Mercurial > audlegacy-plugins
view src/timidity/libtimidity/instrum.c @ 1062:1d65dcdb714c trunk
[svn] - bump version of plugins and version requirements per plugin api v2
author | nenolod |
---|---|
date | Thu, 24 May 2007 02:12:43 -0700 |
parents | 3da1b8942b8b |
children | f14d11bf9cbb |
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/* TiMidity -- Experimental MIDI to WAVE converter Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi> This program is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details. You should have received a copy of the GNU General Public License along with this program; if not, write to the Free Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. instrum.c Code to load and unload GUS-compatible instrument patches. */ #if HAVE_CONFIG_H # include <config.h> #endif #include "audacious/vfs.h" #include <string.h> #include <stdlib.h> #include "timidity.h" #include "timidity_internal.h" #include "options.h" #include "common.h" #include "instrum.h" #include "instrum_dls.h" #include "resample.h" #include "tables.h" static void free_instrument(MidInstrument *ip) { MidSample *sp; int i; if (!ip) return; for (i=0; i<ip->samples; i++) { sp=&(ip->sample[i]); free(sp->data); } free(ip->sample); free(ip); } static void free_bank(MidSong *song, int dr, int b) { int i; MidToneBank *bank=((dr) ? song->drumset[b] : song->tonebank[b]); for (i=0; i<128; i++) if (bank->instrument[i]) { /* Not that this could ever happen, of course */ if (bank->instrument[i] != MAGIC_LOAD_INSTRUMENT) free_instrument(bank->instrument[i]); bank->instrument[i]=0; } } static sint32 convert_envelope_rate(MidSong *song, uint8 rate) { sint32 r; r = 3 - ((rate >> 6) & 0x3); r *= 3; r = (sint32) (rate & 0x3f) << r; /* 6.9 fixed point */ /* 15.15 fixed point. */ r = ((r * 44100) / song->rate) * song->control_ratio; #ifdef FAST_DECAY return r << 10; #else return r << 9; #endif } static sint32 convert_envelope_offset(uint8 offset) { /* This is not too good... Can anyone tell me what these values mean? Are they GUS-style "exponential" volumes? And what does that mean? */ /* 15.15 fixed point */ return offset << (7+15); } static sint32 convert_tremolo_sweep(MidSong *song, uint8 sweep) { if (!sweep) return 0; return ((song->control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) / (song->rate * sweep); } static sint32 convert_vibrato_sweep(MidSong *song, uint8 sweep, sint32 vib_control_ratio) { if (!sweep) return 0; return (sint32) (FSCALE((double) (vib_control_ratio) * SWEEP_TUNING, SWEEP_SHIFT) / (double)(song->rate * sweep)); /* this was overflowing with seashore.pat ((vib_control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) / (song->rate * sweep); */ } static sint32 convert_tremolo_rate(MidSong *song, uint8 rate) { return ((SINE_CYCLE_LENGTH * song->control_ratio * rate) << RATE_SHIFT) / (TREMOLO_RATE_TUNING * song->rate); } static sint32 convert_vibrato_rate(MidSong *song, uint8 rate) { /* Return a suitable vibrato_control_ratio value */ return (VIBRATO_RATE_TUNING * song->rate) / (rate * 2 * MID_VIBRATO_SAMPLE_INCREMENTS); } static void reverse_data(sint16 *sp, sint32 ls, sint32 le) { sint16 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 MidInstrument *load_instrument(MidSong *song, char *name, int percussion, int panning, int amp, int note_to_use, int strip_loop, int strip_envelope, int strip_tail) { MidInstrument *ip; MidSample *sp; VFSFile *fp; char tmp[1024]; int i,j,noluck=0; static char *patch_ext[] = PATCH_EXT_LIST; if (!name) return 0; /* Open patch file */ if ((fp=open_file(name)) == NULL) { noluck=1; /* Try with various extensions */ for (i=0; patch_ext[i]; i++) { if (strlen(name)+strlen(patch_ext[i])<1024) { strcpy(tmp, name); strcat(tmp, patch_ext[i]); if ((fp=open_file(tmp)) != NULL) { noluck=0; break; } } } } if (noluck) { DEBUG_MSG("Instrument `%s' can't be found.\n", name); return 0; } DEBUG_MSG("Loading instrument %s\n", tmp); /* Read some headers and do cursory sanity checks. There are loads of magic offsets. This could be rewritten... */ if ((239 != vfs_fread(tmp, 1, 239, fp)) || (memcmp(tmp, "GF1PATCH110\0ID#000002", 22) && memcmp(tmp, "GF1PATCH100\0ID#000002", 22))) /* don't know what the differences are */ { DEBUG_MSG("%s: not an instrument\n", name); return 0; } if (tmp[82] != 1 && tmp[82] != 0) /* instruments. To some patch makers, 0 means 1 */ { DEBUG_MSG("Can't handle patches with %d instruments\n", tmp[82]); return 0; } if (tmp[151] != 1 && tmp[151] != 0) /* layers. What's a layer? */ { DEBUG_MSG("Can't handle instruments with %d layers\n", tmp[151]); return 0; } ip=safe_malloc(sizeof(MidInstrument)); ip->samples = tmp[198]; ip->sample = safe_malloc(sizeof(MidSample) * ip->samples); for (i=0; i<ip->samples; i++) { uint8 fractions; sint32 tmplong; uint16 tmpshort; uint8 tmpchar; #define READ_CHAR(thing) \ if (1 != vfs_fread(&tmpchar, 1, 1, fp)) goto fail; \ thing = tmpchar; #define READ_SHORT(thing) \ if (1 != vfs_fread(&tmpshort, 2, 1, fp)) goto fail; \ thing = SWAPLE16(tmpshort); #define READ_LONG(thing) \ if (1 != vfs_fread(&tmplong, 4, 1, fp)) goto fail; \ thing = SWAPLE32(tmplong); vfs_fseek(fp, 7, SEEK_CUR); /* Skip the wave name */ if (1 != vfs_fread(&fractions, 1, 1, fp)) { fail: DEBUG_MSG("Error reading sample %d\n", i); for (j=0; j<i; j++) free(ip->sample[j].data); free(ip->sample); free(ip); return 0; } sp=&(ip->sample[i]); 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); sp->low_vel = 0; sp->high_vel = 127; vfs_fseek(fp, 2, SEEK_CUR); /* Why have a "root frequency" and then * "tuning"?? */ READ_CHAR(tmp[0]); if (panning==-1) sp->panning = (tmp[0] * 8 + 4) & 0x7f; else sp->panning=(uint8)(panning & 0x7F); /* envelope, tremolo, and vibrato */ if (18 != vfs_fread(tmp, 1, 18, fp)) goto fail; if (!tmp[13] || !tmp[14]) { sp->tremolo_sweep_increment= sp->tremolo_phase_increment=sp->tremolo_depth=0; DEBUG_MSG(" * no tremolo\n"); } else { sp->tremolo_sweep_increment=convert_tremolo_sweep(song, tmp[12]); sp->tremolo_phase_increment=convert_tremolo_rate(song, tmp[13]); sp->tremolo_depth=tmp[14]; DEBUG_MSG(" * tremolo: sweep %d, phase %d, depth %d\n", sp->tremolo_sweep_increment, sp->tremolo_phase_increment, sp->tremolo_depth); } if (!tmp[16] || !tmp[17]) { sp->vibrato_sweep_increment= sp->vibrato_control_ratio=sp->vibrato_depth=0; DEBUG_MSG(" * no vibrato\n"); } else { sp->vibrato_control_ratio=convert_vibrato_rate(song, tmp[16]); sp->vibrato_sweep_increment= convert_vibrato_sweep(song, tmp[15], sp->vibrato_control_ratio); sp->vibrato_depth=tmp[17]; DEBUG_MSG(" * vibrato: sweep %d, ctl %d, depth %d\n", sp->vibrato_sweep_increment, sp->vibrato_control_ratio, sp->vibrato_depth); } READ_CHAR(sp->modes); vfs_fseek(fp, 40, SEEK_CUR); /* skip the useless scale frequency, scale factor (what's it mean?), and reserved space */ /* Mark this as a fixed-pitch instrument if such a deed is desired. */ if (note_to_use!=-1) sp->note_to_use=(uint8)(note_to_use); else sp->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))) { DEBUG_MSG(" - Removing loop and/or sustain\n"); sp->modes &=~(MODES_SUSTAIN | MODES_LOOPING | MODES_PINGPONG | MODES_REVERSE); } if (strip_envelope==1) { #ifdef DEBUG if (sp->modes & MODES_ENVELOPE) DEBUG_MSG(" - Removing envelope\n"); #endif 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); DEBUG_MSG(" - No loop, removing sustain and envelope\n"); } 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; DEBUG_MSG(" - Weirdness, removing envelope\n"); } 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; DEBUG_MSG(" - No sustain, removing envelope\n"); } } for (j=0; j<6; j++) { sp->envelope_rate[j]= convert_envelope_rate(song, tmp[j]); sp->envelope_offset[j]= convert_envelope_offset(tmp[6+j]); } /* Then read the sample data */ sp->data = safe_malloc(sp->data_length); if (1 != vfs_fread(sp->data, sp->data_length, 1, fp)) goto fail; if (!(sp->modes & MODES_16BIT)) /* convert to 16-bit data */ { sint32 i=sp->data_length; uint8 *cp=(uint8 *)(sp->data); uint16 *tmp,*new; tmp=new=safe_malloc(sp->data_length*2); while (i--) *tmp++ = (uint16)(*cp++) << 8; cp=(uint8 *)(sp->data); sp->data = (sample_t *)new; free(cp); sp->data_length *= 2; sp->loop_start *= 2; sp->loop_end *= 2; } #ifndef LITTLE_ENDIAN else /* convert to machine byte order */ { sint32 i=sp->data_length/2; sint16 *tmp=(sint16 *)sp->data,s; while (i--) { s=SWAPLE16(*tmp); *tmp++=s; } } #endif if (sp->modes & MODES_UNSIGNED) /* convert to signed data */ { sint32 i=sp->data_length/2; sint16 *tmp=(sint16 *)sp->data; while (i--) *tmp++ ^= 0x8000; } /* Reverse reverse loops and pass them off as normal loops */ if (sp->modes & MODES_REVERSE) { sint32 t; /* The GUS apparently plays reverse loops by reversing the whole sample. We do the same because the GUS does not SUCK. */ DEBUG_MSG("Reverse loop in %s\n", name); reverse_data((sint16 *)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 */ } #ifdef ADJUST_SAMPLE_VOLUMES if (amp!=-1) sp->volume=(float)((amp) / 100.0); 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. */ sint32 i=sp->data_length/2; sint16 maxamp=0,a; sint16 *tmp=(sint16 *)sp->data; while (i--) { a=*tmp++; if (a<0) a=-a; if (a>maxamp) maxamp=a; } sp->volume=(float)(32768.0 / maxamp); DEBUG_MSG(" * volume comp: %f\n", sp->volume); } #else if (amp!=-1) sp->volume=(double)(amp) / 100.0; else sp->volume=1.0; #endif sp->data_length /= 2; /* These are in bytes. Convert into samples. */ sp->loop_start /= 2; sp->loop_end /= 2; /* 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(song, sp); if (strip_tail==1) { /* Let's not really, just say we did. */ DEBUG_MSG(" - Stripping tail\n"); sp->data_length = sp->loop_end; } } vfs_fclose(fp); return ip; } static int fill_bank(MidSong *song, int dr, int b) { int i, errors=0; MidToneBank *bank=((dr) ? song->drumset[b] : song->tonebank[b]); if (!bank) { DEBUG_MSG("Huh. Tried to load instruments in non-existent %s %d\n", (dr) ? "drumset" : "tone bank", b); return 0; } for (i=0; i<128; i++) { if (bank->instrument[i]==MAGIC_LOAD_INSTRUMENT) { bank->instrument[i]=load_instrument_dls(song, dr, b, i); if (bank->instrument[i]) { continue; } if (!(bank->tone[i].name)) { DEBUG_MSG("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"); if (b!=0) { /* Mark the corresponding instrument in the default bank / drumset for loading (if it isn't already) */ if (!dr) { if (!(song->tonebank[0]->instrument[i])) song->tonebank[0]->instrument[i] = MAGIC_LOAD_INSTRUMENT; } else { if (!(song->drumset[0]->instrument[i])) song->drumset[0]->instrument[i] = MAGIC_LOAD_INSTRUMENT; } } bank->instrument[i] = 0; errors++; } else if (!(bank->instrument[i] = 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 ))) { DEBUG_MSG("Couldn't load instrument %s (%s %d, program %d)\n", bank->tone[i].name, (dr)? "drum set" : "tone bank", b, i); errors++; } } } return errors; } int load_missing_instruments(MidSong *song) { int i=128,errors=0; while (i--) { if (song->tonebank[i]) errors+=fill_bank(song,0,i); if (song->drumset[i]) errors+=fill_bank(song,1,i); } return errors; } void free_instruments(MidSong *song) { int i=128; while(i--) { if (song->tonebank[i]) free_bank(song, 0, i); if (song->drumset[i]) free_bank(song, 1, i); } } int set_default_instrument(MidSong *song, char *name) { MidInstrument *ip; if (!(ip=load_instrument(song, name, 0, -1, -1, -1, 0, 0, 0))) return -1; song->default_instrument = ip; song->default_program = SPECIAL_PROGRAM; return 0; }