/*
    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 Perl Artistic License, available in COPYING.
 */

#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\0ID#000002", 22) &&
       memcmp(tmp, "GF1PATCH100\0ID#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\0" 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;
}