/* TiMidity -- Experimental MIDI to WAVE converter Copyright (C) 1995 Tuukka Toivonen 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 #include #include #include #include "timidity.h" #include "m_swap.h" #include "files.h" #include "templates.h" #include "gf1patch.h" namespace Timidity { extern 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 convert_tremolo_sweep(Renderer *song, BYTE sweep) { if (sweep == 0) return 0; return int(((song->control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) / (song->rate * sweep)); } int convert_vibrato_sweep(Renderer *song, BYTE sweep, int vib_control_ratio) { if (sweep == 0) return 0; return (int) (FSCALE((double) (vib_control_ratio) * SWEEP_TUNING, SWEEP_SHIFT) / (song->rate * sweep)); /* this was overflowing with seashore.pat ((vib_control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) / (song->rate * sweep); */ } int convert_tremolo_rate(Renderer *song, BYTE rate) { return int(((song->control_ratio * rate) << RATE_SHIFT) / (TREMOLO_RATE_TUNING * song->rate)); } int convert_vibrato_rate(Renderer *song, BYTE rate) { /* Return a suitable vibrato_control_ratio value */ return int((VIBRATO_RATE_TUNING * song->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 */ static Instrument *load_instrument(Renderer *song, const char *name, int percussion, int panning, int amp, int note_to_use, int strip_loop, int strip_envelope, int strip_tail) { Instrument *ip; Sample *sp; FileReader *fp; GF1PatchHeader header; GF1InstrumentData idata; GF1LayerData layer_data; GF1PatchData patch_data; int i, j; bool noluck = false; if (!name) return 0; /* Open patch file */ if ((fp = open_filereader(name, openmode, NULL)) == NULL) { /* Try with various extensions */ FString tmp = name; tmp += ".pat"; if ((fp = open_filereader(tmp, openmode, NULL)) == NULL) { #ifdef __unix__ // Windows isn't case-sensitive. tmp.ToUpper(); if ((fp = open_filereader(tmp, openmode, NULL)) == NULL) #endif { noluck = true; } } } if (noluck) { cmsg(CMSG_ERROR, VERB_NORMAL, "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); delete fp; return 0; } if (strncmp(header.Header, GF1_HEADER_TEXT, HEADER_SIZE - 4) != 0) { cmsg(CMSG_ERROR, VERB_NORMAL, "%s: Not an instrument.\n", name); delete fp; return 0; } if (strcmp(header.Header + 8, "110") < 0) { cmsg(CMSG_ERROR, VERB_NORMAL, "%s: Is an old and unsupported patch version.\n", name); delete fp; 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); delete fp; 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); delete fp; 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"); delete fp; 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; delete fp; 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; } song->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(song, patch_data.TremoloSweep); sp->tremolo_phase_increment = convert_tremolo_rate(song, 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(song, patch_data.VibratoRate); sp->vibrato_sweep_increment = convert_vibrato_sweep(song, 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); } sp->modes |= PATCH_T_NO_LOOP; } 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]; } } sp->modes |= PATCH_T_NO_ENVELOPE; } else if (strip_envelope != 0) { /* Have to make a guess. */ if (!(sp->modes & (PATCH_LOOPEN | PATCH_BIDIR | PATCH_BACKWARD))) { /* No loop? Then what's there to sustain? No envelope needed either... */ sp->modes |= PATCH_T_NO_ENVELOPE; cmsg(CMSG_INFO, VERB_DEBUG, " - No loop, removing sustain and envelope\n"); } else if (memcmp(patch_data.EnvelopeRate, "??????", 6) == 0 || patch_data.EnvelopeOffset[GF1_RELEASEC] >= 100) { /* Envelope rates all maxed out? Envelope end at a high "offset"? That's a weird envelope. Take it out. */ sp->modes |= PATCH_T_NO_ENVELOPE; cmsg(CMSG_INFO, VERB_DEBUG, " - Weirdness, removing envelope\n"); } else if (!(sp->modes & PATCH_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 |= PATCH_T_NO_ENVELOPE; cmsg(CMSG_INFO, VERB_DEBUG, " - No sustain, removing envelope\n"); } } if (!(sp->modes & PATCH_NO_SRELEASE)) { // TiMidity thinks that this is an envelope enable flag. sp->modes |= PATCH_T_NO_ENVELOPE; } 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] = clamp(patch_data.EnvelopeOffset[j], 5, 251); } /* 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 */ } if (amp != -1) { sp->volume = (amp) / 100.f; } 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. (This is ignored unless midi_timiditylike is turned on.) */ int i; sample_t maxamp = 0, a; sample_t *tmp; for (i = sp->data_length, tmp = sp->data; i; --i) { a = abs(*tmp++); if (a > maxamp) maxamp = a; } 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; } }