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removed files already in miscDeps submodule
[nvdajp/nvdajpmiscdep.git] / include / espeak / src / klatt.cpp
diff --git a/include/espeak/src/klatt.cpp b/include/espeak/src/klatt.cpp
deleted file mode 100644 (file)
index 5c7da27..0000000
+++ /dev/null
@@ -1,1413 +0,0 @@
-
-/***************************************************************************
- *   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] = 0;
-               }
-
-               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] * 2;
-                       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] * 2;
-                       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