/*  ---------------------------------------------------------------  *//*      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,                                int mlsa_speed_param); cst_wave *mlsa_resynthesis(const cst_track *params,                            const cst_track *str,                            cst_cg_db *cg_db,                           cst_audio_streaming_info *asi,                           int mlsa_speed_param){    /* 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,mlsa_speed_param);     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,                                int mlsa_speed_param){    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;    if ((num_mcep > mlsa_speed_param) &&        ((num_mcep - mlsa_speed_param) > 4))        /* Basically ignore some of the higher coeffs */        /* It'll sound worse, but it will be faster */        num_mcep -= mlsa_speed_param;    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(void){    /* 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;  register int i;       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); 

Lines 1–500 of 642. Load more