instrum.c
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上传日期:2022-07-21
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文件大小:25k
- /*
- 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.
- */
- #include <stdio.h>
- #include <string.h>
- #include <stdlib.h>
- #include "config.h"
- #include "common.h"
- #include "instrum.h"
- #include "playmidi.h"
- #include "output.h"
- #include "ctrlmode.h"
- #include "resample.h"
- #include "tables.h"
- #include "filter.h"
- /* Some functions get aggravated if not even the standard banks are
- available. */
- static ToneBank standard_tonebank, standard_drumset;
- ToneBank
- *tonebank[MAXBANK]={&standard_tonebank},
- *drumset[MAXBANK]={&standard_drumset};
- /* This is a special instrument, used for all melodic programs */
- InstrumentLayer *default_instrument=0;
- /* This is only used for tracks that don't specify a program */
- int default_program=DEFAULT_PROGRAM;
- int antialiasing_allowed=0;
- #ifdef FAST_DECAY
- int fast_decay=1;
- #else
- int fast_decay=0;
- #endif
- int current_tune_number = 0;
- int last_tune_purged = 0;
- int current_patch_memory = 0;
- int max_patch_memory = 60000000;
- static void purge_as_required(void);
- static void free_instrument(Instrument *ip)
- {
- Sample *sp;
- int i;
- if (!ip) return;
- if (!ip->contents)
- for (i=0; i<ip->samples; i++)
- {
- sp=&(ip->sample[i]);
- if (sp->data) free(sp->data);
- }
- free(ip->sample);
- if (!ip->contents)
- for (i=0; i<ip->right_samples; i++)
- {
- sp=&(ip->right_sample[i]);
- if (sp->data) free(sp->data);
- }
- if (ip->right_sample)
- free(ip->right_sample);
- free(ip);
- }
- static void free_layer(InstrumentLayer *lp)
- {
- InstrumentLayer *next;
- current_patch_memory -= lp->size;
- for (; lp; lp = next)
- {
- next = lp->next;
- free_instrument(lp->instrument);
- free(lp);
- }
- }
- static void free_bank(int dr, int b)
- {
- int i;
- ToneBank *bank=((dr) ? drumset[b] : tonebank[b]);
- for (i=0; i<MAXPROG; i++)
- {
- if (bank->tone[i].layer)
- {
- /* Not that this could ever happen, of course */
- if (bank->tone[i].layer != MAGIC_LOAD_INSTRUMENT)
- {
- free_layer(bank->tone[i].layer);
- bank->tone[i].layer=NULL;
- bank->tone[i].last_used=-1;
- }
- }
- if (bank->tone[i].name)
- {
- free(bank->tone[i].name);
- bank->tone[i].name = NULL;
- }
- }
- }
- static void free_old_bank(int dr, int b, int how_old)
- {
- int i;
- ToneBank *bank=((dr) ? drumset[b] : tonebank[b]);
- for (i=0; i<MAXPROG; i++)
- if (bank->tone[i].layer && bank->tone[i].last_used < how_old)
- {
- if (bank->tone[i].layer != MAGIC_LOAD_INSTRUMENT)
- {
- ctl->cmsg(CMSG_INFO, VERB_DEBUG,
- "Unloading %s %s[%d,%d] - last used %d.",
- (dr)? "drum" : "inst", bank->tone[i].name,
- i, b, bank->tone[i].last_used);
- free_layer(bank->tone[i].layer);
- bank->tone[i].layer=NULL;
- bank->tone[i].last_used=-1;
- }
- }
- }
- int32 convert_envelope_rate_attack(uint8 rate, uint8 fastness)
- {
- int32 r;
- r=3-((rate>>6) & 0x3);
- r*=3;
- r = (int32)(rate & 0x3f) << r; /* 6.9 fixed point */
- /* 15.15 fixed point. */
- return (((r * 44100) / play_mode->rate) * control_ratio)
- << 10;
- }
- int32 convert_envelope_rate(uint8 rate)
- {
- int32 r;
- r=3-((rate>>6) & 0x3);
- r*=3;
- r = (int32)(rate & 0x3f) << r; /* 6.9 fixed point */
- /* 15.15 fixed point. */
- return (((r * 44100) / play_mode->rate) * control_ratio)
- << ((fast_decay) ? 10 : 9);
- }
- int32 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);
- }
- int32 convert_tremolo_sweep(uint8 sweep)
- {
- if (!sweep)
- return 0;
- return
- ((control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) /
- (play_mode->rate * sweep);
- }
- int32 convert_vibrato_sweep(uint8 sweep, int32 vib_control_ratio)
- {
- if (!sweep)
- return 0;
- return
- (int32) (FSCALE((double) (vib_control_ratio) * SWEEP_TUNING, SWEEP_SHIFT)
- / (double)(play_mode->rate * sweep));
- /* this was overflowing with seashore.pat
- ((vib_control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) /
- (play_mode->rate * sweep); */
- }
- int32 convert_tremolo_rate(uint8 rate)
- {
- return
- ((SINE_CYCLE_LENGTH * control_ratio * rate) << RATE_SHIFT) /
- (TREMOLO_RATE_TUNING * play_mode->rate);
- }
- int32 convert_vibrato_rate(uint8 rate)
- {
- /* Return a suitable vibrato_control_ratio value */
- return
- (VIBRATO_RATE_TUNING * play_mode->rate) /
- (rate * 2 * VIBRATO_SAMPLE_INCREMENTS);
- }
- static void reverse_data(int16 *sp, int32 ls, int32 le)
- {
- int16 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 InstrumentLayer *load_instrument(const char *name, int font_type, int percussion,
- int panning, int amp, int cfg_tuning, int note_to_use,
- int strip_loop, int strip_envelope,
- int strip_tail, int bank, int gm_num, int sf_ix)
- {
- InstrumentLayer *lp, *lastlp, *headlp = 0;
- Instrument *ip;
- FILE *fp;
- uint8 tmp[1024];
- int i,j,noluck=0;
- #ifdef PATCH_EXT_LIST
- static char *patch_ext[] = PATCH_EXT_LIST;
- #endif
- int sf2flag = 0;
- int right_samples = 0;
- int stereo_channels = 1, stereo_layer;
- int vlayer_list[19][4], vlayer, vlayer_count = 0;
- if (!name) return 0;
-
- /* Open patch file */
- if ((fp=open_file(name, 1, OF_NORMAL)) == NULL)
- {
- noluck=1;
- #ifdef PATCH_EXT_LIST
- /* Try with various extensions */
- for (i=0; patch_ext[i]; i++)
- {
- if (strlen(name)+strlen(patch_ext[i])<PATH_MAX)
- {
- char path[PATH_MAX];
- strcpy(path, name);
- strcat(path, patch_ext[i]);
- if ((fp=open_file(path, 1, OF_NORMAL)) != NULL)
- {
- noluck=0;
- break;
- }
- }
- }
- #endif
- }
-
- if (noluck)
- {
- ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
- "Instrument `%s' can't be found.", name);
- return 0;
- }
-
- /*ctl->cmsg(CMSG_INFO, VERB_NOISY, "Loading instrument %s", current_filename);*/
-
- /* Read some headers and do cursory sanity checks. There are loads
- of magic offsets. This could be rewritten... */
- if ((239 != fread(tmp, 1, 239, fp)) ||
- (memcmp(tmp, "GF1PATCH110 ID#000002", 22) &&
- memcmp(tmp, "GF1PATCH100 ID#000002", 22))) /* don't know what the
- differences are */
- {
- ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "%s: not an instrument", name);
- return 0;
- }
- /* patch layout:
- * bytes: info: starts at offset:
- * 22 id (see above) 0
- * 60 copyright 22
- * 1 instruments 82
- * 1 voices 83
- * 1 channels 84
- * 2 number of waveforms 85
- * 2 master volume 87
- * 4 datasize 89
- * 36 reserved, but now: 93
- * 7 "SF2EXT " id 93
- * 1 right samples 100
- * 28 reserved 101
- * 2 instrument number 129
- * 16 instrument name 131
- * 4 instrument size 147
- * 1 number of layers 151
- * 40 reserved 152
- * 1 layer duplicate 192
- * 1 layer number 193
- * 4 layer size 194
- * 1 number of samples 198
- * 40 reserved 199
- * 239
- * THEN, for each sample, see below
- */
- if (!memcmp(tmp + 93, "SF2EXT", 6))
- {
- sf2flag = 1;
- vlayer_count = tmp[152];
- }
- if (tmp[82] != 1 && tmp[82] != 0) /* instruments. To some patch makers,
- 0 means 1 */
- {
- ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
- "Can't handle patches with %d instruments", tmp[82]);
- 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]);
- return 0;
- }
-
- if (sf2flag && vlayer_count > 0) {
- for (i = 0; i < 9; i++)
- for (j = 0; j < 4; j++)
- vlayer_list[i][j] = tmp[153+i*4+j];
- for (i = 9; i < 19; i++)
- for (j = 0; j < 4; j++)
- vlayer_list[i][j] = tmp[199+(i-9)*4+j];
- }
- else {
- for (i = 0; i < 19; i++)
- for (j = 0; j < 4; j++)
- vlayer_list[i][j] = 0;
- vlayer_list[0][0] = 0;
- vlayer_list[0][1] = 127;
- vlayer_list[0][2] = tmp[198];
- vlayer_list[0][3] = 0;
- vlayer_count = 1;
- }
- lastlp = 0;
- for (vlayer = 0; vlayer < vlayer_count; vlayer++) {
- lp=(InstrumentLayer *)safe_malloc(sizeof(InstrumentLayer));
- lp->size = sizeof(InstrumentLayer);
- lp->lo = vlayer_list[vlayer][0];
- lp->hi = vlayer_list[vlayer][1];
- ip=(Instrument *)safe_malloc(sizeof(Instrument));
- lp->size += sizeof(Instrument);
- lp->instrument = ip;
- lp->next = 0;
- if (lastlp) lastlp->next = lp;
- else headlp = lp;
- lastlp = lp;
- if (sf2flag) ip->type = INST_SF2;
- else ip->type = INST_GUS;
- ip->samples = vlayer_list[vlayer][2];
- ip->sample = (Sample *)safe_malloc(sizeof(Sample) * ip->samples);
- lp->size += sizeof(Sample) * ip->samples;
- ip->left_samples = ip->samples;
- ip->left_sample = ip->sample;
- right_samples = vlayer_list[vlayer][3];
- ip->right_samples = right_samples;
- if (right_samples)
- {
- ip->right_sample = (Sample *)safe_malloc(sizeof(Sample) * right_samples);
- lp->size += sizeof(Sample) * right_samples;
- stereo_channels = 2;
- }
- else ip->right_sample = 0;
- ip->contents = 0;
- ctl->cmsg(CMSG_INFO, VERB_NOISY, "%s%s[%d,%d] %s(%d-%d layer %d of %d)",
- (percussion)? " ":"", name,
- (percussion)? note_to_use : gm_num, bank,
- (right_samples)? "(2) " : "",
- lp->lo, lp->hi, vlayer+1, vlayer_count);
- for (stereo_layer = 0; stereo_layer < stereo_channels; stereo_layer++)
- {
- int sample_count = 0;
- if (stereo_layer == 0) sample_count = ip->left_samples;
- else if (stereo_layer == 1) sample_count = ip->right_samples;
- for (i=0; i < sample_count; i++)
- {
- uint8 fractions;
- int32 tmplong;
- uint16 tmpshort;
- uint16 sample_volume = 0;
- uint8 tmpchar;
- Sample *sp = 0;
- uint8 sf2delay = 0;
- #define READ_CHAR(thing)
- if (1 != fread(&tmpchar, 1, 1, fp)) goto fail;
- thing = tmpchar;
- #define READ_SHORT(thing)
- if (1 != fread(&tmpshort, 2, 1, fp)) goto fail;
- thing = LE_SHORT(tmpshort);
- #define READ_LONG(thing)
- if (1 != fread(&tmplong, 4, 1, fp)) goto fail;
- thing = LE_LONG(tmplong);
- /*
- * 7 sample name
- * 1 fractions
- * 4 length
- * 4 loop start
- * 4 loop end
- * 2 sample rate
- * 4 low frequency
- * 4 high frequency
- * 2 finetune
- * 1 panning
- * 6 envelope rates |
- * 6 envelope offsets | 18 bytes
- * 3 tremolo sweep, rate, depth |
- * 3 vibrato sweep, rate, depth |
- * 1 sample mode
- * 2 scale frequency
- * 2 scale factor
- * 2 sample volume (??)
- * 34 reserved
- * Now: 1 delay
- * 33 reserved
- */
- skip(fp, 7); /* Skip the wave name */
- if (1 != fread(&fractions, 1, 1, fp))
- {
- fail:
- ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "Error reading sample %d", i);
- if (stereo_layer == 1)
- {
- for (j=0; j<i; j++)
- free(ip->right_sample[j].data);
- free(ip->right_sample);
- i = ip->left_samples;
- }
- for (j=0; j<i; j++)
- free(ip->left_sample[j].data);
- free(ip->left_sample);
- free(ip);
- free(lp);
- return 0;
- }
- if (stereo_layer == 0) sp=&(ip->left_sample[i]);
- else if (stereo_layer == 1) sp=&(ip->right_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);
- skip(fp, 2); /* 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);
- sp->resonance=0;
- sp->cutoff_freq=0;
- sp->reverberation=0;
- sp->chorusdepth=0;
- sp->exclusiveClass=0;
- sp->keyToModEnvHold=0;
- sp->keyToModEnvDecay=0;
- sp->keyToVolEnvHold=0;
- sp->keyToVolEnvDecay=0;
- if (cfg_tuning)
- {
- double tune_factor = (double)(cfg_tuning)/1200.0;
- tune_factor = pow(2.0, tune_factor);
- sp->root_freq = (uint32)( tune_factor * (double)sp->root_freq );
- }
- /* envelope, tremolo, and vibrato */
- if (18 != fread(tmp, 1, 18, fp)) goto fail;
- if (!tmp[13] || !tmp[14])
- {
- sp->tremolo_sweep_increment=
- sp->tremolo_phase_increment=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=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);
- READ_SHORT(sp->freq_center);
- READ_SHORT(sp->freq_scale);
- if (sf2flag)
- {
- READ_SHORT(sample_volume);
- READ_CHAR(sf2delay);
- READ_CHAR(sp->exclusiveClass);
- skip(fp, 32);
- }
- else
- {
- skip(fp, 36);
- }
- /* 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)))
- {
- ctl->cmsg(CMSG_INFO, VERB_DEBUG, " - Removing loop and/or sustain");
- sp->modes &=~(MODES_SUSTAIN | MODES_LOOPING |
- MODES_PINGPONG | MODES_REVERSE);
- }
- 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");
- }
- }
- sp->attenuation = 0;
- for (j=ATTACK; j<DELAY; j++)
- {
- sp->envelope_rate[j]=
- (j<3)? convert_envelope_rate_attack(tmp[j], 11) : convert_envelope_rate(tmp[j]);
- sp->envelope_offset[j]=
- convert_envelope_offset(tmp[6+j]);
- }
- if (sf2flag)
- {
- if (sf2delay > 5) sf2delay = 5;
- sp->envelope_rate[DELAY] = (int32)( (sf2delay*play_mode->rate) / 1000 );
- }
- else
- {
- sp->envelope_rate[DELAY]=0;
- }
- sp->envelope_offset[DELAY]=0;
- for (j=ATTACK; j<DELAY; j++)
- {
- sp->modulation_rate[j]=sp->envelope_rate[j];
- sp->modulation_offset[j]=sp->envelope_offset[j];
- }
- sp->modulation_rate[DELAY] = sp->modulation_offset[DELAY] = 0;
- sp->modEnvToFilterFc=0;
- sp->modEnvToPitch=0;
- sp->lfo_sweep_increment = 0;
- sp->lfo_phase_increment = 0;
- sp->modLfoToFilterFc = 0;
- sp->vibrato_delay = 0;
- /* Then read the sample data */
- if (sp->data_length/2 > MAX_SAMPLE_SIZE)
- {
- goto fail;
- }
- sp->data = safe_malloc(sp->data_length + 1);
- lp->size += sp->data_length + 1;
- if (1 != fread(sp->data, sp->data_length, 1, fp))
- goto fail;
-
- if (!(sp->modes & MODES_16BIT)) /* convert to 16-bit data */
- {
- int32 i=sp->data_length;
- uint8 *cp=(uint8 *)(sp->data);
- uint16 *tmp,*newdta;
- tmp=newdta=safe_malloc(sp->data_length*2 + 2);
- while (i--)
- *tmp++ = (uint16)(*cp++) << 8;
- cp=(uint8 *)(sp->data);
- sp->data = (sample_t *)newdta;
- free(cp);
- sp->data_length *= 2;
- sp->loop_start *= 2;
- sp->loop_end *= 2;
- }
- #if SDL_BYTEORDER == SDL_BIG_ENDIAN
- else
- /* convert to machine byte order */
- {
- int32 i=sp->data_length/2;
- int16 *tmp=(int16 *)sp->data,s;
- while (i--)
- {
- s=LE_SHORT(*tmp);
- *tmp++=s;
- }
- }
- #endif
-
- if (sp->modes & MODES_UNSIGNED) /* convert to signed data */
- {
- int32 i=sp->data_length/2;
- int16 *tmp=(int16 *)sp->data;
- while (i--)
- *tmp++ ^= 0x8000;
- }
- /* Reverse reverse loops and pass them off as normal loops */
- if (sp->modes & MODES_REVERSE)
- {
- int32 t;
- /* The GUS apparently plays reverse loops by reversing the
- whole sample. We do the same because the GUS does not SUCK. */
- ctl->cmsg(CMSG_WARNING, VERB_NORMAL, "Reverse loop in %s", name);
- reverse_data((int16 *)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 */
- }
- /* If necessary do some anti-aliasing filtering */
- if (antialiasing_allowed)
- antialiasing(sp,play_mode->rate);
- #ifdef ADJUST_SAMPLE_VOLUMES
- if (amp!=-1)
- sp->volume=(FLOAT_T)((amp) / 100.0);
- else if (sf2flag)
- sp->volume=(FLOAT_T)((sample_volume) / 255.0);
- 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. */
- uint32 i, numsamps=sp->data_length/2;
- uint32 higher=0, highcount=0;
- int16 maxamp=0,a;
- int16 *tmp=(int16 *)sp->data;
- i = numsamps;
- while (i--)
- {
- a=*tmp++;
- if (a<0) a=-a;
- if (a>maxamp)
- maxamp=a;
- }
- tmp=(int16 *)sp->data;
- i = numsamps;
- while (i--)
- {
- a=*tmp++;
- if (a<0) a=-a;
- if (a > 3*maxamp/4)
- {
- higher += a;
- highcount++;
- }
- }
- if (highcount) higher /= highcount;
- else higher = 10000;
- sp->volume = (32768.0 * 0.875) / (double)higher ;
- ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * volume comp: %f", 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;
- sp->data[sp->data_length] = sp->data[sp->data_length-1];
- /* Then fractional samples */
- sp->data_length <<= FRACTION_BITS;
- sp->loop_start <<= FRACTION_BITS;
- sp->loop_end <<= FRACTION_BITS;
- /* trim off zero data at end */
- {
- int ls = sp->loop_start>>FRACTION_BITS;
- int le = sp->loop_end>>FRACTION_BITS;
- int se = sp->data_length>>FRACTION_BITS;
- while (se > 1 && !sp->data[se-1]) se--;
- if (le > se) le = se;
- if (ls >= le) sp->modes &= ~MODES_LOOPING;
- sp->loop_end = le<<FRACTION_BITS;
- sp->data_length = se<<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);
- #ifdef LOOKUP_HACK
- /* Squash the 16-bit data into 8 bits. */
- {
- uint8 *gulp,*ulp;
- int16 *swp;
- int l=sp->data_length >> FRACTION_BITS;
- gulp=ulp=safe_malloc(l+1);
- swp=(int16 *)sp->data;
- while(l--)
- *ulp++ = (*swp++ >> 8) & 0xFF;
- free(sp->data);
- sp->data=(sample_t *)gulp;
- }
- #endif
-
- if (strip_tail==1)
- {
- /* Let's not really, just say we did. */
- ctl->cmsg(CMSG_INFO, VERB_DEBUG, " - Stripping tail");
- sp->data_length = sp->loop_end;
- }
- } /* end of sample loop */
- } /* end of stereo layer loop */
- } /* end of vlayer loop */
- close_file(fp);
- return headlp;
- }
- static int fill_bank(int dr, int b)
- {
- int i, errors=0;
- ToneBank *bank=((dr) ? drumset[b] : tonebank[b]);
- if (!bank)
- {
- ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
- "Huh. Tried to load instruments in non-existent %s %d",
- (dr) ? "drumset" : "tone bank", b);
- return 0;
- }
- for (i=0; i<MAXPROG; i++)
- {
- if (bank->tone[i].layer==MAGIC_LOAD_INSTRUMENT)
- {
- if (!(bank->tone[i].name))
- {
- ctl->cmsg(CMSG_WARNING, (b!=0) ? VERB_VERBOSE : VERB_NORMAL,
- "No instrument mapped to %s %d, program %d%s",
- (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 (!(standard_tonebank.tone[i].layer))
- standard_tonebank.tone[i].layer=
- MAGIC_LOAD_INSTRUMENT;
- }
- else
- {
- if (!(standard_drumset.tone[i].layer))
- standard_drumset.tone[i].layer=
- MAGIC_LOAD_INSTRUMENT;
- }
- }
- bank->tone[i].layer=0;
- errors++;
- }
- else if (!(bank->tone[i].layer=
- load_instrument(bank->tone[i].name,
- bank->tone[i].font_type,
- (dr) ? 1 : 0,
- bank->tone[i].pan,
- bank->tone[i].amp,
- bank->tone[i].tuning,
- (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,
- b,
- ((dr) ? i + 128 : i),
- bank->tone[i].sf_ix
- )))
- {
- ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
- "Couldn't load instrument %s (%s %d, program %d)",
- bank->tone[i].name,
- (dr)? "drum set" : "tone bank", b, i);
- errors++;
- }
- else
- { /* it's loaded now */
- bank->tone[i].last_used = current_tune_number;
- current_patch_memory += bank->tone[i].layer->size;
- purge_as_required();
- if (current_patch_memory > max_patch_memory) {
- ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
- "Not enough memory to load instrument %s (%s %d, program %d)",
- bank->tone[i].name,
- (dr)? "drum set" : "tone bank", b, i);
- errors++;
- free_layer(bank->tone[i].layer);
- bank->tone[i].layer=0;
- bank->tone[i].last_used=-1;
- }
- #if 0
- if (check_for_rc()) {
- free_layer(bank->tone[i].layer);
- bank->tone[i].layer=0;
- bank->tone[i].last_used=-1;
- return 0;
- }
- #endif
- }
- }
- }
- return errors;
- }
- static void free_old_instruments(int how_old)
- {
- int i=MAXBANK;
- while(i--)
- {
- if (tonebank[i])
- free_old_bank(0, i, how_old);
- if (drumset[i])
- free_old_bank(1, i, how_old);
- }
- }
- static void purge_as_required(void)
- {
- if (!max_patch_memory) return;
- while (last_tune_purged < current_tune_number
- && current_patch_memory > max_patch_memory)
- {
- last_tune_purged++;
- free_old_instruments(last_tune_purged);
- }
- }
- int load_missing_instruments(void)
- {
- int i=MAXBANK,errors=0;
- while (i--)
- {
- if (tonebank[i])
- errors+=fill_bank(0,i);
- if (drumset[i])
- errors+=fill_bank(1,i);
- }
- current_tune_number++;
- return errors;
- }
- void free_instruments(void)
- {
- int i=128;
- while(i--)
- {
- if (tonebank[i])
- free_bank(0,i);
- if (drumset[i])
- free_bank(1,i);
- }
- }
- int set_default_instrument(const char *name)
- {
- InstrumentLayer *lp;
- /* if (!(lp=load_instrument(name, 0, -1, -1, -1, 0, 0, 0))) */
- if (!(lp=load_instrument(name, FONT_NORMAL, 0, -1, -1, 0, -1, -1, -1, -1, 0, -1, -1)))
- return -1;
- if (default_instrument)
- free_layer(default_instrument);
- default_instrument=lp;
- default_program=SPECIAL_PROGRAM;
- return 0;
- }