void
sound_frame( void )
{
long count;
if( !sound_enabled )
return;
/* overlay AY sound */
sound_ay_overlay();
blip_buffer_end_frame( left_buf, machine_current->timings.tstates_per_frame );
if( sound_stereo_ay != SOUND_STEREO_AY_NONE ) {
blip_buffer_end_frame( right_buf, machine_current->timings.tstates_per_frame );
/* Read left channel into even samples, right channel into odd samples:
LRLRLRLRLR... */
count = blip_buffer_read_samples( left_buf, samples, sound_framesiz, 1 );
blip_buffer_read_samples( right_buf, samples + 1, count, 1 );
count <<= 1;
} else {
count = blip_buffer_read_samples( left_buf, samples, sound_framesiz, BLIP_BUFFER_DEF_STEREO );
}
if( settings_current.sound )
sound_lowlevel_frame( samples, count );
if( movie_recording )
movie_add_sound( samples, count );
ay_change_count = 0;
}
void sound_frame(void)
{
unsigned char *ptr;
int f;
if(!sound_enabled) return;
//if(zx81.vsyncsound)
{
ptr=sound_buf+(sound_stereo?sound_fillpos*2:sound_fillpos);
for(f=sound_fillpos;f<sound_framesiz;f++)
{
BEEPER_OLDVAL_ADJUST;
*ptr++=sound_oldval;
if(sound_stereo)
*ptr++=sound_oldval;
}
}
//else
/* must be AY then, so `zero' buffer ready for it */
// memset(sound_buf,128,sound_framesiz*(sound_stereo+1));
if(zx81.aysound )
sound_ay_overlay();
sound_lowlevel_frame(sound_buf,sound_framesiz*(sound_stereo+1));
SoundOP_Update(sound_buf,sound_framesiz*(sound_stereo+1));
sound_oldpos=-1;
sound_fillpos=0;
sound_ptr=sound_buf;
ay_change_count=0;
}
void sound_frame( CAY8910 *_this )
{
#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(_this);
#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
_this->ay_change_count = 0;
}
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(_this);
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;
}
}
