Merge branch 'development'

Conflicts:
	src/klatt.cpp

dictsource/en_list

 pastry		peIstri
 pate		pat2eI
 ?3 patent	pat@nt
+pathe		paTeI
 pathetic	$alt3
 pathos		peIT0s
 patio		patIoU
 veg		vEdZ
 vegetable	vEdZI2t@b@L
 vehicle		vi@k@L
+vega		vi:g@
 veggy		vEdZI
 vengeance	vEndZ@ns
 ventriloquist	$2

dictsource/hy_list

 // ?1  West Armenian
 
 // names of letters
-_ա	aIb
-բ	ben
-գ	gim
-դ	da
-ե	jetS
-զ	za
-_է	e
-_ը	'@t#
-թ	t#o
-ժ	Ze
-ի	ini
-լ	lyn
-խ	Xe
-ծ	tsa
-կ	ken
-հ	ho
-ձ	dza
-ղ	r"at
-ճ	ce
-մ	men
-յ	hi
-ն	nu
-շ	Sa
-ո	vo
-չ	tSa
-պ	pe
-ջ	dZe
-ռ	R2a
-ս	se
-վ	vev
-տ	tyn
-ր	re
-ց	ts#o
-ւ	hyn
-փ	p#yr
-ք	k#e
-և	jev
-_օ	o
-ֆ	fe
+
+_ա	a:
+բ	b'@
+գ	g'@
+դ	d'@
+ե	j'e
+զ	z'@
+_է	'e:
+_ը	'@:
+թ	t#'@
+ժ	Z'@
+ի	'i:
+լ	l'@
+խ	X'@
+ծ	ts'@
+կ	k'@
+հ	h'@
+ձ	dz'@
+ղ	r"'@
+ճ	tS'@
+մ	m'@
+յ	j'@
+ն	n'@
+շ	S'@
+ո	v'o
+չ	tS#'@
+պ	p'@
+ջ	dZ'@
+ռ	R2'@
+ս	s'@
+վ	v'@
+տ	t'@
+ր	r'@
+ց	ts#'@
+ւ	'u:
+փ	p#'@
+ք	k#'@
+և	j'ev
+_օ	'o:
+ֆ	f'@
+
+// Note: Western Armenian consonants are transformed in hy-west voice file
+?1 _ա	aIb
+?1 բ	ben
+?1 գ	gim
+?1 դ	da
+?1 ե	jetS
+?1 զ	za
+?1 _է	e
+?1 _ը	'@t#
+?1 թ	t#o
+?1 ժ	Ze
+?1 ի	ini
+?1 լ	lyn
+?1 խ	Xe
+?1 ծ	tsa
+?1 կ	ken
+?1 հ	ho
+?1 ձ	dza
+?1 ղ	r"at
+?1 ճ	tSe
+?1 մ	men
+?1 յ	hi
+?1 ն	nu
+?1 շ	Sa
+?1 ո	vo
+?1 չ	tS#a
+?1 պ	pe
+?1 ջ	dZe
+?1 ռ	R2a
+?1 ս	se
+?1 վ	vev
+?1 տ	tyn
+?1 ր	re
+?1 ց	ts#o
+?1 ւ	hyn
+?1 փ	p#yr
+?1 ք	k#e
+?1 և	jev
+?1 _օ	o
+?1 ֆ	fe
+
 
 _cap	m,ets_|
 _??	g,iR

dictsource/hy_rules

      _) ղ (C     r"@
 
 .group ճ
-        ճ        c
-     _) ճ (C     c@
+        ճ        tS
+     _) ճ (C     tS@
 
 .group մ
         մ        m
 ?1      ոյ (_    o
 
 .group չ
-        չ        tS
-     _) չ (C     tS@
+        չ        tS#
+     _) չ (C     tS#@
 
 .group պ
         պ        p

dictsource/pt_list

 alcateia	$alt
 alcova		$alt2
 alegro		$alt2 $noun
-alferes		$alt
 almoço		$alt $verb
 ameba		$alt
 amores		$alt2
 farofa		$alt
 farolete	$alt2
 febre		$alt
-fera		$alt
 feitora		$alt2
 fezes		$alt
 filete		$alt2
 osso		$alt2
 ossos		$alt
 ostra		$alt2
-ovelha		$alt2
 ovos		$alt
 pachorra	$alt2
 paexa		paeSa
 pelo		$alt $verb
 pereba		$alt
 peroba		$alt
+peses	$alt
 peseta		$alt2
 peso		$alt $verb
 piloto		$alt2 $noun

dictsource/pt_rules

 ?2      el (K      EU
         e (lh+     e
         e (ll+     E
-     v) e (lhA_    E
+
 
         e (bL02_   E
 
         e (lL04_   E
 
      i) e (r_      E
-    @s) e (r_      E
+
     qu) e (r_      E
 
     ib) e (rL04_   E
     iv) e (rL04_   E
    pod) e (rA_     e
 
-    @s) e (rL04_   E
+    @s) e (L05_    E
 
     sp) e (rL04_   E
-     f) e (rL02_   E
+
+     f) e (rL04_   E
 
         e (quL02_  E
         e (rbL04_  E
         e (rsL04_  E
 
         e (sL02_   E
-
         e (ssL04_  E
         e (ss_     E         // stress and english words
 ?2   C) ee (_      i
 ?1   v) e (l_      ,E        // eg: possível, amovível, disponível, etc...
 ?1   l) e (ta      'E        // eg: bicicleta, atleta.
 
-    sf) e (ra_     E         // esfera, biosfera
     ef) e (tuL02_  e
      t) e (tra     E
 
    óCC) e (CA_     e
     úC) e (CA_     e
     úC) el (_      eU
+        ection (_  'EkS&~N
 
 //sort
     _d) e (L05_    E
    _vi) e (L05_    E
    end) e (L05_    e         // entenderam, etc.
    fiz) e (L05_    E
-   pus) e (L05_    E         // com- im- de-
    tiv) e (L05_    E         // es-
    viv) e (L05_    e
   _com) e (L05_    e
  _exag) e (L05_    E
  _houv) e (L05_    E
  _mulh) e (L05_    E
- _quis) e (L05_    E
  _soub) e (L05_    E
  _talh) e (L05_    E
 _escond) e (L05_   e
 _interromp) e (L07_ e
 _troux) e (L07_    E
   _ens) e (bL03_   E
+    _v) e (lhL03_  E
 _amanc) e (bL03_   E
     qu) e (brL04_  E
   _cel) e (brL04_  E

espeak-data/voices/!v/klatt4

+language variant
+name klatt4
+klatt 4
+ 

espeak-data/voices/asia/hy-west

 replace 00 b  p#
 replace 00 d  t#
 replace 00 dz ts#
-replace 00 dZ tS
+replace 00 dZ tS#
 replace 00 g  k#
 
 replace 00 p  b
 replace 00 t  d
 replace 00 ts dz
-replace 00 c  dZ
+replace 00 tS dZ
 replace 00 k  g
 
 replace 00 R2 R  // ??

phsource/ph_armenian

 endphoneme
 
 
-phoneme c
-  import_phoneme consonants/c2
+phoneme tS
+  vls pla afr sibilant
+  voicingswitch dZ
+  lengthmod 2
+  Vowelin f1=0  f2=2300 200 400  f3=-100 80
+
+  IF nextPh(isPause2) THEN
+    WAV(ustop/tsh_unasp, 60)
+  ENDIF
+  WAV(ustop/tsh_unasp)
 endphoneme
 
+phoneme tS#
+  import_phoneme base/tS
+endphoneme
 
 phoneme k
   import_phoneme consonants/k-

phsource/ph_hindi

   lengthmod 2
   ipa ʈU+02B0
   voicingswitch d.#
-  Vowelin f1=0  f2=1800 -300 300  f3=-400 80
+  Vowelin f1=0  f2=1800 -300 300  f3=-200 80
   Vowelout f1=0 f2=1800 -300 250  f3=-400 80  rms=20 colr=2
   
   IF nextPhW(t.#) THEN
     ChangePhoneme(t.)
   ENDIF
-  WAV(ustop/th_rfx)
+  WAV(ustop/th_rfx2, 80)
 endphoneme
 
 phoneme d.
   lengthmod 5
   ipa ɖU+02B0
   voicingswitch t.#
-  Vowelin f1=1  f2=1800 -300 300  f3=-150 80
+  Vowelin f1=1  f2=1800 -300 300  f3=-400 80
   Vowelout f1=1 f2=1800 -300 300  f3=-400 80 colr=2
 
   IF PreVoicing THEN
     FMT(d/xdr)
   ENDIF
 
-  FMT(d/dr2, 50) addWav(ustop/th_rfx2, 80)
+  FMT(d/dr) addWav(ustop/t)
+//  FMT(d/dr) addWav(ustop/th_rfx)
 endphoneme
 
 

phsource/ph_turkish

 
 
 phoneme R
-  liquid trill rhotic
+  liquid trill
   lengthmod 6
   IF prevPhW(isVowel) AND nextPhW(isVowel) THEN
     ChangePhoneme(*)

phsource/phonemes

 
 
 phoneme R
-  liquid rhotic
+  liquid
   trill
   lengthmod 6
   ipa r
 endphoneme
 
 phoneme R2   // this is [R] from Slovak/Czech
-  liquid rhotic
+  liquid
   trill
   lengthmod 6
   ipa r
 endphoneme
 
 phoneme R3  // Afrikaans
-  liquid rhotic
+  liquid
   trill  
   lengthmod 6
   length 60
 
 
 phoneme r"     // uvular trill
-  vcd uvl frc rhotic
+  vcd uvl frc
   ipa ʀ
   lengthmod 6
   IF nextPh(isNotVowel) THEN

src/klatt.cpp

  *               <http://www.gnu.org/licenses/>.                           *
  ***************************************************************************/
 
-// See URL: ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/synthesis/klatt.3.04.tar.gz  
+// See URL: ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/synthesis/klatt.3.04.tar.gz
 
 #include "StdAfx.h"
 
 /* function prototypes for functions private to this file */
 
 static void flutter(klatt_frame_ptr);
-static double sampled_source (void);
+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 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 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
 
 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;
 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;
 
 
 
-/* 
+/*
 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 
+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) 
+Output = (rnz.a * input) + (rnz.b * oldin1) + (rnz.c * oldin2)
 */
 
 #ifdef deleted
 	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;
 
 This function adds F0 flutter, as specified in:
 
-"Analysis, synthesis and perception of voice quality variations among 
+"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
 	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);
 voice.
 */
 
-static double sampled_source()
+static double sampled_source(int source_num)
 {
 	int itemp;
 	double ftemp;
 	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 = kt_globals.natural_samples[itemp];
-		next_value = kt_globals.natural_samples[itemp+1];
-	
+
+		current_value = samples[itemp];
+		next_value = samples[itemp+1];
+
 		diff_value = (double) next_value - (double) current_value;
 		diff_value = diff_value * temp_diff;
-	
-		result = kt_globals.natural_samples[itemp] + diff_value;
+
+		result = samples[itemp] + diff_value;
 		result = result * kt_globals.sample_factor;
 	}
 	else
 
 
 
-/* 
+/*
 function PARWAVE
 
 Converts synthesis parameters to a waveform.
 */
 
 
-static int parwave(klatt_frame_ptr frame) 
+static int parwave(klatt_frame_ptr frame)
 {
 	double temp;
 	int value;
 
 	/* MAIN LOOP, for each output sample of current frame: */
 
-	for (kt_globals.ns=0; kt_globals.ns<kt_globals.nspfr; kt_globals.ns++) 
+	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) 
+
+		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 
+			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++) 
+
+		for (n4=0; n4<4; n4++)
 		{
 			switch(kt_globals.glsource)
 			{
 				voice = impulsive_source();
 				break;
 			case NATURAL:
-				voice = natural_source();	
+				voice = natural_source();
 				break;
 			case SAMPLED:
-				voice = sampled_source();
+				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) 
+			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 
+			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. 
+
+		/*
+			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) 
+
+
+		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 
+			Nasal antiresonator, then formants FNP, F5, F4, F3, F2, F1
 		*/
 
 		out=0;
 			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. 
+			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;
 
 		{
 			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) 
+		if (kt_globals.outsl != 0)
 		{
 			switch(kt_globals.outsl)
 			{
 			case 2:
 				out = aspiration;
 				break;
-			case 3: 
+			case 3:
 				out = frics;
 				break;
 			case 4:
 			int z2;
 			signed char c;
 			int sample;
-	
+
 			z2 = 0;
 			if(wdata.mix_wavefile_ix < wdata.n_mix_wavefile)
 			{
 		{
 			value = -32768;
 		}
-	
+
 		if (value > 32767)
 		{
 			value =  32767;
 }
 
 
-/* 
+/*
 function FRAME_INIT
 
 Use parameters from the input frame to set up resonator coefficients.
 */
 
-static void frame_init(klatt_frame_ptr frame) 
+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};
 	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);
 	}
 
 	Gain0_tmp = frame->Gain0 - 3;
-	if (Gain0_tmp <= 0) 
+	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++)
 	{
 		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;
 	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 
+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 impulsive_source()
 {
 	static double doublet[] = {0.0,13000000.0,-13000000.0};
 	static double vwave;
-	
-	if (kt_globals.nper < 3) 
+
+	if (kt_globals.nper < 3)
 	{
 		vwave = doublet[kt_globals.nper];
 	}
-	else 
+	else
 	{
 		vwave = 0.0;
 	}
-	
+
 	return(resonator(&(kt_globals.rsn[RGL]),vwave));
 }
 
 spectral zero around 800 Hz, magic constants a,b reset pitch synchronously.
 */
 
-static double natural_source() 
+static double natural_source()
 {
 	double lgtemp;
 	static double vwave;
-	
-	if (kt_globals.nper < kt_globals.nopen) 
+
+	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 
+	else
 	{
 		vwave = 0.0;
 		return(0.0);
   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) 
+static void pitch_synch_par_reset(klatt_frame_ptr frame)
 {
 	long temp;
 	double temp1;
 	static long skew;
-	static short B0[224] = 
+	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, 
+		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, 
+		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) 
+
+	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) 
+		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)) 
+
+		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) 
+
+		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) 
+		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) 
+		if (skew >= 0)
 		{
 			skew = frame->Kskew;
 		}
-		else 
+		else
 		{
 			skew = - frame->Kskew;
 		}
-	
+
 		/* Add skewness to closed portion of voicing period */
 		kt_globals.T0 = kt_globals.T0 + skew;
 		skew = - skew;
 	}
-	else 
+	else
 	{
 		kt_globals.T0 = 4;                     /* Default for f0 undefined */
 		kt_globals.amp_voice = 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)) 
+
+	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) 
+
+		if (kt_globals.decay > 0.0)
 		{
 			kt_globals.onemd = 1.0 - kt_globals.decay;
 		}
-		else 
+		else
 		{
 			kt_globals.onemd = 1.0;
 		}
 /*
 function SETABC
 
-Convert formant freqencies and bandwidth into resonator difference 
+Convert formant freqencies and bandwidth into resonator difference
 equation constants.
 */
 
 {
 	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 
+Convert formant freqencies and bandwidth into anti-resonator difference
 equation constants.
 */
 
 {
 	double r;
 	double arg;
-	
+
 	f = -f;
 
 //NOTE, changes made 30.09.2011 for Reece Dunn <[email protected]>
 	/* 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).
 }
 
 
-/* 
+/*
 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) 
+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) 
+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);
 }
 
                                 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 double DBtoLIN(long dB)
 {
-	static short amptable[88] = 
+	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,
 		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);
 }
 
 static double klattp1[N_KLATTP];
 static double klattp_inc[N_KLATTP];
 
-static int scale_wav_tab[] = {45,38,45,45};   // scale output from different voicing sources
 
 
 
 
 	if(wvoice != NULL)
 	{
-		if((wvoice->klattv[0] > 0) && (wvoice->klattv[0] <=3 ))
+		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];
 		{
 			if(qix >= N_WCMDQ) qix = 0;
 			if(qix == wcmdq_tail) break;
-	
+
 			cmd = wcmdq[qix][0];
 			if(cmd==WCMD_KLATT)
 			{
 		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);
 	}
 }
 
 void KlattInit()
 {
-#define NUMBER_OF_SAMPLES 100
 
-	static short natural_samples[NUMBER_OF_SAMPLES]=
-	{
-		-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
-	};
 	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};
 
 	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;

src/klatt.h

 
 
 #define CASCADE_PARALLEL 1         /* Type of synthesis model */
-#define ALL_PARALLEL     2 
+#define ALL_PARALLEL     2
 
 #define IMPULSIVE        1         /* Type of voicing source */
 #define NATURAL          2
 #define SAMPLED          3
+#define SAMPLED2         4
 
 #define PI               3.1415927
 
   resonator_t rsn_next[N_RSN];
 
 } klatt_global_t, *klatt_global_ptr;
-  
+
 /* Structure for Klatt Parameters */
 
 #define F_NZ   0  // nasal zero formant

src/synthdata.cpp

 #include "translate.h"
 #include "wave.h"
 
-const char *version_string = "1.47.10a  03.May.13";
+const char *version_string = "1.47.10b  03.May.13";
 const int version_phdata  = 0x014709;
 
 int option_device_number = -1;