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Diffstat (limited to 'lib/rbcodec/codecs/libspeex/preprocess.c')
| -rw-r--r-- | lib/rbcodec/codecs/libspeex/preprocess.c | 1185 |
1 files changed, 1185 insertions, 0 deletions
diff --git a/lib/rbcodec/codecs/libspeex/preprocess.c b/lib/rbcodec/codecs/libspeex/preprocess.c new file mode 100644 index 0000000000..07a2ad3479 --- /dev/null +++ b/lib/rbcodec/codecs/libspeex/preprocess.c @@ -0,0 +1,1185 @@ +/* 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-speex.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; +} |