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1007 lines
27 KiB
C++
1007 lines
27 KiB
C++
/* sound.c: Sound support
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Copyright (c) 2000-2007 Russell Marks, Matan Ziv-Av, Philip Kendall,
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Fredrick Meunier
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This program is free software; you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation; either version 2 of the License, or
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(at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License along
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with this program; if not, write to the Free Software Foundation, Inc.,
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51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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Author contact information:
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E-mail: philip-fuse@shadowmagic.org.uk
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*/
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// [AppleWin-TC] From FUSE's sound.c module
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#include "StdAfx.h"
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#pragma hdrstop
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#include <windows.h>
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#include <stdio.h>
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#include <crtdbg.h>
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#include "AY8910.h"
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#include "Common.h"
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#include "Structs.h"
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#include "Applewin.h" // For g_fh
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#include "Mockingboard.h" // For g_uTimer1IrqCount
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/* The AY white noise RNG algorithm is based on info from MAME's ay8910.c -
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* MAME's licence explicitly permits free use of info (even encourages it).
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*/
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/* NB: I know some of this stuff looks fairly CPU-hogging.
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* For example, the AY code tracks changes with sub-frame timing
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* in a rather hairy way, and there's subsampling here and there.
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* But if you measure the CPU use, it doesn't actually seem
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* very high at all. And I speak as a Cyrix owner. :-)
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*/
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libspectrum_signed_word** g_ppSoundBuffers; // Used to pass param to sound_ay_overlay()
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/* configuration */
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//int sound_enabled = 0; /* Are we currently using the sound card */
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//int sound_enabled_ever = 0; /* if it's *ever* been in use; see
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// sound_ay_write() and sound_ay_reset() */
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//int sound_stereo = 0; /* true for stereo *output sample* (only) */
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//int sound_stereo_ay_abc = 0; /* (AY stereo) true for ABC stereo, else ACB */
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//int sound_stereo_ay_narrow = 0; /* (AY stereo) true for narrow AY st. sep. */
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//int sound_stereo_ay = 0; /* local copy of settings_current.stereo_ay */
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//int sound_stereo_beeper = 0; /* and settings_current.stereo_beeper */
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/* assume all three tone channels together match the beeper volume (ish).
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* Must be <=127 for all channels; 50+2+(24*3) = 124.
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* (Now scaled up for 16-bit.)
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*/
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//#define AMPL_BEEPER ( 50 * 256)
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//#define AMPL_TAPE ( 2 * 256 )
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//#define AMPL_AY_TONE ( 24 * 256 ) /* three of these */
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#define AMPL_AY_TONE ( 42 * 256 ) // 42*3 = 126
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/* max. number of sub-frame AY port writes allowed;
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* given the number of port writes theoretically possible in a
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* 50th I think this should be plenty.
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*/
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//#define AY_CHANGE_MAX 8000 // [TC] Moved into AY8910.h
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///* frequency to generate sound at for hifi sound */
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//#define HIFI_FREQ 88200
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#ifdef HAVE_SAMPLERATE
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static SRC_STATE *src_state;
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#endif /* #ifdef HAVE_SAMPLERATE */
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int sound_generator_framesiz;
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int sound_framesiz;
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static int sound_generator_freq;
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static int sound_channels;
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static unsigned int ay_tone_levels[16];
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//static libspectrum_signed_word *sound_buf, *tape_buf;
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//static float *convert_input_buffer, *convert_output_buffer;
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#if 0
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/* beeper stuff */
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static int sound_oldpos[2], sound_fillpos[2];
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static int sound_oldval[2], sound_oldval_orig[2];
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#endif
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#if 0
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#define STEREO_BUF_SIZE 4096
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static int pstereobuf[ STEREO_BUF_SIZE ];
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static int pstereobufsiz, pstereopos;
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static int psgap = 250;
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static int rstereobuf_l[ STEREO_BUF_SIZE ], rstereobuf_r[ STEREO_BUF_SIZE ];
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static int rstereopos, rchan1pos, rchan2pos, rchan3pos;
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#endif
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// Statics:
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double CAY8910::m_fCurrentCLK_AY8910 = 0.0;
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CAY8910::CAY8910() :
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// Init the statics that were in sound_ay_overlay()
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rng(1),
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noise_toggle(0),
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env_first(1), env_rev(0), env_counter(15)
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{
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m_fCurrentCLK_AY8910 = g_fCurrentCLK6502;
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};
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void CAY8910::sound_ay_init( void )
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{
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/* AY output doesn't match the claimed levels; these levels are based
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* on the measurements posted to comp.sys.sinclair in Dec 2001 by
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* Matthew Westcott, adjusted as I described in a followup to his post,
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* then scaled to 0..0xffff.
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*/
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static const int levels[16] = {
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0x0000, 0x0385, 0x053D, 0x0770,
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0x0AD7, 0x0FD5, 0x15B0, 0x230C,
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0x2B4C, 0x43C1, 0x5A4B, 0x732F,
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0x9204, 0xAFF1, 0xD921, 0xFFFF
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};
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int f;
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/* scale the values down to fit */
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for( f = 0; f < 16; f++ )
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ay_tone_levels[f] = ( levels[f] * AMPL_AY_TONE + 0x8000 ) / 0xffff;
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ay_noise_tick = ay_noise_period = 0;
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ay_env_internal_tick = ay_env_tick = ay_env_period = 0;
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ay_tone_subcycles = ay_env_subcycles = 0;
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for( f = 0; f < 3; f++ )
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ay_tone_tick[f] = ay_tone_high[f] = 0, ay_tone_period[f] = 1;
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ay_change_count = 0;
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}
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void CAY8910::sound_init( const char *device )
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{
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// static int first_init = 1;
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// int f, ret;
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float hz;
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#ifdef HAVE_SAMPLERATE
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int error;
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#endif /* #ifdef HAVE_SAMPLERATE */
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/* if we don't have any sound I/O code compiled in, don't do sound */
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#ifdef NO_SOUND
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return;
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#endif
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#if 0
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if( !( !sound_enabled && settings_current.sound &&
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settings_current.emulation_speed == 100 ) )
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return;
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sound_stereo_ay = settings_current.stereo_ay;
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sound_stereo_beeper = settings_current.stereo_beeper;
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/* only try for stereo if we need it */
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if( sound_stereo_ay || sound_stereo_beeper )
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sound_stereo = 1;
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ret =
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sound_lowlevel_init( device, &settings_current.sound_freq,
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&sound_stereo );
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if( ret )
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return;
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#endif
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#if 0
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/* important to override these settings if not using stereo
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* (it would probably be confusing to mess with the stereo
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* settings in settings_current though, which is why we make copies
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* rather than using the real ones).
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*/
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if( !sound_stereo ) {
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sound_stereo_ay = 0;
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sound_stereo_beeper = 0;
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}
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sound_enabled = sound_enabled_ever = 1;
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sound_channels = ( sound_stereo ? 2 : 1 );
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#endif
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sound_channels = 3; // 3 mono channels: ABC
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// hz = ( float ) machine_current->timings.processor_speed /
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// machine_current->timings.tstates_per_frame;
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hz = 50;
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// sound_generator_freq =
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// settings_current.sound_hifi ? HIFI_FREQ : settings_current.sound_freq;
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sound_generator_freq = SPKR_SAMPLE_RATE;
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sound_generator_framesiz = sound_generator_freq / (int)hz;
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#if 0
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if( ( sound_buf = (libspectrum_signed_word*) malloc( sizeof( libspectrum_signed_word ) *
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sound_generator_framesiz * sound_channels ) ) ==
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NULL
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|| ( tape_buf =
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malloc( sizeof( libspectrum_signed_word ) *
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sound_generator_framesiz ) ) == NULL ) {
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if( sound_buf ) {
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free( sound_buf );
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sound_buf = NULL;
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}
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sound_end();
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return;
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}
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#endif
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// sound_framesiz = ( float ) settings_current.sound_freq / hz;
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sound_framesiz = sound_generator_freq / (int)hz;
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#ifdef HAVE_SAMPLERATE
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if( settings_current.sound_hifi ) {
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if( ( convert_input_buffer = malloc( sizeof( float ) *
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sound_generator_framesiz *
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sound_channels ) ) == NULL
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|| ( convert_output_buffer =
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malloc( sizeof( float ) * sound_framesiz * sound_channels ) ) ==
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NULL ) {
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if( convert_input_buffer ) {
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free( convert_input_buffer );
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convert_input_buffer = NULL;
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}
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sound_end();
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return;
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}
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}
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src_state = src_new( SRC_SINC_MEDIUM_QUALITY, sound_channels, &error );
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if( error ) {
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ui_error( UI_ERROR_ERROR,
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"error initialising sample rate converter %s",
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src_strerror( error ) );
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sound_end();
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return;
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}
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#endif /* #ifdef HAVE_SAMPLERATE */
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/* if we're resuming, we need to be careful about what
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* gets reset. The minimum we can do is the beeper
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* buffer positions, so that's here.
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*/
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#if 0
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sound_oldpos[0] = sound_oldpos[1] = -1;
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sound_fillpos[0] = sound_fillpos[1] = 0;
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#endif
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/* this stuff should only happen on the initial call.
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* (We currently assume the new sample rate will be the
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* same as the previous one, hence no need to recalculate
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* things dependent on that.)
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*/
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#if 0
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if( first_init ) {
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first_init = 0;
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for( f = 0; f < 2; f++ )
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sound_oldval[f] = sound_oldval_orig[f] = 0;
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}
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#endif
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#if 0
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if( sound_stereo_beeper ) {
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for( f = 0; f < STEREO_BUF_SIZE; f++ )
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pstereobuf[f] = 0;
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pstereopos = 0;
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pstereobufsiz = ( sound_generator_freq * psgap ) / 22000;
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}
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if( sound_stereo_ay ) {
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int pos =
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( sound_stereo_ay_narrow ? 3 : 6 ) * sound_generator_freq / 8000;
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for( f = 0; f < STEREO_BUF_SIZE; f++ )
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rstereobuf_l[f] = rstereobuf_r[f] = 0;
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rstereopos = 0;
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/* the actual ACB/ABC bit :-) */
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rchan1pos = -pos;
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if( sound_stereo_ay_abc )
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rchan2pos = 0, rchan3pos = pos;
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else
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rchan2pos = pos, rchan3pos = 0;
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}
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#endif
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#if 0
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ay_tick_incr = ( int ) ( 65536. *
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libspectrum_timings_ay_speed( machine_current->
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machine ) /
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sound_generator_freq );
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#endif
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ay_tick_incr = ( int ) ( 65536. * m_fCurrentCLK_AY8910 / sound_generator_freq ); // [TC]
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}
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#if 0
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void
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sound_pause( void )
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{
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if( sound_enabled )
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sound_end();
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}
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void
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sound_unpause( void )
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{
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/* No sound if fastloading in progress */
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if( settings_current.fastload && tape_is_playing() )
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return;
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sound_init( settings_current.sound_device );
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}
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#endif
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void CAY8910::sound_end( void )
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{
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#if 0
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if( sound_enabled ) {
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if( sound_buf ) {
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free( sound_buf );
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sound_buf = NULL;
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free( tape_buf );
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tape_buf = NULL;
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}
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if( convert_input_buffer ) {
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free( convert_input_buffer );
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convert_input_buffer = NULL;
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}
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if( convert_output_buffer ) {
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free( convert_output_buffer );
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convert_output_buffer = NULL;
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}
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#ifdef HAVE_SAMPLERATE
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if( src_state )
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src_state = src_delete( src_state );
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#endif /* #ifdef HAVE_SAMPLERATE */
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sound_lowlevel_end();
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sound_enabled = 0;
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}
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#endif
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#if 0
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if( sound_buf ) {
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free( sound_buf );
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sound_buf = NULL;
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}
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#endif
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}
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#if 0
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/* write sample to buffer as pseudo-stereo */
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static void
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sound_write_buf_pstereo( libspectrum_signed_word * out, int c )
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{
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int bl = ( c - pstereobuf[ pstereopos ] ) / 2;
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int br = ( c + pstereobuf[ pstereopos ] ) / 2;
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if( bl < -AMPL_BEEPER )
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bl = -AMPL_BEEPER;
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if( br < -AMPL_BEEPER )
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br = -AMPL_BEEPER;
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if( bl > AMPL_BEEPER )
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bl = AMPL_BEEPER;
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if( br > AMPL_BEEPER )
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br = AMPL_BEEPER;
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*out = bl;
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out[1] = br;
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pstereobuf[ pstereopos ] = c;
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pstereopos++;
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if( pstereopos >= pstereobufsiz )
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pstereopos = 0;
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}
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#endif
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/* not great having this as a macro to inline it, but it's only
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* a fairly short routine, and it saves messing about.
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* (XXX ummm, possibly not so true any more :-))
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*/
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#define AY_GET_SUBVAL( chan ) \
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( level * 2 * ay_tone_tick[ chan ] / tone_count )
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#define AY_DO_TONE( var, chan ) \
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( var ) = 0; \
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is_low = 0; \
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if( level ) { \
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if( ay_tone_high[ chan ] ) \
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( var ) = ( level ); \
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else { \
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( var ) = -( level ); \
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is_low = 1; \
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} \
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} \
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\
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ay_tone_tick[ chan ] += tone_count; \
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count = 0; \
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while( ay_tone_tick[ chan ] >= ay_tone_period[ chan ] ) { \
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count++; \
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ay_tone_tick[ chan ] -= ay_tone_period[ chan ]; \
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ay_tone_high[ chan ] = !ay_tone_high[ chan ]; \
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\
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/* has to be here, unfortunately... */ \
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if( count == 1 && level && ay_tone_tick[ chan ] < tone_count ) { \
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if( is_low ) \
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( var ) += AY_GET_SUBVAL( chan ); \
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else \
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( var ) -= AY_GET_SUBVAL( chan ); \
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} \
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} \
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\
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/* if it's changed more than once during the sample, we can't */ \
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/* represent it faithfully. So, just hope it's a sample. */ \
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/* (That said, this should also help avoid aliasing noise.) */ \
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if( count > 1 ) \
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( var ) = -( level )
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#if 0
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/* add val, correctly delayed on either left or right buffer,
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* to add the AY stereo positioning. This doesn't actually put
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* anything directly in sound_buf, though.
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*/
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#define GEN_STEREO( pos, val ) \
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if( ( pos ) < 0 ) { \
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rstereobuf_l[ rstereopos ] += ( val ); \
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rstereobuf_r[ ( rstereopos - pos ) % STEREO_BUF_SIZE ] += ( val ); \
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} else { \
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rstereobuf_l[ ( rstereopos + pos ) % STEREO_BUF_SIZE ] += ( val ); \
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rstereobuf_r[ rstereopos ] += ( val ); \
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}
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#endif
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/* bitmasks for envelope */
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#define AY_ENV_CONT 8
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#define AY_ENV_ATTACK 4
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#define AY_ENV_ALT 2
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#define AY_ENV_HOLD 1
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#define HZ_COMMON_DENOMINATOR 50
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void CAY8910::sound_ay_overlay( void )
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{
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int tone_level[3];
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int mixer, envshape;
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int f, g, level, count;
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// libspectrum_signed_word *ptr;
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struct ay_change_tag *change_ptr = ay_change;
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int changes_left = ay_change_count;
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int reg, r;
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int is_low;
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int chan1, chan2, chan3;
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unsigned int tone_count, noise_count;
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libspectrum_dword sfreq, cpufreq;
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///* If no AY chip, don't produce any AY sound (!) */
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// if( !machine_current->capabilities & LIBSPECTRUM_MACHINE_CAPABILITY_AY )
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// return;
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/* convert change times to sample offsets, use common denominator of 50 to
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avoid overflowing a dword */
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sfreq = sound_generator_freq / HZ_COMMON_DENOMINATOR;
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// cpufreq = machine_current->timings.processor_speed / HZ_COMMON_DENOMINATOR;
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cpufreq = (libspectrum_dword) (m_fCurrentCLK_AY8910 / HZ_COMMON_DENOMINATOR); // [TC]
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for( f = 0; f < ay_change_count; f++ )
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ay_change[f].ofs = (USHORT) (( ay_change[f].tstates * sfreq ) / cpufreq); // [TC] Added cast
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libspectrum_signed_word* pBuf1 = g_ppSoundBuffers[0];
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libspectrum_signed_word* pBuf2 = g_ppSoundBuffers[1];
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libspectrum_signed_word* pBuf3 = g_ppSoundBuffers[2];
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// for( f = 0, ptr = sound_buf; f < sound_generator_framesiz; f++ ) {
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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();
|
|
}
|