perlin
[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_ui.h"
14 #include "vg/vg_console.h"
15 #include "vg/vg_store.h"
16 #include "vg/vg_profiler.h"
17 #include "vg/vg_audio_synth_bird.h"
18
19 #include <sys/time.h>
20 #include <math.h>
21
22 #ifdef __GNUC__
23 #ifndef __clang__
24 #pragma GCC push_options
25 #pragma GCC optimize ("O3")
26 #pragma GCC diagnostic push
27 #pragma GCC diagnostic ignored "-Wdeprecated-declarations"
28 #endif
29 #endif
30
31 #define STB_VORBIS_MAX_CHANNELS 2
32 #include "submodules/stb/stb_vorbis.c"
33 #undef L
34 #undef R
35 #undef C
36
37 #ifdef __GNUC__
38 #ifndef __clang__
39 #pragma GCC pop_options
40 #pragma GCC diagnostic pop
41 #endif
42 #endif
43
44 #define AUDIO_FRAME_SIZE 512
45 #define AUDIO_MIX_FRAME_SIZE 256
46
47 #define AUDIO_CHANNELS 32
48 #define AUDIO_LFOS 8
49 #define AUDIO_FILTERS 16
50 #define AUDIO_FLAG_LOOP 0x1
51 #define AUDIO_FLAG_SPACIAL_3D 0x4
52 #define AUDIO_FLAG_AUTO_START 0x8
53
54 /* Vorbis will ALWAYS use the maximum amount of channels it can */
55 //#define AUDIO_FLAG_MONO 0x100 NOTE: This is the default, so its not used
56 //#define AUDIO_FLAG_STEREO 0x200
57 //#define AUDIO_FLAG_VORBIS 0x400
58 //#define AUDIO_FLAG_BIRD_SYNTH 0x800
59
60 #define AUDIO_FLAG_FORMAT 0x1E00
61
62 enum audio_format
63 {
64 k_audio_format_mono = 0x000u,
65 k_audio_format_stereo = 0x200u,
66 k_audio_format_vorbis = 0x400u,
67 k_audio_format_none0 = 0x600u,
68 k_audio_format_none1 = 0x800u,
69 k_audio_format_none2 = 0xA00u,
70 k_audio_format_none3 = 0xC00u,
71 k_audio_format_none4 = 0xE00u,
72
73 k_audio_format_bird = 0x1000u,
74 k_audio_format_none5 = 0x1200u,
75 k_audio_format_none6 = 0x1400u,
76 k_audio_format_none7 = 0x1600u,
77 k_audio_format_none8 = 0x1800u,
78 k_audio_format_none9 = 0x1A00u,
79 k_audio_format_none10 = 0x1C00u,
80 k_audio_format_none11 = 0x1E00u,
81 };
82
83 #define AUDIO_DECODE_SIZE (1024*256) /* 256 kb decoding buffers */
84 #define AUDIO_MUTE_VOLUME 0.0f
85 #define AUDIO_BASE_VOLUME 1.0f
86
87 typedef struct audio_clip audio_clip;
88 typedef struct audio_channel audio_channel;
89 typedef struct audio_lfo audio_lfo;
90
91 struct audio_clip
92 {
93 const char *path;
94 u32 flags;
95
96 u32 size;
97 void *data;
98 };
99
100 static struct vg_audio_system
101 {
102 SDL_AudioDeviceID sdl_output_device;
103
104 void *audio_pool,
105 *decode_buffer;
106 u32 samples_last;
107
108 /* synchro */
109 int sync_locked;
110
111 SDL_mutex *mux_checker,
112 *mux_sync;
113
114 struct audio_lfo
115 {
116 u32 time, time_startframe;
117 float sqrt_polynomial_coefficient;
118
119 struct
120 {
121 enum lfo_wave_type
122 {
123 k_lfo_triangle,
124 k_lfo_square,
125 k_lfo_saw,
126 k_lfo_polynomial_bipolar
127 }
128 wave_type;
129
130 u32 period;
131 float polynomial_coefficient;
132 }
133 _, editable_state;
134 u32 editble_state_write_mask;
135 }
136 oscillators[ AUDIO_LFOS ];
137
138 struct audio_channel
139 {
140 int allocated;
141 char name[32]; /* only editable while allocated == 0 */
142 audio_clip *source; /* ... */
143 u32 flags; /* ... */
144 u32 colour; /* ... */
145
146 /* internal non-readable state
147 * -----------------------------*/
148 u32 cursor, source_length;
149
150 float volume_movement_start,
151 pan_movement_start;
152
153 u32 volume_movement,
154 pan_movement;
155
156 union
157 {
158 struct synth_bird *bird_handle;
159 stb_vorbis *vorbis_handle;
160 };
161
162 stb_vorbis_alloc vorbis_alloc;
163
164 enum channel_activity
165 {
166 k_channel_activity_reset, /* will advance if allocated==1, to wake */
167 k_channel_activity_wake, /* will advance to either of next two */
168 k_channel_activity_alive,
169 k_channel_activity_end,
170 k_channel_activity_error
171 }
172 activity,
173 readable_activity;
174
175 /*
176 * editable structure, can be modified inside _lock and _unlock
177 * the edit mask tells which to copy into internal _, or to discard
178 * ----------------------------------------------------------------------
179 */
180 struct channel_state
181 {
182 int relinquished;
183
184 float volume, /* current volume */
185 volume_target, /* target volume */
186 pan,
187 pan_target,
188 sampling_rate;
189
190 u32 volume_rate,
191 pan_rate;
192
193 v4f spacial_falloff; /* xyz, range */
194
195 audio_lfo *lfo;
196 float lfo_amount;
197 }
198 _, editable_state;
199 u32 editble_state_write_mask;
200 }
201 channels[ AUDIO_CHANNELS ];
202
203 /* System queue, and access from thread 0 */
204 int debug_ui, debug_ui_3d;
205
206 v3f listener_pos,
207 listener_ears,
208 listener_velocity;
209
210 float volume,
211 volume_target,
212 volume_target_internal,
213 volume_console;
214 }
215 vg_audio = { .volume_console = 1.0f };
216
217 #include "vg/vg_audio_dsp.h"
218
219 static struct vg_profile
220 _vg_prof_audio_decode = {.mode = k_profile_mode_accum,
221 .name = "[T2] audio_decode()"},
222 _vg_prof_audio_mix = {.mode = k_profile_mode_accum,
223 .name = "[T2] audio_mix()"},
224 _vg_prof_dsp = {.mode = k_profile_mode_accum,
225 .name = "[T2] dsp_process()"},
226 vg_prof_audio_decode,
227 vg_prof_audio_mix,
228 vg_prof_audio_dsp;
229
230 /*
231 * These functions are called from the main thread and used to prevent bad
232 * access. TODO: They should be no-ops in release builds.
233 */
234 VG_STATIC int audio_lock_checker_load(void)
235 {
236 int value;
237 SDL_LockMutex( vg_audio.mux_checker );
238 value = vg_audio.sync_locked;
239 SDL_UnlockMutex( vg_audio.mux_checker );
240 return value;
241 }
242
243 VG_STATIC void audio_lock_checker_store( int value )
244 {
245 SDL_LockMutex( vg_audio.mux_checker );
246 vg_audio.sync_locked = value;
247 SDL_UnlockMutex( vg_audio.mux_checker );
248 }
249
250 VG_STATIC void audio_require_lock(void)
251 {
252 if( audio_lock_checker_load() )
253 return;
254
255 vg_error( "Modifying sound effects systems requires locking\n" );
256 abort();
257 }
258
259 VG_STATIC void audio_lock(void)
260 {
261 SDL_LockMutex( vg_audio.mux_sync );
262 audio_lock_checker_store(1);
263 }
264
265 VG_STATIC void audio_unlock(void)
266 {
267 audio_lock_checker_store(0);
268 SDL_UnlockMutex( vg_audio.mux_sync );
269 }
270
271 VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int frame_count );
272 VG_STATIC void vg_audio_init(void)
273 {
274 vg_audio.mux_checker = SDL_CreateMutex();
275 vg_audio.mux_sync = SDL_CreateMutex();
276
277 /* TODO: Move here? */
278 vg_var_push( (struct vg_var){
279 .name = "debug_audio",
280 .data = &vg_audio.debug_ui,
281 .data_type = k_var_dtype_i32,
282 .opt_i32 = { .min=0, .max=1, .clamp=1 },
283 .persistent = 1
284 });
285
286 vg_var_push( (struct vg_var){
287 .name = "volume",
288 .data = &vg_audio.volume_console,
289 .data_type = k_var_dtype_f32,
290 .opt_f32 = { .min=0.0f, .max=2.0f, .clamp=1 },
291 .persistent = 1
292 });
293
294 /* allocate memory */
295
296 /* 32mb fixed */
297 vg_audio.audio_pool =
298 vg_create_linear_allocator( vg_mem.rtmemory, 1024*1024*32,
299 VG_MEMORY_SYSTEM );
300
301 /* fixed */
302 u32 decode_size = AUDIO_DECODE_SIZE * AUDIO_CHANNELS;
303 vg_audio.decode_buffer = vg_linear_alloc( vg_mem.rtmemory, decode_size );
304
305 vg_dsp_init();
306
307 SDL_AudioSpec spec_desired, spec_got;
308 spec_desired.callback = audio_mixer_callback;
309 spec_desired.channels = 2;
310 spec_desired.format = AUDIO_F32;
311 spec_desired.freq = 44100;
312 spec_desired.padding = 0;
313 spec_desired.samples = AUDIO_FRAME_SIZE;
314 spec_desired.silence = 0;
315 spec_desired.size = 0;
316 spec_desired.userdata = NULL;
317
318 vg_audio.sdl_output_device =
319 SDL_OpenAudioDevice( NULL, 0, &spec_desired, &spec_got,0 );
320
321 if( vg_audio.sdl_output_device )
322 {
323 SDL_PauseAudioDevice( vg_audio.sdl_output_device, 0 );
324 }
325 else
326 {
327 vg_fatal_exit_loop(
328 "SDL_OpenAudioDevice failed. Your default audio device must support:\n"
329 " Frequency: 44100 hz\n"
330 " Buffer size: 512\n"
331 " Channels: 2\n"
332 " Format: s16 or f32\n" );
333 }
334
335 vg_success( "Ready\n" );
336 }
337
338 VG_STATIC void vg_audio_free(void)
339 {
340 vg_dsp_free();
341 SDL_CloseAudioDevice( vg_audio.sdl_output_device );
342 }
343
344 /*
345 * thread 1
346 */
347
348 #define AUDIO_EDIT_VOLUME_SLOPE 0x1
349 #define AUDIO_EDIT_VOLUME 0x2
350 #define AUDIO_EDIT_LFO_PERIOD 0x4
351 #define AUDIO_EDIT_LFO_WAVE 0x8
352 #define AUDIO_EDIT_LFO_ATTACHMENT 0x10
353 #define AUDIO_EDIT_SPACIAL 0x20
354 #define AUDIO_EDIT_OWNERSHIP 0x40
355 #define AUDIO_EDIT_SAMPLING_RATE 0x80
356
357 static audio_channel *audio_request_channel( audio_clip *clip, u32 flags )
358 {
359 for( int i=0; i<AUDIO_CHANNELS; i++ )
360 {
361 audio_channel *ch = &vg_audio.channels[i];
362
363 if( !ch->allocated )
364 {
365 ch->source = clip;
366 ch->flags = flags;
367 ch->colour = 0x00333333;
368
369 if( (ch->source->flags & AUDIO_FLAG_FORMAT) == k_audio_format_bird )
370 strcpy( ch->name, "[array]" );
371 else
372 strncpy( ch->name, clip->path, 31 );
373
374 ch->allocated = 1;
375
376 ch->editable_state.relinquished = 0;
377 ch->editable_state.volume = 1.0f;
378 ch->editable_state.volume_target = 1.0f;
379 ch->editable_state.pan = 0.0f;
380 ch->editable_state.pan_target = 0.0f;
381 ch->editable_state.volume_rate = 0;
382 ch->editable_state.pan_rate = 0;
383 v4_copy((v4f){0.0f,0.0f,0.0f,1.0f},ch->editable_state.spacial_falloff);
384 ch->editable_state.lfo = NULL;
385 ch->editable_state.lfo_amount = 0.0f;
386 ch->editable_state.sampling_rate = 1.0f;
387 ch->editble_state_write_mask = 0x00;
388 return ch;
389 }
390 }
391
392 return NULL;
393 }
394
395 static int audio_channel_finished( audio_channel *ch )
396 {
397 if( ch->readable_activity == k_channel_activity_end )
398 return 1;
399 else
400 return 0;
401 }
402
403 static audio_channel *audio_relinquish_channel( audio_channel *ch )
404 {
405 ch->editable_state.relinquished = 1;
406 ch->editble_state_write_mask |= AUDIO_EDIT_OWNERSHIP;
407 return NULL;
408 }
409
410 static void audio_channel_slope_volume( audio_channel *ch, float length,
411 float new_volume )
412 {
413 ch->editable_state.volume_target = new_volume;
414 ch->editable_state.volume_rate = length * 44100.0f;
415 ch->editble_state_write_mask |= AUDIO_EDIT_VOLUME_SLOPE;
416 }
417
418 static void audio_channel_set_sampling_rate( audio_channel *ch, float rate )
419 {
420 ch->editable_state.sampling_rate = rate;
421 ch->editble_state_write_mask |= AUDIO_EDIT_SAMPLING_RATE;
422 }
423
424 static void audio_channel_edit_volume( audio_channel *ch,
425 float new_volume, int instant )
426 {
427 if( instant )
428 {
429 ch->editable_state.volume = new_volume;
430 ch->editble_state_write_mask |= AUDIO_EDIT_VOLUME;
431 }
432 else
433 {
434 audio_channel_slope_volume( ch, 0.05f, new_volume );
435 }
436 }
437
438 static audio_channel *audio_channel_fadeout( audio_channel *ch, float length )
439 {
440 audio_channel_slope_volume( ch, length, 0.0f );
441 return audio_relinquish_channel( ch );
442 }
443
444 static void audio_channel_fadein( audio_channel *ch, float length )
445 {
446 audio_channel_edit_volume( ch, 0.0f, 1 );
447 audio_channel_slope_volume( ch, length, 1.0f );
448 }
449
450 static audio_channel *audio_channel_crossfade( audio_channel *ch,
451 audio_clip *new_clip,
452 float length, u32 flags )
453 {
454 u32 cursor = 0;
455
456 if( ch )
457 ch = audio_channel_fadeout( ch, length );
458
459 audio_channel *replacement = audio_request_channel( new_clip, flags );
460
461 if( replacement )
462 audio_channel_fadein( replacement, length );
463
464 return replacement;
465 }
466
467 static void audio_channel_sidechain_lfo( audio_channel *ch, int lfo_id,
468 float amount )
469 {
470 ch->editable_state.lfo = &vg_audio.oscillators[ lfo_id ];
471 ch->editable_state.lfo_amount = amount;
472 ch->editble_state_write_mask |= AUDIO_EDIT_LFO_ATTACHMENT;
473 }
474
475 static void audio_channel_set_spacial( audio_channel *ch, v3f co, float range )
476 {
477 if( ch->flags & AUDIO_FLAG_SPACIAL_3D ){
478 v3_copy( co, ch->editable_state.spacial_falloff );
479
480 if( range == 0.0f )
481 ch->editable_state.spacial_falloff[3] = 1.0f;
482 else
483 ch->editable_state.spacial_falloff[3] = 1.0f/range;
484
485 ch->editble_state_write_mask |= AUDIO_EDIT_SPACIAL;
486 }
487 else{
488 vg_warn( "Tried to set spacialization paramaters for 2D channel (%s)\n",
489 ch->name );
490 }
491 }
492
493 static int audio_oneshot_3d( audio_clip *clip, v3f position,
494 float range, float volume )
495 {
496 audio_channel *ch = audio_request_channel( clip, AUDIO_FLAG_SPACIAL_3D );
497
498 if( ch ){
499 audio_channel_set_spacial( ch, position, range );
500 audio_channel_edit_volume( ch, volume, 1 );
501 ch = audio_relinquish_channel( ch );
502
503 return 1;
504 }
505 else
506 return 0;
507 }
508
509 static int audio_oneshot( audio_clip *clip, float volume, float pan )
510 {
511 audio_channel *ch = audio_request_channel( clip, 0x00 );
512
513 if( ch ){
514 audio_channel_edit_volume( ch, volume, 1 );
515 ch = audio_relinquish_channel( ch );
516
517 return 1;
518 }
519 else
520 return 0;
521 }
522
523 static void audio_set_lfo_wave( int id, enum lfo_wave_type type,
524 float coefficient )
525 {
526 audio_lfo *lfo = &vg_audio.oscillators[ id ];
527 lfo->editable_state.polynomial_coefficient = coefficient;
528 lfo->editable_state.wave_type = type;
529
530 lfo->editble_state_write_mask |= AUDIO_EDIT_LFO_WAVE;
531 }
532
533 static void audio_set_lfo_frequency( int id, float freq )
534 {
535 audio_lfo *lfo = &vg_audio.oscillators[ id ];
536 lfo->editable_state.period = 44100.0f / freq;
537 lfo->editble_state_write_mask |= AUDIO_EDIT_LFO_PERIOD;
538 }
539
540
541 /*
542 * Committers
543 * -----------------------------------------------------------------------------
544 */
545 static int audio_channel_load_source( audio_channel *ch )
546 {
547 u32 format = ch->source->flags & AUDIO_FLAG_FORMAT;
548
549 if( format == k_audio_format_vorbis )
550 {
551 /* Setup vorbis decoder */
552 u32 index = ch - vg_audio.channels;
553
554 u8 *buf = (u8*)vg_audio.decode_buffer,
555 *loc = &buf[AUDIO_DECODE_SIZE*index];
556
557 stb_vorbis_alloc alloc = {
558 .alloc_buffer = (char *)loc,
559 .alloc_buffer_length_in_bytes = AUDIO_DECODE_SIZE
560 };
561
562 int err;
563 stb_vorbis *decoder = stb_vorbis_open_memory(
564 ch->source->data,
565 ch->source->size, &err, &alloc );
566
567 if( !decoder )
568 {
569 vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
570 ch->source->path, err );
571 return 0;
572 }
573 else
574 {
575 ch->source_length = stb_vorbis_stream_length_in_samples( decoder );
576 ch->vorbis_handle = decoder;
577 }
578 }
579 else if( format == k_audio_format_bird )
580 {
581 u32 index = ch - vg_audio.channels;
582
583 u8 *buf = (u8*)vg_audio.decode_buffer;
584 struct synth_bird *loc = (void *)&buf[AUDIO_DECODE_SIZE*index];
585
586 memcpy( loc, ch->source->data, ch->source->size );
587 synth_bird_reset( loc );
588
589 ch->bird_handle = loc;
590 ch->source_length = synth_bird_get_length_in_samples( loc );
591 }
592 else if( format == k_audio_format_stereo )
593 {
594 ch->source_length = ch->source->size / 2;
595 }
596 else
597 {
598 ch->source_length = ch->source->size;
599 }
600
601 return 1;
602 }
603
604 VG_STATIC void audio_decode_uncompressed_mono( i16 *src, u32 count, float *dst )
605 {
606 for( u32 i=0; i<count; i++ )
607 {
608 dst[ i*2 + 0 ] = ((float)src[i]) * (1.0f/32767.0f);
609 dst[ i*2 + 1 ] = ((float)src[i]) * (1.0f/32767.0f);
610 }
611 }
612
613 /*
614 * adapted from stb_vorbis.h, since the original does not handle mono->stereo
615 */
616 VG_STATIC int
617 stb_vorbis_get_samples_float_interleaved_stereo( stb_vorbis *f, float *buffer,
618 int len )
619 {
620 int n = 0,
621 c = VG_MIN( 1, f->channels - 1 );
622
623 while( n < len )
624 {
625 int k = f->channel_buffer_end - f->channel_buffer_start;
626
627 if( n+k >= len )
628 k = len - n;
629
630 for( int j=0; j < k; ++j )
631 {
632 *buffer++ = f->channel_buffers[ 0 ][f->channel_buffer_start+j];
633 *buffer++ = f->channel_buffers[ c ][f->channel_buffer_start+j];
634 }
635
636 n += k;
637 f->channel_buffer_start += k;
638
639 if( n == len )
640 break;
641
642 if( !stb_vorbis_get_frame_float( f, NULL, NULL ))
643 break;
644 }
645
646 return n;
647 }
648
649 /*
650 * ........ more wrecked code sorry!
651 */
652 VG_STATIC int
653 stb_vorbis_get_samples_i16_downmixed( stb_vorbis *f, i16 *buffer, int len )
654 {
655 int n = 0,
656 c = VG_MIN( 1, f->channels - 1 );
657
658 while( n < len ) {
659 int k = f->channel_buffer_end - f->channel_buffer_start;
660
661 if( n+k >= len )
662 k = len - n;
663
664 for( int j=0; j < k; ++j ) {
665 float sl = f->channel_buffers[ 0 ][f->channel_buffer_start+j],
666 sr = f->channel_buffers[ c ][f->channel_buffer_start+j];
667
668 *buffer++ = vg_clampf( 0.5f*(sl+sr), -1.0f, 1.0f ) * 32767.0f;
669 //*buffer++ = vg_clampf( sr, -1.0f, 1.0f ) * 32767.0f;
670 }
671
672 n += k;
673 f->channel_buffer_start += k;
674
675 if( n == len )
676 break;
677
678 if( !stb_vorbis_get_frame_float( f, NULL, NULL ))
679 break;
680 }
681
682 return n;
683 }
684
685 static inline float audio_lfo_pull_sample( audio_lfo *lfo )
686 {
687 lfo->time ++;
688
689 if( lfo->time >= lfo->_.period )
690 lfo->time = 0;
691
692 float t = lfo->time;
693 t /= (float)lfo->_.period;
694
695 if( lfo->_.wave_type == k_lfo_polynomial_bipolar ){
696 /*
697 * #
698 * # #
699 * # #
700 * # #
701 * ### # ###
702 * ## #
703 * # #
704 * # #
705 * ##
706 */
707
708 t *= 2.0f;
709 t -= 1.0f;
710
711 return (( 2.0f * lfo->sqrt_polynomial_coefficient * t ) /
712 /* --------------------------------------- */
713 ( 1.0f + lfo->_.polynomial_coefficient * t*t )
714
715 ) * (1.0f-fabsf(t));
716 }
717 else{
718 return 0.0f;
719 }
720 }
721
722 static void audio_channel_get_samples( audio_channel *ch,
723 u32 count, float *buf )
724 {
725 vg_profile_begin( &_vg_prof_audio_decode );
726
727 u32 remaining = count;
728 u32 buffer_pos = 0;
729
730 u32 format = ch->source->flags & AUDIO_FLAG_FORMAT;
731
732 while( remaining ){
733 u32 samples_this_run = VG_MIN(remaining, ch->source_length - ch->cursor);
734 remaining -= samples_this_run;
735
736 float *dst = &buf[ buffer_pos * 2 ];
737
738 if( format == k_audio_format_stereo ){
739 for( int i=0;i<samples_this_run; i++ ){
740 dst[i*2+0] = 0.0f;
741 dst[i*2+1] = 0.0f;
742 }
743 }
744 else if( format == k_audio_format_vorbis ){
745 int read_samples = stb_vorbis_get_samples_float_interleaved_stereo(
746 ch->vorbis_handle,
747 dst,
748 samples_this_run );
749
750 if( read_samples != samples_this_run ){
751 vg_warn( "Invalid samples read (%s)\n", ch->source->path );
752
753 for( int i=0; i<samples_this_run; i++ ){
754 dst[i*2+0] = 0.0f;
755 dst[i*2+1] = 0.0f;
756 }
757 }
758 }
759 else if( format == k_audio_format_bird ){
760 synth_bird_generate_samples( ch->bird_handle, dst, samples_this_run );
761 }
762 else{
763 i16 *src_buffer = ch->source->data,
764 *src = &src_buffer[ch->cursor];
765
766 audio_decode_uncompressed_mono( src, samples_this_run, dst );
767 }
768
769 ch->cursor += samples_this_run;
770 buffer_pos += samples_this_run;
771
772 if( (ch->flags & AUDIO_FLAG_LOOP) && remaining ){
773 if( format == k_audio_format_vorbis )
774 stb_vorbis_seek_start( ch->vorbis_handle );
775 else if( format == k_audio_format_bird )
776 synth_bird_reset( ch->bird_handle );
777
778 ch->cursor = 0;
779 continue;
780 }
781 else
782 break;
783 }
784
785 while( remaining ){
786 buf[ buffer_pos*2 + 0 ] = 0.0f;
787 buf[ buffer_pos*2 + 1 ] = 0.0f;
788 buffer_pos ++;
789
790 remaining --;
791 }
792
793 vg_profile_end( &_vg_prof_audio_decode );
794 }
795
796 static void audio_channel_mix( audio_channel *ch, float *buffer )
797 {
798 float framevol_l = 1.0f,
799 framevol_r = 1.0f;
800
801 float frame_samplerate = ch->_.sampling_rate;
802
803 if( ch->flags & AUDIO_FLAG_SPACIAL_3D ){
804 v3f delta;
805 v3_sub( ch->_.spacial_falloff, vg_audio.listener_pos, delta );
806
807 float dist = v3_length( delta ),
808 vol = vg_maxf( 0.0f, 1.0f - ch->_.spacial_falloff[3]*dist );
809
810 if( dist <= 0.01f ){
811
812 }
813 else{
814 v3_muls( delta, 1.0f/dist, delta );
815 float pan = v3_dot( vg_audio.listener_ears, delta );
816 vol = powf( vol, 5.0f );
817
818 framevol_l *= (vol * 0.5f) * (1.0f - pan);
819 framevol_r *= (vol * 0.5f) * (1.0f + pan);
820
821 const float vs = 100.0f;
822 float doppler = (vs+v3_dot(delta,vg_audio.listener_velocity))/vs;
823 doppler = vg_clampf( doppler, 0.6f, 1.4f );
824
825 if( fabsf(doppler-1.0f) > 0.01f )
826 frame_samplerate *= doppler;
827 }
828
829 if( !vg_validf( framevol_l ) ) vg_fatal_exit_loop( "NaN left channel" );
830 if( !vg_validf( framevol_r ) ) vg_fatal_exit_loop( "NaN right channel" );
831 if( !vg_validf( frame_samplerate ) )
832 vg_fatal_exit_loop( "NaN sample rate" );
833 }
834
835 u32 buffer_length = AUDIO_MIX_FRAME_SIZE;
836 if( frame_samplerate != 1.0f ){
837 float l = ceilf( (float)(AUDIO_MIX_FRAME_SIZE) * frame_samplerate );
838 buffer_length = l+1;
839 }
840
841 float pcf[ AUDIO_MIX_FRAME_SIZE * 2 * 2 ];
842
843 audio_channel_get_samples( ch, buffer_length, pcf );
844
845 vg_profile_begin( &_vg_prof_audio_mix );
846
847 float volume_movement = ch->volume_movement;
848 float const fvolume_rate = vg_maxf( 1.0f, ch->_.volume_rate );
849 const float inv_volume_rate = 1.0f/fvolume_rate;
850
851 float volume = ch->_.volume;
852 const float volume_start = ch->volume_movement_start;
853 const float volume_target = ch->_.volume_target;
854
855 for( u32 j=0; j<AUDIO_MIX_FRAME_SIZE; j++ ){
856 /*
857 * there is some REALLY weird behaviour with minss,
858 * i cannot begin to guess what the cause is, but the bahaviour when
859 * the second argument is not 1.0 would seemingly tripple or up to
860 * eight times this routine.
861 *
862 * the times it would happen are when moving from empty space into areas
863 * with geometry. in the bvh for skate rift.
864 *
865 * it should be completely unrelated to this, but somehow -- it is
866 * effecting the speed of minss. and severely at that too.
867 **/
868
869 volume_movement += 1.0f;
870 float movement_t = volume_movement * inv_volume_rate;
871 movement_t = vg_minf( volume_movement, 1.0f );
872 volume = vg_lerpf( volume_start, volume_target, movement_t );
873
874 float vol_norm = volume * volume;
875
876 if( ch->_.lfo )
877 vol_norm *= 1.0f + audio_lfo_pull_sample(ch->_.lfo) * ch->_.lfo_amount;
878
879 float vol_l = vol_norm * framevol_l,
880 vol_r = vol_norm * framevol_r,
881 sample_l,
882 sample_r;
883
884 if( frame_samplerate != 1.0f )
885 {
886 /* absolutely garbage resampling, but it will do
887 */
888
889 float sample_index = frame_samplerate * (float)j;
890 float t = vg_fractf( sample_index );
891
892 u32 i0 = floorf( sample_index ),
893 i1 = i0+1;
894
895 sample_l = pcf[ i0*2+0 ]*(1.0f-t) + pcf[ i1*2+0 ]*t;
896 sample_r = pcf[ i0*2+1 ]*(1.0f-t) + pcf[ i1*2+1 ]*t;
897 }
898 else
899 {
900 sample_l = pcf[ j*2+0 ];
901 sample_r = pcf[ j*2+1 ];
902 }
903
904 buffer[ j*2+0 ] += sample_l * vol_l;
905 buffer[ j*2+1 ] += sample_r * vol_r;
906 }
907
908 ch->volume_movement += AUDIO_MIX_FRAME_SIZE;
909 ch->volume_movement = VG_MIN( ch->volume_movement, ch->_.volume_rate );
910 ch->_.volume = volume;
911
912 vg_profile_end( &_vg_prof_audio_mix );
913 }
914
915 VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count )
916 {
917 /*
918 * Copy data and move edit flags to commit flags
919 * ------------------------------------------------------------- */
920 audio_lock();
921 for( int i=0; i<AUDIO_CHANNELS; i++ ){
922 audio_channel *ch = &vg_audio.channels[i];
923
924 if( !ch->allocated )
925 continue;
926
927 if( ch->activity == k_channel_activity_alive ){
928 if( (ch->cursor >= ch->source_length) &&
929 !(ch->flags & AUDIO_FLAG_LOOP) )
930 {
931 ch->activity = k_channel_activity_end;
932 }
933 }
934
935 /* process relinquishments */
936 if( (ch->activity != k_channel_activity_reset) && ch->_.relinquished ){
937 if( (ch->activity == k_channel_activity_end)
938 || (ch->_.volume == 0.0f)
939 || (ch->activity == k_channel_activity_error) )
940 {
941 ch->_.relinquished = 0;
942 ch->allocated = 0;
943 ch->activity = k_channel_activity_reset;
944 continue;
945 }
946 }
947
948 /* process new channels */
949 if( ch->activity == k_channel_activity_reset ){
950 ch->_ = ch->editable_state;
951 ch->cursor = 0;
952 ch->source_length = 0;
953 ch->activity = k_channel_activity_wake;
954 }
955
956 if( ch->editble_state_write_mask & AUDIO_EDIT_OWNERSHIP )
957 ch->_.relinquished = ch->editable_state.relinquished;
958 else
959 ch->editable_state.relinquished = ch->_.relinquished;
960
961
962 if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME ){
963 ch->_.volume = ch->editable_state.volume;
964 ch->_.volume_target = ch->editable_state.volume;
965 }
966 else{
967 ch->editable_state.volume = ch->_.volume;
968 }
969
970
971 if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME_SLOPE ){
972 ch->volume_movement_start = ch->_.volume;
973 ch->volume_movement = 0;
974
975 ch->_.volume_target = ch->editable_state.volume_target;
976 ch->_.volume_rate = ch->editable_state.volume_rate;
977 }
978 else{
979 ch->editable_state.volume_target = ch->_.volume_target;
980 ch->editable_state.volume_rate = ch->_.volume_rate;
981 }
982
983
984 if( ch->editble_state_write_mask & AUDIO_EDIT_SAMPLING_RATE )
985 ch->_.sampling_rate = ch->editable_state.sampling_rate;
986 else
987 ch->editable_state.sampling_rate = ch->_.sampling_rate;
988
989
990 if( ch->editble_state_write_mask & AUDIO_EDIT_LFO_ATTACHMENT ){
991 ch->_.lfo = ch->editable_state.lfo;
992 ch->_.lfo_amount = ch->editable_state.lfo_amount;
993 }
994 else{
995 ch->editable_state.lfo = ch->_.lfo;
996 ch->editable_state.lfo_amount = ch->_.lfo_amount;
997 }
998
999
1000 if( ch->editble_state_write_mask & AUDIO_EDIT_SPACIAL )
1001 v4_copy( ch->editable_state.spacial_falloff,ch->_.spacial_falloff );
1002 else
1003 v4_copy( ch->_.spacial_falloff,ch->editable_state.spacial_falloff );
1004
1005
1006 /* currently readonly, i guess */
1007 ch->editable_state.pan_target = ch->_.pan_target;
1008 ch->editable_state.pan = ch->_.pan;
1009 ch->editble_state_write_mask = 0x00;
1010 }
1011
1012 for( int i=0; i<AUDIO_LFOS; i++ ){
1013 audio_lfo *lfo = &vg_audio.oscillators[ i ];
1014
1015 if( lfo->editble_state_write_mask & AUDIO_EDIT_LFO_WAVE ){
1016 lfo->_.wave_type = lfo->editable_state.wave_type;
1017
1018 if( lfo->_.wave_type == k_lfo_polynomial_bipolar )
1019 {
1020 lfo->_.polynomial_coefficient =
1021 lfo->editable_state.polynomial_coefficient;
1022 lfo->sqrt_polynomial_coefficient =
1023 sqrtf(lfo->_.polynomial_coefficient);
1024 }
1025 }
1026
1027 if( lfo->editble_state_write_mask & AUDIO_EDIT_LFO_PERIOD ){
1028 if( lfo->_.period ){
1029 float t = lfo->time;
1030 t/= (float)lfo->_.period;
1031
1032 lfo->_.period = lfo->editable_state.period;
1033 lfo->time = lfo->_.period * t;
1034 }
1035 else{
1036 lfo->time = 0;
1037 lfo->_.period = lfo->editable_state.period;
1038 }
1039 }
1040
1041 lfo->editble_state_write_mask = 0x00;
1042 }
1043
1044 dsp_update_tunings();
1045 audio_unlock();
1046
1047 /*
1048 * Process spawns
1049 * ------------------------------------------------------------- */
1050 for( int i=0; i<AUDIO_CHANNELS; i++ )
1051 {
1052 audio_channel *ch = &vg_audio.channels[i];
1053
1054 if( ch->activity == k_channel_activity_wake )
1055 {
1056 if( audio_channel_load_source( ch ) )
1057 ch->activity = k_channel_activity_alive;
1058 else
1059 ch->activity = k_channel_activity_error;
1060 }
1061 }
1062
1063 /*
1064 * Mix everything
1065 * -------------------------------------------------------- */
1066 int frame_count = byte_count/(2*sizeof(float));
1067
1068 /* Clear buffer */
1069 float *pOut32F = (float *)stream;
1070 for( int i=0; i<frame_count*2; i ++ )
1071 pOut32F[i] = 0.0f;
1072
1073 for( int i=0; i<AUDIO_LFOS; i++ )
1074 {
1075 audio_lfo *lfo = &vg_audio.oscillators[i];
1076 lfo->time_startframe = lfo->time;
1077 }
1078
1079 for( int i=0; i<AUDIO_CHANNELS; i ++ )
1080 {
1081 audio_channel *ch = &vg_audio.channels[i];
1082
1083 if( ch->activity == k_channel_activity_alive )
1084 {
1085 if( ch->_.lfo )
1086 ch->_.lfo->time = ch->_.lfo->time_startframe;
1087
1088 u32 remaining = frame_count,
1089 subpos = 0;
1090
1091 while( remaining )
1092 {
1093 audio_channel_mix( ch, pOut32F+subpos );
1094 remaining -= AUDIO_MIX_FRAME_SIZE;
1095 subpos += AUDIO_MIX_FRAME_SIZE*2;
1096 }
1097 }
1098 }
1099
1100 vg_profile_begin( &_vg_prof_dsp );
1101
1102 for( int i=0; i<frame_count; i++ )
1103 vg_dsp_process( pOut32F + i*2, pOut32F + i*2 );
1104
1105 vg_profile_end( &_vg_prof_dsp );
1106
1107 audio_lock();
1108
1109 for( int i=0; i<AUDIO_CHANNELS; i ++ )
1110 {
1111 audio_channel *ch = &vg_audio.channels[i];
1112 ch->readable_activity = ch->activity;
1113 }
1114
1115 /* Profiling information
1116 * ----------------------------------------------- */
1117 vg_profile_increment( &_vg_prof_audio_decode );
1118 vg_profile_increment( &_vg_prof_audio_mix );
1119 vg_profile_increment( &_vg_prof_dsp );
1120
1121 vg_prof_audio_mix = _vg_prof_audio_mix;
1122 vg_prof_audio_decode = _vg_prof_audio_decode;
1123 vg_prof_audio_dsp = _vg_prof_dsp;
1124
1125 vg_audio.samples_last = frame_count;
1126
1127 if( vg_audio.debug_ui )
1128 {
1129 vg_dsp_update_texture();
1130 }
1131
1132 audio_unlock();
1133 }
1134
1135 VG_STATIC void audio_clip_load( audio_clip *clip, void *lin_alloc )
1136 {
1137 if( lin_alloc == NULL )
1138 lin_alloc = vg_audio.audio_pool;
1139
1140 /* load in directly */
1141 u32 format = clip->flags & AUDIO_FLAG_FORMAT;
1142
1143 /* TODO: This contains audio_lock() and unlock, but i don't know why
1144 * can probably remove them. Low priority to check this */
1145
1146 if( format == k_audio_format_vorbis )
1147 {
1148 audio_lock();
1149 clip->data = vg_file_read( lin_alloc, clip->path, &clip->size );
1150 audio_unlock();
1151
1152 if( !clip->data )
1153 vg_fatal_exit_loop( "Audio failed to load" );
1154
1155 float mb = (float)(clip->size) / (1024.0f*1024.0f);
1156 vg_info( "Loaded audio clip '%s' (%.1fmb)\n", clip->path, mb );
1157 }
1158 else if( format == k_audio_format_stereo )
1159 {
1160 vg_fatal_exit_loop( "Unsupported format (Stereo uncompressed)" );
1161 }
1162 else if( format == k_audio_format_bird )
1163 {
1164 u32 len = strlen( clip->path ),
1165 size = synth_bird_memory_requirement( len );
1166
1167 if( size > AUDIO_DECODE_SIZE )
1168 vg_fatal_exit_loop( "Bird code too long\n" );
1169
1170 clip->size = size;
1171 clip->data = vg_linear_alloc( lin_alloc, size );
1172
1173 synth_bird_load( clip->data, clip->path, len );
1174 }
1175 else
1176 {
1177 vg_linear_clear( vg_mem.scratch );
1178 u32 fsize;
1179
1180 stb_vorbis_alloc alloc = {
1181 .alloc_buffer = vg_linear_alloc( vg_mem.scratch, AUDIO_DECODE_SIZE ),
1182 .alloc_buffer_length_in_bytes = AUDIO_DECODE_SIZE
1183 };
1184
1185 void *filedata = vg_file_read( vg_mem.scratch, clip->path, &fsize );
1186
1187 int err;
1188 stb_vorbis *decoder = stb_vorbis_open_memory(
1189 filedata, fsize, &err, &alloc );
1190
1191 if( !decoder )
1192 {
1193 vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
1194 clip->path, err );
1195 vg_fatal_exit_loop( "Vorbis decode error" );
1196 }
1197
1198 /* only mono is supported in uncompressed */
1199 u32 length_samples = stb_vorbis_stream_length_in_samples( decoder ),
1200 data_size = length_samples * sizeof(i16);
1201
1202 audio_lock();
1203 clip->data = vg_linear_alloc( lin_alloc, vg_align8(data_size) );
1204 clip->size = length_samples;
1205 audio_unlock();
1206
1207 int read_samples = stb_vorbis_get_samples_i16_downmixed(
1208 decoder, clip->data, length_samples );
1209
1210 if( read_samples != length_samples )
1211 vg_fatal_exit_loop( "Decode error" );
1212
1213 float mb = (float)(data_size) / (1024.0f*1024.0f);
1214 vg_info( "Loaded audio clip '%s' (%.1fmb) %u samples\n", clip->path, mb,
1215 length_samples );
1216 }
1217 }
1218
1219 VG_STATIC void audio_clip_loadn( audio_clip *arr, int count, void *lin_alloc )
1220 {
1221 for( int i=0; i<count; i++ )
1222 audio_clip_load( &arr[i], lin_alloc );
1223 }
1224
1225 VG_STATIC void audio_require_clip_loaded( audio_clip *clip )
1226 {
1227 if( clip->data && clip->size )
1228 return;
1229
1230 audio_unlock();
1231 vg_fatal_exit_loop( "Must load audio clip before playing! \n" );
1232 }
1233
1234 /*
1235 * Debugging
1236 */
1237
1238 VG_STATIC void audio_debug_ui( m4x4f mtx_pv )
1239 {
1240 if( !vg_audio.debug_ui )
1241 return;
1242
1243 audio_lock();
1244
1245 glBindTexture( GL_TEXTURE_2D, vg_dsp.view_texture );
1246 glTexSubImage2D( GL_TEXTURE_2D, 0, 0, 0, 256, 256,
1247 GL_RGBA, GL_UNSIGNED_BYTE,
1248 vg_dsp.view_texture_buffer );
1249
1250 /*
1251 * Profiler
1252 * -----------------------------------------------------------------------
1253 */
1254
1255 float budget = ((double)vg_audio.samples_last / 44100.0) * 1000.0;
1256 vg_profile_drawn( (struct vg_profile *[]){ &vg_prof_audio_decode,
1257 &vg_prof_audio_mix,
1258 &vg_prof_audio_dsp}, 3,
1259 budget, (ui_rect){ 4, VG_PROFILE_SAMPLE_COUNT*2 + 8,
1260 512, 0 }, 3 );
1261
1262
1263 char perf[128];
1264
1265 /* Draw UI */
1266 vg_uictx.cursor[0] = 512 + 8;
1267 vg_uictx.cursor[1] = VG_PROFILE_SAMPLE_COUNT*2+8+24+12+12;
1268 vg_uictx.cursor[2] = 150;
1269 vg_uictx.cursor[3] = 12;
1270
1271 ui_rect view_thing = { 4, vg.window_y-512-4, 512, 512 };
1272 ui_push_image( view_thing, vg_dsp.view_texture );
1273
1274 float mb1 = 1024.0f*1024.0f,
1275 usage = vg_linear_get_cur( vg_audio.audio_pool ) / mb1,
1276 total = vg_linear_get_capacity( vg_audio.audio_pool ) / mb1,
1277 percent = (usage/total) * 100.0f;
1278
1279 snprintf( perf, 127, "Mem: %.1f/%.1fmb (%.1f%%)\n", usage, total, percent );
1280
1281 ui_text( vg_uictx.cursor, perf, 1, 0 );
1282 vg_uictx.cursor[1] += 20;
1283
1284 ui_rect overlap_buffer[ AUDIO_CHANNELS ];
1285 u32 overlap_length = 0;
1286
1287 /* Draw audio stack */
1288 for( int i=0; i<AUDIO_CHANNELS; i ++ )
1289 {
1290 audio_channel *ch = &vg_audio.channels[i];
1291
1292 vg_uictx.cursor[2] = 400;
1293 vg_uictx.cursor[3] = 18;
1294
1295 ui_new_node();
1296
1297 if( !ch->allocated )
1298 {
1299 ui_fill_rect( vg_uictx.cursor, 0x50333333 );
1300
1301 ui_end_down();
1302 vg_uictx.cursor[1] += 1;
1303 continue;
1304 }
1305
1306 const char *formats[] =
1307 {
1308 " mono ",
1309 " stereo ",
1310 " vorbis ",
1311 " none0 ",
1312 " none1 ",
1313 " none2 ",
1314 " none3 ",
1315 " none4 ",
1316 "synth:bird",
1317 " none5 ",
1318 " none6 ",
1319 " none7 ",
1320 " none8 ",
1321 " none9 ",
1322 " none10 ",
1323 " none11 ",
1324 };
1325
1326 const char *activties[] =
1327 {
1328 "reset",
1329 "wake ",
1330 "alive",
1331 "end ",
1332 "error"
1333 };
1334
1335 u32 format_index = (ch->source->flags & AUDIO_FLAG_FORMAT)>>9;
1336
1337 snprintf( perf, 127, "%02d %c%c%cD %s [%s] %4.2fv'%s'",
1338 i,
1339 (ch->editable_state.relinquished)? 'r': '_',
1340 0? 'r': '_',
1341 0? '3': '2',
1342 formats[format_index],
1343 activties[ch->readable_activity],
1344 ch->editable_state.volume,
1345 ch->name );
1346
1347 ui_fill_rect( vg_uictx.cursor, 0xa0000000 | ch->colour );
1348
1349 vg_uictx.cursor[0] += 2;
1350 vg_uictx.cursor[1] += 2;
1351 ui_text( vg_uictx.cursor, perf, 1, 0 );
1352
1353 ui_end_down();
1354 vg_uictx.cursor[1] += 1;
1355
1356 if( AUDIO_FLAG_SPACIAL_3D )
1357 {
1358 v4f wpos;
1359 v3_copy( ch->editable_state.spacial_falloff, wpos );
1360
1361 wpos[3] = 1.0f;
1362 m4x4_mulv( mtx_pv, wpos, wpos );
1363
1364 if( wpos[3] > 0.0f )
1365 {
1366 v2_muls( wpos, (1.0f/wpos[3]) * 0.5f, wpos );
1367 v2_add( wpos, (v2f){ 0.5f, 0.5f }, wpos );
1368
1369 ui_rect wr;
1370 wr[0] = wpos[0] * vg.window_x;
1371 wr[1] = (1.0f-wpos[1]) * vg.window_y;
1372 wr[2] = 100;
1373 wr[3] = 17;
1374
1375 for( int j=0; j<12; j++ )
1376 {
1377 int collide = 0;
1378 for( int k=0; k<overlap_length; k++ )
1379 {
1380 ui_px *wk = overlap_buffer[k];
1381 if( ((wr[0] <= wk[0]+wk[2]) && (wr[0]+wr[2] >= wk[0])) &&
1382 ((wr[1] <= wk[1]+wk[3]) && (wr[1]+wr[3] >= wk[1])) )
1383 {
1384 collide = 1;
1385 break;
1386 }
1387 }
1388
1389 if( !collide )
1390 break;
1391 else
1392 wr[1] += 18;
1393 }
1394
1395 ui_text( wr, perf, 1, 0 );
1396
1397 ui_rect_copy( wr, overlap_buffer[ overlap_length ++ ] );
1398 }
1399 }
1400 }
1401
1402 audio_unlock();
1403 }
1404
1405 #endif /* VG_AUDIO_H */