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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
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it under the terms of the Perl Artistic License, available in COPYING.
*/
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#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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{
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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(const char *name, int font_type, int percussion,
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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++)
{
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if (strlen(name)+strlen(patch_ext[i])<PATH_MAX)
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{
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char path[PATH_MAX];
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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) \
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if ((size_t)1 != fread(&tmpchar, 1, 1, fp)) goto fail; \
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thing = tmpchar;
#define READ_SHORT(thing) \
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if ((size_t)1 != fread(&tmpshort, 2, 1, fp)) goto fail; \
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thing = LE_SHORT(tmpshort);
#define READ_LONG(thing) \
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if ((size_t)1 != fread(&tmplong, 4, 1, fp)) goto fail; \
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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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free(lp);
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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(tmpshort);
sp->freq_center = (uint8)tmpshort;
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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");
}
}
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sp->attenuation = 0;
for (j=ATTACK; j<DELAY; j++)
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{
sp->envelope_rate[j]=
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(j<3)? convert_envelope_rate_attack(tmp[j], 11) : convert_envelope_rate(tmp[j]);
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sp->envelope_offset[j]=
convert_envelope_offset(tmp[6+j]);
}
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
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;
667
668
/* Then read the sample data */
669
670
671
672
if (sp->data_length/2 > MAX_SAMPLE_SIZE)
{
goto fail;
}
673
sp->data = (sample_t *)safe_malloc(sp->data_length + 1);
674
675
lp->size += sp->data_length + 1;
676
677
678
679
680
681
682
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);
683
uint16 *tmp,*newdta;
684
tmp=newdta=(uint16 *)safe_malloc(sp->data_length*2 + 2);
685
686
687
while (i--)
*tmp++ = (uint16)(*cp++) << 8;
cp=(uint8 *)(sp->data);
688
sp->data = (sample_t *)newdta;
689
690
691
692
693
free(cp);
sp->data_length *= 2;
sp->loop_start *= 2;
sp->loop_end *= 2;
}
694
#if SDL_BYTEORDER == SDL_BIG_ENDIAN
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
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
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)
741
742
743
sp->volume=(FLOAT_T)((amp) / 100.0);
else if (sf2flag)
sp->volume=(FLOAT_T)((sample_volume) / 255.0);
744
745
746
747
748
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. */
749
750
uint32 i, numsamps=sp->data_length/2;
uint32 higher=0, highcount=0;
751
752
int16 maxamp=0,a;
int16 *tmp=(int16 *)sp->data;
753
i = numsamps;
754
755
756
757
758
759
760
while (i--)
{
a=*tmp++;
if (a<0) a=-a;
if (a>maxamp)
maxamp=a;
}
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
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 ;
776
777
778
779
780
781
782
783
784
785
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. */
786
787
788
sp->loop_start /= 2;
sp->loop_end /= 2;
789
sp->data[sp->data_length] = sp->data[sp->data_length-1];
790
791
792
793
794
795
/* Then fractional samples */
sp->data_length <<= FRACTION_BITS;
sp->loop_start <<= FRACTION_BITS;
sp->loop_end <<= FRACTION_BITS;
796
797
798
799
800
801
802
803
804
805
806
807
/* 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;
}
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
/* 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;
}
841
842
843
844
} /* end of sample loop */
} /* end of stereo layer loop */
} /* end of vlayer loop */
845
846
close_file(fp);
847
return headlp;
848
849
850
851
852
853
854
855
856
857
858
859
860
}
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;
}
861
for (i=0; i<MAXPROG; i++)
862
{
863
if (bank->tone[i].layer==MAGIC_LOAD_INSTRUMENT)
864
865
866
867
868
869
870
871
872
873
874
875
876
{
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)
{
877
878
if (!(standard_tonebank.tone[i].layer))
standard_tonebank.tone[i].layer=
879
880
881
882
MAGIC_LOAD_INSTRUMENT;
}
else
{
883
884
if (!(standard_drumset.tone[i].layer))
standard_drumset.tone[i].layer=
885
886
887
MAGIC_LOAD_INSTRUMENT;
}
}
888
bank->tone[i].layer=0;
889
890
errors++;
}
891
else if (!(bank->tone[i].layer=
892
load_instrument(bank->tone[i].name,
893
bank->tone[i].font_type,
894
895
896
(dr) ? 1 : 0,
bank->tone[i].pan,
bank->tone[i].amp,
897
bank->tone[i].tuning,
898
(bank->tone[i].note!=-1) ?
899
900
bank->tone[i].note :
((dr) ? i : -1),
901
902
903
904
905
906
(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),
907
908
909
910
911
bank->tone[i].strip_tail,
b,
((dr) ? i + 128 : i),
bank->tone[i].sf_ix
)))
912
913
914
915
916
917
918
{
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++;
}
919
920
921
922
923
924
925
926
927
928
929
930
931
932
933
934
935
936
937
938
939
940
941
942
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
}
943
944
945
946
947
}
}
return errors;
}
948
949
950
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
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);
}
}
973
974
int load_missing_instruments(void)
{
975
int i=MAXBANK,errors=0;
976
977
978
979
980
981
982
while (i--)
{
if (tonebank[i])
errors+=fill_bank(0,i);
if (drumset[i])
errors+=fill_bank(1,i);
}
983
current_tune_number++;
984
985
986
987
988
989
990
991
992
993
994
995
996
997
998
return errors;
}
void free_instruments(void)
{
int i=128;
while(i--)
{
if (tonebank[i])
free_bank(0,i);
if (drumset[i])
free_bank(1,i);
}
}
999
int set_default_instrument(const char *name)
1000
{