static void sound_ay_overlay(CAY8910 *_this)
{
int tone_level[3];
int mixer, envshape;
int f, g, level, count;
// libspectrum_signed_word *ptr;
struct ay_change_tag *change_ptr = _this->ay_change;
int changes_left = _this->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 < _this->ay_change_count; f++ )
_this->ay_change[f].ofs = (uint16_t) (( _this->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 ) {
_this->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 */
_this->ay_tone_period[r] = ( _this->sound_ay_registers[ reg & ~1 ] |
( _this->sound_ay_registers[ reg | 1 ] & 15 ) << 8 );
if( !_this->ay_tone_period[r] )
_this->ay_tone_period[r]++;
/* important to get this right, otherwise e.g. Ghouls 'n' Ghosts
* has really scratchy, horrible-sounding vibrato.
*/
if( _this->ay_tone_tick[r] >= _this->ay_tone_period[r] * 2 )
_this->ay_tone_tick[r] %= _this->ay_tone_period[r] * 2;
break;
case 6:
_this->ay_noise_tick = 0;
_this->ay_noise_period = ( _this->sound_ay_registers[ reg ] & 31 );
break;
case 11:
case 12:
/* this one *isn't* fixed-point */
_this->ay_env_period =
_this->sound_ay_registers[11] | ( _this->sound_ay_registers[12] << 8 );
break;
case 13:
_this->ay_env_internal_tick = _this->ay_env_tick = _this->ay_env_subcycles = 0;
_this->env_first = 1;
_this->env_rev = 0;
_this->env_counter = ( _this->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[ _this->sound_ay_registers[ 8 + g ] & 15 ];
/* envelope */
envshape = _this->sound_ay_registers[13];
level = ay_tone_levels[ _this->env_counter ];
for( g = 0; g < 3; g++ )
if( _this->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.
*/
_this->ay_env_subcycles += _this->ay_tick_incr;
noise_count = 0;
while( _this->ay_env_subcycles >= ( 16 << 16 ) ) {
_this->ay_env_subcycles -= ( 16 << 16 );
noise_count++;
_this->ay_env_tick++;
while( _this->ay_env_tick >= _this->ay_env_period ) {
_this->ay_env_tick -= _this->ay_env_period;
/* do a 1/16th-of-period incr/decr if needed */
if( _this->env_first ||
( ( envshape & AY_ENV_CONT ) && !( envshape & AY_ENV_HOLD ) ) ) {
if( _this->env_rev )
_this->env_counter -= ( envshape & AY_ENV_ATTACK ) ? 1 : -1;
else
_this->env_counter += ( envshape & AY_ENV_ATTACK ) ? 1 : -1;
if( _this->env_counter < 0 )
_this->env_counter = 0;
if( _this->env_counter > 15 )
_this->env_counter = 15;
}
_this->ay_env_internal_tick++;
while( _this->ay_env_internal_tick >= 16 ) {
_this->ay_env_internal_tick -= 16;
/* end of cycle */
if( !( envshape & AY_ENV_CONT ) )
_this->env_counter = 0;
else {
if( envshape & AY_ENV_HOLD ) {
if( _this->env_first && ( envshape & AY_ENV_ALT ) )
_this->env_counter = ( _this->env_counter ? 0 : 15 );
} else {
/* non-hold */
if( envshape & AY_ENV_ALT )
_this->env_rev = !_this->env_rev;
else
_this->env_counter = ( envshape & AY_ENV_ATTACK ) ? 0 : 15;
}
}
_this->env_first = 0;
}
/* don't keep trying if period is zero */
if( !_this->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 = _this->sound_ay_registers[7];
_this->ay_tone_subcycles += _this->ay_tick_incr;
tone_count = _this->ay_tone_subcycles >> ( 3 + 16 );
_this->ay_tone_subcycles &= ( 8 << 16 ) - 1;
if( ( mixer & 1 ) == 0 ) {
level = chan1;
AY_DO_TONE( chan1, 0 );
}
if( ( mixer & 0x08 ) == 0 && _this->noise_toggle )
chan1 = 0;
if( ( mixer & 2 ) == 0 ) {
level = chan2;
AY_DO_TONE( chan2, 1 );
}
if( ( mixer & 0x10 ) == 0 && _this->noise_toggle )
chan2 = 0;
if( ( mixer & 4 ) == 0 ) {
level = chan3;
AY_DO_TONE( chan3, 2 );
}
if( ( mixer & 0x20 ) == 0 && _this->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 */
_this->ay_noise_tick += noise_count;
while( _this->ay_noise_tick >= _this->ay_noise_period ) {
_this->ay_noise_tick -= _this->ay_noise_period;
if( ( _this->rng & 1 ) ^ ( ( _this->rng & 2 ) ? 1 : 0 ) )
_this->noise_toggle = !_this->noise_toggle;
/* rng is 17-bit shift reg, bit 0 is output.
* input is bit 0 xor bit 2.
*/
_this->rng |= ( ( _this->rng & 1 ) ^ ( ( _this->rng & 4 ) ? 1 : 0 ) ) ? 0x20000 : 0;
_this->rng >>= 1;
/* don't keep trying if period is zero */
if( !_this->ay_noise_period )
break;
}
}
}
void sound_ay_overlay(void)
{
signed short *ptr;
int tone_level[3];
int mixer,envshape;
int f,g,level,count;
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;
/* 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 */
for(f=0;f<ay_change_count;f++)
ay_change[f].ofs=(ay_change[f].tstates*sound_freq)/
(tsmax*50);
for(f=0,ptr=sound_buf;f<sound_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]<<1)
ay_tone_tick[r]%=ay_tone_period[r]<<1;
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;
//Seleuco: cuidado si ponemos pseudostereo es el codigo de abajo.
/* write the sample(s) */
//(*ptr++)+=chan1+chan2+chan3;
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;
}
}
}
