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get rid of -fno-common
[gpsp.git] / sound.c
CommitLineData
2823a4c8 1/* gameplaySP
2 *
3 * Copyright (C) 2006 Exophase <exophase@gmail.com>
4 *
5 * This program is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU General Public License as
7 * published by the Free Software Foundation; either version 2 of
8 * the License, or (at your option) any later version.
9 *
10 * This program is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * General Public License for more details.
14 *
15 * You should have received a copy of the GNU General Public License
16 * along with this program; if not, write to the Free Software
17 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
18 */
19
20
21#include "common.h"
22#include <SDL.h>
23u32 global_enable_audio = 1;
24
25direct_sound_struct direct_sound_channel[2];
26gbc_sound_struct gbc_sound_channel[4];
27
2823a4c8 28u32 sound_frequency = 44100;
2823a4c8 29
2823a4c8 30SDL_mutex *sound_mutex;
bbba3209 31static SDL_cond *sound_cv;
2823a4c8 32
2455b6a3 33#ifdef PSP_BUILD
2823a4c8 34u32 audio_buffer_size_number = 1;
2455b6a3 35#else
36u32 audio_buffer_size_number = 8;
2823a4c8 37#endif
38
bbba3209 39u32 sound_on;
40static u32 audio_buffer_size;
41static s16 sound_buffer[BUFFER_SIZE];
42static u32 sound_buffer_base;
2823a4c8 43
bbba3209 44static u32 sound_last_cpu_ticks;
45static fixed16_16 gbc_sound_tick_step;
2823a4c8 46
bbba3209 47static u32 sound_exit_flag;
2455b6a3 48
2823a4c8 49// Queue 1, 2, or 4 samples to the top of the DS FIFO, wrap around circularly
50
51#define sound_timer_queue(size, value) \
52 *((s##size *)(ds->fifo + ds->fifo_top)) = value; \
53 ds->fifo_top = (ds->fifo_top + 1) % 32; \
54
55void sound_timer_queue8(u32 channel, u8 value)
56{
57 direct_sound_struct *ds = direct_sound_channel + channel;
58 sound_timer_queue(8, value);
59}
60
61void sound_timer_queue16(u32 channel, u16 value)
62{
63 direct_sound_struct *ds = direct_sound_channel + channel;
64 sound_timer_queue(8, value & 0xFF);
65 sound_timer_queue(8, value >> 8);
66}
67
68void sound_timer_queue32(u32 channel, u32 value)
69{
70 direct_sound_struct *ds = direct_sound_channel + channel;
71
72 sound_timer_queue(8, value & 0xFF);
73 sound_timer_queue(8, (value >> 8) & 0xFF);
74 sound_timer_queue(8, (value >> 16) & 0xFF);
75 sound_timer_queue(8, value >> 24);
76}
77
78// Unqueue 1 sample from the base of the DS FIFO and place it on the audio
79// buffer for as many samples as necessary. If the DS FIFO is 16 bytes or
80// smaller and if DMA is enabled for the sound channel initiate a DMA transfer
81// to the DS FIFO.
82
83#define render_sample_null() \
84
85#define render_sample_left() \
86 sound_buffer[buffer_index] += current_sample + \
87 fp16_16_to_u32((next_sample - current_sample) * fifo_fractional) \
88
89#define render_sample_right() \
90 sound_buffer[buffer_index + 1] += current_sample + \
91 fp16_16_to_u32((next_sample - current_sample) * fifo_fractional) \
92
93#define render_sample_both() \
94 dest_sample = current_sample + \
95 fp16_16_to_u32((next_sample - current_sample) * fifo_fractional); \
96 sound_buffer[buffer_index] += dest_sample; \
97 sound_buffer[buffer_index + 1] += dest_sample \
98
99#define render_samples(type) \
100 while(fifo_fractional <= 0xFFFF) \
101 { \
102 render_sample_##type(); \
103 fifo_fractional += frequency_step; \
104 buffer_index = (buffer_index + 2) % BUFFER_SIZE; \
105 } \
106
107void sound_timer(fixed16_16 frequency_step, u32 channel)
108{
109 direct_sound_struct *ds = direct_sound_channel + channel;
110
111 fixed16_16 fifo_fractional = ds->fifo_fractional;
112 u32 buffer_index = ds->buffer_index;
113 s16 current_sample, next_sample, dest_sample;
114
115 current_sample = ds->fifo[ds->fifo_base] << 4;
116 ds->fifo_base = (ds->fifo_base + 1) % 32;
117 next_sample = ds->fifo[ds->fifo_base] << 4;
118
119 if(sound_on == 1)
120 {
121 if(ds->volume == DIRECT_SOUND_VOLUME_50)
122 {
123 current_sample >>= 1;
124 next_sample >>= 1;
125 }
126
127 switch(ds->status)
128 {
129 case DIRECT_SOUND_INACTIVE:
130 render_samples(null);
131 break;
132
133 case DIRECT_SOUND_RIGHT:
134 render_samples(right);
135 break;
136
137 case DIRECT_SOUND_LEFT:
138 render_samples(left);
139 break;
140
141 case DIRECT_SOUND_LEFTRIGHT:
142 render_samples(both);
143 break;
144 }
145 }
146 else
147 {
148 render_samples(null);
149 }
150
151 ds->buffer_index = buffer_index;
152 ds->fifo_fractional = fp16_16_fractional_part(fifo_fractional);
153
154 if(((ds->fifo_top - ds->fifo_base) % 32) <= 16)
155 {
156 if(dma[1].direct_sound_channel == channel)
157 dma_transfer(dma + 1);
158
159 if(dma[2].direct_sound_channel == channel)
160 dma_transfer(dma + 2);
161 }
162}
163
164void sound_reset_fifo(u32 channel)
165{
166 direct_sound_struct *ds = direct_sound_channel;
167
168 memset(ds->fifo, 0, 32);
169}
170
171// Initial pattern data = 4bits (signed)
172// Channel volume = 12bits
173// Envelope volume = 14bits
174// Master volume = 2bits
175
176// Recalculate left and right volume as volume changes.
177// To calculate the current sample, use (sample * volume) >> 16
178
179// Square waves range from -8 (low) to 7 (high)
180
181s8 square_pattern_duty[4][8] =
182{
183 { 0xF8, 0xF8, 0xF8, 0xF8, 0x07, 0xF8, 0xF8, 0xF8 },
184 { 0xF8, 0xF8, 0xF8, 0xF8, 0x07, 0x07, 0xF8, 0xF8 },
185 { 0xF8, 0xF8, 0x07, 0x07, 0x07, 0x07, 0xF8, 0xF8 },
186 { 0x07, 0x07, 0x07, 0x07, 0xF8, 0xF8, 0x07, 0x07 },
187};
188
189s8 wave_samples[64];
190
191u32 noise_table15[1024];
192u32 noise_table7[4];
193
194u32 gbc_sound_master_volume_table[4] = { 1, 2, 4, 0 };
195
196u32 gbc_sound_channel_volume_table[8] =
197{
198 fixed_div(0, 7, 12),
199 fixed_div(1, 7, 12),
200 fixed_div(2, 7, 12),
201 fixed_div(3, 7, 12),
202 fixed_div(4, 7, 12),
203 fixed_div(5, 7, 12),
204 fixed_div(6, 7, 12),
205 fixed_div(7, 7, 12)
206};
207
208u32 gbc_sound_envelope_volume_table[16] =
209{
210 fixed_div(0, 15, 14),
211 fixed_div(1, 15, 14),
212 fixed_div(2, 15, 14),
213 fixed_div(3, 15, 14),
214 fixed_div(4, 15, 14),
215 fixed_div(5, 15, 14),
216 fixed_div(6, 15, 14),
217 fixed_div(7, 15, 14),
218 fixed_div(8, 15, 14),
219 fixed_div(9, 15, 14),
220 fixed_div(10, 15, 14),
221 fixed_div(11, 15, 14),
222 fixed_div(12, 15, 14),
223 fixed_div(13, 15, 14),
224 fixed_div(14, 15, 14),
225 fixed_div(15, 15, 14)
226};
227
228u32 gbc_sound_buffer_index = 0;
229u32 gbc_sound_last_cpu_ticks = 0;
230u32 gbc_sound_partial_ticks = 0;
231
232u32 gbc_sound_master_volume_left;
233u32 gbc_sound_master_volume_right;
234u32 gbc_sound_master_volume;
235
236#define update_volume_channel_envelope(channel) \
237 volume_##channel = gbc_sound_envelope_volume_table[envelope_volume] * \
238 gbc_sound_channel_volume_table[gbc_sound_master_volume_##channel] * \
239 gbc_sound_master_volume_table[gbc_sound_master_volume] \
240
241#define update_volume_channel_noenvelope(channel) \
242 volume_##channel = gs->wave_volume * \
243 gbc_sound_channel_volume_table[gbc_sound_master_volume_##channel] * \
244 gbc_sound_master_volume_table[gbc_sound_master_volume] \
245
246#define update_volume(type) \
247 update_volume_channel_##type(left); \
248 update_volume_channel_##type(right) \
249
250#define update_tone_sweep() \
251 if(gs->sweep_status) \
252 { \
253 u32 sweep_ticks = gs->sweep_ticks - 1; \
254 \
255 if(sweep_ticks == 0) \
256 { \
257 u32 rate = gs->rate; \
258 \
259 if(gs->sweep_direction) \
260 rate = rate - (rate >> gs->sweep_shift); \
261 else \
262 rate = rate + (rate >> gs->sweep_shift); \
263 \
264 if(rate > 2048) \
265 rate = 2048; \
266 \
267 frequency_step = float_to_fp16_16(((131072.0 / (2048 - rate)) * 8.0) / \
268 sound_frequency); \
269 \
270 gs->frequency_step = frequency_step; \
271 gs->rate = rate; \
272 \
273 sweep_ticks = gs->sweep_initial_ticks; \
274 } \
275 gs->sweep_ticks = sweep_ticks; \
276 } \
277
278#define update_tone_nosweep() \
279
280#define update_tone_envelope() \
281 if(gs->envelope_status) \
282 { \
283 u32 envelope_ticks = gs->envelope_ticks - 1; \
284 envelope_volume = gs->envelope_volume; \
285 \
286 if(envelope_ticks == 0) \
287 { \
288 if(gs->envelope_direction) \
289 { \
290 if(envelope_volume != 15) \
291 envelope_volume = gs->envelope_volume + 1; \
292 } \
293 else \
294 { \
295 if(envelope_volume != 0) \
296 envelope_volume = gs->envelope_volume - 1; \
297 } \
298 \
299 update_volume(envelope); \
300 \
301 gs->envelope_volume = envelope_volume; \
302 gs->envelope_ticks = gs->envelope_initial_ticks; \
303 } \
304 else \
305 { \
306 gs->envelope_ticks = envelope_ticks; \
307 } \
308 } \
309
310#define update_tone_noenvelope() \
311
312#define gbc_sound_synchronize() \
313 while(((gbc_sound_buffer_index - sound_buffer_base) % BUFFER_SIZE) > \
314 (audio_buffer_size * 2)) \
315 { \
316 SDL_CondWait(sound_cv, sound_mutex); \
317 } \
318
319#define update_tone_counters(envelope_op, sweep_op) \
320 tick_counter += gbc_sound_tick_step; \
321 if(tick_counter > 0xFFFF) \
322 { \
323 if(gs->length_status) \
324 { \
325 u32 length_ticks = gs->length_ticks - 1; \
326 gs->length_ticks = length_ticks; \
327 \
328 if(length_ticks == 0) \
329 { \
330 gs->active_flag = 0; \
331 break; \
332 } \
333 } \
334 \
335 update_tone_##envelope_op(); \
336 update_tone_##sweep_op(); \
337 \
338 tick_counter &= 0xFFFF; \
339 } \
340
341#define gbc_sound_render_sample_right() \
342 sound_buffer[buffer_index + 1] += (current_sample * volume_right) >> 22 \
343
344#define gbc_sound_render_sample_left() \
345 sound_buffer[buffer_index] += (current_sample * volume_left) >> 22 \
346
347#define gbc_sound_render_sample_both() \
348 gbc_sound_render_sample_right(); \
349 gbc_sound_render_sample_left() \
350
351#define gbc_sound_render_samples(type, sample_length, envelope_op, sweep_op) \
352 for(i = 0; i < buffer_ticks; i++) \
353 { \
354 current_sample = \
355 sample_data[fp16_16_to_u32(sample_index) % sample_length]; \
356 gbc_sound_render_sample_##type(); \
357 \
358 sample_index += frequency_step; \
359 buffer_index = (buffer_index + 2) % BUFFER_SIZE; \
360 \
361 update_tone_counters(envelope_op, sweep_op); \
362 } \
363
364#define gbc_noise_wrap_full 32767
365
366#define gbc_noise_wrap_half 126
367
368#define get_noise_sample_full() \
369 current_sample = \
370 ((s32)(noise_table15[fp16_16_to_u32(sample_index) >> 5] << \
371 (fp16_16_to_u32(sample_index) & 0x1F)) >> 31) & 0x0F \
372
373#define get_noise_sample_half() \
374 current_sample = \
375 ((s32)(noise_table7[fp16_16_to_u32(sample_index) >> 5] << \
376 (fp16_16_to_u32(sample_index) & 0x1F)) >> 31) & 0x0F \
377
378#define gbc_sound_render_noise(type, noise_type, envelope_op, sweep_op) \
379 for(i = 0; i < buffer_ticks; i++) \
380 { \
381 get_noise_sample_##noise_type(); \
382 gbc_sound_render_sample_##type(); \
383 \
384 sample_index += frequency_step; \
385 \
386 if(sample_index >= u32_to_fp16_16(gbc_noise_wrap_##noise_type)) \
387 sample_index -= u32_to_fp16_16(gbc_noise_wrap_##noise_type); \
388 \
389 buffer_index = (buffer_index + 2) % BUFFER_SIZE; \
390 update_tone_counters(envelope_op, sweep_op); \
391 } \
392
393#define gbc_sound_render_channel(type, sample_length, envelope_op, sweep_op) \
394 buffer_index = gbc_sound_buffer_index; \
395 sample_index = gs->sample_index; \
396 frequency_step = gs->frequency_step; \
397 tick_counter = gs->tick_counter; \
398 \
399 update_volume(envelope_op); \
400 \
401 switch(gs->status) \
402 { \
403 case GBC_SOUND_INACTIVE: \
404 break; \
405 \
406 case GBC_SOUND_LEFT: \
407 gbc_sound_render_##type(left, sample_length, envelope_op, sweep_op); \
408 break; \
409 \
410 case GBC_SOUND_RIGHT: \
411 gbc_sound_render_##type(right, sample_length, envelope_op, sweep_op); \
412 break; \
413 \
414 case GBC_SOUND_LEFTRIGHT: \
415 gbc_sound_render_##type(both, sample_length, envelope_op, sweep_op); \
416 break; \
417 } \
418 \
419 gs->sample_index = sample_index; \
420 gs->tick_counter = tick_counter; \
421
422#define gbc_sound_load_wave_ram(bank) \
423 wave_bank = wave_samples + (bank * 32); \
424 for(i = 0, i2 = 0; i < 16; i++, i2 += 2) \
425 { \
426 current_sample = wave_ram[i]; \
427 wave_bank[i2] = (((current_sample >> 4) & 0x0F) - 8); \
428 wave_bank[i2 + 1] = ((current_sample & 0x0F) - 8); \
429 } \
430
431void synchronize_sound()
432{
433 SDL_LockMutex(sound_mutex);
434
435 gbc_sound_synchronize();
436
437 SDL_UnlockMutex(sound_mutex);
438}
439
440void update_gbc_sound(u32 cpu_ticks)
441{
442 fixed16_16 buffer_ticks = float_to_fp16_16(((float)(cpu_ticks -
443 gbc_sound_last_cpu_ticks) * sound_frequency) / 16777216.0);
444 u32 i, i2;
445 gbc_sound_struct *gs = gbc_sound_channel;
446 fixed16_16 sample_index, frequency_step;
447 fixed16_16 tick_counter;
448 u32 buffer_index;
449 s32 volume_left, volume_right;
450 u32 envelope_volume;
451 s32 current_sample;
452 u32 sound_status = address16(io_registers, 0x84) & 0xFFF0;
453 s8 *sample_data;
454 s8 *wave_bank;
455 u8 *wave_ram = ((u8 *)io_registers) + 0x90;
456
457 gbc_sound_partial_ticks += fp16_16_fractional_part(buffer_ticks);
458 buffer_ticks = fp16_16_to_u32(buffer_ticks);
459
460 if(gbc_sound_partial_ticks > 0xFFFF)
461 {
462 buffer_ticks += 1;
463 gbc_sound_partial_ticks &= 0xFFFF;
464 }
465
466 SDL_LockMutex(sound_mutex);
467 if(synchronize_flag)
468 {
469 if(((gbc_sound_buffer_index - sound_buffer_base) % BUFFER_SIZE) >
470 (audio_buffer_size * 3 / 2))
471 {
472 while(((gbc_sound_buffer_index - sound_buffer_base) % BUFFER_SIZE) >
473 (audio_buffer_size * 3 / 2))
474 {
475 SDL_CondWait(sound_cv, sound_mutex);
476 }
477
478#ifdef PSP_BUILD
479 if(current_frameskip_type == auto_frameskip)
480 {
481 sceDisplayWaitVblankStart();
482 real_frame_count = 0;
483 virtual_frame_count = 0;
484 }
2455b6a3 485#else
2823a4c8 486 if(current_frameskip_type == auto_frameskip)
487 {
2455b6a3 488/*
2823a4c8 489 u64 current_ticks;
490 u64 next_ticks;
491 get_ticks_us(&current_ticks);
492
493 next_ticks = ((current_ticks + 16666) / 16667) * 16667;
494 delay_us(next_ticks - current_ticks);
495
496 get_ticks_us(&frame_count_initial_timestamp);
2455b6a3 497*/
498 /* prevent frameskip, or it will cause more audio,
499 * then more waiting here, then frame skip again, ... */
500 num_skipped_frames = 100;
2823a4c8 501 }
502#endif
503
2823a4c8 504 }
505 }
506 if(sound_on == 1)
507 {
508 gs = gbc_sound_channel + 0;
509 if(gs->active_flag)
510 {
511 sound_status |= 0x01;
512 sample_data = gs->sample_data;
513 envelope_volume = gs->envelope_volume;
514 gbc_sound_render_channel(samples, 8, envelope, sweep);
515 }
516
517 gs = gbc_sound_channel + 1;
518 if(gs->active_flag)
519 {
520 sound_status |= 0x02;
521 sample_data = gs->sample_data;
522 envelope_volume = gs->envelope_volume;
523 gbc_sound_render_channel(samples, 8, envelope, nosweep);
524 }
525
526 gs = gbc_sound_channel + 2;
527 if(gbc_sound_wave_update)
528 {
529 if(gs->wave_bank == 1)
530 {
531 gbc_sound_load_wave_ram(1);
532 }
533 else
534 {
535 gbc_sound_load_wave_ram(0);
536 }
537
538 gbc_sound_wave_update = 0;
539 }
540
541 if((gs->active_flag) && (gs->master_enable))
542 {
543 sound_status |= 0x04;
544 sample_data = wave_samples;
545 if(gs->wave_type == 0)
546 {
547 if(gs->wave_bank == 1)
548 sample_data += 32;
549
550 gbc_sound_render_channel(samples, 32, noenvelope, nosweep);
551 }
552 else
553 {
554 gbc_sound_render_channel(samples, 64, noenvelope, nosweep);
555 }
556 }
557
558 gs = gbc_sound_channel + 3;
559 if(gs->active_flag)
560 {
561 sound_status |= 0x08;
562 envelope_volume = gs->envelope_volume;
563
564 if(gs->noise_type == 1)
565 {
566 gbc_sound_render_channel(noise, half, envelope, nosweep);
567 }
568 else
569 {
570 gbc_sound_render_channel(noise, full, envelope, nosweep);
571 }
572 }
573 }
574
575 address16(io_registers, 0x84) = sound_status;
576
2823a4c8 577 gbc_sound_last_cpu_ticks = cpu_ticks;
578 gbc_sound_buffer_index =
579 (gbc_sound_buffer_index + (buffer_ticks * 2)) % BUFFER_SIZE;
e9c7b137 580
581 SDL_UnlockMutex(sound_mutex);
582
583 SDL_CondSignal(sound_cv);
2823a4c8 584}
585
586#define sound_copy_normal() \
587 current_sample = source[i] \
588
589#define sound_copy(source_offset, length, render_type) \
590 _length = (length) / 2; \
591 source = (s16 *)(sound_buffer + source_offset); \
592 for(i = 0; i < _length; i++) \
593 { \
594 sound_copy_##render_type(); \
595 if(current_sample > 2047) \
596 current_sample = 2047; \
597 if(current_sample < -2048) \
598 current_sample = -2048; \
599 \
600 stream_base[i] = current_sample << 4; \
601 source[i] = 0; \
602 } \
603
604#define sound_copy_null(source_offset, length) \
605 _length = (length) / 2; \
606 source = (s16 *)(sound_buffer + source_offset); \
607 for(i = 0; i < _length; i++) \
608 { \
609 stream_base[i] = 0; \
610 source[i] = 0; \
611 } \
612
613
614void sound_callback(void *userdata, Uint8 *stream, int length)
615{
616 u32 sample_length = length / 2;
617 u32 _length;
618 u32 i;
619 s16 *stream_base = (s16 *)stream;
620 s16 *source;
621 s32 current_sample;
622
623 SDL_LockMutex(sound_mutex);
624
625 while(((gbc_sound_buffer_index - sound_buffer_base) % BUFFER_SIZE) <
2455b6a3 626 length && !sound_exit_flag)
2823a4c8 627 {
628 SDL_CondWait(sound_cv, sound_mutex);
629 }
630
631 if(global_enable_audio)
632 {
633 if((sound_buffer_base + sample_length) >= BUFFER_SIZE)
634 {
635 u32 partial_length = (BUFFER_SIZE - sound_buffer_base) * 2;
636 sound_copy(sound_buffer_base, partial_length, normal);
637 source = (s16 *)sound_buffer;
638 sound_copy(0, length - partial_length, normal);
639 sound_buffer_base = (length - partial_length) / 2;
640 }
641 else
642 {
643 sound_copy(sound_buffer_base, length, normal);
644 sound_buffer_base += sample_length;
645 }
646 }
647 else
648 {
649 if((sound_buffer_base + sample_length) >= BUFFER_SIZE)
650 {
651 u32 partial_length = (BUFFER_SIZE - sound_buffer_base) * 2;
652 sound_copy_null(sound_buffer_base, partial_length);
653 source = (s16 *)sound_buffer;
654 sound_copy(0, length - partial_length, normal);
655 sound_buffer_base = (length - partial_length) / 2;
656 }
657 else
658 {
659 sound_copy_null(sound_buffer_base, length);
660 sound_buffer_base += sample_length;
661 }
662 }
663
664 SDL_CondSignal(sound_cv);
665
666 SDL_UnlockMutex(sound_mutex);
667}
668
669// Special thanks to blarrg for the LSFR frequency used in Meridian, as posted
670// on the forum at http://meridian.overclocked.org:
671// http://meridian.overclocked.org/cgi-bin/wwwthreads/showpost.pl?Board=merid
672// angeneraldiscussion&Number=2069&page=0&view=expanded&mode=threaded&sb=4
673// Hope you don't mind me borrowing it ^_-
674
675void init_noise_table(u32 *table, u32 period, u32 bit_length)
676{
677 u32 shift_register = 0xFF;
678 u32 mask = ~(1 << bit_length);
679 s32 table_pos, bit_pos;
680 u32 current_entry;
681 u32 table_period = (period + 31) / 32;
682
683 // Bits are stored in reverse order so they can be more easily moved to
684 // bit 31, for sign extended shift down.
685
686 for(table_pos = 0; table_pos < table_period; table_pos++)
687 {
688 current_entry = 0;
689 for(bit_pos = 31; bit_pos >= 0; bit_pos--)
690 {
691 current_entry |= (shift_register & 0x01) << bit_pos;
692
693 shift_register =
694 ((1 & (shift_register ^ (shift_register >> 1))) << bit_length) |
695 ((shift_register >> 1) & mask);
696 }
697
698 table[table_pos] = current_entry;
699 }
700}
701
702void reset_sound()
703{
704 direct_sound_struct *ds = direct_sound_channel;
705 gbc_sound_struct *gs = gbc_sound_channel;
706 u32 i;
707
708 sound_on = 0;
709 sound_buffer_base = 0;
710 sound_last_cpu_ticks = 0;
711 memset(sound_buffer, 0, audio_buffer_size);
712
713 for(i = 0; i < 2; i++, ds++)
714 {
715 ds->buffer_index = 0;
716 ds->status = DIRECT_SOUND_INACTIVE;
717 ds->fifo_top = 0;
718 ds->fifo_base = 0;
719 ds->fifo_fractional = 0;
720 ds->last_cpu_ticks = 0;
721 memset(ds->fifo, 0, 32);
722 }
723
724 gbc_sound_buffer_index = 0;
725 gbc_sound_last_cpu_ticks = 0;
726 gbc_sound_partial_ticks = 0;
727
728 gbc_sound_master_volume_left = 0;
729 gbc_sound_master_volume_right = 0;
730 gbc_sound_master_volume = 0;
731 memset(wave_samples, 0, 64);
732
733 for(i = 0; i < 4; i++, gs++)
734 {
735 gs->status = GBC_SOUND_INACTIVE;
736 gs->sample_data = square_pattern_duty[2];
737 gs->active_flag = 0;
738 }
739}
740
741void sound_exit()
742{
743 gbc_sound_buffer_index =
744 (sound_buffer_base + audio_buffer_size) % BUFFER_SIZE;
745 SDL_PauseAudio(1);
2455b6a3 746 sound_exit_flag = 1;
2823a4c8 747 SDL_CondSignal(sound_cv);
2455b6a3 748 SDL_CloseAudio();
2823a4c8 749}
750
751void init_sound()
752{
bbba3209 753 SDL_AudioSpec sound_settings;
754
2823a4c8 755#ifdef PSP_BUILD
756 audio_buffer_size = (audio_buffer_size_number * 1024) + 3072;
2455b6a3 757#else
2823a4c8 758 audio_buffer_size = 16 << audio_buffer_size_number;
759// audio_buffer_size = 16384;
2823a4c8 760#endif
761
762 SDL_AudioSpec desired_spec =
763 {
764 sound_frequency,
765 AUDIO_S16,
766 2,
767 0,
768 audio_buffer_size / 4,
769 0,
770 0,
771 sound_callback,
772 NULL
773 };
774
775 gbc_sound_tick_step =
776 float_to_fp16_16(256.0 / sound_frequency);
777
778 init_noise_table(noise_table15, 32767, 14);
779 init_noise_table(noise_table7, 127, 6);
780
781 reset_sound();
782
2823a4c8 783 sound_mutex = SDL_CreateMutex();
784 sound_cv = SDL_CreateCond();
2455b6a3 785
786 SDL_OpenAudio(&desired_spec, &sound_settings);
787 sound_frequency = sound_settings.freq;
788 audio_buffer_size = sound_settings.size;
789 u32 i = audio_buffer_size / 16;
790 for (audio_buffer_size_number = 0; i && (i & 1) == 0; i >>= 1)
791 audio_buffer_size_number++;
792#ifndef PSP_BUILD
793 printf("audio: freq %d, size %d\n", sound_frequency, audio_buffer_size);
794#endif
795
2823a4c8 796 SDL_PauseAudio(0);
797}
798
799#define sound_savestate_builder(type) \
800void sound_##type##_savestate(file_tag_type savestate_file) \
801{ \
802 file_##type##_variable(savestate_file, sound_on); \
803 file_##type##_variable(savestate_file, sound_buffer_base); \
804 file_##type##_variable(savestate_file, sound_last_cpu_ticks); \
805 file_##type##_variable(savestate_file, gbc_sound_buffer_index); \
806 file_##type##_variable(savestate_file, gbc_sound_last_cpu_ticks); \
807 file_##type##_variable(savestate_file, gbc_sound_partial_ticks); \
808 file_##type##_variable(savestate_file, gbc_sound_master_volume_left); \
809 file_##type##_variable(savestate_file, gbc_sound_master_volume_right); \
810 file_##type##_variable(savestate_file, gbc_sound_master_volume); \
811 file_##type##_array(savestate_file, wave_samples); \
812 file_##type##_array(savestate_file, direct_sound_channel); \
813 file_##type##_array(savestate_file, gbc_sound_channel); \
814} \
815
816sound_savestate_builder(read);
817sound_savestate_builder(write_mem);
818
gpSP fork tuned for ARM Linux