diff options
Diffstat (limited to 'navit/support/espeak/klatt.c')
| -rw-r--r-- | navit/support/espeak/klatt.c | 662 |
1 files changed, 386 insertions, 276 deletions
diff --git a/navit/support/espeak/klatt.c b/navit/support/espeak/klatt.c index 8cdbf782b..c191534a5 100644 --- a/navit/support/espeak/klatt.c +++ b/navit/support/espeak/klatt.c @@ -22,7 +22,7 @@ * <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" @@ -58,19 +58,69 @@ static int sample_count; /* 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 @@ -81,7 +131,7 @@ 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; @@ -92,11 +142,11 @@ static double resonator(resonator_ptr r, double input) 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; @@ -105,13 +155,13 @@ static double resonator2(resonator_ptr r, double input) -/* +/* 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 @@ -129,7 +179,7 @@ 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; @@ -143,7 +193,7 @@ function FLUTTER 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 @@ -155,7 +205,7 @@ 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); @@ -178,7 +228,7 @@ Allows the use of a glottal excitation waveform sampled from a real voice. */ -static double sampled_source() +static double sampled_source(int source_num) { int itemp; double ftemp; @@ -187,23 +237,35 @@ static double sampled_source() 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 @@ -216,16 +278,17 @@ static double sampled_source() -/* +/* 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; double outbypas; double out; long n4; @@ -241,23 +304,22 @@ static int parwave(klatt_frame_ptr frame) static double sourc; int ix; - frame_init(frame); /* get parameters for next frame of speech */ - flutter(frame); /* add f0 flutter */ -#ifdef deleted +#ifdef LOG_FRAMES +if(option_log_frames) { FILE *f; - f=fopen("klatt_log","a"); - fprintf(f,"%4dhz %2dAV %4d %3d, %4d %3d, %4d %3d, %4d %3d, %4d, %3d, %4d %3d TLT=%2d\n",frame->F0hz10,frame->AVdb, - frame->F1hz,frame->B1hz,frame->F2hz,frame->B2hz,frame->F3hz,frame->B3hz,frame->F4hz,frame->B4hz,frame->F5hz,frame->B5hz,frame->F6hz,frame->B6hz,frame->TLTdb); + 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++) + 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); @@ -266,22 +328,22 @@ static int parwave(klatt_frame_ptr frame) 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) { @@ -289,64 +351,67 @@ static int parwave(klatt_frame_ptr frame) 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; @@ -363,20 +428,20 @@ static int parwave(klatt_frame_ptr frame) 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; @@ -384,14 +449,14 @@ static int parwave(klatt_frame_ptr frame) { 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) { @@ -401,7 +466,7 @@ static int parwave(klatt_frame_ptr frame) case 2: out = aspiration; break; - case 3: + case 3: out = frics; break; case 4: @@ -421,7 +486,7 @@ static int parwave(klatt_frame_ptr frame) #endif out = resonator(&(kt_globals.rsn[Rout]),out); - temp = (out * wdata.amplitude * kt_globals.amp_gain0) ; /* Convert back to integer */ + temp = (int)(out * wdata.amplitude * kt_globals.amp_gain0) ; /* Convert back to integer */ // mix with a recorded WAV if required for this phoneme @@ -429,7 +494,7 @@ static int parwave(klatt_frame_ptr frame) int z2; signed char c; int sample; - + z2 = 0; if(wdata.mix_wavefile_ix < wdata.n_mix_wavefile) { @@ -458,18 +523,27 @@ static int parwave(klatt_frame_ptr frame) temp = (temp * kt_globals.fadeout) / 64; } - if (temp < -32768.0) + value = (int)temp + ((echo_buf[echo_tail++]*echo_amp) >> 8); + if(echo_tail >= N_ECHO_BUF) + echo_tail=0; + + if (value < -32768) { - temp = -32768.0; + value = -32768; } - - if (temp > 32767.0) + + if (value > 32767) { - temp = 32767.0; + value = 32767; } - - *out_ptr++ = (int)(temp); // **JSD - *out_ptr++ = (int)(temp) >> 8; + + *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) { @@ -482,47 +556,52 @@ static int parwave(klatt_frame_ptr frame) -/* -function PARWAVE_INIT - -Initialises all parameters used in parwave, sets resonator internal memory -to zero. -*/ -static void reset_resonators() +void KlattReset(int control) { int r_ix; - for(r_ix=0; r_ix < N_RSN; 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; } } -static void parwave_init() -{ - 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])); - kt_globals.nper = 0; - kt_globals.T0 = 0; - kt_globals.nopen = 0; - kt_globals.nmod = 0; - - reset_resonators(); -} - -/* +/* 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}; @@ -530,13 +609,13 @@ static void frame_init(klatt_frame_ptr frame) 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); @@ -549,14 +628,14 @@ static void frame_init(klatt_frame_ptr frame) } 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=0; ix<=8; ix++) + for(ix=1; ix<=9; ix++) { // formants 1 to 8, plus nasal pole setabc(frame->Fhz[ix],frame->Bhz[ix],&(kt_globals.rsn[ix])); @@ -564,7 +643,7 @@ static void frame_init(klatt_frame_ptr frame) 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; @@ -578,19 +657,19 @@ static void frame_init(klatt_frame_ptr frame) 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])); - + } @@ -598,27 +677,27 @@ static void frame_init(klatt_frame_ptr frame) /* 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)); } @@ -631,20 +710,20 @@ 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() +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); @@ -678,125 +757,125 @@ Assume voicing waveform V(t) has form: k1 t**2 - k2 t**3 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; @@ -805,20 +884,20 @@ static void pitch_synch_par_reset(klatt_frame_ptr frame) 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; } @@ -830,7 +909,7 @@ static void pitch_synch_par_reset(klatt_frame_ptr frame) /* function SETABC -Convert formant freqencies and bandwidth into resonator difference +Convert formant freqencies and bandwidth into resonator difference equation constants. */ @@ -839,18 +918,18 @@ 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; } @@ -859,7 +938,7 @@ static void setabc(long int f, long int bw, resonator_ptr rp) /* function SETZEROABC -Convert formant freqencies and bandwidth into anti-resonator difference +Convert formant freqencies and bandwidth into anti-resonator difference equation constants. */ @@ -867,56 +946,62 @@ static void setzeroabc(long int f, long int bw, resonator_ptr rp) { double r; double arg; - + f = -f; - - if(f>=0) - { - f = -1; - } - + +//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) */ - rp->a = 1.0 / rp->a; - rp->c *= -rp->a; - rp->b *= -rp->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) +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); } @@ -933,16 +1018,16 @@ Conversion table, db to linear, 87 dB --> 32767 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, @@ -953,12 +1038,12 @@ static double DBtoLIN(long dB) 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); } @@ -973,15 +1058,15 @@ static int klattp[N_KLATTP]; 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 -int Wavegen_Klatt(int resume) +static int Wavegen_Klatt(int resume) {//========================== int pk; int x; int ix; + int fade; if(resume==0) { @@ -1005,7 +1090,7 @@ int Wavegen_Klatt(int resume) } for(ix=1; ix < 7; ix++) { - kt_frame.Ap[ix] = 0; + kt_frame.Ap[ix] = peaks[ix].ap; } kt_frame.AVdb = klattp[KLATT_AV]; @@ -1035,7 +1120,7 @@ int Wavegen_Klatt(int resume) for(ix=0; ix < N_KLATTP; ix++) { klattp1[ix] += klattp_inc[ix]; - klattp[ix] = (int)(klattp1[ix]); + klattp[ix] = (int)klattp1[ix]; } for(ix=0; ix<=6; ix++) @@ -1057,22 +1142,37 @@ int Wavegen_Klatt(int resume) 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); + return(1); // output buffer is full } } - if(end_wave == 1) + 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 = 64; + kt_globals.fadeout = fade; end_wave = 0; - sample_count -= 64; - kt_globals.nspfr = 64; + sample_count -= fade; + kt_globals.nspfr = fade; if(parwave(&kt_frame) == 1) { - return(1); + return(1); // output buffer is full } } @@ -1080,17 +1180,18 @@ int Wavegen_Klatt(int resume) } -void SetSynth_Klatt(int length, int modn, frame_t *fr1, frame_t *fr2, voice_t *v, int control) +static 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] <=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]; @@ -1110,11 +1211,22 @@ void SetSynth_Klatt(int length, int modn, frame_t *fr1, frame_t *fr2, voice_t *v { 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)) @@ -1122,47 +1234,44 @@ void SetSynth_Klatt(int length, int modn, frame_t *fr1, frame_t *fr2, voice_t *v } } +#ifdef LOG_FRAMES +if(option_log_frames) { -//FILE *f; -//f=fopen("klatt_log","a"); -//fprintf(f,"len %4d (%3d %4d %4d) (%3d %4d %4d)\n",length,fr1->ffreq[1],fr1->ffreq[2],fr1->ffreq[3],fr2->ffreq[1],fr2->ffreq[2],fr2->ffreq[3]); -//fclose(f); + 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) { - if(wdata.prev_was_synth == 0) - { - // A break, not following on from another synthesized sound. - // Reset the synthesizer - //reset_resonators(&kt_globals); - parwave_init(); - } - else + for(ix=1; ix<6; ix++) { - if((prev_fr.ffreq[1] != fr1->ffreq[1]) || (prev_fr.ffreq[2] != fr1->ffreq[2])) + if(prev_fr.ffreq[ix] != fr1->ffreq[ix]) { - - // fade out to avoid a click, but only up to the end of output buffer - ix = (out_end - out_ptr)/2; - if(ix > 64) - ix = 64; - kt_globals.fadeout = ix; - kt_globals.nspfr = ix; - parwave(&kt_frame); - - //reset_resonators(&kt_globals); - parwave_init(); + // Discontinuity in formants. + // end_wave was set in SetSynth_Klatt() to fade out the previous frame + KlattReset(0); + break; } } - wdata.prev_was_synth = 1; memcpy(&prev_fr,fr2,sizeof(prev_fr)); } - if(fr2->frflags & FRFLAG_BREAK) - { -// fr2 = fr1; -// reset_resonators(&kt_globals); - } for(ix=0; ix<N_KLATTP; ix++) { @@ -1187,7 +1296,7 @@ void SetSynth_Klatt(int length, int modn, frame_t *fr1, frame_t *fr2, voice_t *v 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); + 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; @@ -1195,7 +1304,7 @@ void SetSynth_Klatt(int length, int modn, frame_t *fr1, frame_t *fr2, voice_t *v { // klatt bandwidth for f1, f2, f3 (others are fixed) peaks[ix].bw1 = fr1->bw[ix] * 2; - peaks[ix].bw = (int)(peaks[ix].bw1); + peaks[ix].bw = (int)peaks[ix].bw1; next = fr2->bw[ix] * 2; peaks[ix].bw_inc = ((next - peaks[ix].bw1) * STEPSIZE) / length; } @@ -1203,8 +1312,14 @@ void SetSynth_Klatt(int length, int modn, frame_t *fr1, frame_t *fr2, voice_t *v // nasal zero frequency peaks[0].freq1 = fr1->klattp[KLATT_FNZ] * 2; - peaks[0].freq = (int)(peaks[0].freq1); + 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; @@ -1217,13 +1332,13 @@ void SetSynth_Klatt(int length, int modn, frame_t *fr1, frame_t *fr2, voice_t *v 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 = (int)peaks[ix].bp1; + next = fr2->klatt_bp[ix] * 4; peaks[ix].bp_inc = ((next - peaks[ix].bp1) * STEPSIZE) / length; peaks[ix].ap1 = fr1->klatt_ap[ix]; // parallal amplitude - peaks[ix].ap = (int)(peaks[ix].ap1); - next = fr2->klatt_ap[ix] * 2; + peaks[ix].ap = (int)peaks[ix].ap1; + next = fr2->klatt_ap[ix]; peaks[ix].ap_inc = ((next - peaks[ix].ap1) * STEPSIZE) / length; } } @@ -1242,18 +1357,7 @@ int Wavegen_Klatt2(int length, int modulation, int resume, frame_t *fr1, frame_t 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}; @@ -1261,6 +1365,12 @@ void KlattInit() 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; @@ -1275,7 +1385,7 @@ void KlattInit() kt_globals.outsl = 0; kt_globals.f0_flutter = 20; - parwave_init(); + KlattReset(2); // set default values for frame parameters for(ix=0; ix<=9; ix++) @@ -1297,7 +1407,7 @@ void KlattInit() kt_frame.Kskew = 0; kt_frame.AB = 0; kt_frame.AVpdb = 0; - kt_frame.Gain0 = 60; // 62 + kt_frame.Gain0 = 62; // 60 } // end of KlattInit #endif // INCLUDE_KLATT |