resample.c
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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
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it under the terms of the Perl Artistic License, available in COPYING.
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resample.c
*/
#if HAVE_CONFIG_H
# include <config.h>
#endif
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#include <math.h>
#include <stdio.h>
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#include <stdlib.h>
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#include "SDL.h"
#include "timidity.h"
#include "options.h"
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#include "common.h"
#include "instrum.h"
#include "playmidi.h"
#include "tables.h"
#include "resample.h"
/*************** resampling with fixed increment *****************/
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static sample_t *rs_plain(MidiSong *song, int v, Sint32 *countptr)
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{
/* Play sample until end, then free the voice. */
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sample_t v1, v2;
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Voice
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*vp=&(song->voice[v]);
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sample_t
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*dest=song->resample_buffer,
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*src=vp->sample->data;
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Sint32
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ofs=vp->sample_offset,
incr=vp->sample_increment,
le=vp->sample->data_length,
count=*countptr;
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Sint32 i;
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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 = (le - ofs) / incr + 1;
if (i > count)
{
i = count;
count = 0;
}
else count -= i;
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while (i--)
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{
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v1 = src[ofs >> FRACTION_BITS];
v2 = src[(ofs >> FRACTION_BITS)+1];
*dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS);
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ofs += incr;
}
if (ofs >= le)
{
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if (ofs == le)
*dest++ = src[ofs >> FRACTION_BITS];
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vp->status=VOICE_FREE;
*countptr-=count+1;
}
vp->sample_offset=ofs; /* Update offset */
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return song->resample_buffer;
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}
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static sample_t *rs_loop(MidiSong *song, Voice *vp, Sint32 count)
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{
/* Play sample until end-of-loop, skip back and continue. */
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sample_t v1, v2;
Sint32
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ofs=vp->sample_offset,
incr=vp->sample_increment,
le=vp->sample->loop_end,
ll=le - vp->sample->loop_start;
sample_t
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*dest=song->resample_buffer,
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*src=vp->sample->data;
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Sint32 i;
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while (count)
{
if (ofs >= le)
/* NOTE: Assumes that ll > incr and that incr > 0. */
ofs -= ll;
/* Precalc how many times we should go through the loop */
i = (le - ofs) / incr + 1;
if (i > count)
{
i = count;
count = 0;
}
else count -= i;
while (i--)
{
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v1 = src[ofs >> FRACTION_BITS];
v2 = src[(ofs >> FRACTION_BITS)+1];
*dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS);
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ofs += incr;
}
}
vp->sample_offset=ofs; /* Update offset */
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return song->resample_buffer;
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}
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static sample_t *rs_bidir(MidiSong *song, Voice *vp, Sint32 count)
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{
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sample_t v1, v2;
Sint32
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ofs=vp->sample_offset,
incr=vp->sample_increment,
le=vp->sample->loop_end,
ls=vp->sample->loop_start;
sample_t
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*dest=song->resample_buffer,
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*src=vp->sample->data;
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Sint32
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le2 = le<<1,
ls2 = ls<<1,
i;
/* Play normally until inside the loop region */
if (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 = (ls - ofs) / incr + 1;
if (i > count)
{
i = count;
count = 0;
}
else count -= i;
while (i--)
{
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v1 = src[ofs >> FRACTION_BITS];
v2 = src[(ofs >> FRACTION_BITS)+1];
*dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS);
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ofs += incr;
}
}
/* Then do the bidirectional looping */
while(count)
{
/* Precalc how many times we should go through the loop */
i = ((incr > 0 ? le : ls) - ofs) / incr + 1;
if (i > count)
{
i = count;
count = 0;
}
else count -= i;
while (i--)
{
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v1 = src[ofs >> FRACTION_BITS];
v2 = src[(ofs >> FRACTION_BITS)+1];
*dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS);
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ofs += incr;
}
if (ofs>=le)
{
/* fold the overshoot back in */
ofs = le2 - ofs;
incr *= -1;
}
else if (ofs <= ls)
{
ofs = ls2 - ofs;
incr *= -1;
}
}
vp->sample_increment=incr;
vp->sample_offset=ofs; /* Update offset */
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return song->resample_buffer;
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}
/*********************** 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;
}
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static Sint32 update_vibrato(MidiSong *song, Voice *vp, int sign)
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{
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Sint32 depth;
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int phase, pb;
double a;
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->sample->vibrato_depth<<7;
if (vp->vibrato_sweep)
{
/* 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;
}
}
a = FSCALE(((double)(vp->sample->sample_rate) *
(double)(vp->frequency)) /
((double)(vp->sample->root_freq) *
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(double)(song->rate)),
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FRACTION_BITS);
pb=(int)((sine(vp->vibrato_phase *
(SINE_CYCLE_LENGTH/(2*VIBRATO_SAMPLE_INCREMENTS)))
* (double)(depth) * VIBRATO_AMPLITUDE_TUNING));
if (pb<0)
{
pb=-pb;
a /= bend_fine[(pb>>5) & 0xFF] * bend_coarse[pb>>13];
}
else
a *= bend_fine[(pb>>5) & 0xFF] * bend_coarse[pb>>13];
/* If the sweep's over, we can store the newly computed sample_increment */
if (!vp->vibrato_sweep)
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vp->vibrato_sample_increment[phase]=(Sint32) a;
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if (sign)
a = -a; /* need to preserve the loop direction */
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return (Sint32) a;
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}
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static sample_t *rs_vib_plain(MidiSong *song, int v, Sint32 *countptr)
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{
/* Play sample until end, then free the voice. */
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sample_t v1, v2;
Voice *vp=&(song->voice[v]);
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sample_t
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*dest=song->resample_buffer,
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*src=vp->sample->data;
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Sint32
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le=vp->sample->data_length,
ofs=vp->sample_offset,
incr=vp->sample_increment,
count=*countptr;
int
cc=vp->vibrato_control_counter;
/* 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;
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incr=update_vibrato(song, vp, 0);
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}
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v1 = src[ofs >> FRACTION_BITS];
v2 = src[(ofs >> FRACTION_BITS)+1];
*dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS);
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ofs += incr;
if (ofs >= le)
{
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if (ofs == le)
*dest++ = src[ofs >> FRACTION_BITS];
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vp->status=VOICE_FREE;
*countptr-=count+1;
break;
}
}
vp->vibrato_control_counter=cc;
vp->sample_increment=incr;
vp->sample_offset=ofs; /* Update offset */
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return song->resample_buffer;
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}
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static sample_t *rs_vib_loop(MidiSong *song, Voice *vp, Sint32 count)
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{
/* Play sample until end-of-loop, skip back and continue. */
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sample_t v1, v2;
Sint32
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ofs=vp->sample_offset,
incr=vp->sample_increment,
le=vp->sample->loop_end,
ll=le - vp->sample->loop_start;
sample_t
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*dest=song->resample_buffer,
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*src=vp->sample->data;
int
cc=vp->vibrato_control_counter;
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Sint32 i;
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int
vibflag=0;
while (count)
{
/* Hopefully the loop is longer than an increment */
if(ofs >= le)
ofs -= ll;
/* Precalc how many times to go through the loop, taking
the vibrato control ratio into account this time. */
i = (le - ofs) / incr + 1;
if(i > count) i = count;
if(i > cc)
{
i = cc;
vibflag = 1;
}
else cc -= i;
count -= i;
while(i--)
{
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v1 = src[ofs >> FRACTION_BITS];
v2 = src[(ofs >> FRACTION_BITS)+1];
*dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS);
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ofs += incr;
}
if(vibflag)
{
cc = vp->vibrato_control_ratio;
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incr = update_vibrato(song, vp, 0);
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vibflag = 0;
}
}
vp->vibrato_control_counter=cc;
vp->sample_increment=incr;
vp->sample_offset=ofs; /* Update offset */
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return song->resample_buffer;
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}
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static sample_t *rs_vib_bidir(MidiSong *song, Voice *vp, Sint32 count)
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{
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sample_t v1, v2;
Sint32
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ofs=vp->sample_offset,
incr=vp->sample_increment,
le=vp->sample->loop_end,
ls=vp->sample->loop_start;
sample_t
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*dest=song->resample_buffer,
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*src=vp->sample->data;
int
cc=vp->vibrato_control_counter;
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Sint32
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le2=le<<1,
ls2=ls<<1,
i;
int
vibflag = 0;
/* Play normally until inside the loop region */
while (count && (ofs <= ls))
{
i = (ls - ofs) / incr + 1;
if (i > count) i = count;
if (i > cc)
{
i = cc;
vibflag = 1;
}
else cc -= i;
count -= i;
while (i--)
{
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v1 = src[ofs >> FRACTION_BITS];
v2 = src[(ofs >> FRACTION_BITS)+1];
*dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS);
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ofs += incr;
}
if (vibflag)
{
cc = vp->vibrato_control_ratio;
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incr = update_vibrato(song, vp, 0);
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vibflag = 0;
}
}
/* Then do the bidirectional looping */
while (count)
{
/* Precalc how many times we should go through the loop */
i = ((incr > 0 ? le : ls) - ofs) / incr + 1;
if(i > count) i = count;
if(i > cc)
{
i = cc;
vibflag = 1;
}
else cc -= i;
count -= i;
while (i--)
{
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v1 = src[ofs >> FRACTION_BITS];
v2 = src[(ofs >> FRACTION_BITS)+1];
*dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS);
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ofs += incr;
}
if (vibflag)
{
cc = vp->vibrato_control_ratio;
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incr = update_vibrato(song, vp, (incr < 0));
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vibflag = 0;
}
if (ofs >= le)
{
/* fold the overshoot back in */
ofs = le2 - ofs;
incr *= -1;
}
else if (ofs <= ls)
{
ofs = ls2 - ofs;
incr *= -1;
}
}
vp->vibrato_control_counter=cc;
vp->sample_increment=incr;
vp->sample_offset=ofs; /* Update offset */
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return song->resample_buffer;
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}
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sample_t *resample_voice(MidiSong *song, int v, Sint32 *countptr)
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{
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Sint32 ofs;
Uint8 modes;
Voice *vp=&(song->voice[v]);
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if (!(vp->sample->sample_rate))
{
/* Pre-resampled data -- just update the offset and check if
we're out of data. */
ofs=vp->sample_offset >> FRACTION_BITS; /* Kind of silly to use
FRACTION_BITS here... */
if (*countptr >= (vp->sample->data_length>>FRACTION_BITS) - ofs)
{
/* Note finished. Free the voice. */
vp->status = VOICE_FREE;
/* Let the caller know how much data we had left */
*countptr = (vp->sample->data_length>>FRACTION_BITS) - ofs;
}
else
vp->sample_offset += *countptr << FRACTION_BITS;
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return vp->sample->data+ofs;
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}
/* Need to resample. Use the proper function. */
modes=vp->sample->modes;
if (vp->vibrato_control_ratio)
{
if ((modes & MODES_LOOPING) &&
((modes & MODES_ENVELOPE) ||
(vp->status==VOICE_ON || vp->status==VOICE_SUSTAINED)))
{
if (modes & MODES_PINGPONG)
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return rs_vib_bidir(song, vp, *countptr);
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else
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return rs_vib_loop(song, vp, *countptr);
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}
else
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return rs_vib_plain(song, v, countptr);
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}
else
{
if ((modes & MODES_LOOPING) &&
((modes & MODES_ENVELOPE) ||
(vp->status==VOICE_ON || vp->status==VOICE_SUSTAINED)))
{
if (modes & MODES_PINGPONG)
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return rs_bidir(song, vp, *countptr);
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else
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return rs_loop(song, vp, *countptr);
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}
else
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return rs_plain(song, v, countptr);
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}
}
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void pre_resample(MidiSong *song, Sample *sp)
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{
double a, xdiff;
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Sint32 incr, ofs, newlen, count;
Sint16 *newdata, *dest, *src = (Sint16 *) sp->data;
Sint16 v1, v2, v3, v4, *vptr;
#ifdef DEBUG_CHATTER
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static const char note_name[12][3] =
{
"C", "C#", "D", "D#", "E", "F", "F#", "G", "G#", "A", "A#", "B"
};
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SNDDBG((" * pre-resampling for note %d (%s%d)\n",
sp->note_to_use,
note_name[sp->note_to_use % 12], (sp->note_to_use & 0x7F) / 12));
#endif
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a = ((double) (sp->sample_rate) * freq_table[(int) (sp->note_to_use)]) /
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((double) (sp->root_freq) * song->rate);
newlen = (Sint32)(sp->data_length / a);
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dest = newdata = safe_malloc(newlen >> (FRACTION_BITS - 1));
count = (newlen >> FRACTION_BITS) - 1;
ofs = incr = (sp->data_length - (1 << FRACTION_BITS)) / count;
if (--count)
*dest++ = src[0];
/* Since we're pre-processing and this doesn't have to be done in
real-time, we go ahead and do the full sliding cubic interpolation. */
while (--count)
{
vptr = src + (ofs >> FRACTION_BITS);
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/*
* Electric Fence to the rescue: Accessing *(vptr - 1) is not a
* good thing to do when vptr <= src. (TiMidity++ has a similar
* safe-guard here.)
*/
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v2 = *vptr;
v3 = *(vptr + 1);
v4 = *(vptr + 2);
xdiff = FSCALENEG(ofs & FRACTION_MASK, FRACTION_BITS);
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*dest++ = (Sint16)(v2 + (xdiff / 6.0) * (-2 * v1 - 3 * v2 + 6 * v3 - v4 +
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xdiff * (3 * (v1 - 2 * v2 + v3) + xdiff * (-v1 + 3 * (v2 - v3) + v4))));
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ofs += incr;
}
if (ofs & FRACTION_MASK)
{
v1 = src[ofs >> FRACTION_BITS];
v2 = src[(ofs >> FRACTION_BITS) + 1];
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*dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS);
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}
else
*dest++ = src[ofs >> FRACTION_BITS];
sp->data_length = newlen;
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sp->loop_start = (Sint32)(sp->loop_start / a);
sp->loop_end = (Sint32)(sp->loop_end / a);
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free(sp->data);
sp->data = (sample_t *) newdata;
sp->sample_rate = 0;
}