/*
    TiMidity++ -- MIDI to WAVE converter and player
    Copyright (C) 1999-2002 Masanao Izumo <mo@goice.co.jp>
    Copyright (C) 1995 Tuukka Toivonen <tt@cgs.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., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

    resample.c
*/

#include <math.h>
#include <stdio.h>
#include <stdlib.h>

#include "timidity.h"
#include "common.h"
#include "instrum.h"
#include "playmidi.h"
#include "tables.h"
#include "resample.h"
#include "recache.h"

namespace TimidityPlus
{


/* for start/end of samples */
static float newt_coeffs[58][58];
static int sample_bounds_min, sample_bounds_max; /* min/max bounds for sample data */

#define DEFAULT_GAUSS_ORDER	25
static float *gauss_table[(1 << FRACTION_BITS)] = { 0 };	/* don't need doubles */
static int gauss_n = DEFAULT_GAUSS_ORDER;


static void initialize_newton_coeffs()
{
	int i, j, n = 57;
	int sign;

	newt_coeffs[0][0] = 1;
	for (i = 0; i <= n; i++)
	{
		newt_coeffs[i][0] = 1;
		newt_coeffs[i][i] = 1;

		if (i > 1)
		{
			newt_coeffs[i][0] = newt_coeffs[i - 1][0] / i;
			newt_coeffs[i][i] = newt_coeffs[i - 1][0] / i;
		}

		for (j = 1; j < i; j++)
		{
			newt_coeffs[i][j] = newt_coeffs[i - 1][j - 1] + newt_coeffs[i - 1][j];

			if (i > 1)
				newt_coeffs[i][j] /= i;
		}
	}
	for (i = 0; i <= n; i++)
		for (j = 0, sign = pow(-1, i); j <= i; j++, sign *= -1)
			newt_coeffs[i][j] *= sign;

}



/* Very fast and accurate table based interpolation.  Better speed and higher
	accuracy than Newton.  This isn't *quite* true Gauss interpolation; it's
	more a slightly modified Gauss interpolation that I accidently stumbled
	upon.  Rather than normalize all x values in the window to be in the range
	[0 to 2*PI], it simply divides them all by 2*PI instead.  I don't know why
	this works, but it does.  Gauss should only work on periodic data with the
	window spanning exactly one period, so it is no surprise that regular Gauss
	interpolation doesn't work too well on general audio data.  But dividing
	the x values by 2*PI magically does.  Any other scaling produces degraded
	results or total garbage.  If anyone can work out the theory behind why
	this works so well (at first glance, it shouldn't ??), please contact me
	(Eric A. Welsh, ewelsh@ccb.wustl.edu), as I would really like to have some
	mathematical justification for doing this.  Despite the lack of any sound
	theoretical basis, this method DOES result in highly accurate interpolation
	(or possibly approximaton, not sure yet if it truly interpolates, but it
	looks like it does).  -N 34 is as high as it can go before errors start
	appearing.  But even at -N 34, it is more accurate than Newton at -N 57.
	-N 34 has no problem running in realtime on my system, but -N 25 is the
	default, since that is the optimal compromise between speed and accuracy.
	I strongly recommend using Gauss interpolation.  It is the highest
	quality interpolation option available, and is much faster than using
	Newton polynomials. */


static resample_t resample_gauss(sample_t *src, splen_t ofs, resample_rec_t *rec)
{
	sample_t *sptr;
	int32_t left, right, temp_n;

	left = (ofs >> FRACTION_BITS);
	right = (rec->data_length >> FRACTION_BITS) - left - 1;
	temp_n = (right << 1) - 1;
	if (temp_n > (left << 1) + 1)
		temp_n = (left << 1) + 1;
	if (temp_n < gauss_n) {
		int ii, jj;
		float xd, y;
		if (temp_n <= 0)
			temp_n = 1;
		xd = ofs & FRACTION_MASK;
		xd /= (1L << FRACTION_BITS);
		xd += temp_n >> 1;
		y = 0;
		sptr = src + (ofs >> FRACTION_BITS) - (temp_n >> 1);
		for (ii = temp_n; ii;) {
			for (jj = 0; jj <= ii; jj++)
				y += sptr[jj] * newt_coeffs[ii][jj];
			y *= xd - --ii;
		}
		y += *sptr;
		return ((y > sample_bounds_max) ? sample_bounds_max :
			((y < sample_bounds_min) ? sample_bounds_min : y));
	}
	else {
		float *gptr, *gend;
		float y;
		y = 0;
		sptr = src + left - (gauss_n >> 1);
		gptr = gauss_table[ofs&FRACTION_MASK];
		if (gauss_n == DEFAULT_GAUSS_ORDER) {
			/* expanding the loop for the default case.
				* this will allow intensive optimization when compiled
				* with SSE2 capability.
				*/
#define do_gauss  y += *(sptr++) * *(gptr++);
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			do_gauss;
			y += *sptr * *gptr;
#undef do_gauss
		}
		else {
			gend = gptr + gauss_n;
			do {
				y += *(sptr++) * *(gptr++);
			} while (gptr <= gend);
		}
		return ((y > sample_bounds_max) ? sample_bounds_max :
			((y < sample_bounds_min) ? sample_bounds_min : y));
	}
}


#define RESAMPLATION *dest++ = resample_gauss(src, ofs, &resrc);

/* exported for recache.c */
resample_t do_resamplation(sample_t *src, splen_t ofs, resample_rec_t *rec)
{
	return resample_gauss(src, ofs, rec);
}

#define PRECALC_LOOP_COUNT(start, end, incr) (int32_t)(((int64_t)((end) - (start) + (incr) - 1)) / (incr))

void initialize_gauss_table(int n)
{
	int m, i, k, n_half = (n >> 1);
	double ck;
	double x, x_inc, xz;
	double z[35], zsin_[34 + 35], *zsin, xzsin[35];
	float *gptr;

	for (i = 0; i <= n; i++)
		z[i] = i / (4 * M_PI);
	zsin = &zsin_[34];
	for (i = -n; i <= n; i++)
		zsin[i] = sin(i / (4 * M_PI));

	x_inc = 1.0 / (1 << FRACTION_BITS);
	gptr = (float*)safe_realloc(gauss_table[0], (n + 1) * sizeof(float)*(1 << FRACTION_BITS));
	for (m = 0, x = 0.0; m < (1 << FRACTION_BITS); m++, x += x_inc)
	{
		xz = (x + n_half) / (4 * M_PI);
		for (i = 0; i <= n; i++)
			xzsin[i] = sin(xz - z[i]);
		gauss_table[m] = gptr;

		for (k = 0; k <= n; k++)
		{
			ck = 1.0;

			for (i = 0; i <= n; i++)
			{
				if (i == k)
					continue;

				ck *= xzsin[i] / zsin[k - i];
			}

			*gptr++ = ck;
		}
	}
}

void free_gauss_table(void)
{
	if (gauss_table[0] != 0)
		free(gauss_table[0]);
	gauss_table[0] = NULL;
}

/* initialize the coefficients of the current resampling algorithm */
void initialize_resampler_coeffs(void)
{
	// Only needs to be done once.
	static bool done = false;
	if (done) return;
	done = true;

	initialize_newton_coeffs();
	initialize_gauss_table(gauss_n);

	sample_bounds_min = -32768;
	sample_bounds_max = 32767;
}


/*************** resampling with fixed increment *****************/

resample_t *Resampler::rs_plain_c(int v, int32_t *countptr)
{
	Voice *vp = &player->voice[v];
	resample_t *dest = resample_buffer + resample_buffer_offset;
	sample_t *src = vp->sample->data;
	int32_t ofs, count = *countptr, i, le;

	le = (int32_t)(vp->sample->loop_end >> FRACTION_BITS);
	ofs = (int32_t)(vp->sample_offset >> FRACTION_BITS);

	i = ofs + count;
	if (i > le)
		i = le;
	count = i - ofs;

	for (i = 0; i < count; i++) {
		dest[i] = src[i + ofs];
	}

	ofs += count;
	if (ofs == le)
	{
		vp->timeout = 1;
		*countptr = count;
	}
	vp->sample_offset = ((splen_t)ofs << FRACTION_BITS);
	return resample_buffer + resample_buffer_offset;
}

resample_t *Resampler::rs_plain(int v, int32_t *countptr)
{
	/* Play sample until end, then free the voice. */
	Voice *vp = &player->voice[v];
	resample_t *dest = resample_buffer + resample_buffer_offset;
	sample_t *src = vp->sample->data;
	splen_t
		ofs = vp->sample_offset,
		ls = 0,
		le = vp->sample->data_length;
	resample_rec_t resrc;
	int32_t count = *countptr, incr = vp->sample_increment;
	int32_t i, j;

	if (vp->cache && incr == (1 << FRACTION_BITS))
		return rs_plain_c(v, countptr);

	resrc.loop_start = ls;
	resrc.loop_end = le;
	resrc.data_length = vp->sample->data_length;
	if (incr < 0) incr = -incr; /* In case we're coming out of a bidir loop */

	/* Precalc how many times we should go through the loop.
		NOTE: Assumes that incr > 0 and that ofs <= le */
	i = PRECALC_LOOP_COUNT(ofs, le, incr);

	if (i > count)
	{
		i = count;
		count = 0;
	}
	else count -= i;

	for (j = 0; j < i; j++)
	{
		RESAMPLATION;
		ofs += incr;
	}

	if (ofs >= le)
	{
		vp->timeout = 1;
		*countptr -= count;
	}

	vp->sample_offset = ofs; /* Update offset */
	return resample_buffer + resample_buffer_offset;
}

resample_t *Resampler::rs_loop_c(Voice *vp, int32_t count)
{
	int32_t
		ofs = (int32_t)(vp->sample_offset >> FRACTION_BITS),
		le = (int32_t)(vp->sample->loop_end >> FRACTION_BITS),
		ll = le - (int32_t)(vp->sample->loop_start >> FRACTION_BITS);
	resample_t *dest = resample_buffer + resample_buffer_offset;
	sample_t *src = vp->sample->data;
	int32_t i, j;

	while (count)
	{
		while (ofs >= le)
			ofs -= ll;
		/* Precalc how many times we should go through the loop */
		i = le - ofs;
		if (i > count)
			i = count;
		count -= i;
		for (j = 0; j < i; j++) {
			dest[j] = src[j + ofs];
		}
		dest += i;
		ofs += i;
	}
	vp->sample_offset = ((splen_t)ofs << FRACTION_BITS);
	return resample_buffer + resample_buffer_offset;
}

resample_t *Resampler::rs_loop(Voice *vp, int32_t count)
{
	/* Play sample until end-of-loop, skip back and continue. */
	splen_t
		ofs = vp->sample_offset,
		ls, le, ll;
	resample_rec_t resrc;
	resample_t *dest = resample_buffer + resample_buffer_offset;
	sample_t *src = vp->sample->data;
	int32_t i, j;
	int32_t incr = vp->sample_increment;

	if (vp->cache && incr == (1 << FRACTION_BITS))
		return rs_loop_c(vp, count);

	resrc.loop_start = ls = vp->sample->loop_start;
	resrc.loop_end = le = vp->sample->loop_end;
	ll = le - ls;
	resrc.data_length = vp->sample->data_length;

	while (count)
	{
		while (ofs >= le) { ofs -= ll; }
		/* Precalc how many times we should go through the loop */
		i = PRECALC_LOOP_COUNT(ofs, le, incr);
		if (i > count) {
			i = count;
			count = 0;
		}
		else { count -= i; }
		for (j = 0; j < i; j++) {
			RESAMPLATION;
			ofs += incr;
		}
	}

	vp->sample_offset = ofs; /* Update offset */
	return resample_buffer + resample_buffer_offset;
}

resample_t *Resampler::rs_bidir(Voice *vp, int32_t count)
{
	int32_t
		ofs = vp->sample_offset,
		le = vp->sample->loop_end,
		ls = vp->sample->loop_start;
	resample_t *dest = resample_buffer + resample_buffer_offset;
	sample_t *src = vp->sample->data;
	int32_t incr = vp->sample_increment;
	resample_rec_t resrc;

	int32_t
		le2 = le << 1,
		ls2 = ls << 1;
	int32_t i, j;
	/* Play normally until inside the loop region */

	resrc.loop_start = ls;
	resrc.loop_end = le;
	resrc.data_length = vp->sample->data_length;

	if (incr > 0 && ofs < ls)
	{
		/* NOTE: Assumes that incr > 0, which is NOT always the case
		when doing bidirectional looping.  I have yet to see a case
		where both ofs <= ls AND incr < 0, however. */
		i = PRECALC_LOOP_COUNT(ofs, ls, incr);
		if (i > count)
		{
			i = count;
			count = 0;
		}
		else count -= i;
		for (j = 0; j < i; j++)
		{
			RESAMPLATION;
			ofs += incr;
		}
	}

	/* Then do the bidirectional looping */

	while (count)
	{
		/* Precalc how many times we should go through the loop */
		i = PRECALC_LOOP_COUNT(ofs, incr > 0 ? le : ls, incr);
		if (i > count)
		{
			i = count;
			count = 0;
		}
		else count -= i;
		for (j = 0; j < i; j++)
		{
			RESAMPLATION;
			ofs += incr;
		}
		if (ofs >= 0 && ofs >= le)
		{
			/* fold the overshoot back in */
			ofs = le2 - ofs;
			incr *= -1;
		}
		else if (ofs <= 0 || ofs <= ls)
		{
			ofs = ls2 - ofs;
			incr *= -1;
		}
	}
	vp->sample_increment = incr;
	vp->sample_offset = ofs; /* Update offset */
	return resample_buffer + resample_buffer_offset;
}

/*********************** vibrato versions ***************************/

/* We only need to compute one half of the vibrato sine cycle */
static int vib_phase_to_inc_ptr(int phase)
{
	if (phase < VIBRATO_SAMPLE_INCREMENTS / 2)
		return VIBRATO_SAMPLE_INCREMENTS / 2 - 1 - phase;
	else if (phase >= 3 * VIBRATO_SAMPLE_INCREMENTS / 2)
		return 5 * VIBRATO_SAMPLE_INCREMENTS / 2 - 1 - phase;
	else
		return phase - VIBRATO_SAMPLE_INCREMENTS / 2;
}

int32_t Resampler::update_vibrato(Voice *vp, int sign)
{
	int32_t depth;
	int phase, pb;
	double a;
	int ch = vp->channel;

	if (vp->vibrato_delay > 0)
	{
		vp->vibrato_delay -= vp->vibrato_control_ratio;
		if (vp->vibrato_delay > 0)
			return vp->sample_increment;
	}

	if (vp->vibrato_phase++ >= 2 * VIBRATO_SAMPLE_INCREMENTS - 1)
		vp->vibrato_phase = 0;
	phase = vib_phase_to_inc_ptr(vp->vibrato_phase);

	if (vp->vibrato_sample_increment[phase])
	{
		if (sign)
			return -vp->vibrato_sample_increment[phase];
		else
			return vp->vibrato_sample_increment[phase];
	}

	/* Need to compute this sample increment. */

	depth = vp->vibrato_depth;
	depth <<= 7;

	if (vp->vibrato_sweep && !player->channel[ch].mod.val)
	{
		/* Need to update sweep */
		vp->vibrato_sweep_position += vp->vibrato_sweep;
		if (vp->vibrato_sweep_position >= (1 << SWEEP_SHIFT))
			vp->vibrato_sweep = 0;
		else
		{
			/* Adjust depth */
			depth *= vp->vibrato_sweep_position;
			depth >>= SWEEP_SHIFT;
		}
	}

	if (vp->sample->inst_type == INST_SF2) {
		pb = (int)((lookup_triangular(vp->vibrato_phase *
			(SINE_CYCLE_LENGTH / (2 * VIBRATO_SAMPLE_INCREMENTS)))
			* (double)(depth)* VIBRATO_AMPLITUDE_TUNING));
	}
	else {
		pb = (int)((lookup_sine(vp->vibrato_phase *
			(SINE_CYCLE_LENGTH / (2 * VIBRATO_SAMPLE_INCREMENTS)))
			* (double)(depth)* VIBRATO_AMPLITUDE_TUNING));
	}

	a = TIM_FSCALE(((double)(vp->sample->sample_rate) *
		(double)(vp->frequency)) /
		((double)(vp->sample->root_freq) *
		(double)(playback_rate)),
		FRACTION_BITS);

	if (pb < 0) {
		pb = -pb;
		a /= bend_fine[(pb >> 5) & 0xFF] * bend_coarse[pb >> 13];
		pb = -pb;
	}
	else {
		a *= bend_fine[(pb >> 5) & 0xFF] * bend_coarse[pb >> 13];
	}
	a += 0.5;

	/* If the sweep's over, we can store the newly computed sample_increment */
	if (!vp->vibrato_sweep || player->channel[ch].mod.val)
		vp->vibrato_sample_increment[phase] = (int32_t)a;

	if (sign)
		a = -a; /* need to preserve the loop direction */

	return (int32_t)a;
}

resample_t *Resampler::rs_vib_plain(int v, int32_t *countptr)
{
	/* Play sample until end, then free the voice. */
	Voice *vp = &player->voice[v];
	resample_t *dest = resample_buffer + resample_buffer_offset;
	sample_t *src = vp->sample->data;
	splen_t
		ls = 0,
		le = vp->sample->data_length,
		ofs = vp->sample_offset;
	resample_rec_t resrc;

	int32_t count = *countptr, incr = vp->sample_increment;
	int cc = vp->vibrato_control_counter;

	resrc.loop_start = ls;
	resrc.loop_end = le;
	resrc.data_length = vp->sample->data_length;
	/* This has never been tested */

	if (incr < 0) incr = -incr; /* In case we're coming out of a bidir loop */

	while (count--)
	{
		if (!cc--)
		{
			cc = vp->vibrato_control_ratio;
			incr = update_vibrato(vp, 0);
		}
		RESAMPLATION;
		ofs += incr;
		if (ofs >= le)
		{
			vp->timeout = 1;
			*countptr -= count;
			break;
		}
	}

	vp->vibrato_control_counter = cc;
	vp->sample_increment = incr;
	vp->sample_offset = ofs; /* Update offset */
	return resample_buffer + resample_buffer_offset;
}

resample_t *Resampler::rs_vib_loop(Voice *vp, int32_t count)
{
	/* Play sample until end-of-loop, skip back and continue. */
	splen_t
		ofs = vp->sample_offset,
		ls = vp->sample->loop_start,
		le = vp->sample->loop_end,
		ll = le - vp->sample->loop_start;
	resample_t *dest = resample_buffer + resample_buffer_offset;
	sample_t *src = vp->sample->data;
	int cc = vp->vibrato_control_counter;
	int32_t incr = vp->sample_increment;
	resample_rec_t resrc;
	int32_t i, j;
	int vibflag = 0;

	resrc.loop_start = ls;
	resrc.loop_end = le;
	resrc.data_length = vp->sample->data_length;

	while (count)
	{
		/* Hopefully the loop is longer than an increment */
		while (ofs >= le) { ofs -= ll; }
		/* Precalc how many times to go through the loop, taking
		the vibrato control ratio into account this time. */
		i = PRECALC_LOOP_COUNT(ofs, le, incr);
		if (i > count) {
			i = count;
		}
		if (i > cc) {
			i = cc;
			vibflag = 1;
		}
		else { cc -= i; }
		count -= i;
		if (vibflag) {
			cc = vp->vibrato_control_ratio;
			incr = update_vibrato(vp, 0);
			vibflag = 0;
		}
		for (j = 0; j < i; j++) {
			RESAMPLATION;
			ofs += incr;
		}
	}

	vp->vibrato_control_counter = cc;
	vp->sample_increment = incr;
	vp->sample_offset = ofs; /* Update offset */
	return resample_buffer + resample_buffer_offset;
}

resample_t *Resampler::rs_vib_bidir(Voice *vp, int32_t count)
{
	int32_t
		ofs = vp->sample_offset,
		le = vp->sample->loop_end,
		ls = vp->sample->loop_start;
	resample_t *dest = resample_buffer + resample_buffer_offset;
	sample_t *src = vp->sample->data;
	int cc = vp->vibrato_control_counter;
	int32_t incr = vp->sample_increment;
	resample_rec_t resrc;


	resrc.loop_start = ls;
	resrc.loop_end = le;
	resrc.data_length = vp->sample->data_length;
	/* Play normally until inside the loop region */

	if (ofs < ls)
	{
		while (count--)
		{
			if (!cc--)
			{
				cc = vp->vibrato_control_ratio;
				incr = update_vibrato(vp, 0);
			}
			RESAMPLATION;
			ofs += incr;
			if (ofs >= ls)
				break;
		}
	}

	/* Then do the bidirectional looping */

	if (count > 0)
		while (count--)
		{
			if (!cc--)
			{
				cc = vp->vibrato_control_ratio;
				incr = update_vibrato(vp, (incr < 0));
			}
			RESAMPLATION;
			ofs += incr;
			if (ofs >= le)
			{
				/* fold the overshoot back in */
				ofs = le - (ofs - le);
				incr = -incr;
			}
			else if (ofs <= ls)
			{
				ofs = ls + (ls - ofs);
				incr = -incr;
			}
		}

	/* Update changed values */
	vp->vibrato_control_counter = cc;
	vp->sample_increment = incr;
	vp->sample_offset = ofs;
	return resample_buffer + resample_buffer_offset;
}

/*********************** portamento versions ***************************/

int Resampler::rs_update_porta(int v)
{
	Voice *vp = &player->voice[v];
	int32_t d;

	d = vp->porta_dpb;
	if (vp->porta_pb < 0)
	{
		if (d > -vp->porta_pb)
			d = -vp->porta_pb;
	}
	else
	{
		if (d > vp->porta_pb)
			d = -vp->porta_pb;
		else
			d = -d;
	}

	vp->porta_pb += d;
	if (vp->porta_pb == 0)
	{
		vp->porta_control_ratio = 0;
		vp->porta_pb = 0;
	}
	player->recompute_freq(v);
	return vp->porta_control_ratio;
}

resample_t *Resampler::porta_resample_voice(int v, int32_t *countptr, int mode)
{
	Voice *vp = &player->voice[v];
	int32_t n = *countptr, i;
	resample_t *(Resampler::*resampler)(int, int32_t *, int);
	int cc = vp->porta_control_counter;
	int loop;

	if (vp->vibrato_control_ratio)
		resampler = &Resampler::vib_resample_voice;
	else
		resampler = &Resampler::normal_resample_voice;
	if (mode != 1)
		loop = 1;
	else
		loop = 0;

	vp->cache = NULL;
	resample_buffer_offset = 0;
	while (resample_buffer_offset < n)
	{
		if (cc == 0)
		{
			if ((cc = rs_update_porta(v)) == 0)
			{
				i = n - resample_buffer_offset;
				(this->*resampler)(v, &i, mode);
				resample_buffer_offset += i;
				break;
			}
		}

		i = n - resample_buffer_offset;
		if (i > cc)
			i = cc;
		(this->*resampler)(v, &i, mode);
		resample_buffer_offset += i;

		if (!loop && (i == 0 || vp->status == VOICE_FREE))
			break;
		cc -= i;
	}
	*countptr = resample_buffer_offset;
	resample_buffer_offset = 0;
	vp->porta_control_counter = cc;
	return resample_buffer;
}

/* interface function */
resample_t *Resampler::vib_resample_voice(int v, int32_t *countptr, int mode)
{
	Voice *vp = &player->voice[v];

	vp->cache = NULL;
	if (mode == 0)
		return rs_vib_loop(vp, *countptr);
	if (mode == 1)
		return rs_vib_plain(v, countptr);
	return rs_vib_bidir(vp, *countptr);
}

/* interface function */
resample_t *Resampler::normal_resample_voice(int v, int32_t *countptr, int mode)
{
	Voice *vp = &player->voice[v];
	if (mode == 0)
		return rs_loop(vp, *countptr);
	if (mode == 1)
		return rs_plain(v, countptr);
	return rs_bidir(vp, *countptr);
}

/* interface function */
resample_t *Resampler::resample_voice(int v, int32_t *countptr)
{
	Voice *vp = &player->voice[v];
	int mode;
	resample_t *result;
	int32_t i;

	if (vp->sample->sample_rate == playback_rate &&
		vp->sample->root_freq == get_note_freq(vp->sample, vp->sample->note_to_use) &&
		vp->frequency == vp->orig_frequency)
	{
		int32_t ofs;

		/* Pre-resampled data -- just update the offset and check if
			we're out of data. */
		ofs = (int32_t)(vp->sample_offset >> FRACTION_BITS); /* Kind of silly to use
								FRACTION_BITS here... */
		if (*countptr >= (int32_t)((vp->sample->data_length >> FRACTION_BITS) - ofs))
		{
			/* Note finished. Free the voice. */
			vp->timeout = 1;

			/* Let the caller know how much data we had left */
			*countptr = (int32_t)(vp->sample->data_length >> FRACTION_BITS) - ofs;
		}
		else
			vp->sample_offset += *countptr << FRACTION_BITS;

		for (i = 0; i < *countptr; i++) {
			resample_buffer[i] = vp->sample->data[i + ofs];
		}
		return resample_buffer;
	}

	mode = vp->sample->modes;
	if ((mode & MODES_LOOPING) &&
		((mode & MODES_ENVELOPE) ||
		(vp->status & (VOICE_ON | VOICE_SUSTAINED))))
	{
		if (mode & MODES_PINGPONG)
		{
			vp->cache = NULL;
			mode = 2;	/* Bidir loop */
		}
		else
			mode = 0;	/* loop */
	}
	else
		mode = 1;	/* no loop */

	if (vp->porta_control_ratio)
		result = porta_resample_voice(v, countptr, mode);
	else if (vp->vibrato_control_ratio)
		result = vib_resample_voice(v, countptr, mode);
	else
		result = normal_resample_voice(v, countptr, mode);

	return result;
}


void pre_resample(Sample * sp)
{
	double a, b;
	splen_t ofs, newlen;
	sample_t *newdata, *dest, *src = (sample_t *)sp->data;
	int32_t i, count, incr, f, x;
	resample_rec_t resrc;

	f = get_note_freq(sp, sp->note_to_use);
	a = b = ((double)(sp->root_freq) * playback_rate) /
		((double)(sp->sample_rate) * f);
	if ((int64_t)sp->data_length * a >= 0x7fffffffL)
	{
		/* Too large to compute */
		ctl_cmsg(CMSG_INFO, VERB_DEBUG, " *** Can't pre-resampling for note %d",
			sp->note_to_use);
		return;
	}
	newlen = (splen_t)(sp->data_length * a);
	count = (newlen >> FRACTION_BITS);
	ofs = incr = (sp->data_length - 1) / (count - 1);

	if ((double)newlen + incr >= 0x7fffffffL)
	{
		/* Too large to compute */
		ctl_cmsg(CMSG_INFO, VERB_DEBUG, " *** Can't pre-resampling for note %d",
			sp->note_to_use);
		return;
	}

	dest = newdata = (sample_t *)safe_malloc((int32_t)(newlen >> (FRACTION_BITS - 1)) + 2);
	dest[newlen >> FRACTION_BITS] = 0;

	*dest++ = src[0];

	resrc.loop_start = 0;
	resrc.loop_end = sp->data_length;
	resrc.data_length = sp->data_length;

	/* Since we're pre-processing and this doesn't have to be done in
		real-time, we go ahead and do the higher order interpolation. */
	for (i = 1; i < count; i++)
	{
		x = resample_gauss(src, ofs, &resrc);
		*dest++ = (int16_t)((x > 32767) ? 32767 : ((x < -32768) ? -32768 : x));
		ofs += incr;
	}

	sp->data_length = newlen;
	sp->loop_start = (splen_t)(sp->loop_start * b);
	sp->loop_end = (splen_t)(sp->loop_end * b);
	free(sp->data);
	sp->data = (sample_t *)newdata;
	sp->root_freq = f;
	sp->sample_rate = playback_rate;
	sp->low_freq = freq_table[0];
	sp->high_freq = freq_table[127];
}

}