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@@ -22,7 +22,7 @@ |
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* <http://www.gnu.org/licenses/>. * |
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***************************************************************************/ |
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// See URL: ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/synthesis/klatt.3.04.tar.gz |
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// See URL: ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/synthesis/klatt.3.04.tar.gz |
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#include "StdAfx.h" |
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@@ -58,19 +58,69 @@ static int sample_count; |
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/* function prototypes for functions private to this file */ |
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static void flutter(klatt_frame_ptr); |
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static double sampled_source (void); |
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static double sampled_source (int); |
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static double impulsive_source (void); |
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static double natural_source (void); |
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static void pitch_synch_par_reset (klatt_frame_ptr); |
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static void pitch_synch_par_reset (klatt_frame_ptr); |
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static double gen_noise (double); |
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static double DBtoLIN (long); |
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static void frame_init (klatt_frame_ptr); |
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static void frame_init (klatt_frame_ptr); |
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static void setabc (long,long,resonator_ptr); |
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static void setzeroabc (long,long,resonator_ptr); |
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static klatt_frame_t kt_frame; |
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static klatt_global_t kt_globals; |
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#define NUMBER_OF_SAMPLES 100 |
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static int scale_wav_tab[] = {45,38,45,45,55}; // scale output from different voicing sources |
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// For testing, this can be overwritten in KlattInit() |
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static short natural_samples2[256]= { |
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2583, 2516, 2450, 2384, 2319, 2254, 2191, 2127, |
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2067, 2005, 1946, 1890, 1832, 1779, 1726, 1675, |
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1626, 1579, 1533, 1491, 1449, 1409, 1372, 1336, |
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1302, 1271, 1239, 1211, 1184, 1158, 1134, 1111, |
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1089, 1069, 1049, 1031, 1013, 996, 980, 965, |
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950, 936, 921, 909, 895, 881, 869, 855, |
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843, 830, 818, 804, 792, 779, 766, 754, |
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740, 728, 715, 702, 689, 676, 663, 651, |
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637, 626, 612, 601, 588, 576, 564, 552, |
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540, 530, 517, 507, 496, 485, 475, 464, |
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454, 443, 434, 424, 414, 404, 394, 385, |
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375, 366, 355, 347, 336, 328, 317, 308, |
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299, 288, 280, 269, 260, 250, 240, 231, |
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220, 212, 200, 192, 181, 172, 161, 152, |
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142, 133, 123, 113, 105, 94, 86, 76, |
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67, 57, 49, 39, 30, 22, 11, 4, |
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-5, -14, -23, -32, -41, -50, -60, -69, |
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-78, -87, -96, -107, -115, -126, -134, -144, |
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-154, -164, -174, -183, -193, -203, -213, -222, |
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-233, -242, -252, -262, -271, -281, -291, -301, |
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-310, -320, -330, -339, -349, -357, -368, -377, |
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-387, -397, -406, -417, -426, -436, -446, -456, |
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-467, -477, -487, -499, -509, -521, -532, -543, |
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-555, -567, -579, -591, -603, -616, -628, -641, |
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-653, -666, -679, -692, -705, -717, -732, -743, |
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-758, -769, -783, -795, -808, -820, -834, -845, |
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-860, -872, -885, -898, -911, -926, -939, -955, |
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-968, -986, -999, -1018, -1034, -1054, -1072, -1094, |
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-1115, -1138, -1162, -1188, -1215, -1244, -1274, -1307, |
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-1340, -1377, -1415, -1453, -1496, -1538, -1584, -1631, |
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-1680, -1732, -1783, -1839, -1894, -1952, -2010, -2072, |
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-2133, -2196, -2260, -2325, -2390, -2456, -2522, -2589, |
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}; |
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static short natural_samples[100]= |
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{ |
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-310,-400,530,356,224,89,23,-10,-58,-16,461,599,536,701,770, |
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605,497,461,560,404,110,224,131,104,-97,155,278,-154,-1165, |
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-598,737,125,-592,41,11,-247,-10,65,92,80,-304,71,167,-1,122, |
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233,161,-43,278,479,485,407,266,650,134,80,236,68,260,269,179, |
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53,140,275,293,296,104,257,152,311,182,263,245,125,314,140,44, |
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203,230,-235,-286,23,107,92,-91,38,464,443,176,98,-784,-2449, |
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-1891,-1045,-1600,-1462,-1384,-1261,-949,-730 |
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}; |
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/* |
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function RESONATOR |
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@@ -81,7 +131,7 @@ is stored in the globals structure. |
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static double resonator(resonator_ptr r, double input) |
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{ |
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double x; |
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x = (double) ((double)r->a * (double)input + (double)r->b * (double)r->p1 + (double)r->c * (double)r->p2); |
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r->p2 = (double)r->p1; |
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r->p1 = (double)x; |
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@@ -92,11 +142,11 @@ static double resonator(resonator_ptr r, double input) |
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static double resonator2(resonator_ptr r, double input) |
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{ |
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double x; |
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x = (double) ((double)r->a * (double)input + (double)r->b * (double)r->p1 + (double)r->c * (double)r->p2); |
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r->p2 = (double)r->p1; |
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r->p1 = (double)x; |
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r->a += r->a_inc; |
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r->b += r->b_inc; |
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r->c += r->c_inc; |
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@@ -105,13 +155,13 @@ static double resonator2(resonator_ptr r, double input) |
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/* |
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/* |
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function ANTIRESONATOR |
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This is a generic anti-resonator function. The code is the same as resonator |
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except that a,b,c need to be set with setzeroabc() and we save inputs in |
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This is a generic anti-resonator function. The code is the same as resonator |
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except that a,b,c need to be set with setzeroabc() and we save inputs in |
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p1/p2 rather than outputs. There is currently only one of these - "rnz" |
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Output = (rnz.a * input) + (rnz.b * oldin1) + (rnz.c * oldin2) |
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Output = (rnz.a * input) + (rnz.b * oldin1) + (rnz.c * oldin2) |
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*/ |
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#ifdef deleted |
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@@ -129,7 +179,7 @@ static double antiresonator2(resonator_ptr r, double input) |
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register double x = (double)r->a * (double)input + (double)r->b * (double)r->p1 + (double)r->c * (double)r->p2; |
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r->p2 = (double)r->p1; |
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r->p1 = (double)input; |
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r->a += r->a_inc; |
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r->b += r->b_inc; |
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r->c += r->c_inc; |
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@@ -143,7 +193,7 @@ function FLUTTER |
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This function adds F0 flutter, as specified in: |
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"Analysis, synthesis and perception of voice quality variations among |
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"Analysis, synthesis and perception of voice quality variations among |
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female and male talkers" D.H. Klatt and L.C. Klatt JASA 87(2) February 1990. |
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Flutter is added by applying a quasi-random element constructed from three |
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@@ -155,7 +205,7 @@ static void flutter(klatt_frame_ptr frame) |
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static int time_count; |
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double delta_f0; |
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double fla,flb,flc,fld,fle; |
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fla = (double) kt_globals.f0_flutter / 50; |
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flb = (double) kt_globals.original_f0 / 100; |
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// flc = sin(2*PI*12.7*time_count); |
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@@ -178,7 +228,7 @@ Allows the use of a glottal excitation waveform sampled from a real |
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voice. |
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*/ |
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static double sampled_source() |
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static double sampled_source(int source_num) |
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{ |
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int itemp; |
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double ftemp; |
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@@ -187,23 +237,35 @@ static double sampled_source() |
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int current_value; |
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int next_value; |
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double temp_diff; |
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short *samples; |
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if(source_num == 0) |
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{ |
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samples = natural_samples; |
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kt_globals.num_samples = 100; |
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} |
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else |
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{ |
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samples = natural_samples2; |
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kt_globals.num_samples = 256; |
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} |
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if(kt_globals.T0!=0) |
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{ |
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ftemp = (double) kt_globals.nper; |
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ftemp = ftemp / kt_globals.T0; |
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ftemp = ftemp * kt_globals.num_samples; |
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itemp = (int) ftemp; |
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temp_diff = ftemp - (double) itemp; |
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current_value = kt_globals.natural_samples[itemp]; |
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next_value = kt_globals.natural_samples[itemp+1]; |
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current_value = samples[itemp]; |
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next_value = samples[itemp+1]; |
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diff_value = (double) next_value - (double) current_value; |
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diff_value = diff_value * temp_diff; |
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result = kt_globals.natural_samples[itemp] + diff_value; |
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result = samples[itemp] + diff_value; |
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result = result * kt_globals.sample_factor; |
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} |
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else |
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@@ -216,14 +278,14 @@ static double sampled_source() |
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/* |
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/* |
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function PARWAVE |
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Converts synthesis parameters to a waveform. |
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*/ |
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static int parwave(klatt_frame_ptr frame) |
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static int parwave(klatt_frame_ptr frame) |
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{ |
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double temp; |
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int value; |
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@@ -257,7 +319,7 @@ if(option_log_frames) |
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/* MAIN LOOP, for each output sample of current frame: */ |
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for (kt_globals.ns=0; kt_globals.ns<kt_globals.nspfr; kt_globals.ns++) |
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for (kt_globals.ns=0; kt_globals.ns<kt_globals.nspfr; kt_globals.ns++) |
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{ |
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/* Get low-passed random number for aspiration and frication noise */ |
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noise = gen_noise(noise); |
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@@ -266,22 +328,22 @@ if(option_log_frames) |
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Amplitude modulate noise (reduce noise amplitude during |
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second half of glottal period) if voicing simultaneously present. |
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*/ |
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if (kt_globals.nper > kt_globals.nmod) |
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if (kt_globals.nper > kt_globals.nmod) |
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{ |
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noise *= (double) 0.5; |
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} |
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/* Compute frication noise */ |
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frics = kt_globals.amp_frica * noise; |
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/* |
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Compute voicing waveform. Run glottal source simulation at 4 |
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times normal sample rate to minimize quantization noise in |
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Compute voicing waveform. Run glottal source simulation at 4 |
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times normal sample rate to minimize quantization noise in |
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period of female voice. |
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*/ |
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for (n4=0; n4<4; n4++) |
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for (n4=0; n4<4; n4++) |
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{ |
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switch(kt_globals.glsource) |
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{ |
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@@ -289,64 +351,67 @@ if(option_log_frames) |
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voice = impulsive_source(); |
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break; |
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case NATURAL: |
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voice = natural_source(); |
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voice = natural_source(); |
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break; |
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case SAMPLED: |
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voice = sampled_source(); |
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voice = sampled_source(0); |
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break; |
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case SAMPLED2: |
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voice = sampled_source(1); |
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break; |
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} |
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/* Reset period when counter 'nper' reaches T0 */ |
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if (kt_globals.nper >= kt_globals.T0) |
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if (kt_globals.nper >= kt_globals.T0) |
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{ |
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kt_globals.nper = 0; |
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pitch_synch_par_reset(frame); |
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} |
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/* |
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Low-pass filter voicing waveform before downsampling from 4*samrate |
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to samrate samples/sec. Resonator f=.09*samrate, bw=.06*samrate |
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to samrate samples/sec. Resonator f=.09*samrate, bw=.06*samrate |
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*/ |
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voice = resonator(&(kt_globals.rsn[RLP]),voice); |
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/* Increment counter that keeps track of 4*samrate samples per sec */ |
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kt_globals.nper++; |
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} |
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/* |
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Tilt spectrum of voicing source down by soft low-pass filtering, amount |
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of tilt determined by TLTdb |
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*/ |
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voice = (voice * kt_globals.onemd) + (vlast * kt_globals.decay); |
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vlast = voice; |
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/* |
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Add breathiness during glottal open phase. Amount of breathiness |
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determined by parameter Aturb Use nrand rather than noise because |
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noise is low-passed. |
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/* |
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Add breathiness during glottal open phase. Amount of breathiness |
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determined by parameter Aturb Use nrand rather than noise because |
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noise is low-passed. |
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*/ |
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if (kt_globals.nper < kt_globals.nopen) |
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if (kt_globals.nper < kt_globals.nopen) |
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{ |
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voice += kt_globals.amp_breth * kt_globals.nrand; |
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} |
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/* Set amplitude of voicing */ |
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glotout = kt_globals.amp_voice * voice; |
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par_glotout = kt_globals.par_amp_voice * voice; |
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/* Compute aspiration amplitude and add to voicing source */ |
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aspiration = kt_globals.amp_aspir * noise; |
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glotout += aspiration; |
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par_glotout += aspiration; |
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/* |
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/* |
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Cascade vocal tract, excited by laryngeal sources. |
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Nasal antiresonator, then formants FNP, F5, F4, F3, F2, F1 |
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Nasal antiresonator, then formants FNP, F5, F4, F3, F2, F1 |
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*/ |
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out=0; |
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@@ -363,20 +428,20 @@ if(option_log_frames) |
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casc_next_in = resonator2(&(kt_globals.rsn[R2c]),casc_next_in); |
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out = resonator2(&(kt_globals.rsn[R1c]),casc_next_in); |
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} |
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/* Excite parallel F1 and FNP by voicing waveform */ |
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sourc = par_glotout; /* Source is voicing plus aspiration */ |
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/* |
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Standard parallel vocal tract Formants F6,F5,F4,F3,F2, |
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outputs added with alternating sign. Sound source for other |
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parallel resonators is frication plus first difference of |
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voicing waveform. |
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Standard parallel vocal tract Formants F6,F5,F4,F3,F2, |
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outputs added with alternating sign. Sound source for other |
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parallel resonators is frication plus first difference of |
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voicing waveform. |
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*/ |
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out += resonator(&(kt_globals.rsn[R1p]),sourc); |
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out += resonator(&(kt_globals.rsn[Rnpp]),sourc); |
|
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|
sourc = frics + par_glotout - glotlast; |
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|
glotlast = par_glotout; |
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|
@@ -384,14 +449,14 @@ if(option_log_frames) |
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|
{ |
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|
out = resonator(&(kt_globals.rsn[ix]),sourc) - out; |
|
|
|
} |
|
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|
outbypas = kt_globals.amp_bypas * sourc; |
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|
out = outbypas - out; |
|
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|
#ifdef deleted |
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|
// for testing |
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|
if (kt_globals.outsl != 0) |
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|
if (kt_globals.outsl != 0) |
|
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|
{ |
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|
switch(kt_globals.outsl) |
|
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|
{ |
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|
@@ -401,7 +466,7 @@ if(option_log_frames) |
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|
case 2: |
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|
out = aspiration; |
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|
break; |
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|
case 3: |
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|
case 3: |
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|
out = frics; |
|
|
|
break; |
|
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|
case 4: |
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|
@@ -429,7 +494,7 @@ if(option_log_frames) |
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int z2; |
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|
signed char c; |
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int sample; |
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|
z2 = 0; |
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|
if(wdata.mix_wavefile_ix < wdata.n_mix_wavefile) |
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|
{ |
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|
@@ -466,7 +531,7 @@ if(option_log_frames) |
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|
{ |
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|
value = -32768; |
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|
} |
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if (value > 32767) |
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{ |
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|
value = 32767; |
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|
@@ -530,13 +595,13 @@ void KlattReset(int control) |
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|
} |
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|
/* |
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|
/* |
|
|
|
function FRAME_INIT |
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|
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|
Use parameters from the input frame to set up resonator coefficients. |
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|
*/ |
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static void frame_init(klatt_frame_ptr frame) |
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|
static void frame_init(klatt_frame_ptr frame) |
|
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|
{ |
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|
double amp_par[7]; |
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static double amp_par_factor[7] = {0.6, 0.4, 0.15, 0.06, 0.04, 0.022, 0.03}; |
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@@ -544,13 +609,13 @@ static void frame_init(klatt_frame_ptr frame) |
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int ix; |
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kt_globals.original_f0 = frame->F0hz10 / 10; |
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frame->AVdb_tmp = frame->AVdb - 7; |
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|
if (frame->AVdb_tmp < 0) |
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|
{ |
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|
frame->AVdb_tmp = 0; |
|
|
|
} |
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|
kt_globals.amp_aspir = DBtoLIN(frame->ASP) * 0.05; |
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|
kt_globals.amp_frica = DBtoLIN(frame->AF) * 0.25; |
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|
kt_globals.par_amp_voice = DBtoLIN(frame->AVpdb); |
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|
@@ -563,12 +628,12 @@ static void frame_init(klatt_frame_ptr frame) |
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|
} |
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|
Gain0_tmp = frame->Gain0 - 3; |
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|
|
if (Gain0_tmp <= 0) |
|
|
|
if (Gain0_tmp <= 0) |
|
|
|
{ |
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|
|
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++) |
|
|
|
{ |
|
|
|
@@ -578,7 +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])); |
|
|
|
|
|
|
|
|
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|
|
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; |
|
|
|
@@ -592,19 +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])); |
|
|
|
|
|
|
|
|
|
|
|
} |
|
|
|
|
|
|
|
|
|
|
|
@@ -612,27 +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)); |
|
|
|
} |
|
|
|
|
|
|
|
@@ -645,20 +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); |
|
|
|
@@ -692,125 +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; |
|
|
|
@@ -819,20 +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; |
|
|
|
} |
|
|
|
@@ -844,7 +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. |
|
|
|
*/ |
|
|
|
|
|
|
|
@@ -853,18 +918,18 @@ static void setabc(long int f, long int bw, resonator_ptr rp) |
|
|
|
{ |
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double r; |
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double arg; |
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/* Let r = exp(-pi bw t) */ |
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arg = kt_globals.minus_pi_t * bw; |
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r = exp(arg); |
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/* Let c = -r**2 */ |
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rp->c = -(r * r); |
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/* Let b = r * 2*cos(2 pi f t) */ |
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arg = kt_globals.two_pi_t * f; |
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rp->b = r * cos(arg) * 2.0; |
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/* Let a = 1.0 - b - c */ |
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rp->a = 1.0 - rp->b - rp->c; |
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} |
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@@ -873,7 +938,7 @@ static void setabc(long int f, long int bw, resonator_ptr rp) |
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/* |
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function SETZEROABC |
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Convert formant freqencies and bandwidth into anti-resonator difference |
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Convert formant freqencies and bandwidth into anti-resonator difference |
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equation constants. |
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*/ |
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@@ -881,7 +946,7 @@ static void setzeroabc(long int f, long int bw, resonator_ptr rp) |
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{ |
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double r; |
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double arg; |
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f = -f; |
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//NOTE, changes made 30.09.2011 for Reece Dunn <[email protected]> |
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@@ -891,17 +956,18 @@ static void setzeroabc(long int f, long int bw, resonator_ptr rp) |
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/* Let r = exp(-pi bw t) */ |
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arg = kt_globals.minus_pi_t * bw; |
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r = exp(arg); |
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/* Let c = -r**2 */ |
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rp->c = -(r * r); |
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/* Let b = r * 2*cos(2 pi f t) */ |
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arg = kt_globals.two_pi_t * f; |
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rp->b = r * cos(arg) * 2.; |
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/* Let a = 1.0 - b - c */ |
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rp->a = 1.0 - rp->b - rp->c; |
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/* Now convert to antiresonator coefficients (a'=1/a, b'=b/a, c'=c/a) */ |
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/* If f == 0 then rp->a gets set to 0 which makes a'=1/a set a', b' and c' to |
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* INF, causing an audible sound spike when triggered (e.g. apiration with the |
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* nasal register set to f=0, bw=0). |
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@@ -916,26 +982,26 @@ static void setzeroabc(long int f, long int bw, resonator_ptr rp) |
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} |
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/* |
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/* |
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function GEN_NOISE |
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Random number generator (return a number between -8191 and +8191) |
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Noise spectrum is tilted down by soft low-pass filter having a pole near |
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the origin in the z-plane, i.e. output = input + (0.75 * lastoutput) |
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Random number generator (return a number between -8191 and +8191) |
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Noise spectrum is tilted down by soft low-pass filter having a pole near |
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the origin in the z-plane, i.e. output = input + (0.75 * lastoutput) |
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*/ |
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static double gen_noise(double noise) |
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static double gen_noise(double noise) |
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{ |
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long temp; |
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static double nlast; |
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temp = (long) getrandom(-8191,8191); |
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kt_globals.nrand = (long) temp; |
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noise = kt_globals.nrand + (0.75 * nlast); |
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nlast = noise; |
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return(noise); |
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} |
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@@ -952,16 +1018,16 @@ Conversion table, db to linear, 87 dB --> 32767 |
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81 dB --> 16384 (6 dB down = 0.5) |
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... |
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0 dB --> 0 |
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The just noticeable difference for a change in intensity of a vowel |
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is approximately 1 dB. Thus all amplitudes are quantized to 1 dB |
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steps. |
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*/ |
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static double DBtoLIN(long dB) |
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static double DBtoLIN(long dB) |
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{ |
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static short amptable[88] = |
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static short amptable[88] = |
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{ |
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0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 7, |
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8, 9, 10, 11, 13, 14, 16, 18, 20, 22, 25, 28, 32, |
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@@ -972,12 +1038,12 @@ static double DBtoLIN(long dB) |
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4096, 4547, 5104, 5751, 6488, 7291, 8192, 9093, 10207, |
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11502, 12976, 14582, 16384, 18350, 20644, 23429, |
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26214, 29491, 32767 }; |
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if ((dB < 0) || (dB > 87)) |
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{ |
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return(0); |
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} |
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return((double)(amptable[dB]) * 0.001); |
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} |
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@@ -992,7 +1058,6 @@ static int klattp[N_KLATTP]; |
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static double klattp1[N_KLATTP]; |
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static double klattp_inc[N_KLATTP]; |
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static int scale_wav_tab[] = {45,38,45,45}; // scale output from different voicing sources |
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@@ -1126,7 +1191,7 @@ void SetSynth_Klatt(int length, int modn, frame_t *fr1, frame_t *fr2, voice_t *v |
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if(wvoice != NULL) |
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{ |
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if((wvoice->klattv[0] > 0) && (wvoice->klattv[0] <=3 )) |
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if((wvoice->klattv[0] > 0) && (wvoice->klattv[0] <=4 )) |
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{ |
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kt_globals.glsource = wvoice->klattv[0]; |
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kt_globals.scale_wav = scale_wav_tab[kt_globals.glsource]; |
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@@ -1146,7 +1211,7 @@ void SetSynth_Klatt(int length, int modn, frame_t *fr1, frame_t *fr2, voice_t *v |
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{ |
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if(qix >= N_WCMDQ) qix = 0; |
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if(qix == wcmdq_tail) break; |
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cmd = wcmdq[qix][0]; |
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if(cmd==WCMD_KLATT) |
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{ |
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@@ -1187,7 +1252,7 @@ if(option_log_frames) |
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fprintf(f_log,"K %3dmS %3d %3d %4d %4d (%2d) to %3d %3d %4d %4d (%2d)\n",length*1000/samplerate, |
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fr1->klattp[KLATT_FNZ]*2,fr1->ffreq[1],fr1->ffreq[2],fr1->ffreq[3], fr1->klattp[KLATT_AV], |
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fr2->klattp[KLATT_FNZ]*2,fr2->ffreq[1],fr2->ffreq[2],fr2->ffreq[3], fr2->klattp[KLATT_AV] ); |
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fclose(f_log); |
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} |
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} |
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@@ -1292,18 +1357,7 @@ int Wavegen_Klatt2(int length, int modulation, int resume, frame_t *fr1, frame_t |
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void KlattInit() |
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{ |
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#define NUMBER_OF_SAMPLES 100 |
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static short natural_samples[NUMBER_OF_SAMPLES]= |
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{ |
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-310,-400,530,356,224,89,23,-10,-58,-16,461,599,536,701,770, |
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605,497,461,560,404,110,224,131,104,-97,155,278,-154,-1165, |
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-598,737,125,-592,41,11,-247,-10,65,92,80,-304,71,167,-1,122, |
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233,161,-43,278,479,485,407,266,650,134,80,236,68,260,269,179, |
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53,140,275,293,296,104,257,152,311,182,263,245,125,314,140,44, |
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203,230,-235,-286,23,107,92,-91,38,464,443,176,98,-784,-2449, |
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-1891,-1045,-1600,-1462,-1384,-1261,-949,-730 |
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}; |
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static short formant_hz[10] = {280,688,1064,2806,3260,3700,6500,7000,8000,280}; |
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static short bandwidth[10] = {89,160,70,160,200,200,500,500,500,89}; |
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static short parallel_amp[10] = { 0,59,59,59,59,59,59,0,0,0}; |
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@@ -1311,6 +1365,12 @@ void KlattInit() |
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int ix; |
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for(ix=0; ix<256; ix++) |
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{ |
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// TEST: Overwrite natural_samples2 |
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// sawtooth wave |
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// natural_samples2[ix] = (128-ix) * 20; |
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} |
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sample_count=0; |
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kt_globals.synthesis_model = CASCADE_PARALLEL; |
Reece H. Dunn
12 years ago