예제 #1
0
/*
	Analog formant filter response :
	H(f) = i f B / (f1^2 - f^2 + i f B)
*/
static int Sound_into_FormantFilter_frame (Sound me, FormantFilter thee, long frame, double bw) {
	Melder_assert (bw > 0);
	autoMatrix pv = Sound_to_spectralpower (me);
	double z1 = pv -> x1;
	double dz = pv -> dx;
	long nf = pv -> nx;

	for (long i = 1; i <= thy ny; i++) {
		double p = 0;
		double fc = thy y1 + (i - 1) * thy dy;
		double *pow = pv -> z[1];
		for (long j = 1; j <= nf; j++) {
			// H(f) = ifB / (fc^2 - f^2 + ifB)
			// H(f)| = fB / sqrt ((fc^2 - f^2)^2 + f^2B^2)
			//|H(f)|^2 = f^2B^2 / ((fc^2 - f^2)^2 + f^2B^2)
			//         = 1 / (((fc^2 - f^2) /fB)^2 + 1)

			double f = z1 + (j - 1) * dz;
			double a = NUMformantfilter_amplitude (fc, bw, f);
			p += a * pow[j];
		}
		thy z[i][frame] = p;
	}
	return 1;
}
static int Sound_into_BarkFilter_frame (Sound me, BarkFilter thee, long frame) {
	autoMatrix pv = Sound_to_spectralpower (me);
	long nf = pv -> nx;
	autoNUMvector<double> z (1, nf);

	for (long j = 1; j <= nf; j++) {
		z[j] = HZTOBARK (pv -> x1 + (j - 1) * pv -> dx);
	}

	for (long i = 1; i <= thy ny; i++) {
		double p = 0;
		double z0 = thy y1 + (i - 1) * thy dy;
		double *pow = pv -> z[1]; // TODO ??
		for (long j = 1; j <= nf; j++) {
			// Sekey & Hanson filter is defined in the power domain.
			// We therefore multiply the power with a (and not a^2).
			// integral (F(z),z=0..25) = 1.58/9

			double a = NUMsekeyhansonfilter_amplitude (z0, z[j]);
			p += a * pow[j] ;
		}
		thy z[i][frame] = p;
	}
	return 1;
}
예제 #3
0
static int Sound_into_MelFilter_frame (Sound me, MelFilter thee, long frame) {
	autoMatrix pv = Sound_to_spectralpower (me);

	double z1 = pv -> x1;
	double dz = pv -> dx;
	long nf = pv -> nx;
	double df = thy dy;
	for (long i = 1; i <= thy ny; i++) {
		double p = 0;
		double fc_mel = thy y1 + (i - 1) * df;
		double fc_hz = MELTOHZ (fc_mel);
		double fl_hz = MELTOHZ (fc_mel - df);
		double fh_hz =  MELTOHZ (fc_mel + df);
		double *pow = pv -> z[1];
		for (long j = 1; j <= nf; j++) {
			// Filter with a triangular filter the power (=amplitude-squared)

			double f = z1 + (j - 1) * dz;
			double a = NUMtriangularfilter_amplitude (fl_hz, fc_hz, fh_hz, f);
			p += a * pow[j] ;
		}
		thy z[i][frame] = p;
	}
	return 1;
}