mirror of
https://github.com/ZDoom/gzdoom.git
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2049 lines
54 KiB
C++
2049 lines
54 KiB
C++
/*
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TiMidity++ -- MIDI to WAVE converter and player
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Copyright (C) 1999-2004 Masanao Izumo <iz@onicos.co.jp>
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Copyright (C) 1995 Tuukka Toivonen <tt@cgs.fi>
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This program is free software; you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation; either version 2 of the License, or
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(at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program; if not, write to the Free Software
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Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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instrum.c
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Code to load and unload GUS-compatible instrument patches.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <math.h>
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#include <string.h>
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#include "timidity.h"
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#include "common.h"
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#include "instrum.h"
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#include "playmidi.h"
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#include "resample.h"
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#include "tables.h"
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#include "filter.h"
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#include "quantity.h"
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#include "freq.h"
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namespace TimidityPlus
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{
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Instruments::Instruments()
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{
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// init one-time global stuff - this should go to the device class once it exists.
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initialize_resampler_coeffs();
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init_tables();
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memset(&standard_tonebank, 0, sizeof(standard_tonebank));
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memset(&standard_drumset, 0, sizeof(standard_drumset));
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memcpy(layer_items, static_layer_items, sizeof(layer_items));
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}
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bool Instruments::load(SoundFontReaderInterface *sf)
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{
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sfreader = sf;
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if (read_config_file(nullptr, 0, 0) == RC_OK)
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{
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init_load_soundfont();
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set_default_instrument();
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return true;
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}
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return false;
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}
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Instruments::~Instruments()
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{
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free_instruments(0);
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free_soundfonts();
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free_tone_bank();
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free_instrument_map();
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if (sfreader != nullptr) delete sfreader;
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}
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void Instruments::free_instrument(Instrument *ip)
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{
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Sample *sp;
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int i;
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if (!ip) return;
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for (i = 0; i<ip->samples; i++)
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{
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sp = &(ip->sample[i]);
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if (sp->data_alloced)
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free(sp->data);
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}
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free(ip->sample);
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free(ip);
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}
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/* calculate ramp rate in fractional unit;
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* diff = 8bit, time = msec
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*/
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int32_t Instruments::calc_rate_i(int diff, double msec)
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{
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double rate;
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if(msec < 6)
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msec = 6;
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if(diff == 0)
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diff = 255;
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diff <<= (7+15);
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rate = ((double)diff / playback_rate) * control_ratio * 1000.0 / msec;
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if(fast_decay)
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rate *= 2;
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return (int32_t)rate;
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}
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/*End of Pseudo Reverb*/
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void Instruments::clear_magic_instruments(void)
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{
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int i, j;
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for (j = 0; j < 128 + map_bank_counter; j++)
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{
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if (tonebank[j])
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{
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ToneBank *bank = tonebank[j];
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for (i = 0; i < 128; i++)
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if (IS_MAGIC_INSTRUMENT(bank->tone[i].instrument))
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bank->tone[i].instrument = NULL;
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}
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if (drumset[j])
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{
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ToneBank *bank = drumset[j];
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for (i = 0; i < 128; i++)
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if (IS_MAGIC_INSTRUMENT(bank->tone[i].instrument))
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bank->tone[i].instrument = NULL;
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}
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}
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}
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int32_t Instruments::convert_envelope_rate(uint8_t rate)
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{
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const int32_t GUS_ENVRATE_MAX = (int32_t)(0x3FFFFFFF >> 9);
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int32_t r;
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r = 3 - ((rate >> 6) & 0x3);
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r *= 3;
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r = (int32_t)(rate & 0x3f) << r; /* 6.9 fixed point */
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/* 15.15 fixed point. */
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r = r * 44100 / playback_rate * control_ratio * (1 << fast_decay);
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if (r > GUS_ENVRATE_MAX) { r = GUS_ENVRATE_MAX; }
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return (r << 9);
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}
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int32_t Instruments::convert_envelope_offset(uint8_t offset)
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{
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/* This is not too good... Can anyone tell me what these values mean?
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Are they GUS-style "exponential" volumes? And what does that mean? */
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/* 15.15 fixed point */
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return offset << (7 + 15);
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}
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int32_t Instruments::convert_tremolo_sweep(uint8_t sweep)
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{
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if (!sweep)
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return 0;
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return
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((control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) /
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(playback_rate * sweep);
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}
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int32_t Instruments::convert_vibrato_sweep(uint8_t sweep, int32_t vib_control_ratio)
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{
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if (!sweep)
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return 0;
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return (int32_t)(TIM_FSCALE((double) (vib_control_ratio)
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* SWEEP_TUNING, SWEEP_SHIFT)
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/ (double)(playback_rate * sweep));
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/* this was overflowing with seashore.pat
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((vib_control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) /
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(playback_rate * sweep); */
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}
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int32_t Instruments::convert_tremolo_rate(uint8_t rate)
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{
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return
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((SINE_CYCLE_LENGTH * control_ratio * rate) << RATE_SHIFT) /
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(TREMOLO_RATE_TUNING * playback_rate);
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}
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int32_t Instruments::convert_vibrato_rate(uint8_t rate)
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{
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/* Return a suitable vibrato_control_ratio value */
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return
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(VIBRATO_RATE_TUNING * playback_rate) /
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(rate * 2 * VIBRATO_SAMPLE_INCREMENTS);
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}
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void Instruments::reverse_data(int16_t *sp, int32_t ls, int32_t le)
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{
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int16_t s, *ep = sp + le;
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int32_t i;
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sp += ls;
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le -= ls;
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le /= 2;
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for (i = 0; i < le; i++)
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{
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s = *sp;
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*sp++ = *ep;
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*ep-- = s;
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}
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}
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int Instruments::name_hash(char *name)
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{
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unsigned int addr = 0;
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while(*name)
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addr += *name++;
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return addr % INSTRUMENT_HASH_SIZE;
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}
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Instrument *Instruments::search_instrument_cache(char *name, int panning, int amp, int note_to_use, int strip_loop, int strip_envelope, int strip_tail)
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{
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struct InstrumentCache *p;
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for (p = instrument_cache[name_hash(name)]; p != NULL; p = p->next)
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{
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if (strcmp(p->name, name) != 0)
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return NULL;
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if (p->panning == panning &&
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p->amp == amp &&
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p->note_to_use == note_to_use &&
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p->strip_loop == strip_loop &&
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p->strip_envelope == strip_envelope &&
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p->strip_tail == strip_tail)
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return p->ip;
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}
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return NULL;
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}
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void Instruments::store_instrument_cache(Instrument *ip, char *name, int panning, int amp, int note_to_use, int strip_loop, int strip_envelope, int strip_tail)
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{
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struct InstrumentCache *p;
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int addr;
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addr = name_hash(name);
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p = (struct InstrumentCache *)safe_malloc(sizeof(struct InstrumentCache));
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p->next = instrument_cache[addr];
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instrument_cache[addr] = p;
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p->name = name;
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p->panning = panning;
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p->amp = amp;
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p->note_to_use = note_to_use;
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p->strip_loop = strip_loop;
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p->strip_envelope = strip_envelope;
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p->strip_tail = strip_tail;
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p->ip = ip;
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}
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static int32_t adjust_tune_freq(int32_t val, float tune)
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{
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if (! tune)
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return val;
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return val / pow(2.0, tune / 12.0);
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}
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static int16_t adjust_scale_tune(int16_t val)
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{
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return 1024 * (double) val / 100 + 0.5;
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}
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static int16_t adjust_fc(int16_t val)
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{
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if (val < 0 || val > playback_rate / 2) {
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return 0;
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} else {
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return val;
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}
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}
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static int16_t adjust_reso(int16_t val)
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{
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if (val < 0 || val > 960) {
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return 0;
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} else {
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return val;
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}
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}
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int32_t Instruments::to_rate(int rate)
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{
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return (rate) ? (int32_t) (0x200 * pow(2.0, rate / 17.0)
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* 44100 / playback_rate * control_ratio) << fast_decay : 0;
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}
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void Instruments::apply_bank_parameter(Instrument *ip, ToneBankElement *tone)
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{
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int i, j;
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Sample *sp;
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if (tone->tunenum)
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for (i = 0; i < ip->samples; i++) {
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sp = &ip->sample[i];
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if (tone->tunenum == 1) {
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sp->low_freq = adjust_tune_freq(sp->low_freq, tone->tune[0]);
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sp->high_freq = adjust_tune_freq(sp->high_freq, tone->tune[0]);
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sp->root_freq = adjust_tune_freq(sp->root_freq, tone->tune[0]);
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} else if (i < tone->tunenum) {
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sp->low_freq = adjust_tune_freq(sp->low_freq, tone->tune[i]);
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sp->high_freq = adjust_tune_freq(sp->high_freq, tone->tune[i]);
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sp->root_freq = adjust_tune_freq(sp->root_freq, tone->tune[i]);
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}
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}
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if (tone->envratenum)
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for (i = 0; i < ip->samples; i++) {
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sp = &ip->sample[i];
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if (tone->envratenum == 1) {
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for (j = 0; j < 6; j++)
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if (tone->envrate[0][j] >= 0)
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sp->envelope_rate[j] = to_rate(tone->envrate[0][j]);
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} else if (i < tone->envratenum) {
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for (j = 0; j < 6; j++)
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if (tone->envrate[i][j] >= 0)
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sp->envelope_rate[j] = to_rate(tone->envrate[i][j]);
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}
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}
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if (tone->envofsnum)
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for (i = 0; i < ip->samples; i++) {
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sp = &ip->sample[i];
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if (tone->envofsnum == 1) {
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for (j = 0; j < 6; j++)
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if (tone->envofs[0][j] >= 0)
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sp->envelope_offset[j] = to_offset_22(tone->envofs[0][j]);
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} else if (i < tone->envofsnum) {
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for (j = 0; j < 6; j++)
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if (tone->envofs[i][j] >= 0)
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sp->envelope_offset[j] = to_offset_22(tone->envofs[i][j]);
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}
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}
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if (tone->tremnum)
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for (i = 0; i < ip->samples; i++) {
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sp = &ip->sample[i];
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if (tone->tremnum == 1) {
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if (IS_QUANTITY_DEFINED(tone->trem[0][0]))
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sp->tremolo_sweep_increment =
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quantity_to_int(&tone->trem[0][0], 0);
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if (IS_QUANTITY_DEFINED(tone->trem[0][1]))
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sp->tremolo_phase_increment =
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quantity_to_int(&tone->trem[0][1], 0);
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if (IS_QUANTITY_DEFINED(tone->trem[0][2]))
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sp->tremolo_depth =
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quantity_to_int(&tone->trem[0][2], 0) << 1;
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} else if (i < tone->tremnum) {
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if (IS_QUANTITY_DEFINED(tone->trem[i][0]))
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sp->tremolo_sweep_increment =
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quantity_to_int(&tone->trem[i][0], 0);
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if (IS_QUANTITY_DEFINED(tone->trem[i][1]))
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sp->tremolo_phase_increment =
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quantity_to_int(&tone->trem[i][1], 0);
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if (IS_QUANTITY_DEFINED(tone->trem[i][2]))
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sp->tremolo_depth =
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quantity_to_int(&tone->trem[i][2], 0) << 1;
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}
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}
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if (tone->vibnum)
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for (i = 0; i < ip->samples; i++) {
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sp = &ip->sample[i];
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if (tone->vibnum == 1) {
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if (IS_QUANTITY_DEFINED(tone->vib[0][1]))
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sp->vibrato_control_ratio =
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quantity_to_int(&tone->vib[0][1], 0);
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if (IS_QUANTITY_DEFINED(tone->vib[0][0]))
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sp->vibrato_sweep_increment =
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quantity_to_int(&tone->vib[0][0],
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sp->vibrato_control_ratio);
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if (IS_QUANTITY_DEFINED(tone->vib[0][2]))
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sp->vibrato_depth = quantity_to_int(&tone->vib[0][2], 0);
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} else if (i < tone->vibnum) {
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if (IS_QUANTITY_DEFINED(tone->vib[i][1]))
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sp->vibrato_control_ratio =
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quantity_to_int(&tone->vib[i][1], 0);
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if (IS_QUANTITY_DEFINED(tone->vib[i][0]))
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sp->vibrato_sweep_increment =
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quantity_to_int(&tone->vib[i][0],
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sp->vibrato_control_ratio);
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if (IS_QUANTITY_DEFINED(tone->vib[i][2]))
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sp->vibrato_depth = quantity_to_int(&tone->vib[i][2], 0);
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}
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}
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if (tone->sclnotenum)
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for (i = 0; i < ip->samples; i++) {
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sp = &ip->sample[i];
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if (tone->sclnotenum == 1)
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sp->scale_freq = tone->sclnote[0];
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else if (i < tone->sclnotenum)
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sp->scale_freq = tone->sclnote[i];
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}
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if (tone->scltunenum)
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for (i = 0; i < ip->samples; i++) {
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sp = &ip->sample[i];
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if (tone->scltunenum == 1)
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sp->scale_factor = adjust_scale_tune(tone->scltune[0]);
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else if (i < tone->scltunenum)
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sp->scale_factor = adjust_scale_tune(tone->scltune[i]);
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}
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if (tone->modenvratenum)
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for (i = 0; i < ip->samples; i++) {
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sp = &ip->sample[i];
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if (tone->modenvratenum == 1) {
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for (j = 0; j < 6; j++)
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if (tone->modenvrate[0][j] >= 0)
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sp->modenv_rate[j] = to_rate(tone->modenvrate[0][j]);
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} else if (i < tone->modenvratenum) {
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for (j = 0; j < 6; j++)
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if (tone->modenvrate[i][j] >= 0)
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sp->modenv_rate[j] = to_rate(tone->modenvrate[i][j]);
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}
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}
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if (tone->modenvofsnum)
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for (i = 0; i < ip->samples; i++) {
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sp = &ip->sample[i];
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if (tone->modenvofsnum == 1) {
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for (j = 0; j < 6; j++)
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if (tone->modenvofs[0][j] >= 0)
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sp->modenv_offset[j] =
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to_offset_22(tone->modenvofs[0][j]);
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} else if (i < tone->modenvofsnum) {
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for (j = 0; j < 6; j++)
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if (tone->modenvofs[i][j] >= 0)
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sp->modenv_offset[j] =
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to_offset_22(tone->modenvofs[i][j]);
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}
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}
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if (tone->envkeyfnum)
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for (i = 0; i < ip->samples; i++) {
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sp = &ip->sample[i];
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if (tone->envkeyfnum == 1) {
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for (j = 0; j < 6; j++)
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if (tone->envkeyf[0][j] != -1)
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sp->envelope_keyf[j] = tone->envkeyf[0][j];
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} else if (i < tone->envkeyfnum) {
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for (j = 0; j < 6; j++)
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if (tone->envkeyf[i][j] != -1)
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sp->envelope_keyf[j] = tone->envkeyf[i][j];
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}
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}
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if (tone->envvelfnum)
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for (i = 0; i < ip->samples; i++) {
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sp = &ip->sample[i];
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if (tone->envvelfnum == 1) {
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for (j = 0; j < 6; j++)
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if (tone->envvelf[0][j] != -1)
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sp->envelope_velf[j] = tone->envvelf[0][j];
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} else if (i < tone->envvelfnum) {
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for (j = 0; j < 6; j++)
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if (tone->envvelf[i][j] != -1)
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sp->envelope_velf[j] = tone->envvelf[i][j];
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}
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}
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if (tone->modenvkeyfnum)
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for (i = 0; i < ip->samples; i++) {
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sp = &ip->sample[i];
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if (tone->modenvkeyfnum == 1) {
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for (j = 0; j < 6; j++)
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if (tone->modenvkeyf[0][j] != -1)
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sp->modenv_keyf[j] = tone->modenvkeyf[0][j];
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} else if (i < tone->modenvkeyfnum) {
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for (j = 0; j < 6; j++)
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if (tone->modenvkeyf[i][j] != -1)
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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_t tmpchar; \
|
|
\
|
|
if (tf_read(&tmpchar, 1, 1, tf) != 1) \
|
|
goto fail; \
|
|
thing = tmpchar; \
|
|
}
|
|
#define READ_SHORT(thing) { \
|
|
uint16_t tmpshort; \
|
|
\
|
|
if (tf_read(&tmpshort, 2, 1, tf) != 1) \
|
|
goto fail; \
|
|
thing = LE_SHORT(tmpshort); \
|
|
}
|
|
#define READ_LONG(thing) { \
|
|
int32_t 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
|
|
*/
|
|
Instrument *Instruments::load_gus_instrument(char *name, ToneBank *bank, int dr, int prog)
|
|
{
|
|
ToneBankElement *tone;
|
|
int amp, note_to_use, panning, strip_envelope, strip_loop, strip_tail;
|
|
Instrument *ip;
|
|
struct timidity_file *tf;
|
|
uint8_t tmp[1024], fractions;
|
|
Sample *sp;
|
|
int i, j, noluck = 0;
|
|
|
|
if (!name)
|
|
return 0;
|
|
|
|
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 */
|
|
tf = open_file(name, sfreader);
|
|
if (!tf)
|
|
{
|
|
int name_len, ext_len;
|
|
static const char *patch_ext[] = { ".pat", 0 };
|
|
|
|
noluck = 1;
|
|
name_len = (int)strlen(name);
|
|
/* Try with various extensions */
|
|
for (i = 0; patch_ext[i]; i++)
|
|
{
|
|
ext_len = (int)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]);
|
|
tf = open_file((char *)tmp, sfreader);
|
|
if (tf)
|
|
{
|
|
noluck = 0;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
if (noluck)
|
|
{
|
|
ctl_cmsg(CMSG_INFO, VERB_DEBUG, "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);
|
|
tf_close(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]);
|
|
tf_close(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]);
|
|
tf_close(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);
|
|
tf_close(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_t)(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_t)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_22(5);
|
|
sp->envelope_rate[3] = calc_rate_i(255, modify_release);
|
|
sp->envelope_offset[4] = to_offset_22(4);
|
|
sp->envelope_rate[4] = to_offset_22(200);
|
|
sp->envelope_offset[5] = to_offset_22(4);
|
|
sp->envelope_rate[5] = to_offset_22(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)) != (int)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 */
|
|
uint16_t *tmp;
|
|
uint8_t *cp = (uint8_t *)sp->data;
|
|
|
|
tmp = (uint16_t *)safe_malloc(sp->data_length * 2 + 4);
|
|
for (splen_t i = 0; i < sp->data_length; i++)
|
|
tmp[i] = (uint16_t)cp[i] << 8;
|
|
sp->data = (sample_t *)tmp;
|
|
free(cp);
|
|
sp->data_length *= 2;
|
|
sp->loop_start *= 2;
|
|
sp->loop_end *= 2;
|
|
}
|
|
#ifdef _BIG_ENDIAN_
|
|
else { /* convert to machine byte order */
|
|
int32_t i;
|
|
int16_t *tmp = (int16_t *)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_t i = sp->data_length / 2;
|
|
int16_t *tmp = (int16_t *)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 t;
|
|
|
|
reverse_data((int16_t *)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_t *)sp->data, sp->data_length / 2,
|
|
sp->sample_rate, playback_rate);
|
|
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_t a, maxamp = 0;
|
|
int16_t *tmp = (int16_t *)sp->data;
|
|
|
|
for (splen_t 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);
|
|
}
|
|
/* 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 && timidity_surround_chorus)
|
|
{
|
|
Freq freq;
|
|
sp->chord = -1;
|
|
sp->root_freq_detected = freq.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);
|
|
}
|
|
|
|
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");
|
|
}
|
|
}
|
|
tf_close(tf);
|
|
store_instrument_cache(ip, name, panning, amp, note_to_use,
|
|
strip_loop, strip_envelope, strip_tail);
|
|
return ip;
|
|
}
|
|
|
|
|
|
Instrument *Instruments::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;
|
|
double volume_max;
|
|
int pan, panning;
|
|
|
|
#if 0
|
|
// This cannot possibly work as implemented.
|
|
if (play_system_mode == GS_SYSTEM_MODE && (b == 64 || b == 65)) {
|
|
if (!dr) /* User Instrument */
|
|
recompute_userinst(b, prog);
|
|
else { /* User Drumset */
|
|
ip = recompute_userdrum(b, prog);
|
|
if (ip != NULL) {
|
|
return ip;
|
|
}
|
|
}
|
|
}
|
|
#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].name == NULL) /* this should not be NULL to play the instrument */
|
|
bank->tone[prog].name = safe_strdup(DYNAMIC_INSTRUMENT_NAME);
|
|
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 */
|
|
ip = load_gus_instrument(bank->tone[prog].name, bank, dr, prog);
|
|
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;
|
|
}
|
|
|
|
int Instruments::fill_bank(int dr, int b, int *rc)
|
|
{
|
|
int i, errors = 0;
|
|
ToneBank *bank = ((dr) ? drumset[b] : tonebank[b]);
|
|
|
|
if (rc != NULL)
|
|
*rc = RC_OK;
|
|
|
|
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)
|
|
{
|
|
// This would be too annoying on 'warning' level.
|
|
ctl_cmsg(CMSG_WARNING, VERB_DEBUG,
|
|
"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 = RC_OK;
|
|
}
|
|
|
|
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 Instruments::load_missing_instruments(int *rc)
|
|
{
|
|
int i = 128 + map_bank_counter, errors = 0;
|
|
if (rc != NULL)
|
|
*rc = RC_OK;
|
|
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;
|
|
}
|
|
|
|
// The precaching code is from ZDoom's Timidity-based GUS emulation
|
|
void Instruments::MarkInstrument(int banknum, int percussion, int instr)
|
|
{
|
|
ToneBank *bank;
|
|
|
|
if (banknum >= 128)
|
|
{
|
|
return;
|
|
}
|
|
if (banknum != 0)
|
|
{
|
|
/* Mark the standard bank in case it's not defined by this one. */
|
|
MarkInstrument(0, percussion, instr);
|
|
}
|
|
if (percussion)
|
|
{
|
|
bank = drumset[banknum];
|
|
}
|
|
else
|
|
{
|
|
bank = tonebank[banknum];
|
|
}
|
|
if (bank == NULL)
|
|
{
|
|
return;
|
|
}
|
|
if (bank->tone[instr].instrument == NULL)
|
|
{
|
|
bank->tone[instr].instrument = MAGIC_LOAD_INSTRUMENT;
|
|
}
|
|
}
|
|
|
|
void Instruments::PrecacheInstruments(const uint16_t *instruments, int count)
|
|
{
|
|
for (int i = 0; i < count; ++i)
|
|
{
|
|
MarkInstrument((instruments[i] >> 7) & 127, instruments[i] >> 14, instruments[i] & 127);
|
|
}
|
|
load_missing_instruments(nullptr);
|
|
}
|
|
|
|
|
|
|
|
void *Instruments::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 Instruments::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_t *)safe_memdup(elm->sclnote,
|
|
elm->sclnotenum * sizeof(int16_t));
|
|
if (elm->scltunenum)
|
|
elm->scltune = (int16_t *)safe_memdup(elm->scltune,
|
|
elm->scltunenum * sizeof(int16_t));
|
|
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_t *)safe_memdup(elm->trempitch,
|
|
elm->trempitchnum * sizeof(int16_t));
|
|
if (elm->tremfcnum)
|
|
elm->tremfc = (int16_t *)safe_memdup(elm->tremfc,
|
|
elm->tremfcnum * sizeof(int16_t));
|
|
if (elm->modpitchnum)
|
|
elm->modpitch = (int16_t *)safe_memdup(elm->modpitch,
|
|
elm->modpitchnum * sizeof(int16_t));
|
|
if (elm->modfcnum)
|
|
elm->modfc = (int16_t *)safe_memdup(elm->modfc,
|
|
elm->modfcnum * sizeof(int16_t));
|
|
if (elm->fcnum)
|
|
elm->fc = (int16_t *)safe_memdup(elm->fc,
|
|
elm->fcnum * sizeof(int16_t));
|
|
if (elm->resonum)
|
|
elm->reso = (int16_t *)safe_memdup(elm->reso,
|
|
elm->resonum * sizeof(int16_t));
|
|
|
|
}
|
|
|
|
/*! Release ToneBank[128 + MAP_BANK_COUNT] */
|
|
void Instruments::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 Instruments::free_tone_bank(void)
|
|
{
|
|
free_tone_bank_list(tonebank);
|
|
free_tone_bank_list(drumset);
|
|
}
|
|
|
|
/*! Release ToneBankElement. */
|
|
void Instruments::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 Instruments::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->tone[j].name && !bank->tone[j].name[0]) /* DYNAMIC_INSTRUMENT_NAME */
|
|
{
|
|
free(bank->tone[j].name);
|
|
bank->tone[j].name = 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;
|
|
if (bank->tone[j].name && !bank->tone[j].name[0]) /* DYNAMIC_INSTRUMENT_NAME */
|
|
{
|
|
free(bank->tone[j].name);
|
|
bank->tone[j].name = NULL;
|
|
}
|
|
}
|
|
#if 0
|
|
if ((drumset[i] != NULL) && (drumset[i]->alt != NULL)) {
|
|
free(drumset[i]->alt);
|
|
drumset[i]->alt = NULL;
|
|
}
|
|
#endif
|
|
}
|
|
|
|
/* 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 Instruments::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 Instruments::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)))
|
|
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 Instruments::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 Instruments::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 Instruments::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 Instruments::instrument_map(int mapID, int *set, int *elem) const
|
|
{
|
|
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 Instruments::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 Instruments::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 *Instruments::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 *Instruments::find_altassign(AlternateAssign *altassign, int note)
|
|
{
|
|
AlternateAssign *p;
|
|
uint32_t 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;
|
|
}
|
|
|
|
Instrument *Instruments::play_midi_load_instrument(int dr, int bk, int prog, bool *pLoad_success)
|
|
{
|
|
ToneBank **bank = (dr) ? drumset : tonebank;
|
|
ToneBankElement *tone;
|
|
Instrument *ip;
|
|
bool load_success = false;
|
|
|
|
if (bank[bk] == NULL)
|
|
alloc_instrument_bank(dr, bk);
|
|
|
|
tone = &bank[bk]->tone[prog];
|
|
/* tone->name is NULL if "soundfont" directive is used, and ip is NULL when not preloaded */
|
|
/* dr: not sure but only drumsets are concerned at the moment */
|
|
if (dr && !tone->name && ((ip = tone->instrument) == MAGIC_LOAD_INSTRUMENT || ip == NULL)
|
|
&& (ip = load_instrument(dr, bk, prog)) != NULL) {
|
|
tone->instrument = ip;
|
|
tone->name = safe_strdup(DYNAMIC_INSTRUMENT_NAME);
|
|
load_success = 1;
|
|
}
|
|
else if (tone->name) {
|
|
/* Instrument is found. */
|
|
if ((ip = tone->instrument) == MAGIC_LOAD_INSTRUMENT
|
|
#ifndef SUPPRESS_CHANNEL_LAYER
|
|
|| ip == NULL /* see also readmidi.c: groom_list(). */
|
|
#endif
|
|
) {
|
|
ip = tone->instrument = load_instrument(dr, bk, prog);
|
|
}
|
|
if (ip == NULL || IS_MAGIC_INSTRUMENT(ip)) {
|
|
tone->instrument = MAGIC_ERROR_INSTRUMENT;
|
|
}
|
|
else {
|
|
load_success = true;
|
|
}
|
|
}
|
|
else {
|
|
/* Instrument is not found.
|
|
Try to load the instrument from bank 0 */
|
|
ToneBankElement *tone0 = &bank[0]->tone[prog];
|
|
if ((ip = tone0->instrument) == NULL
|
|
|| ip == MAGIC_LOAD_INSTRUMENT)
|
|
ip = tone0->instrument = load_instrument(dr, 0, prog);
|
|
if (ip == NULL || IS_MAGIC_INSTRUMENT(ip)) {
|
|
tone0->instrument = MAGIC_ERROR_INSTRUMENT;
|
|
}
|
|
else {
|
|
copy_tone_bank_element(tone, tone0);
|
|
tone->instrument = ip;
|
|
load_success = 1;
|
|
}
|
|
}
|
|
|
|
*pLoad_success = load_success;
|
|
|
|
if (ip == MAGIC_ERROR_INSTRUMENT)
|
|
return NULL;
|
|
|
|
return ip;
|
|
}
|
|
|
|
|
|
|
|
//void recompute_userinst_altassign(int bank, int group);
|
|
|
|
|
|
/*! initialize GS user drumset. */
|
|
void Instruments::init_userdrum()
|
|
{
|
|
int i;
|
|
|
|
free_userdrum();
|
|
|
|
for (i = 0; i<2; i++) { /* allocate alternative assign */
|
|
memset(&alt[i], 0, sizeof(AlternateAssign));
|
|
alloc_instrument_bank(1, 64 + i);
|
|
drumset[64 + i]->alt = &alt[i];
|
|
}
|
|
}
|
|
|
|
|
|
/*! free GS user drumset. */
|
|
void Instruments::free_userdrum()
|
|
{
|
|
UserDrumset *p, *next;
|
|
|
|
for (p = userdrum_first; p != NULL; p = next) {
|
|
next = p->next;
|
|
free(p);
|
|
}
|
|
userdrum_first = userdrum_last = NULL;
|
|
}
|
|
|
|
/*! free GS user instrument. */
|
|
void Instruments::free_userinst()
|
|
{
|
|
UserInstrument *p, *next;
|
|
|
|
for (p = userinst_first; p != NULL; p = next) {
|
|
next = p->next;
|
|
free(p);
|
|
}
|
|
userinst_first = userinst_last = NULL;
|
|
}
|
|
|
|
|
|
|
|
|
|
/*! recompute GS user instrument. */
|
|
/*! get pointer to requested GS user instrument.
|
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if it's not found, allocate a new item first. */
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Instruments::UserInstrument *Instruments::get_userinst(int bank, int prog)
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{
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UserInstrument *p;
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for (p = userinst_first; p != NULL; p = p->next) {
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if (p->bank == bank && p->prog == prog) { return p; }
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}
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p = (UserInstrument *)safe_malloc(sizeof(UserInstrument));
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memset(p, 0, sizeof(UserInstrument));
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p->next = NULL;
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if (userinst_first == NULL) {
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userinst_first = p;
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userinst_last = p;
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}
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else {
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userinst_last->next = p;
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userinst_last = p;
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}
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p->bank = bank;
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p->prog = prog;
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return p;
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}
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void Instruments::recompute_userinst(int bank, int prog)
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{
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auto p = get_userinst(bank, prog);
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auto source_bank = p->source_bank;
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auto source_prog = p->source_prog;
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free_tone_bank_element(&tonebank[bank]->tone[prog]);
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if (tonebank[source_bank]) {
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if (tonebank[source_bank]->tone[source_prog].name)
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{
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copy_tone_bank_element(&tonebank[bank]->tone[prog], &tonebank[source_bank]->tone[source_prog]);
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}
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else if (tonebank[0]->tone[source_prog].name)
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{
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copy_tone_bank_element(&tonebank[bank]->tone[prog], &tonebank[0]->tone[source_prog]);
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}
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}
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}
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/*! get pointer to requested GS user drumset.
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if it's not found, allocate a new item first. */
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Instruments::UserDrumset *Instruments::get_userdrum(int bank, int prog)
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{
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UserDrumset *p;
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for (p = userdrum_first; p != NULL; p = p->next) {
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if (p->bank == bank && p->prog == prog) { return p; }
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}
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p = (UserDrumset *)safe_malloc(sizeof(UserDrumset));
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memset(p, 0, sizeof(UserDrumset));
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p->next = NULL;
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if (userdrum_first == NULL) {
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userdrum_first = p;
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userdrum_last = p;
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}
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else {
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userdrum_last->next = p;
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userdrum_last = p;
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}
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p->bank = bank;
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p->prog = prog;
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return p;
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}
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/*! recompute GS user drumset. */
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Instrument *Instruments::recompute_userdrum(int bank, int prog)
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{
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Instrument *ip = NULL;
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auto p = get_userdrum(bank, prog);
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auto source_note = p->source_note;
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auto source_prog = p->source_prog;
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free_tone_bank_element(&drumset[bank]->tone[prog]);
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if (drumset[source_prog]) {
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ToneBankElement *source_tone = &drumset[source_prog]->tone[source_note];
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if (source_tone->name == NULL /* NULL if "soundfont" directive is used */
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&& source_tone->instrument == NULL) {
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if ((ip = load_instrument(1, source_prog, source_note)) == NULL) {
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ip = MAGIC_ERROR_INSTRUMENT;
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}
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source_tone->instrument = ip;
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}
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if (source_tone->name)
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{
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copy_tone_bank_element(&drumset[bank]->tone[prog], source_tone);
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}
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else if (drumset[0]->tone[source_note].name)
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{
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copy_tone_bank_element(&drumset[bank]->tone[prog], &drumset[0]->tone[source_note]);
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}
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else {
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ctl_cmsg(CMSG_WARNING, VERB_NORMAL, "Referring user drum set %d, note %d not found - this instrument will not be heard as expected", bank, prog);
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}
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}
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return ip;
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}
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/*! convert GS user drumset assign groups to internal "alternate assign". */
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void Instruments::recompute_userdrum_altassign(int bank, int group)
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{
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int number = 0, i;
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char *params[131], param[10];
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ToneBank *bk;
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UserDrumset *p;
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for (p = userdrum_first; p != NULL; p = p->next) {
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if (p->assign_group == group) {
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sprintf(param, "%d", p->prog);
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params[number] = safe_strdup(param);
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number++;
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}
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}
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params[number] = NULL;
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alloc_instrument_bank(1, bank);
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bk = drumset[bank];
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bk->alt = add_altassign_string(bk->alt, params, number);
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for (i = number - 1; i >= 0; i--)
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free(params[i]);
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}
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}
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