/* Copyright (C) 2003 Epic Games Written by Jean-Marc Valin 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. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include "speex/speex_preprocess.h" #include "misc.h" #include "smallft.h" #define max(a,b) ((a) > (b) ? (a) : (b)) #define min(a,b) ((a) < (b) ? (a) : (b)) #ifndef M_PI #define M_PI 3.14159263 #endif #define SQRT_M_PI_2 0.88623 #define LOUDNESS_EXP 2.5 #define NB_BANDS 8 #define SPEEX_PROB_START_DEFAULT 0.35f #define SPEEX_PROB_CONTINUE_DEFAULT 0.20f #define ZMIN .1 #define ZMAX .316 #define ZMIN_1 10 #define LOG_MIN_MAX_1 0.86859 static void conj_window(float *w, int len) { int i; for (i=0;i19) return 1+.1296/x; frac = 2*x-integer; return ((1-frac)*table[ind] + frac*table[ind+1])/sqrt(x+.0001f); } static inline float qcurve(float x) { return 1.f/(1.f+.1f/(x*x)); } SpeexPreprocessState *speex_preprocess_state_init(int frame_size, int sampling_rate) { int i; int N, N3, N4; 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->agc_enabled = 0; st->agc_level = 8000; st->vad_enabled = 0; st->dereverb_enabled = 0; st->reverb_decay = .5; st->reverb_level = .2; st->speech_prob_start = SPEEX_PROB_START_DEFAULT; st->speech_prob_continue = SPEEX_PROB_CONTINUE_DEFAULT; st->frame = (float*)speex_alloc(2*N*sizeof(float)); st->ps = (float*)speex_alloc(N*sizeof(float)); st->gain2 = (float*)speex_alloc(N*sizeof(float)); st->window = (float*)speex_alloc(2*N*sizeof(float)); st->noise = (float*)speex_alloc(N*sizeof(float)); st->reverb_estimate = (float*)speex_alloc(N*sizeof(float)); st->old_ps = (float*)speex_alloc(N*sizeof(float)); st->gain = (float*)speex_alloc(N*sizeof(float)); st->prior = (float*)speex_alloc(N*sizeof(float)); st->post = (float*)speex_alloc(N*sizeof(float)); st->loudness_weight = (float*)speex_alloc(N*sizeof(float)); st->inbuf = (float*)speex_alloc(N3*sizeof(float)); st->outbuf = (float*)speex_alloc(N3*sizeof(float)); st->echo_noise = (float*)speex_alloc(N*sizeof(float)); st->S = (float*)speex_alloc(N*sizeof(float)); st->Smin = (float*)speex_alloc(N*sizeof(float)); st->Stmp = (float*)speex_alloc(N*sizeof(float)); st->update_prob = (float*)speex_alloc(N*sizeof(float)); st->zeta = (float*)speex_alloc(N*sizeof(float)); st->Zpeak = 0; st->Zlast = 0; st->noise_bands = (float*)speex_alloc(NB_BANDS*sizeof(float)); st->noise_bands2 = (float*)speex_alloc(NB_BANDS*sizeof(float)); st->speech_bands = (float*)speex_alloc(NB_BANDS*sizeof(float)); st->speech_bands2 = (float*)speex_alloc(NB_BANDS*sizeof(float)); st->noise_bandsN = st->speech_bandsN = 1; conj_window(st->window, 2*N3); for (i=2*N3;i<2*st->ps_size;i++) st->window[i]=1; 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;inoise[i]=1e4; st->reverb_estimate[i]=0.; st->old_ps[i]=1e4; st->gain[i]=1; st->post[i]=1; st->prior[i]=1; } for (i=0;iinbuf[i]=0; st->outbuf[i]=0; } for (i=0;iloudness_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->speech_prob = 0; st->last_speech = 1000; st->loudness = pow(6000,LOUDNESS_EXP); st->loudness2 = 6000; st->nb_loudness_adapt = 0; st->fft_lookup = (struct drft_lookup*)speex_alloc(sizeof(struct drft_lookup)); spx_drft_init(st->fft_lookup,2*N); st->nb_adapt=0; st->consec_noise=0; st->nb_preprocess=0; return st; } void speex_preprocess_state_destroy(SpeexPreprocessState *st) { speex_free(st->frame); speex_free(st->ps); speex_free(st->gain2); 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); speex_free(st->loudness_weight); speex_free(st->echo_noise); speex_free(st->S); speex_free(st->Smin); speex_free(st->Stmp); speex_free(st->update_prob); speex_free(st->zeta); speex_free(st->noise_bands); speex_free(st->noise_bands2); speex_free(st->speech_bands); speex_free(st->speech_bands2); speex_free(st->inbuf); speex_free(st->outbuf); spx_drft_clear(st->fft_lookup); speex_free(st->fft_lookup); speex_free(st); } static void update_noise(SpeexPreprocessState *st, float *ps, spx_int32_t *echo) { int i; float beta; st->nb_adapt++; beta=1.0f/st->nb_adapt; if (beta < .05f) beta=.05f; if (!echo) { for (i=0;ips_size;i++) st->noise[i] = (1.f-beta)*st->noise[i] + beta*ps[i]; } else { for (i=0;ips_size;i++) st->noise[i] = (1.f-beta)*st->noise[i] + beta*max(1.f,ps[i]-st->frame_size*st->frame_size*1.0*echo[i]); #if 0 for (i=0;ips_size;i++) st->noise[i] = 0; #endif } } static int speex_compute_vad(SpeexPreprocessState *st, float *ps, float mean_prior, float mean_post) { int i, is_speech=0; int N = st->ps_size; float scale=.5f/N; /* FIXME: Clean this up a bit */ { float bands[NB_BANDS]; int j; float p0, p1; float tot_loudness=0; float x = sqrt(mean_post); for (i=5;ips[i] * st->loudness_weight[i]; } for (i=0;ispeech_prob + .01*(1-st->speech_prob); p1 *= .01*st->speech_prob + .99*(1-st->speech_prob); st->speech_prob = p0/(p1+p0); */ if (st->noise_bandsN < 50 || st->speech_bandsN < 50) { if (mean_post > 5.f) { float adapt = 1./st->speech_bandsN++; if (adapt<.005f) adapt = .005f; for (i=0;ispeech_bands[i] = (1.f-adapt)*st->speech_bands[i] + adapt*bands[i]; /*st->speech_bands2[i] = (1-adapt)*st->speech_bands2[i] + adapt*bands[i]*bands[i];*/ st->speech_bands2[i] = (1.f-adapt)*st->speech_bands2[i] + adapt*(bands[i]-st->speech_bands[i])*(bands[i]-st->speech_bands[i]); } } else { float adapt = 1./st->noise_bandsN++; if (adapt<.005f) adapt = .005f; for (i=0;inoise_bands[i] = (1.f-adapt)*st->noise_bands[i] + adapt*bands[i]; /*st->noise_bands2[i] = (1-adapt)*st->noise_bands2[i] + adapt*bands[i]*bands[i];*/ st->noise_bands2[i] = (1.f-adapt)*st->noise_bands2[i] + adapt*(bands[i]-st->noise_bands[i])*(bands[i]-st->noise_bands[i]); } } } p0=p1=1; for (i=0;inoise_bands2[i] - st->noise_bands[i]*st->noise_bands[i]; speech_var = 1.01*st->speech_bands2[i] - st->speech_bands[i]*st->speech_bands[i];*/ noise_var = st->noise_bands2[i]; speech_var = st->speech_bands2[i]; if (noise_var < .1f) noise_var = .1f; if (speech_var < .1f) speech_var = .1f; /*speech_var = sqrt(speech_var*noise_var); noise_var = speech_var;*/ if (noise_var < .05f*speech_var) noise_var = .05f*speech_var; if (speech_var < .05f*noise_var) speech_var = .05f*noise_var; if (bands[i] < st->noise_bands[i]) speech_var = noise_var; if (bands[i] > st->speech_bands[i]) noise_var = speech_var; speech_mean = st->speech_bands[i]; noise_mean = st->noise_bands[i]; if (noise_mean < speech_mean - 5.f) noise_mean = speech_mean - 5.f; tmp1 = exp(-.5f*(bands[i]-speech_mean)*(bands[i]-speech_mean)/speech_var)/sqrt(2.f*M_PI*speech_var); tmp2 = exp(-.5f*(bands[i]-noise_mean)*(bands[i]-noise_mean)/noise_var)/sqrt(2.f*M_PI*noise_var); /*fprintf (stderr, "%f ", (float)(p0/(.01+p0+p1)));*/ /*fprintf (stderr, "%f ", (float)(bands[i]));*/ pr = tmp1/(1e-25+tmp1+tmp2); /*if (bands[i] < st->noise_bands[i]) pr=.01; if (bands[i] > st->speech_bands[i] && pr < .995) pr=.995;*/ if (pr>.999f) pr=.999f; if (pr<.001f) pr=.001f; /*fprintf (stderr, "%f ", pr);*/ p0 *= pr; p1 *= (1-pr); } p0 = pow(p0,.2); p1 = pow(p1,.2); #if 1 p0 *= 2.f; p0=p0/(p1+p0); if (st->last_speech>20) { float tmp = sqrt(tot_loudness)/st->loudness2; tmp = 1.f-exp(-10.f*tmp); if (p0>tmp) p0=tmp; } p1=1-p0; #else if (sqrt(tot_loudness) < .6f*st->loudness2 && p0>15.f*p1) p0=15.f*p1; if (sqrt(tot_loudness) < .45f*st->loudness2 && p0>7.f*p1) p0=7.f*p1; if (sqrt(tot_loudness) < .3f*st->loudness2 && p0>3.f*p1) p0=3.f*p1; if (sqrt(tot_loudness) < .15f*st->loudness2 && p0>p1) p0=p1; /*fprintf (stderr, "%f %f ", (float)(sqrt(tot_loudness) /( .25*st->loudness2)), p0/(p1+p0));*/ #endif p0 *= .99f*st->speech_prob + .01f*(1-st->speech_prob); p1 *= .01f*st->speech_prob + .99f*(1-st->speech_prob); st->speech_prob = p0/(1e-25f+p1+p0); /*fprintf (stderr, "%f %f %f ", tot_loudness, st->loudness2, st->speech_prob);*/ if (st->speech_prob > st->speech_prob_start || (st->last_speech < 20 && st->speech_prob > st->speech_prob_continue)) { is_speech = 1; st->last_speech = 0; } else { st->last_speech++; if (st->last_speech<20) is_speech = 1; } if (st->noise_bandsN > 50 && st->speech_bandsN > 50) { if (mean_post > 5) { float adapt = 1./st->speech_bandsN++; if (adapt<.005f) adapt = .005f; for (i=0;ispeech_bands[i] = (1-adapt)*st->speech_bands[i] + adapt*bands[i]; /*st->speech_bands2[i] = (1-adapt)*st->speech_bands2[i] + adapt*bands[i]*bands[i];*/ st->speech_bands2[i] = (1-adapt)*st->speech_bands2[i] + adapt*(bands[i]-st->speech_bands[i])*(bands[i]-st->speech_bands[i]); } } else { float adapt = 1./st->noise_bandsN++; if (adapt<.005f) adapt = .005f; for (i=0;inoise_bands[i] = (1-adapt)*st->noise_bands[i] + adapt*bands[i]; /*st->noise_bands2[i] = (1-adapt)*st->noise_bands2[i] + adapt*bands[i]*bands[i];*/ st->noise_bands2[i] = (1-adapt)*st->noise_bands2[i] + adapt*(bands[i]-st->noise_bands[i])*(bands[i]-st->noise_bands[i]); } } } } return is_speech; } static void speex_compute_agc(SpeexPreprocessState *st, float mean_prior) { int i; int N = st->ps_size; float scale=.5f/N; float agc_gain; int freq_start, freq_end; float active_bands = 0; freq_start = (int)(300.0f*2*N/st->sampling_rate); freq_end = (int)(2000.0f*2*N/st->sampling_rate); for (i=freq_start;iS[i] > 20.f*st->Smin[i]+1000.f) active_bands+=1; } active_bands /= (freq_end-freq_start+1); if (active_bands > .2f) { float loudness=0.f; float rate, rate2=.2f; st->nb_loudness_adapt++; rate=2.0f/(1+st->nb_loudness_adapt); if (rate < .05f) rate = .05f; if (rate < .1f && pow(loudness, LOUDNESS_EXP) > st->loudness) rate = .1f; if (rate < .2f && pow(loudness, LOUDNESS_EXP) > 3.f*st->loudness) rate = .2f; if (rate < .4f && pow(loudness, LOUDNESS_EXP) > 10.f*st->loudness) rate = .4f; for (i=2;ips[i] * st->gain2[i] * st->gain2[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))*/ st->loudness = (1-rate)*st->loudness + (rate)*pow(loudness, LOUDNESS_EXP); st->loudness2 = (1-rate2)*st->loudness2 + rate2*pow(st->loudness, 1.0f/LOUDNESS_EXP); loudness = pow(st->loudness, 1.0f/LOUDNESS_EXP); /*fprintf (stderr, "%f %f %f\n", loudness, st->loudness2, rate);*/ } agc_gain = st->agc_level/st->loudness2; /*fprintf (stderr, "%f %f %f %f\n", active_bands, st->loudness, st->loudness2, agc_gain);*/ if (agc_gain>200) agc_gain = 200; for (i=0;igain2[i] *= agc_gain; } 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; float *ps=st->ps; /* 'Build' input frame */ for (i=0;iframe[i]=st->inbuf[i]; for (i=0;iframe_size;i++) st->frame[N3+i]=x[i]; /* Update inbuf */ for (i=0;iinbuf[i]=x[N4+i]; /* Windowing */ for (i=0;i<2*N;i++) st->frame[i] *= st->window[i]; /* Perform FFT */ spx_drft_forward(st->fft_lookup, st->frame); /* Power spectrum */ ps[0]=1; for (i=1;iframe[2*i-1]*st->frame[2*i-1] + st->frame[2*i]*st->frame[2*i]; } static void update_noise_prob(SpeexPreprocessState *st) { int i; int N = st->ps_size; for (i=1;iS[i] = 100.f+ .8f*st->S[i] + .05f*st->ps[i-1]+.1f*st->ps[i]+.05f*st->ps[i+1]; if (st->nb_preprocess<1) { for (i=1;iSmin[i] = st->Stmp[i] = st->S[i]+100.f; } if (st->nb_preprocess%200==0) { for (i=1;iSmin[i] = min(st->Stmp[i], st->S[i]); st->Stmp[i] = st->S[i]; } } else { for (i=1;iSmin[i] = min(st->Smin[i], st->S[i]); st->Stmp[i] = min(st->Stmp[i], st->S[i]); } } for (i=1;iupdate_prob[i] *= .2f; if (st->S[i] > 2.5*st->Smin[i]) st->update_prob[i] += .8f; /*fprintf (stderr, "%f ", st->S[i]/st->Smin[i]);*/ /*fprintf (stderr, "%f ", st->update_prob[i]);*/ } } #define NOISE_OVERCOMPENS 1.4 int speex_preprocess(SpeexPreprocessState *st, spx_int16_t *x, spx_int32_t *echo) { int i; int is_speech=1; float mean_post=0; float mean_prior=0; int N = st->ps_size; int N3 = 2*N - st->frame_size; int N4 = st->frame_size - N3; float scale=.5f/N; float *ps=st->ps; float Zframe=0, Pframe; preprocess_analysis(st, x); update_noise_prob(st); st->nb_preprocess++; /* Noise estimation always updated for the 20 first times */ if (st->nb_adapt<10) { update_noise(st, ps, echo); } /* Deal with residual echo if provided */ if (echo) for (i=1;iecho_noise[i] = (.3f*st->echo_noise[i] + st->frame_size*st->frame_size*1.0*echo[i]); /* Compute a posteriori SNR */ for (i=1;inoise[i] + st->echo_noise[i] + st->reverb_estimate[i]; st->post[i] = ps[i]/tot_noise - 1.f; if (st->post[i]>100.f) st->post[i]=100.f; /*if (st->post[i]<0) st->post[i]=0;*/ mean_post+=st->post[i]; } mean_post /= N; if (mean_post<0.f) mean_post=0.f; /* Special case for first frame */ if (st->nb_adapt==1) for (i=1;iold_ps[i] = ps[i]; /* Compute a priori SNR */ { /* A priori update rate */ for (i=1;iprior[i]*st->prior[i]/((1+st->prior[i])*(1+st->prior[i])); float tot_noise = 1.f+ NOISE_OVERCOMPENS*st->noise[i] + st->echo_noise[i] + st->reverb_estimate[i]; /* A priori SNR update */ st->prior[i] = gamma*max(0.0f,st->post[i]) + (1.f-gamma)* (.8*st->gain[i]*st->gain[i]*st->old_ps[i]/tot_noise + .2*st->prior[i]); if (st->prior[i]>100.f) st->prior[i]=100.f; mean_prior+=st->prior[i]; } } mean_prior /= N; #if 0 for (i=0;iprior[i]); } fprintf (stderr, "\n"); #endif /*fprintf (stderr, "%f %f\n", mean_prior,mean_post);*/ if (st->nb_preprocess>=20) { int do_update = 0; float noise_ener=0, sig_ener=0; /* If SNR is low (both a priori and a posteriori), update the noise estimate*/ /*if (mean_prior<.23 && mean_post < .5)*/ if (mean_prior<.23f && mean_post < .5f) do_update = 1; for (i=1;inoise[i]; sig_ener += ps[i]; } if (noise_ener > 3.f*sig_ener) do_update = 1; /*do_update = 0;*/ if (do_update) { st->consec_noise++; } else { st->consec_noise=0; } } if (st->vad_enabled) is_speech = speex_compute_vad(st, ps, mean_prior, mean_post); if (st->consec_noise>=3) { update_noise(st, st->old_ps, echo); } else { for (i=1;iupdate_prob[i]<.5f/* || st->ps[i] < st->noise[i]*/) { if (echo) st->noise[i] = .95f*st->noise[i] + .05f*max(1.0f,st->ps[i]-st->frame_size*st->frame_size*1.0*echo[i]); else st->noise[i] = .95f*st->noise[i] + .05f*st->ps[i]; } } } for (i=1;izeta[i] = .7f*st->zeta[i] + .3f*st->prior[i]; } { int freq_start = (int)(300.0f*2.f*N/st->sampling_rate); int freq_end = (int)(2000.0f*2.f*N/st->sampling_rate); for (i=freq_start;izeta[i]; } Zframe /= (freq_end-freq_start); } st->Zlast = Zframe; Pframe = qcurve(Zframe); /*fprintf (stderr, "%f\n", Pframe);*/ /* Compute gain according to the Ephraim-Malah algorithm */ for (i=1;iprior[i]/(1.0001f+st->prior[i]); theta = (1.f+st->post[i])*prior_ratio; if (i==1 || i==N-1) zeta1 = st->zeta[i]; else zeta1 = .25f*st->zeta[i-1] + .5f*st->zeta[i] + .25f*st->zeta[i+1]; P1 = qcurve (zeta1); /* FIXME: add global prob (P2) */ q = 1-Pframe*P1; q = 1-P1; if (q>.95f) q=.95f; p=1.f/(1.f + (q/(1.f-q))*(1.f+st->prior[i])*exp(-theta)); /*p=1;*/ /* Optimal estimator for loudness domain */ MM = hypergeom_gain(theta); st->gain[i] = prior_ratio * MM; /*Put some (very arbitraty) limit on the gain*/ if (st->gain[i]>2.f) { st->gain[i]=2.f; } 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]; if (st->denoise_enabled) { /*st->gain2[i] = p*p*st->gain[i];*/ st->gain2[i]=(p*sqrt(st->gain[i])+.2*(1-p)) * (p*sqrt(st->gain[i])+.2*(1-p)); /*st->gain2[i] = pow(st->gain[i], p) * pow(.1f,1.f-p);*/ } else { st->gain2[i]=1.f; } } st->gain2[0]=st->gain[0]=0.f; st->gain2[N-1]=st->gain[N-1]=0.f; /* for (i=30;igain[i] = st->gain2[i]*st->gain2[i] + (1-st->gain2[i])*.333*(.6*st->gain2[i-1]+st->gain2[i]+.6*st->gain2[i+1]+.4*st->gain2[i-2]+.4*st->gain2[i+2]); } for (i=30;igain2[i] = st->gain[i]; */ if (st->agc_enabled) speex_compute_agc(st, mean_prior); #if 0 if (!is_speech) { for (i=0;igain2[i] = 0; } #if 0 else { for (i=0;igain2[i] = 1; } #endif #endif /* Apply computed gain */ for (i=1;iframe[2*i-1] *= st->gain2[i]; st->frame[2*i] *= st->gain2[i]; } /* Get rid of the DC and very low frequencies */ st->frame[0]=0; st->frame[1]=0; st->frame[2]=0; /* Nyquist frequency is mostly useless too */ st->frame[2*N-1]=0; /* Inverse FFT with 1/N scaling */ spx_drft_backward(st->fft_lookup, st->frame); for (i=0;i<2*N;i++) st->frame[i] *= scale; { 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; } } for (i=0;i<2*N;i++) st->frame[i] *= st->window[i]; /* Perform overlap and add */ for (i=0;ioutbuf[i] + st->frame[i]; for (i=0;iframe[N3+i]; /* Update outbuf */ for (i=0;ioutbuf[i] = st->frame[st->frame_size+i]; /* Save old power spectrum */ for (i=1;iold_ps[i] = ps[i]; return is_speech; } void speex_preprocess_estimate_update(SpeexPreprocessState *st, spx_int16_t *x, spx_int32_t *echo) { int i; int N = st->ps_size; int N3 = 2*N - st->frame_size; float *ps=st->ps; preprocess_analysis(st, x); update_noise_prob(st); st->nb_preprocess++; for (i=1;iupdate_prob[i]<.5f || st->ps[i] < st->noise[i]) { if (echo) st->noise[i] = .95f*st->noise[i] + .1f*max(1.0f,st->ps[i]-st->frame_size*st->frame_size*1.0*echo[i]); else st->noise[i] = .95f*st->noise[i] + .1f*st->ps[i]; } } for (i=0;ioutbuf[i] = x[st->frame_size-N3+i]*st->window[st->frame_size+i]; /* Save old power spectrum */ for (i=1;iold_ps[i] = ps[i]; for (i=1;ireverb_estimate[i] *= st->reverb_decay; } 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 = (*(int*)ptr); break; case SPEEX_PREPROCESS_GET_DENOISE: (*(int*)ptr) = st->denoise_enabled; break; case SPEEX_PREPROCESS_SET_AGC: st->agc_enabled = (*(int*)ptr); break; case SPEEX_PREPROCESS_GET_AGC: (*(int*)ptr) = st->agc_enabled; break; 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; case SPEEX_PREPROCESS_SET_VAD: st->vad_enabled = (*(int*)ptr); break; case SPEEX_PREPROCESS_GET_VAD: (*(int*)ptr) = st->vad_enabled; break; case SPEEX_PREPROCESS_SET_DEREVERB: st->dereverb_enabled = (*(int*)ptr); for (i=0;ips_size;i++) st->reverb_estimate[i]=0; break; case SPEEX_PREPROCESS_GET_DEREVERB: (*(int*)ptr) = st->dereverb_enabled; break; case SPEEX_PREPROCESS_SET_DEREVERB_LEVEL: st->reverb_level = (*(float*)ptr); break; case SPEEX_PREPROCESS_GET_DEREVERB_LEVEL: (*(float*)ptr) = st->reverb_level; break; case SPEEX_PREPROCESS_SET_DEREVERB_DECAY: st->reverb_decay = (*(float*)ptr); break; case SPEEX_PREPROCESS_GET_DEREVERB_DECAY: (*(float*)ptr) = st->reverb_decay; break; case SPEEX_PREPROCESS_SET_PROB_START: st->speech_prob_start = (*(int*)ptr) / 100.0; if ( st->speech_prob_start > 1 || st->speech_prob_start < 0 ) st->speech_prob_start = SPEEX_PROB_START_DEFAULT; break; case SPEEX_PREPROCESS_GET_PROB_START: (*(int*)ptr) = st->speech_prob_start * 100; break; case SPEEX_PREPROCESS_SET_PROB_CONTINUE: st->speech_prob_continue = (*(int*)ptr) / 100.0; if ( st->speech_prob_continue > 1 || st->speech_prob_continue < 0 ) st->speech_prob_continue = SPEEX_PROB_CONTINUE_DEFAULT; break; case SPEEX_PREPROCESS_GET_PROB_CONTINUE: (*(int*)ptr) = st->speech_prob_continue * 100; break; default: speex_warning_int("Unknown speex_preprocess_ctl request: ", request); return -1; } return 0; }