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espeak-ng/src/klatt.cpp
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jonsd 9d17fb5482 [1.47.11a] 
Fix: Flush stderr() after espeak_CompileDictionary().
Fix: parallel formant parameters for Klatt synthesizer.
Changes for Malayalam numbers.
Language changes: en-us,  nl, pt, sl, ta, ur.


git-svn-id: https://espeak.svn.sourceforge.net/svnroot/espeak/trunk@359 d46cf337-b52f-0410-862d-fd96e6ae7743
2013-05-18 11:25:44 +00:00

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/***************************************************************************
* Copyright (C) 2008 by Jonathan Duddington *
* email: jonsd@users.sourceforge.net *
* *
* Based on a re-implementation by: *
* (c) 1993,94 Jon Iles and Nick Ing-Simmons *
* of the Klatt cascade-parallel formant synthesizer *
* *
* This program is free software; you can redistribute it and/or modify *
* it under the terms of the GNU General Public License as published by *
* the Free Software Foundation; either version 3 of the License, or *
* (at your option) any later version. *
* *
* This program is distributed in the hope that it will be useful, *
* but WITHOUT ANY WARRANTY; without even the implied warranty of *
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the *
* GNU General Public License for more details. *
* *
* You should have received a copy of the GNU General Public License *
* along with this program; if not, see: *
* <http://www.gnu.org/licenses/>. *
***************************************************************************/
// See URL: ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/synthesis/klatt.3.04.tar.gz
#include "StdAfx.h"
#include <stdio.h>
#include <stdlib.h>
#include <math.h>
#include <string.h>
#include "speak_lib.h"
#include "speech.h"
#include "klatt.h"
#include "phoneme.h"
#include "synthesize.h"
#include "voice.h"
#ifdef INCLUDE_KLATT // conditional compilation for the whole file
extern unsigned char *out_ptr; // **JSD
extern unsigned char *out_start;
extern unsigned char *out_end;
extern WGEN_DATA wdata;
static int nsamples;
static int sample_count;
#ifdef _MSC_VER
#define getrandom(min,max) ((rand()%(int)(((max)+1)-(min)))+(min))
#else
#define getrandom(min,max) ((rand()%(long)(((max)+1)-(min)))+(min))
#endif
/* function prototypes for functions private to this file */
static void flutter(klatt_frame_ptr);
static double sampled_source (int);
static double impulsive_source (void);
static double natural_source (void);
static void pitch_synch_par_reset (klatt_frame_ptr);
static double gen_noise (double);
static double DBtoLIN (long);
static void frame_init (klatt_frame_ptr);
static void setabc (long,long,resonator_ptr);
static void setzeroabc (long,long,resonator_ptr);
static klatt_frame_t kt_frame;
static klatt_global_t kt_globals;
#define NUMBER_OF_SAMPLES 100
static int scale_wav_tab[] = {45,38,45,45,55}; // scale output from different voicing sources
// For testing, this can be overwritten in KlattInit()
static short natural_samples2[256]= {
2583, 2516, 2450, 2384, 2319, 2254, 2191, 2127,
2067, 2005, 1946, 1890, 1832, 1779, 1726, 1675,
1626, 1579, 1533, 1491, 1449, 1409, 1372, 1336,
1302, 1271, 1239, 1211, 1184, 1158, 1134, 1111,
1089, 1069, 1049, 1031, 1013, 996, 980, 965,
950, 936, 921, 909, 895, 881, 869, 855,
843, 830, 818, 804, 792, 779, 766, 754,
740, 728, 715, 702, 689, 676, 663, 651,
637, 626, 612, 601, 588, 576, 564, 552,
540, 530, 517, 507, 496, 485, 475, 464,
454, 443, 434, 424, 414, 404, 394, 385,
375, 366, 355, 347, 336, 328, 317, 308,
299, 288, 280, 269, 260, 250, 240, 231,
220, 212, 200, 192, 181, 172, 161, 152,
142, 133, 123, 113, 105, 94, 86, 76,
67, 57, 49, 39, 30, 22, 11, 4,
-5, -14, -23, -32, -41, -50, -60, -69,
-78, -87, -96, -107, -115, -126, -134, -144,
-154, -164, -174, -183, -193, -203, -213, -222,
-233, -242, -252, -262, -271, -281, -291, -301,
-310, -320, -330, -339, -349, -357, -368, -377,
-387, -397, -406, -417, -426, -436, -446, -456,
-467, -477, -487, -499, -509, -521, -532, -543,
-555, -567, -579, -591, -603, -616, -628, -641,
-653, -666, -679, -692, -705, -717, -732, -743,
-758, -769, -783, -795, -808, -820, -834, -845,
-860, -872, -885, -898, -911, -926, -939, -955,
-968, -986, -999, -1018, -1034, -1054, -1072, -1094,
-1115, -1138, -1162, -1188, -1215, -1244, -1274, -1307,
-1340, -1377, -1415, -1453, -1496, -1538, -1584, -1631,
-1680, -1732, -1783, -1839, -1894, -1952, -2010, -2072,
-2133, -2196, -2260, -2325, -2390, -2456, -2522, -2589,
};
static short natural_samples[100]=
{
-310,-400,530,356,224,89,23,-10,-58,-16,461,599,536,701,770,
605,497,461,560,404,110,224,131,104,-97,155,278,-154,-1165,
-598,737,125,-592,41,11,-247,-10,65,92,80,-304,71,167,-1,122,
233,161,-43,278,479,485,407,266,650,134,80,236,68,260,269,179,
53,140,275,293,296,104,257,152,311,182,263,245,125,314,140,44,
203,230,-235,-286,23,107,92,-91,38,464,443,176,98,-784,-2449,
-1891,-1045,-1600,-1462,-1384,-1261,-949,-730
};
/*
function RESONATOR
This is a generic resonator function. Internal memory for the resonator
is stored in the globals structure.
*/
static double resonator(resonator_ptr r, double input)
{
double x;
x = (double) ((double)r->a * (double)input + (double)r->b * (double)r->p1 + (double)r->c * (double)r->p2);
r->p2 = (double)r->p1;
r->p1 = (double)x;
return (double)x;
}
static double resonator2(resonator_ptr r, double input)
{
double x;
x = (double) ((double)r->a * (double)input + (double)r->b * (double)r->p1 + (double)r->c * (double)r->p2);
r->p2 = (double)r->p1;
r->p1 = (double)x;
r->a += r->a_inc;
r->b += r->b_inc;
r->c += r->c_inc;
return (double)x;
}
/*
function ANTIRESONATOR
This is a generic anti-resonator function. The code is the same as resonator
except that a,b,c need to be set with setzeroabc() and we save inputs in
p1/p2 rather than outputs. There is currently only one of these - "rnz"
Output = (rnz.a * input) + (rnz.b * oldin1) + (rnz.c * oldin2)
*/
#ifdef deleted
static double antiresonator(resonator_ptr r, double input)
{
register double x = (double)r->a * (double)input + (double)r->b * (double)r->p1 + (double)r->c * (double)r->p2;
r->p2 = (double)r->p1;
r->p1 = (double)input;
return (double)x;
}
#endif
static double antiresonator2(resonator_ptr r, double input)
{
register double x = (double)r->a * (double)input + (double)r->b * (double)r->p1 + (double)r->c * (double)r->p2;
r->p2 = (double)r->p1;
r->p1 = (double)input;
r->a += r->a_inc;
r->b += r->b_inc;
r->c += r->c_inc;
return (double)x;
}
/*
function FLUTTER
This function adds F0 flutter, as specified in:
"Analysis, synthesis and perception of voice quality variations among
female and male talkers" D.H. Klatt and L.C. Klatt JASA 87(2) February 1990.
Flutter is added by applying a quasi-random element constructed from three
slowly varying sine waves.
*/
static void flutter(klatt_frame_ptr frame)
{
static int time_count;
double delta_f0;
double fla,flb,flc,fld,fle;
fla = (double) kt_globals.f0_flutter / 50;
flb = (double) kt_globals.original_f0 / 100;
// flc = sin(2*PI*12.7*time_count);
// fld = sin(2*PI*7.1*time_count);
// fle = sin(2*PI*4.7*time_count);
flc = sin(PI*12.7*time_count); // because we are calling flutter() more frequently, every 2.9mS
fld = sin(PI*7.1*time_count);
fle = sin(PI*4.7*time_count);
delta_f0 = fla * flb * (flc + fld + fle) * 10;
frame->F0hz10 = frame->F0hz10 + (long) delta_f0;
time_count++;
}
/*
function SAMPLED_SOURCE
Allows the use of a glottal excitation waveform sampled from a real
voice.
*/
static double sampled_source(int source_num)
{
int itemp;
double ftemp;
double result;
double diff_value;
int current_value;
int next_value;
double temp_diff;
short *samples;
if(source_num == 0)
{
samples = natural_samples;
kt_globals.num_samples = 100;
}
else
{
samples = natural_samples2;
kt_globals.num_samples = 256;
}
if(kt_globals.T0!=0)
{
ftemp = (double) kt_globals.nper;
ftemp = ftemp / kt_globals.T0;
ftemp = ftemp * kt_globals.num_samples;
itemp = (int) ftemp;
temp_diff = ftemp - (double) itemp;
current_value = samples[itemp];
next_value = samples[itemp+1];
diff_value = (double) next_value - (double) current_value;
diff_value = diff_value * temp_diff;
result = samples[itemp] + diff_value;
result = result * kt_globals.sample_factor;
}
else
{
result = 0;
}
return(result);
}
/*
function PARWAVE
Converts synthesis parameters to a waveform.
*/
static int parwave(klatt_frame_ptr frame)
{
double temp;
int value;
double outbypas;
double out;
long n4;
double frics;
double glotout;
double aspiration;
double casc_next_in;
double par_glotout;
static double noise;
static double voice;
static double vlast;
static double glotlast;
static double sourc;
int ix;
flutter(frame); /* add f0 flutter */
#ifdef LOG_FRAMES
if(option_log_frames)
{
FILE *f;
f=fopen("log-klatt","a");
fprintf(f,"%4dhz %2dAV %4d %3d, %4d %3d, %4d %3d, %4d %3d, %4d, %3d, FNZ=%3d TLT=%2d\n",frame->F0hz10,frame->AVdb,
frame->Fhz[1],frame->Bhz[1],frame->Fhz[2],frame->Bhz[2],frame->Fhz[3],frame->Bhz[3],frame->Fhz[4],frame->Bhz[4],frame->Fhz[5],frame->Bhz[5],frame->Fhz[0],frame->TLTdb);
fclose(f);
}
#endif
/* MAIN LOOP, for each output sample of current frame: */
for (kt_globals.ns=0; kt_globals.ns<kt_globals.nspfr; kt_globals.ns++)
{
/* Get low-passed random number for aspiration and frication noise */
noise = gen_noise(noise);
/*
Amplitude modulate noise (reduce noise amplitude during
second half of glottal period) if voicing simultaneously present.
*/
if (kt_globals.nper > kt_globals.nmod)
{
noise *= (double) 0.5;
}
/* Compute frication noise */
frics = kt_globals.amp_frica * noise;
/*
Compute voicing waveform. Run glottal source simulation at 4
times normal sample rate to minimize quantization noise in
period of female voice.
*/
for (n4=0; n4<4; n4++)
{
switch(kt_globals.glsource)
{
case IMPULSIVE:
voice = impulsive_source();
break;
case NATURAL:
voice = natural_source();
break;
case SAMPLED:
voice = sampled_source(0);
break;
case SAMPLED2:
voice = sampled_source(1);
break;
}
/* Reset period when counter 'nper' reaches T0 */
if (kt_globals.nper >= kt_globals.T0)
{
kt_globals.nper = 0;
pitch_synch_par_reset(frame);
}
/*
Low-pass filter voicing waveform before downsampling from 4*samrate
to samrate samples/sec. Resonator f=.09*samrate, bw=.06*samrate
*/
voice = resonator(&(kt_globals.rsn[RLP]),voice);
/* Increment counter that keeps track of 4*samrate samples per sec */
kt_globals.nper++;
}
/*
Tilt spectrum of voicing source down by soft low-pass filtering, amount
of tilt determined by TLTdb
*/
voice = (voice * kt_globals.onemd) + (vlast * kt_globals.decay);
vlast = voice;
/*
Add breathiness during glottal open phase. Amount of breathiness
determined by parameter Aturb Use nrand rather than noise because
noise is low-passed.
*/
if (kt_globals.nper < kt_globals.nopen)
{
voice += kt_globals.amp_breth * kt_globals.nrand;
}
/* Set amplitude of voicing */
glotout = kt_globals.amp_voice * voice;
par_glotout = kt_globals.par_amp_voice * voice;
/* Compute aspiration amplitude and add to voicing source */
aspiration = kt_globals.amp_aspir * noise;
glotout += aspiration;
par_glotout += aspiration;
/*
Cascade vocal tract, excited by laryngeal sources.
Nasal antiresonator, then formants FNP, F5, F4, F3, F2, F1
*/
out=0;
if(kt_globals.synthesis_model != ALL_PARALLEL)
{
casc_next_in = antiresonator2(&(kt_globals.rsn[Rnz]),glotout);
casc_next_in = resonator(&(kt_globals.rsn[Rnpc]),casc_next_in);
casc_next_in = resonator(&(kt_globals.rsn[R8c]),casc_next_in);
casc_next_in = resonator(&(kt_globals.rsn[R7c]),casc_next_in);
casc_next_in = resonator(&(kt_globals.rsn[R6c]),casc_next_in);
casc_next_in = resonator2(&(kt_globals.rsn[R5c]),casc_next_in);
casc_next_in = resonator2(&(kt_globals.rsn[R4c]),casc_next_in);
casc_next_in = resonator2(&(kt_globals.rsn[R3c]),casc_next_in);
casc_next_in = resonator2(&(kt_globals.rsn[R2c]),casc_next_in);
out = resonator2(&(kt_globals.rsn[R1c]),casc_next_in);
}
/* Excite parallel F1 and FNP by voicing waveform */
sourc = par_glotout; /* Source is voicing plus aspiration */
/*
Standard parallel vocal tract Formants F6,F5,F4,F3,F2,
outputs added with alternating sign. Sound source for other
parallel resonators is frication plus first difference of
voicing waveform.
*/
out += resonator(&(kt_globals.rsn[R1p]),sourc);
out += resonator(&(kt_globals.rsn[Rnpp]),sourc);
sourc = frics + par_glotout - glotlast;
glotlast = par_glotout;
for(ix=R2p; ix<=R6p; ix++)
{
out = resonator(&(kt_globals.rsn[ix]),sourc) - out;
}
outbypas = kt_globals.amp_bypas * sourc;
out = outbypas - out;
#ifdef deleted
// for testing
if (kt_globals.outsl != 0)
{
switch(kt_globals.outsl)
{
case 1:
out = voice;
break;
case 2:
out = aspiration;
break;
case 3:
out = frics;
break;
case 4:
out = glotout;
break;
case 5:
out = par_glotout;
break;
case 6:
out = outbypas;
break;
case 7:
out = sourc;
break;
}
}
#endif
out = resonator(&(kt_globals.rsn[Rout]),out);
temp = (int)(out * wdata.amplitude * kt_globals.amp_gain0) ; /* Convert back to integer */
// mix with a recorded WAV if required for this phoneme
{
int z2;
signed char c;
int sample;
z2 = 0;
if(wdata.mix_wavefile_ix < wdata.n_mix_wavefile)
{
if(wdata.mix_wave_scale == 0)
{
// a 16 bit sample
c = wdata.mix_wavefile[wdata.mix_wavefile_ix+1];
sample = wdata.mix_wavefile[wdata.mix_wavefile_ix] + (c * 256);
wdata.mix_wavefile_ix += 2;
}
else
{
// a 8 bit sample, scaled
sample = (signed char)wdata.mix_wavefile[wdata.mix_wavefile_ix++] * wdata.mix_wave_scale;
}
z2 = sample * wdata.amplitude_v / 1024;
z2 = (z2 * wdata.mix_wave_amp)/40;
temp += z2;
}
}
// if fadeout is set, fade to zero over 64 samples, to avoid clicks at end of synthesis
if(kt_globals.fadeout > 0)
{
kt_globals.fadeout--;
temp = (temp * kt_globals.fadeout) / 64;
}
value = (int)temp + ((echo_buf[echo_tail++]*echo_amp) >> 8);
if(echo_tail >= N_ECHO_BUF)
echo_tail=0;
if (value < -32768)
{
value = -32768;
}
if (value > 32767)
{
value = 32767;
}
*out_ptr++ = value;
*out_ptr++ = value >> 8;
echo_buf[echo_head++] = value;
if(echo_head >= N_ECHO_BUF)
echo_head = 0;
sample_count++;
if(out_ptr >= out_end)
{
return(1);
}
}
return(0);
} // end of parwave
void KlattReset(int control)
{
int r_ix;
if(control == 2)
{
//Full reset
kt_globals.FLPhz = (950 * kt_globals.samrate) / 10000;
kt_globals.BLPhz = (630 * kt_globals.samrate) / 10000;
kt_globals.minus_pi_t = -PI / kt_globals.samrate;
kt_globals.two_pi_t = -2.0 * kt_globals.minus_pi_t;
setabc(kt_globals.FLPhz,kt_globals.BLPhz,&(kt_globals.rsn[RLP]));
}
if(control > 0)
{
kt_globals.nper = 0;
kt_globals.T0 = 0;
kt_globals.nopen = 0;
kt_globals.nmod = 0;
for(r_ix=RGL; r_ix < N_RSN; r_ix++)
{
kt_globals.rsn[r_ix].p1 = 0;
kt_globals.rsn[r_ix].p2 = 0;
}
}
for(r_ix=0; r_ix <= R6p; r_ix++)
{
kt_globals.rsn[r_ix].p1 = 0;
kt_globals.rsn[r_ix].p2 = 0;
}
}
/*
function FRAME_INIT
Use parameters from the input frame to set up resonator coefficients.
*/
static void frame_init(klatt_frame_ptr frame)
{
double amp_par[7];
static double amp_par_factor[7] = {0.6, 0.4, 0.15, 0.06, 0.04, 0.022, 0.03};
long Gain0_tmp;
int ix;
kt_globals.original_f0 = frame->F0hz10 / 10;
frame->AVdb_tmp = frame->AVdb - 7;
if (frame->AVdb_tmp < 0)
{
frame->AVdb_tmp = 0;
}
kt_globals.amp_aspir = DBtoLIN(frame->ASP) * 0.05;
kt_globals.amp_frica = DBtoLIN(frame->AF) * 0.25;
kt_globals.par_amp_voice = DBtoLIN(frame->AVpdb);
kt_globals.amp_bypas = DBtoLIN(frame->AB) * 0.05;
for(ix=0; ix <= 6; ix++)
{
// parallel amplitudes F1 to F6, and parallel nasal pole
amp_par[ix] = DBtoLIN(frame->Ap[ix]) * amp_par_factor[ix];
}
Gain0_tmp = frame->Gain0 - 3;
if (Gain0_tmp <= 0)
{
Gain0_tmp = 57;
}
kt_globals.amp_gain0 = DBtoLIN(Gain0_tmp) / kt_globals.scale_wav;
/* Set coefficients of variable cascade resonators */
for(ix=1; ix<=9; ix++)
{
// formants 1 to 8, plus nasal pole
setabc(frame->Fhz[ix],frame->Bhz[ix],&(kt_globals.rsn[ix]));
if(ix <= 5)
{
setabc(frame->Fhz_next[ix],frame->Bhz_next[ix],&(kt_globals.rsn_next[ix]));
kt_globals.rsn[ix].a_inc = (kt_globals.rsn_next[ix].a - kt_globals.rsn[ix].a) / 64.0;
kt_globals.rsn[ix].b_inc = (kt_globals.rsn_next[ix].b - kt_globals.rsn[ix].b) / 64.0;
kt_globals.rsn[ix].c_inc = (kt_globals.rsn_next[ix].c - kt_globals.rsn[ix].c) / 64.0;
}
}
// nasal zero anti-resonator
setzeroabc(frame->Fhz[F_NZ],frame->Bhz[F_NZ],&(kt_globals.rsn[Rnz]));
setzeroabc(frame->Fhz_next[F_NZ],frame->Bhz_next[F_NZ],&(kt_globals.rsn_next[Rnz]));
kt_globals.rsn[F_NZ].a_inc = (kt_globals.rsn_next[F_NZ].a - kt_globals.rsn[F_NZ].a) / 64.0;
kt_globals.rsn[F_NZ].b_inc = (kt_globals.rsn_next[F_NZ].b - kt_globals.rsn[F_NZ].b) / 64.0;
kt_globals.rsn[F_NZ].c_inc = (kt_globals.rsn_next[F_NZ].c - kt_globals.rsn[F_NZ].c) / 64.0;
/* Set coefficients of parallel resonators, and amplitude of outputs */
for(ix=0; ix<=6; ix++)
{
setabc(frame->Fhz[ix],frame->Bphz[ix],&(kt_globals.rsn[Rparallel+ix]));
kt_globals.rsn[Rparallel+ix].a *= amp_par[ix];
}
/* output low-pass filter */
setabc((long)0.0,(long)(kt_globals.samrate/2),&(kt_globals.rsn[Rout]));
}
/*
function IMPULSIVE_SOURCE
Generate a low pass filtered train of impulses as an approximation of
a natural excitation waveform. Low-pass filter the differentiated impulse
with a critically-damped second-order filter, time constant proportional
to Kopen.
*/
static double impulsive_source()
{
static double doublet[] = {0.0,13000000.0,-13000000.0};
static double vwave;
if (kt_globals.nper < 3)
{
vwave = doublet[kt_globals.nper];
}
else
{
vwave = 0.0;
}
return(resonator(&(kt_globals.rsn[RGL]),vwave));
}
/*
function NATURAL_SOURCE
Vwave is the differentiated glottal flow waveform, there is a weak
spectral zero around 800 Hz, magic constants a,b reset pitch synchronously.
*/
static double natural_source()
{
double lgtemp;
static double vwave;
if (kt_globals.nper < kt_globals.nopen)
{
kt_globals.pulse_shape_a -= kt_globals.pulse_shape_b;
vwave += kt_globals.pulse_shape_a;
lgtemp=vwave * 0.028;
return(lgtemp);
}
else
{
vwave = 0.0;
return(0.0);
}
}
/*
function PITCH_SYNC_PAR_RESET
Reset selected parameters pitch-synchronously.
Constant B0 controls shape of glottal pulse as a function
of desired duration of open phase N0
(Note that N0 is specified in terms of 40,000 samples/sec of speech)
Assume voicing waveform V(t) has form: k1 t**2 - k2 t**3
If the radiation characterivative, a temporal derivative
is folded in, and we go from continuous time to discrete
integers n: dV/dt = vwave[n]
= sum over i=1,2,...,n of { a - (i * b) }
= a n - b/2 n**2
where the constants a and b control the detailed shape
and amplitude of the voicing waveform over the open
potion of the voicing cycle "nopen".
Let integral of dV/dt have no net dc flow --> a = (b * nopen) / 3
Let maximum of dUg(n)/dn be constant --> b = gain / (nopen * nopen)
meaning as nopen gets bigger, V has bigger peak proportional to n
Thus, to generate the table below for 40 <= nopen <= 263:
B0[nopen - 40] = 1920000 / (nopen * nopen)
*/
static void pitch_synch_par_reset(klatt_frame_ptr frame)
{
long temp;
double temp1;
static long skew;
static short B0[224] =
{
1200,1142,1088,1038, 991, 948, 907, 869, 833, 799, 768, 738, 710, 683, 658,
634, 612, 590, 570, 551, 533, 515, 499, 483, 468, 454, 440, 427, 415, 403,
391, 380, 370, 360, 350, 341, 332, 323, 315, 307, 300, 292, 285, 278, 272,
265, 259, 253, 247, 242, 237, 231, 226, 221, 217, 212, 208, 204, 199, 195,
192, 188, 184, 180, 177, 174, 170, 167, 164, 161, 158, 155, 153, 150, 147,
145, 142, 140, 137, 135, 133, 131, 128, 126, 124, 122, 120, 119, 117, 115,
113,111, 110, 108, 106, 105, 103, 102, 100, 99, 97, 96, 95, 93, 92, 91, 90,
88, 87, 86, 85, 84, 83, 82, 80, 79, 78, 77, 76, 75, 75, 74, 73, 72, 71,
70, 69, 68, 68, 67, 66, 65, 64, 64, 63, 62, 61, 61, 60, 59, 59, 58, 57,
57, 56, 56, 55, 55, 54, 54, 53, 53, 52, 52, 51, 51, 50, 50, 49, 49, 48, 48,
47, 47, 46, 46, 45, 45, 44, 44, 43, 43, 42, 42, 41, 41, 41, 41, 40, 40,
39, 39, 38, 38, 38, 38, 37, 37, 36, 36, 36, 36, 35, 35, 35, 35, 34, 34,33,
33, 33, 33, 32, 32, 32, 32, 31, 31, 31, 31, 30, 30, 30, 30, 29, 29, 29, 29,
28, 28, 28, 28, 27, 27
};
if (frame->F0hz10 > 0)
{
/* T0 is 4* the number of samples in one pitch period */
kt_globals.T0 = (40 * kt_globals.samrate) / frame->F0hz10;
kt_globals.amp_voice = DBtoLIN(frame->AVdb_tmp);
/* Duration of period before amplitude modulation */
kt_globals.nmod = kt_globals.T0;
if (frame->AVdb_tmp > 0)
{
kt_globals.nmod >>= 1;
}
/* Breathiness of voicing waveform */
kt_globals.amp_breth = DBtoLIN(frame->Aturb) * 0.1;
/* Set open phase of glottal period where 40 <= open phase <= 263 */
kt_globals.nopen = 4 * frame->Kopen;
if ((kt_globals.glsource == IMPULSIVE) && (kt_globals.nopen > 263))
{
kt_globals.nopen = 263;
}
if (kt_globals.nopen >= (kt_globals.T0-1))
{
// printf("Warning: glottal open period cannot exceed T0, truncated\n");
kt_globals.nopen = kt_globals.T0 - 2;
}
if (kt_globals.nopen < 40)
{
/* F0 max = 1000 Hz */
// printf("Warning: minimum glottal open period is 10 samples.\n");
// printf("truncated, nopen = %d\n",kt_globals.nopen);
kt_globals.nopen = 40;
}
/* Reset a & b, which determine shape of "natural" glottal waveform */
kt_globals.pulse_shape_b = B0[kt_globals.nopen-40];
kt_globals.pulse_shape_a = (kt_globals.pulse_shape_b * kt_globals.nopen) * 0.333;
/* Reset width of "impulsive" glottal pulse */
temp = kt_globals.samrate / kt_globals.nopen;
setabc((long)0,temp,&(kt_globals.rsn[RGL]));
/* Make gain at F1 about constant */
temp1 = kt_globals.nopen *.00833;
kt_globals.rsn[RGL].a *= temp1 * temp1;
/*
Truncate skewness so as not to exceed duration of closed phase
of glottal period.
*/
temp = kt_globals.T0 - kt_globals.nopen;
if (frame->Kskew > temp)
{
// printf("Kskew duration=%d > glottal closed period=%d, truncate\n", frame->Kskew, kt_globals.T0 - kt_globals.nopen);
frame->Kskew = temp;
}
if (skew >= 0)
{
skew = frame->Kskew;
}
else
{
skew = - frame->Kskew;
}
/* Add skewness to closed portion of voicing period */
kt_globals.T0 = kt_globals.T0 + skew;
skew = - skew;
}
else
{
kt_globals.T0 = 4; /* Default for f0 undefined */
kt_globals.amp_voice = 0.0;
kt_globals.nmod = kt_globals.T0;
kt_globals.amp_breth = 0.0;
kt_globals.pulse_shape_a = 0.0;
kt_globals.pulse_shape_b = 0.0;
}
/* Reset these pars pitch synchronously or at update rate if f0=0 */
if ((kt_globals.T0 != 4) || (kt_globals.ns == 0))
{
/* Set one-pole low-pass filter that tilts glottal source */
kt_globals.decay = (0.033 * frame->TLTdb);
if (kt_globals.decay > 0.0)
{
kt_globals.onemd = 1.0 - kt_globals.decay;
}
else
{
kt_globals.onemd = 1.0;
}
}
}
/*
function SETABC
Convert formant freqencies and bandwidth into resonator difference
equation constants.
*/
static void setabc(long int f, long int bw, resonator_ptr rp)
{
double r;
double arg;
/* Let r = exp(-pi bw t) */
arg = kt_globals.minus_pi_t * bw;
r = exp(arg);
/* Let c = -r**2 */
rp->c = -(r * r);
/* Let b = r * 2*cos(2 pi f t) */
arg = kt_globals.two_pi_t * f;
rp->b = r * cos(arg) * 2.0;
/* Let a = 1.0 - b - c */
rp->a = 1.0 - rp->b - rp->c;
}
/*
function SETZEROABC
Convert formant freqencies and bandwidth into anti-resonator difference
equation constants.
*/
static void setzeroabc(long int f, long int bw, resonator_ptr rp)
{
double r;
double arg;
f = -f;
//NOTE, changes made 30.09.2011 for Reece Dunn <msclrhd@googlemail.com>
// fix a sound spike when f=0
/* First compute ordinary resonator coefficients */
/* Let r = exp(-pi bw t) */
arg = kt_globals.minus_pi_t * bw;
r = exp(arg);
/* Let c = -r**2 */
rp->c = -(r * r);
/* Let b = r * 2*cos(2 pi f t) */
arg = kt_globals.two_pi_t * f;
rp->b = r * cos(arg) * 2.;
/* Let a = 1.0 - b - c */
rp->a = 1.0 - rp->b - rp->c;
/* Now convert to antiresonator coefficients (a'=1/a, b'=b/a, c'=c/a) */
/* If f == 0 then rp->a gets set to 0 which makes a'=1/a set a', b' and c' to
* INF, causing an audible sound spike when triggered (e.g. apiration with the
* nasal register set to f=0, bw=0).
*/
if (rp->a != 0)
{
/* Now convert to antiresonator coefficients (a'=1/a, b'=b/a, c'=c/a) */
rp->a = 1.0 / rp->a;
rp->c *= -rp->a;
rp->b *= -rp->a;
}
}
/*
function GEN_NOISE
Random number generator (return a number between -8191 and +8191)
Noise spectrum is tilted down by soft low-pass filter having a pole near
the origin in the z-plane, i.e. output = input + (0.75 * lastoutput)
*/
static double gen_noise(double noise)
{
long temp;
static double nlast;
temp = (long) getrandom(-8191,8191);
kt_globals.nrand = (long) temp;
noise = kt_globals.nrand + (0.75 * nlast);
nlast = noise;
return(noise);
}
/*
function DBTOLIN
Convert from decibels to a linear scale factor
Conversion table, db to linear, 87 dB --> 32767
86 dB --> 29491 (1 dB down = 0.5**1/6)
...
81 dB --> 16384 (6 dB down = 0.5)
...
0 dB --> 0
The just noticeable difference for a change in intensity of a vowel
is approximately 1 dB. Thus all amplitudes are quantized to 1 dB
steps.
*/
static double DBtoLIN(long dB)
{
static short amptable[88] =
{
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 7,
8, 9, 10, 11, 13, 14, 16, 18, 20, 22, 25, 28, 32,
35, 40, 45, 51, 57, 64, 71, 80, 90, 101, 114, 128,
142, 159, 179, 202, 227, 256, 284, 318, 359, 405,
455, 512, 568, 638, 719, 881, 911, 1024, 1137, 1276,
1438, 1622, 1823, 2048, 2273, 2552, 2875, 3244, 3645,
4096, 4547, 5104, 5751, 6488, 7291, 8192, 9093, 10207,
11502, 12976, 14582, 16384, 18350, 20644, 23429,
26214, 29491, 32767 };
if ((dB < 0) || (dB > 87))
{
return(0);
}
return((double)(amptable[dB]) * 0.001);
}
extern voice_t *wvoice;
static klatt_peaks_t peaks[N_PEAKS];
static int end_wave;
static int klattp[N_KLATTP];
static double klattp1[N_KLATTP];
static double klattp_inc[N_KLATTP];
int Wavegen_Klatt(int resume)
{//==========================
int pk;
int x;
int ix;
int fade;
if(resume==0)
{
sample_count = 0;
}
while(sample_count < nsamples)
{
kt_frame.F0hz10 = (wdata.pitch * 10) / 4096;
// formants F6,F7,F8 are fixed values for cascade resonators, set in KlattInit()
// but F6 is used for parallel resonator
// F0 is used for the nasal zero
for(ix=0; ix < 6; ix++)
{
kt_frame.Fhz[ix] = peaks[ix].freq;
if(ix < 4)
{
kt_frame.Bhz[ix] = peaks[ix].bw;
}
}
for(ix=1; ix < 7; ix++)
{
kt_frame.Ap[ix] = peaks[ix].ap;
}
kt_frame.AVdb = klattp[KLATT_AV];
kt_frame.AVpdb = klattp[KLATT_AVp];
kt_frame.AF = klattp[KLATT_Fric];
kt_frame.AB = klattp[KLATT_FricBP];
kt_frame.ASP = klattp[KLATT_Aspr];
kt_frame.Aturb = klattp[KLATT_Turb];
kt_frame.Kskew = klattp[KLATT_Skew];
kt_frame.TLTdb = klattp[KLATT_Tilt];
kt_frame.Kopen = klattp[KLATT_Kopen];
// advance formants
for(pk=0; pk<N_PEAKS; pk++)
{
peaks[pk].freq1 += peaks[pk].freq_inc;
peaks[pk].freq = (int)peaks[pk].freq1;
peaks[pk].bw1 += peaks[pk].bw_inc;
peaks[pk].bw = (int)peaks[pk].bw1;
peaks[pk].bp1 += peaks[pk].bp_inc;
peaks[pk].bp = (int)peaks[pk].bp1;
peaks[pk].ap1 += peaks[pk].ap_inc;
peaks[pk].ap = (int)peaks[pk].ap1;
}
// advance other parameters
for(ix=0; ix < N_KLATTP; ix++)
{
klattp1[ix] += klattp_inc[ix];
klattp[ix] = (int)klattp1[ix];
}
for(ix=0; ix<=6; ix++)
{
kt_frame.Fhz_next[ix] = peaks[ix].freq;
if(ix < 4)
{
kt_frame.Bhz_next[ix] = peaks[ix].bw;
}
}
// advance the pitch
wdata.pitch_ix += wdata.pitch_inc;
if((ix = wdata.pitch_ix>>8) > 127) ix = 127;
x = wdata.pitch_env[ix] * wdata.pitch_range;
wdata.pitch = (x>>8) + wdata.pitch_base;
kt_globals.nspfr = (nsamples - sample_count);
if(kt_globals.nspfr > STEPSIZE)
kt_globals.nspfr = STEPSIZE;
frame_init(&kt_frame); /* get parameters for next frame of speech */
if(parwave(&kt_frame) == 1)
{
return(1); // output buffer is full
}
}
if(end_wave > 0)
{
#ifdef deleted
if(end_wave == 2)
{
fade = (kt_globals.T0 - kt_globals.nper)/4; // samples until end of current cycle
if(fade < 64)
fade = 64;
}
else
#endif
{
fade = 64; // not followd by formant synthesis
}
// fade out to avoid a click
kt_globals.fadeout = fade;
end_wave = 0;
sample_count -= fade;
kt_globals.nspfr = fade;
if(parwave(&kt_frame) == 1)
{
return(1); // output buffer is full
}
}
return(0);
}
void SetSynth_Klatt(int length, int modn, frame_t *fr1, frame_t *fr2, voice_t *v, int control)
{//===========================================================================================
int ix;
DOUBLEX next;
int qix;
int cmd;
frame_t *fr3;
static frame_t prev_fr;
if(wvoice != NULL)
{
if((wvoice->klattv[0] > 0) && (wvoice->klattv[0] <=4 ))
{
kt_globals.glsource = wvoice->klattv[0];
kt_globals.scale_wav = scale_wav_tab[kt_globals.glsource];
}
kt_globals.f0_flutter = wvoice->flutter/32;
}
end_wave = 0;
if(control & 2)
{
end_wave = 1; // fadeout at the end
}
if(control & 1)
{
end_wave = 1;
for(qix=wcmdq_head+1;;qix++)
{
if(qix >= N_WCMDQ) qix = 0;
if(qix == wcmdq_tail) break;
cmd = wcmdq[qix][0];
if(cmd==WCMD_KLATT)
{
end_wave = 0; // next wave generation is from another spectrum
fr3 = (frame_t *)wcmdq[qix][2];
for(ix=1; ix<6; ix++)
{
if(fr3->ffreq[ix] != fr2->ffreq[ix])
{
// there is a discontinuity in formants
end_wave = 2;
break;
}
}
break;
}
if((cmd==WCMD_WAVE) || (cmd==WCMD_PAUSE))
break; // next is not from spectrum, so continue until end of wave cycle
}
}
#ifdef LOG_FRAMES
if(option_log_frames)
{
FILE *f_log;
f_log=fopen("log-espeakedit","a");
if(f_log != NULL)
{
fprintf(f_log,"K %3dmS %3d %3d %4d %4d %4d %4d (%2d) to %3d %3d %4d %4d %4d %4d (%2d)\n",length*1000/samplerate,
fr1->klattp[KLATT_FNZ]*2,fr1->ffreq[1],fr1->ffreq[2],fr1->ffreq[3],fr1->ffreq[4],fr1->ffreq[5], fr1->klattp[KLATT_AV],
fr2->klattp[KLATT_FNZ]*2,fr2->ffreq[1],fr2->ffreq[2],fr2->ffreq[3],fr1->ffreq[4],fr1->ffreq[5], fr2->klattp[KLATT_AV] );
fclose(f_log);
}
f_log=fopen("log-klatt","a");
if(f_log != NULL)
{
fprintf(f_log,"K %3dmS %3d %3d %4d %4d (%2d) to %3d %3d %4d %4d (%2d)\n",length*1000/samplerate,
fr1->klattp[KLATT_FNZ]*2,fr1->ffreq[1],fr1->ffreq[2],fr1->ffreq[3], fr1->klattp[KLATT_AV],
fr2->klattp[KLATT_FNZ]*2,fr2->ffreq[1],fr2->ffreq[2],fr2->ffreq[3], fr2->klattp[KLATT_AV] );
fclose(f_log);
}
}
#endif
if(control & 1)
{
for(ix=1; ix<6; ix++)
{
if(prev_fr.ffreq[ix] != fr1->ffreq[ix])
{
// Discontinuity in formants.
// end_wave was set in SetSynth_Klatt() to fade out the previous frame
KlattReset(0);
break;
}
}
memcpy(&prev_fr,fr2,sizeof(prev_fr));
}
for(ix=0; ix<N_KLATTP; ix++)
{
if((ix >= 5) && ((fr1->frflags & FRFLAG_KLATT) == 0))
{
klattp1[ix] = klattp[ix] = 0;
klattp_inc[ix] = 0;
}
else
{
klattp1[ix] = klattp[ix] = fr1->klattp[ix];
klattp_inc[ix] = (double)((fr2->klattp[ix] - klattp[ix]) * STEPSIZE)/length;
}
// get klatt parameter adjustments for the voice
// if((ix>0) && (ix < KLATT_AVp))
// klattp1[ix] = klattp[ix] = (klattp[ix] + wvoice->klattv[ix]);
}
nsamples = length;
for(ix=1; ix < 6; ix++)
{
peaks[ix].freq1 = (fr1->ffreq[ix] * v->freq[ix] / 256.0) + v->freqadd[ix];
peaks[ix].freq = (int)peaks[ix].freq1;
next = (fr2->ffreq[ix] * v->freq[ix] / 256.0) + v->freqadd[ix];
peaks[ix].freq_inc = ((next - peaks[ix].freq1) * STEPSIZE) / length;
if(ix < 4)
{
// klatt bandwidth for f1, f2, f3 (others are fixed)
peaks[ix].bw1 = fr1->bw[ix] * 2;
peaks[ix].bw = (int)peaks[ix].bw1;
next = fr2->bw[ix] * 2;
peaks[ix].bw_inc = ((next - peaks[ix].bw1) * STEPSIZE) / length;
}
}
// nasal zero frequency
peaks[0].freq1 = fr1->klattp[KLATT_FNZ] * 2;
if(peaks[0].freq1 == 0)
peaks[0].freq1 = kt_frame.Fhz[F_NP]; // if no nasal zero, set it to same freq as nasal pole
peaks[0].freq = (int)peaks[0].freq1;
next = fr2->klattp[KLATT_FNZ] * 2;
if(next == 0)
next = kt_frame.Fhz[F_NP];
peaks[0].freq_inc = ((next - peaks[0].freq1) * STEPSIZE) / length;
peaks[0].bw1 = 89;
peaks[0].bw = 89;
peaks[0].bw_inc = 0;
if(fr1->frflags & FRFLAG_KLATT)
{
// the frame contains additional parameters for parallel resonators
for(ix=1; ix < 7; ix++)
{
peaks[ix].bp1 = fr1->klatt_bp[ix] * 4; // parallel bandwidth
peaks[ix].bp = (int)peaks[ix].bp1;
next = fr2->klatt_bp[ix] * 4;
peaks[ix].bp_inc = ((next - peaks[ix].bp1) * STEPSIZE) / length;
peaks[ix].ap1 = fr1->klatt_ap[ix]; // parallal amplitude
peaks[ix].ap = (int)peaks[ix].ap1;
next = fr2->klatt_ap[ix];
peaks[ix].ap_inc = ((next - peaks[ix].ap1) * STEPSIZE) / length;
}
}
} // end of SetSynth_Klatt
int Wavegen_Klatt2(int length, int modulation, int resume, frame_t *fr1, frame_t *fr2)
{//===================================================================================
if(resume==0)
SetSynth_Klatt(length, modulation, fr1, fr2, wvoice, 1);
return(Wavegen_Klatt(resume));
}
void KlattInit()
{
static short formant_hz[10] = {280,688,1064,2806,3260,3700,6500,7000,8000,280};
static short bandwidth[10] = {89,160,70,160,200,200,500,500,500,89};
static short parallel_amp[10] = { 0,59,59,59,59,59,59,0,0,0};
static short parallel_bw[10] = {59,59,89,149,200,200,500,0,0,0};
int ix;
for(ix=0; ix<256; ix++)
{
// TEST: Overwrite natural_samples2
// sawtooth wave
// natural_samples2[ix] = (128-ix) * 20;
}
sample_count=0;
kt_globals.synthesis_model = CASCADE_PARALLEL;
kt_globals.samrate = 22050;
kt_globals.glsource = IMPULSIVE; // IMPULSIVE, NATURAL, SAMPLED
kt_globals.scale_wav = scale_wav_tab[kt_globals.glsource];
kt_globals.natural_samples = natural_samples;
kt_globals.num_samples = NUMBER_OF_SAMPLES;
kt_globals.sample_factor = 3.0;
kt_globals.nspfr = (kt_globals.samrate * 10) / 1000;
kt_globals.outsl = 0;
kt_globals.f0_flutter = 20;
KlattReset(2);
// set default values for frame parameters
for(ix=0; ix<=9; ix++)
{
kt_frame.Fhz[ix] = formant_hz[ix];
kt_frame.Bhz[ix] = bandwidth[ix];
kt_frame.Ap[ix] = parallel_amp[ix];
kt_frame.Bphz[ix] = parallel_bw[ix];
}
kt_frame.Bhz_next[F_NZ] = bandwidth[F_NZ];
kt_frame.F0hz10 = 1000;
kt_frame.AVdb = 59; // 59
kt_frame.ASP = 0;
kt_frame.Kopen = 40; // 40
kt_frame.Aturb = 0;
kt_frame.TLTdb = 0;
kt_frame.AF =50;
kt_frame.Kskew = 0;
kt_frame.AB = 0;
kt_frame.AVpdb = 0;
kt_frame.Gain0 = 62; // 60
} // end of KlattInit
#endif // INCLUDE_KLATT
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