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AppleWin/source/AY8910.cpp
tomcw ea23def601 More header refactoring
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/* sound.c: Sound support
Copyright (c) 2000-2007 Russell Marks, Matan Ziv-Av, Philip Kendall,
Fredrick Meunier
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
*/
// [AppleWin-TC] From FUSE's sound.c module
#include "StdAfx.h"
#include <windows.h>
#include <stdio.h>
#include <crtdbg.h>
#include "AY8910.h"
#include "Applewin.h" // For g_fh
#include "Mockingboard.h" // For g_uTimer1IrqCount
/* 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. :-)
*/
libspectrum_signed_word** g_ppSoundBuffers; // Used to pass param to sound_ay_overlay()
/* 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 */
#define AMPL_AY_TONE ( 42 * 256 ) // 42*3 = 126
/* 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 // [TC] Moved into AY8910.h
///* 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;
#if 0
/* beeper stuff */
static int sound_oldpos[2], sound_fillpos[2];
static int sound_oldval[2], sound_oldval_orig[2];
#endif
#if 0
#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;
#endif
// Statics:
double CAY8910::m_fCurrentCLK_AY8910 = 0.0;
CAY8910::CAY8910() :
// Init the statics that were in sound_ay_overlay()
rng(1),
noise_toggle(0),
env_first(1), env_rev(0), env_counter(15)
{
m_fCurrentCLK_AY8910 = g_fCurrentCLK6502;
};
void CAY8910::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 CAY8910::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 0
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;
#endif
#if 0
/* 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 );
#endif
sound_channels = 3; // 3 mono channels: ABC
// hz = ( float ) machine_current->timings.processor_speed /
// machine_current->timings.tstates_per_frame;
hz = 50;
// sound_generator_freq =
// settings_current.sound_hifi ? HIFI_FREQ : settings_current.sound_freq;
sound_generator_freq = SPKR_SAMPLE_RATE;
sound_generator_framesiz = sound_generator_freq / (int)hz;
#if 0
if( ( sound_buf = (libspectrum_signed_word*) 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;
}
#endif
// sound_framesiz = ( float ) settings_current.sound_freq / hz;
sound_framesiz = sound_generator_freq / (int)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.
*/
#if 0
sound_oldpos[0] = sound_oldpos[1] = -1;
sound_fillpos[0] = sound_fillpos[1] = 0;
#endif
/* 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 0
if( first_init ) {
first_init = 0;
for( f = 0; f < 2; f++ )
sound_oldval[f] = sound_oldval_orig[f] = 0;
}
#endif
#if 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;
}
#endif
#if 0
ay_tick_incr = ( int ) ( 65536. *
libspectrum_timings_ay_speed( machine_current->
machine ) /
sound_generator_freq );
#endif
ay_tick_incr = ( int ) ( 65536. * m_fCurrentCLK_AY8910 / sound_generator_freq ); // [TC]
}
#if 0
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 );
}
#endif
void CAY8910::sound_end( void )
{
#if 0
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;
}
#endif
#if 0
if( sound_buf ) {
free( sound_buf );
sound_buf = NULL;
}
#endif
}
#if 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;
}
#endif
/* 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 )
#if 0
/* 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 ); \
}
#endif
/* 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
void CAY8910::sound_ay_overlay( void )
{
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;
cpufreq = (libspectrum_dword) (m_fCurrentCLK_AY8910 / HZ_COMMON_DENOMINATOR); // [TC]
for( f = 0; f < ay_change_count; f++ )
ay_change[f].ofs = (USHORT) (( ay_change[f].tstates * sfreq ) / cpufreq); // [TC] Added cast
libspectrum_signed_word* pBuf1 = g_ppSoundBuffers[0];
libspectrum_signed_word* pBuf2 = g_ppSoundBuffers[1];
libspectrum_signed_word* pBuf3 = g_ppSoundBuffers[2];
// for( f = 0, ptr = sound_buf; f < sound_generator_framesiz; f++ ) {
for( f = 0; 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) */
*pBuf1++ = chan1; // [TC]
*pBuf2++ = chan2; // [TC]
*pBuf3++ = chan3; // [TC]
#if 0
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;
}
}
#endif
/* 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;
}
}
}
// AppleWin:TC Holding down ScrollLock will result in lots of AY changes /ay_change_count/
// - since sound_ay_overlay() is called to consume them.
/* don't make the change immediately; record it for later,
* to be made by sound_frame() (via sound_ay_overlay()).
*/
void CAY8910::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 CAY8910::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;
}
#if 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 )
#endif
#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 CAY8910::sound_frame( void )
{
#if 0
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;
}
#endif
/* 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 */
#if 0
sound_lowlevel_frame( sound_buf,
sound_generator_framesiz * sound_channels );
#endif
#if 0
sound_oldpos[0] = sound_oldpos[1] = -1;
sound_fillpos[0] = sound_fillpos[1] = 0;
#endif
ay_change_count = 0;
}
#if 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;
}
#endif
///////////////////////////////////////////////////////////////////////////////
// AY8910 interface
#include "CPU.h" // For g_nCumulativeCycles
static CAY8910 g_AY8910[MAX_8910];
static unsigned __int64 g_uLastCumulativeCycles = 0;
void _AYWriteReg(int chip, int r, int v)
{
libspectrum_dword uOffset = (libspectrum_dword) (g_nCumulativeCycles - g_uLastCumulativeCycles);
g_AY8910[chip].sound_ay_write(r, v, uOffset);
}
void AY8910_reset(int chip)
{
// Don't reset the AY CLK, as this is a property of the card (MB/Phasor), not the AY chip
g_AY8910[chip].sound_ay_reset(); // Calls: sound_ay_init();
}
void AY8910UpdateSetCycles()
{
g_uLastCumulativeCycles = g_nCumulativeCycles;
}
void AY8910Update(int chip, INT16** buffer, int nNumSamples)
{
AY8910UpdateSetCycles();
sound_generator_framesiz = nNumSamples;
g_ppSoundBuffers = buffer;
g_AY8910[chip].sound_frame();
}
void AY8910_InitAll(int nClock, int nSampleRate)
{
for (UINT i=0; i<MAX_8910; i++)
{
g_AY8910[i].sound_init(NULL); // Inits mainly static members (except ay_tick_incr)
g_AY8910[i].sound_ay_init();
}
}
void AY8910_InitClock(int nClock)
{
CAY8910::SetCLK( (double)nClock );
for (UINT i=0; i<MAX_8910; i++)
{
g_AY8910[i].sound_init(NULL); // ay_tick_incr is dependent on AY_CLK
}
}
BYTE* AY8910_GetRegsPtr(UINT uChip)
{
if(uChip >= MAX_8910)
return NULL;
return g_AY8910[uChip].GetAYRegsPtr();
}
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