/*

    TiMidity -- Experimental MIDI to WAVE converter
    Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>

    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.

   filter.c: written by Vincent Pagel ( pagel@loria.fr ) 
 
   implements fir antialiasing filter : should help when setting sample
   rates as low as 8Khz.

   April 95
      - first draft

   22/5/95
      - modify "filter" so that it simulate leading and trailing 0 in the buffer
   */

#include "config.h"
#include "common.h"
#include "instrum.h"
#include "filter.h"

/*  bessel  function   */
static float
ino (float x)
{
	float y, de, e, sde;
	int i;
	
	y = x / 2;
	e = 1.0;
	de = 1.0;
	i = 1;
	do
    {
		de = de * y / (float) i;
		sde = de * de;
		e += sde;
    }
	while (!((e * 1.0e-08 - sde > 0) || (i++ > 25)));
	return (e);
}

/* Kaiser Window (symetric) */
static void
kaiser (float *w, int n, float beta)
{
	float xind, xi;
	int i;
	
	// KSK Optimize here? xind = (2*n - 1) * (2*n - 1);
	xind = (float) ((2 * n - 1) * (2 * n - 1));
	for (i = 0; i < n; i++)
    {
		xi = (float) i + (float) 0.5;
		w[i] = ino ((float) (beta * sqrt ((double) (1. - 4 * xi * xi / xind))))
			/ ino ((float) beta);
    }
}

/*
* fir coef in g, cuttoff frequency in fc
*/
static void
designfir (float *g, float fc)
{
	int i;
	float xi, omega, att, beta;
	float w[ORDER2];
	
	for (i = 0; i < ORDER2; i++)
    {
		xi = (float) i + (float) 0.5;
		omega = (float) PI *xi;
		g[i] = (float) (sin ((double) omega * fc) / omega);
    }
	
	att = 40.;			/* attenuation  in  db */
	beta = (float) (exp (log ((double) 0.58417 * (att - 20.96)) * 0.4) + 0.07886
		* (att - 20.96));
	kaiser (w, ORDER2, beta);
	
	/* Matrix product */
	for (i = 0; i < ORDER2; i++)
		g[i] = g[i] * w[i];
}

/*
* FIR filtering -> apply the filter given by coef[] to the data buffer
* Note that we simulate leading and trailing 0 at the border of the 
* data buffer
*/
static void
filter (sample_t * result, sample_t * data, int32b length, float coef[])
{
	int32b sample, i, sample_window;
	int16b peak = 0;
	float sum;
	
	/* Simulate leading 0 at the begining of the buffer */
	for (sample = 0; sample < ORDER2; sample++)
    {
		sum = 0.0;
		sample_window = sample - ORDER2;
		
		for (i = 0; i < ORDER; i++)
			sum += (float) (coef[i] *
			((sample_window < 0) ? 0.0 : data[sample_window++]));
		
		/* Saturation ??? */
		if (sum > 32767.)
		{
			sum = 32767.;
			peak++;
		}
		if (sum < -32768.)
		{
			sum = -32768;
			peak++;
		}
		result[sample] = (sample_t) sum;
    }
	
	/* The core of the buffer  */
	for (sample = ORDER2; sample < length - ORDER + ORDER2; sample++)
    {
		sum = 0.0;
		sample_window = sample - ORDER2;
		
		for (i = 0; i < ORDER; i++)
			sum += data[sample_window++] * coef[i];
		
		/* Saturation ??? */
		if (sum > 32767.)
		{
			sum = 32767.;
			peak++;
		}
		if (sum < -32768.)
		{
			sum = -32768;
			peak++;
		}
		result[sample] = (sample_t) sum;
    }
	
	/* Simulate 0 at the end of the buffer */
	for (sample = length - ORDER + ORDER2; sample < length; sample++)
    {
		sum = 0.0;
		sample_window = sample - ORDER2;
		
		for (i = 0; i < ORDER; i++)
			sum += (float) (coef[i] *
			((sample_window >= length) ? 0.0 :
		data[sample_window++]));
		
		/* Saturation ??? */
		if (sum > 32767.)
		{
			sum = 32767.;
			peak++;
		}
		if (sum < -32768.)
		{
			sum = -32768;
			peak++;
		}
		result[sample] = (sample_t) sum;
    }
	
	/* Ignore this for now
	if (peak)
		ctl.cmsg (CMSG_ERROR, VERB_NORMAL,
	       TEXT ("Saturation %2.3f %%."), 100.0 * peak / (float) length);
		   */
}

/***********************************************************************/
/* Prevent aliasing by filtering any freq above the output_rate        */
/*                                                                     */
/* I don't worry about looping point -> they will remain soft if they  */
/* were already                                                        */
/***********************************************************************/
void
antialiasing (Sample * sp, int32b output_rate)
{
	sample_t *temp;
	int i;
	float fir_symetric[ORDER];
	float fir_coef[ORDER2];
	float freq_cut;		/* cutoff frequency [0..1.0] FREQ_CUT/SAMP_FREQ */
	
	
	/* No oversampling  */
	if (output_rate >= sp->sample_rate)
		return;
	
	freq_cut = (float) output_rate / (float) sp->sample_rate;
	
	designfir (fir_coef, freq_cut);
	
	/* Make the filter symetric */
	for (i = 0; i < ORDER2; i++)
		fir_symetric[ORDER - 1 - i] = fir_symetric[i] = fir_coef[ORDER2 - 1 - i];
	
	/* We apply the filter we have designed on a copy of the patch */
	temp = (sample_t *)safe_malloc (sp->data_length);
	memcpy (temp, sp->data, sp->data_length);
	
	filter (sp->data, temp, sp->data_length / sizeof (sample_t), fir_symetric);
	
	free (temp);
}
