/*
 * August 24, 1998
 * Copyright (C) 1998 Juergen Mueller And Sundry Contributors
 * This source code is freely redistributable and may be used for
 * any purpose.  This copyright notice must be maintained. 
 * Juergen Mueller And Sundry Contributors are not responsible for 
 * the consequences of using this software.
 */

/*
 * Echos effect for dsp.
 * 
 * Flow diagram scheme for n delays ( 1 <= n <= MAX_ECHOS ):
 *
 *                                                    * gain-in  ___
 * ibuff --+--------------------------------------------------->|   |
 *         |                                          * decay 1 |   |
 *         |               +----------------------------------->|   |
 *         |               |                          * decay 2 | + |
 *         |               |             +--------------------->|   |
 *         |               |             |            * decay n |   |
 *         |    _________  |  _________  |     _________   +--->|___|
 *         |   |         | | |         | |    |         |  |      | 
 *         +-->| delay 1 |-+-| delay 2 |-+...-| delay n |--+      | * gain-out
 *             |_________|   |_________|      |_________|         |
 *                                                                +----->obuff
 *
 * Usage: 
 *   echos gain-in gain-out delay-1 decay-1 [delay-2 decay-2 ... delay-n decay-n]
 *
 * Where:
 *   gain-in, decay-1 ... decay-n :  0.0 ... 1.0      volume
 *   gain-out :  0.0 ...      volume
 *   delay-1 ... delay-n :  > 0.0 msec
 *
 * Note:
 *   when decay is close to 1.0, the samples can begin clipping and the output
 *   can saturate! 
 *
 * Hint:
 *   1 / out-gain > gain-in ( 1 + decay-1 + ... + decay-n )
 *
*/

/*
 * Sound Tools reverb effect file.
 */

#include <stdlib.h> /* Harmless, and prototypes atof() etc. --dgc */
#ifdef HAVE_MALLOC_H
#include <malloc.h>
#endif
#include <math.h>
#include "st.h"

#define DELAY_BUFSIZ ( 50L * MAXRATE )
#define MAX_ECHOS 7	/* 24 bit x ( 1 + MAX_ECHOS ) = */
			/* 24 bit x 8 = 32 bit !!!	*/

/* Private data for SKEL file */
typedef struct echosstuff {
	int	counter[MAX_ECHOS];			
	int	num_delays;
	double	*delay_buf;
	float	in_gain, out_gain;
	float	delay[MAX_ECHOS], decay[MAX_ECHOS];
	long	samples[MAX_ECHOS], pointer[MAX_ECHOS], sumsamples;
} *echos_t;

/* Private data for SKEL file */


/* If we are not carefull with the output volume */
LONG echos_clip24(l)
LONG l;
{
	if (l >= ((LONG)1 << 24))
		return ((LONG)1 << 24) - 1;
	else if (l <= -((LONG)1 << 24))
		return -((LONG)1 << 24) + 1;
	else
		return l;
}



/*
 * Process options
 */
void echos_getopts(effp, n, argv) 
eff_t effp;
int n;
char **argv;
{
	echos_t echos = (echos_t) effp->priv;
	int i;

	echos->num_delays = 0;

	if ((n < 4) || (n % 2))
	    fail("Usage: echos gain-in gain-out delay decay [ delay decay ... ]");

	i = 0;
	sscanf(argv[i++], "%f", &echos->in_gain);
	sscanf(argv[i++], "%f", &echos->out_gain);
	while (i < n) {
		/* Linux bug and it's cleaner. */
		sscanf(argv[i++], "%f", &echos->delay[echos->num_delays]);
		sscanf(argv[i++], "%f", &echos->decay[echos->num_delays]);
		echos->num_delays++;
		if ( echos->num_delays > MAX_ECHOS )
			fail("echos: to many delays, use less than %i delays",
				MAX_ECHOS);
	}
	echos->sumsamples = 0;
}

/*
 * Prepare for processing.
 */
void echos_start(effp)
eff_t effp;
{
	echos_t echos = (echos_t) effp->priv;
	int i;
	float sum_in_volume;
	long j;

	if ( echos->in_gain < 0.0 )
		fail("echos: gain-in must be positive!\n");
	if ( echos->in_gain > 1.0 )
		fail("echos: gain-in must be less than 1.0!\n");
	if ( echos->out_gain < 0.0 )
		fail("echos: gain-in must be positive!\n");
	for ( i = 0; i < echos->num_delays; i++ ) {
		echos->samples[i] = echos->delay[i] * effp->ininfo.rate / 1000.0;
		if ( echos->samples[i] < 1 )
		    fail("echos: delay must be positive!\n");
		if ( echos->samples[i] > DELAY_BUFSIZ )
			fail("echos: delay must be less than %g seconds!\n",
				DELAY_BUFSIZ / (float) effp->ininfo.rate );
		if ( echos->decay[i] < 0.0 )
		    fail("echos: decay must be positive!\n" );
		if ( echos->decay[i] > 1.0 )
		    fail("echos: decay must be less than 1.0!\n" );
		echos->counter[i] = 0;
		echos->pointer[i] = echos->sumsamples;
		echos->sumsamples += echos->samples[i];
	}
	if (! (echos->delay_buf = (double *) malloc(sizeof (double) * echos->sumsamples)))
		fail("echos: Cannot malloc %d bytes!\n", 
			sizeof(long) * echos->sumsamples);
	for ( j = 0; j < echos->samples[i]; ++j )
		echos->delay_buf[j] = 0.0;
	/* Be nice and check the hint with warning, if... */
	sum_in_volume = 1.0;
	for ( i = 0; i < echos->num_delays; i++ ) 
		sum_in_volume += echos->decay[i];
	if ( sum_in_volume * echos->in_gain > 1.0 / echos->out_gain )
		warn("echos: warning >>> gain-out can cause saturation of output <<<");
}

/*
 * Processed signed long samples from ibuf to obuf.
 * Return number of samples processed.
 */

void echos_flow(effp, ibuf, obuf, isamp, osamp)
eff_t effp;
LONG *ibuf, *obuf;
int *isamp, *osamp;
{
	echos_t echos = (echos_t) effp->priv;
	int len, done;
	int j;
	
	double d_in, d_out;
	LONG out;

	len = ((*isamp > *osamp) ? *osamp : *isamp);
	for(done = 0; done < len; done++) {
		/* Store delays as 24-bit signed longs */
		d_in = (double) *ibuf++ / 256;
		/* Compute output first */
		d_out = d_in * echos->in_gain;
		for ( j = 0; j < echos->num_delays; j++ ) {
			d_out += echos->delay_buf[echos->counter[j] + echos->pointer[j]] * echos->decay[j];
		}
		/* Adjust the output volume and size to 24 bit */
		d_out = d_out * echos->out_gain;
		out = echos_clip24((LONG) d_out);
		*obuf++ = out * 256;
		/* Mix decay of delays and input */
		for ( j = 0; j < echos->num_delays; j++ ) {
			if ( j == 0 )
				echos->delay_buf[echos->counter[j] + echos->pointer[j]] = d_in;
			else
				echos->delay_buf[echos->counter[j] + echos->pointer[j]] = 
				   echos->delay_buf[echos->counter[j-1] + echos->pointer[j-1]] + d_in;
		}
		/* Adjust the counters */
		for ( j = 0; j < echos->num_delays; j++ )
			echos->counter[j] = 
			   ( echos->counter[j] + 1 ) % echos->samples[j];
	}
	/* processed all samples */
}

/*
 * Drain out reverb lines. 
 */
void echos_drain(effp, obuf, osamp)
eff_t effp;
LONG *obuf;
int *osamp;
{
	echos_t echos = (echos_t) effp->priv;
	double d_in, d_out;
	LONG out;
	int j;
	long done;

	done = 0;
	/* drain out delay samples */
	while ( ( done < *osamp ) && ( done < echos->sumsamples ) ) {
		d_in = 0;
		d_out = 0;
		for ( j = 0; j < echos->num_delays; j++ ) {
			d_out += echos->delay_buf[echos->counter[j] + echos->pointer[j]] * echos->decay[j];
		}
		/* Adjust the output volume and size to 24 bit */
		d_out = d_out * echos->out_gain;
		out = echos_clip24((LONG) d_out);
		*obuf++ = out * 256;
		/* Mix decay of delays and input */
		for ( j = 0; j < echos->num_delays; j++ ) {
			if ( j == 0 )
				echos->delay_buf[echos->counter[j] + echos->pointer[j]] = d_in;
			else
				echos->delay_buf[echos->counter[j] + echos->pointer[j]] = 
				   echos->delay_buf[echos->counter[j-1] + echos->pointer[j-1]];
		}
		/* Adjust the counters */
		for ( j = 0; j < echos->num_delays; j++ )
			echos->counter[j] = 
			   ( echos->counter[j] + 1 ) % echos->samples[j];
		done++;
		echos->sumsamples--;
	};
	/* samples played, it remains */
	*osamp = done;
}

/*
 * Clean up reverb effect.
 */
void echos_stop(effp)
eff_t effp;
{
	echos_t echos = (echos_t) effp->priv;

	free((char *) echos->delay_buf);
	echos->delay_buf = (double *) -1;   /* guaranteed core dump */
}