sound.c

/* sound.c: Sound support
   Copyright (c) 2000-2007 Russell Marks, Matan Ziv-Av, Philip Kendall,
                           Fredrick Meunier

   $Id: sound.c 3186 2007-10-03 19:07:13Z zubzero $

   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.,
   51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.

   Author contact information:

   E-mail: philip-fuse@shadowmagic.org.uk

*/

/* The AY white noise RNG algorithm is based on info from MAME's ay8910.c -
 * MAME's licence explicitly permits free use of info (even encourages it).
 */

/* NB: I know some of this stuff looks fairly CPU-hogging.
 * For example, the AY code tracks changes with sub-frame timing
 * in a rather hairy way, and there's subsampling here and there.
 * But if you measure the CPU use, it doesn't actually seem
 * very high at all. And I speak as a Cyrix owner. :-)
 */

#include <config.h>

#include <libspectrum.h>
#ifdef HAVE_SAMPLERATE
#include <samplerate.h>
#endif /* #ifdef HAVE_SAMPLERATE */

#include "fuse.h"
#include "machine.h"
#include "settings.h"
#include "sound.h"
#include "tape.h"
#include "ui/ui.h"

/* Do we have any of our sound devices available? */

/* configuration */
int sound_enabled = 0;		/* Are we currently using the sound card */
int sound_enabled_ever = 0;	/* if it's *ever* been in use; see
				   sound_ay_write() and sound_ay_reset() */
int sound_stereo = 0;		/* true for stereo *output sample* (only) */
int sound_stereo_ay_abc = 0;	/* (AY stereo) true for ABC stereo, else ACB */
int sound_stereo_ay_narrow = 0;	/* (AY stereo) true for narrow AY st. sep. */

int sound_stereo_ay = 0;	/* local copy of settings_current.stereo_ay */
int sound_stereo_beeper = 0;	/* and settings_current.stereo_beeper */


/* assume all three tone channels together match the beeper volume (ish).
 * Must be <=127 for all channels; 50+2+(24*3) = 124.
 * (Now scaled up for 16-bit.)
 */
#define AMPL_BEEPER		( 50 * 256)
#define AMPL_TAPE		( 2 * 256 )
#define AMPL_AY_TONE		( 24 * 256 )	/* three of these */

/* max. number of sub-frame AY port writes allowed;
 * given the number of port writes theoretically possible in a
 * 50th I think this should be plenty.
 */
#define AY_CHANGE_MAX		8000

/* frequency to generate sound at for hifi sound */
#define HIFI_FREQ              88200

#ifdef HAVE_SAMPLERATE
static SRC_STATE *src_state;
#endif /* #ifdef HAVE_SAMPLERATE */

int sound_generator_framesiz;
int sound_framesiz;

static int sound_generator_freq;

static int sound_channels;

static unsigned int ay_tone_levels[16];

static libspectrum_signed_word *sound_buf, *tape_buf;
static float *convert_input_buffer, *convert_output_buffer;

/* beeper stuff */
static int sound_oldpos[2], sound_fillpos[2];
static int sound_oldval[2], sound_oldval_orig[2];

/* foo_subcycles are fixed-point with low 16 bits as fractional part.
 * The other bits count as the chip does.
 */
static unsigned int ay_tone_tick[3], ay_tone_high[3], ay_noise_tick;
static unsigned int ay_tone_subcycles, ay_env_subcycles;
static unsigned int ay_env_internal_tick, ay_env_tick;
static unsigned int ay_tick_incr;
static unsigned int ay_tone_period[3], ay_noise_period, ay_env_period;

static int beeper_last_subpos[2] = { 0, 0 };

/* Local copy of the AY registers */
static libspectrum_byte sound_ay_registers[16];

struct ay_change_tag
{
  libspectrum_dword tstates;
  unsigned short ofs;
  unsigned char reg, val;
};

static struct ay_change_tag ay_change[ AY_CHANGE_MAX ];
static int ay_change_count;


#define STEREO_BUF_SIZE 4096

static int pstereobuf[ STEREO_BUF_SIZE ];
static int pstereobufsiz, pstereopos;
static int psgap = 250;
static int rstereobuf_l[ STEREO_BUF_SIZE ], rstereobuf_r[ STEREO_BUF_SIZE ];
static int rstereopos, rchan1pos, rchan2pos, rchan3pos;


static void
sound_ay_init( void )
{
/* AY output doesn't match the claimed levels; these levels are based
 * on the measurements posted to comp.sys.sinclair in Dec 2001 by
 * Matthew Westcott, adjusted as I described in a followup to his post,
 * then scaled to 0..0xffff.
 */
  static const int levels[16] = {
    0x0000, 0x0385, 0x053D, 0x0770,
    0x0AD7, 0x0FD5, 0x15B0, 0x230C,
    0x2B4C, 0x43C1, 0x5A4B, 0x732F,
    0x9204, 0xAFF1, 0xD921, 0xFFFF
  };
  int f;

/* scale the values down to fit */
  for( f = 0; f < 16; f++ )
    ay_tone_levels[f] = ( levels[f] * AMPL_AY_TONE + 0x8000 ) / 0xffff;

  ay_noise_tick = ay_noise_period = 0;
  ay_env_internal_tick = ay_env_tick = ay_env_period = 0;
  ay_tone_subcycles = ay_env_subcycles = 0;
  for( f = 0; f < 3; f++ )
    ay_tone_tick[f] = ay_tone_high[f] = 0, ay_tone_period[f] = 1;

  ay_change_count = 0;
}


void
sound_init( const char *device )
{
  static int first_init = 1;
  int f, ret;
  float hz;
#ifdef HAVE_SAMPLERATE
  int error;
#endif /* #ifdef HAVE_SAMPLERATE */

/* if we don't have any sound I/O code compiled in, don't do sound */
#ifdef NO_SOUND
  return;
#endif

  if( !( !sound_enabled && settings_current.sound &&
	 settings_current.emulation_speed == 100 ) )
    return;

  sound_stereo_ay = settings_current.stereo_ay;
  sound_stereo_beeper = settings_current.stereo_beeper;

/* only try for stereo if we need it */
  if( sound_stereo_ay || sound_stereo_beeper )
    sound_stereo = 1;
  ret =
    sound_lowlevel_init( device, &settings_current.sound_freq,
			 &sound_stereo );

  if( ret )
    return;

/* important to override these settings if not using stereo
 * (it would probably be confusing to mess with the stereo
 * settings in settings_current though, which is why we make copies
 * rather than using the real ones).
 */
  if( !sound_stereo ) {
    sound_stereo_ay = 0;
    sound_stereo_beeper = 0;
  }

  sound_enabled = sound_enabled_ever = 1;

  sound_channels = ( sound_stereo ? 2 : 1 );

  hz = ( float ) machine_current->timings.processor_speed /
    machine_current->timings.tstates_per_frame;

  sound_generator_freq =
    settings_current.sound_hifi ? HIFI_FREQ : settings_current.sound_freq;
  sound_generator_framesiz = sound_generator_freq / hz;

  if( ( sound_buf = malloc( sizeof( libspectrum_signed_word ) *
			    sound_generator_framesiz * sound_channels ) ) ==
      NULL
      || ( tape_buf =
	   malloc( sizeof( libspectrum_signed_word ) *
		   sound_generator_framesiz ) ) == NULL ) {
    if( sound_buf ) {
      free( sound_buf );
      sound_buf = NULL;
    }
    sound_end();
    return;
  }

  sound_framesiz = ( float ) settings_current.sound_freq / hz;

#ifdef HAVE_SAMPLERATE
  if( settings_current.sound_hifi ) {
    if( ( convert_input_buffer = malloc( sizeof( float ) *
					 sound_generator_framesiz *
					 sound_channels ) ) == NULL
	|| ( convert_output_buffer =
	     malloc( sizeof( float ) * sound_framesiz * sound_channels ) ) ==
	NULL ) {
      if( convert_input_buffer ) {
	free( convert_input_buffer );
	convert_input_buffer = NULL;
      }
      sound_end();
      return;
    }
  }

  src_state = src_new( SRC_SINC_MEDIUM_QUALITY, sound_channels, &error );
  if( error ) {
    ui_error( UI_ERROR_ERROR,
	      "error initialising sample rate converter %s",
	      src_strerror( error ) );
    sound_end();
    return;
  }
#endif /* #ifdef HAVE_SAMPLERATE */

/* if we're resuming, we need to be careful about what
 * gets reset. The minimum we can do is the beeper
 * buffer positions, so that's here.
 */
  sound_oldpos[0] = sound_oldpos[1] = -1;
  sound_fillpos[0] = sound_fillpos[1] = 0;

/* this stuff should only happen on the initial call.
 * (We currently assume the new sample rate will be the
 * same as the previous one, hence no need to recalculate
 * things dependent on that.)
 */
  if( first_init ) {
    first_init = 0;

    for( f = 0; f < 2; f++ )
      sound_oldval[f] = sound_oldval_orig[f] = 0;
  }

  if( sound_stereo_beeper ) {
    for( f = 0; f < STEREO_BUF_SIZE; f++ )
      pstereobuf[f] = 0;
    pstereopos = 0;
    pstereobufsiz = ( sound_generator_freq * psgap ) / 22000;
  }

  if( sound_stereo_ay ) {
    int pos =
      ( sound_stereo_ay_narrow ? 3 : 6 ) * sound_generator_freq / 8000;

    for( f = 0; f < STEREO_BUF_SIZE; f++ )
      rstereobuf_l[f] = rstereobuf_r[f] = 0;
    rstereopos = 0;

    /* the actual ACB/ABC bit :-) */
    rchan1pos = -pos;
    if( sound_stereo_ay_abc )
      rchan2pos = 0, rchan3pos = pos;
    else
      rchan2pos = pos, rchan3pos = 0;
  }

  ay_tick_incr = ( int ) ( 65536. *
			   libspectrum_timings_ay_speed( machine_current->
							 machine ) /
			   sound_generator_freq );
}

void
sound_pause( void )
{
  if( sound_enabled )
    sound_end();
}


void
sound_unpause( void )
{
/* No sound if fastloading in progress */
  if( settings_current.fastload && tape_is_playing() )
    return;

  sound_init( settings_current.sound_device );
}


void
sound_end( void )
{
  if( sound_enabled ) {
    if( sound_buf ) {
      free( sound_buf );
      sound_buf = NULL;
      free( tape_buf );
      tape_buf = NULL;
    }
    if( convert_input_buffer ) {
      free( convert_input_buffer );
      convert_input_buffer = NULL;
    }
    if( convert_output_buffer ) {
      free( convert_output_buffer );
      convert_output_buffer = NULL;
    }
#ifdef HAVE_SAMPLERATE
    if( src_state )
      src_state = src_delete( src_state );
#endif /* #ifdef HAVE_SAMPLERATE */
    sound_lowlevel_end();
    sound_enabled = 0;
  }
}


/* write sample to buffer as pseudo-stereo */
static void
sound_write_buf_pstereo( libspectrum_signed_word * out, int c )
{
  int bl = ( c - pstereobuf[ pstereopos ] ) / 2;
  int br = ( c + pstereobuf[ pstereopos ] ) / 2;

  if( bl < -AMPL_BEEPER )
    bl = -AMPL_BEEPER;
  if( br < -AMPL_BEEPER )
    br = -AMPL_BEEPER;
  if( bl > AMPL_BEEPER )
    bl = AMPL_BEEPER;
  if( br > AMPL_BEEPER )
    br = AMPL_BEEPER;

  *out = bl;
  out[1] = br;

  pstereobuf[ pstereopos ] = c;
  pstereopos++;
  if( pstereopos >= pstereobufsiz )
    pstereopos = 0;
}



/* not great having this as a macro to inline it, but it's only
 * a fairly short routine, and it saves messing about.
 * (XXX ummm, possibly not so true any more :-))
 */
#define AY_GET_SUBVAL( chan ) \
  ( level * 2 * ay_tone_tick[ chan ] / tone_count )

#define AY_DO_TONE( var, chan ) \
  ( var ) = 0;								\
  is_low = 0;								\
  if( level ) {								\
    if( ay_tone_high[ chan ] )						\
      ( var ) = ( level );						\
    else {								\
      ( var ) = -( level );						\
      is_low = 1;							\
    }									\
  }									\
  									\
  ay_tone_tick[ chan ] += tone_count;					\
  count = 0;								\
  while( ay_tone_tick[ chan ] >= ay_tone_period[ chan ] ) {		\
    count++;								\
    ay_tone_tick[ chan ] -= ay_tone_period[ chan ];			\
    ay_tone_high[ chan ] = !ay_tone_high[ chan ];			\
    									\
    /* has to be here, unfortunately... */				\
    if( count == 1 && level && ay_tone_tick[ chan ] < tone_count ) {	\
      if( is_low )							\
        ( var ) += AY_GET_SUBVAL( chan );				\
      else								\
        ( var ) -= AY_GET_SUBVAL( chan );				\
      }									\
    }									\
  									\
  /* if it's changed more than once during the sample, we can't */	\
  /* represent it faithfully. So, just hope it's a sample.      */	\
  /* (That said, this should also help avoid aliasing noise.)   */	\
  if( count > 1 )							\
    ( var ) = -( level )


/* add val, correctly delayed on either left or right buffer,
 * to add the AY stereo positioning. This doesn't actually put
 * anything directly in sound_buf, though.
 */
#define GEN_STEREO( pos, val ) \
  if( ( pos ) < 0 ) {							\
    rstereobuf_l[ rstereopos ] += ( val );				\
    rstereobuf_r[ ( rstereopos - pos ) % STEREO_BUF_SIZE ] += ( val );	\
  } else {								\
    rstereobuf_l[ ( rstereopos + pos ) % STEREO_BUF_SIZE ] += ( val );	\
    rstereobuf_r[ rstereopos ] += ( val );				\
  }



/* bitmasks for envelope */
#define AY_ENV_CONT	8
#define AY_ENV_ATTACK	4
#define AY_ENV_ALT	2
#define AY_ENV_HOLD	1

#define HZ_COMMON_DENOMINATOR 50

static void
sound_ay_overlay( void )
{
  static int rng = 1;
  static int noise_toggle = 0;
  static int env_first = 1, env_rev = 0, env_counter = 15;
  int tone_level[3];
  int mixer, envshape;
  int f, g, level, count;
  libspectrum_signed_word *ptr;
  struct ay_change_tag *change_ptr = ay_change;
  int changes_left = ay_change_count;
  int reg, r;
  int is_low;
  int chan1, chan2, chan3;
  unsigned int tone_count, noise_count;
  libspectrum_dword sfreq, cpufreq;

/* If no AY chip, don't produce any AY sound (!) */
  if( !machine_current->capabilities & LIBSPECTRUM_MACHINE_CAPABILITY_AY )
    return;

/* convert change times to sample offsets, use common denominator of 50 to
   avoid overflowing a dword */
  sfreq = sound_generator_freq / HZ_COMMON_DENOMINATOR;
  cpufreq = machine_current->timings.processor_speed / HZ_COMMON_DENOMINATOR;
  for( f = 0; f < ay_change_count; f++ )
    ay_change[f].ofs = ( ay_change[f].tstates * sfreq ) / cpufreq;

  for( f = 0, ptr = sound_buf; f < sound_generator_framesiz; f++ ) {
    /* update ay registers. All this sub-frame change stuff
     * is pretty hairy, but how else would you handle the
     * samples in Robocop? :-) It also clears up some other
     * glitches.
     */
    while( changes_left && f >= change_ptr->ofs ) {
      sound_ay_registers[ reg = change_ptr->reg ] = change_ptr->val;
      change_ptr++;
      changes_left--;

      /* fix things as needed for some register changes */
      switch ( reg ) {
      case 0:
      case 1:
      case 2:
      case 3:
      case 4:
      case 5:
	r = reg >> 1;
	/* a zero-len period is the same as 1 */
	ay_tone_period[r] = ( sound_ay_registers[ reg & ~1 ] |
			      ( sound_ay_registers[ reg | 1 ] & 15 ) << 8 );
	if( !ay_tone_period[r] )
	  ay_tone_period[r]++;

	/* important to get this right, otherwise e.g. Ghouls 'n' Ghosts
	 * has really scratchy, horrible-sounding vibrato.
	 */
	if( ay_tone_tick[r] >= ay_tone_period[r] * 2 )
	  ay_tone_tick[r] %= ay_tone_period[r] * 2;
	break;
      case 6:
	ay_noise_tick = 0;
	ay_noise_period = ( sound_ay_registers[ reg ] & 31 );
	break;
      case 11:
      case 12:
	/* this one *isn't* fixed-point */
	ay_env_period =
	  sound_ay_registers[11] | ( sound_ay_registers[12] << 8 );
	break;
      case 13:
	ay_env_internal_tick = ay_env_tick = ay_env_subcycles = 0;
	env_first = 1;
	env_rev = 0;
	env_counter = ( sound_ay_registers[13] & AY_ENV_ATTACK ) ? 0 : 15;
	break;
      }
    }

    /* the tone level if no enveloping is being used */
    for( g = 0; g < 3; g++ )
      tone_level[g] = ay_tone_levels[ sound_ay_registers[ 8 + g ] & 15 ];

    /* envelope */
    envshape = sound_ay_registers[13];
    level = ay_tone_levels[ env_counter ];

    for( g = 0; g < 3; g++ )
      if( sound_ay_registers[ 8 + g ] & 16 )
	tone_level[g] = level;

    /* envelope output counter gets incr'd every 16 AY cycles.
     * Has to be a while, as this is sub-output-sample res.
     */
    ay_env_subcycles += ay_tick_incr;
    noise_count = 0;
    while( ay_env_subcycles >= ( 16 << 16 ) ) {
      ay_env_subcycles -= ( 16 << 16 );
      noise_count++;
      ay_env_tick++;
      while( ay_env_tick >= ay_env_period ) {
	ay_env_tick -= ay_env_period;

	/* do a 1/16th-of-period incr/decr if needed */
	if( env_first ||
	    ( ( envshape & AY_ENV_CONT ) && !( envshape & AY_ENV_HOLD ) ) ) {
	  if( env_rev )
	    env_counter -= ( envshape & AY_ENV_ATTACK ) ? 1 : -1;
	  else
	    env_counter += ( envshape & AY_ENV_ATTACK ) ? 1 : -1;
	  if( env_counter < 0 )
	    env_counter = 0;
	  if( env_counter > 15 )
	    env_counter = 15;
	}

	ay_env_internal_tick++;
	while( ay_env_internal_tick >= 16 ) {
	  ay_env_internal_tick -= 16;

	  /* end of cycle */
	  if( !( envshape & AY_ENV_CONT ) )
	    env_counter = 0;
	  else {
	    if( envshape & AY_ENV_HOLD ) {
	      if( env_first && ( envshape & AY_ENV_ALT ) )
		env_counter = ( env_counter ? 0 : 15 );
	    } else {
	      /* non-hold */
	      if( envshape & AY_ENV_ALT )
		env_rev = !env_rev;
	      else
		env_counter = ( envshape & AY_ENV_ATTACK ) ? 0 : 15;
	    }
	  }

	  env_first = 0;
	}

	/* don't keep trying if period is zero */
	if( !ay_env_period )
	  break;
      }
    }

    /* generate tone+noise... or neither.
     * (if no tone/noise is selected, the chip just shoves the
     * level out unmodified. This is used by some sample-playing
     * stuff.)
     */
    chan1 = tone_level[0];
    chan2 = tone_level[1];
    chan3 = tone_level[2];
    mixer = sound_ay_registers[7];

    ay_tone_subcycles += ay_tick_incr;
    tone_count = ay_tone_subcycles >> ( 3 + 16 );
    ay_tone_subcycles &= ( 8 << 16 ) - 1;

    if( ( mixer & 1 ) == 0 ) {
      level = chan1;
      AY_DO_TONE( chan1, 0 );
    }
    if( ( mixer & 0x08 ) == 0 && noise_toggle )
      chan1 = 0;

    if( ( mixer & 2 ) == 0 ) {
      level = chan2;
      AY_DO_TONE( chan2, 1 );
    }
    if( ( mixer & 0x10 ) == 0 && noise_toggle )
      chan2 = 0;

    if( ( mixer & 4 ) == 0 ) {
      level = chan3;
      AY_DO_TONE( chan3, 2 );
    }
    if( ( mixer & 0x20 ) == 0 && noise_toggle )
      chan3 = 0;

    /* write the sample(s) */
    if( !sound_stereo ) {
      /* mono */
      ( *ptr++ ) += chan1 + chan2 + chan3;
    } else {
      if( !sound_stereo_ay ) {
	/* stereo output, but mono AY sound; still,
	 * incr separately in case of beeper pseudostereo.
	 */
	( *ptr++ ) += chan1 + chan2 + chan3;
	( *ptr++ ) += chan1 + chan2 + chan3;
      } else {
	/* stereo with ACB/ABC AY positioning.
	 * Here we use real stereo positions for the channels.
	 * Just because, y'know, it's cool and stuff. No, really. :-)
	 * This is a little tricky, as it works by delaying sounds
	 * on the left or right channels to model the delay you get
	 * in the real world when sounds originate at different places.
	 */
	GEN_STEREO( rchan1pos, chan1 );
	GEN_STEREO( rchan2pos, chan2 );
	GEN_STEREO( rchan3pos, chan3 );
	( *ptr++ ) += rstereobuf_l[ rstereopos ];
	( *ptr++ ) += rstereobuf_r[ rstereopos ];
	rstereobuf_l[ rstereopos ] = rstereobuf_r[ rstereopos ] = 0;
	rstereopos++;
	if( rstereopos >= STEREO_BUF_SIZE )
	  rstereopos = 0;
      }
    }

    /* update noise RNG/filter */
    ay_noise_tick += noise_count;
    while( ay_noise_tick >= ay_noise_period ) {
      ay_noise_tick -= ay_noise_period;

      if( ( rng & 1 ) ^ ( ( rng & 2 ) ? 1 : 0 ) )
	noise_toggle = !noise_toggle;

      /* rng is 17-bit shift reg, bit 0 is output.
       * input is bit 0 xor bit 2.
       */
      rng |= ( ( rng & 1 ) ^ ( ( rng & 4 ) ? 1 : 0 ) ) ? 0x20000 : 0;
      rng >>= 1;

      /* don't keep trying if period is zero */
      if( !ay_noise_period )
	break;
    }
  }
}


/* don't make the change immediately; record it for later,
 * to be made by sound_frame() (via sound_ay_overlay()).
 */
void
sound_ay_write( int reg, int val, libspectrum_dword now )
{
  if( ay_change_count < AY_CHANGE_MAX ) {
    ay_change[ ay_change_count ].tstates = now;
    ay_change[ ay_change_count ].reg = ( reg & 15 );
    ay_change[ ay_change_count ].val = val;
    ay_change_count++;
  }
}


/* no need to call this initially, but should be called
 * on reset otherwise.
 */
void
sound_ay_reset( void )
{
  int f;

/* recalculate timings based on new machines ay clock */
  sound_ay_init();

  ay_change_count = 0;
  for( f = 0; f < 16; f++ )
    sound_ay_write( f, 0, 0 );
  for( f = 0; f < 3; f++ )
    ay_tone_high[f] = 0;
  ay_tone_subcycles = ay_env_subcycles = 0;
}


/* write stereo or mono beeper sample, and incr ptr */
#define SOUND_WRITE_BUF_BEEPER( ptr, val ) \
  do {							\
    if( sound_stereo_beeper ) {				\
      sound_write_buf_pstereo( ( ptr ), ( val ) );	\
      ( ptr ) += 2;					\
    } else {						\
      *( ptr )++ = ( val );				\
      if( sound_stereo )				\
        *( ptr )++ = ( val );				\
    }							\
  } while(0)

/* the tape version works by writing to a separate mono buffer,
 * which gets added after being generated.
 */
#define SOUND_WRITE_BUF( is_tape, ptr, val ) \
  if( is_tape )					\
    *( ptr )++ = ( val );			\
  else						\
    SOUND_WRITE_BUF_BEEPER( ptr, val )

#ifdef HAVE_SAMPLERATE
static void
sound_resample( void )
{
  int error;
  SRC_DATA data;

  data.data_in = convert_input_buffer;
  data.input_frames = sound_generator_framesiz;
  data.data_out = convert_output_buffer;
  data.output_frames = sound_framesiz;
  data.src_ratio =
    ( double ) settings_current.sound_freq / sound_generator_freq;
  data.end_of_input = 0;

  src_short_to_float_array( ( const short * ) sound_buf, convert_input_buffer,
			    sound_generator_framesiz * sound_channels );

  while( data.input_frames ) {
    error = src_process( src_state, &data );
    if( error ) {
      ui_error( UI_ERROR_ERROR, "hifi sound downsample error %s",
		src_strerror( error ) );
      sound_end();
      return;
    }

    src_float_to_short_array( convert_output_buffer, ( short * ) sound_buf,
			      data.output_frames_gen * sound_channels );

    sound_lowlevel_frame( sound_buf,
			  data.output_frames_gen * sound_channels );

    data.data_in += data.input_frames_used * sound_channels;
    data.input_frames -= data.input_frames_used;
  }
}
#endif /* #ifdef HAVE_SAMPLERATE */

void
sound_frame( void )
{
  libspectrum_signed_word *ptr, *tptr;
  int f, bchan;
  int ampl = AMPL_BEEPER;

  if( !sound_enabled )
    return;

/* fill in remaining beeper/tape sound */
  ptr =
    sound_buf + ( sound_stereo ? sound_fillpos[0] * 2 : sound_fillpos[0] );
  for( bchan = 0; bchan < 2; bchan++ ) {
    for( f = sound_fillpos[ bchan ]; f < sound_generator_framesiz; f++ )
      SOUND_WRITE_BUF( bchan, ptr, sound_oldval[ bchan ] );

    ptr = tape_buf + sound_fillpos[1];
    ampl = AMPL_TAPE;
  }

/* overlay tape sound */
  ptr = sound_buf;
  tptr = tape_buf;
  for( f = 0; f < sound_generator_framesiz; f++, tptr++ ) {
    ( *ptr++ ) += *tptr;
    if( sound_stereo )
      ( *ptr++ ) += *tptr;
  }

/* overlay AY sound */
  sound_ay_overlay();

#ifdef HAVE_SAMPLERATE
/* resample from generated frequency down to output frequency if required */
  if( settings_current.sound_hifi )
    sound_resample();
  else
#endif /* #ifdef HAVE_SAMPLERATE */
    sound_lowlevel_frame( sound_buf,
			  sound_generator_framesiz * sound_channels );

  sound_oldpos[0] = sound_oldpos[1] = -1;
  sound_fillpos[0] = sound_fillpos[1] = 0;

  ay_change_count = 0;
}


/* two beepers are supported - the real beeper (call with is_tape==0)
 * and a `fake' beeper which lets you hear when a tape is being played.
 */
void
sound_beeper( int is_tape, int on )
{
  libspectrum_signed_word *ptr;
  int newpos, subpos;
  int val, subval;
  int f;
  int bchan = ( is_tape ? 1 : 0 );
  int ampl = ( is_tape ? AMPL_TAPE : AMPL_BEEPER );
  int vol = ampl * 2;

  if( !sound_enabled )
    return;

  val = ( on ? -ampl : ampl );

  if( val == sound_oldval_orig[ bchan ] )
    return;

/* XXX a lookup table might help here, but would need to regenerate it
 * whenever cycles_per_frame were changed (i.e. when machine type changed).
 */
  newpos =
    ( tstates * sound_generator_framesiz ) /
    machine_current->timings.tstates_per_frame;
  subpos =
    ( ( ( libspectrum_signed_qword ) tstates ) * sound_generator_framesiz *
      vol ) / ( machine_current->timings.tstates_per_frame ) - vol * newpos;

/* if we already wrote here, adjust the level.
 */
  if( newpos == sound_oldpos[ bchan ] ) {
    /* adjust it as if the rest of the sample period were all in
     * the new state. (Often it will be, but if not, we'll fix
     * it later by doing this again.)
     */
    if( on )
      beeper_last_subpos[ bchan ] += vol - subpos;
    else
      beeper_last_subpos[ bchan ] -= vol - subpos;
  } else
    beeper_last_subpos[ bchan ] = ( on ? vol - subpos : subpos );

  subval = ampl - beeper_last_subpos[ bchan ];

  if( newpos >= 0 ) {
    /* fill gap from previous position */
    if( is_tape )
      ptr = tape_buf + sound_fillpos[1];
    else
      ptr =
	sound_buf +
	( sound_stereo ? sound_fillpos[0] * 2 : sound_fillpos[0] );

    for( f = sound_fillpos[ bchan ];
	 f < newpos && f < sound_generator_framesiz;
	 f++ )
      SOUND_WRITE_BUF( bchan, ptr, sound_oldval[ bchan ] );

    if( newpos < sound_generator_framesiz ) {
      /* newpos may be less than sound_fillpos, so... */
      if( is_tape )
	ptr = tape_buf + newpos;
      else
	ptr = sound_buf + ( sound_stereo ? newpos * 2 : newpos );

      /* write subsample value */
      SOUND_WRITE_BUF( bchan, ptr, subval );
    }
  }

  sound_oldpos[ bchan ] = newpos;
  sound_fillpos[ bchan ] = newpos + 1;
  sound_oldval[ bchan ] = sound_oldval_orig[ bchan ] = val;
}