/*
	ToneStack.h
	
	Copyright 2006-9
		David Yeh <dtyeh@ccrma.stanford.edu> (implementation)
		Tim Goetze <tim@quitte.de> (cosmetics)

	Tone Stack emulation.

*/
/*
	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., 59 Temple Place - Suite 330, Boston, MA
	02111-1307, USA or point your web browser to http://www.gnu.org.
*/

#ifndef _DSP_TONESTACK_H_
#define _DSP_TONESTACK_H_

#include "util.h"
#include "windows.h"
#include "TDFII.h"
#include "LatFilt.h"
#include "tonestack/tables.h"

namespace DSP {

typedef struct {
	double R1, R2, R3, R4;
	double C1, C2, C3;
} TSParameters;

#define TS_N_PRESETS (sizeof (DSP::ToneStack::presets) / \
				sizeof (DSP::TSParameters))

class ToneStack 
{
  private:
		enum { Order = 3 };

		double c;   // BT coef

		double b1t, b1m, b1l, b1d,
				b2t, b2m2, b2m, b2l, b2lm, b2d,
				b3lm, b3m2, b3m, b3t, b3tm, b3tl,
				a0, a1d, a1m, a1l, a2m, a2lm, a2m2, a2l, a2d,
				a3lm, a3m2, a3m, a3l, a3d;      // intermediate calculations

		struct {
				double b1;
				double b2;
				double b3;
				double a1;
				double a2;
				double a3;
		} acoef; // analog coefficients

		// digital coefficients
		double dcoef_a[Order + 1];
		double dcoef_b[Order + 1];

		TDFII<Order> filter;
		
	public:
		int model;

		static TSParameters presets[]; /* in ../ToneStack.cc */
		static int n_presets;

		ToneStack() 
			{
				model = -1;
				setparams (presets[0]);
			}
		
		void init (double _fs)
			{
				c = 2 * _fs;
			}

		void activate (sample_t ** ports)
			{
				filter.reset();
			}

		/* pass in pointer to ports and relative index of first eq band control */
		void start_cycle (sample_t ** ports, int bassindex = 1)
		{
			int m = clamp<int> ((int) *ports[0], 0, n_presets - 1);
			if (m != model)
			{
				model = m;
				setparams (presets[model]);
				filter.reset();
			}
			updatecoefs (ports + bassindex);
		}


		void setparams (TSParameters & p)
			{
				double R1 = p.R1, R2 = p.R2, R3 = p.R3, R4 = p.R4;
				double C1 = p.C1, C2 = p.C2, C3 = p.C3;

				b1t  = C1*R1;
				b1m  = C3*R3;
				b1l  = C1*R2 + C2*R2;
				b1d  = C1*R3 + C2*R3;
				b2t  = C1*C2*R1*R4 + C1*C3*R1*R4;
				b2m2 = -(C1*C3*R3*R3 + C2*C3*R3*R3);
				b2m  = C1*C3*R1*R3 + C1*C3*R3*R3 + C2*C3*R3*R3;
				b2l  = C1*C2*R1*R2 + C1*C2*R2*R4 + C1*C3*R2*R4;
				b2lm = C1*C3*R2*R3 + C2*C3*R2*R3;
				b2d  = C1*C2*R1*R3 + C1*C2*R3*R4 + C1*C3*R3*R4;
				b3lm = C1*C2*C3*R1*R2*R3 + C1*C2*C3*R2*R3*R4;
				b3m2 = -(C1*C2*C3*R1*R3*R3 + C1*C2*C3*R3*R3*R4);
				b3m  = C1*C2*C3*R1*R3*R3 + C1*C2*C3*R3*R3*R4;
				b3t  = C1*C2*C3*R1*R3*R4;
				b3tm = -b3t;
				b3tl = C1*C2*C3*R1*R2*R4;
				a0   = 1;
				a1d  = C1*R1 + C1*R3 + C2*R3 + C2*R4 + C3*R4;
				a1m  = C3*R3;
				a1l  = C1*R2 + C2*R2;
				a2m  = C1*C3*R1*R3 - C2*C3*R3*R4 + C1*C3*R3*R3 + C2*C3*R3*R3;
				a2lm = C1*C3*R2*R3 + C2*C3*R2*R3;
				a2m2 = -(C1*C3*R3*R3 + C2*C3*R3*R3);
				a2l  = C1*C2*R2*R4 + C1*C2*R1*R2 + C1*C3*R2*R4 + C2*C3*R2*R4;
				a2d  = C1*C2*R1*R4 + C1*C3*R1*R4 + C1*C2*R3*R4
						+ C1*C2*R1*R3 + C1*C3*R3*R4 + C2*C3*R3*R4;
				a3lm = C1*C2*C3*R1*R2*R3 + C1*C2*C3*R2*R3*R4;
				a3m2 = -(C1*C2*C3*R1*R3*R3 + C1*C2*C3*R3*R3*R4);
				a3m  = C1*C2*C3*R3*R3*R4 + C1*C2*C3*R1*R3*R3 - C1*C2*C3*R1*R3*R4;
				a3l  = C1*C2*C3*R1*R2*R4;
				a3d  = C1*C2*C3*R1*R3*R4;

				filter.reset();
			}

		inline void updatecoefs (sample_t ** ports) 
			{
				/* range checks on input */
				double b = clamp<double> (*ports[0], 0, 1);
				double m = clamp<double> (*ports[1], 0, 1);
				double t = clamp<double> (*ports[2], 0, 1);

				m = (m - 1) * 3.5;
				m = pow (10, m);
		
				acoef.a1 = a1d + m*a1m + b*a1l;
				acoef.a2 = m*a2m + b*m*a2lm + m*m*a2m2 + b*a2l + a2d;
				acoef.a3 = b*m*a3lm + m*m*a3m2 + m*a3m + b*a3l + a3d;
				dcoef_a[0] = -1 - acoef.a1*c - acoef.a2*c*c - acoef.a3*c*c*c; // sets scale
				dcoef_a[1] = -3 - acoef.a1*c + acoef.a2*c*c + 3*acoef.a3*c*c*c;
				dcoef_a[2] = -3 + acoef.a1*c + acoef.a2*c*c - 3*acoef.a3*c*c*c;
				dcoef_a[3] = -1 + acoef.a1*c - acoef.a2*c*c + acoef.a3*c*c*c;

				acoef.b1 = t*b1t + m*b1m + b*b1l + b1d;
				acoef.b2 = t*b2t + m*m*b2m2 + m*b2m + b*b2l + b*m*b2lm + b2d;
				acoef.b3 = b*m*b3lm + m*m*b3m2 + m*b3m + t*b3t + t*m*b3tm + t*b*b3tl;
				dcoef_b[0] = - acoef.b1*c - acoef.b2*c*c - acoef.b3*c*c*c;
				dcoef_b[1] = - acoef.b1*c + acoef.b2*c*c + 3*acoef.b3*c*c*c;
				dcoef_b[2] = acoef.b1*c + acoef.b2*c*c - 3*acoef.b3*c*c*c;
				dcoef_b[3] = acoef.b1*c - acoef.b2*c*c + acoef.b3*c*c*c;

				for (int i=1; i<=3; ++i)
					filter.a[i] = dcoef_a[i] / dcoef_a[0];
				
				for (int i=0; i<=3; ++i)
					filter.b[i] = dcoef_b[i] / dcoef_a[0];
			} 

		// actualy do the DFII filtering, one sample at a time
		inline sample_t process (sample_t x)
			{
				return filter.process (x);
			}
};

/* /////////////////////////////////////////////////////////////////////// */

/* 
	hardcode this, known size memory blocks
	extern double* KS;    // 25 x 25 x 3
	extern double* VS;    // 25 x 25 x 25 x 4
	extern double KS[NSTEPS][NSTEPS][TSORDER]; //[bass][mid][coefs]
	extern double VS[NSTEPS][NSTEPS][NSTEPS][TSORDER+1]; //[bass][mid][treb][coefs]
 */

class ToneStackLT 
{
  private:
		enum { Order = 3, Steps = 25 };

		// digital coefficients
		double *kcoef;
		double *vcoef;
		LatFilt<Order> filter;

	public:
		ToneStackLT() 
			{
				setparams (250000, 1000000, 25000, 56000, 0.25e-9, 20e-9, 20e-9);
			}

		void init (double _fs)
			{ }

		void activate (sample_t ** ports)
			{
				filter.reset();
			}
	
		void setparams
			(double R1, double R2, double R3, double R4,
							double C1, double C2, double C3) {
		}

		void updatecoefs (sample_t ** ports)
			{
				double b = min (Steps - 1, max (*ports[0] * (Steps - 1), 0));
				double m = min (Steps - 1, max (*ports[1] * (Steps - 1), 0));
				double t = min (Steps - 1, max (*ports[2] * (Steps - 1), 0));

				int bi = (int) b;
				int mi = (int) m;
				int ti = (int) t;

				kcoef = DSP::ToneStackKS +  (mi * Steps + bi) * Order;
				vcoef = DSP::ToneStackVS + ((mi * Steps + bi) * Steps + ti) * (Order + 1);
				
				for (int i = 0; i < Order; ++i) 
						filter.set_ki (kcoef[i], i);
						
				for (int i = 0; i < Order + 1; ++i) 
						filter.set_vi (vcoef[i], i);
			}

		// actualy do the DFII filtering, one sample at a time
		inline sample_t process (sample_t x)
			{
				return filter.process (x);
			}
};

} /* namespace DSP */

#endif /* _DSP_TONESTACK_H_ */
