
/***************************************************************************
 *   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
