static autoSound synthesize_pulses_lpc (Manipulation me) {
try {
if (! my lpc) {
if (! my sound) Melder_throw (U"Missing original sound.");
autoSound sound10k = Sound_resample (my sound.get(), 10000.0, 50);
my lpc = Sound_to_LPC_burg (sound10k.get(), 20, 0.025, 0.01, 50.0);
}
if (! my pulses) Melder_throw (U"Missing pulses analysis.");
autoSound train = PointProcess_to_Sound_pulseTrain (my pulses.get(), 1.0 / my lpc -> samplingPeriod, 0.7, 0.05, 30);
train -> dx = my lpc -> samplingPeriod; // to be exact
Sound_PointProcess_fillVoiceless (train.get(), my pulses.get());
autoSound result = LPC_and_Sound_filter (my lpc.get(), train.get(), true);
NUMdeemphasize_f (result -> z [1], result -> nx, result -> dx, 50.0);
Vector_scale (result.get(), 0.99);
return result;
} catch (MelderError) {
Melder_throw (me, U": LPC synthesis not performed.");
}
}
void Sound_filterWithFormants (Sound me, double tmin, double tmax,
int numberOfFormants, double formant [], double bandwidth [])
{
try {
for (long channel = 1; channel <= my ny; channel ++) {
if (tmax <= tmin) { tmin = my xmin; tmax = my xmax; } // autowindowing
long itmin, itmax;
long n = Sampled_getWindowSamples (me, tmin, tmax, & itmin, & itmax);
if (n <= 2)
Melder_throw (U"Sound too short.");
double *amplitude = my z [channel] + itmin - 1; // base 1
NUMdeemphasize_f (amplitude, n, my dx, 50.0);
for (int iformant = 1; iformant <= numberOfFormants; iformant ++) {
NUMfilterSecondOrderSection_fb (amplitude, n, my dx, formant [iformant], bandwidth [iformant]);
}
}
Matrix_scaleAbsoluteExtremum (me, 0.99);
} catch (MelderError) {
Melder_throw (me, U": not filtered.");
}
}
