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d4563767ee
GUS patch flags, envelopes, and volume levels, while trying to be closer to TiMidity++ than original TiMidity. - Renamed timidity_config and timidity_voices to midi_config and midi_voices respectively. SVN r959 (trunk)
747 lines
20 KiB
C++
747 lines
20 KiB
C++
/*
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TiMidity -- Experimental MIDI to WAVE converter
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Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
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This library is free software; you can redistribute it and/or
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modify it under the terms of the GNU Lesser General Public
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License as published by the Free Software Foundation; either
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version 2.1 of the License, or (at your option) any later version.
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This library 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 GNU
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Lesser General Public License for more details.
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You should have received a copy of the GNU Lesser General Public
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License along with this library; 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 <string.h>
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#include <stdlib.h>
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#include <math.h>
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#include "timidity.h"
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#include "m_swap.h"
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#include "files.h"
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#include "templates.h"
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#include "gf1patch.h"
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namespace Timidity
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{
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extern Instrument *load_instrument_dls(Renderer *song, int drum, int bank, int instrument);
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Instrument::Instrument()
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: samples(0), sample(NULL)
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{
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}
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Instrument::~Instrument()
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{
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Sample *sp;
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int i;
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for (i = samples, sp = &(sample[0]); i != 0; i--, sp++)
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{
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if (sp->type == INST_GUS && sp->data != NULL)
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{
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free(sp->data);
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}
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}
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free(sample);
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}
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ToneBank::ToneBank()
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{
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tone = new ToneBankElement[128];;
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for (int i = 0; i < MAXPROG; ++i)
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{
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instrument[i] = 0;
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}
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}
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ToneBank::~ToneBank()
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{
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delete[] tone;
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for (int i = 0; i < MAXPROG; i++)
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{
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if (instrument[i] != NULL && instrument[i] != MAGIC_LOAD_INSTRUMENT)
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{
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delete instrument[i];
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instrument[i] = NULL;
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}
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}
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}
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int convert_tremolo_sweep(Renderer *song, BYTE sweep)
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{
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if (sweep == 0)
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return 0;
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return
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int(((song->control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) / (song->rate * sweep));
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}
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int convert_vibrato_sweep(Renderer *song, BYTE sweep, int vib_control_ratio)
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{
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if (sweep == 0)
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return 0;
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return
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(int) (FSCALE((double) (vib_control_ratio) * SWEEP_TUNING, SWEEP_SHIFT) / (song->rate * sweep));
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/* this was overflowing with seashore.pat
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((vib_control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) / (song->rate * sweep);
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*/
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}
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int convert_tremolo_rate(Renderer *song, BYTE rate)
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{
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return
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int(((song->control_ratio * rate) << RATE_SHIFT) / (TREMOLO_RATE_TUNING * song->rate));
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}
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int convert_vibrato_rate(Renderer *song, BYTE 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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int((VIBRATO_RATE_TUNING * song->rate) / (rate * 2 * VIBRATO_SAMPLE_INCREMENTS));
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}
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static void reverse_data(sample_t *sp, int ls, int le)
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{
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sample_t s, *ep = sp + le;
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sp += ls;
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le -= ls;
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le /= 2;
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while (le--)
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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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/*
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If panning or note_to_use != -1, it will be used for all samples,
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instead of the sample-specific values in the instrument file.
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For note_to_use, any value <0 or >127 will be forced to 0.
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For other parameters, 1 means yes, 0 means no, other values are
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undefined.
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TODO: do reverse loops right */
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static Instrument *load_instrument(Renderer *song, const char *name, int percussion,
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int panning, int amp, int note_to_use,
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int strip_loop, int strip_envelope,
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int strip_tail)
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{
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Instrument *ip;
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Sample *sp;
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FileReader *fp;
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GF1PatchHeader header;
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GF1InstrumentData idata;
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GF1LayerData layer_data;
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GF1PatchData patch_data;
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int i, j;
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bool noluck = false;
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if (!name) return 0;
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/* Open patch file */
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if ((fp = open_filereader(name, openmode, NULL)) == NULL)
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{
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/* Try with various extensions */
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FString tmp = name;
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tmp += ".pat";
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if ((fp = open_filereader(tmp, openmode, NULL)) == NULL)
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{
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#ifdef unix // Windows isn't case-sensitive.
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tmp.ToUpper();
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if ((fp = open_filereader(tmp, openmode, NULL)) == NULL)
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#endif
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{
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noluck = true;
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}
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}
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}
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if (noluck)
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{
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cmsg(CMSG_ERROR, VERB_NORMAL, "Instrument `%s' can't be found.\n", name);
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return 0;
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}
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cmsg(CMSG_INFO, VERB_NOISY, "Loading instrument %s\n", name);
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/* Read some headers and do cursory sanity checks. */
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if (sizeof(header) != fp->Read(&header, sizeof(header)))
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{
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failread:
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cmsg(CMSG_ERROR, VERB_NORMAL, "%s: Error reading instrument.\n", name);
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delete fp;
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return 0;
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}
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if (strncmp(header.Header, GF1_HEADER_TEXT, HEADER_SIZE - 4) != 0)
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{
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cmsg(CMSG_ERROR, VERB_NORMAL, "%s: Not an instrument.\n", name);
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delete fp;
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return 0;
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}
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if (strcmp(header.Header + 8, "110") < 0)
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{
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cmsg(CMSG_ERROR, VERB_NORMAL, "%s: Is an old and unsupported patch version.\n", name);
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delete fp;
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return 0;
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}
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if (sizeof(idata) != fp->Read(&idata, sizeof(idata)))
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{
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goto failread;
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}
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header.WaveForms = LittleShort(header.WaveForms);
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header.MasterVolume = LittleShort(header.MasterVolume);
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header.DataSize = LittleLong(header.DataSize);
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idata.Instrument = LittleShort(idata.Instrument);
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if (header.Instruments != 1 && header.Instruments != 0) /* instruments. To some patch makers, 0 means 1 */
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{
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cmsg(CMSG_ERROR, VERB_NORMAL, "Can't handle patches with %d instruments.\n", header.Instruments);
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delete fp;
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return 0;
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}
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if (idata.Layers != 1 && idata.Layers != 0) /* layers. What's a layer? */
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{
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cmsg(CMSG_ERROR, VERB_NORMAL, "Can't handle instruments with %d layers.\n", idata.Layers);
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delete fp;
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return 0;
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}
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if (sizeof(layer_data) != fp->Read(&layer_data, sizeof(layer_data)))
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{
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goto failread;
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}
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if (layer_data.Samples == 0)
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{
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cmsg(CMSG_ERROR, VERB_NORMAL, "Instrument has 0 samples.\n");
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delete fp;
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return 0;
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}
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ip = new Instrument;
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ip->samples = layer_data.Samples;
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ip->sample = (Sample *)safe_malloc(sizeof(Sample) * layer_data.Samples);
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memset(ip->sample, 0, sizeof(Sample) * layer_data.Samples);
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for (i = 0; i < layer_data.Samples; ++i)
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{
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if (sizeof(patch_data) != fp->Read(&patch_data, sizeof(patch_data)))
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{
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fail:
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cmsg(CMSG_ERROR, VERB_NORMAL, "Error reading sample %d.\n", i);
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delete ip;
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delete fp;
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return 0;
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}
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sp = &(ip->sample[i]);
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sp->data_length = LittleLong(patch_data.WaveSize);
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sp->loop_start = LittleLong(patch_data.StartLoop);
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sp->loop_end = LittleLong(patch_data.EndLoop);
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sp->sample_rate = LittleShort(patch_data.SampleRate);
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sp->low_freq = float(LittleLong(patch_data.LowFrequency));
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sp->high_freq = float(LittleLong(patch_data.HighFrequency)) + 0.9999f;
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sp->root_freq = float(LittleLong(patch_data.RootFrequency));
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sp->high_vel = 127;
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sp->velocity = -1;
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sp->type = INST_GUS;
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// Expand to SF2 range.
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if (panning == -1)
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{
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sp->panning = (patch_data.Balance & 0x0F) * 1000 / 15 - 500;
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}
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else
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{
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sp->panning = (panning & 0x7f) * 1000 / 127 - 500;
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}
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song->compute_pan((sp->panning + 500) / 1000.0, INST_GUS, sp->left_offset, sp->right_offset);
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/* tremolo */
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if (patch_data.TremoloRate == 0 || patch_data.TremoloDepth == 0)
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{
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sp->tremolo_sweep_increment = 0;
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sp->tremolo_phase_increment = 0;
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sp->tremolo_depth = 0;
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cmsg(CMSG_INFO, VERB_DEBUG, " * no tremolo\n");
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}
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else
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{
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sp->tremolo_sweep_increment = convert_tremolo_sweep(song, patch_data.TremoloSweep);
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sp->tremolo_phase_increment = convert_tremolo_rate(song, patch_data.TremoloRate);
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sp->tremolo_depth = patch_data.TremoloDepth;
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cmsg(CMSG_INFO, VERB_DEBUG, " * tremolo: sweep %d, phase %d, depth %d\n",
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sp->tremolo_sweep_increment, sp->tremolo_phase_increment, sp->tremolo_depth);
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}
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/* vibrato */
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if (patch_data.VibratoRate == 0 || patch_data.VibratoDepth == 0)
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{
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sp->vibrato_sweep_increment = 0;
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sp->vibrato_control_ratio = 0;
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sp->vibrato_depth = 0;
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cmsg(CMSG_INFO, VERB_DEBUG, " * no vibrato\n");
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}
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else
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{
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sp->vibrato_control_ratio = convert_vibrato_rate(song, patch_data.VibratoRate);
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sp->vibrato_sweep_increment = convert_vibrato_sweep(song, patch_data.VibratoSweep, sp->vibrato_control_ratio);
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sp->vibrato_depth = patch_data.VibratoDepth;
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cmsg(CMSG_INFO, VERB_DEBUG, " * vibrato: sweep %d, ctl %d, depth %d\n",
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sp->vibrato_sweep_increment, sp->vibrato_control_ratio, sp->vibrato_depth);
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}
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sp->modes = patch_data.Modes;
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/* Mark this as a fixed-pitch instrument if such a deed is desired. */
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if (note_to_use != -1)
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{
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sp->scale_note = note_to_use;
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sp->scale_factor = 0;
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}
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else
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{
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sp->scale_note = LittleShort(patch_data.ScaleFrequency);
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sp->scale_factor = LittleShort(patch_data.ScaleFactor);
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if (sp->scale_factor <= 2)
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{
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sp->scale_factor *= 1024;
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}
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else if (sp->scale_factor > 2048)
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{
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sp->scale_factor = 1024;
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}
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if (sp->scale_factor != 1024)
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{
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cmsg(CMSG_INFO, VERB_DEBUG, " * Scale: note %d, factor %d\n",
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sp->scale_note, sp->scale_factor);
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}
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}
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#if 0
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/* seashore.pat in the Midia patch set has no Sustain. I don't
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understand why, and fixing it by adding the Sustain flag to
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all looped patches probably breaks something else. We do it
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anyway. */
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if (sp->modes & PATCH_LOOPEN)
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{
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sp->modes |= PATCH_SUSTAIN;
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}
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#endif
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/* [RH] Alas, eawpats has percussion instruments with bad envelopes. :(
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* (See cymchina.pat for one example of this sadness.)
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* Do this logic for instruments without a description, only. Hopefully that
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* catches all the patches that need it without including any extra.
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*/
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for (j = 0; j < DESC_SIZE; ++j)
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{
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if (header.Description[j] != 0)
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break;
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}
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/* Strip any loops and envelopes we're permitted to */
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/* [RH] (But PATCH_BACKWARD isn't a loop flag at all!) */
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if ((strip_loop == 1) &&
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(sp->modes & (PATCH_SUSTAIN | PATCH_LOOPEN | PATCH_BIDIR | PATCH_BACKWARD)))
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{
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cmsg(CMSG_INFO, VERB_DEBUG, " - Removing loop and/or sustain\n");
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if (j == DESC_SIZE)
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{
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sp->modes &= ~(PATCH_SUSTAIN | PATCH_LOOPEN | PATCH_BIDIR | PATCH_BACKWARD);
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}
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sp->modes |= PATCH_T_NO_LOOP;
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}
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if (strip_envelope == 1)
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{
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cmsg(CMSG_INFO, VERB_DEBUG, " - Removing envelope\n");
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/* [RH] The envelope isn't really removed, but this is the way the standard
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* Gravis patches get that effect: All rates at maximum, and all offsets at
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* a constant level.
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*/
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if (j == DESC_SIZE)
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{
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int k;
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for (k = 1; k < ENVELOPES; ++k)
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{ /* Find highest offset. */
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if (patch_data.EnvelopeOffset[k] > patch_data.EnvelopeOffset[0])
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{
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patch_data.EnvelopeOffset[0] = patch_data.EnvelopeOffset[k];
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}
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}
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for (k = 0; k < ENVELOPES; ++k)
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{
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patch_data.EnvelopeRate[k] = 63;
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patch_data.EnvelopeOffset[k] = patch_data.EnvelopeOffset[0];
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}
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}
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sp->modes |= PATCH_T_NO_ENVELOPE;
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}
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else if (strip_envelope != 0)
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{
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/* Have to make a guess. */
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if (!(sp->modes & (PATCH_LOOPEN | PATCH_BIDIR | PATCH_BACKWARD)))
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{
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/* No loop? Then what's there to sustain? No envelope needed either... */
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sp->modes |= PATCH_T_NO_ENVELOPE;
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cmsg(CMSG_INFO, VERB_DEBUG, " - No loop, removing sustain and envelope\n");
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}
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else if (memcmp(patch_data.EnvelopeRate, "??????", 6) == 0 || patch_data.EnvelopeOffset[GF1_RELEASEC] >= 100)
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{
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/* Envelope rates all maxed out? Envelope end at a high "offset"?
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That's a weird envelope. Take it out. */
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sp->modes |= PATCH_T_NO_ENVELOPE;
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cmsg(CMSG_INFO, VERB_DEBUG, " - Weirdness, removing envelope\n");
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}
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else if (!(sp->modes & PATCH_SUSTAIN))
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{
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/* No sustain? Then no envelope. I don't know if this is
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justified, but patches without sustain usually don't need the
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envelope either... at least the Gravis ones. They're mostly
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drums. I think. */
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sp->modes |= PATCH_T_NO_ENVELOPE;
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cmsg(CMSG_INFO, VERB_DEBUG, " - No sustain, removing envelope\n");
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}
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}
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if (!(sp->modes & PATCH_NO_SRELEASE))
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{ // TiMidity thinks that this is an envelope enable flag.
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sp->modes |= PATCH_T_NO_ENVELOPE;
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}
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for (j = 0; j < 6; j++)
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{
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sp->envelope.gf1.rate[j] = patch_data.EnvelopeRate[j];
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/* [RH] GF1NEW clamps the offsets to the range [5,251], so we do too. */
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sp->envelope.gf1.offset[j] = clamp<BYTE>(patch_data.EnvelopeOffset[j], 5, 251);
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}
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/* Then read the sample data */
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if (((sp->modes & PATCH_16) && sp->data_length/2 > MAX_SAMPLE_SIZE) ||
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(!(sp->modes & PATCH_16) && sp->data_length > MAX_SAMPLE_SIZE))
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{
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goto fail;
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}
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sp->data = (sample_t *)safe_malloc(sp->data_length);
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if (sp->data_length != fp->Read(sp->data, sp->data_length))
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goto fail;
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convert_sample_data(sp, sp->data);
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/* Reverse reverse loops and pass them off as normal loops */
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if (sp->modes & PATCH_BACKWARD)
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{
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int t;
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/* The GUS apparently plays reverse loops by reversing the
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whole sample. We do the same because the GUS does not SUCK. */
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cmsg(CMSG_WARNING, VERB_NORMAL, "Reverse loop in %s\n", name);
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reverse_data((sample_t *)sp->data, 0, sp->data_length);
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sp->data[sp->data_length] = sp->data[sp->data_length - 1];
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t = sp->loop_start;
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sp->loop_start = sp->data_length - sp->loop_end;
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sp->loop_end = sp->data_length - t;
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sp->modes &= ~PATCH_BACKWARD;
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sp->modes |= PATCH_LOOPEN; /* just in case */
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}
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if (amp != -1)
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{
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sp->volume = (amp) / 100.f;
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}
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else
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{
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/* Try to determine a volume scaling factor for the sample.
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This is a very crude adjustment, but things sound more
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balanced with it. Still, this should be a runtime option.
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(This is ignored unless midi_timiditylike is turned on.) */
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int i;
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sample_t maxamp = 0, a;
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sample_t *tmp;
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for (i = sp->data_length, tmp = sp->data; i; --i)
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{
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a = abs(*tmp++);
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if (a > maxamp)
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maxamp = a;
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}
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sp->volume = 1 / maxamp;
|
|
cmsg(CMSG_INFO, VERB_DEBUG, " * volume comp: %f\n", sp->volume);
|
|
}
|
|
|
|
/* Then fractional samples */
|
|
sp->data_length <<= FRACTION_BITS;
|
|
sp->loop_start <<= FRACTION_BITS;
|
|
sp->loop_end <<= FRACTION_BITS;
|
|
|
|
/* Adjust for fractional loop points. */
|
|
sp->loop_start |= (patch_data.Fractions & 0x0F) << (FRACTION_BITS-4);
|
|
sp->loop_end |= (patch_data.Fractions & 0xF0) << (FRACTION_BITS-4-4);
|
|
|
|
/* If this instrument will always be played on the same note,
|
|
and it's not looped, we can resample it now. */
|
|
if (sp->scale_factor == 0 && !(sp->modes & PATCH_LOOPEN))
|
|
{
|
|
pre_resample(song, sp);
|
|
}
|
|
|
|
if (strip_tail == 1)
|
|
{
|
|
/* Let's not really, just say we did. */
|
|
cmsg(CMSG_INFO, VERB_DEBUG, " - Stripping tail\n");
|
|
sp->data_length = sp->loop_end;
|
|
}
|
|
}
|
|
delete fp;
|
|
return ip;
|
|
}
|
|
|
|
void convert_sample_data(Sample *sp, const void *data)
|
|
{
|
|
/* convert everything to 32-bit floating point data */
|
|
sample_t *newdata = NULL;
|
|
|
|
switch (sp->modes & (PATCH_16 | PATCH_UNSIGNED))
|
|
{
|
|
case 0:
|
|
{ /* 8-bit, signed */
|
|
SBYTE *cp = (SBYTE *)data;
|
|
newdata = (sample_t *)safe_malloc((sp->data_length + 1) * sizeof(sample_t));
|
|
for (int i = 0; i < sp->data_length; ++i)
|
|
{
|
|
if (cp[i] < 0)
|
|
{
|
|
newdata[i] = float(cp[i]) / 128.f;
|
|
}
|
|
else
|
|
{
|
|
newdata[i] = float(cp[i]) / 127.f;
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
|
|
case PATCH_UNSIGNED:
|
|
{ /* 8-bit, unsigned */
|
|
BYTE *cp = (BYTE *)data;
|
|
newdata = (sample_t *)safe_malloc((sp->data_length + 1) * sizeof(sample_t));
|
|
for (int i = 0; i < sp->data_length; ++i)
|
|
{
|
|
int c = cp[i] - 128;
|
|
if (c < 0)
|
|
{
|
|
newdata[i] = float(c) / 128.f;
|
|
}
|
|
else
|
|
{
|
|
newdata[i] = float(c) / 127.f;
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
|
|
case PATCH_16:
|
|
{ /* 16-bit, signed */
|
|
SWORD *cp = (SWORD *)data;
|
|
/* Convert these to samples */
|
|
sp->data_length >>= 1;
|
|
sp->loop_start >>= 1;
|
|
sp->loop_end >>= 1;
|
|
newdata = (sample_t *)safe_malloc((sp->data_length + 1) * sizeof(sample_t));
|
|
for (int i = 0; i < sp->data_length; ++i)
|
|
{
|
|
int c = LittleShort(cp[i]);
|
|
if (c < 0)
|
|
{
|
|
newdata[i] = float(c) / 32768.f;
|
|
}
|
|
else
|
|
{
|
|
newdata[i] = float(c) / 32767.f;
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
|
|
case PATCH_16 | PATCH_UNSIGNED:
|
|
{ /* 16-bit, unsigned */
|
|
WORD *cp = (WORD *)data;
|
|
/* Convert these to samples */
|
|
sp->data_length >>= 1;
|
|
sp->loop_start >>= 1;
|
|
sp->loop_end >>= 1;
|
|
newdata = (sample_t *)safe_malloc((sp->data_length + 1) * sizeof(sample_t));
|
|
for (int i = 0; i < sp->data_length; ++i)
|
|
{
|
|
int c = LittleShort(cp[i]) - 32768;
|
|
if (c < 0)
|
|
{
|
|
newdata[i] = float(c) / 32768.f;
|
|
}
|
|
else
|
|
{
|
|
newdata[i] = float(c) / 32767.f;
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
/* Duplicate the final sample for linear interpolation. */
|
|
newdata[sp->data_length] = newdata[sp->data_length - 1];
|
|
if (sp->data != NULL)
|
|
{
|
|
free(sp->data);
|
|
}
|
|
sp->data = newdata;
|
|
}
|
|
|
|
static int fill_bank(Renderer *song, int dr, int b)
|
|
{
|
|
int i, errors = 0;
|
|
ToneBank *bank = ((dr) ? drumset[b] : tonebank[b]);
|
|
if (bank == NULL)
|
|
{
|
|
cmsg(CMSG_ERROR, VERB_NORMAL,
|
|
"Huh. Tried to load instruments in non-existent %s %d\n",
|
|
(dr) ? "drumset" : "tone bank", b);
|
|
return 0;
|
|
}
|
|
for (i = 0; i < MAXPROG; i++)
|
|
{
|
|
if (bank->instrument[i] == MAGIC_LOAD_INSTRUMENT)
|
|
{
|
|
bank->instrument[i] = NULL;
|
|
bank->instrument[i] = load_instrument_dls(song, dr, b, i);
|
|
if (bank->instrument[i] != NULL)
|
|
{
|
|
continue;
|
|
}
|
|
Instrument *ip;
|
|
ip = load_instrument_font_order(song, 0, dr, b, i);
|
|
if (ip == NULL)
|
|
{
|
|
if (bank->tone[i].fontbank >= 0)
|
|
{
|
|
ip = load_instrument_font(song, bank->tone[i].name, dr, b, i);
|
|
}
|
|
else
|
|
{
|
|
ip = load_instrument(song, bank->tone[i].name,
|
|
(dr) ? 1 : 0,
|
|
bank->tone[i].pan,
|
|
bank->tone[i].amp,
|
|
(bank->tone[i].note != -1) ? bank->tone[i].note : ((dr) ? i : -1),
|
|
(bank->tone[i].strip_loop != -1) ? bank->tone[i].strip_loop : ((dr) ? 1 : -1),
|
|
(bank->tone[i].strip_envelope != -1) ? bank->tone[i].strip_envelope : ((dr) ? 1 : -1),
|
|
bank->tone[i].strip_tail);
|
|
}
|
|
if (ip == NULL)
|
|
{
|
|
ip = load_instrument_font_order(song, 1, dr, b, i);
|
|
}
|
|
}
|
|
bank->instrument[i] = ip;
|
|
if (ip == NULL)
|
|
{
|
|
if (bank->tone[i].name.IsEmpty())
|
|
{
|
|
cmsg(CMSG_WARNING, (b != 0) ? VERB_VERBOSE : VERB_NORMAL,
|
|
"No instrument mapped to %s %d, program %d%s\n",
|
|
(dr) ? "drum set" : "tone bank", b, i,
|
|
(b != 0) ? "" : " - this instrument will not be heard");
|
|
}
|
|
else
|
|
{
|
|
cmsg(CMSG_ERROR, VERB_NORMAL,
|
|
"Couldn't load instrument %s (%s %d, program %d)\n",
|
|
bank->tone[i].name.GetChars(),
|
|
(dr) ? "drum set" : "tone bank", b, i);
|
|
}
|
|
if (b != 0)
|
|
{
|
|
/* Mark the corresponding instrument in the default
|
|
bank / drumset for loading (if it isn't already) */
|
|
if (((dr) ? drumset[0] : tonebank[0])->instrument[i] != NULL)
|
|
{
|
|
((dr) ? drumset[0] : tonebank[0])->instrument[i] = MAGIC_LOAD_INSTRUMENT;
|
|
}
|
|
}
|
|
errors++;
|
|
}
|
|
}
|
|
}
|
|
return errors;
|
|
}
|
|
|
|
int Renderer::load_missing_instruments()
|
|
{
|
|
int i = MAXBANK, errors = 0;
|
|
while (i--)
|
|
{
|
|
if (tonebank[i] != NULL)
|
|
errors += fill_bank(this, 0,i);
|
|
if (drumset[i] != NULL)
|
|
errors += fill_bank(this, 1,i);
|
|
}
|
|
return errors;
|
|
}
|
|
|
|
void free_instruments()
|
|
{
|
|
int i = MAXBANK;
|
|
while (i--)
|
|
{
|
|
if (tonebank[i] != NULL)
|
|
{
|
|
delete tonebank[i];
|
|
tonebank[i] = NULL;
|
|
}
|
|
if (drumset[i] != NULL)
|
|
{
|
|
delete drumset[i];
|
|
drumset[i] = NULL;
|
|
}
|
|
}
|
|
}
|
|
|
|
int Renderer::set_default_instrument(const char *name)
|
|
{
|
|
Instrument *ip;
|
|
if ((ip = load_instrument(this, name, 0, -1, -1, -1, 0, 0, 0)) == NULL)
|
|
{
|
|
return -1;
|
|
}
|
|
if (default_instrument != NULL)
|
|
{
|
|
delete default_instrument;
|
|
}
|
|
default_instrument = ip;
|
|
default_program = SPECIAL_PROGRAM;
|
|
return 0;
|
|
}
|
|
|
|
}
|