Google Git
Sign in
fuchsia / third_party / flite / 62a4aaf0888fbe8b3d746997236b4a42bb774674 / . / src / cg / cst_mlsa.c
blob: cc491cff7ea378259859fdf5f61d957e30a72f7f [file] [log] [blame]
/* --------------------------------------------------------------- */
/* The HMM-Based Speech Synthesis System (HTS): version 1.1b */
/* HTS Working Group */
/* */
/* Department of Computer Science */
/* Nagoya Institute of Technology */
/* and */
/* Interdisciplinary Graduate School of Science and Engineering */
/* Tokyo Institute of Technology */
/* Copyright (c) 2001-2003 */
/* All Rights Reserved. */
/* */
/* Permission is hereby granted, free of charge, to use and */
/* distribute this software and its documentation without */
/* restriction, including without limitation the rights to use, */
/* copy, modify, merge, publish, distribute, sublicense, and/or */
/* sell copies of this work, and to permit persons to whom this */
/* work is furnished to do so, subject to the following conditions: */
/* */
/* 1. The code must retain the above copyright notice, this list */
/* of conditions and the following disclaimer. */
/* */
/* 2. Any modifications must be clearly marked as such. */
/* */
/* NAGOYA INSTITUTE OF TECHNOLOGY, TOKYO INSITITUTE OF TECHNOLOGY, */
/* HTS WORKING GROUP, AND THE CONTRIBUTORS TO THIS WORK DISCLAIM */
/* ALL WARRANTIES WITH REGARD TO THIS SOFTWARE, INCLUDING ALL */
/* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS, IN NO EVENT */
/* SHALL NAGOYA INSTITUTE OF TECHNOLOGY, TOKYO INSITITUTE OF */
/* TECHNOLOGY, HTS WORKING GROUP, NOR THE CONTRIBUTORS BE LIABLE */
/* FOR ANY SPECIAL, INDIRECT OR CONSEQUENTIAL DAMAGES OR ANY */
/* DAMAGES WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, */
/* WHETHER IN AN ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTUOUS */
/* ACTION, ARISING OUT OF OR IN CONNECTION WITH THE USE OR */
/* PERFORMANCE OF THIS SOFTWARE. */
/* */
/* --------------------------------------------------------------- */
/* This is Zen's MLSA filter as ported by Toda to festvox vc */
/* and back ported into hts/festival so we can do MLSA filtering */
/* If I took more time I could probably make this use the same as */
/* as the other code in this directory -- awb@cs.cmu.edu 03JAN06 */
/* --------------------------------------------------------------- */
/* and then ported into Flite (November 2007 awb@cs.cmu.edu) */
/* with some speed uptimizations */
/*********************************************************************/
/* */
/* Mel-cepstral vocoder (pulse/noise excitation & MLSA filter) */
/* 2003/12/26 by Heiga Zen */
/* */
/* Extracted from HTS and slightly modified */
/* by Tomoki Toda (tomoki@ics.nitech.ac.jp) */
/* June 2004 */
/* Integrate as a Voice Conversion module */
/* */
/*-------------------------------------------------------------------*/
#include "cst_alloc.h"
#include "cst_string.h"
#include "cst_math.h"
#include "cst_track.h"
#include "cst_wave.h"
#include "cst_audio.h"
#ifdef ANDROID
#define SPEED_HACK
#endif
#ifdef UNDER_CE
#define SPEED_HACK
/* This is one of those other things you shouldn't do, but it makes
CG voices in flowm fast enough on my phone */
#define double float
#endif
#include "cst_vc.h"
#include "cst_cg.h"
#include "cst_mlsa.h"
static cst_wave *synthesis_body(const cst_track *params,
const cst_track *str,
double fs, double framem,
cst_cg_db *cg_db,
cst_audio_streaming_info *asi);
cst_wave *mlsa_resynthesis(const cst_track *params,
const cst_track *str,
cst_cg_db *cg_db,
cst_audio_streaming_info *asi)
{
/* Resynthesizes a wave from given track */
cst_wave *wave = 0;
int sr = cg_db->sample_rate;
double shift;
if (params->num_frames > 1)
shift = 1000.0*(params->times[1]-params->times[0]);
else
shift = 5.0;
wave = synthesis_body(params,str,sr,shift,cg_db,asi);
return wave;
}
static cst_wave *synthesis_body(const cst_track *params, /* f0 + mcep */
const cst_track *str,
double fs, /* sampling frequency (Hz) */
double framem, /* frame size */
cst_cg_db *cg_db,
cst_audio_streaming_info *asi)
{
long t, pos;
int framel, i;
double f0;
VocoderSetup vs;
cst_wave *wave = 0;
double *mcep;
int stream_mark;
int rc = CST_AUDIO_STREAM_CONT;
int num_mcep;
double ffs = fs;
num_mcep = params->num_channels-1;
/* For SPEED_HACK we could reduce num_mcep, and it will run faster */
/* num_mcep -= 10; */
framel = (int)(0.5 + (framem * ffs / 1000.0)); /* 80 for 16KHz */
init_vocoder(ffs, framel, num_mcep, &vs, cg_db);
if (str != NULL)
vs.gauss = MFALSE;
/* synthesize waveforms by MLSA filter */
wave = new_wave();
cst_wave_resize(wave,params->num_frames * framel,1);
wave->sample_rate = fs;
mcep = cst_alloc(double,num_mcep+1);
for (t = 0, stream_mark = pos = 0;
(rc == CST_AUDIO_STREAM_CONT) && (t < params->num_frames);
t++)
{
f0 = (double)params->frames[t][0];
for (i=1; i<num_mcep+1; i++)
mcep[i-1] = params->frames[t][i];
mcep[i-1] = 0;
if (str)
vocoder(f0, mcep, str->frames[t], num_mcep, cg_db, &vs, wave, &pos);
else
vocoder(f0, mcep, NULL, num_mcep, cg_db, &vs, wave, &pos);
if (asi && (pos-stream_mark > asi->min_buffsize))
{
rc=(*asi->asc)(wave,stream_mark,pos-stream_mark,0,asi);
stream_mark = pos;
}
}
wave->num_samples = pos;
if (asi && (rc == CST_AUDIO_STREAM_CONT))
{ /* drain the last part of the waveform */
(*asi->asc)(wave,stream_mark,pos-stream_mark,1,asi);
}
/* memory free */
cst_free(mcep);
free_vocoder(&vs);
if (rc == CST_AUDIO_STREAM_STOP)
{
delete_wave(wave);
return NULL;
}
else
return wave;
}
static void init_vocoder(double fs, int framel, int m,
VocoderSetup *vs, cst_cg_db *cg_db)
{
/* initialize global parameter */
vs->fprd = framel;
vs->iprd = 1;
vs->seed = 1;
#ifdef SPEED_HACK
/* This makes it about 25% faster and sounds basically the same */
vs->pd = 4;
#else
vs->pd = 5;
#endif
vs->next =1;
vs->gauss = MTRUE;
/* Pade' approximants */
vs->pade[ 0]=1.0;
vs->pade[ 1]=1.0; vs->pade[ 2]=0.0;
vs->pade[ 3]=1.0; vs->pade[ 4]=0.0; vs->pade[ 5]=0.0;
vs->pade[ 6]=1.0; vs->pade[ 7]=0.0; vs->pade[ 8]=0.0; vs->pade[ 9]=0.0;
vs->pade[10]=1.0; vs->pade[11]=0.4999273; vs->pade[12]=0.1067005; vs->pade[13]=0.01170221; vs->pade[14]=0.0005656279;
vs->pade[15]=1.0; vs->pade[16]=0.4999391; vs->pade[17]=0.1107098; vs->pade[18]=0.01369984; vs->pade[19]=0.0009564853;
vs->pade[20]=0.00003041721;
vs->rate = fs;
vs->c = cst_alloc(double,3 * (m + 1) + 3 * (vs->pd + 1) + vs->pd * (m + 2));
vs->p1 = -1;
vs->sw = 0;
vs->x = 0x55555555;
/* for postfiltering */
vs->mc = NULL;
vs->o = 0;
vs->d = NULL;
vs->irleng= 64;
// for MIXED EXCITATION
vs->ME_order = cg_db->ME_order;
vs->ME_num = cg_db->ME_num;
vs->hpulse = cst_alloc(double,vs->ME_order);
vs->hnoise = cst_alloc(double,vs->ME_order);
vs->xpulsesig = cst_alloc(double,vs->ME_order);
vs->xnoisesig = cst_alloc(double,vs->ME_order);
vs->h = cg_db->me_h;
return;
}
static double plus_or_minus_one()
{
/* Randomly return 1 or -1 */
/* not sure rand() is portable */
if (rand() > RAND_MAX/2.0)
return 1.0;
else
return -1.0;
}
static void vocoder(double p, double *mc,
const float *str,
int m, cst_cg_db *cg_db,
VocoderSetup *vs, cst_wave *wav, long *pos)
{
double inc, x, e1, e2;
int i, j, k;
double xpulse, xnoise;
double fxpulse, fxnoise;
float gain=1.0;
if (cg_db->gain != 0.0)
gain = cg_db->gain;
if (str != NULL) /* MIXED-EXCITATION */
{
/* Copy in str's and build hpulse and hnoise for this frame */
for (i=0; i<vs->ME_order; i++)
{
vs->hpulse[i] = vs->hnoise[i] = 0.0;
for (j=0; j<vs->ME_num; j++)
{
vs->hpulse[i] += str[j] * vs->h[j][i];
vs->hnoise[i] += (1 - str[j]) * vs->h[j][i];
}
}
}
if (p != 0.0)
p = vs->rate / p; /* f0 -> pitch */
if (vs->p1 < 0) {
if (vs->gauss & (vs->seed != 1))
vs->next = srnd((unsigned)vs->seed);
vs->p1 = p;
vs->pc = vs->p1;
vs->cc = vs->c + m + 1;
vs->cinc = vs->cc + m + 1;
vs->d1 = vs->cinc + m + 1;
mc2b(mc, vs->c, m, cg_db->mlsa_alpha);
if (cg_db->mlsa_beta > 0.0 && m > 1) {
e1 = b2en(vs->c, m, cg_db->mlsa_alpha, vs);
vs->c[1] -= cg_db->mlsa_beta * cg_db->mlsa_alpha * mc[2];
for (k=2;k<=m;k++)
vs->c[k] *= (1.0 + cg_db->mlsa_beta);
e2 = b2en(vs->c, m, cg_db->mlsa_alpha, vs);
vs->c[0] += log(e1/e2)/2;
}
return;
}
mc2b(mc, vs->cc, m, cg_db->mlsa_alpha);
if (cg_db->mlsa_beta>0.0 && m > 1) {
e1 = b2en(vs->cc, m, cg_db->mlsa_alpha, vs);
vs->cc[1] -= cg_db->mlsa_beta * cg_db->mlsa_alpha * mc[2];
for (k = 2; k <= m; k++)
vs->cc[k] *= (1.0 + cg_db->mlsa_beta);
e2 = b2en(vs->cc, m, cg_db->mlsa_alpha, vs);
vs->cc[0] += log(e1 / e2) / 2.0;
}
for (k=0; k<=m; k++)
vs->cinc[k] = (vs->cc[k] - vs->c[k]) *
(double)vs->iprd / (double)vs->fprd;
if (vs->p1!=0.0 && p!=0.0) {
inc = (p - vs->p1) * (double)vs->iprd / (double)vs->fprd;
} else {
inc = 0.0;
vs->pc = p;
vs->p1 = 0.0;
}
for (j = vs->fprd, i = (vs->iprd + 1) / 2; j--;) {
if (vs->p1 == 0.0) {
if (vs->gauss)
x = (double) nrandom(vs);
else
x = plus_or_minus_one();
if (str != NULL) /* MIXED EXCITATION */
{
xnoise = x;
xpulse = 0.0;
}
} else {
if ((vs->pc += 1.0) >= vs->p1) {
x = sqrt (vs->p1);
vs->pc = vs->pc - vs->p1;
} else
x = 0.0;
if (str != NULL) /* MIXED EXCITATION */
{
xpulse = x;
xnoise = plus_or_minus_one();
}
}
/* MIXED EXCITATION */
/* The real work -- apply shaping filters to pulse and noise */
if (str != NULL)
{
fxpulse = fxnoise = 0.0;
for (k=vs->ME_order-1; k>0; k--)
{
fxpulse += vs->hpulse[k] * vs->xpulsesig[k];
fxnoise += vs->hnoise[k] * vs->xnoisesig[k];
vs->xpulsesig[k] = vs->xpulsesig[k-1];
vs->xnoisesig[k] = vs->xnoisesig[k-1];
}
fxpulse += vs->hpulse[0] * xpulse;
fxnoise += vs->hnoise[0] * xnoise;
vs->xpulsesig[0] = xpulse;
vs->xnoisesig[0] = xnoise;
x = fxpulse + fxnoise; /* excitation is pulse plus noise */
}
if (cg_db->sample_rate == 8000)
/* 8KHz voices are too quiet: this is probably not general */
x *= exp(vs->c[0])*2.0;
else
x *= exp(vs->c[0])*gain;
x = mlsadf(x, vs->c, m, cg_db->mlsa_alpha, vs->pd, vs->d1, vs);
wav->samples[*pos] = (short)x;
*pos += 1;
if (!--i) {
vs->p1 += inc;
for (k = 0; k <= m; k++) vs->c[k] += vs->cinc[k];
i = vs->iprd;
}
}
vs->p1 = p;
memmove(vs->c,vs->cc,sizeof(double)*(m+1));
return;
}
static double mlsadf(double x, double *b, int m, double a, int pd, double *d, VocoderSetup *vs)
{
vs->ppade = &(vs->pade[pd*(pd+1)/2]);
x = mlsadf1 (x, b, m, a, pd, d, vs);
x = mlsadf2 (x, b, m, a, pd, &d[2*(pd+1)], vs);
return(x);
}
static double mlsadf1(double x, double *b, int m, double a, int pd, double *d, VocoderSetup *vs)
{
double v, out = 0.0, *pt, aa;
register int i;
aa = 1 - a*a;
pt = &d[pd+1];
for (i=pd; i>=1; i--) {
d[i] = aa*pt[i-1] + a*d[i];
pt[i] = d[i] * b[1];
v = pt[i] * vs->ppade[i];
x += (1 & i) ? v : -v;
out += v;
}
pt[0] = x;
out += x;
return(out);
}
static double mlsadf2 (double x, double *b, int m, double a, int pd, double *d, VocoderSetup *vs)
{
double v, out = 0.0, *pt;
// double aa;
register int i;
//aa = 1 - a*a;
pt = &d[pd * (m+2)];
for (i=pd; i>=1; i--) {
pt[i] = mlsafir (pt[i-1], b, m, a, &d[(i-1)*(m+2)]);
v = pt[i] * vs->ppade[i];
x += (1&i) ? v : -v;
out += v;
}
pt[0] = x;
out += x;
return(out);
}
static double mlsafir (double x, double *b, int m, double a, double *d)
{
double y = 0.0;
double aa;
register int i;
aa = 1 - a*a;
d[0] = x;
d[1] = aa*d[0] + a*d[1];
for (i=2; i<= m; i++) {
d[i] = d[i] + a*(d[i+1]-d[i-1]);
y += d[i]*b[i];
}
for (i=m+1; i>1; i--)
d[i] = d[i-1];
return(y);
}
static double nrandom (VocoderSetup *vs)
{
if (vs->sw == 0) {
vs->sw = 1;
do {
vs->r1 = 2.0 * rnd(&vs->next) - 1.0;
vs->r2 = 2.0 * rnd(&vs->next) - 1.0;
vs->s = vs->r1 * vs->r1 + vs->r2 * vs->r2;
} while (vs->s > 1 || vs->s == 0);
vs->s = sqrt (-2 * log(vs->s) / vs->s);
return(vs->r1*vs->s);
}
else {
vs->sw = 0;
return (vs->r2*vs->s);
}
}
static double rnd (unsigned long *next)
{
double r;
*next = *next * 1103515245L + 12345;
r = (*next / 65536L) % 32768L;
return(r/RANDMAX);
}
static unsigned long srnd ( unsigned long seed )
{
return(seed);
}
/* mc2b : transform mel-cepstrum to MLSA digital fillter coefficients */
static void mc2b (double *mc, double *b, int m, double a)
{
b[m] = mc[m];
for (m--; m>=0; m--)
b[m] = mc[m] - a * b[m+1];
return;
}
static double b2en (double *b, int m, double a, VocoderSetup *vs)
{
double en;
int k;
if (vs->o<m) {
if (vs->mc != NULL)
cst_free(vs->mc);
vs->mc = cst_alloc(double,(m + 1) + 2 * vs->irleng);
vs->cep = vs->mc + m+1;
vs->ir = vs->cep + vs->irleng;
}
b2mc(b, vs->mc, m, a);
freqt(vs->mc, m, vs->cep, vs->irleng-1, -a, vs);
c2ir(vs->cep, vs->irleng, vs->ir, vs->irleng);
en = 0.0;
for (k=0;k<vs->irleng;k++)
en += vs->ir[k] * vs->ir[k];
return(en);
}
/* b2bc : transform MLSA digital filter coefficients to mel-cepstrum */
static void b2mc (double *b, double *mc, int m, double a)
{
double d, o;
d = mc[m] = b[m];
for (m--; m>=0; m--) {
o = b[m] + a * d;
d = b[m];
mc[m] = o;
}
return;
}
/* freqt : frequency transformation */
static void freqt (double *c1, int m1, double *c2, int m2, double a, VocoderSetup *vs)
{
register int i, j;
double b;
if (vs->d==NULL) {
vs->size = m2;
vs->d = cst_alloc(double,vs->size + vs->size + 2);
vs->g = vs->d+vs->size+1;
}
if (m2>vs->size) {
cst_free(vs->d);
vs->size = m2;
vs->d = cst_alloc(double,vs->size + vs->size + 2);
vs->g = vs->d+vs->size+1;
}
b = 1-a*a;
for (i=0; i<m2+1; i++)
vs->g[i] = 0.0;
for (i=-m1; i<=0; i++) {
if (0 <= m2)
vs->g[0] = c1[-i]+a*(vs->d[0]=vs->g[0]);
if (1 <= m2)
vs->g[1] = b*vs->d[0]+a*(vs->d[1]=vs->g[1]);
for (j=2; j<=m2; j++)
vs->g[j] = vs->d[j-1]+a*((vs->d[j]=vs->g[j])-vs->g[j-1]);
}
memmove(c2,vs->g,sizeof(double)*(m2+1));
return;
}
/* c2ir : The minimum phase impulse response is evaluated from the minimum phase cepstrum */
static void c2ir (double *c, int nc, double *h, int leng)
{
register int n, k, upl;
double d;
h[0] = exp(c[0]);
for (n=1; n<leng; n++) {
d = 0;
upl = (n>=nc) ? nc-1 : n;
for (k=1; k<=upl; k++)
d += k*c[k]*h[n-k];
h[n] = d/n;
}
return;
}
static void free_vocoder(VocoderSetup *vs)
{
cst_free(vs->c);
cst_free(vs->mc);
cst_free(vs->d);
vs->c = NULL;
vs->mc = NULL;
vs->d = NULL;
vs->ppade = NULL;
vs->cc = NULL;
vs->cinc = NULL;
vs->d1 = NULL;
vs->g = NULL;
vs->cep = NULL;
vs->ir = NULL;
cst_free(vs->hpulse);
cst_free(vs->hnoise);
cst_free(vs->xpulsesig);
cst_free(vs->xnoisesig);
return;
}