/* Transform a masking curve (power spectrum) into a pole-zero filter */
void curve_to_lpc(VorbisPsy *psy, float *curve, float *awk1, float *awk2, int ord)
{
int i;
float ac[psy->n];
float tmp;
int len = psy->n >> 1;
for (i=0;i<2*len;i++)
ac[i] = 0;
for (i=1;i<len;i++)
ac[2*i-1] = curve[i];
ac[0] = curve[0];
ac[2*len-1] = curve[len-1];
spx_drft_backward(&psy->lookup, ac);
_spx_lpc(awk1, ac, ord);
tmp = 1.;
for (i=0;i<ord;i++)
{
tmp *= .99;
awk1[i] *= tmp;
}
#if 0
for (i=0;i<ord;i++)
awk2[i] = 0;
#else
/* Use the second (awk2) filter to correct the first one */
for (i=0;i<2*len;i++)
ac[i] = 0;
for (i=0;i<ord;i++)
ac[i+1] = awk1[i];
ac[0] = 1;
spx_drft_forward(&psy->lookup, ac);
/* Compute (power) response of awk1 (all zero) */
ac[0] *= ac[0];
for (i=1;i<len;i++)
ac[i] = ac[2*i-1]*ac[2*i-1] + ac[2*i]*ac[2*i];
ac[len] = ac[2*len-1]*ac[2*len-1];
/* Compute correction required */
for (i=0;i<len;i++)
curve[i] = 1. / (1e-6f+curve[i]*ac[i]);
for (i=0;i<2*len;i++)
ac[i] = 0;
for (i=1;i<len;i++)
ac[2*i-1] = curve[i];
ac[0] = curve[0];
ac[2*len-1] = curve[len-1];
spx_drft_backward(&psy->lookup, ac);
_spx_lpc(awk2, ac, ord);
tmp = 1;
for (i=0;i<ord;i++)
{
tmp *= .99;
awk2[i] *= tmp;
}
#endif
}
/** Performs echo cancellation on a frame */
void speex_echo_cancel(SpeexEchoState *st, short *ref, short *echo, short *out, float *Yout)
{
int i,j,m;
int N,M;
float scale;
float ESR;
float SER;
float Sry=0,Srr=0,Syy=0,Sey=0,See=0,Sxx=0;
float leak_estimate;
leak_estimate = .1+(.9/(1+2*st->sum_adapt));
N = st->window_size;
M = st->M;
scale = 1.0f/N;
st->cancel_count++;
/* Copy input data to buffer */
for (i=0; i<st->frame_size; i++)
{
st->x[i] = st->x[i+st->frame_size];
st->x[i+st->frame_size] = echo[i];
st->d[i] = st->d[i+st->frame_size];
st->d[i+st->frame_size] = ref[i];
}
/* Shift memory: this could be optimized eventually*/
for (i=0; i<N*(M-1); i++)
st->X[i]=st->X[i+N];
/* Copy new echo frame */
for (i=0; i<N; i++)
st->X[(M-1)*N+i]=st->x[i];
/* Convert x (echo input) to frequency domain */
spx_drft_forward(st->fft_lookup, &st->X[(M-1)*N]);
/* Compute filter response Y */
for (i=0; i<N; i++)
st->Y[i] = 0;
for (j=0; j<M; j++)
spectral_mul_accum(&st->X[j*N], &st->W[j*N], st->Y, N);
/* Convert Y (filter response) to time domain */
for (i=0; i<N; i++)
st->y[i] = st->Y[i];
spx_drft_backward(st->fft_lookup, st->y);
for (i=0; i<N; i++)
st->y[i] *= scale;
/* Transform d (reference signal) to frequency domain */
for (i=0; i<N; i++)
st->D[i]=st->d[i];
spx_drft_forward(st->fft_lookup, st->D);
/* Compute error signal (signal with echo removed) */
for (i=0; i<st->frame_size; i++)
{
float tmp_out;
tmp_out = (float)ref[i] - st->y[i+st->frame_size];
st->E[i] = 0;
st->E[i+st->frame_size] = tmp_out;
/* Saturation */
if (tmp_out>32767)
tmp_out = 32767;
else if (tmp_out<-32768)
tmp_out = -32768;
out[i] = tmp_out;
}
/* This bit of code provides faster adaptation by doing a projection of the previous gradient on the
"MMSE surface" */
if (1)
{
float Sge, Sgg, Syy;
float gain;
Syy = inner_prod(st->y+st->frame_size, st->y+st->frame_size, st->frame_size);
for (i=0; i<N; i++)
st->Y2[i] = 0;
for (j=0; j<M; j++)
spectral_mul_accum(&st->X[j*N], &st->PHI[j*N], st->Y2, N);
for (i=0; i<N; i++)
st->y2[i] = st->Y2[i];
spx_drft_backward(st->fft_lookup, st->y2);
for (i=0; i<N; i++)
st->y2[i] *= scale;
Sge = inner_prod(st->y2+st->frame_size, st->E+st->frame_size, st->frame_size);
Sgg = inner_prod(st->y2+st->frame_size, st->y2+st->frame_size, st->frame_size);
/* Compute projection gain */
gain = Sge/(N+.03*Syy+Sgg);
if (gain>2)
gain = 2;
if (gain < -2)
gain = -2;
/* Apply gain to weights, echo estimates, output */
for (i=0; i<N; i++)
st->Y[i] += gain*st->Y2[i];
for (i=0; i<st->frame_size; i++)
{
st->y[i+st->frame_size] += gain*st->y2[i+st->frame_size];
st->E[i+st->frame_size] -= gain*st->y2[i+st->frame_size];
}
for (i=0; i<M*N; i++)
st->W[i] += gain*st->PHI[i];
}
/* Compute power spectrum of output (D-Y) and filter response (Y) */
for (i=0; i<N; i++)
st->D[i] -= st->Y[i];
power_spectrum(st->D, st->Rf, N);
power_spectrum(st->Y, st->Yf, N);
/* Compute frequency-domain adaptation mask */
for (j=0; j<=st->frame_size; j++)
{
float r;
r = leak_estimate*st->Yf[j] / (1+st->Rf[j]);
if (r>1)
r = 1;
st->fratio[j] = r;
}
/* Compute a bunch of correlations */
Sry = inner_prod(st->y+st->frame_size, st->d+st->frame_size, st->frame_size);
Sey = inner_prod(st->y+st->frame_size, st->E+st->frame_size, st->frame_size);
See = inner_prod(st->E+st->frame_size, st->E+st->frame_size, st->frame_size);
Syy = inner_prod(st->y+st->frame_size, st->y+st->frame_size, st->frame_size);
Srr = inner_prod(st->d+st->frame_size, st->d+st->frame_size, st->frame_size);
Sxx = inner_prod(st->x+st->frame_size, st->x+st->frame_size, st->frame_size);
/* Compute smoothed cross-correlation and energy */
st->Sey = .98*st->Sey + .02*Sey;
st->Syy = .98*st->Syy + .02*Syy;
st->See = .98*st->See + .02*See;
/* Check if filter is completely mis-adapted (if so, reset filter) */
if (st->Sey/(1+st->Syy + .01*st->See) < -1)
{
/*fprintf (stderr, "reset at %d\n", st->cancel_count);*/
speex_echo_state_reset(st);
return;
}
SER = Srr / (1+Sxx);
ESR = leak_estimate*Syy / (1+See);
if (ESR>1)
ESR = 1;
#if 1
/* If over-cancellation (creating echo with 180 phase) damp filter */
if (st->Sey/(1+st->Syy) < -.1 && (ESR > .3))
{
for (i=0; i<M*N; i++)
st->W[i] *= .95;
st->Sey *= .5;
/*fprintf (stderr, "corrected down\n");*/
}
#endif
#if 1
/* If under-cancellation (leaving echo with 0 phase) scale filter up */
if (st->Sey/(1+st->Syy) > .1 && (ESR > .1 || SER < 10))
{
for (i=0; i<M*N; i++)
st->W[i] *= 1.05;
st->Sey *= .5;
/*fprintf (stderr, "corrected up %d\n", st->cancel_count);*/
}
#endif
/* We consider that the filter is adapted if the following is true*/
if (ESR>.6 && st->sum_adapt > 1)
{
/*if (!st->adapted)
fprintf(stderr, "Adapted at %d %f\n", st->cancel_count, st->sum_adapt);*/
st->adapted = 1;
}
/* Update frequency-dependent energy ratio with the total energy ratio */
for (i=0; i<=st->frame_size; i++)
{
st->fratio[i] = (.2*ESR+.8*min(.005+ESR,st->fratio[i]));
}
if (st->adapted)
{
st->adapt_rate = .95f/(2+M);
} else {
/* Temporary adaption rate if filter is not adapted correctly */
if (SER<.1)
st->adapt_rate =.8/(2+M);
else if (SER<1)
st->adapt_rate =.4/(2+M);
else if (SER<10)
st->adapt_rate =.2/(2+M);
else if (SER<30)
st->adapt_rate =.08/(2+M);
else
st->adapt_rate = 0;
}
/* How much have we adapted so far? */
st->sum_adapt += st->adapt_rate;
/* Compute echo power in each frequency bin */
{
float ss = 1.0f/st->cancel_count;
if (ss < .3/M)
ss=.3/M;
power_spectrum(&st->X[(M-1)*N], st->Xf, N);
/* Smooth echo energy estimate over time */
for (j=0; j<=st->frame_size; j++)
st->power[j] = (1-ss)*st->power[j] + ss*st->Xf[j];
/* Combine adaptation rate to the the inverse energy estimate */
if (st->adapted)
{
/* If filter is adapted, include the frequency-dependent ratio too */
for (i=0; i<=st->frame_size; i++)
st->power_1[i] = st->adapt_rate*st->fratio[i] /(1.f+st->power[i]);
} else {
for (i=0; i<=st->frame_size; i++)
st->power_1[i] = st->adapt_rate/(1.f+st->power[i]);
}
}
/* Convert error to frequency domain */
spx_drft_forward(st->fft_lookup, st->E);
/* Do some regularization (prevents problems when system is ill-conditoned) */
for (m=0; m<M; m++)
for (i=0; i<N; i++)
st->W[m*N+i] *= 1-st->regul[i]*ESR;
/* Compute weight gradient */
for (j=0; j<M; j++)
{
weighted_spectral_mul_conj(st->power_1, &st->X[j*N], st->E, st->PHI+N*j, N);
}
/* Gradient descent */
for (i=0; i<M*N; i++)
st->W[i] += st->PHI[i];
/* AUMDF weight constraint */
for (j=0; j<M; j++)
{
/* Remove the "if" to make this an MDF filter */
if (st->cancel_count%M == j)
{
spx_drft_backward(st->fft_lookup, &st->W[j*N]);
for (i=0; i<N; i++)
st->W[j*N+i]*=scale;
for (i=st->frame_size; i<N; i++)
{
st->W[j*N+i]=0;
}
spx_drft_forward(st->fft_lookup, &st->W[j*N]);
}
}
/* Compute spectrum of estimated echo for use in an echo post-filter (if necessary)*/
if (Yout)
{
if (st->adapted)
{
/* If the filter is adapted, take the filtered echo */
for (i=0; i<st->frame_size; i++)
st->last_y[i] = st->last_y[st->frame_size+i];
for (i=0; i<st->frame_size; i++)
st->last_y[st->frame_size+i] = st->y[st->frame_size+i];
} else {
/* If filter isn't adapted yet, all we can do is take the echo signal directly */
for (i=0; i<N; i++)
st->last_y[i] = st->x[i];
}
/* Apply hanning window (should pre-compute it)*/
for (i=0; i<N; i++)
st->Yps[i] = (.5-.5*cos(2*M_PI*i/N))*st->last_y[i];
/* Compute power spectrum of the echo */
spx_drft_forward(st->fft_lookup, st->Yps);
power_spectrum(st->Yps, st->Yps, N);
/* Estimate residual echo */
for (i=0; i<=st->frame_size; i++)
Yout[i] = 2*leak_estimate*st->Yps[i];
}
}
int speex_preprocess(SpeexPreprocessState *st, spx_int16_t *x, float *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;i<N;i++)
st->echo_noise[i] = (.3f*st->echo_noise[i] + echo[i]);
/* Compute a posteriori SNR */
for (i=1;i<N;i++)
{
st->post[i] = ps[i]/(1.f+NOISE_OVERCOMPENS*st->noise[i]+st->echo_noise[i]+st->reverb_estimate[i]) - 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;i<N;i++)
st->old_ps[i] = ps[i];
/* Compute a priori SNR */
{
/* A priori update rate */
float gamma;
float min_gamma=0.12f;
gamma = 1.0f/st->nb_preprocess;
/*Make update rate smaller when there's no speech*/
#if 0
if (mean_post<3.5 && mean_prior < 1)
min_gamma *= (mean_post+.5);
else
min_gamma *= 4.;
#else
min_gamma = .1f*fabs(mean_prior - mean_post)*fabs(mean_prior - mean_post);
if (min_gamma>.15f)
min_gamma = .15f;
if (min_gamma<.02f)
min_gamma = .02f;
#endif
/*min_gamma = .08;*/
/*if (gamma<min_gamma)*/
gamma=min_gamma;
gamma = .1;
for (i=1;i<N;i++)
{
/* A priori SNR update */
st->prior[i] = gamma*max(0.0f,st->post[i]) +
(1.f-gamma)*st->gain[i]*st->gain[i]*st->old_ps[i]/(1.f+NOISE_OVERCOMPENS*st->noise[i]+st->echo_noise[i]+st->reverb_estimate[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;i<N;i++)
{
fprintf (stderr, "%f ", st->prior[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;i<N;i++)
{
noise_ener += st->noise[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;i<N-1;i++)
{
if (st->update_prob[i]<.5f || st->ps[i] < st->noise[i])
{
if (echo)
st->noise[i] = .90f*st->noise[i] + .1f*max(1.0f,st->ps[i]-echo[i]);
else
st->noise[i] = .90f*st->noise[i] + .1f*st->ps[i];
}
}
}
for (i=1;i<N;i++)
{
st->zeta[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;i<freq_end;i++)
{
Zframe += st->zeta[i];
}
}
Zframe /= N;
if (Zframe<ZMIN)
{
Pframe = 0;
} else {
if (Zframe > 1.5f*st->Zlast)
{
Pframe = 1.f;
st->Zpeak = Zframe;
if (st->Zpeak > 10.f)
st->Zpeak = 10.f;
if (st->Zpeak < 1.f)
st->Zpeak = 1.f;
} else {
if (Zframe < st->Zpeak*ZMIN)
{
Pframe = 0;
} else if (Zframe > st->Zpeak*ZMAX)
{
Pframe = 1;
} else {
Pframe = log(Zframe/(st->Zpeak*ZMIN)) / log(ZMAX/ZMIN);
}
}
}
st->Zlast = Zframe;
/*fprintf (stderr, "%f\n", Pframe);*/
/* Compute gain according to the Ephraim-Malah algorithm */
for (i=1;i<N;i++)
{
float MM;
float theta;
float prior_ratio;
float p, q;
float zeta1;
float P1;
prior_ratio = st->prior[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];
if (zeta1<ZMIN)
P1 = 0.f;
else if (zeta1>ZMAX)
P1 = 1.f;
else
P1 = LOG_MIN_MAX_1 * log(ZMIN_1*zeta1);
/*P1 = log(zeta1/ZMIN)/log(ZMAX/ZMIN);*/
/* 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;*/
#if 0
/* log-spectral magnitude estimator */
if (theta<6)
MM = 0.74082*pow(theta+1,.61)/sqrt(.0001+theta);
else
MM=1;
#else
/* Optimal estimator for loudness domain */
MM = hypergeom_gain(theta);
#endif
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];
} else {
st->gain2[i]=1.f;
}
}
st->gain2[0]=st->gain[0]=0.f;
st->gain2[N-1]=st->gain[N-1]=0.f;
if (st->agc_enabled)
speex_compute_agc(st, mean_prior);
#if 0
if (!is_speech)
{
for (i=0;i<N;i++)
st->gain2[i] = 0;
}
#if 0
else {
for (i=0;i<N;i++)
st->gain2[i] = 1;
}
#endif
#endif
/* Apply computed gain */
for (i=1;i<N;i++)
{
st->frame[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;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];
/* Save old power spectrum */
for (i=1;i<N;i++)
st->old_ps[i] = ps[i];
return is_speech;
}
EXPORT void speex_decorrelate(SpeexDecorrState *st, const spx_int16_t *in, spx_int16_t *out, int strength)
{
int ch;
float amount;
if (strength<0)
strength = 0;
if (strength>100)
strength = 100;
amount = .01*strength;
for (ch=0;ch<st->channels;ch++)
{
int i;
int N=2*st->frame_size;
float beta, beta2;
float *x;
float max_alpha = 0;
float *buff;
float *ring;
int ringID;
int order;
float alpha;
buff = st->buff+ch*2*st->frame_size;
ring = st->ring[ch];
ringID = st->ringID[ch];
order = st->order[ch];
alpha = st->alpha[ch];
for (i=0;i<st->frame_size;i++)
buff[i] = buff[i+st->frame_size];
for (i=0;i<st->frame_size;i++)
buff[i+st->frame_size] = in[i*st->channels+ch];
x = buff+st->frame_size;
beta = 1.-.3*amount*amount;
if (amount>1)
beta = 1-sqrt(.4*amount);
else
beta = 1-0.63246*amount;
if (beta<0)
beta = 0;
beta2 = beta;
for (i=0;i<st->frame_size;i++)
{
st->y[i] = alpha*(x[i-ALLPASS_ORDER+order]-beta*x[i-ALLPASS_ORDER+order-1])*st->vorbis_win[st->frame_size+i+order]
+ x[i-ALLPASS_ORDER]*st->vorbis_win[st->frame_size+i]
- alpha*(ring[ringID]
- beta*ring[ringID+1>=order?0:ringID+1]);
ring[ringID++]=st->y[i];
st->y[i] *= st->vorbis_win[st->frame_size+i];
if (ringID>=order)
ringID=0;
}
order = order+(irand(&st->seed)%3)-1;
if (order < 5)
order = 5;
if (order > 10)
order = 10;
/*order = 5+(irand(&st->seed)%6);*/
max_alpha = pow(.96+.04*(amount-1),order);
if (max_alpha > .98/(1.+beta2))
max_alpha = .98/(1.+beta2);
alpha = alpha + .4*uni_rand(&st->seed);
if (alpha > max_alpha)
alpha = max_alpha;
if (alpha < -max_alpha)
alpha = -max_alpha;
for (i=0;i<ALLPASS_ORDER;i++)
ring[i] = 0;
ringID = 0;
for (i=0;i<st->frame_size;i++)
{
float tmp = alpha*(x[i-ALLPASS_ORDER+order]-beta*x[i-ALLPASS_ORDER+order-1])*st->vorbis_win[i+order]
+ x[i-ALLPASS_ORDER]*st->vorbis_win[i]
- alpha*(ring[ringID]
- beta*ring[ringID+1>=order?0:ringID+1]);
ring[ringID++]=tmp;
tmp *= st->vorbis_win[i];
if (ringID>=order)
ringID=0;
st->y[i] += tmp;
}
#ifdef VORBIS_PSYCHO
float frame[N];
float scale = 1./N;
for (i=0;i<2*st->frame_size;i++)
frame[i] = buff[i];
//float coef = .5*0.78130;
float coef = M_PI*0.075063 * 0.93763 * amount * .8 * 0.707;
compute_curve(st->psy, buff, st->curve);
for (i=1;i<st->frame_size;i++)
{
float x1,x2;
float gain;
do {
x1 = uni_rand(&st->seed);
x2 = uni_rand(&st->seed);
} while (x1*x1+x2*x2 > 1.);
gain = coef*sqrt(.1+st->curve[i]);
frame[2*i-1] = gain*x1;
frame[2*i] = gain*x2;
}
frame[0] = coef*uni_rand(&st->seed)*sqrt(.1+st->curve[0]);
frame[2*st->frame_size-1] = coef*uni_rand(&st->seed)*sqrt(.1+st->curve[st->frame_size-1]);
spx_drft_backward(&st->lookup,frame);
for (i=0;i<2*st->frame_size;i++)
frame[i] *= st->vorbis_win[i];
#endif
for (i=0;i<st->frame_size;i++)
{
#ifdef VORBIS_PSYCHO
float tmp = st->y[i] + frame[i] + st->wola_mem[i];
st->wola_mem[i] = frame[i+st->frame_size];
#else
float tmp = st->y[i];
#endif
if (tmp>32767)
tmp = 32767;
if (tmp < -32767)
tmp = -32767;
out[i*st->channels+ch] = tmp;
}
st->ringID[ch] = ringID;
st->order[ch] = order;
st->alpha[ch] = alpha;
}
}
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;i<N;i++)
st->echo_noise[i] = (.3f*st->echo_noise[i] + st->frame_size*st->frame_size*1.0*echo[i]);
/* Compute a posteriori SNR */
for (i=1;i<N;i++)
{
float tot_noise = 1.f+ NOISE_OVERCOMPENS*st->noise[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;i<N;i++)
st->old_ps[i] = ps[i];
/* Compute a priori SNR */
{
/* A priori update rate */
for (i=1;i<N;i++)
{
float gamma = .15+.85*st->prior[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;i<N;i++)
{
fprintf (stderr, "%f ", st->prior[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;i<N;i++)
{
noise_ener += st->noise[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;i<N-1;i++)
{
if (st->update_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;i<N;i++)
{
st->zeta[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;i<freq_end;i++)
{
Zframe += st->zeta[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;i<N;i++)
{
float MM;
float theta;
float prior_ratio;
float p, q;
float zeta1;
float P1;
prior_ratio = st->prior[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;i<N-2;i++)
{
st->gain[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;i<N-2;i++)
st->gain2[i] = st->gain[i];
*/
if (st->agc_enabled)
speex_compute_agc(st, mean_prior);
#if 0
if (!is_speech)
{
for (i=0;i<N;i++)
st->gain2[i] = 0;
}
#if 0
else {
for (i=0;i<N;i++)
st->gain2[i] = 1;
}
#endif
#endif
/* Apply computed gain */
for (i=1;i<N;i++)
{
st->frame[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;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];
/* Save old power spectrum */
for (i=1;i<N;i++)
st->old_ps[i] = ps[i];
return is_speech;
}
