/* TiMidity++ -- MIDI to WAVE converter and player Copyright (C) 1999-2004 Masanao Izumo Copyright (C) 1995 Tuukka Toivonen 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 instrum.c Code to load and unload GUS-compatible instrument patches. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif /* HAVE_CONFIG_H */ #include #include #include #ifndef NO_STRING_H #include #else #include #endif #include "timidity.h" #include "common.h" #include "instrum.h" #include "playmidi.h" #include "readmidi.h" #include "output.h" #include "controls.h" #include "resample.h" #include "tables.h" #include "filter.h" #include "quantity.h" #include "freq.h" #define INSTRUMENT_HASH_SIZE 128 struct InstrumentCache { char *name; int panning, amp, note_to_use, strip_loop, strip_envelope, strip_tail; Instrument *ip; struct InstrumentCache *next; }; static struct InstrumentCache *instrument_cache[INSTRUMENT_HASH_SIZE]; /* Some functions get aggravated if not even the standard banks are available. */ static ToneBank standard_tonebank, standard_drumset; ToneBank *tonebank[128 + MAP_BANK_COUNT] = {&standard_tonebank}, *drumset[128 + MAP_BANK_COUNT] = {&standard_drumset}; /* bank mapping (mapped bank) */ struct bank_map_elem { int16 used, mapid; int bankno; }; static struct bank_map_elem map_bank[MAP_BANK_COUNT], map_drumset[MAP_BANK_COUNT]; static int map_bank_counter; /* This is a special instrument, used for all melodic programs */ Instrument *default_instrument=0; SpecialPatch *special_patch[NSPECIAL_PATCH]; int progbase = 0; struct inst_map_elem { int set, elem, mapped; }; static struct inst_map_elem *inst_map_table[NUM_INST_MAP][128]; /* This is only used for tracks that don't specify a program */ int default_program[MAX_CHANNELS]; char *default_instrument_name = NULL; int antialiasing_allowed=0; #ifdef FAST_DECAY int fast_decay=1; #else int fast_decay=0; #endif /*Pseudo Reverb*/ int32 modify_release; /** below three functinos are imported from sndfont.c **/ /* convert from 8bit value to fractional offset (15.15) */ static int32 to_offset(int offset) { return (int32)offset << (7+15); } /* calculate ramp rate in fractional unit; * diff = 8bit, time = msec */ static int32 calc_rate(int diff, double msec) { double rate; if(msec < 6) msec = 6; if(diff == 0) diff = 255; diff <<= (7+15); rate = ((double)diff / play_mode->rate) * control_ratio * 1000.0 / msec; if(fast_decay) rate *= 2; return (int32)rate; } /*End of Pseudo Reverb*/ void free_instrument(Instrument *ip) { Sample *sp; int i; if (!ip) return; for (i=0; isamples; i++) { sp=&(ip->sample[i]); if(sp->data_alloced) free(sp->data); } free(ip->sample); free(ip); } void clear_magic_instruments(void) { int i, j; for(j = 0; j < 128 + map_bank_counter; j++) { if(tonebank[j]) { ToneBank *bank = tonebank[j]; for(i = 0; i < 128; i++) if(IS_MAGIC_INSTRUMENT(bank->tone[i].instrument)) bank->tone[i].instrument = NULL; } if(drumset[j]) { ToneBank *bank = drumset[j]; for(i = 0; i < 128; i++) if(IS_MAGIC_INSTRUMENT(bank->tone[i].instrument)) bank->tone[i].instrument = NULL; } } } #define GUS_ENVRATE_MAX (int32)(0x3FFFFFFF >> 9) static int32 convert_envelope_rate(uint8 rate) { int32 r; r=3-((rate>>6) & 0x3); r*=3; r = (int32)(rate & 0x3f) << r; /* 6.9 fixed point */ /* 15.15 fixed point. */ r = r * 44100 / play_mode->rate * control_ratio * (1 << fast_decay); if(r > GUS_ENVRATE_MAX) {r = GUS_ENVRATE_MAX;} return (r << 9); } static int32 convert_envelope_offset(uint8 offset) { /* 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); } static int32 convert_tremolo_sweep(uint8 sweep) { if (!sweep) return 0; return ((control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) / (play_mode->rate * sweep); } static int32 convert_vibrato_sweep(uint8 sweep, int32 vib_control_ratio) { if (!sweep) return 0; return (int32)(TIM_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); */ } static int32 convert_tremolo_rate(uint8 rate) { return ((SINE_CYCLE_LENGTH * control_ratio * rate) << RATE_SHIFT) / (TREMOLO_RATE_TUNING * play_mode->rate); } static int32 convert_vibrato_rate(uint8 rate) { /* 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; int32 i; sp+=ls; le-=ls; le/=2; for(i = 0; i < le; i++) { s=*sp; *sp++=*ep; *ep--=s; } } static int name_hash(char *name) { unsigned int addr = 0; while(*name) addr += *name++; return addr % INSTRUMENT_HASH_SIZE; } static Instrument *search_instrument_cache(char *name, int panning, int amp, int note_to_use, int strip_loop, int strip_envelope, int strip_tail) { struct InstrumentCache *p; for(p = instrument_cache[name_hash(name)]; p != NULL; p = p->next) { if(strcmp(p->name, name) != 0) return NULL; if(p->panning == panning && p->amp == amp && p->note_to_use == note_to_use && p->strip_loop == strip_loop && p->strip_envelope == strip_envelope && p->strip_tail == strip_tail) return p->ip; } return NULL; } static void store_instrument_cache(Instrument *ip, char *name, int panning, int amp, int note_to_use, int strip_loop, int strip_envelope, int strip_tail) { struct InstrumentCache *p; int addr; addr = name_hash(name); p = (struct InstrumentCache *)safe_malloc(sizeof(struct InstrumentCache)); p->next = instrument_cache[addr]; instrument_cache[addr] = p; p->name = name; p->panning = panning; p->amp = amp; p->note_to_use = note_to_use; p->strip_loop = strip_loop; p->strip_envelope = strip_envelope; p->strip_tail = strip_tail; p->ip = ip; } static int32 adjust_tune_freq(int32 val, float tune) { if (! tune) return val; return val / pow(2.0, tune / 12.0); } static int16 adjust_scale_tune(int16 val) { return 1024 * (double) val / 100 + 0.5; } static int16 adjust_fc(int16 val) { if (val < 0 || val > play_mode->rate / 2) { return 0; } else { return val; } } static int16 adjust_reso(int16 val) { if (val < 0 || val > 960) { return 0; } else { return val; } } static int32 to_rate(int rate) { return (rate) ? (int32) (0x200 * pow(2.0, rate / 17.0) * 44100 / play_mode->rate * control_ratio) << fast_decay : 0; } #if 0 static int32 to_control(int control) { return (int32) (0x2000 / pow(2.0, control / 31.0)); } #endif static void apply_bank_parameter(Instrument *ip, ToneBankElement *tone) { int i, j; Sample *sp; if (tone->tunenum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->tunenum == 1) { sp->low_freq = adjust_tune_freq(sp->low_freq, tone->tune[0]); sp->high_freq = adjust_tune_freq(sp->high_freq, tone->tune[0]); sp->root_freq = adjust_tune_freq(sp->root_freq, tone->tune[0]); } else if (i < tone->tunenum) { sp->low_freq = adjust_tune_freq(sp->low_freq, tone->tune[i]); sp->high_freq = adjust_tune_freq(sp->high_freq, tone->tune[i]); sp->root_freq = adjust_tune_freq(sp->root_freq, tone->tune[i]); } } if (tone->envratenum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->envratenum == 1) { for (j = 0; j < 6; j++) if (tone->envrate[0][j] >= 0) sp->envelope_rate[j] = to_rate(tone->envrate[0][j]); } else if (i < tone->envratenum) { for (j = 0; j < 6; j++) if (tone->envrate[i][j] >= 0) sp->envelope_rate[j] = to_rate(tone->envrate[i][j]); } } if (tone->envofsnum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->envofsnum == 1) { for (j = 0; j < 6; j++) if (tone->envofs[0][j] >= 0) sp->envelope_offset[j] = to_offset(tone->envofs[0][j]); } else if (i < tone->envofsnum) { for (j = 0; j < 6; j++) if (tone->envofs[i][j] >= 0) sp->envelope_offset[j] = to_offset(tone->envofs[i][j]); } } if (tone->tremnum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->tremnum == 1) { if (IS_QUANTITY_DEFINED(tone->trem[0][0])) sp->tremolo_sweep_increment = quantity_to_int(&tone->trem[0][0], 0); if (IS_QUANTITY_DEFINED(tone->trem[0][1])) sp->tremolo_phase_increment = quantity_to_int(&tone->trem[0][1], 0); if (IS_QUANTITY_DEFINED(tone->trem[0][2])) sp->tremolo_depth = quantity_to_int(&tone->trem[0][2], 0) << 1; } else if (i < tone->tremnum) { if (IS_QUANTITY_DEFINED(tone->trem[i][0])) sp->tremolo_sweep_increment = quantity_to_int(&tone->trem[i][0], 0); if (IS_QUANTITY_DEFINED(tone->trem[i][1])) sp->tremolo_phase_increment = quantity_to_int(&tone->trem[i][1], 0); if (IS_QUANTITY_DEFINED(tone->trem[i][2])) sp->tremolo_depth = quantity_to_int(&tone->trem[i][2], 0) << 1; } } if (tone->vibnum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->vibnum == 1) { if (IS_QUANTITY_DEFINED(tone->vib[0][1])) sp->vibrato_control_ratio = quantity_to_int(&tone->vib[0][1], 0); if (IS_QUANTITY_DEFINED(tone->vib[0][0])) sp->vibrato_sweep_increment = quantity_to_int(&tone->vib[0][0], sp->vibrato_control_ratio); if (IS_QUANTITY_DEFINED(tone->vib[0][2])) sp->vibrato_depth = quantity_to_int(&tone->vib[0][2], 0); } else if (i < tone->vibnum) { if (IS_QUANTITY_DEFINED(tone->vib[i][1])) sp->vibrato_control_ratio = quantity_to_int(&tone->vib[i][1], 0); if (IS_QUANTITY_DEFINED(tone->vib[i][0])) sp->vibrato_sweep_increment = quantity_to_int(&tone->vib[i][0], sp->vibrato_control_ratio); if (IS_QUANTITY_DEFINED(tone->vib[i][2])) sp->vibrato_depth = quantity_to_int(&tone->vib[i][2], 0); } } if (tone->sclnotenum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->sclnotenum == 1) sp->scale_freq = tone->sclnote[0]; else if (i < tone->sclnotenum) sp->scale_freq = tone->sclnote[i]; } if (tone->scltunenum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->scltunenum == 1) sp->scale_factor = adjust_scale_tune(tone->scltune[0]); else if (i < tone->scltunenum) sp->scale_factor = adjust_scale_tune(tone->scltune[i]); } if (tone->modenvratenum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->modenvratenum == 1) { for (j = 0; j < 6; j++) if (tone->modenvrate[0][j] >= 0) sp->modenv_rate[j] = to_rate(tone->modenvrate[0][j]); } else if (i < tone->modenvratenum) { for (j = 0; j < 6; j++) if (tone->modenvrate[i][j] >= 0) sp->modenv_rate[j] = to_rate(tone->modenvrate[i][j]); } } if (tone->modenvofsnum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->modenvofsnum == 1) { for (j = 0; j < 6; j++) if (tone->modenvofs[0][j] >= 0) sp->modenv_offset[j] = to_offset(tone->modenvofs[0][j]); } else if (i < tone->modenvofsnum) { for (j = 0; j < 6; j++) if (tone->modenvofs[i][j] >= 0) sp->modenv_offset[j] = to_offset(tone->modenvofs[i][j]); } } if (tone->envkeyfnum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->envkeyfnum == 1) { for (j = 0; j < 6; j++) if (tone->envkeyf[0][j] != -1) sp->envelope_keyf[j] = tone->envkeyf[0][j]; } else if (i < tone->envkeyfnum) { for (j = 0; j < 6; j++) if (tone->envkeyf[i][j] != -1) sp->envelope_keyf[j] = tone->envkeyf[i][j]; } } if (tone->envvelfnum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->envvelfnum == 1) { for (j = 0; j < 6; j++) if (tone->envvelf[0][j] != -1) sp->envelope_velf[j] = tone->envvelf[0][j]; } else if (i < tone->envvelfnum) { for (j = 0; j < 6; j++) if (tone->envvelf[i][j] != -1) sp->envelope_velf[j] = tone->envvelf[i][j]; } } if (tone->modenvkeyfnum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->modenvkeyfnum == 1) { for (j = 0; j < 6; j++) if (tone->modenvkeyf[0][j] != -1) sp->modenv_keyf[j] = tone->modenvkeyf[0][j]; } else if (i < tone->modenvkeyfnum) { for (j = 0; j < 6; j++) if (tone->modenvkeyf[i][j] != -1) sp->modenv_keyf[j] = tone->modenvkeyf[i][j]; } } if (tone->modenvvelfnum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->modenvvelfnum == 1) { for (j = 0; j < 6; j++) if (tone->modenvvelf[0][j] != -1) sp->modenv_velf[j] = tone->modenvvelf[0][j]; } else if (i < tone->modenvvelfnum) { for (j = 0; j < 6; j++) if (tone->modenvvelf[i][j] != -1) sp->modenv_velf[j] = tone->modenvvelf[i][j]; } } if (tone->trempitchnum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->trempitchnum == 1) sp->tremolo_to_pitch = tone->trempitch[0]; else if (i < tone->trempitchnum) sp->tremolo_to_pitch = tone->trempitch[i]; } if (tone->tremfcnum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->tremfcnum == 1) sp->tremolo_to_fc = tone->tremfc[0]; else if (i < tone->tremfcnum) sp->tremolo_to_fc = tone->tremfc[i]; } if (tone->modpitchnum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->modpitchnum == 1) sp->modenv_to_pitch = tone->modpitch[0]; else if (i < tone->modpitchnum) sp->modenv_to_pitch = tone->modpitch[i]; } if (tone->modfcnum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->modfcnum == 1) sp->modenv_to_fc = tone->modfc[0]; else if (i < tone->modfcnum) sp->modenv_to_fc = tone->modfc[i]; } if (tone->fcnum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->fcnum == 1) sp->cutoff_freq = adjust_fc(tone->fc[0]); else if (i < tone->fcnum) sp->cutoff_freq = adjust_fc(tone->fc[i]); } if (tone->resonum) for (i = 0; i < ip->samples; i++) { sp = &ip->sample[i]; if (tone->resonum == 1) sp->resonance = adjust_reso(tone->reso[0]); else if (i < tone->resonum) sp->resonance = adjust_reso(tone->reso[i]); } } #define READ_CHAR(thing) { \ uint8 tmpchar; \ \ if (tf_read(&tmpchar, 1, 1, tf) != 1) \ goto fail; \ thing = tmpchar; \ } #define READ_SHORT(thing) { \ uint16 tmpshort; \ \ if (tf_read(&tmpshort, 2, 1, tf) != 1) \ goto fail; \ thing = LE_SHORT(tmpshort); \ } #define READ_LONG(thing) { \ int32 tmplong; \ \ if (tf_read(&tmplong, 4, 1, tf) != 1) \ goto fail; \ thing = LE_LONG(tmplong); \ } /* 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 */ static Instrument *load_gus_instrument(char *name, ToneBank *bank, int dr, int prog, char *infomsg) { ToneBankElement *tone; int amp, note_to_use, panning, strip_envelope, strip_loop, strip_tail; Instrument *ip; struct timidity_file *tf; uint8 tmp[1024], fractions; Sample *sp; int i, j, noluck = 0; if (! name) return 0; if (infomsg != NULL) ctl->cmsg(CMSG_INFO, VERB_NOISY, "%s: %s", infomsg, name); else ctl->cmsg(CMSG_INFO, VERB_NOISY, "Loading instrument %s", name); if (bank) { tone = &bank->tone[prog]; amp = tone->amp; note_to_use = (tone->note != -1) ? tone->note : ((dr) ? prog : -1); panning = tone->pan; strip_envelope = (tone->strip_envelope != -1) ? tone->strip_envelope : ((dr) ? 1 : -1); strip_loop = (tone->strip_loop != -1) ? tone->strip_loop : ((dr) ? 1 : -1); strip_tail = tone->strip_tail; } else { tone = NULL; amp = note_to_use = panning = -1; strip_envelope = strip_loop = strip_tail = 0; } if (tone && tone->tunenum == 0 && tone->envratenum == 0 && tone->envofsnum == 0 && tone->tremnum == 0 && tone->vibnum == 0 && tone->sclnotenum == 0 && tone->scltunenum == 0 && tone->modenvratenum == 0 && tone->modenvofsnum == 0 && tone->envkeyfnum == 0 && tone->envvelfnum == 0 && tone->modenvkeyfnum == 0 && tone->modenvvelfnum == 0 && tone->trempitchnum == 0 && tone->tremfcnum == 0 && tone->modpitchnum == 0 && tone->modfcnum == 0 && tone->fcnum == 0 && tone->resonum == 0) if ((ip = search_instrument_cache(name, panning, amp, note_to_use, strip_loop, strip_envelope, strip_tail)) != NULL) { ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * Cached"); return ip; } /* Open patch file */ if (! (tf = open_file(name, 2, OF_NORMAL))) { #ifdef PATCH_EXT_LIST int name_len, ext_len; static char *patch_ext[] = PATCH_EXT_LIST; #endif noluck = 1; #ifdef PATCH_EXT_LIST name_len = strlen(name); /* Try with various extensions */ for (i = 0; patch_ext[i]; i++) { ext_len = strlen(patch_ext[i]); if (name_len + ext_len < 1024) { if (name_len >= ext_len && strcmp(name + name_len - ext_len, patch_ext[i]) == 0) continue; /* duplicated ext. */ strcpy((char *) tmp, name); strcat((char *) tmp, patch_ext[i]); if ((tf = open_file((char *) tmp, 1, OF_NORMAL))) { noluck = 0; break; } } } #endif } if (noluck) { ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "Instrument `%s' can't be found.", name); return 0; } /* Read some headers and do cursory sanity checks. There are loads * of magic offsets. This could be rewritten... */ tmp[0] = tf_getc(tf); if (tmp[0] == '\0') { /* for Mac binary */ skip(tf, 127); tmp[0] = tf_getc(tf); } if ((tf_read(tmp + 1, 1, 238, tf) != 238) || (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); close_file(tf); return 0; } /* instruments. To some patch makers, 0 means 1 */ if (tmp[82] != 1 && tmp[82] != 0) { ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "Can't handle patches with %d instruments", tmp[82]); close_file(tf); 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]); close_file(tf); return 0; } ip = (Instrument *) safe_malloc(sizeof(Instrument)); ip->type = INST_GUS; ip->samples = tmp[198]; ip->sample = (Sample *) safe_malloc(sizeof(Sample) * ip->samples); memset(ip->sample, 0, sizeof(Sample) * ip->samples); for (i = 0; i < ip->samples; i++) { skip(tf, 7); /* Skip the wave name */ if (tf_read(&fractions, 1, 1, tf) != 1) { fail: ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "Error reading sample %d", i); for (j = 0; j < i; j++) free(ip->sample[j].data); free(ip->sample); free(ip); close_file(tf); return 0; } sp = &(ip->sample[i]); sp->low_vel = 0; sp->high_vel = 127; sp->cutoff_freq = sp->resonance = 0; sp->tremolo_to_pitch = sp->tremolo_to_fc = 0; sp->modenv_to_pitch = sp->modenv_to_fc = 0; sp->vel_to_fc = sp->key_to_fc = sp->vel_to_resonance = 0; sp->envelope_velf_bpo = sp->modenv_velf_bpo = 64; sp->vel_to_fc_threshold = 64; sp->key_to_fc_bpo = 60; sp->envelope_delay = sp->modenv_delay = 0; sp->tremolo_delay = sp->vibrato_delay = 0; sp->inst_type = INST_GUS; sp->sample_type = SF_SAMPLETYPE_MONO; sp->sf_sample_link = -1; sp->sf_sample_index = 0; memset(sp->envelope_velf, 0, sizeof(sp->envelope_velf)); memset(sp->envelope_keyf, 0, sizeof(sp->envelope_keyf)); memset(sp->modenv_velf, 0, sizeof(sp->modenv_velf)); memset(sp->modenv_keyf, 0, sizeof(sp->modenv_keyf)); memset(sp->modenv_rate, 0, sizeof(sp->modenv_rate)); memset(sp->modenv_offset, 0, sizeof(sp->modenv_offset)); 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(tf, 2); /* Why have a "root frequency" and then "tuning"?? */ READ_CHAR(tmp[0]); ctl->cmsg(CMSG_INFO, VERB_DEBUG, "Rate/Low/Hi/Root = %d/%d/%d/%d", sp->sample_rate, sp->low_freq, sp->high_freq, sp->root_freq); if (panning == -1) /* 0x07 and 0x08 are both center panning */ sp->panning = ((tmp[0] - ((tmp[0] < 8) ? 7 : 8)) * 63) / 7 + 64; else sp->panning = (uint8) (panning & 0x7f); /* envelope, tremolo, and vibrato */ if (tf_read(tmp, 1, 18, tf) != 18) goto fail; if (! tmp[13] || ! tmp[14]) { sp->tremolo_sweep_increment = sp->tremolo_phase_increment = 0; 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 = 0; 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); ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * mode: 0x%02x", sp->modes); READ_SHORT(sp->scale_freq); READ_SHORT(sp->scale_factor); skip(tf, 36); /* skip reserved space */ /* Mark this as a fixed-pitch instrument if such a deed is desired. */ sp->note_to_use = (note_to_use != -1) ? (uint8) 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))) { sp->modes &= ~(MODES_SUSTAIN | MODES_LOOPING | MODES_PINGPONG | MODES_REVERSE); ctl->cmsg(CMSG_INFO, VERB_DEBUG, " - Removing loop and/or sustain"); } 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"); } } for (j = 0; j < 6; j++) { sp->envelope_rate[j]= convert_envelope_rate(tmp[j]); sp->envelope_offset[j] = convert_envelope_offset(tmp[j + 6]); } /* this envelope seems to give reverb like effects to most patches * use the same method as soundfont */ if (modify_release) { sp->envelope_offset[3] = to_offset(5); sp->envelope_rate[3] = calc_rate(255, modify_release); sp->envelope_offset[4] = to_offset(4); sp->envelope_rate[4] = to_offset(200); sp->envelope_offset[5] = to_offset(4); sp->envelope_rate[5] = to_offset(200); } /* Then read the sample data */ sp->data = (sample_t *) safe_malloc(sp->data_length + 4); sp->data_alloced = 1; if ((j = tf_read(sp->data, 1, sp->data_length, tf)) != sp->data_length) { ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "Too small this patch length: %d < %d", j, sp->data_length); goto fail; } if (! (sp->modes & MODES_16BIT)) { /* convert to 16-bit data */ int32 i; uint16 *tmp; uint8 *cp = (uint8 *) sp->data; tmp = (uint16 *) safe_malloc(sp->data_length * 2 + 4); for (i = 0; i < sp->data_length; i++) tmp[i] = (uint16) cp[i] << 8; sp->data = (sample_t *) tmp; free(cp); sp->data_length *= 2; sp->loop_start *= 2; sp->loop_end *= 2; } #ifndef LITTLE_ENDIAN else { /* convert to machine byte order */ int32 i; int16 *tmp = (int16 *) sp->data, s; for (i = 0; i < sp->data_length / 2; i++) s = LE_SHORT(tmp[i]), tmp[i] = 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 loops and pass them off as normal loops */ if (sp->modes & MODES_REVERSE) { /* The GUS apparently plays reverse loops by reversing the * whole sample. We do the same because the GUS does not SUCK. */ int32 t; 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 */ ctl->cmsg(CMSG_WARNING, VERB_NORMAL, "Reverse loop in %s", name); } /* If necessary do some anti-aliasing filtering */ if (antialiasing_allowed) antialiasing((int16 *) sp->data, sp->data_length / 2, sp->sample_rate, play_mode->rate); #ifdef ADJUST_SAMPLE_VOLUMES if (amp != -1) sp->volume = (double) amp / 100; 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. */ int32 i, a, maxamp = 0; int16 *tmp = (int16 *) sp->data; for (i = 0; i < sp->data_length / 2; i++) if ((a = abs(tmp[i])) > maxamp) maxamp = a; sp->volume = 32768 / (double) maxamp; ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * volume comp: %f", sp->volume); } #else sp->volume = (amp != -1) ? (double) amp / 100 : 1.0; #endif /* These are in bytes. Convert into samples. */ sp->data_length /= 2; sp->loop_start /= 2; sp->loop_end /= 2; /* The sample must be padded out by 2 extra sample, so that * round off errors in the offsets used in interpolation will not * cause a "pop" by reading random data beyond data_length */ sp->data[sp->data_length] = sp->data[sp->data_length + 1] = 0; /* Remove abnormal loops which cause pop noise * in long sustain stage */ if (! (sp->modes & MODES_LOOPING)) { sp->loop_start = sp->data_length - 1; sp->loop_end = sp->data_length; sp->data[sp->data_length - 1] = 0; } /* Then fractional samples */ sp->data_length <<= FRACTION_BITS; sp->loop_start <<= FRACTION_BITS; sp->loop_end <<= FRACTION_BITS; /* 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); /* do pitch detection on drums if surround chorus is used */ if (dr && opt_surround_chorus) { sp->chord = -1; sp->root_freq_detected = freq_fourier(sp, &(sp->chord)); sp->transpose_detected = assign_pitch_to_freq(sp->root_freq_detected) - assign_pitch_to_freq(sp->root_freq / 1024.0); } #ifdef LOOKUP_HACK squash_sample_16to8(sp); #endif if (strip_tail == 1) { /* Let's not really, just say we did. */ sp->data_length = sp->loop_end; ctl->cmsg(CMSG_INFO, VERB_DEBUG, " - Stripping tail"); } } close_file(tf); store_instrument_cache(ip, name, panning, amp, note_to_use, strip_loop, strip_envelope, strip_tail); return ip; } #ifdef LOOKUP_HACK /*! Squash the 16-bit data into 8 bits. */ void squash_sample_16to8(Sample *sp) { uint8 *gulp, *ulp; int16 *swp; int l = sp->data_length >> FRACTION_BITS; gulp = ulp = (uint8 *)safe_malloc(l + 1); swp = (int16 *)sp->data; while (l--) *ulp++ = (*swp++ >> 8) & 0xff; free(sp->data); sp->data = (sample_t *)gulp; } #endif Instrument *load_instrument(int dr, int b, int prog) { ToneBank *bank = ((dr) ? drumset[b] : tonebank[b]); Instrument *ip; int i, font_bank, font_preset, font_keynote; extern Instrument *extract_sample_file(char *); FLOAT_T volume_max; int pan, panning; char infomsg[256]; #ifndef CFG_FOR_SF if (play_system_mode == GS_SYSTEM_MODE && (b == 64 || b == 65)) if (! dr) /* User Instrument */ recompute_userinst(b, prog); else /* User Drumset */ recompute_userdrum(b, prog); #endif if (bank->tone[prog].instype == 1 || bank->tone[prog].instype == 2) { if (bank->tone[prog].instype == 1) { /* Font extention */ font_bank = bank->tone[prog].font_bank; font_preset = bank->tone[prog].font_preset; font_keynote = bank->tone[prog].font_keynote; ip = extract_soundfont(bank->tone[prog].name, font_bank, font_preset, font_keynote); } else /* Sample extension */ ip = extract_sample_file(bank->tone[prog].name); /* amp tuning */ if (ip != NULL && bank->tone[prog].amp != -1) { for (i = 0, volume_max = 0; i < ip->samples; i++) if (volume_max < ip->sample[i].volume) volume_max = ip->sample[i].volume; if (volume_max != 0) for (i = 0; i < ip->samples; i++) ip->sample[i].volume *= bank->tone[prog].amp / 100.0 / volume_max; } /* panning */ if (ip != NULL && bank->tone[prog].pan != -1) { pan = ((int) bank->tone[prog].pan & 0x7f) - 64; for (i = 0; i < ip->samples; i++) { panning = (int) ip->sample[i].panning + pan; panning = (panning < 0) ? 0 : ((panning > 127) ? 127 : panning); ip->sample[i].panning = panning; } } /* note to use */ if (ip != NULL && bank->tone[prog].note != -1) for (i = 0; i < ip->samples; i++) ip->sample[i].root_freq = freq_table[bank->tone[prog].note & 0x7f]; /* filter key-follow */ if (ip != NULL && bank->tone[prog].key_to_fc != 0) for (i = 0; i < ip->samples; i++) ip->sample[i].key_to_fc = bank->tone[prog].key_to_fc; /* filter velocity-follow */ if (ip != NULL && bank->tone[prog].vel_to_fc != 0) for (i = 0; i < ip->samples; i++) ip->sample[i].key_to_fc = bank->tone[prog].vel_to_fc; /* resonance velocity-follow */ if (ip != NULL && bank->tone[prog].vel_to_resonance != 0) for (i = 0; i < ip->samples; i++) ip->sample[i].vel_to_resonance = bank->tone[prog].vel_to_resonance; /* strip tail */ if (ip != NULL && bank->tone[prog].strip_tail == 1) for (i = 0; i < ip->samples; i++) ip->sample[i].data_length = ip->sample[i].loop_end; if (ip != NULL) { i = (dr) ? 0 : prog; if (bank->tone[i].comment) free(bank->tone[i].comment); bank->tone[i].comment = safe_strdup(ip->instname); apply_bank_parameter(ip, &bank->tone[prog]); } return ip; } if (! dr) { font_bank = b; font_preset = prog; font_keynote = -1; } else { font_bank = 128; font_preset = b; font_keynote = prog; } /* preload soundfont */ ip = load_soundfont_inst(0, font_bank, font_preset, font_keynote); if (ip != NULL) { if (bank->tone[prog].comment) free(bank->tone[prog].comment); bank->tone[prog].comment = safe_strdup(ip->instname); } if (ip == NULL) { /* load GUS/patch file */ if (! dr) sprintf(infomsg, "Tonebank %d %d", b, prog + progbase); else sprintf(infomsg, "Drumset %d %d(%s)", b + progbase, prog, note_name[prog % 12]); ip = load_gus_instrument(bank->tone[prog].name, bank, dr, prog, infomsg); if (ip == NULL) { /* no patch; search soundfont again */ ip = load_soundfont_inst(1, font_bank, font_preset, font_keynote); if (ip != NULL) { if (bank->tone[0].comment) free(bank->tone[0].comment); bank->tone[0].comment = safe_strdup(ip->instname); } } } if (ip != NULL) apply_bank_parameter(ip, &bank->tone[prog]); return ip; } static int fill_bank(int dr, int b, int *rc) { int i, errors = 0; ToneBank *bank=((dr) ? drumset[b] : tonebank[b]); if(rc != NULL) *rc = RC_NONE; for(i = 0; i < 128; i++) { if(bank->tone[i].instrument == MAGIC_LOAD_INSTRUMENT) { if(!(bank->tone[i].name)) { bank->tone[i].instrument = load_instrument(dr, b, i); if(bank->tone[i].instrument == NULL) { ctl->cmsg(CMSG_WARNING, (b != 0) ? VERB_VERBOSE : VERB_NORMAL, "No instrument mapped to %s %d, program %d%s", dr ? "drum set" : "tone bank", dr ? b+progbase : b, dr ? i : i+progbase, (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) { if(!(standard_tonebank.tone[i].instrument)) standard_tonebank.tone[i].instrument = MAGIC_LOAD_INSTRUMENT; } else { if(!(standard_drumset.tone[i].instrument)) standard_drumset.tone[i].instrument = MAGIC_LOAD_INSTRUMENT; } bank->tone[i].instrument = 0; } else bank->tone[i].instrument = MAGIC_ERROR_INSTRUMENT; errors++; } } else { if(rc != NULL) { *rc = check_apply_control(); if(RC_IS_SKIP_FILE(*rc)) return errors; } bank->tone[i].instrument = load_instrument(dr, b, i); if(!bank->tone[i].instrument) { ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "Couldn't load instrument %s " "(%s %d, program %d)", bank->tone[i].name, dr ? "drum set" : "tone bank", dr ? b+progbase : b, dr ? i : i+progbase); errors++; } } } } return errors; } int load_missing_instruments(int *rc) { int i = 128 + map_bank_counter, errors = 0; if(rc != NULL) *rc = RC_NONE; while (i--) { if (tonebank[i]) errors+=fill_bank(0,i,rc); if(rc != NULL && RC_IS_SKIP_FILE(*rc)) return errors; if (drumset[i]) errors+=fill_bank(1,i,rc); if(rc != NULL && RC_IS_SKIP_FILE(*rc)) return errors; } return errors; } static void *safe_memdup(void *s, size_t size) { return memcpy(safe_malloc(size), s, size); } /*! Copy ToneBankElement src to elm. The original elm is released. */ void copy_tone_bank_element(ToneBankElement *elm, const ToneBankElement *src) { int i; free_tone_bank_element(elm); memcpy(elm, src, sizeof(ToneBankElement)); if (elm->name) elm->name = safe_strdup(elm->name); if (elm->tunenum) elm->tune = (float *) safe_memdup(elm->tune, elm->tunenum * sizeof(float)); if (elm->envratenum) { elm->envrate = (int **) safe_memdup(elm->envrate, elm->envratenum * sizeof(int *)); for (i = 0; i < elm->envratenum; i++) elm->envrate[i] = (int *) safe_memdup(elm->envrate[i], 6 * sizeof(int)); } if (elm->envofsnum) { elm->envofs = (int **) safe_memdup(elm->envofs, elm->envofsnum * sizeof(int *)); for (i = 0; i < elm->envofsnum; i++) elm->envofs[i] = (int *) safe_memdup(elm->envofs[i], 6 * sizeof(int)); } if (elm->tremnum) { elm->trem = (Quantity **) safe_memdup(elm->trem, elm->tremnum * sizeof(Quantity *)); for (i = 0; i < elm->tremnum; i++) elm->trem[i] = (Quantity *) safe_memdup(elm->trem[i], 3 * sizeof(Quantity)); } if (elm->vibnum) { elm->vib = (Quantity **) safe_memdup(elm->vib, elm->vibnum * sizeof(Quantity *)); for (i = 0; i < elm->vibnum; i++) elm->vib[i] = (Quantity *) safe_memdup(elm->vib[i], 3 * sizeof(Quantity)); } if (elm->sclnotenum) elm->sclnote = (int16 *) safe_memdup(elm->sclnote, elm->sclnotenum * sizeof(int16)); if (elm->scltunenum) elm->scltune = (int16 *) safe_memdup(elm->scltune, elm->scltunenum * sizeof(int16)); if (elm->comment) elm->comment = safe_strdup(elm->comment); if (elm->modenvratenum) { elm->modenvrate = (int **) safe_memdup(elm->modenvrate, elm->modenvratenum * sizeof(int *)); for (i = 0; i < elm->modenvratenum; i++) elm->modenvrate[i] = (int *) safe_memdup(elm->modenvrate[i], 6 * sizeof(int)); } if (elm->modenvofsnum) { elm->modenvofs = (int **) safe_memdup(elm->modenvofs, elm->modenvofsnum * sizeof(int *)); for (i = 0; i < elm->modenvofsnum; i++) elm->modenvofs[i] = (int *) safe_memdup(elm->modenvofs[i], 6 * sizeof(int)); } if (elm->envkeyfnum) { elm->envkeyf = (int **) safe_memdup(elm->envkeyf, elm->envkeyfnum * sizeof(int *)); for (i = 0; i < elm->envkeyfnum; i++) elm->envkeyf[i] = (int *) safe_memdup(elm->envkeyf[i], 6 * sizeof(int)); } if (elm->envvelfnum) { elm->envvelf = (int **) safe_memdup(elm->envvelf, elm->envvelfnum * sizeof(int *)); for (i = 0; i < elm->envvelfnum; i++) elm->envvelf[i] = (int *) safe_memdup(elm->envvelf[i], 6 * sizeof(int)); } if (elm->modenvkeyfnum) { elm->modenvkeyf = (int **) safe_memdup(elm->modenvkeyf, elm->modenvkeyfnum * sizeof(int *)); for (i = 0; i < elm->modenvkeyfnum; i++) elm->modenvkeyf[i] = (int *) safe_memdup(elm->modenvkeyf[i], 6 * sizeof(int)); } if (elm->modenvvelfnum) { elm->modenvvelf = (int **) safe_memdup(elm->modenvvelf, elm->modenvvelfnum * sizeof(int *)); for (i = 0; i < elm->modenvvelfnum; i++) elm->modenvvelf[i] = (int *) safe_memdup(elm->modenvvelf[i], 6 * sizeof(int)); } if (elm->trempitchnum) elm->trempitch = (int16 *) safe_memdup(elm->trempitch, elm->trempitchnum * sizeof(int16)); if (elm->tremfcnum) elm->tremfc = (int16 *) safe_memdup(elm->tremfc, elm->tremfcnum * sizeof(int16)); if (elm->modpitchnum) elm->modpitch = (int16 *) safe_memdup(elm->modpitch, elm->modpitchnum * sizeof(int16)); if (elm->modfcnum) elm->modfc = (int16 *) safe_memdup(elm->modfc, elm->modfcnum * sizeof(int16)); if (elm->fcnum) elm->fc = (int16 *) safe_memdup(elm->fc, elm->fcnum * sizeof(int16)); if (elm->resonum) elm->reso = (int16 *) safe_memdup(elm->reso, elm->resonum * sizeof(int16)); } /*! Release ToneBank[128 + MAP_BANK_COUNT] */ static void free_tone_bank_list(ToneBank *tb[]) { int i, j; ToneBank *bank; for (i = 0; i < 128 + map_bank_counter; i++) { bank = tb[i]; if (!bank) continue; for (j = 0; j < 128; j++) free_tone_bank_element(&bank->tone[j]); if (i > 0) { free(bank); tb[i] = NULL; } } } /*! Release tonebank and drumset */ void free_tone_bank(void) { free_tone_bank_list(tonebank); free_tone_bank_list(drumset); } /*! Release ToneBankElement. */ void free_tone_bank_element(ToneBankElement *elm) { elm->instype = 0; if (elm->name) free(elm->name); elm->name = NULL; if (elm->tune) free(elm->tune); elm->tune = NULL, elm->tunenum = 0; if (elm->envratenum) free_ptr_list(elm->envrate, elm->envratenum); elm->envrate = NULL, elm->envratenum = 0; if (elm->envofsnum) free_ptr_list(elm->envofs, elm->envofsnum); elm->envofs = NULL, elm->envofsnum = 0; if (elm->tremnum) free_ptr_list(elm->trem, elm->tremnum); elm->trem = NULL, elm->tremnum = 0; if (elm->vibnum) free_ptr_list(elm->vib, elm->vibnum); elm->vib = NULL, elm->vibnum = 0; if (elm->sclnote) free(elm->sclnote); elm->sclnote = NULL, elm->sclnotenum = 0; if (elm->scltune) free(elm->scltune); elm->scltune = NULL, elm->scltunenum = 0; if (elm->comment) free(elm->comment); elm->comment = NULL; if (elm->modenvratenum) free_ptr_list(elm->modenvrate, elm->modenvratenum); elm->modenvrate = NULL, elm->modenvratenum = 0; if (elm->modenvofsnum) free_ptr_list(elm->modenvofs, elm->modenvofsnum); elm->modenvofs = NULL, elm->modenvofsnum = 0; if (elm->envkeyfnum) free_ptr_list(elm->envkeyf, elm->envkeyfnum); elm->envkeyf = NULL, elm->envkeyfnum = 0; if (elm->envvelfnum) free_ptr_list(elm->envvelf, elm->envvelfnum); elm->envvelf = NULL, elm->envvelfnum = 0; if (elm->modenvkeyfnum) free_ptr_list(elm->modenvkeyf, elm->modenvkeyfnum); elm->modenvkeyf = NULL, elm->modenvkeyfnum = 0; if (elm->modenvvelfnum) free_ptr_list(elm->modenvvelf, elm->modenvvelfnum); elm->modenvvelf = NULL, elm->modenvvelfnum = 0; if (elm->trempitch) free(elm->trempitch); elm->trempitch = NULL, elm->trempitchnum = 0; if (elm->tremfc) free(elm->tremfc); elm->tremfc = NULL, elm->tremfcnum = 0; if (elm->modpitch) free(elm->modpitch); elm->modpitch = NULL, elm->modpitchnum = 0; if (elm->modfc) free(elm->modfc); elm->modfc = NULL, elm->modfcnum = 0; if (elm->fc) free(elm->fc); elm->fc = NULL, elm->fcnum = 0; if (elm->reso) free(elm->reso); elm->reso = NULL, elm->resonum = 0; } void free_instruments(int reload_default_inst) { int i = 128 + map_bank_counter, j; struct InstrumentCache *p; ToneBank *bank; Instrument *ip; struct InstrumentCache *default_entry; int default_entry_addr; clear_magic_instruments(); /* Free soundfont instruments */ while(i--) { /* Note that bank[*]->tone[j].instrument may pointer to bank[0]->tone[j].instrument. See play_midi_load_instrument() at playmidi.c for the implementation */ if((bank = tonebank[i]) != NULL) for(j = 127; j >= 0; j--) { ip = bank->tone[j].instrument; if(ip != NULL && ip->type == INST_SF2 && (i == 0 || ip != tonebank[0]->tone[j].instrument)) free_instrument(ip); bank->tone[j].instrument = NULL; } if((bank = drumset[i]) != NULL) for(j = 127; j >= 0; j--) { ip = bank->tone[j].instrument; if(ip != NULL && ip->type == INST_SF2 && (i == 0 || ip != drumset[0]->tone[j].instrument)) free_instrument(ip); bank->tone[j].instrument = NULL; } } /* Free GUS/patch instruments */ default_entry = NULL; default_entry_addr = 0; for(i = 0; i < INSTRUMENT_HASH_SIZE; i++) { p = instrument_cache[i]; while(p != NULL) { if(!reload_default_inst && p->ip == default_instrument) { default_entry = p; default_entry_addr = i; p = p->next; } else { struct InstrumentCache *tmp; tmp = p; p = p->next; free_instrument(tmp->ip); free(tmp); } } instrument_cache[i] = NULL; } if(reload_default_inst) set_default_instrument(NULL); else if(default_entry) { default_entry->next = NULL; instrument_cache[default_entry_addr] = default_entry; } } void free_special_patch(int id) { int i, j, start, end; if(id >= 0) start = end = id; else { start = 0; end = NSPECIAL_PATCH - 1; } for(i = start; i <= end; i++) if(special_patch[i] != NULL) { Sample *sp; int n; if(special_patch[i]->name != NULL) free(special_patch[i]->name); special_patch[i]->name = NULL; n = special_patch[i]->samples; sp = special_patch[i]->sample; if(sp) { for(j = 0; j < n; j++) if(sp[j].data_alloced && sp[j].data) free(sp[j].data); free(sp); } free(special_patch[i]); special_patch[i] = NULL; } } int set_default_instrument(char *name) { Instrument *ip; int i; static char *last_name; if(name == NULL) { name = last_name; if(name == NULL) return 0; } if(!(ip = load_gus_instrument(name, NULL, 0, 0, NULL))) return -1; if(default_instrument) free_instrument(default_instrument); default_instrument = ip; for(i = 0; i < MAX_CHANNELS; i++) default_program[i] = SPECIAL_PROGRAM; last_name = name; return 0; } /*! search mapped bank. returns negative value indicating free bank if not found, 0 if no free bank was available */ int find_instrument_map_bank(int dr, int map, int bk) { struct bank_map_elem *bm; int i; if (map == INST_NO_MAP) return 0; bm = dr ? map_drumset : map_bank; for(i = 0; i < MAP_BANK_COUNT; i++) { if (!bm[i].used) return -(128 + i); else if (bm[i].mapid == map && bm[i].bankno == bk) return 128 + i; } return 0; } /*! allocate mapped bank if needed. returns -1 if allocation failed. */ int alloc_instrument_map_bank(int dr, int map, int bk) { struct bank_map_elem *bm; int i; if (map == INST_NO_MAP) { alloc_instrument_bank(dr, bk); return bk; } i = find_instrument_map_bank(dr, map, bk); if (i == 0) return -1; if (i < 0) { i = -i - 128; bm = dr ? map_drumset : map_bank; bm[i].used = 1; bm[i].mapid = map; bm[i].bankno = bk; if (map_bank_counter < i + 1) map_bank_counter = i + 1; i += 128; alloc_instrument_bank(dr, i); } return i; } void alloc_instrument_bank(int dr, int bk) { ToneBank *b; if(dr) { if((b = drumset[bk]) == NULL) { b = drumset[bk] = (ToneBank *)safe_malloc(sizeof(ToneBank)); memset(b, 0, sizeof(ToneBank)); } } else { if((b = tonebank[bk]) == NULL) { b = tonebank[bk] = (ToneBank *)safe_malloc(sizeof(ToneBank)); memset(b, 0, sizeof(ToneBank)); } } } /* Instrument alias map - Written by Masanao Izumo */ int instrument_map(int mapID, int *set, int *elem) { int s, e; struct inst_map_elem *p; if(mapID == INST_NO_MAP) return 0; /* No map */ s = *set; e = *elem; p = inst_map_table[mapID][s]; if(p != NULL && p[e].mapped) { *set = p[e].set; *elem = p[e].elem; return 1; } if(s != 0) { p = inst_map_table[mapID][0]; if(p != NULL && p[e].mapped) { *set = p[e].set; *elem = p[e].elem; } return 2; } return 0; } void set_instrument_map(int mapID, int set_from, int elem_from, int set_to, int elem_to) { struct inst_map_elem *p; p = inst_map_table[mapID][set_from]; if(p == NULL) { p = (struct inst_map_elem *) safe_malloc(128 * sizeof(struct inst_map_elem)); memset(p, 0, 128 * sizeof(struct inst_map_elem)); inst_map_table[mapID][set_from] = p; } p[elem_from].set = set_to; p[elem_from].elem = elem_to; p[elem_from].mapped = 1; } void free_instrument_map(void) { int i, j; for(i = 0; i < map_bank_counter; i++) map_bank[i].used = map_drumset[i].used = 0; /* map_bank_counter = 0; never shrinks rather than assuming tonebank was already freed */ for (i = 0; i < NUM_INST_MAP; i++) { for (j = 0; j < 128; j++) { struct inst_map_elem *map; map = inst_map_table[i][j]; if (map) { free(map); inst_map_table[i][j] = NULL; } } } } /* Alternate assign - Written by Masanao Izumo */ AlternateAssign *add_altassign_string(AlternateAssign *old, char **params, int n) { int i, j; char *p; int beg, end; AlternateAssign *alt; if(n == 0) return old; if(!strcmp(*params, "clear")) { while(old) { AlternateAssign *next; next = old->next; free(old); old = next; } params++; n--; if(n == 0) return NULL; } alt = (AlternateAssign *)safe_malloc(sizeof(AlternateAssign)); memset(alt, 0, sizeof(AlternateAssign)); for(i = 0; i < n; i++) { p = params[i]; if(*p == '-') { beg = 0; p++; } else beg = atoi(p); if((p = strchr(p, '-')) != NULL) { if(p[1] == '\0') end = 127; else end = atoi(p + 1); } else end = beg; if(beg > end) { int t; t = beg; beg = end; end = t; } if(beg < 0) beg = 0; if(end > 127) end = 127; for(j = beg; j <= end; j++) alt->bits[(j >> 5) & 0x3] |= 1 << (j & 0x1F); } alt->next = old; return alt; } AlternateAssign *find_altassign(AlternateAssign *altassign, int note) { AlternateAssign *p; uint32 mask; int idx; mask = 1 << (note & 0x1F); idx = (note >> 5) & 0x3; for(p = altassign; p != NULL; p = p->next) if(p->bits[idx] & mask) return p; return NULL; }