view decoders/timidity/mix.c @ 474:c66080364dff

Most decoders now report total sample play time, now. Technically, this breaks binary compatibility with the 1.0 branch, since it extends the Sound_Sample struct, but most (all?) programs are just passing pointers allocated by SDL_sound around, and might be okay. Source-level compatibility is not broken...yet! :) --ryan. -------- Original Message -------- Subject: SDL_sound patch: Finding total length of time of sound file. Date: Sun, 26 Jan 2003 09:31:17 -0800 (PST) Hi Ryan, I am working with Eric Wing and helping him modify SDL_sound. AS part of our efforts in improving and enhancing SDL_sound, we like to submit this patch. We modified the codecs to find the total time of a sound file. Below is the explanation of the patch. The patch is appended as an attachment to this email. * MOTIVATION: We needed the ability to get the total play time of a sample (And we noticed that we're not the only ones). Since SDL_sound blocks direct access to the specific decoders, there is no way for a user to know this information short of decoding the whole thing. Because of this, we believe this will be a useful addition, even though the accuracy may not be perfect (subject to each decoder) or the information may not always be available. * CONTRIBUTORS: Wesley Leong (modified the majority of the codecs and verified the results) Eric Wing (showed everyone how to do modify codec, modified mikmod) Wang Lam (modified a handful of codecs, researched into specs and int overflow) Ahilan Anantha (modified a few codecs and helped with integer math) * GENERAL ISSUES: We chose the value to be milliseconds as an Sint32. Milliseconds because that's what Sound_Seek takes as a parameter and -1 to allow for instances/codecs where the value could not be determined. We are not sure if this is the final convention you want, so we are willing to work with you on this. We also expect the total_time field to be set on open and never again modified by SDL_sound. Users may access it directly much like the sample buffer and buffer_size. We thought about recomputing the time on DecodeAll, but since users may seek or decode small chunks first, not all the data may be there. So this is better done by the user. This may be good information to document. Currently, all the main codecs are implemented except for QuickTime.
author Ryan C. Gordon <icculus@icculus.org>
date Sat, 08 May 2004 08:19:50 +0000
parents 2d887640d300
children
line wrap: on
line source

/*

    TiMidity -- Experimental MIDI to WAVE converter
    Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>

    Suddenly, you realize that this program is free software; you get
    an overwhelming urge to 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 another 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.
    I bet they'll be amazed.

    mix.c */

#if HAVE_CONFIG_H
#  include <config.h>
#endif

#include <math.h>
#include <stdio.h>
#include <stdlib.h>

#include "SDL_sound.h"

#define __SDL_SOUND_INTERNAL__
#include "SDL_sound_internal.h"

#include "timidity.h"
#include "options.h"
#include "instrum.h"
#include "playmidi.h"
#include "output.h"
#include "tables.h"
#include "resample.h"
#include "mix.h"

/* Returns 1 if envelope runs out */
int recompute_envelope(MidiSong *song, int v)
{
  int stage;

  stage = song->voice[v].envelope_stage;

  if (stage>5)
    {
      /* Envelope ran out. */
      song->voice[v].status = VOICE_FREE;
      return 1;
    }

  if (song->voice[v].sample->modes & MODES_ENVELOPE)
    {
      if (song->voice[v].status==VOICE_ON || song->voice[v].status==VOICE_SUSTAINED)
	{
	  if (stage>2)
	    {
	      /* Freeze envelope until note turns off. Trumpets want this. */
	      song->voice[v].envelope_increment=0;
	      return 0;
	    }
	}
    }
  song->voice[v].envelope_stage=stage+1;

  if (song->voice[v].envelope_volume==song->voice[v].sample->envelope_offset[stage])
    return recompute_envelope(song, v);
  song->voice[v].envelope_target = song->voice[v].sample->envelope_offset[stage];
  song->voice[v].envelope_increment = song->voice[v].sample->envelope_rate[stage];
  if (song->voice[v].envelope_target < song->voice[v].envelope_volume)
    song->voice[v].envelope_increment = -song->voice[v].envelope_increment;
  return 0;
}

void apply_envelope_to_amp(MidiSong *song, int v)
{
  float lamp = song->voice[v].left_amp, ramp;
  Sint32 la,ra;
  if (song->voice[v].panned == PANNED_MYSTERY)
    {
      ramp = song->voice[v].right_amp;
      if (song->voice[v].tremolo_phase_increment)
	{
	  lamp *= song->voice[v].tremolo_volume;
	  ramp *= song->voice[v].tremolo_volume;
	}
      if (song->voice[v].sample->modes & MODES_ENVELOPE)
	{
	  lamp *= (float)vol_table[song->voice[v].envelope_volume>>23];
	  ramp *= (float)vol_table[song->voice[v].envelope_volume>>23];
	}

      la = (Sint32)FSCALE(lamp,AMP_BITS);
      
      if (la>MAX_AMP_VALUE)
	la=MAX_AMP_VALUE;

      ra = (Sint32)FSCALE(ramp,AMP_BITS);
      if (ra>MAX_AMP_VALUE)
	ra=MAX_AMP_VALUE;
      
      song->voice[v].left_mix = la;
      song->voice[v].right_mix = ra;
    }
  else
    {
      if (song->voice[v].tremolo_phase_increment)
	lamp *= song->voice[v].tremolo_volume;
      if (song->voice[v].sample->modes & MODES_ENVELOPE)
	lamp *= (float)vol_table[song->voice[v].envelope_volume>>23];

      la = (Sint32)FSCALE(lamp,AMP_BITS);

      if (la>MAX_AMP_VALUE)
	la=MAX_AMP_VALUE;

      song->voice[v].left_mix = la;
    }
}

static int update_envelope(MidiSong *song, int v)
{
  song->voice[v].envelope_volume += song->voice[v].envelope_increment;
  /* Why is there no ^^ operator?? */
  if (((song->voice[v].envelope_increment < 0) &&
       (song->voice[v].envelope_volume <= song->voice[v].envelope_target)) ||
      ((song->voice[v].envelope_increment > 0) &&
	   (song->voice[v].envelope_volume >= song->voice[v].envelope_target)))
    {
      song->voice[v].envelope_volume = song->voice[v].envelope_target;
      if (recompute_envelope(song, v))
	return 1;
    }
  return 0;
}

static void update_tremolo(MidiSong *song, int v)
{
  Sint32 depth = song->voice[v].sample->tremolo_depth << 7;

  if (song->voice[v].tremolo_sweep)
    {
      /* Update sweep position */

      song->voice[v].tremolo_sweep_position += song->voice[v].tremolo_sweep;
      if (song->voice[v].tremolo_sweep_position >= (1 << SWEEP_SHIFT))
	song->voice[v].tremolo_sweep=0; /* Swept to max amplitude */
      else
	{
	  /* Need to adjust depth */
	  depth *= song->voice[v].tremolo_sweep_position;
	  depth >>= SWEEP_SHIFT;
	}
    }

  song->voice[v].tremolo_phase += song->voice[v].tremolo_phase_increment;

  /* if (song->voice[v].tremolo_phase >= (SINE_CYCLE_LENGTH<<RATE_SHIFT))
     song->voice[v].tremolo_phase -= SINE_CYCLE_LENGTH<<RATE_SHIFT;  */

  song->voice[v].tremolo_volume = (float) 
    (1.0 - FSCALENEG((sine(song->voice[v].tremolo_phase >> RATE_SHIFT) + 1.0)
		    * depth * TREMOLO_AMPLITUDE_TUNING,
		    17));

  /* I'm not sure about the +1.0 there -- it makes tremoloed voices'
     volumes on average the lower the higher the tremolo amplitude. */
}

/* Returns 1 if the note died */
static int update_signal(MidiSong *song, int v)
{
  if (song->voice[v].envelope_increment && update_envelope(song, v))
    return 1;

  if (song->voice[v].tremolo_phase_increment)
    update_tremolo(song, v);

  apply_envelope_to_amp(song, v);
  return 0;
}

#define MIXATION(a)	*lp++ += (a)*s;

static void mix_mystery_signal(MidiSong *song, sample_t *sp, Sint32 *lp, int v,
			       int count)
{
  Voice *vp = song->voice + v;
  final_volume_t 
    left=vp->left_mix, 
    right=vp->right_mix;
  int cc;
  sample_t s;

  if (!(cc = vp->control_counter))
    {
      cc = song->control_ratio;
      if (update_signal(song, v))
	return;	/* Envelope ran out */
      left = vp->left_mix;
      right = vp->right_mix;
    }
  
  while (count)
    if (cc < count)
      {
	count -= cc;
	while (cc--)
	  {
	    s = *sp++;
	    MIXATION(left);
	    MIXATION(right);
	  }
	cc = song->control_ratio;
	if (update_signal(song, v))
	  return;	/* Envelope ran out */
	left = vp->left_mix;
	right = vp->right_mix;
      }
    else
      {
	vp->control_counter = cc - count;
	while (count--)
	  {
	    s = *sp++;
	    MIXATION(left);
	    MIXATION(right);
	  }
	return;
      }
}

static void mix_center_signal(MidiSong *song, sample_t *sp, Sint32 *lp, int v,
			      int count)
{
  Voice *vp = song->voice + v;
  final_volume_t 
    left=vp->left_mix;
  int cc;
  sample_t s;

  if (!(cc = vp->control_counter))
    {
      cc = song->control_ratio;
      if (update_signal(song, v))
	return;	/* Envelope ran out */
      left = vp->left_mix;
    }
  
  while (count)
    if (cc < count)
      {
	count -= cc;
	while (cc--)
	  {
	    s = *sp++;
	    MIXATION(left);
	    MIXATION(left);
	  }
	cc = song->control_ratio;
	if (update_signal(song, v))
	  return;	/* Envelope ran out */
	left = vp->left_mix;
      }
    else
      {
	vp->control_counter = cc - count;
	while (count--)
	  {
	    s = *sp++;
	    MIXATION(left);
	    MIXATION(left);
	  }
	return;
      }
}

static void mix_single_signal(MidiSong *song, sample_t *sp, Sint32 *lp, int v,
			      int count)
{
  Voice *vp = song->voice + v;
  final_volume_t 
    left=vp->left_mix;
  int cc;
  sample_t s;
  
  if (!(cc = vp->control_counter))
    {
      cc = song->control_ratio;
      if (update_signal(song, v))
	return;	/* Envelope ran out */
      left = vp->left_mix;
    }
  
  while (count)
    if (cc < count)
      {
	count -= cc;
	while (cc--)
	  {
	    s = *sp++;
	    MIXATION(left);
	    lp++;
	  }
	cc = song->control_ratio;
	if (update_signal(song, v))
	  return;	/* Envelope ran out */
	left = vp->left_mix;
      }
    else
      {
	vp->control_counter = cc - count;
	while (count--)
	  {
	    s = *sp++;
	    MIXATION(left);
	    lp++;
	  }
	return;
      }
}

static void mix_mono_signal(MidiSong *song, sample_t *sp, Sint32 *lp, int v,
			    int count)
{
  Voice *vp = song->voice + v;
  final_volume_t 
    left=vp->left_mix;
  int cc;
  sample_t s;
  
  if (!(cc = vp->control_counter))
    {
      cc = song->control_ratio;
      if (update_signal(song, v))
	return;	/* Envelope ran out */
      left = vp->left_mix;
    }
  
  while (count)
    if (cc < count)
      {
	count -= cc;
	while (cc--)
	  {
	    s = *sp++;
	    MIXATION(left);
	  }
	cc = song->control_ratio;
	if (update_signal(song, v))
	  return;	/* Envelope ran out */
	left = vp->left_mix;
      }
    else
      {
	vp->control_counter = cc - count;
	while (count--)
	  {
	    s = *sp++;
	    MIXATION(left);
	  }
	return;
      }
}

static void mix_mystery(MidiSong *song, sample_t *sp, Sint32 *lp, int v, int count)
{
  final_volume_t 
    left = song->voice[v].left_mix, 
    right = song->voice[v].right_mix;
  sample_t s;
  
  while (count--)
    {
      s = *sp++;
      MIXATION(left);
      MIXATION(right);
    }
}

static void mix_center(MidiSong *song, sample_t *sp, Sint32 *lp, int v, int count)
{
  final_volume_t 
    left = song->voice[v].left_mix;
  sample_t s;
  
  while (count--)
    {
      s = *sp++;
      MIXATION(left);
      MIXATION(left);
    }
}

static void mix_single(MidiSong *song, sample_t *sp, Sint32 *lp, int v, int count)
{
  final_volume_t 
    left = song->voice[v].left_mix;
  sample_t s;
  
  while (count--)
    {
      s = *sp++;
      MIXATION(left);
      lp++;
    }
}

static void mix_mono(MidiSong *song, sample_t *sp, Sint32 *lp, int v, int count)
{
  final_volume_t 
    left = song->voice[v].left_mix;
  sample_t s;
  
  while (count--)
    {
      s = *sp++;
      MIXATION(left);
    }
}

/* Ramp a note out in c samples */
static void ramp_out(MidiSong *song, sample_t *sp, Sint32 *lp, int v, Sint32 c)
{

  /* should be final_volume_t, but Uint8 gives trouble. */
  Sint32 left, right, li, ri;

  sample_t s=0; /* silly warning about uninitialized s */

  /* Fix by James Caldwell */
  if ( c == 0 ) c = 1;
  
  left=song->voice[v].left_mix;
  li=-(left/c);
  if (!li) li=-1;

  /* printf("Ramping out: left=%d, c=%d, li=%d\n", left, c, li); */

  if (!(song->encoding & PE_MONO))
    {
      if (song->voice[v].panned==PANNED_MYSTERY)
	{
	  right=song->voice[v].right_mix;
	  ri=-(right/c);
	  while (c--)
	    {
	      left += li;
	      if (left<0)
		left=0;
	      right += ri;
	      if (right<0)
		right=0;
	      s=*sp++;
	      MIXATION(left);
	      MIXATION(right);
	    }
	}
      else if (song->voice[v].panned==PANNED_CENTER)
	{
	  while (c--)
	    {
	      left += li;
	      if (left<0)
		return;
	      s=*sp++;	
	      MIXATION(left);
	      MIXATION(left);
	    }
	}
      else if (song->voice[v].panned==PANNED_LEFT)
	{
	  while (c--)
	    {
	      left += li;
	      if (left<0)
		return;
	      s=*sp++;
	      MIXATION(left);
	      lp++;
	    }
	}
      else if (song->voice[v].panned==PANNED_RIGHT)
	{
	  while (c--)
	    {
	      left += li;
	      if (left<0)
		return;
	      s=*sp++;
	      lp++;
	      MIXATION(left);
	    }
	}
    }
  else
    {
      /* Mono output.  */
      while (c--)
	{
	  left += li;
	  if (left<0)
	    return;
	  s=*sp++;
	  MIXATION(left);
	}
    }
}


/**************** interface function ******************/

void mix_voice(MidiSong *song, Sint32 *buf, int v, Sint32 c)
{
  Voice *vp = song->voice + v;
  sample_t *sp;
  if (vp->status==VOICE_DIE)
    {
      if (c>=MAX_DIE_TIME)
	c=MAX_DIE_TIME;
      sp=resample_voice(song, v, &c);
      ramp_out(song, sp, buf, v, c);
      vp->status=VOICE_FREE;
    }
  else
    {
      sp=resample_voice(song, v, &c);
      if (song->encoding & PE_MONO)
	{
	  /* Mono output. */
	  if (vp->envelope_increment || vp->tremolo_phase_increment)
	    mix_mono_signal(song, sp, buf, v, c);
	  else
	    mix_mono(song, sp, buf, v, c);
	}
      else
	{
	  if (vp->panned == PANNED_MYSTERY)
	    {
	      if (vp->envelope_increment || vp->tremolo_phase_increment)
		mix_mystery_signal(song, sp, buf, v, c);
	      else
		mix_mystery(song, sp, buf, v, c);
	    }
	  else if (vp->panned == PANNED_CENTER)
	    {
	      if (vp->envelope_increment || vp->tremolo_phase_increment)
		mix_center_signal(song, sp, buf, v, c);
	      else
		mix_center(song, sp, buf, v, c);
	    }
	  else
	    { 
	      /* It's either full left or full right. In either case,
		 every other sample is 0. Just get the offset right: */
	      if (vp->panned == PANNED_RIGHT) buf++;
	      
	      if (vp->envelope_increment || vp->tremolo_phase_increment)
		mix_single_signal(song, sp, buf, v, c);
	      else 
		mix_single(song, sp, buf, v, c);
	    }
	}
    }
}