gzdoom-gles/libraries/timidity/instrum.cpp
2019-09-24 23:43:44 +02:00

729 lines
18 KiB
C++

/*
TiMidity -- Experimental MIDI to WAVE converter
Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
instrum.c
Code to load and unload GUS-compatible instrument patches.
*/
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <math.h>
#include <memory>
#include <algorithm>
#include "timidity.h"
#include "gf1patch.h"
#include "timidity_file.h"
#include "t_swap.h"
#include "common.h"
#include "instrum.h"
#include "playmidi.h"
namespace Timidity
{
Instrument *load_instrument_dls(Renderer *song, int drum, int bank, int instrument);
Instrument::Instrument()
: samples(0), sample(NULL)
{
}
Instrument::~Instrument()
{
Sample *sp;
int i;
for (i = samples, sp = &(sample[0]); i != 0; i--, sp++)
{
if (sp->type == INST_GUS && sp->data != NULL)
{
free(sp->data);
}
}
free(sample);
}
ToneBank::ToneBank()
{
tone = new ToneBankElement[128];;
for (int i = 0; i < MAXPROG; ++i)
{
instrument[i] = 0;
}
}
ToneBank::~ToneBank()
{
delete[] tone;
for (int i = 0; i < MAXPROG; i++)
{
if (instrument[i] != NULL && instrument[i] != MAGIC_LOAD_INSTRUMENT)
{
delete instrument[i];
instrument[i] = NULL;
}
}
}
int Renderer::convert_tremolo_sweep(uint8_t sweep)
{
if (sweep == 0)
return 0;
return
int(((control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) / (rate * sweep));
}
int Renderer::convert_vibrato_sweep(uint8_t sweep, int vib_control_ratio)
{
if (sweep == 0)
return 0;
return
(int)(FSCALE((double)(vib_control_ratio)* SWEEP_TUNING, SWEEP_SHIFT) / (rate * sweep));
/* this was overflowing with seashore.pat
((vib_control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) / (song->rate * sweep);
*/
}
int Renderer::convert_tremolo_rate(uint8_t rate)
{
return
int(((control_ratio * rate) << RATE_SHIFT) / (TREMOLO_RATE_TUNING * rate));
}
int Renderer::convert_vibrato_rate(uint8_t rate)
{
/* Return a suitable vibrato_control_ratio value */
return
int((VIBRATO_RATE_TUNING * rate) / (rate * 2 * VIBRATO_SAMPLE_INCREMENTS));
}
static void reverse_data(sample_t *sp, int ls, int le)
{
sample_t s, *ep = sp + le;
sp += ls;
le -= ls;
le /= 2;
while (le--)
{
s = *sp;
*sp++ = *ep;
*ep-- = s;
}
}
/*
If panning or note_to_use != -1, it will be used for all samples,
instead of the sample-specific values in the instrument file.
For note_to_use, any value <0 or >127 will be forced to 0.
For other parameters, 1 means yes, 0 means no, other values are
undefined.
TODO: do reverse loops right */
Instrument *Renderer::load_instrument(const char *name, int percussion,
int panning, int note_to_use,
int strip_loop, int strip_envelope,
int strip_tail)
{
Instrument *ip;
Sample *sp;
GF1PatchHeader header;
GF1InstrumentData idata;
GF1LayerData layer_data;
GF1PatchData patch_data;
int i, j;
bool noluck = false;
auto reader = instruments->sfreader;
if (!name || reader == nullptr) return nullptr;
/* Open patch file */
auto fp = reader->open_timidity_file(name);
if (!fp)
{
/* Try with various extensions */
std::string tmp = name;
tmp += ".pat";
fp = reader->open_timidity_file(tmp.c_str());
if (!fp)
{
#ifndef _WIN32 // Windows isn't case-sensitive.
std::transform(tmp.begin(), tmp.end(), tmp.begin(), [](unsigned char c){ return toupper(c); } );
fp = reader->open_timidity_file(tmp.c_str());
if (!fp)
#endif
{
noluck = true;
}
}
}
if (noluck)
{
cmsg(CMSG_ERROR, VERB_DEBUG, "Instrument `%s' can't be found.\n", name);
return 0;
}
cmsg(CMSG_INFO, VERB_NOISY, "Loading instrument %s\n", name);
/* Read some headers and do cursory sanity checks. */
if (sizeof(header) != fp->read(&header, sizeof(header)))
{
failread:
cmsg(CMSG_ERROR, VERB_NORMAL, "%s: Error reading instrument.\n", name);
fp->close();
return 0;
}
if (strncmp(header.Header, GF1_HEADER_TEXT, HEADER_SIZE - 4) != 0)
{
cmsg(CMSG_ERROR, VERB_NORMAL, "%s: Not an instrument.\n", name);
fp->close();
return 0;
}
if (strcmp(header.Header + 8, "110") < 0)
{
cmsg(CMSG_ERROR, VERB_NORMAL, "%s: Is an old and unsupported patch version.\n", name);
fp->close();
return 0;
}
if (sizeof(idata) != fp->read(&idata, sizeof(idata)))
{
goto failread;
}
header.WaveForms = LittleShort(header.WaveForms);
header.MasterVolume = LittleShort(header.MasterVolume);
header.DataSize = LittleLong(header.DataSize);
idata.Instrument = LittleShort(idata.Instrument);
if (header.Instruments != 1 && header.Instruments != 0) /* instruments. To some patch makers, 0 means 1 */
{
cmsg(CMSG_ERROR, VERB_NORMAL, "Can't handle patches with %d instruments.\n", header.Instruments);
fp->close();
return 0;
}
if (idata.Layers != 1 && idata.Layers != 0) /* layers. What's a layer? */
{
cmsg(CMSG_ERROR, VERB_NORMAL, "Can't handle instruments with %d layers.\n", idata.Layers);
fp->close();
return 0;
}
if (sizeof(layer_data) != fp->read(&layer_data, sizeof(layer_data)))
{
goto failread;
}
if (layer_data.Samples == 0)
{
cmsg(CMSG_ERROR, VERB_NORMAL, "Instrument has 0 samples.\n");
fp->close();
return 0;
}
ip = new Instrument;
ip->samples = layer_data.Samples;
ip->sample = (Sample *)safe_malloc(sizeof(Sample) * layer_data.Samples);
memset(ip->sample, 0, sizeof(Sample) * layer_data.Samples);
for (i = 0; i < layer_data.Samples; ++i)
{
if (sizeof(patch_data) != fp->read(&patch_data, sizeof(patch_data)))
{
fail:
cmsg(CMSG_ERROR, VERB_NORMAL, "Error reading sample %d.\n", i);
delete ip;
fp->close();
return 0;
}
sp = &(ip->sample[i]);
sp->data_length = LittleLong(patch_data.WaveSize);
sp->loop_start = LittleLong(patch_data.StartLoop);
sp->loop_end = LittleLong(patch_data.EndLoop);
sp->sample_rate = LittleShort(patch_data.SampleRate);
sp->low_freq = float(LittleLong(patch_data.LowFrequency));
sp->high_freq = float(LittleLong(patch_data.HighFrequency)) + 0.9999f;
sp->root_freq = float(LittleLong(patch_data.RootFrequency));
sp->high_vel = 127;
sp->velocity = -1;
sp->type = INST_GUS;
// Expand to SF2 range.
if (panning == -1)
{
sp->panning = (patch_data.Balance & 0x0F) * 1000 / 15 - 500;
}
else
{
sp->panning = (panning & 0x7f) * 1000 / 127 - 500;
}
compute_pan((sp->panning + 500) / 1000.0, INST_GUS, sp->left_offset, sp->right_offset);
/* tremolo */
if (patch_data.TremoloRate == 0 || patch_data.TremoloDepth == 0)
{
sp->tremolo_sweep_increment = 0;
sp->tremolo_phase_increment = 0;
sp->tremolo_depth = 0;
cmsg(CMSG_INFO, VERB_DEBUG, " * no tremolo\n");
}
else
{
sp->tremolo_sweep_increment = convert_tremolo_sweep(patch_data.TremoloSweep);
sp->tremolo_phase_increment = convert_tremolo_rate(patch_data.TremoloRate);
sp->tremolo_depth = patch_data.TremoloDepth;
cmsg(CMSG_INFO, VERB_DEBUG, " * tremolo: sweep %d, phase %d, depth %d\n",
sp->tremolo_sweep_increment, sp->tremolo_phase_increment, sp->tremolo_depth);
}
/* vibrato */
if (patch_data.VibratoRate == 0 || patch_data.VibratoDepth == 0)
{
sp->vibrato_sweep_increment = 0;
sp->vibrato_control_ratio = 0;
sp->vibrato_depth = 0;
cmsg(CMSG_INFO, VERB_DEBUG, " * no vibrato\n");
}
else
{
sp->vibrato_control_ratio = convert_vibrato_rate(patch_data.VibratoRate);
sp->vibrato_sweep_increment = convert_vibrato_sweep(patch_data.VibratoSweep, sp->vibrato_control_ratio);
sp->vibrato_depth = patch_data.VibratoDepth;
cmsg(CMSG_INFO, VERB_DEBUG, " * vibrato: sweep %d, ctl %d, depth %d\n",
sp->vibrato_sweep_increment, sp->vibrato_control_ratio, sp->vibrato_depth);
}
sp->modes = patch_data.Modes;
/* Mark this as a fixed-pitch instrument if such a deed is desired. */
if (note_to_use != -1)
{
sp->scale_note = note_to_use;
sp->scale_factor = 0;
}
else
{
sp->scale_note = LittleShort(patch_data.ScaleFrequency);
sp->scale_factor = LittleShort(patch_data.ScaleFactor);
if (sp->scale_factor <= 2)
{
sp->scale_factor *= 1024;
}
else if (sp->scale_factor > 2048)
{
sp->scale_factor = 1024;
}
if (sp->scale_factor != 1024)
{
cmsg(CMSG_INFO, VERB_DEBUG, " * Scale: note %d, factor %d\n",
sp->scale_note, sp->scale_factor);
}
}
#if 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 & PATCH_LOOPEN)
{
sp->modes |= PATCH_SUSTAIN;
}
#endif
/* [RH] Alas, eawpats has percussion instruments with bad envelopes. :(
* (See cymchina.pat for one example of this sadness.)
* Do this logic for instruments without a description, only. Hopefully that
* catches all the patches that need it without including any extra.
*/
for (j = 0; j < DESC_SIZE; ++j)
{
if (header.Description[j] != 0)
break;
}
/* Strip any loops and envelopes we're permitted to */
/* [RH] (But PATCH_BACKWARD isn't a loop flag at all!) */
if ((strip_loop == 1) &&
(sp->modes & (PATCH_SUSTAIN | PATCH_LOOPEN | PATCH_BIDIR | PATCH_BACKWARD)))
{
cmsg(CMSG_INFO, VERB_DEBUG, " - Removing loop and/or sustain\n");
if (j == DESC_SIZE)
{
sp->modes &= ~(PATCH_SUSTAIN | PATCH_LOOPEN | PATCH_BIDIR | PATCH_BACKWARD);
}
}
if (strip_envelope == 1)
{
cmsg(CMSG_INFO, VERB_DEBUG, " - Removing envelope\n");
/* [RH] The envelope isn't really removed, but this is the way the standard
* Gravis patches get that effect: All rates at maximum, and all offsets at
* a constant level.
*/
if (j == DESC_SIZE)
{
int k;
for (k = 1; k < ENVELOPES; ++k)
{ /* Find highest offset. */
if (patch_data.EnvelopeOffset[k] > patch_data.EnvelopeOffset[0])
{
patch_data.EnvelopeOffset[0] = patch_data.EnvelopeOffset[k];
}
}
for (k = 0; k < ENVELOPES; ++k)
{
patch_data.EnvelopeRate[k] = 63;
patch_data.EnvelopeOffset[k] = patch_data.EnvelopeOffset[0];
}
}
}
for (j = 0; j < 6; j++)
{
sp->envelope.gf1.rate[j] = patch_data.EnvelopeRate[j];
/* [RH] GF1NEW clamps the offsets to the range [5,251], so we do too. */
sp->envelope.gf1.offset[j] = std::max<uint8_t>(5, std::min<uint8_t>(251, patch_data.EnvelopeOffset[j]));
}
/* Then read the sample data */
if (((sp->modes & PATCH_16) && sp->data_length/2 > MAX_SAMPLE_SIZE) ||
(!(sp->modes & PATCH_16) && sp->data_length > MAX_SAMPLE_SIZE))
{
goto fail;
}
sp->data = (sample_t *)safe_malloc(sp->data_length);
if (sp->data_length != fp->read(sp->data, sp->data_length))
goto fail;
convert_sample_data(sp, sp->data);
/* Reverse reverse loops and pass them off as normal loops */
if (sp->modes & PATCH_BACKWARD)
{
int t;
/* The GUS apparently plays reverse loops by reversing the
whole sample. We do the same because the GUS does not SUCK. */
cmsg(CMSG_WARNING, VERB_NORMAL, "Reverse loop in %s\n", name);
reverse_data((sample_t *)sp->data, 0, sp->data_length);
sp->data[sp->data_length] = sp->data[sp->data_length - 1];
t = sp->loop_start;
sp->loop_start = sp->data_length - sp->loop_end;
sp->loop_end = sp->data_length - t;
sp->modes &= ~PATCH_BACKWARD;
sp->modes |= PATCH_LOOPEN; /* just in case */
}
/* 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(this, 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;
}
}
fp->close();
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 */
int8_t *cp = (int8_t *)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 */
uint8_t *cp = (uint8_t *)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 */
int16_t *cp = (int16_t *)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 */
auto *cp = (uint16_t *)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;
}
int Renderer::fill_bank(int dr, int b)
{
int i, errors = 0;
ToneBank *bank = ((dr) ? instruments->drumset[b] : instruments->tonebank[b]);
if (bank == NULL)
{
cmsg(CMSG_ERROR, VERB_NORMAL,
"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(this, dr, b, i);
if (bank->instrument[i] != NULL)
{
continue;
}
Instrument *ip;
ip = load_instrument_font_order(0, dr, b, i);
if (ip == NULL)
{
if (bank->tone[i].fontbank >= 0)
{
ip = load_instrument_font(bank->tone[i].name.c_str(), dr, b, i);
}
else
{
ip = load_instrument(bank->tone[i].name.c_str(),
(dr) ? 1 : 0,
bank->tone[i].pan,
(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(1, dr, b, i);
}
}
bank->instrument[i] = ip;
if (ip == NULL)
{
if (bank->tone[i].name.length() == 0)
{
cmsg(CMSG_WARNING, (b != 0) ? VERB_VERBOSE : VERB_DEBUG,
"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_DEBUG,
"Couldn't load instrument %s (%s %d, program %d)\n",
bank->tone[i].name.c_str(),
(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) ? instruments->drumset[0] : instruments->tonebank[0])->instrument[i] != NULL)
{
((dr) ? instruments->drumset[0] : instruments->tonebank[0])->instrument[i] = MAGIC_LOAD_INSTRUMENT;
}
}
errors++;
}
}
}
return errors;
}
int Renderer::load_missing_instruments()
{
int i = MAXBANK, errors = 0;
while (i--)
{
if (instruments->tonebank[i] != NULL)
errors += fill_bank(0, i);
if (instruments->drumset[i] != NULL)
errors += fill_bank(1, i);
}
return errors;
}
void Instruments::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;
}
}
}
Instruments::Instruments(SoundFontReaderInterface *reader)
{
sfreader = reader;
tonebank[0] = new ToneBank;
drumset[0] = new ToneBank;
}
Instruments::~Instruments()
{
free_instruments();
font_freeall();
for (int i = 0; i < MAXBANK; ++i)
{
if (tonebank[i] != NULL)
{
delete tonebank[i];
tonebank[i] = NULL;
}
if (drumset[i] != NULL)
{
delete drumset[i];
drumset[i] = NULL;
}
}
if (sfreader != nullptr) delete sfreader;
sfreader = nullptr;
}
int Renderer::set_default_instrument(const char *name)
{
Instrument *ip;
if ((ip = load_instrument(name, 0, -1, -1, 0, 0, 0)) == NULL)
{
return -1;
}
if (default_instrument != NULL)
{
delete default_instrument;
}
default_instrument = ip;
default_program = SPECIAL_PROGRAM;
return 0;
}
}