1 /* Copyright (C) 2021-2023 Harry Godden (hgn) - All Rights Reserved */
9 #include "vg/vg_stdint.h"
10 #include "vg/vg_platform.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"
20 #pragma GCC push_options
21 #pragma GCC optimize ("O3")
22 #pragma GCC diagnostic push
23 #pragma GCC diagnostic ignored "-Wdeprecated-declarations"
27 #define STB_VORBIS_MAX_CHANNELS 2
28 #include "submodules/stb/stb_vorbis.c"
35 #pragma GCC pop_options
36 #pragma GCC diagnostic pop
40 #define AUDIO_FRAME_SIZE 512
41 #define AUDIO_MIX_FRAME_SIZE 256
43 #define AUDIO_CHANNELS 32
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
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
,
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
,
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
77 typedef struct audio_clip audio_clip
;
78 typedef struct audio_channel audio_channel
;
79 typedef struct audio_lfo audio_lfo
;
82 union { /* TODO oof.. */
97 struct vg_audio_system
{
98 SDL_AudioDeviceID sdl_output_device
;
107 SDL_SpinLock sl_checker
,
111 u32 time
, time_startframe
;
112 float sqrt_polynomial_coefficient
;
119 k_lfo_polynomial_bipolar
124 float polynomial_coefficient
;
127 u32 editble_state_write_mask
;
129 oscillators
[ AUDIO_LFOS
];
131 struct audio_channel
{
136 char name
[32]; /* only editable while allocated == 0 */
137 audio_clip
*source
; /* ... */
139 u32 colour
; /* ... */
141 /* internal non-readable state
142 * -----------------------------*/
143 u32 cursor
, source_length
;
145 float volume_movement_start
,
152 struct synth_bird
*bird_handle
;
153 stb_vorbis
*vorbis_handle
;
156 stb_vorbis_alloc vorbis_alloc
;
158 enum channel_activity
{
159 k_channel_activity_reset
, /* will advance if allocated==1, to wake */
160 k_channel_activity_wake
, /* will advance to either of next two */
161 k_channel_activity_alive
,
162 k_channel_activity_end
,
163 k_channel_activity_error
169 * editable structure, can be modified inside _lock and _unlock
170 * the edit mask tells which to copy into internal _, or to discard
171 * ----------------------------------------------------------------------
173 struct channel_state
{
176 float volume
, /* current volume */
177 volume_target
, /* target volume */
185 v4f spacial_falloff
; /* xyz, range */
191 u32 editble_state_write_mask
;
193 channels
[ AUDIO_CHANNELS
];
195 int debug_ui
, debug_ui_3d
, debug_dsp
;
197 v3f internal_listener_pos
,
198 internal_listener_ears
,
199 internal_listener_velocity
,
201 external_listener_pos
,
202 external_listener_ears
,
203 external_lister_velocity
;
205 float internal_global_volume
,
206 external_global_volume
;
208 static vg_audio
= { .external_global_volume
= 1.0f
};
210 #include "vg/vg_audio_dsp.h"
212 static struct vg_profile
213 _vg_prof_audio_decode
= {.mode
= k_profile_mode_accum
,
214 .name
= "[T2] audio_decode()"},
215 _vg_prof_audio_mix
= {.mode
= k_profile_mode_accum
,
216 .name
= "[T2] audio_mix()"},
217 _vg_prof_dsp
= {.mode
= k_profile_mode_accum
,
218 .name
= "[T2] dsp_process()"},
219 vg_prof_audio_decode
,
224 * These functions are called from the main thread and used to prevent bad
225 * access. TODO: They should be no-ops in release builds.
227 static int audio_lock_checker_load(void)
230 SDL_AtomicLock( &vg_audio
.sl_checker
);
231 value
= vg_audio
.sync_locked
;
232 SDL_AtomicUnlock( &vg_audio
.sl_checker
);
236 static void audio_lock_checker_store( int value
)
238 SDL_AtomicLock( &vg_audio
.sl_checker
);
239 vg_audio
.sync_locked
= value
;
240 SDL_AtomicUnlock( &vg_audio
.sl_checker
);
243 static void audio_require_lock(void)
245 if( audio_lock_checker_load() )
248 vg_error( "Modifying sound effects systems requires locking\n" );
252 static void audio_lock(void)
254 SDL_AtomicLock( &vg_audio
.sl_sync
);
255 audio_lock_checker_store(1);
258 static void audio_unlock(void)
260 audio_lock_checker_store(0);
261 SDL_AtomicUnlock( &vg_audio
.sl_sync
);
264 static void audio_mixer_callback( void *user
, u8
*stream
, int frame_count
);
265 static void vg_audio_init(void)
267 /* TODO: Move here? */
268 vg_console_reg_var( "debug_audio", &vg_audio
.debug_ui
,
269 k_var_dtype_i32
, VG_VAR_CHEAT
);
270 vg_console_reg_var( "debug_dsp", &vg_audio
.debug_dsp
,
271 k_var_dtype_i32
, VG_VAR_CHEAT
);
272 vg_console_reg_var( "volume", &vg_audio
.external_global_volume
,
273 k_var_dtype_f32
, VG_VAR_PERSISTENT
);
275 /* allocate memory */
277 vg_audio
.audio_pool
=
278 vg_create_linear_allocator( vg_mem
.rtmemory
, 1024*1024*32,
282 u32 decode_size
= AUDIO_DECODE_SIZE
* AUDIO_CHANNELS
;
283 vg_audio
.decode_buffer
= vg_linear_alloc( vg_mem
.rtmemory
, decode_size
);
287 SDL_AudioSpec spec_desired
, spec_got
;
288 spec_desired
.callback
= audio_mixer_callback
;
289 spec_desired
.channels
= 2;
290 spec_desired
.format
= AUDIO_F32
;
291 spec_desired
.freq
= 44100;
292 spec_desired
.padding
= 0;
293 spec_desired
.samples
= AUDIO_FRAME_SIZE
;
294 spec_desired
.silence
= 0;
295 spec_desired
.size
= 0;
296 spec_desired
.userdata
= NULL
;
298 vg_audio
.sdl_output_device
=
299 SDL_OpenAudioDevice( NULL
, 0, &spec_desired
, &spec_got
,0 );
301 if( vg_audio
.sdl_output_device
){
302 SDL_PauseAudioDevice( vg_audio
.sdl_output_device
, 0 );
306 "SDL_OpenAudioDevice failed. Your default audio device must support:\n"
307 " Frequency: 44100 hz\n"
308 " Buffer size: 512\n"
310 " Format: s16 or f32\n" );
314 static void vg_audio_free(void)
317 SDL_CloseAudioDevice( vg_audio
.sdl_output_device
);
324 #define AUDIO_EDIT_VOLUME_SLOPE 0x1
325 #define AUDIO_EDIT_VOLUME 0x2
326 #define AUDIO_EDIT_LFO_PERIOD 0x4
327 #define AUDIO_EDIT_LFO_WAVE 0x8
328 #define AUDIO_EDIT_LFO_ATTACHMENT 0x10
329 #define AUDIO_EDIT_SPACIAL 0x20
330 #define AUDIO_EDIT_OWNERSHIP 0x40
331 #define AUDIO_EDIT_SAMPLING_RATE 0x80
333 static void audio_channel_init( audio_channel
*ch
, audio_clip
*clip
,
335 audio_require_lock();
340 ch
->colour
= 0x00333333;
342 if( (ch
->source
->flags
& AUDIO_FLAG_FORMAT
) == k_audio_format_bird
)
343 strcpy( ch
->name
, "[array]" );
344 else if( (ch
->source
->flags
& AUDIO_FLAG_FORMAT
) == k_audio_format_gen
)
345 strcpy( ch
->name
, "[program]" );
347 vg_strncpy( clip
->path
, ch
->name
, 32, k_strncpy_always_add_null
);
351 ch
->editable_state
.relinquished
= 0;
352 ch
->editable_state
.volume
= 1.0f
;
353 ch
->editable_state
.volume_target
= 1.0f
;
354 ch
->editable_state
.pan
= 0.0f
;
355 ch
->editable_state
.pan_target
= 0.0f
;
356 ch
->editable_state
.volume_rate
= 0;
357 ch
->editable_state
.pan_rate
= 0;
358 v4_copy((v4f
){0.0f
,0.0f
,0.0f
,1.0f
},ch
->editable_state
.spacial_falloff
);
359 ch
->editable_state
.lfo
= NULL
;
360 ch
->editable_state
.lfo_amount
= 0.0f
;
361 ch
->editable_state
.sampling_rate
= 1.0f
;
362 ch
->editble_state_write_mask
= 0x00;
365 static void audio_channel_group( audio_channel
*ch
, u16 group
)
367 audio_require_lock();
369 ch
->colour
= (((u32
)group
* 29986577) & 0x00ffffff) | 0xff000000;
372 static void audio_channel_world( audio_channel
*ch
, u8 world_id
)
374 audio_require_lock();
375 ch
->world_id
= world_id
;
378 static audio_channel
*audio_get_first_idle_channel(void)
380 audio_require_lock();
381 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
382 audio_channel
*ch
= &vg_audio
.channels
[i
];
384 if( !ch
->allocated
){
392 static audio_channel
*audio_get_group_idle_channel( u16 group
, u32 max_count
)
394 audio_require_lock();
396 audio_channel
*dest
= NULL
;
398 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
399 audio_channel
*ch
= &vg_audio
.channels
[i
];
402 if( ch
->group
== group
){
412 if( dest
&& (count
< max_count
) ){
419 static audio_channel
*audio_get_group_first_active_channel( u16 group
)
421 audio_require_lock();
422 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
423 audio_channel
*ch
= &vg_audio
.channels
[i
];
424 if( ch
->allocated
&& (ch
->group
== group
) )
430 static int audio_channel_finished( audio_channel
*ch
)
432 audio_require_lock();
433 if( ch
->readable_activity
== k_channel_activity_end
)
439 static audio_channel
*audio_relinquish_channel( audio_channel
*ch
)
441 audio_require_lock();
442 ch
->editable_state
.relinquished
= 1;
443 ch
->editble_state_write_mask
|= AUDIO_EDIT_OWNERSHIP
;
447 static void audio_channel_slope_volume( audio_channel
*ch
, float length
,
450 audio_require_lock();
451 ch
->editable_state
.volume_target
= new_volume
;
452 ch
->editable_state
.volume_rate
= length
* 44100.0f
;
453 ch
->editble_state_write_mask
|= AUDIO_EDIT_VOLUME_SLOPE
;
456 static void audio_channel_set_sampling_rate( audio_channel
*ch
, float rate
)
458 audio_require_lock();
459 ch
->editable_state
.sampling_rate
= rate
;
460 ch
->editble_state_write_mask
|= AUDIO_EDIT_SAMPLING_RATE
;
463 static void audio_channel_edit_volume( audio_channel
*ch
,
464 float new_volume
, int instant
)
466 audio_require_lock();
468 ch
->editable_state
.volume
= new_volume
;
469 ch
->editble_state_write_mask
|= AUDIO_EDIT_VOLUME
;
472 audio_channel_slope_volume( ch
, 0.05f
, new_volume
);
476 static audio_channel
*audio_channel_fadeout( audio_channel
*ch
, float length
)
478 audio_require_lock();
479 audio_channel_slope_volume( ch
, length
, 0.0f
);
480 return audio_relinquish_channel( ch
);
483 static void audio_channel_fadein( audio_channel
*ch
, float length
)
485 audio_require_lock();
486 audio_channel_edit_volume( ch
, 0.0f
, 1 );
487 audio_channel_slope_volume( ch
, length
, 1.0f
);
490 static audio_channel
*audio_channel_crossfade( audio_channel
*ch
,
491 audio_clip
*new_clip
,
492 float length
, u32 flags
)
494 audio_require_lock();
498 ch
= audio_channel_fadeout( ch
, length
);
500 audio_channel
*replacement
= audio_get_first_idle_channel();
503 audio_channel_init( replacement
, new_clip
, flags
);
504 audio_channel_fadein( replacement
, length
);
510 static void audio_channel_sidechain_lfo( audio_channel
*ch
, int lfo_id
,
513 audio_require_lock();
514 ch
->editable_state
.lfo
= &vg_audio
.oscillators
[ lfo_id
];
515 ch
->editable_state
.lfo_amount
= amount
;
516 ch
->editble_state_write_mask
|= AUDIO_EDIT_LFO_ATTACHMENT
;
519 static void audio_channel_set_spacial( audio_channel
*ch
, v3f co
, float range
)
521 audio_require_lock();
522 if( ch
->flags
& AUDIO_FLAG_SPACIAL_3D
){
523 v3_copy( co
, ch
->editable_state
.spacial_falloff
);
526 ch
->editable_state
.spacial_falloff
[3] = 1.0f
;
528 ch
->editable_state
.spacial_falloff
[3] = 1.0f
/range
;
530 ch
->editble_state_write_mask
|= AUDIO_EDIT_SPACIAL
;
533 vg_warn( "Tried to set spacialization paramaters for 2D channel (%s)\n",
538 static int audio_oneshot_3d( audio_clip
*clip
, v3f position
,
539 float range
, float volume
)
541 audio_require_lock();
542 audio_channel
*ch
= audio_get_first_idle_channel();
545 audio_channel_init( ch
, clip
, AUDIO_FLAG_SPACIAL_3D
);
546 audio_channel_set_spacial( ch
, position
, range
);
547 audio_channel_edit_volume( ch
, volume
, 1 );
548 ch
= audio_relinquish_channel( ch
);
556 static int audio_oneshot( audio_clip
*clip
, float volume
, float pan
)
558 audio_require_lock();
559 audio_channel
*ch
= audio_get_first_idle_channel();
562 audio_channel_init( ch
, clip
, 0x00 );
563 audio_channel_edit_volume( ch
, volume
, 1 );
564 ch
= audio_relinquish_channel( ch
);
572 static void audio_set_lfo_wave( int id
, enum lfo_wave_type type
,
575 audio_require_lock();
576 audio_lfo
*lfo
= &vg_audio
.oscillators
[ id
];
577 lfo
->editable_state
.polynomial_coefficient
= coefficient
;
578 lfo
->editable_state
.wave_type
= type
;
580 lfo
->editble_state_write_mask
|= AUDIO_EDIT_LFO_WAVE
;
583 static void audio_set_lfo_frequency( int id
, float freq
)
585 audio_require_lock();
586 audio_lfo
*lfo
= &vg_audio
.oscillators
[ id
];
587 lfo
->editable_state
.period
= 44100.0f
/ freq
;
588 lfo
->editble_state_write_mask
|= AUDIO_EDIT_LFO_PERIOD
;
594 * -----------------------------------------------------------------------------
596 static int audio_channel_load_source( audio_channel
*ch
)
598 u32 format
= ch
->source
->flags
& AUDIO_FLAG_FORMAT
;
600 if( format
== k_audio_format_vorbis
){
601 /* Setup vorbis decoder */
602 u32 index
= ch
- vg_audio
.channels
;
604 u8
*buf
= (u8
*)vg_audio
.decode_buffer
,
605 *loc
= &buf
[AUDIO_DECODE_SIZE
*index
];
607 stb_vorbis_alloc alloc
= {
608 .alloc_buffer
= (char *)loc
,
609 .alloc_buffer_length_in_bytes
= AUDIO_DECODE_SIZE
613 stb_vorbis
*decoder
= stb_vorbis_open_memory(
615 ch
->source
->size
, &err
, &alloc
);
618 vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
619 ch
->source
->path
, err
);
623 ch
->source_length
= stb_vorbis_stream_length_in_samples( decoder
);
624 ch
->vorbis_handle
= decoder
;
627 else if( format
== k_audio_format_bird
){
628 u32 index
= ch
- vg_audio
.channels
;
630 u8
*buf
= (u8
*)vg_audio
.decode_buffer
;
631 struct synth_bird
*loc
= (void *)&buf
[AUDIO_DECODE_SIZE
*index
];
633 memcpy( loc
, ch
->source
->data
, ch
->source
->size
);
634 synth_bird_reset( loc
);
636 ch
->bird_handle
= loc
;
637 ch
->source_length
= synth_bird_get_length_in_samples( loc
);
639 else if( format
== k_audio_format_stereo
){
640 ch
->source_length
= ch
->source
->size
/ 2;
642 else if( format
== k_audio_format_gen
){
643 ch
->source_length
= 0xffffffff;
646 ch
->source_length
= ch
->source
->size
;
652 static void audio_decode_uncompressed_mono( i16
*src
, u32 count
, float *dst
)
654 for( u32 i
=0; i
<count
; i
++ ){
655 dst
[ i
*2 + 0 ] = ((float)src
[i
]) * (1.0f
/32767.0f
);
656 dst
[ i
*2 + 1 ] = ((float)src
[i
]) * (1.0f
/32767.0f
);
661 * adapted from stb_vorbis.h, since the original does not handle mono->stereo
664 stb_vorbis_get_samples_float_interleaved_stereo( stb_vorbis
*f
, float *buffer
,
668 c
= VG_MIN( 1, f
->channels
- 1 );
671 int k
= f
->channel_buffer_end
- f
->channel_buffer_start
;
676 for( int j
=0; j
< k
; ++j
) {
677 *buffer
++ = f
->channel_buffers
[ 0 ][f
->channel_buffer_start
+j
];
678 *buffer
++ = f
->channel_buffers
[ c
][f
->channel_buffer_start
+j
];
682 f
->channel_buffer_start
+= k
;
687 if( !stb_vorbis_get_frame_float( f
, NULL
, NULL
))
695 * ........ more wrecked code sorry!
698 stb_vorbis_get_samples_i16_downmixed( stb_vorbis
*f
, i16
*buffer
, int len
)
701 c
= VG_MIN( 1, f
->channels
- 1 );
704 int k
= f
->channel_buffer_end
- f
->channel_buffer_start
;
709 for( int j
=0; j
< k
; ++j
) {
710 float sl
= f
->channel_buffers
[ 0 ][f
->channel_buffer_start
+j
],
711 sr
= f
->channel_buffers
[ c
][f
->channel_buffer_start
+j
];
713 *buffer
++ = vg_clampf( 0.5f
*(sl
+sr
), -1.0f
, 1.0f
) * 32767.0f
;
714 //*buffer++ = vg_clampf( sr, -1.0f, 1.0f ) * 32767.0f;
718 f
->channel_buffer_start
+= k
;
723 if( !stb_vorbis_get_frame_float( f
, NULL
, NULL
))
730 static inline float audio_lfo_pull_sample( audio_lfo
*lfo
)
734 if( lfo
->time
>= lfo
->_
.period
)
738 t
/= (float)lfo
->_
.period
;
740 if( lfo
->_
.wave_type
== k_lfo_polynomial_bipolar
){
756 return (( 2.0f
* lfo
->sqrt_polynomial_coefficient
* t
) /
757 /* --------------------------------------- */
758 ( 1.0f
+ lfo
->_
.polynomial_coefficient
* t
*t
)
767 static void audio_channel_get_samples( audio_channel
*ch
,
768 u32 count
, float *buf
)
770 vg_profile_begin( &_vg_prof_audio_decode
);
772 u32 remaining
= count
;
775 u32 format
= ch
->source
->flags
& AUDIO_FLAG_FORMAT
;
778 u32 samples_this_run
= VG_MIN(remaining
, ch
->source_length
- ch
->cursor
);
779 remaining
-= samples_this_run
;
781 float *dst
= &buf
[ buffer_pos
* 2 ];
783 if( format
== k_audio_format_stereo
){
784 for( int i
=0;i
<samples_this_run
; i
++ ){
789 else if( format
== k_audio_format_vorbis
){
790 int read_samples
= stb_vorbis_get_samples_float_interleaved_stereo(
795 if( read_samples
!= samples_this_run
){
796 vg_warn( "Invalid samples read (%s)\n", ch
->source
->path
);
798 for( int i
=0; i
<samples_this_run
; i
++ ){
804 else if( format
== k_audio_format_bird
){
805 synth_bird_generate_samples( ch
->bird_handle
, dst
, samples_this_run
);
807 else if( format
== k_audio_format_gen
){
808 void (*fn
)( void *data
, f32
*buf
, u32 count
) = ch
->source
->func
;
809 fn( ch
->source
->data
, dst
, samples_this_run
);
812 i16
*src_buffer
= ch
->source
->data
,
813 *src
= &src_buffer
[ch
->cursor
];
815 audio_decode_uncompressed_mono( src
, samples_this_run
, dst
);
818 ch
->cursor
+= samples_this_run
;
819 buffer_pos
+= samples_this_run
;
821 if( (ch
->flags
& AUDIO_FLAG_LOOP
) && remaining
){
822 if( format
== k_audio_format_vorbis
)
823 stb_vorbis_seek_start( ch
->vorbis_handle
);
824 else if( format
== k_audio_format_bird
)
825 synth_bird_reset( ch
->bird_handle
);
835 buf
[ buffer_pos
*2 + 0 ] = 0.0f
;
836 buf
[ buffer_pos
*2 + 1 ] = 0.0f
;
842 vg_profile_end( &_vg_prof_audio_decode
);
845 static void audio_channel_mix( audio_channel
*ch
, float *buffer
)
847 float framevol_l
= vg_audio
.internal_global_volume
,
848 framevol_r
= vg_audio
.internal_global_volume
;
850 float frame_samplerate
= ch
->_
.sampling_rate
;
852 if( ch
->flags
& AUDIO_FLAG_SPACIAL_3D
){
854 v3_sub( ch
->_
.spacial_falloff
, vg_audio
.internal_listener_pos
, delta
);
856 float dist
= v3_length( delta
),
857 vol
= vg_maxf( 0.0f
, 1.0f
- ch
->_
.spacial_falloff
[3]*dist
);
863 v3_muls( delta
, 1.0f
/dist
, delta
);
864 float pan
= v3_dot( vg_audio
.internal_listener_ears
, delta
);
865 vol
= powf( vol
, 5.0f
);
867 framevol_l
*= (vol
* 0.5f
) * (1.0f
- pan
);
868 framevol_r
*= (vol
* 0.5f
) * (1.0f
+ pan
);
870 if( !(ch
->source
->flags
& AUDIO_FLAG_NO_DOPPLER
) ){
871 const float vs
= 323.0f
;
873 float dv
= v3_dot(delta
,vg_audio
.internal_listener_velocity
);
874 float doppler
= (vs
+dv
)/vs
;
875 doppler
= vg_clampf( doppler
, 0.6f
, 1.4f
);
877 if( fabsf(doppler
-1.0f
) > 0.01f
)
878 frame_samplerate
*= doppler
;
882 if( !vg_validf( framevol_l
) ||
883 !vg_validf( framevol_r
) ||
884 !vg_validf( frame_samplerate
) ){
885 vg_fatal_error( "Invalid sampling conditions.\n"
886 "This crash is to protect your ears.\n"
887 " channel: %p (%s)\n"
890 " listener: %.2f %.2f %.2f [%.2f %.2f %.2f]\n",
891 ch
, ch
->name
, frame_samplerate
,
892 framevol_l
, framevol_r
,
893 vg_audio
.internal_listener_pos
[0],
894 vg_audio
.internal_listener_pos
[1],
895 vg_audio
.internal_listener_pos
[2],
896 vg_audio
.internal_listener_ears
[0],
897 vg_audio
.internal_listener_ears
[1],
898 vg_audio
.internal_listener_ears
[2]
903 u32 buffer_length
= AUDIO_MIX_FRAME_SIZE
;
904 if( frame_samplerate
!= 1.0f
){
905 float l
= ceilf( (float)(AUDIO_MIX_FRAME_SIZE
) * frame_samplerate
);
909 float pcf
[ AUDIO_MIX_FRAME_SIZE
* 2 * 2 ];
911 audio_channel_get_samples( ch
, buffer_length
, pcf
);
913 vg_profile_begin( &_vg_prof_audio_mix
);
915 float volume_movement
= ch
->volume_movement
;
916 float const fvolume_rate
= vg_maxf( 1.0f
, ch
->_
.volume_rate
);
917 const float inv_volume_rate
= 1.0f
/fvolume_rate
;
919 float volume
= ch
->_
.volume
;
920 const float volume_start
= ch
->volume_movement_start
;
921 const float volume_target
= ch
->_
.volume_target
;
923 for( u32 j
=0; j
<AUDIO_MIX_FRAME_SIZE
; j
++ ){
924 volume_movement
+= 1.0f
;
925 float movement_t
= volume_movement
* inv_volume_rate
;
926 movement_t
= vg_minf( movement_t
, 1.0f
);
927 volume
= vg_lerpf( volume_start
, volume_target
, movement_t
);
929 float vol_norm
= volume
* volume
;
932 vol_norm
*= 1.0f
+ audio_lfo_pull_sample(ch
->_
.lfo
) * ch
->_
.lfo_amount
;
934 float vol_l
= vol_norm
* framevol_l
,
935 vol_r
= vol_norm
* framevol_r
,
939 if( frame_samplerate
!= 1.0f
){
940 /* absolutely garbage resampling, but it will do
943 float sample_index
= frame_samplerate
* (float)j
;
944 float t
= vg_fractf( sample_index
);
946 u32 i0
= floorf( sample_index
),
949 sample_l
= pcf
[ i0
*2+0 ]*(1.0f
-t
) + pcf
[ i1
*2+0 ]*t
;
950 sample_r
= pcf
[ i0
*2+1 ]*(1.0f
-t
) + pcf
[ i1
*2+1 ]*t
;
953 sample_l
= pcf
[ j
*2+0 ];
954 sample_r
= pcf
[ j
*2+1 ];
957 buffer
[ j
*2+0 ] += sample_l
* vol_l
;
958 buffer
[ j
*2+1 ] += sample_r
* vol_r
;
961 ch
->volume_movement
+= AUDIO_MIX_FRAME_SIZE
;
962 ch
->volume_movement
= VG_MIN( ch
->volume_movement
, ch
->_
.volume_rate
);
963 ch
->_
.volume
= volume
;
965 vg_profile_end( &_vg_prof_audio_mix
);
968 static void audio_mixer_callback( void *user
, u8
*stream
, int byte_count
)
971 * Copy data and move edit flags to commit flags
972 * ------------------------------------------------------------- */
975 v3_copy( vg_audio
.external_listener_pos
, vg_audio
.internal_listener_pos
);
976 v3_copy( vg_audio
.external_listener_ears
, vg_audio
.internal_listener_ears
);
977 v3_copy( vg_audio
.external_lister_velocity
,
978 vg_audio
.internal_listener_velocity
);
979 vg_audio
.internal_global_volume
= vg_audio
.external_global_volume
;
981 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
982 audio_channel
*ch
= &vg_audio
.channels
[i
];
987 if( ch
->activity
== k_channel_activity_alive
){
988 if( (ch
->cursor
>= ch
->source_length
) &&
989 !(ch
->flags
& AUDIO_FLAG_LOOP
) )
991 ch
->activity
= k_channel_activity_end
;
995 /* process relinquishments */
996 if( (ch
->activity
!= k_channel_activity_reset
) && ch
->_
.relinquished
){
997 if( (ch
->activity
== k_channel_activity_end
)
998 || (ch
->_
.volume
== 0.0f
)
999 || (ch
->activity
== k_channel_activity_error
) )
1001 ch
->_
.relinquished
= 0;
1003 ch
->activity
= k_channel_activity_reset
;
1008 /* process new channels */
1009 if( ch
->activity
== k_channel_activity_reset
){
1010 ch
->_
= ch
->editable_state
;
1012 ch
->source_length
= 0;
1013 ch
->activity
= k_channel_activity_wake
;
1016 if( ch
->editble_state_write_mask
& AUDIO_EDIT_OWNERSHIP
)
1017 ch
->_
.relinquished
= ch
->editable_state
.relinquished
;
1019 ch
->editable_state
.relinquished
= ch
->_
.relinquished
;
1022 if( ch
->editble_state_write_mask
& AUDIO_EDIT_VOLUME
){
1023 ch
->_
.volume
= ch
->editable_state
.volume
;
1024 ch
->_
.volume_target
= ch
->editable_state
.volume
;
1027 ch
->editable_state
.volume
= ch
->_
.volume
;
1031 if( ch
->editble_state_write_mask
& AUDIO_EDIT_VOLUME_SLOPE
){
1032 ch
->volume_movement_start
= ch
->_
.volume
;
1033 ch
->volume_movement
= 0;
1035 ch
->_
.volume_target
= ch
->editable_state
.volume_target
;
1036 ch
->_
.volume_rate
= ch
->editable_state
.volume_rate
;
1039 ch
->editable_state
.volume_target
= ch
->_
.volume_target
;
1040 ch
->editable_state
.volume_rate
= ch
->_
.volume_rate
;
1044 if( ch
->editble_state_write_mask
& AUDIO_EDIT_SAMPLING_RATE
)
1045 ch
->_
.sampling_rate
= ch
->editable_state
.sampling_rate
;
1047 ch
->editable_state
.sampling_rate
= ch
->_
.sampling_rate
;
1050 if( ch
->editble_state_write_mask
& AUDIO_EDIT_LFO_ATTACHMENT
){
1051 ch
->_
.lfo
= ch
->editable_state
.lfo
;
1052 ch
->_
.lfo_amount
= ch
->editable_state
.lfo_amount
;
1055 ch
->editable_state
.lfo
= ch
->_
.lfo
;
1056 ch
->editable_state
.lfo_amount
= ch
->_
.lfo_amount
;
1060 if( ch
->editble_state_write_mask
& AUDIO_EDIT_SPACIAL
)
1061 v4_copy( ch
->editable_state
.spacial_falloff
,ch
->_
.spacial_falloff
);
1063 v4_copy( ch
->_
.spacial_falloff
,ch
->editable_state
.spacial_falloff
);
1066 /* currently readonly, i guess */
1067 ch
->editable_state
.pan_target
= ch
->_
.pan_target
;
1068 ch
->editable_state
.pan
= ch
->_
.pan
;
1069 ch
->editble_state_write_mask
= 0x00;
1072 for( int i
=0; i
<AUDIO_LFOS
; i
++ ){
1073 audio_lfo
*lfo
= &vg_audio
.oscillators
[ i
];
1075 if( lfo
->editble_state_write_mask
& AUDIO_EDIT_LFO_WAVE
){
1076 lfo
->_
.wave_type
= lfo
->editable_state
.wave_type
;
1078 if( lfo
->_
.wave_type
== k_lfo_polynomial_bipolar
){
1079 lfo
->_
.polynomial_coefficient
=
1080 lfo
->editable_state
.polynomial_coefficient
;
1081 lfo
->sqrt_polynomial_coefficient
=
1082 sqrtf(lfo
->_
.polynomial_coefficient
);
1086 if( lfo
->editble_state_write_mask
& AUDIO_EDIT_LFO_PERIOD
){
1087 if( lfo
->_
.period
){
1088 float t
= lfo
->time
;
1089 t
/= (float)lfo
->_
.period
;
1091 lfo
->_
.period
= lfo
->editable_state
.period
;
1092 lfo
->time
= lfo
->_
.period
* t
;
1096 lfo
->_
.period
= lfo
->editable_state
.period
;
1100 lfo
->editble_state_write_mask
= 0x00;
1103 dsp_update_tunings();
1108 * ------------------------------------------------------------- */
1109 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1110 audio_channel
*ch
= &vg_audio
.channels
[i
];
1112 if( ch
->activity
== k_channel_activity_wake
){
1113 if( audio_channel_load_source( ch
) )
1114 ch
->activity
= k_channel_activity_alive
;
1116 ch
->activity
= k_channel_activity_error
;
1122 * -------------------------------------------------------- */
1123 int frame_count
= byte_count
/(2*sizeof(float));
1126 float *pOut32F
= (float *)stream
;
1127 for( int i
=0; i
<frame_count
*2; i
++ )
1130 for( int i
=0; i
<AUDIO_LFOS
; i
++ ){
1131 audio_lfo
*lfo
= &vg_audio
.oscillators
[i
];
1132 lfo
->time_startframe
= lfo
->time
;
1135 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1136 audio_channel
*ch
= &vg_audio
.channels
[i
];
1138 if( ch
->activity
== k_channel_activity_alive
){
1140 ch
->_
.lfo
->time
= ch
->_
.lfo
->time_startframe
;
1142 u32 remaining
= frame_count
,
1146 audio_channel_mix( ch
, pOut32F
+subpos
);
1147 remaining
-= AUDIO_MIX_FRAME_SIZE
;
1148 subpos
+= AUDIO_MIX_FRAME_SIZE
*2;
1153 vg_profile_begin( &_vg_prof_dsp
);
1155 for( int i
=0; i
<frame_count
; i
++ )
1156 vg_dsp_process( pOut32F
+ i
*2, pOut32F
+ i
*2 );
1158 vg_profile_end( &_vg_prof_dsp
);
1162 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1163 audio_channel
*ch
= &vg_audio
.channels
[i
];
1164 ch
->readable_activity
= ch
->activity
;
1167 /* Profiling information
1168 * ----------------------------------------------- */
1169 vg_profile_increment( &_vg_prof_audio_decode
);
1170 vg_profile_increment( &_vg_prof_audio_mix
);
1171 vg_profile_increment( &_vg_prof_dsp
);
1173 vg_prof_audio_mix
= _vg_prof_audio_mix
;
1174 vg_prof_audio_decode
= _vg_prof_audio_decode
;
1175 vg_prof_audio_dsp
= _vg_prof_dsp
;
1177 vg_audio
.samples_last
= frame_count
;
1179 if( vg_audio
.debug_dsp
){
1180 vg_dsp_update_texture();
1186 static void audio_clip_load( audio_clip
*clip
, void *lin_alloc
)
1188 if( lin_alloc
== NULL
)
1189 lin_alloc
= vg_audio
.audio_pool
;
1191 #ifdef VG_AUDIO_FORCE_COMPRESSED
1193 if( (clip
->flags
& AUDIO_FLAG_FORMAT
) != k_audio_format_bird
){
1194 clip
->flags
&= ~AUDIO_FLAG_FORMAT
;
1195 clip
->flags
|= k_audio_format_vorbis
;
1200 /* load in directly */
1201 u32 format
= clip
->flags
& AUDIO_FLAG_FORMAT
;
1203 /* TODO: This contains audio_lock() and unlock, but i don't know why
1204 * can probably remove them. Low priority to check this */
1206 /* TODO: packed files for vorbis etc, should take from data if its not not
1207 * NULL when we get the clip
1210 if( format
== k_audio_format_vorbis
){
1212 vg_fatal_error( "No path specified, embeded vorbis unsupported" );
1216 clip
->data
= vg_file_read( lin_alloc
, clip
->path
, &clip
->size
);
1220 vg_fatal_error( "Audio failed to load" );
1222 float mb
= (float)(clip
->size
) / (1024.0f
*1024.0f
);
1223 vg_info( "Loaded audio clip '%s' (%.1fmb)\n", clip
->path
, mb
);
1225 else if( format
== k_audio_format_stereo
){
1226 vg_fatal_error( "Unsupported format (Stereo uncompressed)" );
1228 else if( format
== k_audio_format_bird
){
1230 vg_fatal_error( "No data, external birdsynth unsupported" );
1233 u32 total_size
= clip
->size
+ sizeof(struct synth_bird
);
1234 total_size
-= sizeof(struct synth_bird_settings
);
1235 total_size
= vg_align8( total_size
);
1237 if( total_size
> AUDIO_DECODE_SIZE
)
1238 vg_fatal_error( "Bird coding too long\n" );
1240 struct synth_bird
*bird
= vg_linear_alloc( lin_alloc
, total_size
);
1241 memcpy( &bird
->settings
, clip
->data
, clip
->size
);
1244 clip
->size
= total_size
;
1246 vg_info( "Loaded bird synthesis pattern (%u bytes)\n", total_size
);
1250 vg_fatal_error( "No path specified, embeded mono unsupported" );
1253 vg_linear_clear( vg_mem
.scratch
);
1256 stb_vorbis_alloc alloc
= {
1257 .alloc_buffer
= vg_linear_alloc( vg_mem
.scratch
, AUDIO_DECODE_SIZE
),
1258 .alloc_buffer_length_in_bytes
= AUDIO_DECODE_SIZE
1261 void *filedata
= vg_file_read( vg_mem
.scratch
, clip
->path
, &fsize
);
1264 stb_vorbis
*decoder
= stb_vorbis_open_memory(
1265 filedata
, fsize
, &err
, &alloc
);
1268 vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
1270 vg_fatal_error( "Vorbis decode error" );
1273 /* only mono is supported in uncompressed */
1274 u32 length_samples
= stb_vorbis_stream_length_in_samples( decoder
),
1275 data_size
= length_samples
* sizeof(i16
);
1278 clip
->data
= vg_linear_alloc( lin_alloc
, vg_align8(data_size
) );
1279 clip
->size
= length_samples
;
1282 int read_samples
= stb_vorbis_get_samples_i16_downmixed(
1283 decoder
, clip
->data
, length_samples
);
1285 if( read_samples
!= length_samples
)
1286 vg_fatal_error( "Decode error" );
1289 float mb
= (float)(data_size
) / (1024.0f
*1024.0f
);
1290 vg_info( "Loaded audio clip '%s' (%.1fmb) %u samples\n", clip
->path
, mb
,
1296 static void audio_clip_loadn( audio_clip
*arr
, int count
, void *lin_alloc
)
1298 for( int i
=0; i
<count
; i
++ )
1299 audio_clip_load( &arr
[i
], lin_alloc
);
1302 static void audio_require_clip_loaded( audio_clip
*clip
)
1304 if( clip
->data
&& clip
->size
)
1308 vg_fatal_error( "Must load audio clip before playing! \n" );
1315 static void audio_debug_ui( m4x4f mtx_pv
)
1317 if( !vg_audio
.debug_ui
)
1322 glBindTexture( GL_TEXTURE_2D
, vg_dsp
.view_texture
);
1323 glTexSubImage2D( GL_TEXTURE_2D
, 0, 0, 0, 256, 256,
1324 GL_RGBA
, GL_UNSIGNED_BYTE
,
1325 vg_dsp
.view_texture_buffer
);
1329 * -----------------------------------------------------------------------
1332 float budget
= ((double)vg_audio
.samples_last
/ 44100.0) * 1000.0;
1333 vg_profile_drawn( (struct vg_profile
*[]){ &vg_prof_audio_decode
,
1335 &vg_prof_audio_dsp
}, 3,
1336 budget
, (ui_rect
){ 4, VG_PROFILE_SAMPLE_COUNT
*2 + 8,
1350 if( vg_audio
.debug_dsp
){
1351 ui_rect view_thing
= { 4, vg
.window_y
-512-4, 512, 512 };
1352 ui_image( view_thing
, vg_dsp
.view_texture
);
1355 ui_rect overlap_buffer
[ AUDIO_CHANNELS
];
1356 u32 overlap_length
= 0;
1358 /* Draw audio stack */
1359 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1360 audio_channel
*ch
= &vg_audio
.channels
[i
];
1363 ui_split( window
, k_ui_axis_h
, 18, 1, row
, window
);
1365 if( !ch
->allocated
){
1366 ui_fill( row
, 0x50333333 );
1370 const char *formats
[] =
1390 const char *activties
[] =
1399 u32 format_index
= (ch
->source
->flags
& AUDIO_FLAG_FORMAT
)>>9;
1401 snprintf( perf
, 127, "%02d[%#04x.%#06x]%c%c%cD %s [%s] %4.2fv'%s'",
1403 ch
->world_id
, ch
->group
,
1404 (ch
->editable_state
.relinquished
)? 'r': '_',
1407 formats
[format_index
],
1408 activties
[ch
->readable_activity
],
1409 ch
->editable_state
.volume
,
1412 ui_fill( row
, 0xa0000000 | ch
->colour
);
1413 ui_text( row
, perf
, 1, k_ui_align_middle_left
, 0 );
1415 if( AUDIO_FLAG_SPACIAL_3D
){
1417 v3_copy( ch
->editable_state
.spacial_falloff
, wpos
);
1420 m4x4_mulv( mtx_pv
, wpos
, wpos
);
1422 if( wpos
[3] > 0.0f
){
1423 v2_muls( wpos
, (1.0f
/wpos
[3]) * 0.5f
, wpos
);
1424 v2_add( wpos
, (v2f
){ 0.5f
, 0.5f
}, wpos
);
1427 wr
[0] = vg_clampf(wpos
[0] * vg
.window_x
, -32000.0f
,32000.0f
);
1428 wr
[1] = vg_clampf((1.0f
-wpos
[1]) * vg
.window_y
,-32000.0f
,32000.0f
);
1432 for( int j
=0; j
<12; j
++ ){
1434 for( int k
=0; k
<overlap_length
; k
++ ){
1435 ui_px
*wk
= overlap_buffer
[k
];
1436 if( ((wr
[0] <= wk
[0]+wk
[2]) && (wr
[0]+wr
[2] >= wk
[0])) &&
1437 ((wr
[1] <= wk
[1]+wk
[3]) && (wr
[1]+wr
[3] >= wk
[1])) )
1450 ui_text( wr
, perf
, 1, k_ui_align_middle_left
, 0 );
1451 rect_copy( wr
, overlap_buffer
[ overlap_length
++ ] );
1459 #endif /* VG_AUDIO_H */