/* Sound_and_FilterBank.cpp
 *
 * Copyright (C) 1993-2012 David Weenink
 *
 * This program is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or (at
 * your option) any later version.
 *
 * This program is distributed in the hope that it will be useful, but
 * WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with this program; if not, write to the Free Software
 * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
 */

/*
 djmw 20010718
 djmw 20020813 GPL header.
 djmw 20041124 Changed call to Sound_to_Spectrum.
 djmw 20070103 Sound interface changes
 djmw 20071107 Errors/warnings text changes
 djmw 20071202 Melder_warning<n>
*/

#include "Sound_and_FilterBank.h"
#include "Sound_extensions.h"
#include "Sound_and_Spectrum.h"
#include "Sound_to_Pitch.h"
#include "Vector.h"
#include "NUM2.h"

#define MIN(m,n) ((m) < (n) ? (m) : (n))
// prototypes
Sound FilterBank_as_Sound (FilterBank me);

/*
	The gaussian(x) = (exp(-48*((i-(n+1)/2)/(n+1))^2)-exp(-12))/(1-exp(-12));
	For power we need thearea under the square of this window:
		Integrate (gaussian(i)^2,i=1..n) =

	(sqrt(Pi)*sqrt(3)*sqrt(2)*erf(2*(n-1)*sqrt(3)*sqrt(2)/(n+1))*(n+1)+
		24*exp(-24)*(n-1)+
	-4*sqrt(Pi)*sqrt(3)*exp(-12)*erf(2*(n-1)*sqrt(3)/(n+1))*(n+1))/
	(24 * (-1+exp(-12))^2)
	To compare with the rectangular window we need to divide this by the
	window width (n-1) x 1^2.
*/
static double gaussian_window_squared_correction (long n) {
	double e12 = exp (-12), denum = (e12 - 1) * (e12 - 1) * 24 * (n - 1);
	double sqrt3 = sqrt (3), sqrt2 = sqrt (2), sqrtpi = sqrt (NUMpi);
	double arg1 = 2 * sqrt3 * (n - 1) / (n + 1), arg2 = arg1 * sqrt2;
	double p2 = sqrtpi * sqrt3 * sqrt2 * (1 - NUMerfcc (arg2)) * (n + 1);
	double p1 = 4 * sqrtpi * sqrt3 * e12 * (1 - NUMerfcc (arg1)) * (n + 1);

	return (p2 - p1 + 24 * (n - 1) * e12 * e12) / denum;
}

static Matrix Sound_to_spectralpower (Sound me) {
	try {
		autoSpectrum s = Sound_to_Spectrum (me, TRUE);
		autoMatrix thee = Matrix_create (s -> xmin, s -> xmax, s -> nx, s -> dx, s -> x1, 1, 1, 1, 1, 1);
		double scale = 2.0 * s -> dx / (my xmax - my xmin);

		// factor '2' because of positive and negative frequencies
		// s -> dx : width of frequncy bin
		// my xmax - my xmin : duration of sound

		double *z = thy z[1], *re = s -> z[1], *im = s -> z[2];
		for (long i = 1; i <= s -> nx; i++) {
			z[i] = scale * (re[i] * re[i] + im[i] * im [i]);
		}

		// Frequency bins at 0 Hz and nyquist don't count for two.

		z[1] *= 0.5;
		z[s -> nx] *= 0.5;
		return thee.transfer();
	} catch (MelderError) {
		Melder_throw (me, ": no Matrix with spectral power created.");
	}
}

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;
}

BarkFilter Sound_to_BarkFilter (Sound me, double analysisWidth, double dt,
                                double f1_bark, double fmax_bark, double df_bark) {
	try {
		double t1, nyquist = 0.5 / my dx, samplingFrequency = 2 * nyquist;
		double windowDuration = 2 * analysisWidth; /* gaussian window */
		double zmax = NUMhertzToBark2 (nyquist);
		double fmin_bark = 0;
		long nt, frameErrorCount = 0;

		// Check defaults.

		if (f1_bark <= 0) {
			f1_bark = 1;
		}
		if (fmax_bark <= 0) {
			fmax_bark = zmax;
		}
		if (df_bark <= 0) {
			df_bark = 1;
		}

		fmax_bark = MIN (fmax_bark, zmax);
		long nf = floor ( (fmax_bark - f1_bark) / df_bark + 0.5);
		if (nf <= 0) {
			Melder_throw ("The combination of filter parameters is not valid.");
		}

		Sampled_shortTermAnalysis (me, windowDuration, dt, & nt, & t1);
		autoSound sframe = Sound_createSimple (1, windowDuration, samplingFrequency);
		autoSound window = Sound_createGaussian (windowDuration, samplingFrequency);
		autoBarkFilter thee = BarkFilter_create (my xmin, my xmax, nt, dt, t1,
		                      fmin_bark, fmax_bark, nf, df_bark, f1_bark);

		autoMelderProgress progess (L"BarkFilter analysis");

		for (long i = 1; i <= nt; i++) {
			double t = Sampled_indexToX (thee.peek(), i);

			Sound_into_Sound (me, sframe.peek(), t - windowDuration / 2);

			Sounds_multiply (sframe.peek(), window.peek());

			if (! Sound_into_BarkFilter_frame (sframe.peek(), thee.peek(), i)) {
				frameErrorCount++;
			}

			if ( (i % 10) == 1) {
				Melder_progress ( (double) i / nt,  L"BarkFilter analysis: frame ",
				                   Melder_integer (i), L" from ", Melder_integer (nt), L"."); therror
			}
		}

		if (frameErrorCount > 0) {
			Melder_warning (L"Analysis results of ", Melder_integer (frameErrorCount), L" frame(s) out of ",
			                Melder_integer (nt), L" will be suspect.");
		}

		double ref = FilterBank_DBREF * gaussian_window_squared_correction (window -> nx);

		NUMdmatrix_to_dBs (thy z, 1, thy ny, 1, thy nx, ref, FilterBank_DBFAC, FilterBank_DBFLOOR);
		return thee.transfer();
	} catch (MelderError) {
		Melder_throw (me, ": no BarkFilter created.");
	}
}


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;
}

MelFilter Sound_to_MelFilter (Sound me, double analysisWidth, double dt,
                              double f1_mel, double fmax_mel, double df_mel) {
	try {
		double t1, samplingFrequency = 1 / my dx, nyquist = 0.5 * samplingFrequency;
		double windowDuration = 2 * analysisWidth; /* gaussian window */
		double fmin_mel = 0;
		double fbottom = HZTOMEL (100.0), fceiling = HZTOMEL (nyquist);
		long nt, frameErrorCount = 0;

		// Check defaults.

		if (fmax_mel <= 0 || fmax_mel > fceiling) {
			fmax_mel = fceiling;
		}
		if (fmax_mel <= f1_mel) {
			f1_mel = fbottom; fmax_mel = fceiling;
		}
		if (f1_mel <= 0) {
			f1_mel = fbottom;
		}
		if (df_mel <= 0) {
			df_mel = 100.0;
		}

		// Determine the number of filters.

		long nf = floor ( (fmax_mel - f1_mel) / df_mel + 0.5);
		fmax_mel = f1_mel + nf * df_mel;

		Sampled_shortTermAnalysis (me, windowDuration, dt, &nt, &t1);
		autoSound sframe = Sound_createSimple (1, windowDuration, samplingFrequency);
		autoSound window = Sound_createGaussian (windowDuration, samplingFrequency);
		autoMelFilter thee = MelFilter_create (my xmin, my xmax, nt, dt, t1, fmin_mel,
		                                       fmax_mel, nf, df_mel, f1_mel);

		autoMelderProgress progress (L"MelFilters analysis");

		for (long i = 1; i <= nt; i++) {
			double t = Sampled_indexToX (thee.peek(), i);
			Sound_into_Sound (me, sframe.peek(), t - windowDuration / 2);
			Sounds_multiply (sframe.peek(), window.peek());
			if (! Sound_into_MelFilter_frame (sframe.peek(), thee.peek(), i)) {
				frameErrorCount++;
			}
			if ( (i % 10) == 1) {
				Melder_progress ( (double) i / nt, L"Frame ", Melder_integer (i), L" out of ",
				                   Melder_integer (nt), L"."); therror
			}
		}

		if (frameErrorCount) Melder_warning (L"Analysis results of ", Melder_integer (frameErrorCount),
			                                     L" frame(s) out of ", Melder_integer (nt), L" will be suspect.");

		// Window correction.

		double ref = FilterBank_DBREF * gaussian_window_squared_correction (window -> nx);

		NUMdmatrix_to_dBs (thy z, 1, thy ny, 1, thy nx, ref, FilterBank_DBFAC, FilterBank_DBFLOOR);
		return thee.transfer();
	} catch (MelderError) {
		Melder_throw (me, ": no MelFilter created.");
	}
}

/*
	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;
}

FormantFilter Sound_to_FormantFilter (Sound me, double analysisWidth,
                                      double dt, double f1_hz, double fmax_hz, double df_hz, double relative_bw,
                                      double minimumPitch, double maximumPitch) {
	try {
		double floor = 80, ceiling = 600;
		if (minimumPitch >= maximumPitch) {
			minimumPitch = floor; maximumPitch = ceiling;
		}
		if (minimumPitch <= 0) {
			minimumPitch = floor;
		}
		if (maximumPitch <= 0) {
			maximumPitch = ceiling;
		}

		autoPitch thee = Sound_to_Pitch (me, dt, minimumPitch, maximumPitch);
		autoFormantFilter ff = Sound_and_Pitch_to_FormantFilter (me, thee.peek(), analysisWidth, dt,
		                       f1_hz, fmax_hz, df_hz, relative_bw);
		return ff.transfer();
	} catch (MelderError) {
		Melder_throw (me, ": no FormantFilter created.");
	}
}

FormantFilter Sound_and_Pitch_to_FormantFilter (Sound me, Pitch thee, double analysisWidth, double dt,
        double f1_hz, double fmax_hz, double df_hz, double relative_bw) {
	try {
		double t1, windowDuration = 2 * analysisWidth; /* gaussian window */
		double nyquist = 0.5 / my dx, samplingFrequency = 2 * nyquist, fmin_hz = 0;
		long nt, f0_undefined = 0;

		if (my xmin > thy xmin || my xmax > thy xmax) Melder_throw
			("The domain of the Sound is not included in the domain of the Pitch.");

		double f0_median = Pitch_getQuantile (thee, thy xmin, thy xmax, 0.5, kPitch_unit_HERTZ);

		if (f0_median == NUMundefined || f0_median == 0) {
			f0_median = 100;
			Melder_warning (L"Pitch values undefined. Bandwith fixed to 100 Hz. ");
		}

		if (f1_hz <= 0) {
			f1_hz = 100;
		}
		if (fmax_hz <= 0) {
			fmax_hz = nyquist;
		}
		if (df_hz <= 0) {
			df_hz = f0_median / 2;
		}
		if (relative_bw <= 0) {
			relative_bw = 1.1;
		}

		fmax_hz = MIN (fmax_hz, nyquist);
		long nf = floor ( (fmax_hz - f1_hz) / df_hz + 0.5);

		Sampled_shortTermAnalysis (me, windowDuration, dt, &nt, &t1);
		autoFormantFilter him = FormantFilter_create (my xmin, my xmax, nt, dt, t1,
		                        fmin_hz, fmax_hz, nf, df_hz, f1_hz);

		// Temporary objects

		autoSound sframe = Sound_createSimple (1, windowDuration, samplingFrequency);
		autoSound window = Sound_createGaussian (windowDuration, samplingFrequency);
		autoMelderProgress progress (L"Sound & Pitch: To FormantFilter");
		for (long i = 1; i <= nt; i++) {
			double t = Sampled_indexToX (him.peek(), i);
			double b, f0 = Pitch_getValueAtTime (thee, t, kPitch_unit_HERTZ, 0);

			if (f0 == NUMundefined || f0 == 0) {
				f0_undefined++; f0 = f0_median;
			}
			b = relative_bw * f0;
			Sound_into_Sound (me, sframe.peek(), t - windowDuration / 2);
			Sounds_multiply (sframe.peek(), window.peek());

			Sound_into_FormantFilter_frame (sframe.peek(), him.peek(), i, b);

			if ( (i % 10) == 1) {
				Melder_progress ( (double) i / nt, L"Frame ", Melder_integer (i), L" out of ",
				                   Melder_integer (nt), L"."); therror
			}
		}

		double ref = FilterBank_DBREF * gaussian_window_squared_correction (window -> nx);
		NUMdmatrix_to_dBs (his z, 1, his ny, 1, his nx, ref, FilterBank_DBFAC, FilterBank_DBFLOOR);
		return him.transfer();
	} catch (MelderError) {
		Melder_throw ("FormantFilter not created from Pitch & FormantFilter.");
	}
}

Sound FilterBank_as_Sound (FilterBank me) {
	try {
		autoSound thee = Sound_create (my ny, my xmin, my xmax, my nx, my dx, my x1);
		for (long i = 1; i <= my ny; i++) {
			for (long j = 1; j <= my nx; j++)
				thy z[i][j] = my z[i][j];
		}
		return thee.transfer();
	} catch (MelderError) {
		Melder_throw (me, ": no Sound created.");
	}
}

Sound FilterBanks_crossCorrelate (FilterBank me, FilterBank thee, enum kSounds_convolve_scaling scaling, enum kSounds_convolve_signalOutsideTimeDomain signalOutsideTimeDomain) {
	try {
		autoSound cc = Sounds_crossCorrelate ((Sound) me, (Sound) thee, scaling, signalOutsideTimeDomain);
		return cc.transfer();
	} catch (MelderError) {
		Melder_throw (me, " and ", thee, " not cross-correlated.");
	}
}

Sound FilterBanks_convolve (FilterBank me, FilterBank thee, enum kSounds_convolve_scaling scaling, enum kSounds_convolve_signalOutsideTimeDomain signalOutsideTimeDomain) {
	try {
		autoSound cc = Sounds_convolve ((Sound) me, (Sound) thee, scaling, signalOutsideTimeDomain);
		return cc.transfer();
	} catch (MelderError) {
		Melder_throw (me, " and ", thee, " not convolved.");
	}
}

/* End of file Sound_and_FilterBank.cpp */
