rq3/reaction
0
0
Fork 0
mirror of https://github.com/ReactionQuake3/reaction.git synced 2024-11-25 05:31:21 +00:00
Code Issues Projects Releases Packages Wiki Activity
reaction/code/libspeex/preprocess.c
Richard Allen b6ae8af89f Moving everything up one folder.
2013-01-04 13:27:22 +00:00

1185 lines
40 KiB
C
Raw Blame History

/* 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;
}
Reference in a new issue View git blame Copy permalink