instrum.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
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.
instrum.c
Code to load and unload GUS-compatible instrument patches.
*/
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include "config.h"
#include "common.h"
#include "instrum.h"
#include "playmidi.h"
#include "output.h"
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#include "ctrlmode.h"
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#include "resample.h"
#include "tables.h"
#include "filter.h"
/* Some functions get aggravated if not even the standard banks are
available. */
static ToneBank standard_tonebank, standard_drumset;
ToneBank
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*tonebank[MAXBANK]={&standard_tonebank},
*drumset[MAXBANK]={&standard_drumset};
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/* This is a special instrument, used for all melodic programs */
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InstrumentLayer *default_instrument=0;
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/* This is only used for tracks that don't specify a program */
int default_program=DEFAULT_PROGRAM;
int antialiasing_allowed=0;
#ifdef FAST_DECAY
int fast_decay=1;
#else
int fast_decay=0;
#endif
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int current_tune_number = 0;
int last_tune_purged = 0;
int current_patch_memory = 0;
int max_patch_memory = 60000000;
static void purge_as_required(void);
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static void free_instrument(Instrument *ip)
{
Sample *sp;
int i;
if (!ip) return;
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if (!ip->contents)
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for (i=0; i<ip->samples; i++)
{
sp=&(ip->sample[i]);
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if (sp->data) free(sp->data);
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}
free(ip->sample);
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if (!ip->contents)
for (i=0; i<ip->right_samples; i++)
{
sp=&(ip->right_sample[i]);
if (sp->data) free(sp->data);
}
if (ip->right_sample)
free(ip->right_sample);
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free(ip);
}
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static void free_layer(InstrumentLayer *lp)
{
InstrumentLayer *next;
current_patch_memory -= lp->size;
for (; lp; lp = next)
{
next = lp->next;
free_instrument(lp->instrument);
free(lp);
}
}
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static void free_bank(int dr, int b)
{
int i;
ToneBank *bank=((dr) ? drumset[b] : tonebank[b]);
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for (i=0; i<MAXPROG; i++)
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if (bank->tone[i].layer)
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{
/* Not that this could ever happen, of course */
if (bank->tone[i].layer != MAGIC_LOAD_INSTRUMENT)
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{
free_layer(bank->tone[i].layer);
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bank->tone[i].layer=NULL;
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bank->tone[i].last_used=-1;
}
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}
if (bank->tone[i].name)
{
free(bank->tone[i].name);
bank->tone[i].name = NULL;
}
}
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}
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static void free_old_bank(int dr, int b, int how_old)
{
int i;
ToneBank *bank=((dr) ? drumset[b] : tonebank[b]);
for (i=0; i<MAXPROG; i++)
if (bank->tone[i].layer && bank->tone[i].last_used < how_old)
{
if (bank->tone[i].layer != MAGIC_LOAD_INSTRUMENT)
{
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
"Unloading %s %s[%d,%d] - last used %d.",
(dr)? "drum" : "inst", bank->tone[i].name,
i, b, bank->tone[i].last_used);
free_layer(bank->tone[i].layer);
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bank->tone[i].layer=NULL;
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bank->tone[i].last_used=-1;
}
}
}
int32 convert_envelope_rate_attack(uint8 rate, uint8 fastness)
{
int32 r;
r=3-((rate>>6) & 0x3);
r*=3;
r = (int32)(rate & 0x3f) << r; /* 6.9 fixed point */
/* 15.15 fixed point. */
return (((r * 44100) / play_mode->rate) * control_ratio)
<< 10;
}
int32 convert_envelope_rate(uint8 rate)
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{
int32 r;
r=3-((rate>>6) & 0x3);
r*=3;
r = (int32)(rate & 0x3f) << r; /* 6.9 fixed point */
/* 15.15 fixed point. */
return (((r * 44100) / play_mode->rate) * control_ratio)
<< ((fast_decay) ? 10 : 9);
}
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int32 convert_envelope_offset(uint8 offset)
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{
/* This is not too good... Can anyone tell me what these values mean?
Are they GUS-style "exponential" volumes? And what does that mean? */
/* 15.15 fixed point */
return offset << (7+15);
}
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int32 convert_tremolo_sweep(uint8 sweep)
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{
if (!sweep)
return 0;
return
((control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) /
(play_mode->rate * sweep);
}
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int32 convert_vibrato_sweep(uint8 sweep, int32 vib_control_ratio)
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{
if (!sweep)
return 0;
return
(int32) (FSCALE((double) (vib_control_ratio) * SWEEP_TUNING, SWEEP_SHIFT)
/ (double)(play_mode->rate * sweep));
/* this was overflowing with seashore.pat
((vib_control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) /
(play_mode->rate * sweep); */
}
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int32 convert_tremolo_rate(uint8 rate)
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{
return
((SINE_CYCLE_LENGTH * control_ratio * rate) << RATE_SHIFT) /
(TREMOLO_RATE_TUNING * play_mode->rate);
}
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int32 convert_vibrato_rate(uint8 rate)
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{
/* Return a suitable vibrato_control_ratio value */
return
(VIBRATO_RATE_TUNING * play_mode->rate) /
(rate * 2 * VIBRATO_SAMPLE_INCREMENTS);
}
static void reverse_data(int16 *sp, int32 ls, int32 le)
{
int16 s, *ep=sp+le;
sp+=ls;
le-=ls;
le/=2;
while (le--)
{
s=*sp;
*sp++=*ep;
*ep--=s;
}
}
/*
If panning or note_to_use != -1, it will be used for all samples,
instead of the sample-specific values in the instrument file.
For note_to_use, any value <0 or >127 will be forced to 0.
For other parameters, 1 means yes, 0 means no, other values are
undefined.
TODO: do reverse loops right */
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static InstrumentLayer *load_instrument(char *name, int font_type, int percussion,
int panning, int amp, int cfg_tuning, int note_to_use,
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int strip_loop, int strip_envelope,
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int strip_tail, int bank, int gm_num, int sf_ix)
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{
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InstrumentLayer *lp, *lastlp, *headlp = 0;
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Instrument *ip;
FILE *fp;
uint8 tmp[1024];
int i,j,noluck=0;
#ifdef PATCH_EXT_LIST
static char *patch_ext[] = PATCH_EXT_LIST;
#endif
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int sf2flag = 0;
int right_samples = 0;
int stereo_channels = 1, stereo_layer;
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int vlayer_list[19][4], vlayer, vlayer_count = 0;
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if (!name) return 0;
/* Open patch file */
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if ((fp=open_file(name, 1, OF_NORMAL)) == NULL)
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{
noluck=1;
#ifdef PATCH_EXT_LIST
/* Try with various extensions */
for (i=0; patch_ext[i]; i++)
{
if (strlen(name)+strlen(patch_ext[i])<1024)
{
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char path[1024];
strcpy(path, name);
strcat(path, patch_ext[i]);
if ((fp=open_file(path, 1, OF_NORMAL)) != NULL)
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{
noluck=0;
break;
}
}
}
#endif
}
if (noluck)
{
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"Instrument `%s' can't be found.", name);
return 0;
}
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/*ctl->cmsg(CMSG_INFO, VERB_NOISY, "Loading instrument %s", current_filename);*/
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/* Read some headers and do cursory sanity checks. There are loads
of magic offsets. This could be rewritten... */
if ((239 != fread(tmp, 1, 239, fp)) ||
(memcmp(tmp, "GF1PATCH110\0ID#000002", 22) &&
memcmp(tmp, "GF1PATCH100\0ID#000002", 22))) /* don't know what the
differences are */
{
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "%s: not an instrument", name);
return 0;
}
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/* patch layout:
* bytes: info: starts at offset:
* 22 id (see above) 0
* 60 copyright 22
* 1 instruments 82
* 1 voices 83
* 1 channels 84
* 2 number of waveforms 85
* 2 master volume 87
* 4 datasize 89
* 36 reserved, but now: 93
* 7 "SF2EXT\0" id 93
* 1 right samples 100
* 28 reserved 101
* 2 instrument number 129
* 16 instrument name 131
* 4 instrument size 147
* 1 number of layers 151
* 40 reserved 152
* 1 layer duplicate 192
* 1 layer number 193
* 4 layer size 194
* 1 number of samples 198
* 40 reserved 199
* 239
* THEN, for each sample, see below
*/
if (!memcmp(tmp + 93, "SF2EXT", 6))
{
sf2flag = 1;
vlayer_count = tmp[152];
}
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if (tmp[82] != 1 && tmp[82] != 0) /* instruments. To some patch makers,
0 means 1 */
{
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"Can't handle patches with %d instruments", tmp[82]);
return 0;
}
if (tmp[151] != 1 && tmp[151] != 0) /* layers. What's a layer? */
{
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"Can't handle instruments with %d layers", tmp[151]);
return 0;
}
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if (sf2flag && vlayer_count > 0) {
for (i = 0; i < 9; i++)
for (j = 0; j < 4; j++)
vlayer_list[i][j] = tmp[153+i*4+j];
for (i = 9; i < 19; i++)
for (j = 0; j < 4; j++)
vlayer_list[i][j] = tmp[199+(i-9)*4+j];
}
else {
for (i = 0; i < 19; i++)
for (j = 0; j < 4; j++)
vlayer_list[i][j] = 0;
vlayer_list[0][0] = 0;
vlayer_list[0][1] = 127;
vlayer_list[0][2] = tmp[198];
vlayer_list[0][3] = 0;
vlayer_count = 1;
}
lastlp = 0;
for (vlayer = 0; vlayer < vlayer_count; vlayer++) {
lp=(InstrumentLayer *)safe_malloc(sizeof(InstrumentLayer));
lp->size = sizeof(InstrumentLayer);
lp->lo = vlayer_list[vlayer][0];
lp->hi = vlayer_list[vlayer][1];
ip=(Instrument *)safe_malloc(sizeof(Instrument));
lp->size += sizeof(Instrument);
lp->instrument = ip;
lp->next = 0;
if (lastlp) lastlp->next = lp;
else headlp = lp;
lastlp = lp;
if (sf2flag) ip->type = INST_SF2;
else ip->type = INST_GUS;
ip->samples = vlayer_list[vlayer][2];
ip->sample = (Sample *)safe_malloc(sizeof(Sample) * ip->samples);
lp->size += sizeof(Sample) * ip->samples;
ip->left_samples = ip->samples;
ip->left_sample = ip->sample;
right_samples = vlayer_list[vlayer][3];
ip->right_samples = right_samples;
if (right_samples)
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{
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ip->right_sample = (Sample *)safe_malloc(sizeof(Sample) * right_samples);
lp->size += sizeof(Sample) * right_samples;
stereo_channels = 2;
}
else ip->right_sample = 0;
ip->contents = 0;
ctl->cmsg(CMSG_INFO, VERB_NOISY, "%s%s[%d,%d] %s(%d-%d layer %d of %d)",
(percussion)? " ":"", name,
(percussion)? note_to_use : gm_num, bank,
(right_samples)? "(2) " : "",
lp->lo, lp->hi, vlayer+1, vlayer_count);
for (stereo_layer = 0; stereo_layer < stereo_channels; stereo_layer++)
{
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int sample_count = 0;
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if (stereo_layer == 0) sample_count = ip->left_samples;
else if (stereo_layer == 1) sample_count = ip->right_samples;
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for (i=0; i < sample_count; i++)
{
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uint8 fractions;
int32 tmplong;
uint16 tmpshort;
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uint16 sample_volume = 0;
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uint8 tmpchar;
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Sample *sp = 0;
uint8 sf2delay = 0;
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#define READ_CHAR(thing) \
if (1 != fread(&tmpchar, 1, 1, fp)) goto fail; \
thing = tmpchar;
#define READ_SHORT(thing) \
if (1 != fread(&tmpshort, 2, 1, fp)) goto fail; \
thing = LE_SHORT(tmpshort);
#define READ_LONG(thing) \
if (1 != fread(&tmplong, 4, 1, fp)) goto fail; \
thing = LE_LONG(tmplong);
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/*
* 7 sample name
* 1 fractions
* 4 length
* 4 loop start
* 4 loop end
* 2 sample rate
* 4 low frequency
* 4 high frequency
* 2 finetune
* 1 panning
* 6 envelope rates |
* 6 envelope offsets | 18 bytes
* 3 tremolo sweep, rate, depth |
* 3 vibrato sweep, rate, depth |
* 1 sample mode
* 2 scale frequency
* 2 scale factor
* 2 sample volume (??)
* 34 reserved
* Now: 1 delay
* 33 reserved
*/
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skip(fp, 7); /* Skip the wave name */
if (1 != fread(&fractions, 1, 1, fp))
{
fail:
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "Error reading sample %d", i);
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if (stereo_layer == 1)
{
for (j=0; j<i; j++)
free(ip->right_sample[j].data);
free(ip->right_sample);
i = ip->left_samples;
}
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for (j=0; j<i; j++)
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free(ip->left_sample[j].data);
free(ip->left_sample);
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free(ip);
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return 0;
}
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if (stereo_layer == 0) sp=&(ip->left_sample[i]);
else if (stereo_layer == 1) sp=&(ip->right_sample[i]);
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READ_LONG(sp->data_length);
READ_LONG(sp->loop_start);
READ_LONG(sp->loop_end);
READ_SHORT(sp->sample_rate);
READ_LONG(sp->low_freq);
READ_LONG(sp->high_freq);
READ_LONG(sp->root_freq);
skip(fp, 2); /* Why have a "root frequency" and then "tuning"?? */
READ_CHAR(tmp[0]);
if (panning==-1)
sp->panning = (tmp[0] * 8 + 4) & 0x7f;
else
sp->panning=(uint8)(panning & 0x7F);
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sp->resonance=0;
sp->cutoff_freq=0;
sp->reverberation=0;
sp->chorusdepth=0;
sp->exclusiveClass=0;
sp->keyToModEnvHold=0;
sp->keyToModEnvDecay=0;
sp->keyToVolEnvHold=0;
sp->keyToVolEnvDecay=0;
if (cfg_tuning)
{
double tune_factor = (double)(cfg_tuning)/1200.0;
tune_factor = pow(2.0, tune_factor);
sp->root_freq = (uint32)( tune_factor * (double)sp->root_freq );
}
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/* envelope, tremolo, and vibrato */
if (18 != fread(tmp, 1, 18, fp)) goto fail;
if (!tmp[13] || !tmp[14])
{
sp->tremolo_sweep_increment=
sp->tremolo_phase_increment=sp->tremolo_depth=0;
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * no tremolo");
}
else
{
sp->tremolo_sweep_increment=convert_tremolo_sweep(tmp[12]);
sp->tremolo_phase_increment=convert_tremolo_rate(tmp[13]);
sp->tremolo_depth=tmp[14];
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" * tremolo: sweep %d, phase %d, depth %d",
sp->tremolo_sweep_increment, sp->tremolo_phase_increment,
sp->tremolo_depth);
}
if (!tmp[16] || !tmp[17])
{
sp->vibrato_sweep_increment=
sp->vibrato_control_ratio=sp->vibrato_depth=0;
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * no vibrato");
}
else
{
sp->vibrato_control_ratio=convert_vibrato_rate(tmp[16]);
sp->vibrato_sweep_increment=
convert_vibrato_sweep(tmp[15], sp->vibrato_control_ratio);
sp->vibrato_depth=tmp[17];
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" * vibrato: sweep %d, ctl %d, depth %d",
sp->vibrato_sweep_increment, sp->vibrato_control_ratio,
sp->vibrato_depth);
}
READ_CHAR(sp->modes);
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READ_SHORT(sp->freq_center);
READ_SHORT(sp->freq_scale);
if (sf2flag)
{
READ_SHORT(sample_volume);
READ_CHAR(sf2delay);
READ_CHAR(sp->exclusiveClass);
skip(fp, 32);
}
else
{
skip(fp, 36);
}
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/* Mark this as a fixed-pitch instrument if such a deed is desired. */
if (note_to_use!=-1)
sp->note_to_use=(uint8)(note_to_use);
else
sp->note_to_use=0;
/* seashore.pat in the Midia patch set has no Sustain. I don't
understand why, and fixing it by adding the Sustain flag to
all looped patches probably breaks something else. We do it
anyway. */
if (sp->modes & MODES_LOOPING)
sp->modes |= MODES_SUSTAIN;
/* Strip any loops and envelopes we're permitted to */
if ((strip_loop==1) &&
(sp->modes & (MODES_SUSTAIN | MODES_LOOPING |
MODES_PINGPONG | MODES_REVERSE)))
{
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " - Removing loop and/or sustain");
sp->modes &=~(MODES_SUSTAIN | MODES_LOOPING |
MODES_PINGPONG | MODES_REVERSE);
}
if (strip_envelope==1)
{
if (sp->modes & MODES_ENVELOPE)
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " - Removing envelope");
sp->modes &= ~MODES_ENVELOPE;
}
else if (strip_envelope != 0)
{
/* Have to make a guess. */
if (!(sp->modes & (MODES_LOOPING | MODES_PINGPONG | MODES_REVERSE)))
{
/* No loop? Then what's there to sustain? No envelope needed
either... */
sp->modes &= ~(MODES_SUSTAIN|MODES_ENVELOPE);
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" - No loop, removing sustain and envelope");
}
else if (!memcmp(tmp, "??????", 6) || tmp[11] >= 100)
{
/* Envelope rates all maxed out? Envelope end at a high "offset"?
That's a weird envelope. Take it out. */
sp->modes &= ~MODES_ENVELOPE;
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" - Weirdness, removing envelope");
}
else if (!(sp->modes & MODES_SUSTAIN))
{
/* No sustain? Then no envelope. I don't know if this is
justified, but patches without sustain usually don't need the
envelope either... at least the Gravis ones. They're mostly
drums. I think. */
sp->modes &= ~MODES_ENVELOPE;
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" - No sustain, removing envelope");
}
}
651
652
653
sp->attenuation = 0;
for (j=ATTACK; j<DELAY; j++)
654
655
{
sp->envelope_rate[j]=
656
(j<3)? convert_envelope_rate_attack(tmp[j], 11) : convert_envelope_rate(tmp[j]);
657
658
659
sp->envelope_offset[j]=
convert_envelope_offset(tmp[6+j]);
}
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
if (sf2flag)
{
if (sf2delay > 5) sf2delay = 5;
sp->envelope_rate[DELAY] = (int32)( (sf2delay*play_mode->rate) / 1000 );
}
else
{
sp->envelope_rate[DELAY]=0;
}
sp->envelope_offset[DELAY]=0;
for (j=ATTACK; j<DELAY; j++)
{
sp->modulation_rate[j]=sp->envelope_rate[j];
sp->modulation_offset[j]=sp->envelope_offset[j];
}
sp->modulation_rate[DELAY] = sp->modulation_offset[DELAY] = 0;
sp->modEnvToFilterFc=0;
sp->modEnvToPitch=0;
sp->lfo_sweep_increment = 0;
sp->lfo_phase_increment = 0;
sp->modLfoToFilterFc = 0;
sp->vibrato_delay = 0;
683
684
/* Then read the sample data */
685
686
687
688
689
690
691
if (sp->data_length/2 > MAX_SAMPLE_SIZE)
{
goto fail;
}
sp->data = safe_malloc(sp->data_length + 1);
lp->size += sp->data_length + 1;
692
693
694
695
696
697
698
if (1 != fread(sp->data, sp->data_length, 1, fp))
goto fail;
if (!(sp->modes & MODES_16BIT)) /* convert to 16-bit data */
{
int32 i=sp->data_length;
uint8 *cp=(uint8 *)(sp->data);
699
700
uint16 *tmp,*newdta;
tmp=newdta=safe_malloc(sp->data_length*2 + 2);
701
702
703
while (i--)
*tmp++ = (uint16)(*cp++) << 8;
cp=(uint8 *)(sp->data);
704
sp->data = (sample_t *)newdta;
705
706
707
708
709
free(cp);
sp->data_length *= 2;
sp->loop_start *= 2;
sp->loop_end *= 2;
}
710
#if SDL_BYTEORDER == SDL_BIG_ENDIAN
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
else
/* convert to machine byte order */
{
int32 i=sp->data_length/2;
int16 *tmp=(int16 *)sp->data,s;
while (i--)
{
s=LE_SHORT(*tmp);
*tmp++=s;
}
}
#endif
if (sp->modes & MODES_UNSIGNED) /* convert to signed data */
{
int32 i=sp->data_length/2;
int16 *tmp=(int16 *)sp->data;
while (i--)
*tmp++ ^= 0x8000;
}
/* Reverse reverse loops and pass them off as normal loops */
if (sp->modes & MODES_REVERSE)
{
int32 t;
/* The GUS apparently plays reverse loops by reversing the
whole sample. We do the same because the GUS does not SUCK. */
ctl->cmsg(CMSG_WARNING, VERB_NORMAL, "Reverse loop in %s", name);
reverse_data((int16 *)sp->data, 0, sp->data_length/2);
t=sp->loop_start;
sp->loop_start=sp->data_length - sp->loop_end;
sp->loop_end=sp->data_length - t;
sp->modes &= ~MODES_REVERSE;
sp->modes |= MODES_LOOPING; /* just in case */
}
/* If necessary do some anti-aliasing filtering */
if (antialiasing_allowed)
antialiasing(sp,play_mode->rate);
#ifdef ADJUST_SAMPLE_VOLUMES
if (amp!=-1)
757
758
759
sp->volume=(FLOAT_T)((amp) / 100.0);
else if (sf2flag)
sp->volume=(FLOAT_T)((sample_volume) / 255.0);
760
761
762
763
764
else
{
/* Try to determine a volume scaling factor for the sample.
This is a very crude adjustment, but things sound more
balanced with it. Still, this should be a runtime option. */
765
766
uint32 i, numsamps=sp->data_length/2;
uint32 higher=0, highcount=0;
767
768
int16 maxamp=0,a;
int16 *tmp=(int16 *)sp->data;
769
i = numsamps;
770
771
772
773
774
775
776
while (i--)
{
a=*tmp++;
if (a<0) a=-a;
if (a>maxamp)
maxamp=a;
}
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
tmp=(int16 *)sp->data;
i = numsamps;
while (i--)
{
a=*tmp++;
if (a<0) a=-a;
if (a > 3*maxamp/4)
{
higher += a;
highcount++;
}
}
if (highcount) higher /= highcount;
else higher = 10000;
sp->volume = (32768.0 * 0.875) / (double)higher ;
792
793
794
795
796
797
798
799
800
801
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * volume comp: %f", sp->volume);
}
#else
if (amp!=-1)
sp->volume=(double)(amp) / 100.0;
else
sp->volume=1.0;
#endif
sp->data_length /= 2; /* These are in bytes. Convert into samples. */
803
804
sp->loop_start /= 2;
sp->loop_end /= 2;
805
sp->data[sp->data_length] = sp->data[sp->data_length-1];
806
807
808
809
810
811
/* Then fractional samples */
sp->data_length <<= FRACTION_BITS;
sp->loop_start <<= FRACTION_BITS;
sp->loop_end <<= FRACTION_BITS;
812
813
814
815
816
817
818
819
820
821
822
823
/* trim off zero data at end */
{
int ls = sp->loop_start>>FRACTION_BITS;
int le = sp->loop_end>>FRACTION_BITS;
int se = sp->data_length>>FRACTION_BITS;
while (se > 1 && !sp->data[se-1]) se--;
if (le > se) le = se;
if (ls >= le) sp->modes &= ~MODES_LOOPING;
sp->loop_end = le<<FRACTION_BITS;
sp->data_length = se<<FRACTION_BITS;
}
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
/* Adjust for fractional loop points. This is a guess. Does anyone
know what "fractions" really stands for? */
sp->loop_start |=
(fractions & 0x0F) << (FRACTION_BITS-4);
sp->loop_end |=
((fractions>>4) & 0x0F) << (FRACTION_BITS-4);
/* If this instrument will always be played on the same note,
and it's not looped, we can resample it now. */
if (sp->note_to_use && !(sp->modes & MODES_LOOPING))
pre_resample(sp);
#ifdef LOOKUP_HACK
/* Squash the 16-bit data into 8 bits. */
{
uint8 *gulp,*ulp;
int16 *swp;
int l=sp->data_length >> FRACTION_BITS;
gulp=ulp=safe_malloc(l+1);
swp=(int16 *)sp->data;
while(l--)
*ulp++ = (*swp++ >> 8) & 0xFF;
free(sp->data);
sp->data=(sample_t *)gulp;
}
#endif
if (strip_tail==1)
{
/* Let's not really, just say we did. */
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " - Stripping tail");
sp->data_length = sp->loop_end;
}
857
858
859
860
} /* end of sample loop */
} /* end of stereo layer loop */
} /* end of vlayer loop */
861
862
close_file(fp);
863
return headlp;
864
865
866
867
868
869
870
871
872
873
874
875
876
}
static int fill_bank(int dr, int b)
{
int i, errors=0;
ToneBank *bank=((dr) ? drumset[b] : tonebank[b]);
if (!bank)
{
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"Huh. Tried to load instruments in non-existent %s %d",
(dr) ? "drumset" : "tone bank", b);
return 0;
}
877
for (i=0; i<MAXPROG; i++)
878
{
879
if (bank->tone[i].layer==MAGIC_LOAD_INSTRUMENT)
880
881
882
883
884
885
886
887
888
889
890
891
892
{
if (!(bank->tone[i].name))
{
ctl->cmsg(CMSG_WARNING, (b!=0) ? VERB_VERBOSE : VERB_NORMAL,
"No instrument mapped to %s %d, program %d%s",
(dr)? "drum set" : "tone bank", b, i,
(b!=0) ? "" : " - this instrument will not be heard");
if (b!=0)
{
/* Mark the corresponding instrument in the default
bank / drumset for loading (if it isn't already) */
if (!dr)
{
893
894
if (!(standard_tonebank.tone[i].layer))
standard_tonebank.tone[i].layer=
895
896
897
898
MAGIC_LOAD_INSTRUMENT;
}
else
{
899
900
if (!(standard_drumset.tone[i].layer))
standard_drumset.tone[i].layer=
901
902
903
MAGIC_LOAD_INSTRUMENT;
}
}
904
bank->tone[i].layer=0;
905
906
errors++;
}
907
else if (!(bank->tone[i].layer=
908
load_instrument(bank->tone[i].name,
909
bank->tone[i].font_type,
910
911
912
(dr) ? 1 : 0,
bank->tone[i].pan,
bank->tone[i].amp,
913
bank->tone[i].tuning,
914
(bank->tone[i].note!=-1) ?
915
916
bank->tone[i].note :
((dr) ? i : -1),
917
918
919
920
921
922
(bank->tone[i].strip_loop!=-1) ?
bank->tone[i].strip_loop :
((dr) ? 1 : -1),
(bank->tone[i].strip_envelope != -1) ?
bank->tone[i].strip_envelope :
((dr) ? 1 : -1),
923
924
925
926
927
bank->tone[i].strip_tail,
b,
((dr) ? i + 128 : i),
bank->tone[i].sf_ix
)))
928
929
930
931
932
933
934
{
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"Couldn't load instrument %s (%s %d, program %d)",
bank->tone[i].name,
(dr)? "drum set" : "tone bank", b, i);
errors++;
}
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
955
956
957
958
else
{ /* it's loaded now */
bank->tone[i].last_used = current_tune_number;
current_patch_memory += bank->tone[i].layer->size;
purge_as_required();
if (current_patch_memory > max_patch_memory) {
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"Not enough memory to load instrument %s (%s %d, program %d)",
bank->tone[i].name,
(dr)? "drum set" : "tone bank", b, i);
errors++;
free_layer(bank->tone[i].layer);
bank->tone[i].layer=0;
bank->tone[i].last_used=-1;
}
#if 0
if (check_for_rc()) {
free_layer(bank->tone[i].layer);
bank->tone[i].layer=0;
bank->tone[i].last_used=-1;
return 0;
}
#endif
}
959
960
961
962
963
}
}
return errors;
}
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
982
983
984
985
986
987
988
static void free_old_instruments(int how_old)
{
int i=MAXBANK;
while(i--)
{
if (tonebank[i])
free_old_bank(0, i, how_old);
if (drumset[i])
free_old_bank(1, i, how_old);
}
}
static void purge_as_required(void)
{
if (!max_patch_memory) return;
while (last_tune_purged < current_tune_number
&& current_patch_memory > max_patch_memory)
{
last_tune_purged++;
free_old_instruments(last_tune_purged);
}
}
989
990
int load_missing_instruments(void)
{
991
int i=MAXBANK,errors=0;
992
993
994
995
996
997
998
while (i--)
{
if (tonebank[i])
errors+=fill_bank(0,i);
if (drumset[i])
errors+=fill_bank(1,i);
}
999
current_tune_number++;
1000
return errors;