diff src/Input/timidity/libtimidity/instrum.c @ 0:13389e613d67 trunk

[svn] - initial import of audacious-plugins tree (lots to do)

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/src/Input/timidity/libtimidity/instrum.c	Mon Sep 18 01:11:49 2006 -0700
@@ -0,0 +1,621 @@
+/*
+
+    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.
+
+*/
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include "libaudacious/vfs.h"
+#include <string.h>
+#include <stdlib.h>
+
+#include "timidity.h"
+#include "timidity_internal.h"
+#include "options.h"
+#include "common.h"
+#include "instrum.h"
+#include "instrum_dls.h"
+#include "resample.h"
+#include "tables.h"
+
+static void free_instrument(MidInstrument *ip)
+{
+  MidSample *sp;
+  int i;
+  if (!ip) return;
+  for (i=0; i<ip->samples; i++)
+    {
+      sp=&(ip->sample[i]);
+      free(sp->data);
+    }
+  free(ip->sample);
+  free(ip);
+}
+
+static void free_bank(MidSong *song, int dr, int b)
+{
+  int i;
+  MidToneBank *bank=((dr) ? song->drumset[b] : song->tonebank[b]);
+  for (i=0; i<128; i++)
+    if (bank->instrument[i])
+      {
+	/* Not that this could ever happen, of course */
+	if (bank->instrument[i] != MAGIC_LOAD_INSTRUMENT)
+	  free_instrument(bank->instrument[i]);
+	bank->instrument[i]=0;
+      }
+}
+
+static sint32 convert_envelope_rate(MidSong *song, uint8 rate)
+{
+  sint32 r;
+  
+  r = 3 - ((rate >> 6) & 0x3);
+  r *= 3;
+  r = (sint32) (rate & 0x3f) << r; /* 6.9 fixed point */
+
+  /* 15.15 fixed point. */
+  r = ((r * 44100) / song->rate) * song->control_ratio;
+
+#ifdef FAST_DECAY
+  return r << 10;
+#else
+  return r << 9;
+#endif
+}
+
+static sint32 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);
+}
+
+static sint32 convert_tremolo_sweep(MidSong *song, uint8 sweep)
+{
+  if (!sweep)
+    return 0;
+
+  return
+    ((song->control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) /
+      (song->rate * sweep);
+}
+
+static sint32 convert_vibrato_sweep(MidSong *song, uint8 sweep,
+				    sint32 vib_control_ratio)
+{
+  if (!sweep)
+    return 0;
+
+  return
+    (sint32) (FSCALE((double) (vib_control_ratio) * SWEEP_TUNING, SWEEP_SHIFT)
+	     / (double)(song->rate * sweep));
+
+  /* this was overflowing with seashore.pat
+
+      ((vib_control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) /
+      (song->rate * sweep); */
+}
+
+static sint32 convert_tremolo_rate(MidSong *song, uint8 rate)
+{
+  return
+    ((SINE_CYCLE_LENGTH * song->control_ratio * rate) << RATE_SHIFT) /
+      (TREMOLO_RATE_TUNING * song->rate);
+}
+
+static sint32 convert_vibrato_rate(MidSong *song, uint8 rate)
+{
+  /* Return a suitable vibrato_control_ratio value */
+  return
+    (VIBRATO_RATE_TUNING * song->rate) / 
+      (rate * 2 * MID_VIBRATO_SAMPLE_INCREMENTS);
+}
+
+static void reverse_data(sint16 *sp, sint32 ls, sint32 le)
+{
+  sint16 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 MidInstrument *load_instrument(MidSong *song, char *name, int percussion,
+				   int panning, int amp, int note_to_use,
+				   int strip_loop, int strip_envelope,
+				   int strip_tail)
+{
+  MidInstrument *ip;
+  MidSample *sp;
+  VFSFile *fp;
+  char tmp[1024];
+  int i,j,noluck=0;
+  static char *patch_ext[] = PATCH_EXT_LIST;
+
+  if (!name) return 0;
+  
+  /* Open patch file */
+  if ((fp=open_file(name)) == NULL)
+    {
+      noluck=1;
+      /* Try with various extensions */
+      for (i=0; patch_ext[i]; i++)
+	{
+	  if (strlen(name)+strlen(patch_ext[i])<1024)
+	    {
+	      strcpy(tmp, name);
+	      strcat(tmp, patch_ext[i]);
+	      if ((fp=open_file(tmp)) != NULL)
+		{
+		  noluck=0;
+		  break;
+		}
+	    }
+	}
+    }
+  
+  if (noluck)
+    {
+      DEBUG_MSG("Instrument `%s' can't be found.\n", name);
+      return 0;
+    }
+      
+  DEBUG_MSG("Loading instrument %s\n", tmp);
+  
+  /* Read some headers and do cursory sanity checks. There are loads
+     of magic offsets. This could be rewritten... */
+
+  if ((239 != vfs_fread(tmp, 1, 239, fp)) ||
+      (memcmp(tmp, "GF1PATCH110\0ID#000002", 22) &&
+       memcmp(tmp, "GF1PATCH100\0ID#000002", 22))) /* don't know what the
+						      differences are */
+    {
+      DEBUG_MSG("%s: not an instrument\n", name);
+      return 0;
+    }
+  
+  if (tmp[82] != 1 && tmp[82] != 0) /* instruments. To some patch makers, 
+				       0 means 1 */
+    {
+      DEBUG_MSG("Can't handle patches with %d instruments\n", tmp[82]);
+      return 0;
+    }
+
+  if (tmp[151] != 1 && tmp[151] != 0) /* layers. What's a layer? */
+    {
+      DEBUG_MSG("Can't handle instruments with %d layers\n", tmp[151]);
+      return 0;
+    }
+  
+  ip=safe_malloc(sizeof(MidInstrument));
+  ip->samples = tmp[198];
+  ip->sample = safe_malloc(sizeof(MidSample) * ip->samples);
+  for (i=0; i<ip->samples; i++)
+    {
+
+      uint8 fractions;
+      sint32 tmplong;
+      uint16 tmpshort;
+      uint8 tmpchar;
+
+#define READ_CHAR(thing) \
+      if (1 != vfs_fread(&tmpchar, 1, 1, fp)) goto fail; \
+      thing = tmpchar;
+#define READ_SHORT(thing) \
+      if (1 != vfs_fread(&tmpshort, 2, 1, fp)) goto fail; \
+      thing = SWAPLE16(tmpshort);
+#define READ_LONG(thing) \
+      if (1 != vfs_fread(&tmplong, 4, 1, fp)) goto fail; \
+      thing = SWAPLE32(tmplong);
+
+      vfs_fseek(fp, 7, SEEK_CUR); /* Skip the wave name */
+
+      if (1 != vfs_fread(&fractions, 1, 1, fp))
+	{
+	fail:
+	  DEBUG_MSG("Error reading sample %d\n", 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);
+      sp->low_vel = 0;
+      sp->high_vel = 127;
+      vfs_fseek(fp, 2, SEEK_CUR); /* 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);
+
+      /* envelope, tremolo, and vibrato */
+      if (18 != vfs_fread(tmp, 1, 18, fp)) goto fail; 
+
+      if (!tmp[13] || !tmp[14])
+	{
+	  sp->tremolo_sweep_increment=
+	    sp->tremolo_phase_increment=sp->tremolo_depth=0;
+	  DEBUG_MSG(" * no tremolo\n");
+	}
+      else
+	{
+	  sp->tremolo_sweep_increment=convert_tremolo_sweep(song, tmp[12]);
+	  sp->tremolo_phase_increment=convert_tremolo_rate(song, tmp[13]);
+	  sp->tremolo_depth=tmp[14];
+	  DEBUG_MSG(" * tremolo: sweep %d, phase %d, depth %d\n",
+	       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;
+	  DEBUG_MSG(" * no vibrato\n");
+	}
+      else
+	{
+	  sp->vibrato_control_ratio=convert_vibrato_rate(song, tmp[16]);
+	  sp->vibrato_sweep_increment=
+	    convert_vibrato_sweep(song, tmp[15], sp->vibrato_control_ratio);
+	  sp->vibrato_depth=tmp[17];
+	  DEBUG_MSG(" * vibrato: sweep %d, ctl %d, depth %d\n",
+	       sp->vibrato_sweep_increment, sp->vibrato_control_ratio,
+	       sp->vibrato_depth);
+
+	}
+
+      READ_CHAR(sp->modes);
+
+      vfs_fseek(fp, 40, SEEK_CUR); /* 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=(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)))
+	{
+	  DEBUG_MSG(" - Removing loop and/or sustain\n");
+	  sp->modes &=~(MODES_SUSTAIN | MODES_LOOPING | 
+			MODES_PINGPONG | MODES_REVERSE);
+	}
+
+      if (strip_envelope==1)
+	{
+#ifdef DEBUG
+	  if (sp->modes & MODES_ENVELOPE)
+	    DEBUG_MSG(" - Removing envelope\n");
+#endif
+	  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);
+	      DEBUG_MSG(" - No loop, removing sustain and envelope\n");
+	    }
+	  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;
+	      DEBUG_MSG(" - Weirdness, removing envelope\n");
+	    }
+	  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;
+	      DEBUG_MSG(" - No sustain, removing envelope\n");
+	    }
+	}
+
+      for (j=0; j<6; j++)
+	{
+	  sp->envelope_rate[j]=
+	    convert_envelope_rate(song, tmp[j]);
+	  sp->envelope_offset[j]= 
+	    convert_envelope_offset(tmp[6+j]);
+	}
+
+      /* Then read the sample data */
+      sp->data = safe_malloc(sp->data_length);
+      if (1 != vfs_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=(uint8 *)(sp->data);
+	  uint16 *tmp,*new;
+	  tmp=new=safe_malloc(sp->data_length*2);
+	  while (i--)
+	    *tmp++ = (uint16)(*cp++) << 8;
+	  cp=(uint8 *)(sp->data);
+	  sp->data = (sample_t *)new;
+	  free(cp);
+	  sp->data_length *= 2;
+	  sp->loop_start *= 2;
+	  sp->loop_end *= 2;
+	}
+#ifndef LITTLE_ENDIAN
+      else
+	/* convert to machine byte order */
+	{
+	  sint32 i=sp->data_length/2;
+	  sint16 *tmp=(sint16 *)sp->data,s;
+	  while (i--)
+	    { 
+	      s=SWAPLE16(*tmp);
+	      *tmp++=s;
+	    }
+	}
+#endif
+      
+      if (sp->modes & MODES_UNSIGNED) /* convert to signed data */
+	{
+	  sint32 i=sp->data_length/2;
+	  sint16 *tmp=(sint16 *)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. */
+
+	  DEBUG_MSG("Reverse loop in %s\n", name);
+	  reverse_data((sint16 *)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 */
+	}
+
+#ifdef ADJUST_SAMPLE_VOLUMES
+      if (amp!=-1)
+	sp->volume=(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=(sint16 *)sp->data;
+	  while (i--)
+	    {
+	      a=*tmp++;
+	      if (a<0) a=-a;
+	      if (a>maxamp)
+		maxamp=a;
+	    }
+	  sp->volume=(float)(32768.0 / maxamp);
+	  DEBUG_MSG(" * volume comp: %f\n", 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;
+
+      /* 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(song, sp);
+
+      if (strip_tail==1)
+	{
+	  /* Let's not really, just say we did. */
+	  DEBUG_MSG(" - Stripping tail\n");
+	  sp->data_length = sp->loop_end;
+	}
+    }
+
+  vfs_fclose(fp);
+  return ip;
+}
+
+static int fill_bank(MidSong *song, int dr, int b)
+{
+  int i, errors=0;
+  MidToneBank *bank=((dr) ? song->drumset[b] : song->tonebank[b]);
+  if (!bank)
+    {
+      DEBUG_MSG("Huh. Tried to load instruments in non-existent %s %d\n",
+	   (dr) ? "drumset" : "tone bank", b);
+      return 0;
+    }
+  for (i=0; i<128; i++)
+    {
+      if (bank->instrument[i]==MAGIC_LOAD_INSTRUMENT)
+	{
+          bank->instrument[i]=load_instrument_dls(song, dr, b, i);
+          if (bank->instrument[i])
+            {
+              continue;
+            }
+	  if (!(bank->tone[i].name))
+	    {
+	      DEBUG_MSG("No instrument mapped to %s %d, program %d%s\n",
+		   (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 (!(song->tonebank[0]->instrument[i]))
+			song->tonebank[0]->instrument[i] =
+			  MAGIC_LOAD_INSTRUMENT;
+		    }
+		  else
+		    {
+		      if (!(song->drumset[0]->instrument[i]))
+			song->drumset[0]->instrument[i] =
+			  MAGIC_LOAD_INSTRUMENT;
+		    }
+		}
+	      bank->instrument[i] = 0;
+	      errors++;
+	    }
+	  else if (!(bank->instrument[i] =
+		     load_instrument(song,
+				     bank->tone[i].name, 
+				     (dr) ? 1 : 0,
+				     bank->tone[i].pan,
+				     bank->tone[i].amp,
+				     (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 )))
+	    {
+	      DEBUG_MSG("Couldn't load instrument %s (%s %d, program %d)\n",
+		   bank->tone[i].name,
+		   (dr)? "drum set" : "tone bank", b, i);
+	      errors++;
+	    }
+	}
+    }
+  return errors;
+}
+
+int load_missing_instruments(MidSong *song)
+{
+  int i=128,errors=0;
+  while (i--)
+    {
+      if (song->tonebank[i])
+	errors+=fill_bank(song,0,i);
+      if (song->drumset[i])
+	errors+=fill_bank(song,1,i);
+    }
+  return errors;
+}
+
+void free_instruments(MidSong *song)
+{
+  int i=128;
+  while(i--)
+    {
+      if (song->tonebank[i])
+	free_bank(song, 0, i);
+      if (song->drumset[i])
+	free_bank(song, 1, i);
+    }
+}
+
+int set_default_instrument(MidSong *song, char *name)
+{
+  MidInstrument *ip;
+  if (!(ip=load_instrument(song, name, 0, -1, -1, -1, 0, 0, 0)))
+    return -1;
+  song->default_instrument = ip;
+  song->default_program = SPECIAL_PROGRAM;
+  return 0;
+}