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add dsp and optional game page
[vg.git] / vg_audio.h
1 /* Copyright (C) 2021-2023 Harry Godden (hgn) - All Rights Reserved */
2
3 #ifndef VG_AUDIO_H
4 #define VG_AUDIO_H
5
6 #define VG_GAME
7
8 #include "vg/vg.h"
9 #include "vg/vg_stdint.h"
10 #include "vg/vg_platform.h"
11 #include "vg/vg_io.h"
12 #include "vg/vg_m.h"
13 #include "vg/vg_console.h"
14 #include "vg/vg_store.h"
15 #include "vg/vg_profiler.h"
16 #include "vg/vg_audio_synth_bird.h"
17
18 #ifdef __GNUC__
19 #ifndef __clang__
20 #pragma GCC push_options
21 #pragma GCC optimize ("O3")
22 #pragma GCC diagnostic push
23 #pragma GCC diagnostic ignored "-Wdeprecated-declarations"
24 #endif
25 #endif
26
27 #define STB_VORBIS_MAX_CHANNELS 2
28 #include "submodules/stb/stb_vorbis.c"
29 #undef L
30 #undef R
31 #undef C
32
33 #ifdef __GNUC__
34 #ifndef __clang__
35 #pragma GCC pop_options
36 #pragma GCC diagnostic pop
37 #endif
38 #endif
39
40 #define AUDIO_FRAME_SIZE 512
41 #define AUDIO_MIX_FRAME_SIZE 256
42
43 #define AUDIO_CHANNELS 32
44 #define AUDIO_LFOS 8
45 #define AUDIO_FILTERS 16
46 #define AUDIO_FLAG_LOOP 0x1
47 #define AUDIO_FLAG_NO_DOPPLER 0x2
48 #define AUDIO_FLAG_SPACIAL_3D 0x4
49 #define AUDIO_FLAG_AUTO_START 0x8
50 #define AUDIO_FLAG_FORMAT 0x1E00
51
52 enum audio_format
53 {
54 k_audio_format_mono = 0x000u,
55 k_audio_format_stereo = 0x200u,
56 k_audio_format_vorbis = 0x400u,
57 k_audio_format_none0 = 0x600u,
58 k_audio_format_none1 = 0x800u,
59 k_audio_format_none2 = 0xA00u,
60 k_audio_format_none3 = 0xC00u,
61 k_audio_format_none4 = 0xE00u,
62
63 k_audio_format_bird = 0x1000u,
64 k_audio_format_gen = 0x1200u,
65 k_audio_format_none6 = 0x1400u,
66 k_audio_format_none7 = 0x1600u,
67 k_audio_format_none8 = 0x1800u,
68 k_audio_format_none9 = 0x1A00u,
69 k_audio_format_none10 = 0x1C00u,
70 k_audio_format_none11 = 0x1E00u,
71 };
72
73 #define AUDIO_DECODE_SIZE (1024*256) /* 256 kb decoding buffers */
74 #define AUDIO_MUTE_VOLUME 0.0f
75 #define AUDIO_BASE_VOLUME 1.0f
76
77 typedef struct audio_clip audio_clip;
78 typedef struct audio_channel audio_channel;
79 typedef struct audio_lfo audio_lfo;
80
81 struct audio_clip{
82 union { /* TODO oof.. */
83 u64 _p64_;
84 const char *path;
85 void *func;
86 };
87
88 u32 flags;
89 u32 size;
90
91 union{
92 u64 _p64;
93 void *data;
94 };
95 };
96
97 struct vg_audio_system{
98 SDL_AudioDeviceID sdl_output_device;
99 vg_str device_choice; /* buffer is null? use default from OS */
100
101 void *audio_pool,
102 *decode_buffer;
103 u32 samples_last;
104
105 /* synchro */
106 int sync_locked;
107
108 SDL_SpinLock sl_checker,
109 sl_sync;
110
111 struct audio_lfo{
112 u32 time, time_startframe;
113 float sqrt_polynomial_coefficient;
114
115 struct{
116 enum lfo_wave_type{
117 k_lfo_triangle,
118 k_lfo_square,
119 k_lfo_saw,
120 k_lfo_polynomial_bipolar
121 }
122 wave_type;
123
124 u32 period;
125 float polynomial_coefficient;
126 }
127 _, editable_state;
128 u32 editble_state_write_mask;
129 }
130 oscillators[ AUDIO_LFOS ];
131
132 struct audio_channel{
133 int allocated;
134 u16 group;
135 u8 world_id;
136
137 char name[32]; /* only editable while allocated == 0 */
138 audio_clip *source; /* ... */
139 u32 flags; /* ... */
140 u32 colour; /* ... */
141
142 /* internal non-readable state
143 * -----------------------------*/
144 u32 cursor, source_length;
145
146 float volume_movement_start,
147 pan_movement_start;
148
149 u32 volume_movement,
150 pan_movement;
151
152 union{
153 struct synth_bird *bird_handle;
154 stb_vorbis *vorbis_handle;
155 };
156
157 stb_vorbis_alloc vorbis_alloc;
158
159 enum channel_activity{
160 k_channel_activity_reset, /* will advance if allocated==1, to wake */
161 k_channel_activity_wake, /* will advance to either of next two */
162 k_channel_activity_alive,
163 k_channel_activity_end,
164 k_channel_activity_error
165 }
166 activity,
167 readable_activity;
168
169 /*
170 * editable structure, can be modified inside _lock and _unlock
171 * the edit mask tells which to copy into internal _, or to discard
172 * ----------------------------------------------------------------------
173 */
174 struct channel_state{
175 int relinquished;
176
177 float volume, /* current volume */
178 volume_target, /* target volume */
179 pan,
180 pan_target,
181 sampling_rate;
182
183 u32 volume_rate,
184 pan_rate;
185
186 v4f spacial_falloff; /* xyz, range */
187
188 audio_lfo *lfo;
189 float lfo_amount;
190 }
191 _, editable_state;
192 u32 editble_state_write_mask;
193 }
194 channels[ AUDIO_CHANNELS ];
195
196 int debug_ui, debug_ui_3d, debug_dsp, dsp_enabled;
197
198 v3f internal_listener_pos,
199 internal_listener_ears,
200 internal_listener_velocity,
201
202 external_listener_pos,
203 external_listener_ears,
204 external_lister_velocity;
205
206 float internal_global_volume,
207 external_global_volume;
208 }
209 static vg_audio = { .external_global_volume = 1.0f, .dsp_enabled = 1 };
210
211 #include "vg/vg_audio_dsp.h"
212
213 static struct vg_profile
214 _vg_prof_audio_decode = {.mode = k_profile_mode_accum,
215 .name = "[T2] audio_decode()"},
216 _vg_prof_audio_mix = {.mode = k_profile_mode_accum,
217 .name = "[T2] audio_mix()"},
218 _vg_prof_dsp = {.mode = k_profile_mode_accum,
219 .name = "[T2] dsp_process()"},
220 vg_prof_audio_decode,
221 vg_prof_audio_mix,
222 vg_prof_audio_dsp;
223
224 /*
225 * These functions are called from the main thread and used to prevent bad
226 * access. TODO: They should be no-ops in release builds.
227 */
228 static int audio_lock_checker_load(void)
229 {
230 int value;
231 SDL_AtomicLock( &vg_audio.sl_checker );
232 value = vg_audio.sync_locked;
233 SDL_AtomicUnlock( &vg_audio.sl_checker );
234 return value;
235 }
236
237 static void audio_lock_checker_store( int value )
238 {
239 SDL_AtomicLock( &vg_audio.sl_checker );
240 vg_audio.sync_locked = value;
241 SDL_AtomicUnlock( &vg_audio.sl_checker );
242 }
243
244 static void audio_require_lock(void)
245 {
246 if( audio_lock_checker_load() )
247 return;
248
249 vg_error( "Modifying sound effects systems requires locking\n" );
250 abort();
251 }
252
253 static void audio_lock(void)
254 {
255 SDL_AtomicLock( &vg_audio.sl_sync );
256 audio_lock_checker_store(1);
257 }
258
259 static void audio_unlock(void)
260 {
261 audio_lock_checker_store(0);
262 SDL_AtomicUnlock( &vg_audio.sl_sync );
263 }
264 static void audio_mixer_callback( void *user, u8 *stream, int frame_count );
265
266 static void vg_audio_device_init(void){
267 SDL_AudioSpec spec_desired, spec_got;
268 spec_desired.callback = audio_mixer_callback;
269 spec_desired.channels = 2;
270 spec_desired.format = AUDIO_F32;
271 spec_desired.freq = 44100;
272 spec_desired.padding = 0;
273 spec_desired.samples = AUDIO_FRAME_SIZE;
274 spec_desired.silence = 0;
275 spec_desired.size = 0;
276 spec_desired.userdata = NULL;
277
278 vg_audio.sdl_output_device =
279 SDL_OpenAudioDevice( vg_audio.device_choice.buffer, 0,
280 &spec_desired, &spec_got,0 );
281
282 vg_info( "Start audio device (%u, F32, %u) @%s\n",
283 spec_desired.freq,
284 AUDIO_FRAME_SIZE,
285 vg_audio.device_choice.buffer );
286
287 if( vg_audio.sdl_output_device ){
288 SDL_PauseAudioDevice( vg_audio.sdl_output_device, 0 );
289 vg_success( "Unpaused device %d.\n", vg_audio.sdl_output_device );
290 }
291 else{
292 vg_error(
293 "SDL_OpenAudioDevice failed. Your default audio device must support:\n"
294 " Frequency: 44100 hz\n"
295 " Buffer size: 512\n"
296 " Channels: 2\n"
297 " Format: s16 or f32\n" );
298 }
299 }
300
301 static void vg_audio_register(void){
302 vg_console_reg_var( "debug_audio", &vg_audio.debug_ui,
303 k_var_dtype_i32, VG_VAR_CHEAT );
304 vg_console_reg_var( "debug_dsp", &vg_audio.debug_dsp,
305 k_var_dtype_i32, VG_VAR_CHEAT );
306 vg_console_reg_var( "volume", &vg_audio.external_global_volume,
307 k_var_dtype_f32, VG_VAR_PERSISTENT );
308 vg_console_reg_var( "vg_audio_device", &vg_audio.device_choice,
309 k_var_dtype_str, VG_VAR_PERSISTENT );
310 vg_console_reg_var( "vg_dsp", &vg_audio.dsp_enabled,
311 k_var_dtype_i32, VG_VAR_PERSISTENT );
312 }
313
314 static void vg_audio_init(void){
315 /* allocate memory */
316 /* 32mb fixed */
317 vg_audio.audio_pool =
318 vg_create_linear_allocator( vg_mem.rtmemory, 1024*1024*32,
319 VG_MEMORY_SYSTEM );
320
321 /* fixed */
322 u32 decode_size = AUDIO_DECODE_SIZE * AUDIO_CHANNELS;
323 vg_audio.decode_buffer = vg_linear_alloc( vg_mem.rtmemory, decode_size );
324
325 vg_dsp_init();
326 vg_audio_device_init();
327 }
328
329 static void vg_audio_free(void)
330 {
331 vg_dsp_free();
332 SDL_CloseAudioDevice( vg_audio.sdl_output_device );
333 }
334
335 /*
336 * thread 1
337 */
338
339 #define AUDIO_EDIT_VOLUME_SLOPE 0x1
340 #define AUDIO_EDIT_VOLUME 0x2
341 #define AUDIO_EDIT_LFO_PERIOD 0x4
342 #define AUDIO_EDIT_LFO_WAVE 0x8
343 #define AUDIO_EDIT_LFO_ATTACHMENT 0x10
344 #define AUDIO_EDIT_SPACIAL 0x20
345 #define AUDIO_EDIT_OWNERSHIP 0x40
346 #define AUDIO_EDIT_SAMPLING_RATE 0x80
347
348 static void audio_channel_init( audio_channel *ch, audio_clip *clip,
349 u32 flags ){
350 audio_require_lock();
351 ch->group = 0;
352 ch->world_id = 0;
353 ch->source = clip;
354 ch->flags = flags;
355 ch->colour = 0x00333333;
356
357 if( (ch->source->flags & AUDIO_FLAG_FORMAT) == k_audio_format_bird )
358 strcpy( ch->name, "[array]" );
359 else if( (ch->source->flags & AUDIO_FLAG_FORMAT) == k_audio_format_gen )
360 strcpy( ch->name, "[program]" );
361 else
362 vg_strncpy( clip->path, ch->name, 32, k_strncpy_always_add_null );
363
364 ch->allocated = 1;
365
366 ch->editable_state.relinquished = 0;
367 ch->editable_state.volume = 1.0f;
368 ch->editable_state.volume_target = 1.0f;
369 ch->editable_state.pan = 0.0f;
370 ch->editable_state.pan_target = 0.0f;
371 ch->editable_state.volume_rate = 0;
372 ch->editable_state.pan_rate = 0;
373 v4_copy((v4f){0.0f,0.0f,0.0f,1.0f},ch->editable_state.spacial_falloff);
374 ch->editable_state.lfo = NULL;
375 ch->editable_state.lfo_amount = 0.0f;
376 ch->editable_state.sampling_rate = 1.0f;
377 ch->editble_state_write_mask = 0x00;
378 }
379
380 static void audio_channel_group( audio_channel *ch, u16 group )
381 {
382 audio_require_lock();
383 ch->group = group;
384 ch->colour = (((u32)group * 29986577) & 0x00ffffff) | 0xff000000;
385 }
386
387 static void audio_channel_world( audio_channel *ch, u8 world_id )
388 {
389 audio_require_lock();
390 ch->world_id = world_id;
391 }
392
393 static audio_channel *audio_get_first_idle_channel(void)
394 {
395 audio_require_lock();
396 for( int i=0; i<AUDIO_CHANNELS; i++ ){
397 audio_channel *ch = &vg_audio.channels[i];
398
399 if( !ch->allocated ){
400 return ch;
401 }
402 }
403
404 return NULL;
405 }
406
407 static audio_channel *audio_get_group_idle_channel( u16 group, u32 max_count )
408 {
409 audio_require_lock();
410 u32 count = 0;
411 audio_channel *dest = NULL;
412
413 for( int i=0; i<AUDIO_CHANNELS; i++ ){
414 audio_channel *ch = &vg_audio.channels[i];
415
416 if( ch->allocated ){
417 if( ch->group == group ){
418 count ++;
419 }
420 }
421 else{
422 if( !dest )
423 dest = ch;
424 }
425 }
426
427 if( dest && (count < max_count) ){
428 return dest;
429 }
430
431 return NULL;
432 }
433
434 static audio_channel *audio_get_group_first_active_channel( u16 group )
435 {
436 audio_require_lock();
437 for( int i=0; i<AUDIO_CHANNELS; i++ ){
438 audio_channel *ch = &vg_audio.channels[i];
439 if( ch->allocated && (ch->group == group) )
440 return ch;
441 }
442 return NULL;
443 }
444
445 static int audio_channel_finished( audio_channel *ch )
446 {
447 audio_require_lock();
448 if( ch->readable_activity == k_channel_activity_end )
449 return 1;
450 else
451 return 0;
452 }
453
454 static audio_channel *audio_relinquish_channel( audio_channel *ch )
455 {
456 audio_require_lock();
457 ch->editable_state.relinquished = 1;
458 ch->editble_state_write_mask |= AUDIO_EDIT_OWNERSHIP;
459 return NULL;
460 }
461
462 static void audio_channel_slope_volume( audio_channel *ch, float length,
463 float new_volume )
464 {
465 audio_require_lock();
466 ch->editable_state.volume_target = new_volume;
467 ch->editable_state.volume_rate = length * 44100.0f;
468 ch->editble_state_write_mask |= AUDIO_EDIT_VOLUME_SLOPE;
469 }
470
471 static void audio_channel_set_sampling_rate( audio_channel *ch, float rate )
472 {
473 audio_require_lock();
474 ch->editable_state.sampling_rate = rate;
475 ch->editble_state_write_mask |= AUDIO_EDIT_SAMPLING_RATE;
476 }
477
478 static void audio_channel_edit_volume( audio_channel *ch,
479 float new_volume, int instant )
480 {
481 audio_require_lock();
482 if( instant ){
483 ch->editable_state.volume = new_volume;
484 ch->editble_state_write_mask |= AUDIO_EDIT_VOLUME;
485 }
486 else{
487 audio_channel_slope_volume( ch, 0.05f, new_volume );
488 }
489 }
490
491 static audio_channel *audio_channel_fadeout( audio_channel *ch, float length )
492 {
493 audio_require_lock();
494 audio_channel_slope_volume( ch, length, 0.0f );
495 return audio_relinquish_channel( ch );
496 }
497
498 static void audio_channel_fadein( audio_channel *ch, float length )
499 {
500 audio_require_lock();
501 audio_channel_edit_volume( ch, 0.0f, 1 );
502 audio_channel_slope_volume( ch, length, 1.0f );
503 }
504
505 static audio_channel *audio_channel_crossfade( audio_channel *ch,
506 audio_clip *new_clip,
507 float length, u32 flags )
508 {
509 audio_require_lock();
510 u32 cursor = 0;
511
512 if( ch )
513 ch = audio_channel_fadeout( ch, length );
514
515 audio_channel *replacement = audio_get_first_idle_channel();
516
517 if( replacement ){
518 audio_channel_init( replacement, new_clip, flags );
519 audio_channel_fadein( replacement, length );
520 }
521
522 return replacement;
523 }
524
525 static void audio_channel_sidechain_lfo( audio_channel *ch, int lfo_id,
526 float amount )
527 {
528 audio_require_lock();
529 ch->editable_state.lfo = &vg_audio.oscillators[ lfo_id ];
530 ch->editable_state.lfo_amount = amount;
531 ch->editble_state_write_mask |= AUDIO_EDIT_LFO_ATTACHMENT;
532 }
533
534 static void audio_channel_set_spacial( audio_channel *ch, v3f co, float range )
535 {
536 audio_require_lock();
537 if( ch->flags & AUDIO_FLAG_SPACIAL_3D ){
538 v3_copy( co, ch->editable_state.spacial_falloff );
539
540 if( range == 0.0f )
541 ch->editable_state.spacial_falloff[3] = 1.0f;
542 else
543 ch->editable_state.spacial_falloff[3] = 1.0f/range;
544
545 ch->editble_state_write_mask |= AUDIO_EDIT_SPACIAL;
546 }
547 else{
548 vg_warn( "Tried to set spacialization paramaters for 2D channel (%s)\n",
549 ch->name );
550 }
551 }
552
553 static int audio_oneshot_3d( audio_clip *clip, v3f position,
554 float range, float volume )
555 {
556 audio_require_lock();
557 audio_channel *ch = audio_get_first_idle_channel();
558
559 if( ch ){
560 audio_channel_init( ch, clip, AUDIO_FLAG_SPACIAL_3D );
561 audio_channel_set_spacial( ch, position, range );
562 audio_channel_edit_volume( ch, volume, 1 );
563 ch = audio_relinquish_channel( ch );
564
565 return 1;
566 }
567 else
568 return 0;
569 }
570
571 static int audio_oneshot( audio_clip *clip, float volume, float pan )
572 {
573 audio_require_lock();
574 audio_channel *ch = audio_get_first_idle_channel();
575
576 if( ch ){
577 audio_channel_init( ch, clip, 0x00 );
578 audio_channel_edit_volume( ch, volume, 1 );
579 ch = audio_relinquish_channel( ch );
580
581 return 1;
582 }
583 else
584 return 0;
585 }
586
587 static void audio_set_lfo_wave( int id, enum lfo_wave_type type,
588 float coefficient )
589 {
590 audio_require_lock();
591 audio_lfo *lfo = &vg_audio.oscillators[ id ];
592 lfo->editable_state.polynomial_coefficient = coefficient;
593 lfo->editable_state.wave_type = type;
594
595 lfo->editble_state_write_mask |= AUDIO_EDIT_LFO_WAVE;
596 }
597
598 static void audio_set_lfo_frequency( int id, float freq )
599 {
600 audio_require_lock();
601 audio_lfo *lfo = &vg_audio.oscillators[ id ];
602 lfo->editable_state.period = 44100.0f / freq;
603 lfo->editble_state_write_mask |= AUDIO_EDIT_LFO_PERIOD;
604 }
605
606
607 /*
608 * Committers
609 * -----------------------------------------------------------------------------
610 */
611 static int audio_channel_load_source( audio_channel *ch )
612 {
613 u32 format = ch->source->flags & AUDIO_FLAG_FORMAT;
614
615 if( format == k_audio_format_vorbis ){
616 /* Setup vorbis decoder */
617 u32 index = ch - vg_audio.channels;
618
619 u8 *buf = (u8*)vg_audio.decode_buffer,
620 *loc = &buf[AUDIO_DECODE_SIZE*index];
621
622 stb_vorbis_alloc alloc = {
623 .alloc_buffer = (char *)loc,
624 .alloc_buffer_length_in_bytes = AUDIO_DECODE_SIZE
625 };
626
627 int err;
628 stb_vorbis *decoder = stb_vorbis_open_memory(
629 ch->source->data,
630 ch->source->size, &err, &alloc );
631
632 if( !decoder ){
633 vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
634 ch->source->path, err );
635 return 0;
636 }
637 else{
638 ch->source_length = stb_vorbis_stream_length_in_samples( decoder );
639 ch->vorbis_handle = decoder;
640 }
641 }
642 else if( format == k_audio_format_bird ){
643 u32 index = ch - vg_audio.channels;
644
645 u8 *buf = (u8*)vg_audio.decode_buffer;
646 struct synth_bird *loc = (void *)&buf[AUDIO_DECODE_SIZE*index];
647
648 memcpy( loc, ch->source->data, ch->source->size );
649 synth_bird_reset( loc );
650
651 ch->bird_handle = loc;
652 ch->source_length = synth_bird_get_length_in_samples( loc );
653 }
654 else if( format == k_audio_format_stereo ){
655 ch->source_length = ch->source->size / 2;
656 }
657 else if( format == k_audio_format_gen ){
658 ch->source_length = 0xffffffff;
659 }
660 else{
661 ch->source_length = ch->source->size;
662 }
663
664 return 1;
665 }
666
667 static void audio_decode_uncompressed_mono( i16 *src, u32 count, float *dst )
668 {
669 for( u32 i=0; i<count; i++ ){
670 dst[ i*2 + 0 ] = ((float)src[i]) * (1.0f/32767.0f);
671 dst[ i*2 + 1 ] = ((float)src[i]) * (1.0f/32767.0f);
672 }
673 }
674
675 /*
676 * adapted from stb_vorbis.h, since the original does not handle mono->stereo
677 */
678 static int
679 stb_vorbis_get_samples_float_interleaved_stereo( stb_vorbis *f, float *buffer,
680 int len )
681 {
682 int n = 0,
683 c = VG_MIN( 1, f->channels - 1 );
684
685 while( n < len ) {
686 int k = f->channel_buffer_end - f->channel_buffer_start;
687
688 if( n+k >= len )
689 k = len - n;
690
691 for( int j=0; j < k; ++j ) {
692 *buffer++ = f->channel_buffers[ 0 ][f->channel_buffer_start+j];
693 *buffer++ = f->channel_buffers[ c ][f->channel_buffer_start+j];
694 }
695
696 n += k;
697 f->channel_buffer_start += k;
698
699 if( n == len )
700 break;
701
702 if( !stb_vorbis_get_frame_float( f, NULL, NULL ))
703 break;
704 }
705
706 return n;
707 }
708
709 /*
710 * ........ more wrecked code sorry!
711 */
712 static int
713 stb_vorbis_get_samples_i16_downmixed( stb_vorbis *f, i16 *buffer, int len )
714 {
715 int n = 0,
716 c = VG_MIN( 1, f->channels - 1 );
717
718 while( n < len ) {
719 int k = f->channel_buffer_end - f->channel_buffer_start;
720
721 if( n+k >= len )
722 k = len - n;
723
724 for( int j=0; j < k; ++j ) {
725 float sl = f->channel_buffers[ 0 ][f->channel_buffer_start+j],
726 sr = f->channel_buffers[ c ][f->channel_buffer_start+j];
727
728 *buffer++ = vg_clampf( 0.5f*(sl+sr), -1.0f, 1.0f ) * 32767.0f;
729 //*buffer++ = vg_clampf( sr, -1.0f, 1.0f ) * 32767.0f;
730 }
731
732 n += k;
733 f->channel_buffer_start += k;
734
735 if( n == len )
736 break;
737
738 if( !stb_vorbis_get_frame_float( f, NULL, NULL ))
739 break;
740 }
741
742 return n;
743 }
744
745 static inline float audio_lfo_pull_sample( audio_lfo *lfo )
746 {
747 lfo->time ++;
748
749 if( lfo->time >= lfo->_.period )
750 lfo->time = 0;
751
752 float t = lfo->time;
753 t /= (float)lfo->_.period;
754
755 if( lfo->_.wave_type == k_lfo_polynomial_bipolar ){
756 /*
757 * #
758 * # #
759 * # #
760 * # #
761 * ### # ###
762 * ## #
763 * # #
764 * # #
765 * ##
766 */
767
768 t *= 2.0f;
769 t -= 1.0f;
770
771 return (( 2.0f * lfo->sqrt_polynomial_coefficient * t ) /
772 /* --------------------------------------- */
773 ( 1.0f + lfo->_.polynomial_coefficient * t*t )
774
775 ) * (1.0f-fabsf(t));
776 }
777 else{
778 return 0.0f;
779 }
780 }
781
782 static void audio_channel_get_samples( audio_channel *ch,
783 u32 count, float *buf )
784 {
785 vg_profile_begin( &_vg_prof_audio_decode );
786
787 u32 remaining = count;
788 u32 buffer_pos = 0;
789
790 u32 format = ch->source->flags & AUDIO_FLAG_FORMAT;
791
792 while( remaining ){
793 u32 samples_this_run = VG_MIN(remaining, ch->source_length - ch->cursor);
794 remaining -= samples_this_run;
795
796 float *dst = &buf[ buffer_pos * 2 ];
797
798 if( format == k_audio_format_stereo ){
799 for( int i=0;i<samples_this_run; i++ ){
800 dst[i*2+0] = 0.0f;
801 dst[i*2+1] = 0.0f;
802 }
803 }
804 else if( format == k_audio_format_vorbis ){
805 int read_samples = stb_vorbis_get_samples_float_interleaved_stereo(
806 ch->vorbis_handle,
807 dst,
808 samples_this_run );
809
810 if( read_samples != samples_this_run ){
811 vg_warn( "Invalid samples read (%s)\n", ch->source->path );
812
813 for( int i=0; i<samples_this_run; i++ ){
814 dst[i*2+0] = 0.0f;
815 dst[i*2+1] = 0.0f;
816 }
817 }
818 }
819 else if( format == k_audio_format_bird ){
820 synth_bird_generate_samples( ch->bird_handle, dst, samples_this_run );
821 }
822 else if( format == k_audio_format_gen ){
823 void (*fn)( void *data, f32 *buf, u32 count ) = ch->source->func;
824 fn( ch->source->data, dst, samples_this_run );
825 }
826 else{
827 i16 *src_buffer = ch->source->data,
828 *src = &src_buffer[ch->cursor];
829
830 audio_decode_uncompressed_mono( src, samples_this_run, dst );
831 }
832
833 ch->cursor += samples_this_run;
834 buffer_pos += samples_this_run;
835
836 if( (ch->flags & AUDIO_FLAG_LOOP) && remaining ){
837 if( format == k_audio_format_vorbis )
838 stb_vorbis_seek_start( ch->vorbis_handle );
839 else if( format == k_audio_format_bird )
840 synth_bird_reset( ch->bird_handle );
841
842 ch->cursor = 0;
843 continue;
844 }
845 else
846 break;
847 }
848
849 while( remaining ){
850 buf[ buffer_pos*2 + 0 ] = 0.0f;
851 buf[ buffer_pos*2 + 1 ] = 0.0f;
852 buffer_pos ++;
853
854 remaining --;
855 }
856
857 vg_profile_end( &_vg_prof_audio_decode );
858 }
859
860 static void audio_channel_mix( audio_channel *ch, float *buffer )
861 {
862 float framevol_l = vg_audio.internal_global_volume,
863 framevol_r = vg_audio.internal_global_volume;
864
865 float frame_samplerate = ch->_.sampling_rate;
866
867 if( ch->flags & AUDIO_FLAG_SPACIAL_3D ){
868 v3f delta;
869 v3_sub( ch->_.spacial_falloff, vg_audio.internal_listener_pos, delta );
870
871 float dist = v3_length( delta ),
872 vol = vg_maxf( 0.0f, 1.0f - ch->_.spacial_falloff[3]*dist );
873
874 if( dist <= 0.01f ){
875
876 }
877 else{
878 v3_muls( delta, 1.0f/dist, delta );
879 float pan = v3_dot( vg_audio.internal_listener_ears, delta );
880 vol = powf( vol, 5.0f );
881
882 framevol_l *= (vol * 0.5f) * (1.0f - pan);
883 framevol_r *= (vol * 0.5f) * (1.0f + pan);
884
885 if( !(ch->source->flags & AUDIO_FLAG_NO_DOPPLER) ){
886 const float vs = 323.0f;
887
888 float dv = v3_dot(delta,vg_audio.internal_listener_velocity);
889 float doppler = (vs+dv)/vs;
890 doppler = vg_clampf( doppler, 0.6f, 1.4f );
891
892 if( fabsf(doppler-1.0f) > 0.01f )
893 frame_samplerate *= doppler;
894 }
895 }
896
897 if( !vg_validf( framevol_l ) ||
898 !vg_validf( framevol_r ) ||
899 !vg_validf( frame_samplerate ) ){
900 vg_fatal_error( "Invalid sampling conditions.\n"
901 "This crash is to protect your ears.\n"
902 " channel: %p (%s)\n"
903 " sample_rate: %f\n"
904 " volume: L%f R%f\n"
905 " listener: %.2f %.2f %.2f [%.2f %.2f %.2f]\n",
906 ch, ch->name, frame_samplerate,
907 framevol_l, framevol_r,
908 vg_audio.internal_listener_pos[0],
909 vg_audio.internal_listener_pos[1],
910 vg_audio.internal_listener_pos[2],
911 vg_audio.internal_listener_ears[0],
912 vg_audio.internal_listener_ears[1],
913 vg_audio.internal_listener_ears[2]
914 );
915 }
916 }
917
918 u32 buffer_length = AUDIO_MIX_FRAME_SIZE;
919 if( frame_samplerate != 1.0f ){
920 float l = ceilf( (float)(AUDIO_MIX_FRAME_SIZE) * frame_samplerate );
921 buffer_length = l+1;
922 }
923
924 float pcf[ AUDIO_MIX_FRAME_SIZE * 2 * 2 ];
925
926 audio_channel_get_samples( ch, buffer_length, pcf );
927
928 vg_profile_begin( &_vg_prof_audio_mix );
929
930 float volume_movement = ch->volume_movement;
931 float const fvolume_rate = vg_maxf( 1.0f, ch->_.volume_rate );
932 const float inv_volume_rate = 1.0f/fvolume_rate;
933
934 float volume = ch->_.volume;
935 const float volume_start = ch->volume_movement_start;
936 const float volume_target = ch->_.volume_target;
937
938 for( u32 j=0; j<AUDIO_MIX_FRAME_SIZE; j++ ){
939 volume_movement += 1.0f;
940 float movement_t = volume_movement * inv_volume_rate;
941 movement_t = vg_minf( movement_t, 1.0f );
942 volume = vg_lerpf( volume_start, volume_target, movement_t );
943
944 float vol_norm = volume * volume;
945
946 if( ch->_.lfo )
947 vol_norm *= 1.0f + audio_lfo_pull_sample(ch->_.lfo) * ch->_.lfo_amount;
948
949 float vol_l = vol_norm * framevol_l,
950 vol_r = vol_norm * framevol_r,
951 sample_l,
952 sample_r;
953
954 if( frame_samplerate != 1.0f ){
955 /* absolutely garbage resampling, but it will do
956 */
957
958 float sample_index = frame_samplerate * (float)j;
959 float t = vg_fractf( sample_index );
960
961 u32 i0 = floorf( sample_index ),
962 i1 = i0+1;
963
964 sample_l = pcf[ i0*2+0 ]*(1.0f-t) + pcf[ i1*2+0 ]*t;
965 sample_r = pcf[ i0*2+1 ]*(1.0f-t) + pcf[ i1*2+1 ]*t;
966 }
967 else{
968 sample_l = pcf[ j*2+0 ];
969 sample_r = pcf[ j*2+1 ];
970 }
971
972 buffer[ j*2+0 ] += sample_l * vol_l;
973 buffer[ j*2+1 ] += sample_r * vol_r;
974 }
975
976 ch->volume_movement += AUDIO_MIX_FRAME_SIZE;
977 ch->volume_movement = VG_MIN( ch->volume_movement, ch->_.volume_rate );
978 ch->_.volume = volume;
979
980 vg_profile_end( &_vg_prof_audio_mix );
981 }
982
983 static void audio_mixer_callback( void *user, u8 *stream, int byte_count ){
984 /*
985 * Copy data and move edit flags to commit flags
986 * ------------------------------------------------------------- */
987 audio_lock();
988 int use_dsp = vg_audio.dsp_enabled;
989
990 v3_copy( vg_audio.external_listener_pos, vg_audio.internal_listener_pos );
991 v3_copy( vg_audio.external_listener_ears, vg_audio.internal_listener_ears );
992 v3_copy( vg_audio.external_lister_velocity,
993 vg_audio.internal_listener_velocity );
994 vg_audio.internal_global_volume = vg_audio.external_global_volume;
995
996 for( int i=0; i<AUDIO_CHANNELS; i++ ){
997 audio_channel *ch = &vg_audio.channels[i];
998
999 if( !ch->allocated )
1000 continue;
1001
1002 if( ch->activity == k_channel_activity_alive ){
1003 if( (ch->cursor >= ch->source_length) &&
1004 !(ch->flags & AUDIO_FLAG_LOOP) )
1005 {
1006 ch->activity = k_channel_activity_end;
1007 }
1008 }
1009
1010 /* process relinquishments */
1011 if( (ch->activity != k_channel_activity_reset) && ch->_.relinquished ){
1012 if( (ch->activity == k_channel_activity_end)
1013 || (ch->_.volume == 0.0f)
1014 || (ch->activity == k_channel_activity_error) )
1015 {
1016 ch->_.relinquished = 0;
1017 ch->allocated = 0;
1018 ch->activity = k_channel_activity_reset;
1019 continue;
1020 }
1021 }
1022
1023 /* process new channels */
1024 if( ch->activity == k_channel_activity_reset ){
1025 ch->_ = ch->editable_state;
1026 ch->cursor = 0;
1027 ch->source_length = 0;
1028 ch->activity = k_channel_activity_wake;
1029 }
1030
1031 if( ch->editble_state_write_mask & AUDIO_EDIT_OWNERSHIP )
1032 ch->_.relinquished = ch->editable_state.relinquished;
1033 else
1034 ch->editable_state.relinquished = ch->_.relinquished;
1035
1036
1037 if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME ){
1038 ch->_.volume = ch->editable_state.volume;
1039 ch->_.volume_target = ch->editable_state.volume;
1040 }
1041 else{
1042 ch->editable_state.volume = ch->_.volume;
1043 }
1044
1045
1046 if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME_SLOPE ){
1047 ch->volume_movement_start = ch->_.volume;
1048 ch->volume_movement = 0;
1049
1050 ch->_.volume_target = ch->editable_state.volume_target;
1051 ch->_.volume_rate = ch->editable_state.volume_rate;
1052 }
1053 else{
1054 ch->editable_state.volume_target = ch->_.volume_target;
1055 ch->editable_state.volume_rate = ch->_.volume_rate;
1056 }
1057
1058
1059 if( ch->editble_state_write_mask & AUDIO_EDIT_SAMPLING_RATE )
1060 ch->_.sampling_rate = ch->editable_state.sampling_rate;
1061 else
1062 ch->editable_state.sampling_rate = ch->_.sampling_rate;
1063
1064
1065 if( ch->editble_state_write_mask & AUDIO_EDIT_LFO_ATTACHMENT ){
1066 ch->_.lfo = ch->editable_state.lfo;
1067 ch->_.lfo_amount = ch->editable_state.lfo_amount;
1068 }
1069 else{
1070 ch->editable_state.lfo = ch->_.lfo;
1071 ch->editable_state.lfo_amount = ch->_.lfo_amount;
1072 }
1073
1074
1075 if( ch->editble_state_write_mask & AUDIO_EDIT_SPACIAL )
1076 v4_copy( ch->editable_state.spacial_falloff,ch->_.spacial_falloff );
1077 else
1078 v4_copy( ch->_.spacial_falloff,ch->editable_state.spacial_falloff );
1079
1080
1081 /* currently readonly, i guess */
1082 ch->editable_state.pan_target = ch->_.pan_target;
1083 ch->editable_state.pan = ch->_.pan;
1084 ch->editble_state_write_mask = 0x00;
1085 }
1086
1087 for( int i=0; i<AUDIO_LFOS; i++ ){
1088 audio_lfo *lfo = &vg_audio.oscillators[ i ];
1089
1090 if( lfo->editble_state_write_mask & AUDIO_EDIT_LFO_WAVE ){
1091 lfo->_.wave_type = lfo->editable_state.wave_type;
1092
1093 if( lfo->_.wave_type == k_lfo_polynomial_bipolar ){
1094 lfo->_.polynomial_coefficient =
1095 lfo->editable_state.polynomial_coefficient;
1096 lfo->sqrt_polynomial_coefficient =
1097 sqrtf(lfo->_.polynomial_coefficient);
1098 }
1099 }
1100
1101 if( lfo->editble_state_write_mask & AUDIO_EDIT_LFO_PERIOD ){
1102 if( lfo->_.period ){
1103 float t = lfo->time;
1104 t/= (float)lfo->_.period;
1105
1106 lfo->_.period = lfo->editable_state.period;
1107 lfo->time = lfo->_.period * t;
1108 }
1109 else{
1110 lfo->time = 0;
1111 lfo->_.period = lfo->editable_state.period;
1112 }
1113 }
1114
1115 lfo->editble_state_write_mask = 0x00;
1116 }
1117
1118 dsp_update_tunings();
1119 audio_unlock();
1120
1121 /*
1122 * Process spawns
1123 * ------------------------------------------------------------- */
1124 for( int i=0; i<AUDIO_CHANNELS; i++ ){
1125 audio_channel *ch = &vg_audio.channels[i];
1126
1127 if( ch->activity == k_channel_activity_wake ){
1128 if( audio_channel_load_source( ch ) )
1129 ch->activity = k_channel_activity_alive;
1130 else
1131 ch->activity = k_channel_activity_error;
1132 }
1133 }
1134
1135 /*
1136 * Mix everything
1137 * -------------------------------------------------------- */
1138 int frame_count = byte_count/(2*sizeof(float));
1139
1140 /* Clear buffer */
1141 float *pOut32F = (float *)stream;
1142 for( int i=0; i<frame_count*2; i ++ )
1143 pOut32F[i] = 0.0f;
1144
1145 for( int i=0; i<AUDIO_LFOS; i++ ){
1146 audio_lfo *lfo = &vg_audio.oscillators[i];
1147 lfo->time_startframe = lfo->time;
1148 }
1149
1150 for( int i=0; i<AUDIO_CHANNELS; i ++ ){
1151 audio_channel *ch = &vg_audio.channels[i];
1152
1153 if( ch->activity == k_channel_activity_alive ){
1154 if( ch->_.lfo )
1155 ch->_.lfo->time = ch->_.lfo->time_startframe;
1156
1157 u32 remaining = frame_count,
1158 subpos = 0;
1159
1160 while( remaining ){
1161 audio_channel_mix( ch, pOut32F+subpos );
1162 remaining -= AUDIO_MIX_FRAME_SIZE;
1163 subpos += AUDIO_MIX_FRAME_SIZE*2;
1164 }
1165 }
1166 }
1167
1168 if( use_dsp ){
1169 vg_profile_begin( &_vg_prof_dsp );
1170 for( int i=0; i<frame_count; i++ )
1171 vg_dsp_process( pOut32F + i*2, pOut32F + i*2 );
1172 vg_profile_end( &_vg_prof_dsp );
1173 }
1174
1175 audio_lock();
1176
1177 for( int i=0; i<AUDIO_CHANNELS; i ++ ){
1178 audio_channel *ch = &vg_audio.channels[i];
1179 ch->readable_activity = ch->activity;
1180 }
1181
1182 /* Profiling information
1183 * ----------------------------------------------- */
1184 vg_profile_increment( &_vg_prof_audio_decode );
1185 vg_profile_increment( &_vg_prof_audio_mix );
1186 vg_profile_increment( &_vg_prof_dsp );
1187
1188 vg_prof_audio_mix = _vg_prof_audio_mix;
1189 vg_prof_audio_decode = _vg_prof_audio_decode;
1190 vg_prof_audio_dsp = _vg_prof_dsp;
1191
1192 vg_audio.samples_last = frame_count;
1193
1194 if( vg_audio.debug_dsp ){
1195 vg_dsp_update_texture();
1196 }
1197
1198 audio_unlock();
1199 }
1200
1201 static void audio_clip_load( audio_clip *clip, void *lin_alloc )
1202 {
1203 if( lin_alloc == NULL )
1204 lin_alloc = vg_audio.audio_pool;
1205
1206 #ifdef VG_AUDIO_FORCE_COMPRESSED
1207
1208 if( (clip->flags & AUDIO_FLAG_FORMAT) != k_audio_format_bird ){
1209 clip->flags &= ~AUDIO_FLAG_FORMAT;
1210 clip->flags |= k_audio_format_vorbis;
1211 }
1212
1213 #endif
1214
1215 /* load in directly */
1216 u32 format = clip->flags & AUDIO_FLAG_FORMAT;
1217
1218 /* TODO: This contains audio_lock() and unlock, but i don't know why
1219 * can probably remove them. Low priority to check this */
1220
1221 /* TODO: packed files for vorbis etc, should take from data if its not not
1222 * NULL when we get the clip
1223 */
1224
1225 if( format == k_audio_format_vorbis ){
1226 if( !clip->path ){
1227 vg_fatal_error( "No path specified, embeded vorbis unsupported" );
1228 }
1229
1230 audio_lock();
1231 clip->data = vg_file_read( lin_alloc, clip->path, &clip->size );
1232 audio_unlock();
1233
1234 if( !clip->data )
1235 vg_fatal_error( "Audio failed to load" );
1236
1237 float mb = (float)(clip->size) / (1024.0f*1024.0f);
1238 vg_info( "Loaded audio clip '%s' (%.1fmb)\n", clip->path, mb );
1239 }
1240 else if( format == k_audio_format_stereo ){
1241 vg_fatal_error( "Unsupported format (Stereo uncompressed)" );
1242 }
1243 else if( format == k_audio_format_bird ){
1244 if( !clip->data ){
1245 vg_fatal_error( "No data, external birdsynth unsupported" );
1246 }
1247
1248 u32 total_size = clip->size + sizeof(struct synth_bird);
1249 total_size -= sizeof(struct synth_bird_settings);
1250 total_size = vg_align8( total_size );
1251
1252 if( total_size > AUDIO_DECODE_SIZE )
1253 vg_fatal_error( "Bird coding too long\n" );
1254
1255 struct synth_bird *bird = vg_linear_alloc( lin_alloc, total_size );
1256 memcpy( &bird->settings, clip->data, clip->size );
1257
1258 clip->data = bird;
1259 clip->size = total_size;
1260
1261 vg_info( "Loaded bird synthesis pattern (%u bytes)\n", total_size );
1262 }
1263 else{
1264 if( !clip->path ){
1265 vg_fatal_error( "No path specified, embeded mono unsupported" );
1266 }
1267
1268 vg_linear_clear( vg_mem.scratch );
1269 u32 fsize;
1270
1271 stb_vorbis_alloc alloc = {
1272 .alloc_buffer = vg_linear_alloc( vg_mem.scratch, AUDIO_DECODE_SIZE ),
1273 .alloc_buffer_length_in_bytes = AUDIO_DECODE_SIZE
1274 };
1275
1276 void *filedata = vg_file_read( vg_mem.scratch, clip->path, &fsize );
1277
1278 int err;
1279 stb_vorbis *decoder = stb_vorbis_open_memory(
1280 filedata, fsize, &err, &alloc );
1281
1282 if( !decoder ){
1283 vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
1284 clip->path, err );
1285 vg_fatal_error( "Vorbis decode error" );
1286 }
1287
1288 /* only mono is supported in uncompressed */
1289 u32 length_samples = stb_vorbis_stream_length_in_samples( decoder ),
1290 data_size = length_samples * sizeof(i16);
1291
1292 audio_lock();
1293 clip->data = vg_linear_alloc( lin_alloc, vg_align8(data_size) );
1294 clip->size = length_samples;
1295 audio_unlock();
1296
1297 int read_samples = stb_vorbis_get_samples_i16_downmixed(
1298 decoder, clip->data, length_samples );
1299
1300 if( read_samples != length_samples )
1301 vg_fatal_error( "Decode error" );
1302
1303 #if 0
1304 float mb = (float)(data_size) / (1024.0f*1024.0f);
1305 vg_info( "Loaded audio clip '%s' (%.1fmb) %u samples\n", clip->path, mb,
1306 length_samples );
1307 #endif
1308 }
1309 }
1310
1311 static void audio_clip_loadn( audio_clip *arr, int count, void *lin_alloc )
1312 {
1313 for( int i=0; i<count; i++ )
1314 audio_clip_load( &arr[i], lin_alloc );
1315 }
1316
1317 static void audio_require_clip_loaded( audio_clip *clip )
1318 {
1319 if( clip->data && clip->size )
1320 return;
1321
1322 audio_unlock();
1323 vg_fatal_error( "Must load audio clip before playing! \n" );
1324 }
1325
1326 /*
1327 * Debugging
1328 */
1329
1330 static void audio_debug_ui(
1331
1332 #ifdef VG_3D
1333 m4x4f
1334 #else
1335 m3x3f
1336 #endif
1337 mtx_pv ){
1338
1339 if( !vg_audio.debug_ui )
1340 return;
1341
1342 audio_lock();
1343
1344 glBindTexture( GL_TEXTURE_2D, vg_dsp.view_texture );
1345 glTexSubImage2D( GL_TEXTURE_2D, 0, 0, 0, 256, 256,
1346 GL_RGBA, GL_UNSIGNED_BYTE,
1347 vg_dsp.view_texture_buffer );
1348
1349 /*
1350 * Profiler
1351 * -----------------------------------------------------------------------
1352 */
1353
1354 float budget = ((double)vg_audio.samples_last / 44100.0) * 1000.0;
1355 vg_profile_drawn( (struct vg_profile *[]){ &vg_prof_audio_decode,
1356 &vg_prof_audio_mix,
1357 &vg_prof_audio_dsp}, 3,
1358 budget, (ui_rect){ 4, VG_PROFILE_SAMPLE_COUNT*2 + 8,
1359 512, 0 }, 3, 0 );
1360
1361
1362 char perf[128];
1363
1364 /* Draw UI */
1365 ui_rect window = {
1366 0,
1367 0,
1368 800,
1369 AUDIO_CHANNELS * 18
1370 };
1371
1372 if( vg_audio.debug_dsp ){
1373 ui_rect view_thing = { 4, vg.window_y-512-4, 512, 512 };
1374 ui_image( view_thing, vg_dsp.view_texture );
1375 }
1376
1377 ui_rect overlap_buffer[ AUDIO_CHANNELS ];
1378 u32 overlap_length = 0;
1379
1380 /* Draw audio stack */
1381 for( int i=0; i<AUDIO_CHANNELS; i ++ ){
1382 audio_channel *ch = &vg_audio.channels[i];
1383
1384 ui_rect row;
1385 ui_split( window, k_ui_axis_h, 18, 1, row, window );
1386
1387 if( !ch->allocated ){
1388 ui_fill( row, 0x50333333 );
1389 continue;
1390 }
1391
1392 const char *formats[] =
1393 {
1394 " mono ",
1395 " stereo ",
1396 " vorbis ",
1397 " none0 ",
1398 " none1 ",
1399 " none2 ",
1400 " none3 ",
1401 " none4 ",
1402 "synth:bird",
1403 " none5 ",
1404 " none6 ",
1405 " none7 ",
1406 " none8 ",
1407 " none9 ",
1408 " none10 ",
1409 " none11 ",
1410 };
1411
1412 const char *activties[] =
1413 {
1414 "reset",
1415 "wake ",
1416 "alive",
1417 "end ",
1418 "error"
1419 };
1420
1421 u32 format_index = (ch->source->flags & AUDIO_FLAG_FORMAT)>>9;
1422
1423 snprintf( perf, 127, "%02d[%#04x.%#06x]%c%c%cD %s [%s] %4.2fv'%s'",
1424 i,
1425 ch->world_id, ch->group,
1426 (ch->editable_state.relinquished)? 'r': '_',
1427 0? 'r': '_',
1428 0? '3': '2',
1429 formats[format_index],
1430 activties[ch->readable_activity],
1431 ch->editable_state.volume,
1432 ch->name );
1433
1434 ui_fill( row, 0xa0000000 | ch->colour );
1435 ui_text( row, perf, 1, k_ui_align_middle_left, 0 );
1436
1437 #ifdef VG_3D
1438 if( AUDIO_FLAG_SPACIAL_3D ){
1439 v4f wpos;
1440 v3_copy( ch->editable_state.spacial_falloff, wpos );
1441
1442 wpos[3] = 1.0f;
1443 m4x4_mulv( mtx_pv, wpos, wpos );
1444
1445 if( wpos[3] > 0.0f ){
1446 v2_muls( wpos, (1.0f/wpos[3]) * 0.5f, wpos );
1447 v2_add( wpos, (v2f){ 0.5f, 0.5f }, wpos );
1448
1449 ui_rect wr;
1450 wr[0] = vg_clampf(wpos[0] * vg.window_x, -32000.0f,32000.0f);
1451 wr[1] = vg_clampf((1.0f-wpos[1]) * vg.window_y,-32000.0f,32000.0f);
1452 wr[2] = 1000;
1453 wr[3] = 17;
1454
1455 for( int j=0; j<12; j++ ){
1456 int collide = 0;
1457 for( int k=0; k<overlap_length; k++ ){
1458 ui_px *wk = overlap_buffer[k];
1459 if( ((wr[0] <= wk[0]+wk[2]) && (wr[0]+wr[2] >= wk[0])) &&
1460 ((wr[1] <= wk[1]+wk[3]) && (wr[1]+wr[3] >= wk[1])) )
1461 {
1462 collide = 1;
1463 break;
1464 }
1465 }
1466
1467 if( !collide )
1468 break;
1469 else
1470 wr[1] += 18;
1471 }
1472
1473 ui_text( wr, perf, 1, k_ui_align_middle_left, 0 );
1474 rect_copy( wr, overlap_buffer[ overlap_length ++ ] );
1475 }
1476 }
1477 #endif
1478 }
1479
1480 audio_unlock();
1481 }
1482
1483 #endif /* VG_AUDIO_H */
Voyager Game Engine