[Fuse-for-macosx-commits] SF.net SVN: fuse-for-macosx:[636] vendor/fuse-emulator/current/fuse
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[Fuse-for-macosx-commits] SF.net SVN: fuse-for-macosx:[636] vendor/fuse-emulator/current/fuse
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From: <fr...@us...> - 2010-05-24 11:59:24
|
Revision: 636
http://fuse-for-macosx.svn.sourceforge.net/fuse-for-macosx/?rev=636&view=rev
Author: fredm
Date: 2010-05-24 11:59:18 +0000 (Mon, 24 May 2010)
Log Message:
-----------
To prepare to load . into vendor/fuse-emulator/current, perform 1
rename.
* vendor/fuse-emulator/current/fuse/sound.c: Renamed from
vendor/fuse-emulator/current/fuse/sound.cpp.
Added Paths:
-----------
vendor/fuse-emulator/current/fuse/sound.c
Removed Paths:
-------------
vendor/fuse-emulator/current/fuse/sound.cpp
Copied: vendor/fuse-emulator/current/fuse/sound.c (from rev 635, vendor/fuse-emulator/current/fuse/sound.cpp)
===================================================================
--- vendor/fuse-emulator/current/fuse/sound.c (rev 0)
+++ vendor/fuse-emulator/current/fuse/sound.c 2010-05-24 11:59:18 UTC (rev 636)
@@ -0,0 +1,652 @@
+/* sound.cpp: Sound support
+ Copyright (c) 2000-2009 Russell Marks, Matan Ziv-Av, Philip Kendall,
+ Fredrick Meunier
+
+ $Id: sound.cpp 4036 2009-06-13 09:57:55Z fredm $
+
+ 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: phi...@sh...
+
+*/
+
+/* 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).
+ */
+
+extern "C" {
+#include "fuse.h"
+#include "machine.h"
+#include "options.h"
+#include "settings.h"
+#include "sound.h"
+#include "tape.h"
+#include "ui/ui.h"
+}
+
+#include "sound/Blip_Buffer.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 = 0; /* local copy of settings_current.stereo_ay */
+
+/* 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 ( 5 * 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
+
+int sound_freq;
+int sound_framesiz;
+
+static int sound_channels;
+
+static unsigned int ay_tone_levels[16];
+
+static unsigned int ay_tone_tick[3], ay_tone_high[3], ay_noise_tick;
+static unsigned int ay_tone_cycles, ay_env_cycles;
+static unsigned int ay_env_internal_tick, ay_env_tick;
+static unsigned int ay_tone_period[3], ay_noise_period, ay_env_period;
+
+/* Local copy of the AY registers */
+static libspectrum_byte sound_ay_registers[16];
+
+struct ay_change_tag
+{
+ libspectrum_dword tstates;
+ unsigned char reg, val;
+};
+
+static struct ay_change_tag ay_change[ AY_CHANGE_MAX ];
+static int ay_change_count;
+
+Blip_Buffer *left_buf;
+Blip_Buffer *right_buf;
+blip_sample_t *samples;
+
+typedef Blip_Synth<blip_good_quality, blip_unscaled> fuse_synth;
+fuse_synth *left_beeper_synth, *right_beeper_synth;
+
+fuse_synth *ay_a_synth, *ay_b_synth, *ay_c_synth;
+fuse_synth *ay_a_synth_r, *ay_b_synth_r, *ay_c_synth_r;
+
+struct speaker_type_tag
+{
+ int bass;
+ double treble;
+};
+
+static struct speaker_type_tag speaker_type[] =
+ { { 200, -47.0 }, { 1000, -67.0 } };
+
+static double
+sound_get_volume( int volume )
+{
+ if( volume < 0 ) volume = 0;
+ else if( volume > 100 ) volume = 100;
+
+ return volume / 100.0;
+}
+
+/* Returns the emulation speed adjusted processor speed */
+libspectrum_dword
+sound_get_effective_processor_speed( void )
+{
+ return machine_current->timings.processor_speed / 100 *
+ settings_current.emulation_speed;
+}
+
+bool
+sound_init_blip( Blip_Buffer **buf, fuse_synth **synth )
+{
+ *buf = new Blip_Buffer();
+ (*buf)->clock_rate( sound_get_effective_processor_speed() );
+ /* Allow up to 1s of playback buffer - this allows us to cope with slowing
+ down to 2% of speed where a single Speccy frame generates just under 1s
+ of sound */
+ if ( (*buf)->set_sample_rate( settings_current.sound_freq, 1000 ) ) {
+ sound_end();
+ ui_error( UI_ERROR_ERROR, "out of memory at %s:%d", __FILE__, __LINE__ );
+ return false;
+ }
+
+ *synth = new fuse_synth();
+
+ (*synth)->volume( sound_get_volume( settings_current.volume_beeper ) );
+ (*synth)->output( *buf );
+
+ (*buf)->bass_freq( speaker_type[ option_enumerate_sound_speaker_type() ].bass );
+ (*synth)->treble_eq( speaker_type[ option_enumerate_sound_speaker_type() ].treble );
+
+ return true;
+}
+
+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_cycles = ay_env_cycles = 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 )
+{
+ int ret;
+ float hz;
+
+ /* Allow sound as long as emulation speed is greater than 2%
+ (less than that and a single Speccy frame generates more
+ than a seconds worth of sound which is bigger than the
+ maximum Blip_Buffer of 1 second) */
+ if( !( !sound_enabled && settings_current.sound &&
+ settings_current.emulation_speed > 1 ) )
+ return;
+
+ sound_stereo_ay = settings_current.stereo_ay;
+
+ /* only try for stereo if we need it */
+ if( sound_stereo_ay )
+ sound_stereo = 1;
+
+ ret =
+ sound_lowlevel_init( device, &settings_current.sound_freq, &sound_stereo );
+
+ if( ret )
+ return;
+
+ if( !sound_init_blip(&left_buf, &left_beeper_synth) ) return;
+ if( sound_stereo && !sound_init_blip(&right_buf, &right_beeper_synth) ) return;
+
+ ay_a_synth = new fuse_synth();
+ ay_a_synth->volume( sound_get_volume( settings_current.volume_ay) );
+ ay_a_synth->output( left_buf );
+ ay_a_synth->treble_eq( speaker_type[ option_enumerate_sound_speaker_type() ].treble );
+
+ ay_b_synth = new fuse_synth();
+ ay_b_synth->volume( sound_get_volume( settings_current.volume_ay) );
+ ay_b_synth->treble_eq( speaker_type[ option_enumerate_sound_speaker_type() ].treble );
+
+ ay_c_synth = new fuse_synth();
+ ay_c_synth->volume( sound_get_volume( settings_current.volume_ay) );
+ ay_c_synth->output( left_buf );
+ ay_c_synth->treble_eq( speaker_type[ option_enumerate_sound_speaker_type() ].treble );
+
+ /* 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;
+ }
+
+ ay_a_synth_r = 0;
+ ay_b_synth_r = 0;
+ ay_c_synth_r = 0;
+
+ if( sound_stereo ) {
+ ay_c_synth_r = new fuse_synth();
+ ay_c_synth_r->volume( sound_get_volume( settings_current.volume_ay ) );
+ ay_c_synth_r->output( right_buf );
+
+ if( sound_stereo_ay ) {
+ /* stereo with ACB stereo. */
+ ay_b_synth->output( right_buf );
+ } else {
+ ay_a_synth_r = new fuse_synth();
+ ay_a_synth_r->volume( sound_get_volume( settings_current.volume_ay ) );
+ ay_a_synth_r->output( right_buf );
+ ay_a_synth_r->treble_eq( speaker_type[ option_enumerate_sound_speaker_type() ].treble );
+
+ ay_b_synth->output( left_buf );
+
+ ay_b_synth_r = new fuse_synth();
+ ay_b_synth_r->volume( sound_get_volume( settings_current.volume_ay ) );
+ ay_b_synth_r->output( right_buf );
+ ay_b_synth_r->treble_eq( speaker_type[ option_enumerate_sound_speaker_type() ].treble );
+ }
+ } else {
+ ay_b_synth->output( left_buf );
+ }
+
+ sound_enabled = sound_enabled_ever = 1;
+
+ sound_channels = ( sound_stereo ? 2 : 1 );
+
+ /* Adjust relative processor speed to deal with adjusting sound generation
+ frequency against emulation speed (more flexible than adjusting generated
+ sample rate) */
+ hz = ( float )sound_get_effective_processor_speed() /
+ machine_current->timings.tstates_per_frame;
+
+ /* Size of audio data we will get from running a single Spectrum frame */
+ sound_framesiz = ( float )settings_current.sound_freq / hz;
+ sound_framesiz++;
+
+ samples = (blip_sample_t *)calloc( sound_framesiz * sound_channels,
+ sizeof(blip_sample_t) );
+}
+
+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 ) {
+ delete left_beeper_synth; left_beeper_synth = 0;
+ delete left_buf; left_buf = 0;
+ delete right_beeper_synth; right_beeper_synth = 0;
+ delete right_buf; right_buf = 0;
+
+ delete ay_a_synth; ay_a_synth = 0;
+ delete ay_b_synth; ay_b_synth = 0;
+ delete ay_c_synth; ay_c_synth = 0;
+ delete ay_a_synth_r; ay_a_synth_r = 0;
+ delete ay_b_synth_r; ay_b_synth_r = 0;
+ delete ay_c_synth_r; ay_c_synth_r = 0;
+
+ sound_lowlevel_end();
+ free( samples );
+ sound_enabled = 0;
+ }
+}
+
+static inline void
+ay_do_tone( int level, unsigned int tone_count, int& var, int chan )
+{
+ var = 0;
+
+ ay_tone_tick[ chan ] += tone_count;
+
+ if( ay_tone_tick[ chan ] >= ay_tone_period[ chan ] ) {
+ ay_tone_tick[ chan ] -= ay_tone_period[ chan ];
+ ay_tone_high[ chan ] = !ay_tone_high[ chan ];
+ }
+
+ if( level ) {
+ if( ay_tone_high[ chan ] )
+ var = level;
+ else {
+ var = -level;
+ }
+ }
+
+ /* The AY output goes from 0 to the maximum volume, so there
+ * is a DC component that is half the maxmum volume.
+ * Robocop uses a high frequency square wave with a tone
+ * period of one to average out to being like a DC offset at
+ * around half the maximum volume. This is used as a base for
+ * the sample playback.
+ * This seems to intefere with our attempt to remove the
+ * returned DC offset, so for now we just ignore the high
+ * frequency wave and hope it's a sample
+ */
+ if( ay_tone_period[ chan ] == 1 ) {
+ var = -level;
+ }
+}
+
+/* bitmasks for envelope */
+#define AY_ENV_CONT 8
+#define AY_ENV_ATTACK 4
+#define AY_ENV_ALT 2
+#define AY_ENV_HOLD 1
+
+/* the AY steps down the external clock by 16 for tone and noise
+ generators */
+#define AY_CLOCK_DIVISOR 16
+/* all Spectrum models and clones with an AY seem to count down the
+ master clock by 2 to drive the AY */
+#define AY_CLOCK_RATIO 2
+
+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 g, level;
+ libspectrum_dword f;
+ struct ay_change_tag *change_ptr = ay_change;
+ int changes_left = ay_change_count;
+ int reg, r;
+ int chan1, chan2, chan3;
+ int last_chan1 = 0, last_chan2 = 0, last_chan3 = 0;
+ unsigned int tone_count, noise_count;
+
+ /* If no AY chip, don't produce any AY sound (!) */
+ if( !( periph_fuller_active || periph_melodik_active ||
+ machine_current->capabilities & LIBSPECTRUM_MACHINE_CAPABILITY_AY ) )
+ return;
+
+ for( f = 0; f < machine_current->timings.tstates_per_frame;
+ f+= AY_CLOCK_DIVISOR * AY_CLOCK_RATIO ) {
+ /* update ay registers. */
+ while( changes_left && f >= change_ptr->tstates ) {
+ 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:
+ ay_env_period =
+ sound_ay_registers[11] | ( sound_ay_registers[12] << 8 );
+ break;
+ case 13:
+ ay_env_internal_tick = ay_env_tick = ay_env_cycles = 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. */
+ ay_env_cycles += AY_CLOCK_DIVISOR;
+ noise_count = 0;
+ while( ay_env_cycles >= 16 ) {
+ ay_env_cycles -= 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_cycles += AY_CLOCK_DIVISOR;
+ tone_count = ay_tone_cycles >> 3;
+ ay_tone_cycles &= 7;
+
+ if( ( mixer & 1 ) == 0 ) {
+ level = chan1;
+ ay_do_tone( level, tone_count, chan1, 0 );
+ }
+ if( ( mixer & 0x08 ) == 0 && noise_toggle )
+ chan1 = 0;
+
+ if( ( mixer & 2 ) == 0 ) {
+ level = chan2;
+ ay_do_tone( level, tone_count, chan2, 1 );
+ }
+ if( ( mixer & 0x10 ) == 0 && noise_toggle )
+ chan2 = 0;
+
+ if( ( mixer & 4 ) == 0 ) {
+ level = chan3;
+ ay_do_tone( level, tone_count, chan3, 2 );
+ }
+ if( ( mixer & 0x20 ) == 0 && noise_toggle )
+ chan3 = 0;
+
+ if( last_chan1 != chan1 ) {
+ ay_a_synth->update( f, chan1 );
+ if( ay_a_synth_r ) ay_a_synth_r->update( f, chan1 );
+ last_chan1 = chan1;
+ }
+ if( last_chan2 != chan2 ) {
+ ay_b_synth->update( f, chan2 );
+ if( ay_b_synth_r ) ay_b_synth_r->update( f, chan2 );
+ last_chan2 = chan2;
+ }
+ if( last_chan3 != chan3 ) {
+ ay_c_synth->update( f, chan3 );
+ if( ay_c_synth_r ) ay_c_synth_r->update( f, chan3 );
+ last_chan3 = chan3;
+ }
+
+ /* 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_cycles = ay_env_cycles = 0;
+}
+
+void
+sound_frame( void )
+{
+ long count;
+
+ if( !sound_enabled )
+ return;
+
+ /* overlay AY sound */
+ sound_ay_overlay();
+
+ left_buf->end_frame( machine_current->timings.tstates_per_frame );
+
+ if( sound_stereo ) {
+ right_buf->end_frame( machine_current->timings.tstates_per_frame );
+
+ /* Read left channel into even samples, right channel into odd samples:
+ LRLRLRLRLR... */
+ count = left_buf->read_samples( samples, sound_framesiz, 1 );
+ right_buf->read_samples( samples + 1, count, 1 );
+ count <<= 1;
+ } else {
+ count = left_buf->read_samples( samples, sound_framesiz );
+ }
+
+ sound_lowlevel_frame( samples, count );
+
+ ay_change_count = 0;
+}
+
+void
+sound_beeper( int on )
+{
+ static int beeper_ampl[] = { 0, AMPL_TAPE, AMPL_BEEPER, AMPL_BEEPER+AMPL_TAPE };
+
+ int val;
+ int ampl;
+
+ if( !sound_enabled )
+ return;
+
+ /* Timex machines have no loading noise */
+ if( tape_is_playing() &&
+ ( !settings_current.sound_load || machine_current->timex ) )
+ on = on & 0x02;
+
+ ampl = beeper_ampl[on];
+
+ val = -beeper_ampl[3] + ampl*2;
+
+ left_beeper_synth->update( tstates, val );
+ if( sound_stereo ) {
+ right_beeper_synth->update( tstates, val );
+ }
+}
Deleted: vendor/fuse-emulator/current/fuse/sound.cpp
===================================================================
--- vendor/fuse-emulator/current/fuse/sound.cpp 2010-04-06 11:24:38 UTC (rev 635)
+++ vendor/fuse-emulator/current/fuse/sound.cpp 2010-05-24 11:59:18 UTC (rev 636)
@@ -1,652 +0,0 @@
-/* sound.cpp: Sound support
- Copyright (c) 2000-2009 Russell Marks, Matan Ziv-Av, Philip Kendall,
- Fredrick Meunier
-
- $Id: sound.cpp 4036 2009-06-13 09:57:55Z fredm $
-
- 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: phi...@sh...
-
-*/
-
-/* 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).
- */
-
-extern "C" {
-#include "fuse.h"
-#include "machine.h"
-#include "options.h"
-#include "settings.h"
-#include "sound.h"
-#include "tape.h"
-#include "ui/ui.h"
-}
-
-#include "sound/Blip_Buffer.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 = 0; /* local copy of settings_current.stereo_ay */
-
-/* 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 ( 5 * 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
-
-int sound_freq;
-int sound_framesiz;
-
-static int sound_channels;
-
-static unsigned int ay_tone_levels[16];
-
-static unsigned int ay_tone_tick[3], ay_tone_high[3], ay_noise_tick;
-static unsigned int ay_tone_cycles, ay_env_cycles;
-static unsigned int ay_env_internal_tick, ay_env_tick;
-static unsigned int ay_tone_period[3], ay_noise_period, ay_env_period;
-
-/* Local copy of the AY registers */
-static libspectrum_byte sound_ay_registers[16];
-
-struct ay_change_tag
-{
- libspectrum_dword tstates;
- unsigned char reg, val;
-};
-
-static struct ay_change_tag ay_change[ AY_CHANGE_MAX ];
-static int ay_change_count;
-
-Blip_Buffer *left_buf;
-Blip_Buffer *right_buf;
-blip_sample_t *samples;
-
-typedef Blip_Synth<blip_good_quality, blip_unscaled> fuse_synth;
-fuse_synth *left_beeper_synth, *right_beeper_synth;
-
-fuse_synth *ay_a_synth, *ay_b_synth, *ay_c_synth;
-fuse_synth *ay_a_synth_r, *ay_b_synth_r, *ay_c_synth_r;
-
-struct speaker_type_tag
-{
- int bass;
- double treble;
-};
-
-static struct speaker_type_tag speaker_type[] =
- { { 200, -47.0 }, { 1000, -67.0 } };
-
-static double
-sound_get_volume( int volume )
-{
- if( volume < 0 ) volume = 0;
- else if( volume > 100 ) volume = 100;
-
- return volume / 100.0;
-}
-
-/* Returns the emulation speed adjusted processor speed */
-libspectrum_dword
-sound_get_effective_processor_speed( void )
-{
- return machine_current->timings.processor_speed / 100 *
- settings_current.emulation_speed;
-}
-
-bool
-sound_init_blip( Blip_Buffer **buf, fuse_synth **synth )
-{
- *buf = new Blip_Buffer();
- (*buf)->clock_rate( sound_get_effective_processor_speed() );
- /* Allow up to 1s of playback buffer - this allows us to cope with slowing
- down to 2% of speed where a single Speccy frame generates just under 1s
- of sound */
- if ( (*buf)->set_sample_rate( settings_current.sound_freq, 1000 ) ) {
- sound_end();
- ui_error( UI_ERROR_ERROR, "out of memory at %s:%d", __FILE__, __LINE__ );
- return false;
- }
-
- *synth = new fuse_synth();
-
- (*synth)->volume( sound_get_volume( settings_current.volume_beeper ) );
- (*synth)->output( *buf );
-
- (*buf)->bass_freq( speaker_type[ option_enumerate_sound_speaker_type() ].bass );
- (*synth)->treble_eq( speaker_type[ option_enumerate_sound_speaker_type() ].treble );
-
- return true;
-}
-
-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_cycles = ay_env_cycles = 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 )
-{
- int ret;
- float hz;
-
- /* Allow sound as long as emulation speed is greater than 2%
- (less than that and a single Speccy frame generates more
- than a seconds worth of sound which is bigger than the
- maximum Blip_Buffer of 1 second) */
- if( !( !sound_enabled && settings_current.sound &&
- settings_current.emulation_speed > 1 ) )
- return;
-
- sound_stereo_ay = settings_current.stereo_ay;
-
- /* only try for stereo if we need it */
- if( sound_stereo_ay )
- sound_stereo = 1;
-
- ret =
- sound_lowlevel_init( device, &settings_current.sound_freq, &sound_stereo );
-
- if( ret )
- return;
-
- if( !sound_init_blip(&left_buf, &left_beeper_synth) ) return;
- if( sound_stereo && !sound_init_blip(&right_buf, &right_beeper_synth) ) return;
-
- ay_a_synth = new fuse_synth();
- ay_a_synth->volume( sound_get_volume( settings_current.volume_ay) );
- ay_a_synth->output( left_buf );
- ay_a_synth->treble_eq( speaker_type[ option_enumerate_sound_speaker_type() ].treble );
-
- ay_b_synth = new fuse_synth();
- ay_b_synth->volume( sound_get_volume( settings_current.volume_ay) );
- ay_b_synth->treble_eq( speaker_type[ option_enumerate_sound_speaker_type() ].treble );
-
- ay_c_synth = new fuse_synth();
- ay_c_synth->volume( sound_get_volume( settings_current.volume_ay) );
- ay_c_synth->output( left_buf );
- ay_c_synth->treble_eq( speaker_type[ option_enumerate_sound_speaker_type() ].treble );
-
- /* 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;
- }
-
- ay_a_synth_r = 0;
- ay_b_synth_r = 0;
- ay_c_synth_r = 0;
-
- if( sound_stereo ) {
- ay_c_synth_r = new fuse_synth();
- ay_c_synth_r->volume( sound_get_volume( settings_current.volume_ay ) );
- ay_c_synth_r->output( right_buf );
-
- if( sound_stereo_ay ) {
- /* stereo with ACB stereo. */
- ay_b_synth->output( right_buf );
- } else {
- ay_a_synth_r = new fuse_synth();
- ay_a_synth_r->volume( sound_get_volume( settings_current.volume_ay ) );
- ay_a_synth_r->output( right_buf );
- ay_a_synth_r->treble_eq( speaker_type[ option_enumerate_sound_speaker_type() ].treble );
-
- ay_b_synth->output( left_buf );
-
- ay_b_synth_r = new fuse_synth();
- ay_b_synth_r->volume( sound_get_volume( settings_current.volume_ay ) );
- ay_b_synth_r->output( right_buf );
- ay_b_synth_r->treble_eq( speaker_type[ option_enumerate_sound_speaker_type() ].treble );
- }
- } else {
- ay_b_synth->output( left_buf );
- }
-
- sound_enabled = sound_enabled_ever = 1;
-
- sound_channels = ( sound_stereo ? 2 : 1 );
-
- /* Adjust relative processor speed to deal with adjusting sound generation
- frequency against emulation speed (more flexible than adjusting generated
- sample rate) */
- hz = ( float )sound_get_effective_processor_speed() /
- machine_current->timings.tstates_per_frame;
-
- /* Size of audio data we will get from running a single Spectrum frame */
- sound_framesiz = ( float )settings_current.sound_freq / hz;
- sound_framesiz++;
-
- samples = (blip_sample_t *)calloc( sound_framesiz * sound_channels,
- sizeof(blip_sample_t) );
-}
-
-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 ) {
- delete left_beeper_synth; left_beeper_synth = 0;
- delete left_buf; left_buf = 0;
- delete right_beeper_synth; right_beeper_synth = 0;
- delete right_buf; right_buf = 0;
-
- delete ay_a_synth; ay_a_synth = 0;
- delete ay_b_synth; ay_b_synth = 0;
- delete ay_c_synth; ay_c_synth = 0;
- delete ay_a_synth_r; ay_a_synth_r = 0;
- delete ay_b_synth_r; ay_b_synth_r = 0;
- delete ay_c_synth_r; ay_c_synth_r = 0;
-
- sound_lowlevel_end();
- free( samples );
- sound_enabled = 0;
- }
-}
-
-static inline void
-ay_do_tone( int level, unsigned int tone_count, int& var, int chan )
-{
- var = 0;
-
- ay_tone_tick[ chan ] += tone_count;
-
- if( ay_tone_tick[ chan ] >= ay_tone_period[ chan ] ) {
- ay_tone_tick[ chan ] -= ay_tone_period[ chan ];
- ay_tone_high[ chan ] = !ay_tone_high[ chan ];
- }
-
- if( level ) {
- if( ay_tone_high[ chan ] )
- var = level;
- else {
- var = -level;
- }
- }
-
- /* The AY output goes from 0 to the maximum volume, so there
- * is a DC component that is half the maxmum volume.
- * Robocop uses a high frequency square wave with a tone
- * period of one to average out to being like a DC offset at
- * around half the maximum volume. This is used as a base for
- * the sample playback.
- * This seems to intefere with our attempt to remove the
- * returned DC offset, so for now we just ignore the high
- * frequency wave and hope it's a sample
- */
- if( ay_tone_period[ chan ] == 1 ) {
- var = -level;
- }
-}
-
-/* bitmasks for envelope */
-#define AY_ENV_CONT 8
-#define AY_ENV_ATTACK 4
-#define AY_ENV_ALT 2
-#define AY_ENV_HOLD 1
-
-/* the AY steps down the external clock by 16 for tone and noise
- generators */
-#define AY_CLOCK_DIVISOR 16
-/* all Spectrum models and clones with an AY seem to count down the
- master clock by 2 to drive the AY */
-#define AY_CLOCK_RATIO 2
-
-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 g, level;
- libspectrum_dword f;
- struct ay_change_tag *change_ptr = ay_change;
- int changes_left = ay_change_count;
- int reg, r;
- int chan1, chan2, chan3;
- int last_chan1 = 0, last_chan2 = 0, last_chan3 = 0;
- unsigned int tone_count, noise_count;
-
- /* If no AY chip, don't produce any AY sound (!) */
- if( !( periph_fuller_active || periph_melodik_active ||
- machine_current->capabilities & LIBSPECTRUM_MACHINE_CAPABILITY_AY ) )
- return;
-
- for( f = 0; f < machine_current->timings.tstates_per_frame;
- f+= AY_CLOCK_DIVISOR * AY_CLOCK_RATIO ) {
- /* update ay registers. */
- while( changes_left && f >= change_ptr->tstates ) {
- 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:
- ay_env_period =
- sound_ay_registers[11] | ( sound_ay_registers[12] << 8 );
- break;
- case 13:
- ay_env_internal_tick = ay_env_tick = ay_env_cycles = 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. */
- ay_env_cycles += AY_CLOCK_DIVISOR;
- noise_count = 0;
- while( ay_env_cycles >= 16 ) {
- ay_env_cycles -= 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_cycles += AY_CLOCK_DIVISOR;
- tone_count = ay_tone_cycles >> 3;
- ay_tone_cycles &= 7;
-
- if( ( mixer & 1 ) == 0 ) {
- level = chan1;
- ay_do_tone( level, tone_count, chan1, 0 );
- }
- if( ( mixer & 0x08 ) == 0 && noise_toggle )
- chan1 = 0;
-
- if( ( mixer & 2 ) == 0 ) {
- level = chan2;
- ay_do_tone( level, tone_count, chan2, 1 );
- }
- if( ( mixer & 0x10 ) == 0 && noise_toggle )
- chan2 = 0;
-
- if( ( mixer & 4 ) == 0 ) {
- level = chan3;
- ay_do_tone( level, tone_count, chan3, 2 );
- }
- if( ( mixer & 0x20 ) == 0 && noise_toggle )
- chan3 = 0;
-
- if( last_chan1 != chan1 ) {
- ay_a_synth->update( f, chan1 );
- if( ay_a_synth_r ) ay_a_synth_r->update( f, chan1 );
- last_chan1 = chan1;
- }
- if( last_chan2 != chan2 ) {
- ay_b_synth->update( f, chan2 );
- if( ay_b_synth_r ) ay_b_synth_r->update( f, chan2 );
- last_chan2 = chan2;
- }
- if( last_chan3 != chan3 ) {
- ay_c_synth->update( f, chan3 );
- if( ay_c_synth_r ) ay_c_synth_r->update( f, chan3 );
- last_chan3 = chan3;
- }
-
- /* 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_cycles = ay_env_cycles = 0;
-}
-
-void
-sound_frame( void )
-{
- long count;
-
- if( !sound_enabled )
- return;
-
- /* overlay AY sound */
- sound_ay_overlay();
-
- left_buf->end_frame( machine_current->timings.tstates_per_frame );
-
- if( sound_stereo ) {
- right_buf->end_frame( machine_current->timings.tstates_per_frame );
-
- /* Read left channel into even samples, right channel into odd samples:
- LRLRLRLRLR... */
- count = left_buf->read_samples( samples, sound_framesiz, 1 );
- right_buf->read_samples( samples + 1, count, 1 );
- count <<= 1;
- } else {
- count = left_buf->read_samples( samples, sound_framesiz );
- }
-
- sound_lowlevel_frame( samples, count );
-
- ay_change_count = 0;
-}
-
-void
-sound_beeper( int on )
-{
- static int beeper_ampl[] = { 0, AMPL_TAPE, AMPL_BEEPER, AMPL_BEEPER+AMPL_TAPE };
-
- int val;
- int ampl;
-
- if( !sound_enabled )
- return;
-
- /* Timex machines have no loading noise */
- if( tape_is_playing() &&
- ( !settings_current.sound_load || machine_current->timex ) )
- on = on & 0x02;
-
- ampl = beeper_ampl[on];
-
- val = -beeper_ampl[3] + ampl*2;
-
- left_beeper_synth->update( tstates, val );
- if( sound_stereo ) {
- right_beeper_synth->update( tstates, val );
- }
-}
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