Esempio n. 1
0
/* 
   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(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;
  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;

  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])<1024)
	    {
              char path[1024];
	      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;

  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;
      uint8 tmpchar;
      Sample *sp;
      uint8 sf2delay;

#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;
	}
#ifndef LITTLE_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;
}
Esempio n. 2
0
/*
	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 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 = pathExpander.openFileReader(name, NULL)) == NULL)
    {
        /* Try with various extensions */
        FString tmp = name;
        tmp += ".pat";
        if ((fp = pathExpander.openFileReader(tmp, NULL)) == NULL)
        {
#ifdef __unix__			// Windows isn't case-sensitive.
            tmp.ToUpper();
            if ((fp = pathExpander.openFileReader(tmp, 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);
            }
        }

        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] = clamp<BYTE>(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 */
        }

        /* 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;
}
/* 
 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(char *name, int percussion,
                                   int panning, int amp, int note_to_use,
                                   int strip_loop, int strip_envelope,
                                   int strip_tail)
{
	ignore_unused_variable_warning(percussion);
	Instrument *ip;
	Sample *sp;
	FILE *fp;
	uint8 tmp[1024];
	int i,j,noluck=0;
#ifdef PATCH_EXT_LIST
	static const char *patch_ext[] = PATCH_EXT_LIST;
#endif

	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])<1024)
			{
				char path[1024];
				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;
	}

	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;
	}

	ip=safe_Malloc<Instrument>();
	ip->samples = tmp[198];
	ip->sample = safe_Malloc<Sample>(ip->samples);
	for (i=0; i<ip->samples; i++)
	{

		uint8 fractions;
		sint32 tmplong;
		uint16 tmpshort;
		uint8 tmpchar;

#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);

		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);
			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);
		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=static_cast<uint8>(panning & 0x7F);

		/* 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);

		skip(fp, 40); /* 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=static_cast<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");
			}
		}

		for (j=0; j<6; j++)
		{
			sp->envelope_rate[j]=
				convert_envelope_rate(tmp[j]);
			sp->envelope_offset[j]= 
				convert_envelope_offset(tmp[6+j]);
		}

		/* Then read the sample data */
		sp->data = safe_Malloc<sample_t>(sp->data_length);
		if (1 != 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=reinterpret_cast<uint8 *>(sp->data);
			uint16 *tmp,*new_dat;
			tmp=new_dat=safe_Malloc<uint16>(sp->data_length);
			while (i--)
				*tmp++ = static_cast<uint16>(*cp++) << 8;
			cp=reinterpret_cast<uint8 *>(sp->data);
			sp->data = reinterpret_cast<sample_t *>(new_dat);
			free(cp);
			sp->data_length *= 2;
			sp->loop_start *= 2;
			sp->loop_end *= 2;
		}
#ifndef TIMIDITY_LITTLE_ENDIAN
		else
			/* convert to machine byte order */
		{
			sint32 i=sp->data_length/2;
			sint16 *tmp=reinterpret_cast<sint16 *>(sp->data),s;
			while (i--)
			{ 
				s=LE_SHORT(*tmp);
				*tmp++=s;
			}
		}
#endif

		if (sp->modes & MODES_UNSIGNED) /* convert to signed data */
		{
			sint32 i=sp->data_length/2;
			sint16 *tmp = 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. */

			ctl->cmsg(CMSG_WARNING, VERB_NORMAL, "Reverse loop in %s", name);
			reverse_data(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=static_cast<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 = sp->data;
			while (i--)
			{
				a=*tmp++;
				if (a<0) a=-a;
				if (a>maxamp)
					maxamp=a;
			}
			sp->volume=static_cast<float>(32768.0 / maxamp);
			ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * volume comp: %f", sp->volume);
		}
#else
		if (amp!=-1)
			sp->volume=static_cast<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(sp);

#ifdef LOOKUP_HACK
		/* Squash the 16-bit data into 8 bits. */
		{
			uint8 *gulp,*ulp;
			sint16 *swp;
			int l=sp->data_length >> FRACTION_BITS;
			gulp=ulp=safe_Malloc<uint8>(l+1);
			swp=(sint16 *)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;
		}
	}

	close_file(fp);
	return ip;
}