Initial import.

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diff --git a/src/libspeex/preprocess.c b/src/libspeex/preprocess.c
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+/* 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;
+   
+   /* 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 */
+   int    nb_loudness_adapt; /**< Number of frames used for loudness adaptation so far */
+   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
+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->nb_loudness_adapt = 0;
+   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;
+}
+
+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)
+   {
+      st->nb_loudness_adapt++;
+      /*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]) > ADD32(st->Smin[i],EXTEND32(20)))
+         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);
+
+int speex_preprocess(SpeexPreprocessState *st, spx_int16_t *x, spx_int32_t *echo)
+{
+   return speex_preprocess_run(st, x);
+}
+
+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 */
+   if (st->vad_enabled)
+   {
+      if (Pframe > st->speech_prob_start || (st->was_speech && Pframe > st->speech_prob_continue))
+      {
+         st->was_speech=1;
+         return 1;
+      } else
+      {
+         st->was_speech=0;
+         return 0;
+      }
+   } else {
+      return 1;
+   }
+}
+
+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]);
+}
+
+
+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:
+      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:
+      ptr = (void*)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;
+#endif
+
+   default:
+      speex_warning_int("Unknown speex_preprocess_ctl request: ", request);
+      return -1;
+   }
+   return 0;
+}