = st->dereverb_enabled;
      break;

   case SPEEX_PREPROCESS_SET_DEREVERB_LEVEL:
      /* FIXME: Re-enable when de-reverberation is actually enabled again */
      /*st->reverb_level = (*(float*)ptr);*/
      break;
   case SPEEX_PREPROCESS_GET_DEREVERB_LEVEL:
      /* FIXME: Re-enable when de-reverberation is actually enabled again */
      /*(*(float*)ptr) = st->reverb_level;*/
      break;
   
   case SPEEX_PREPROCESS_SET_DEREVERB_DECAY:
      /* FIXME: Re-enable when de-reverberation is actually enabled again */
      /*st->reverb_decay = (*(float*)ptr);*/
      break;
   case SPEEX_PREPROCESS_GET_DEREVERB_DECAY:
      /* FIXME: Re-enable when de-reverberation is actually enabled again */
      /*(*(float*)ptr) = st->reverb_decay;*/
      break;

   case SPEEX_PREPROCESS_SET_PROB_START:
      *(spx_int32_t*)ptr = MIN32(100,MAX32(0, *(spx_int32_t*)ptr));
      st->speech_prob_start = DIV32_16(MULT16_16(Q15ONE,*(spx_int32_t*)ptr), 100);
      break;
   case SPEEX_PREPROCESS_GET_PROB_START:
      (*(spx_int32_t*)ptr) = MULT16_16_Q15(st->speech_prob_start, 100);
      break;

   case SPEEX_PREPROCESS_SET_PROB_CONTINUE:
      *(spx_int32_t*)ptr = MIN32(100,MAX32(0, *(spx_int32_t*)ptr));
      st->speech_prob_continue = DIV32_16(MULT16_16(Q15ONE,*(spx_int32_t*)ptr), 100);
      break;
   case SPEEX_PREPROCESS_GET_PROB_CONTINUE:
      (*(spx_int32_t*)ptr) = MULT16_16_Q15(st->speech_prob_continue, 100);
      break;

   case SPEEX_PREPROCESS_SET_NOISE_SUPPRESS:
      st->noise_suppress = -ABS(*(spx_int32_t*)ptr);
      break;
   case SPEEX_PREPROCESS_GET_NOISE_SUPPRESS:
      (*(spx_int32_t*)ptr) = st->noise_suppress;
      break;
   case SPEEX_PREPROCESS_SET_ECHO_SUPPRESS:
      st->echo_suppress = -ABS(*(spx_int32_t*)ptr);
      break;
   case SPEEX_PREPROCESS_GET_ECHO_SUPPRESS:
      (*(spx_int32_t*)ptr) = st->echo_suppress;
      break;
   case SPEEX_PREPROCESS_SET_ECHO_SUPPRESS_ACTIVE:
      st->echo_suppress_active = -ABS(*(spx_int32_t*)ptr);
      break;
   case SPEEX_PREPROCESS_GET_ECHO_SUPPRESS_ACTIVE:
      (*(spx_int32_t*)ptr) = st->echo_suppress_active;
      break;
   case SPEEX_PREPROCESS_SET_ECHO_STATE:
      st->echo_state = (SpeexEchoState*)ptr;
      break;
   case SPEEX_PREPROCESS_GET_ECHO_STATE:
      (*(SpeexEchoState**)ptr) = (SpeexEchoState*)st->echo_state;
      break;
#ifndef FIXED_POINT
   case SPEEX_PREPROCESS_GET_AGC_LOUDNESS:
      (*(spx_int32_t*)ptr) = pow(st->loudness, 1.0/LOUDNESS_EXP);
      break;
   case SPEEX_PREPROCESS_GET_AGC_GAIN:
      (*(spx_int32_t*)ptr) = floor(.5+8.6858*log(st->agc_gain));
      break;
#endif
   case SPEEX_PREPROCESS_GET_PSD_SIZE:
   case SPEEX_PREPROCESS_GET_NOISE_PSD_SIZE:
      (*(spx_int32_t*)ptr) = st->ps_size;
      break;
   case SPEEX_PREPROCESS_GET_PSD:
      for(i=0;i<st->ps_size;i++)
      	((spx_int32_t *)ptr)[i] = (spx_int32_t) st->ps[i];
      break;
   case SPEEX_PREPROCESS_GET_NOISE_PSD:
      for(i=0;i<st->ps_size;i++)
      	((spx_int32_t *)ptr)[i] = (spx_int32_t) PSHR32(st->noise[i], NOISE_SHIFT);
      break;
   case SPEEX_PREPROCESS_GET_PROB:
      (*(spx_int32_t*)ptr) = MULT16_16_Q15(st->speech_prob, 100);
      break;
#ifndef FIXED_POINT
   case SPEEX_PREPROCESS_SET_AGC_TARGET:
      st->agc_level = (*(spx_int32_t*)ptr);
      if (st->agc_level<1)
         st->agc_level=1;
      if (st->agc_level>32768)
         st->agc_level=32768;
      break;
   case SPEEX_PREPROCESS_GET_AGC_TARGET:
      (*(spx_int32_t*)ptr) = st->agc_level;
      break;
#endif
   default:
      speex_warning_int("Unknown speex_preprocess_ctl request: ", request);
      return -1;
   }
   return 0;
}

#ifdef FIXED_DEBUG
long long spx_mips=0;
#endif