instrum.c

SDL_mixer 是一个基于 SDL 的混音器

/*

    TiMidity -- Experimental MIDI to WAVE converter
    Copyright (C) 1995 Tuukka Toivonen <[e-mail]>

    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(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;