1 /* Copyright (C) 2021-2023 Harry Godden (hgn) - All Rights Reserved */
9 #include "vg/vg_stdint.h"
10 #include "vg/vg_platform.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"
21 #pragma GCC push_options
22 #pragma GCC optimize ("O3")
23 #pragma GCC diagnostic push
24 #pragma GCC diagnostic ignored "-Wdeprecated-declarations"
28 #define STB_VORBIS_MAX_CHANNELS 2
29 #include "submodules/stb/stb_vorbis.c"
36 #pragma GCC pop_options
37 #pragma GCC diagnostic pop
41 #define AUDIO_FRAME_SIZE 512
42 #define AUDIO_MIX_FRAME_SIZE 256
44 #define AUDIO_CHANNELS 32
46 #define AUDIO_FILTERS 16
47 #define AUDIO_FLAG_LOOP 0x1
48 #define AUDIO_FLAG_NO_DOPPLER 0x2
49 #define AUDIO_FLAG_SPACIAL_3D 0x4
50 #define AUDIO_FLAG_AUTO_START 0x8
51 #define AUDIO_FLAG_FORMAT 0x1E00
55 k_audio_format_mono
= 0x000u
,
56 k_audio_format_stereo
= 0x200u
,
57 k_audio_format_vorbis
= 0x400u
,
58 k_audio_format_none0
= 0x600u
,
59 k_audio_format_none1
= 0x800u
,
60 k_audio_format_none2
= 0xA00u
,
61 k_audio_format_none3
= 0xC00u
,
62 k_audio_format_none4
= 0xE00u
,
64 k_audio_format_bird
= 0x1000u
,
65 k_audio_format_none5
= 0x1200u
,
66 k_audio_format_none6
= 0x1400u
,
67 k_audio_format_none7
= 0x1600u
,
68 k_audio_format_none8
= 0x1800u
,
69 k_audio_format_none9
= 0x1A00u
,
70 k_audio_format_none10
= 0x1C00u
,
71 k_audio_format_none11
= 0x1E00u
,
74 #define AUDIO_DECODE_SIZE (1024*256) /* 256 kb decoding buffers */
75 #define AUDIO_MUTE_VOLUME 0.0f
76 #define AUDIO_BASE_VOLUME 1.0f
78 typedef struct audio_clip audio_clip
;
79 typedef struct audio_channel audio_channel
;
80 typedef struct audio_lfo audio_lfo
;
83 union { /* TODO oof.. */
97 static 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 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 VG_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 VG_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 VG_STATIC
void audio_require_lock(void)
245 if( audio_lock_checker_load() )
248 vg_error( "Modifying sound effects systems requires locking\n" );
252 VG_STATIC
void audio_lock(void)
254 SDL_AtomicLock( &vg_audio
.sl_sync
);
255 audio_lock_checker_store(1);
258 VG_STATIC
void audio_unlock(void)
260 audio_lock_checker_store(0);
261 SDL_AtomicUnlock( &vg_audio
.sl_sync
);
264 VG_STATIC
void audio_mixer_callback( void *user
, u8
*stream
, int frame_count
);
265 VG_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 VG_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
, u32 flags
)
339 ch
->colour
= 0x00333333;
341 if( (ch
->source
->flags
& AUDIO_FLAG_FORMAT
) == k_audio_format_bird
)
342 strcpy( ch
->name
, "[array]" );
344 vg_strncpy( clip
->path
, ch
->name
, 32, k_strncpy_always_add_null
);
348 ch
->editable_state
.relinquished
= 0;
349 ch
->editable_state
.volume
= 1.0f
;
350 ch
->editable_state
.volume_target
= 1.0f
;
351 ch
->editable_state
.pan
= 0.0f
;
352 ch
->editable_state
.pan_target
= 0.0f
;
353 ch
->editable_state
.volume_rate
= 0;
354 ch
->editable_state
.pan_rate
= 0;
355 v4_copy((v4f
){0.0f
,0.0f
,0.0f
,1.0f
},ch
->editable_state
.spacial_falloff
);
356 ch
->editable_state
.lfo
= NULL
;
357 ch
->editable_state
.lfo_amount
= 0.0f
;
358 ch
->editable_state
.sampling_rate
= 1.0f
;
359 ch
->editble_state_write_mask
= 0x00;
362 static void audio_channel_group( audio_channel
*ch
, u16 group
)
365 ch
->colour
= (((u32
)group
* 29986577) & 0x00ffffff) | 0xff000000;
368 static void audio_channel_world( audio_channel
*ch
, u8 world_id
)
370 ch
->world_id
= world_id
;
373 static audio_channel
*audio_get_first_idle_channel(void)
375 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
376 audio_channel
*ch
= &vg_audio
.channels
[i
];
378 if( !ch
->allocated
){
386 static audio_channel
*audio_get_group_idle_channel( u16 group
, u32 max_count
)
389 audio_channel
*dest
= NULL
;
391 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
392 audio_channel
*ch
= &vg_audio
.channels
[i
];
395 if( ch
->group
== group
){
405 if( dest
&& (count
< max_count
) ){
412 static audio_channel
*audio_get_group_first_active_channel( u16 group
)
414 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
415 audio_channel
*ch
= &vg_audio
.channels
[i
];
416 if( ch
->allocated
&& (ch
->group
== group
) )
422 static int audio_channel_finished( audio_channel
*ch
)
424 if( ch
->readable_activity
== k_channel_activity_end
)
430 static audio_channel
*audio_relinquish_channel( audio_channel
*ch
)
432 ch
->editable_state
.relinquished
= 1;
433 ch
->editble_state_write_mask
|= AUDIO_EDIT_OWNERSHIP
;
437 static void audio_channel_slope_volume( audio_channel
*ch
, float length
,
440 ch
->editable_state
.volume_target
= new_volume
;
441 ch
->editable_state
.volume_rate
= length
* 44100.0f
;
442 ch
->editble_state_write_mask
|= AUDIO_EDIT_VOLUME_SLOPE
;
445 static void audio_channel_set_sampling_rate( audio_channel
*ch
, float rate
)
447 ch
->editable_state
.sampling_rate
= rate
;
448 ch
->editble_state_write_mask
|= AUDIO_EDIT_SAMPLING_RATE
;
451 static void audio_channel_edit_volume( audio_channel
*ch
,
452 float new_volume
, int instant
)
455 ch
->editable_state
.volume
= new_volume
;
456 ch
->editble_state_write_mask
|= AUDIO_EDIT_VOLUME
;
459 audio_channel_slope_volume( ch
, 0.05f
, new_volume
);
463 static audio_channel
*audio_channel_fadeout( audio_channel
*ch
, float length
)
465 audio_channel_slope_volume( ch
, length
, 0.0f
);
466 return audio_relinquish_channel( ch
);
469 static void audio_channel_fadein( audio_channel
*ch
, float length
)
471 audio_channel_edit_volume( ch
, 0.0f
, 1 );
472 audio_channel_slope_volume( ch
, length
, 1.0f
);
475 static audio_channel
*audio_channel_crossfade( audio_channel
*ch
,
476 audio_clip
*new_clip
,
477 float length
, u32 flags
)
482 ch
= audio_channel_fadeout( ch
, length
);
484 audio_channel
*replacement
= audio_get_first_idle_channel();
487 audio_channel_init( replacement
, new_clip
, flags
);
488 audio_channel_fadein( replacement
, length
);
494 static void audio_channel_sidechain_lfo( audio_channel
*ch
, int lfo_id
,
497 ch
->editable_state
.lfo
= &vg_audio
.oscillators
[ lfo_id
];
498 ch
->editable_state
.lfo_amount
= amount
;
499 ch
->editble_state_write_mask
|= AUDIO_EDIT_LFO_ATTACHMENT
;
502 static void audio_channel_set_spacial( audio_channel
*ch
, v3f co
, float range
)
504 if( ch
->flags
& AUDIO_FLAG_SPACIAL_3D
){
505 v3_copy( co
, ch
->editable_state
.spacial_falloff
);
508 ch
->editable_state
.spacial_falloff
[3] = 1.0f
;
510 ch
->editable_state
.spacial_falloff
[3] = 1.0f
/range
;
512 ch
->editble_state_write_mask
|= AUDIO_EDIT_SPACIAL
;
515 vg_warn( "Tried to set spacialization paramaters for 2D channel (%s)\n",
520 static int audio_oneshot_3d( audio_clip
*clip
, v3f position
,
521 float range
, float volume
)
523 audio_channel
*ch
= audio_get_first_idle_channel();
526 audio_channel_init( ch
, clip
, AUDIO_FLAG_SPACIAL_3D
);
527 audio_channel_set_spacial( ch
, position
, range
);
528 audio_channel_edit_volume( ch
, volume
, 1 );
529 ch
= audio_relinquish_channel( ch
);
537 static int audio_oneshot( audio_clip
*clip
, float volume
, float pan
)
539 audio_channel
*ch
= audio_get_first_idle_channel();
542 audio_channel_init( ch
, clip
, 0x00 );
543 audio_channel_edit_volume( ch
, volume
, 1 );
544 ch
= audio_relinquish_channel( ch
);
552 static void audio_set_lfo_wave( int id
, enum lfo_wave_type type
,
555 audio_lfo
*lfo
= &vg_audio
.oscillators
[ id
];
556 lfo
->editable_state
.polynomial_coefficient
= coefficient
;
557 lfo
->editable_state
.wave_type
= type
;
559 lfo
->editble_state_write_mask
|= AUDIO_EDIT_LFO_WAVE
;
562 static void audio_set_lfo_frequency( int id
, float freq
)
564 audio_lfo
*lfo
= &vg_audio
.oscillators
[ id
];
565 lfo
->editable_state
.period
= 44100.0f
/ freq
;
566 lfo
->editble_state_write_mask
|= AUDIO_EDIT_LFO_PERIOD
;
572 * -----------------------------------------------------------------------------
574 static int audio_channel_load_source( audio_channel
*ch
)
576 u32 format
= ch
->source
->flags
& AUDIO_FLAG_FORMAT
;
578 if( format
== k_audio_format_vorbis
){
579 /* Setup vorbis decoder */
580 u32 index
= ch
- vg_audio
.channels
;
582 u8
*buf
= (u8
*)vg_audio
.decode_buffer
,
583 *loc
= &buf
[AUDIO_DECODE_SIZE
*index
];
585 stb_vorbis_alloc alloc
= {
586 .alloc_buffer
= (char *)loc
,
587 .alloc_buffer_length_in_bytes
= AUDIO_DECODE_SIZE
591 stb_vorbis
*decoder
= stb_vorbis_open_memory(
593 ch
->source
->size
, &err
, &alloc
);
596 vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
597 ch
->source
->path
, err
);
601 ch
->source_length
= stb_vorbis_stream_length_in_samples( decoder
);
602 ch
->vorbis_handle
= decoder
;
605 else if( format
== k_audio_format_bird
){
606 u32 index
= ch
- vg_audio
.channels
;
608 u8
*buf
= (u8
*)vg_audio
.decode_buffer
;
609 struct synth_bird
*loc
= (void *)&buf
[AUDIO_DECODE_SIZE
*index
];
611 memcpy( loc
, ch
->source
->data
, ch
->source
->size
);
612 synth_bird_reset( loc
);
614 ch
->bird_handle
= loc
;
615 ch
->source_length
= synth_bird_get_length_in_samples( loc
);
617 else if( format
== k_audio_format_stereo
){
618 ch
->source_length
= ch
->source
->size
/ 2;
621 ch
->source_length
= ch
->source
->size
;
627 VG_STATIC
void audio_decode_uncompressed_mono( i16
*src
, u32 count
, float *dst
)
629 for( u32 i
=0; i
<count
; i
++ ){
630 dst
[ i
*2 + 0 ] = ((float)src
[i
]) * (1.0f
/32767.0f
);
631 dst
[ i
*2 + 1 ] = ((float)src
[i
]) * (1.0f
/32767.0f
);
636 * adapted from stb_vorbis.h, since the original does not handle mono->stereo
639 stb_vorbis_get_samples_float_interleaved_stereo( stb_vorbis
*f
, float *buffer
,
643 c
= VG_MIN( 1, f
->channels
- 1 );
646 int k
= f
->channel_buffer_end
- f
->channel_buffer_start
;
651 for( int j
=0; j
< k
; ++j
) {
652 *buffer
++ = f
->channel_buffers
[ 0 ][f
->channel_buffer_start
+j
];
653 *buffer
++ = f
->channel_buffers
[ c
][f
->channel_buffer_start
+j
];
657 f
->channel_buffer_start
+= k
;
662 if( !stb_vorbis_get_frame_float( f
, NULL
, NULL
))
670 * ........ more wrecked code sorry!
673 stb_vorbis_get_samples_i16_downmixed( stb_vorbis
*f
, i16
*buffer
, int len
)
676 c
= VG_MIN( 1, f
->channels
- 1 );
679 int k
= f
->channel_buffer_end
- f
->channel_buffer_start
;
684 for( int j
=0; j
< k
; ++j
) {
685 float sl
= f
->channel_buffers
[ 0 ][f
->channel_buffer_start
+j
],
686 sr
= f
->channel_buffers
[ c
][f
->channel_buffer_start
+j
];
688 *buffer
++ = vg_clampf( 0.5f
*(sl
+sr
), -1.0f
, 1.0f
) * 32767.0f
;
689 //*buffer++ = vg_clampf( sr, -1.0f, 1.0f ) * 32767.0f;
693 f
->channel_buffer_start
+= k
;
698 if( !stb_vorbis_get_frame_float( f
, NULL
, NULL
))
705 static inline float audio_lfo_pull_sample( audio_lfo
*lfo
)
709 if( lfo
->time
>= lfo
->_
.period
)
713 t
/= (float)lfo
->_
.period
;
715 if( lfo
->_
.wave_type
== k_lfo_polynomial_bipolar
){
731 return (( 2.0f
* lfo
->sqrt_polynomial_coefficient
* t
) /
732 /* --------------------------------------- */
733 ( 1.0f
+ lfo
->_
.polynomial_coefficient
* t
*t
)
742 static void audio_channel_get_samples( audio_channel
*ch
,
743 u32 count
, float *buf
)
745 vg_profile_begin( &_vg_prof_audio_decode
);
747 u32 remaining
= count
;
750 u32 format
= ch
->source
->flags
& AUDIO_FLAG_FORMAT
;
753 u32 samples_this_run
= VG_MIN(remaining
, ch
->source_length
- ch
->cursor
);
754 remaining
-= samples_this_run
;
756 float *dst
= &buf
[ buffer_pos
* 2 ];
758 if( format
== k_audio_format_stereo
){
759 for( int i
=0;i
<samples_this_run
; i
++ ){
764 else if( format
== k_audio_format_vorbis
){
765 int read_samples
= stb_vorbis_get_samples_float_interleaved_stereo(
770 if( read_samples
!= samples_this_run
){
771 vg_warn( "Invalid samples read (%s)\n", ch
->source
->path
);
773 for( int i
=0; i
<samples_this_run
; i
++ ){
779 else if( format
== k_audio_format_bird
){
780 synth_bird_generate_samples( ch
->bird_handle
, dst
, samples_this_run
);
783 i16
*src_buffer
= ch
->source
->data
,
784 *src
= &src_buffer
[ch
->cursor
];
786 audio_decode_uncompressed_mono( src
, samples_this_run
, dst
);
789 ch
->cursor
+= samples_this_run
;
790 buffer_pos
+= samples_this_run
;
792 if( (ch
->flags
& AUDIO_FLAG_LOOP
) && remaining
){
793 if( format
== k_audio_format_vorbis
)
794 stb_vorbis_seek_start( ch
->vorbis_handle
);
795 else if( format
== k_audio_format_bird
)
796 synth_bird_reset( ch
->bird_handle
);
806 buf
[ buffer_pos
*2 + 0 ] = 0.0f
;
807 buf
[ buffer_pos
*2 + 1 ] = 0.0f
;
813 vg_profile_end( &_vg_prof_audio_decode
);
816 static void audio_channel_mix( audio_channel
*ch
, float *buffer
)
818 float framevol_l
= vg_audio
.internal_global_volume
,
819 framevol_r
= vg_audio
.internal_global_volume
;
821 float frame_samplerate
= ch
->_
.sampling_rate
;
823 if( ch
->flags
& AUDIO_FLAG_SPACIAL_3D
){
825 v3_sub( ch
->_
.spacial_falloff
, vg_audio
.internal_listener_pos
, delta
);
827 float dist
= v3_length( delta
),
828 vol
= vg_maxf( 0.0f
, 1.0f
- ch
->_
.spacial_falloff
[3]*dist
);
834 v3_muls( delta
, 1.0f
/dist
, delta
);
835 float pan
= v3_dot( vg_audio
.internal_listener_ears
, delta
);
836 vol
= powf( vol
, 5.0f
);
838 framevol_l
*= (vol
* 0.5f
) * (1.0f
- pan
);
839 framevol_r
*= (vol
* 0.5f
) * (1.0f
+ pan
);
841 if( !(ch
->source
->flags
& AUDIO_FLAG_NO_DOPPLER
) ){
842 const float vs
= 323.0f
;
844 float dv
= v3_dot(delta
,vg_audio
.internal_listener_velocity
);
845 float doppler
= (vs
+dv
)/vs
;
846 doppler
= vg_clampf( doppler
, 0.6f
, 1.4f
);
848 if( fabsf(doppler
-1.0f
) > 0.01f
)
849 frame_samplerate
*= doppler
;
853 if( !vg_validf( framevol_l
) ) vg_fatal_error( "NaN left channel" );
854 if( !vg_validf( framevol_r
) ) vg_fatal_error( "NaN right channel" );
855 if( !vg_validf( frame_samplerate
) )
856 vg_fatal_error( "NaN sample rate" );
859 u32 buffer_length
= AUDIO_MIX_FRAME_SIZE
;
860 if( frame_samplerate
!= 1.0f
){
861 float l
= ceilf( (float)(AUDIO_MIX_FRAME_SIZE
) * frame_samplerate
);
865 float pcf
[ AUDIO_MIX_FRAME_SIZE
* 2 * 2 ];
867 audio_channel_get_samples( ch
, buffer_length
, pcf
);
869 vg_profile_begin( &_vg_prof_audio_mix
);
871 float volume_movement
= ch
->volume_movement
;
872 float const fvolume_rate
= vg_maxf( 1.0f
, ch
->_
.volume_rate
);
873 const float inv_volume_rate
= 1.0f
/fvolume_rate
;
875 float volume
= ch
->_
.volume
;
876 const float volume_start
= ch
->volume_movement_start
;
877 const float volume_target
= ch
->_
.volume_target
;
879 for( u32 j
=0; j
<AUDIO_MIX_FRAME_SIZE
; j
++ ){
880 volume_movement
+= 1.0f
;
881 float movement_t
= volume_movement
* inv_volume_rate
;
882 movement_t
= vg_minf( movement_t
, 1.0f
);
883 volume
= vg_lerpf( volume_start
, volume_target
, movement_t
);
885 float vol_norm
= volume
* volume
;
888 vol_norm
*= 1.0f
+ audio_lfo_pull_sample(ch
->_
.lfo
) * ch
->_
.lfo_amount
;
890 float vol_l
= vol_norm
* framevol_l
,
891 vol_r
= vol_norm
* framevol_r
,
895 if( frame_samplerate
!= 1.0f
){
896 /* absolutely garbage resampling, but it will do
899 float sample_index
= frame_samplerate
* (float)j
;
900 float t
= vg_fractf( sample_index
);
902 u32 i0
= floorf( sample_index
),
905 sample_l
= pcf
[ i0
*2+0 ]*(1.0f
-t
) + pcf
[ i1
*2+0 ]*t
;
906 sample_r
= pcf
[ i0
*2+1 ]*(1.0f
-t
) + pcf
[ i1
*2+1 ]*t
;
909 sample_l
= pcf
[ j
*2+0 ];
910 sample_r
= pcf
[ j
*2+1 ];
913 buffer
[ j
*2+0 ] += sample_l
* vol_l
;
914 buffer
[ j
*2+1 ] += sample_r
* vol_r
;
917 ch
->volume_movement
+= AUDIO_MIX_FRAME_SIZE
;
918 ch
->volume_movement
= VG_MIN( ch
->volume_movement
, ch
->_
.volume_rate
);
919 ch
->_
.volume
= volume
;
921 vg_profile_end( &_vg_prof_audio_mix
);
924 VG_STATIC
void audio_mixer_callback( void *user
, u8
*stream
, int byte_count
)
927 * Copy data and move edit flags to commit flags
928 * ------------------------------------------------------------- */
931 v3_copy( vg_audio
.external_listener_pos
, vg_audio
.internal_listener_pos
);
932 v3_copy( vg_audio
.external_listener_ears
, vg_audio
.internal_listener_ears
);
933 v3_copy( vg_audio
.external_lister_velocity
,
934 vg_audio
.internal_listener_velocity
);
935 vg_audio
.internal_global_volume
= vg_audio
.external_global_volume
;
937 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
938 audio_channel
*ch
= &vg_audio
.channels
[i
];
943 if( ch
->activity
== k_channel_activity_alive
){
944 if( (ch
->cursor
>= ch
->source_length
) &&
945 !(ch
->flags
& AUDIO_FLAG_LOOP
) )
947 ch
->activity
= k_channel_activity_end
;
951 /* process relinquishments */
952 if( (ch
->activity
!= k_channel_activity_reset
) && ch
->_
.relinquished
){
953 if( (ch
->activity
== k_channel_activity_end
)
954 || (ch
->_
.volume
== 0.0f
)
955 || (ch
->activity
== k_channel_activity_error
) )
957 ch
->_
.relinquished
= 0;
959 ch
->activity
= k_channel_activity_reset
;
964 /* process new channels */
965 if( ch
->activity
== k_channel_activity_reset
){
966 ch
->_
= ch
->editable_state
;
968 ch
->source_length
= 0;
969 ch
->activity
= k_channel_activity_wake
;
972 if( ch
->editble_state_write_mask
& AUDIO_EDIT_OWNERSHIP
)
973 ch
->_
.relinquished
= ch
->editable_state
.relinquished
;
975 ch
->editable_state
.relinquished
= ch
->_
.relinquished
;
978 if( ch
->editble_state_write_mask
& AUDIO_EDIT_VOLUME
){
979 ch
->_
.volume
= ch
->editable_state
.volume
;
980 ch
->_
.volume_target
= ch
->editable_state
.volume
;
983 ch
->editable_state
.volume
= ch
->_
.volume
;
987 if( ch
->editble_state_write_mask
& AUDIO_EDIT_VOLUME_SLOPE
){
988 ch
->volume_movement_start
= ch
->_
.volume
;
989 ch
->volume_movement
= 0;
991 ch
->_
.volume_target
= ch
->editable_state
.volume_target
;
992 ch
->_
.volume_rate
= ch
->editable_state
.volume_rate
;
995 ch
->editable_state
.volume_target
= ch
->_
.volume_target
;
996 ch
->editable_state
.volume_rate
= ch
->_
.volume_rate
;
1000 if( ch
->editble_state_write_mask
& AUDIO_EDIT_SAMPLING_RATE
)
1001 ch
->_
.sampling_rate
= ch
->editable_state
.sampling_rate
;
1003 ch
->editable_state
.sampling_rate
= ch
->_
.sampling_rate
;
1006 if( ch
->editble_state_write_mask
& AUDIO_EDIT_LFO_ATTACHMENT
){
1007 ch
->_
.lfo
= ch
->editable_state
.lfo
;
1008 ch
->_
.lfo_amount
= ch
->editable_state
.lfo_amount
;
1011 ch
->editable_state
.lfo
= ch
->_
.lfo
;
1012 ch
->editable_state
.lfo_amount
= ch
->_
.lfo_amount
;
1016 if( ch
->editble_state_write_mask
& AUDIO_EDIT_SPACIAL
)
1017 v4_copy( ch
->editable_state
.spacial_falloff
,ch
->_
.spacial_falloff
);
1019 v4_copy( ch
->_
.spacial_falloff
,ch
->editable_state
.spacial_falloff
);
1022 /* currently readonly, i guess */
1023 ch
->editable_state
.pan_target
= ch
->_
.pan_target
;
1024 ch
->editable_state
.pan
= ch
->_
.pan
;
1025 ch
->editble_state_write_mask
= 0x00;
1028 for( int i
=0; i
<AUDIO_LFOS
; i
++ ){
1029 audio_lfo
*lfo
= &vg_audio
.oscillators
[ i
];
1031 if( lfo
->editble_state_write_mask
& AUDIO_EDIT_LFO_WAVE
){
1032 lfo
->_
.wave_type
= lfo
->editable_state
.wave_type
;
1034 if( lfo
->_
.wave_type
== k_lfo_polynomial_bipolar
){
1035 lfo
->_
.polynomial_coefficient
=
1036 lfo
->editable_state
.polynomial_coefficient
;
1037 lfo
->sqrt_polynomial_coefficient
=
1038 sqrtf(lfo
->_
.polynomial_coefficient
);
1042 if( lfo
->editble_state_write_mask
& AUDIO_EDIT_LFO_PERIOD
){
1043 if( lfo
->_
.period
){
1044 float t
= lfo
->time
;
1045 t
/= (float)lfo
->_
.period
;
1047 lfo
->_
.period
= lfo
->editable_state
.period
;
1048 lfo
->time
= lfo
->_
.period
* t
;
1052 lfo
->_
.period
= lfo
->editable_state
.period
;
1056 lfo
->editble_state_write_mask
= 0x00;
1059 dsp_update_tunings();
1064 * ------------------------------------------------------------- */
1065 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1066 audio_channel
*ch
= &vg_audio
.channels
[i
];
1068 if( ch
->activity
== k_channel_activity_wake
){
1069 if( audio_channel_load_source( ch
) )
1070 ch
->activity
= k_channel_activity_alive
;
1072 ch
->activity
= k_channel_activity_error
;
1078 * -------------------------------------------------------- */
1079 int frame_count
= byte_count
/(2*sizeof(float));
1082 float *pOut32F
= (float *)stream
;
1083 for( int i
=0; i
<frame_count
*2; i
++ )
1086 for( int i
=0; i
<AUDIO_LFOS
; i
++ ){
1087 audio_lfo
*lfo
= &vg_audio
.oscillators
[i
];
1088 lfo
->time_startframe
= lfo
->time
;
1091 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1092 audio_channel
*ch
= &vg_audio
.channels
[i
];
1094 if( ch
->activity
== k_channel_activity_alive
){
1096 ch
->_
.lfo
->time
= ch
->_
.lfo
->time_startframe
;
1098 u32 remaining
= frame_count
,
1102 audio_channel_mix( ch
, pOut32F
+subpos
);
1103 remaining
-= AUDIO_MIX_FRAME_SIZE
;
1104 subpos
+= AUDIO_MIX_FRAME_SIZE
*2;
1109 vg_profile_begin( &_vg_prof_dsp
);
1111 for( int i
=0; i
<frame_count
; i
++ )
1112 vg_dsp_process( pOut32F
+ i
*2, pOut32F
+ i
*2 );
1114 vg_profile_end( &_vg_prof_dsp
);
1118 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1119 audio_channel
*ch
= &vg_audio
.channels
[i
];
1120 ch
->readable_activity
= ch
->activity
;
1123 /* Profiling information
1124 * ----------------------------------------------- */
1125 vg_profile_increment( &_vg_prof_audio_decode
);
1126 vg_profile_increment( &_vg_prof_audio_mix
);
1127 vg_profile_increment( &_vg_prof_dsp
);
1129 vg_prof_audio_mix
= _vg_prof_audio_mix
;
1130 vg_prof_audio_decode
= _vg_prof_audio_decode
;
1131 vg_prof_audio_dsp
= _vg_prof_dsp
;
1133 vg_audio
.samples_last
= frame_count
;
1135 if( vg_audio
.debug_dsp
){
1136 vg_dsp_update_texture();
1142 VG_STATIC
void audio_clip_load( audio_clip
*clip
, void *lin_alloc
)
1144 if( lin_alloc
== NULL
)
1145 lin_alloc
= vg_audio
.audio_pool
;
1147 /* load in directly */
1148 u32 format
= clip
->flags
& AUDIO_FLAG_FORMAT
;
1150 /* TODO: This contains audio_lock() and unlock, but i don't know why
1151 * can probably remove them. Low priority to check this */
1153 /* TODO: packed files for vorbis etc, should take from data if its not not
1154 * NULL when we get the clip
1157 if( format
== k_audio_format_vorbis
){
1159 vg_fatal_error( "No path specified, embeded vorbis unsupported" );
1163 clip
->data
= vg_file_read( lin_alloc
, clip
->path
, &clip
->size
);
1167 vg_fatal_error( "Audio failed to load" );
1169 float mb
= (float)(clip
->size
) / (1024.0f
*1024.0f
);
1170 vg_info( "Loaded audio clip '%s' (%.1fmb)\n", clip
->path
, mb
);
1172 else if( format
== k_audio_format_stereo
){
1173 vg_fatal_error( "Unsupported format (Stereo uncompressed)" );
1175 else if( format
== k_audio_format_bird
){
1177 vg_fatal_error( "No data, external birdsynth unsupported" );
1180 u32 total_size
= clip
->size
+ sizeof(struct synth_bird
);
1181 total_size
-= sizeof(struct synth_bird_settings
);
1182 total_size
= vg_align8( total_size
);
1184 if( total_size
> AUDIO_DECODE_SIZE
)
1185 vg_fatal_error( "Bird coding too long\n" );
1187 struct synth_bird
*bird
= vg_linear_alloc( lin_alloc
, total_size
);
1188 memcpy( &bird
->settings
, clip
->data
, clip
->size
);
1191 clip
->size
= total_size
;
1193 vg_info( "Loaded bird synthesis pattern (%u bytes)\n", total_size
);
1197 vg_fatal_error( "No path specified, embeded mono unsupported" );
1200 vg_linear_clear( vg_mem
.scratch
);
1203 stb_vorbis_alloc alloc
= {
1204 .alloc_buffer
= vg_linear_alloc( vg_mem
.scratch
, AUDIO_DECODE_SIZE
),
1205 .alloc_buffer_length_in_bytes
= AUDIO_DECODE_SIZE
1208 void *filedata
= vg_file_read( vg_mem
.scratch
, clip
->path
, &fsize
);
1211 stb_vorbis
*decoder
= stb_vorbis_open_memory(
1212 filedata
, fsize
, &err
, &alloc
);
1215 vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
1217 vg_fatal_error( "Vorbis decode error" );
1220 /* only mono is supported in uncompressed */
1221 u32 length_samples
= stb_vorbis_stream_length_in_samples( decoder
),
1222 data_size
= length_samples
* sizeof(i16
);
1225 clip
->data
= vg_linear_alloc( lin_alloc
, vg_align8(data_size
) );
1226 clip
->size
= length_samples
;
1229 int read_samples
= stb_vorbis_get_samples_i16_downmixed(
1230 decoder
, clip
->data
, length_samples
);
1232 if( read_samples
!= length_samples
)
1233 vg_fatal_error( "Decode error" );
1235 float mb
= (float)(data_size
) / (1024.0f
*1024.0f
);
1236 vg_info( "Loaded audio clip '%s' (%.1fmb) %u samples\n", clip
->path
, mb
,
1241 VG_STATIC
void audio_clip_loadn( audio_clip
*arr
, int count
, void *lin_alloc
)
1243 for( int i
=0; i
<count
; i
++ )
1244 audio_clip_load( &arr
[i
], lin_alloc
);
1247 VG_STATIC
void audio_require_clip_loaded( audio_clip
*clip
)
1249 if( clip
->data
&& clip
->size
)
1253 vg_fatal_error( "Must load audio clip before playing! \n" );
1260 VG_STATIC
void audio_debug_ui( m4x4f mtx_pv
)
1262 if( !vg_audio
.debug_ui
)
1267 glBindTexture( GL_TEXTURE_2D
, vg_dsp
.view_texture
);
1268 glTexSubImage2D( GL_TEXTURE_2D
, 0, 0, 0, 256, 256,
1269 GL_RGBA
, GL_UNSIGNED_BYTE
,
1270 vg_dsp
.view_texture_buffer
);
1274 * -----------------------------------------------------------------------
1277 float budget
= ((double)vg_audio
.samples_last
/ 44100.0) * 1000.0;
1279 vg_profile_drawn( (struct vg_profile
*[]){ &vg_prof_audio_decode
,
1281 &vg_prof_audio_dsp
}, 3,
1282 budget
, (ui_rect
){ 4, VG_PROFILE_SAMPLE_COUNT
*2 + 8,
1297 if( vg_audio
.debug_dsp
){
1298 ui_rect view_thing
= { 4, vg
.window_y
-512-4, 512, 512 };
1299 ui_image( view_thing
, vg_dsp
.view_texture
);
1302 ui_rect overlap_buffer
[ AUDIO_CHANNELS
];
1303 u32 overlap_length
= 0;
1305 /* Draw audio stack */
1306 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1307 audio_channel
*ch
= &vg_audio
.channels
[i
];
1310 ui_split( window
, k_ui_axis_h
, 18, 1, row
, window
);
1312 if( !ch
->allocated
){
1313 ui_fill( row
, 0x50333333 );
1317 const char *formats
[] =
1337 const char *activties
[] =
1346 u32 format_index
= (ch
->source
->flags
& AUDIO_FLAG_FORMAT
)>>9;
1348 snprintf( perf
, 127, "%02d[%#04x.%#06x]%c%c%cD %s [%s] %4.2fv'%s'",
1350 ch
->world_id
, ch
->group
,
1351 (ch
->editable_state
.relinquished
)? 'r': '_',
1354 formats
[format_index
],
1355 activties
[ch
->readable_activity
],
1356 ch
->editable_state
.volume
,
1359 ui_fill( row
, 0xa0000000 | ch
->colour
);
1360 ui_text( row
, perf
, 1, k_ui_align_middle_left
, 0 );
1362 if( AUDIO_FLAG_SPACIAL_3D
){
1364 v3_copy( ch
->editable_state
.spacial_falloff
, wpos
);
1367 m4x4_mulv( mtx_pv
, wpos
, wpos
);
1369 if( wpos
[3] > 0.0f
){
1370 v2_muls( wpos
, (1.0f
/wpos
[3]) * 0.5f
, wpos
);
1371 v2_add( wpos
, (v2f
){ 0.5f
, 0.5f
}, wpos
);
1374 wr
[0] = vg_clampf(wpos
[0] * vg
.window_x
, -32000.0f
,32000.0f
);
1375 wr
[1] = vg_clampf((1.0f
-wpos
[1]) * vg
.window_y
,-32000.0f
,32000.0f
);
1379 for( int j
=0; j
<12; j
++ ){
1381 for( int k
=0; k
<overlap_length
; k
++ ){
1382 ui_px
*wk
= overlap_buffer
[k
];
1383 if( ((wr
[0] <= wk
[0]+wk
[2]) && (wr
[0]+wr
[2] >= wk
[0])) &&
1384 ((wr
[1] <= wk
[1]+wk
[3]) && (wr
[1]+wr
[3] >= wk
[1])) )
1397 ui_text( wr
, perf
, 1, k_ui_align_middle_left
, 0 );
1398 rect_copy( wr
, overlap_buffer
[ overlap_length
++ ] );
1406 #endif /* VG_AUDIO_H */