instrum.c

SDL_mixer 是一个基于 SDL 的混音器

      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(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;
}