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"
24 #pragma GCC push_options
25 #pragma GCC optimize ("O3")
26 #pragma GCC diagnostic push
27 #pragma GCC diagnostic ignored "-Wdeprecated-declarations"
31 #define STB_VORBIS_MAX_CHANNELS 2
32 #include "submodules/stb/stb_vorbis.c"
39 #pragma GCC pop_options
40 #pragma GCC diagnostic pop
44 #define AUDIO_FRAME_SIZE 512
45 #define AUDIO_MIX_FRAME_SIZE 256
47 #define AUDIO_CHANNELS 32
49 #define AUDIO_FILTERS 16
50 #define AUDIO_FLAG_LOOP 0x1
51 #define AUDIO_FLAG_NO_DOPPLER 0x2
52 #define AUDIO_FLAG_SPACIAL_3D 0x4
53 #define AUDIO_FLAG_AUTO_START 0x8
54 #define AUDIO_FLAG_FORMAT 0x1E00
58 k_audio_format_mono
= 0x000u
,
59 k_audio_format_stereo
= 0x200u
,
60 k_audio_format_vorbis
= 0x400u
,
61 k_audio_format_none0
= 0x600u
,
62 k_audio_format_none1
= 0x800u
,
63 k_audio_format_none2
= 0xA00u
,
64 k_audio_format_none3
= 0xC00u
,
65 k_audio_format_none4
= 0xE00u
,
67 k_audio_format_bird
= 0x1000u
,
68 k_audio_format_none5
= 0x1200u
,
69 k_audio_format_none6
= 0x1400u
,
70 k_audio_format_none7
= 0x1600u
,
71 k_audio_format_none8
= 0x1800u
,
72 k_audio_format_none9
= 0x1A00u
,
73 k_audio_format_none10
= 0x1C00u
,
74 k_audio_format_none11
= 0x1E00u
,
77 #define AUDIO_DECODE_SIZE (1024*256) /* 256 kb decoding buffers */
78 #define AUDIO_MUTE_VOLUME 0.0f
79 #define AUDIO_BASE_VOLUME 1.0f
81 typedef struct audio_clip audio_clip
;
82 typedef struct audio_channel audio_channel
;
83 typedef struct audio_lfo audio_lfo
;
92 static struct vg_audio_system
{
93 SDL_AudioDeviceID sdl_output_device
;
102 SDL_SpinLock sl_checker
,
106 u32 time
, time_startframe
;
107 float sqrt_polynomial_coefficient
;
114 k_lfo_polynomial_bipolar
119 float polynomial_coefficient
;
122 u32 editble_state_write_mask
;
124 oscillators
[ AUDIO_LFOS
];
126 struct audio_channel
{
130 char name
[32]; /* only editable while allocated == 0 */
131 audio_clip
*source
; /* ... */
133 u32 colour
; /* ... */
135 /* internal non-readable state
136 * -----------------------------*/
137 u32 cursor
, source_length
;
139 float volume_movement_start
,
146 struct synth_bird
*bird_handle
;
147 stb_vorbis
*vorbis_handle
;
150 stb_vorbis_alloc vorbis_alloc
;
152 enum channel_activity
{
153 k_channel_activity_reset
, /* will advance if allocated==1, to wake */
154 k_channel_activity_wake
, /* will advance to either of next two */
155 k_channel_activity_alive
,
156 k_channel_activity_end
,
157 k_channel_activity_error
163 * editable structure, can be modified inside _lock and _unlock
164 * the edit mask tells which to copy into internal _, or to discard
165 * ----------------------------------------------------------------------
167 struct channel_state
{
170 float volume
, /* current volume */
171 volume_target
, /* target volume */
179 v4f spacial_falloff
; /* xyz, range */
185 u32 editble_state_write_mask
;
187 channels
[ AUDIO_CHANNELS
];
189 int debug_ui
, debug_ui_3d
, debug_dsp
;
191 v3f internal_listener_pos
,
192 internal_listener_ears
,
193 internal_listener_velocity
,
195 external_listener_pos
,
196 external_listener_ears
,
197 external_lister_velocity
;
199 float internal_global_volume
,
200 external_global_volume
;
202 vg_audio
= { .external_global_volume
= 1.0f
};
204 #include "vg/vg_audio_dsp.h"
206 static struct vg_profile
207 _vg_prof_audio_decode
= {.mode
= k_profile_mode_accum
,
208 .name
= "[T2] audio_decode()"},
209 _vg_prof_audio_mix
= {.mode
= k_profile_mode_accum
,
210 .name
= "[T2] audio_mix()"},
211 _vg_prof_dsp
= {.mode
= k_profile_mode_accum
,
212 .name
= "[T2] dsp_process()"},
213 vg_prof_audio_decode
,
218 * These functions are called from the main thread and used to prevent bad
219 * access. TODO: They should be no-ops in release builds.
221 VG_STATIC
int audio_lock_checker_load(void)
224 SDL_AtomicLock( &vg_audio
.sl_checker
);
225 value
= vg_audio
.sync_locked
;
226 SDL_AtomicUnlock( &vg_audio
.sl_checker
);
230 VG_STATIC
void audio_lock_checker_store( int value
)
232 SDL_AtomicLock( &vg_audio
.sl_checker
);
233 vg_audio
.sync_locked
= value
;
234 SDL_AtomicUnlock( &vg_audio
.sl_checker
);
237 VG_STATIC
void audio_require_lock(void)
239 if( audio_lock_checker_load() )
242 vg_error( "Modifying sound effects systems requires locking\n" );
246 VG_STATIC
void audio_lock(void)
248 SDL_AtomicLock( &vg_audio
.sl_sync
);
249 audio_lock_checker_store(1);
252 VG_STATIC
void audio_unlock(void)
254 audio_lock_checker_store(0);
255 SDL_AtomicUnlock( &vg_audio
.sl_sync
);
258 VG_STATIC
void audio_mixer_callback( void *user
, u8
*stream
, int frame_count
);
259 VG_STATIC
void vg_audio_init(void)
261 /* TODO: Move here? */
262 vg_console_reg_var( "debug_audio", &vg_audio
.debug_ui
,
263 k_var_dtype_i32
, VG_VAR_CHEAT
);
264 vg_console_reg_var( "debug_dsp", &vg_audio
.debug_dsp
,
265 k_var_dtype_i32
, VG_VAR_CHEAT
);
266 vg_console_reg_var( "volume", &vg_audio
.external_global_volume
,
267 k_var_dtype_f32
, VG_VAR_PERSISTENT
);
269 /* allocate memory */
271 vg_audio
.audio_pool
=
272 vg_create_linear_allocator( vg_mem
.rtmemory
, 1024*1024*32,
276 u32 decode_size
= AUDIO_DECODE_SIZE
* AUDIO_CHANNELS
;
277 vg_audio
.decode_buffer
= vg_linear_alloc( vg_mem
.rtmemory
, decode_size
);
281 SDL_AudioSpec spec_desired
, spec_got
;
282 spec_desired
.callback
= audio_mixer_callback
;
283 spec_desired
.channels
= 2;
284 spec_desired
.format
= AUDIO_F32
;
285 spec_desired
.freq
= 44100;
286 spec_desired
.padding
= 0;
287 spec_desired
.samples
= AUDIO_FRAME_SIZE
;
288 spec_desired
.silence
= 0;
289 spec_desired
.size
= 0;
290 spec_desired
.userdata
= NULL
;
292 vg_audio
.sdl_output_device
=
293 SDL_OpenAudioDevice( NULL
, 0, &spec_desired
, &spec_got
,0 );
295 if( vg_audio
.sdl_output_device
){
296 SDL_PauseAudioDevice( vg_audio
.sdl_output_device
, 0 );
300 "SDL_OpenAudioDevice failed. Your default audio device must support:\n"
301 " Frequency: 44100 hz\n"
302 " Buffer size: 512\n"
304 " Format: s16 or f32\n" );
307 vg_success( "Ready\n" );
310 VG_STATIC
void vg_audio_free(void)
313 SDL_CloseAudioDevice( vg_audio
.sdl_output_device
);
320 #define AUDIO_EDIT_VOLUME_SLOPE 0x1
321 #define AUDIO_EDIT_VOLUME 0x2
322 #define AUDIO_EDIT_LFO_PERIOD 0x4
323 #define AUDIO_EDIT_LFO_WAVE 0x8
324 #define AUDIO_EDIT_LFO_ATTACHMENT 0x10
325 #define AUDIO_EDIT_SPACIAL 0x20
326 #define AUDIO_EDIT_OWNERSHIP 0x40
327 #define AUDIO_EDIT_SAMPLING_RATE 0x80
329 static void audio_channel_init( audio_channel
*ch
, audio_clip
*clip
, u32 flags
)
334 ch
->colour
= 0x00333333;
336 if( (ch
->source
->flags
& AUDIO_FLAG_FORMAT
) == k_audio_format_bird
)
337 strcpy( ch
->name
, "[array]" );
339 strncpy( ch
->name
, clip
->path
, 31 );
343 ch
->editable_state
.relinquished
= 0;
344 ch
->editable_state
.volume
= 1.0f
;
345 ch
->editable_state
.volume_target
= 1.0f
;
346 ch
->editable_state
.pan
= 0.0f
;
347 ch
->editable_state
.pan_target
= 0.0f
;
348 ch
->editable_state
.volume_rate
= 0;
349 ch
->editable_state
.pan_rate
= 0;
350 v4_copy((v4f
){0.0f
,0.0f
,0.0f
,1.0f
},ch
->editable_state
.spacial_falloff
);
351 ch
->editable_state
.lfo
= NULL
;
352 ch
->editable_state
.lfo_amount
= 0.0f
;
353 ch
->editable_state
.sampling_rate
= 1.0f
;
354 ch
->editble_state_write_mask
= 0x00;
357 static void audio_channel_group( audio_channel
*ch
, u32 group
)
360 ch
->colour
= ((group
* 29986577) & 0x00ffffff) | 0xff000000;
363 static audio_channel
*audio_get_first_idle_channel(void)
365 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
366 audio_channel
*ch
= &vg_audio
.channels
[i
];
368 if( !ch
->allocated
){
376 static audio_channel
*audio_get_group_idle_channel( u32 group
, u32 max_count
)
379 audio_channel
*dest
= NULL
;
381 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
382 audio_channel
*ch
= &vg_audio
.channels
[i
];
385 if( ch
->group
== group
){
395 if( dest
&& (count
< max_count
) ){
402 static audio_channel
*audio_get_group_first_active_channel( u32 group
)
404 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
405 audio_channel
*ch
= &vg_audio
.channels
[i
];
406 if( ch
->allocated
&& (ch
->group
== group
) )
412 static int audio_channel_finished( audio_channel
*ch
)
414 if( ch
->readable_activity
== k_channel_activity_end
)
420 static audio_channel
*audio_relinquish_channel( audio_channel
*ch
)
422 ch
->editable_state
.relinquished
= 1;
423 ch
->editble_state_write_mask
|= AUDIO_EDIT_OWNERSHIP
;
427 static void audio_channel_slope_volume( audio_channel
*ch
, float length
,
430 ch
->editable_state
.volume_target
= new_volume
;
431 ch
->editable_state
.volume_rate
= length
* 44100.0f
;
432 ch
->editble_state_write_mask
|= AUDIO_EDIT_VOLUME_SLOPE
;
435 static void audio_channel_set_sampling_rate( audio_channel
*ch
, float rate
)
437 ch
->editable_state
.sampling_rate
= rate
;
438 ch
->editble_state_write_mask
|= AUDIO_EDIT_SAMPLING_RATE
;
441 static void audio_channel_edit_volume( audio_channel
*ch
,
442 float new_volume
, int instant
)
445 ch
->editable_state
.volume
= new_volume
;
446 ch
->editble_state_write_mask
|= AUDIO_EDIT_VOLUME
;
449 audio_channel_slope_volume( ch
, 0.05f
, new_volume
);
453 static audio_channel
*audio_channel_fadeout( audio_channel
*ch
, float length
)
455 audio_channel_slope_volume( ch
, length
, 0.0f
);
456 return audio_relinquish_channel( ch
);
459 static void audio_channel_fadein( audio_channel
*ch
, float length
)
461 audio_channel_edit_volume( ch
, 0.0f
, 1 );
462 audio_channel_slope_volume( ch
, length
, 1.0f
);
465 static audio_channel
*audio_channel_crossfade( audio_channel
*ch
,
466 audio_clip
*new_clip
,
467 float length
, u32 flags
)
472 ch
= audio_channel_fadeout( ch
, length
);
474 audio_channel
*replacement
= audio_get_first_idle_channel();
477 audio_channel_init( replacement
, new_clip
, flags
);
478 audio_channel_fadein( replacement
, length
);
484 static void audio_channel_sidechain_lfo( audio_channel
*ch
, int lfo_id
,
487 ch
->editable_state
.lfo
= &vg_audio
.oscillators
[ lfo_id
];
488 ch
->editable_state
.lfo_amount
= amount
;
489 ch
->editble_state_write_mask
|= AUDIO_EDIT_LFO_ATTACHMENT
;
492 static void audio_channel_set_spacial( audio_channel
*ch
, v3f co
, float range
)
494 if( ch
->flags
& AUDIO_FLAG_SPACIAL_3D
){
495 v3_copy( co
, ch
->editable_state
.spacial_falloff
);
498 ch
->editable_state
.spacial_falloff
[3] = 1.0f
;
500 ch
->editable_state
.spacial_falloff
[3] = 1.0f
/range
;
502 ch
->editble_state_write_mask
|= AUDIO_EDIT_SPACIAL
;
505 vg_warn( "Tried to set spacialization paramaters for 2D channel (%s)\n",
510 static int audio_oneshot_3d( audio_clip
*clip
, v3f position
,
511 float range
, float volume
)
513 audio_channel
*ch
= audio_get_first_idle_channel();
516 audio_channel_init( ch
, clip
, AUDIO_FLAG_SPACIAL_3D
);
517 audio_channel_set_spacial( ch
, position
, range
);
518 audio_channel_edit_volume( ch
, volume
, 1 );
519 ch
= audio_relinquish_channel( ch
);
527 static int audio_oneshot( audio_clip
*clip
, float volume
, float pan
)
529 audio_channel
*ch
= audio_get_first_idle_channel();
532 audio_channel_init( ch
, clip
, 0x00 );
533 audio_channel_edit_volume( ch
, volume
, 1 );
534 ch
= audio_relinquish_channel( ch
);
542 static void audio_set_lfo_wave( int id
, enum lfo_wave_type type
,
545 audio_lfo
*lfo
= &vg_audio
.oscillators
[ id
];
546 lfo
->editable_state
.polynomial_coefficient
= coefficient
;
547 lfo
->editable_state
.wave_type
= type
;
549 lfo
->editble_state_write_mask
|= AUDIO_EDIT_LFO_WAVE
;
552 static void audio_set_lfo_frequency( int id
, float freq
)
554 audio_lfo
*lfo
= &vg_audio
.oscillators
[ id
];
555 lfo
->editable_state
.period
= 44100.0f
/ freq
;
556 lfo
->editble_state_write_mask
|= AUDIO_EDIT_LFO_PERIOD
;
562 * -----------------------------------------------------------------------------
564 static int audio_channel_load_source( audio_channel
*ch
)
566 u32 format
= ch
->source
->flags
& AUDIO_FLAG_FORMAT
;
568 if( format
== k_audio_format_vorbis
){
569 /* Setup vorbis decoder */
570 u32 index
= ch
- vg_audio
.channels
;
572 u8
*buf
= (u8
*)vg_audio
.decode_buffer
,
573 *loc
= &buf
[AUDIO_DECODE_SIZE
*index
];
575 stb_vorbis_alloc alloc
= {
576 .alloc_buffer
= (char *)loc
,
577 .alloc_buffer_length_in_bytes
= AUDIO_DECODE_SIZE
581 stb_vorbis
*decoder
= stb_vorbis_open_memory(
583 ch
->source
->size
, &err
, &alloc
);
586 vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
587 ch
->source
->path
, err
);
591 ch
->source_length
= stb_vorbis_stream_length_in_samples( decoder
);
592 ch
->vorbis_handle
= decoder
;
595 else if( format
== k_audio_format_bird
){
596 u32 index
= ch
- vg_audio
.channels
;
598 u8
*buf
= (u8
*)vg_audio
.decode_buffer
;
599 struct synth_bird
*loc
= (void *)&buf
[AUDIO_DECODE_SIZE
*index
];
601 memcpy( loc
, ch
->source
->data
, ch
->source
->size
);
602 synth_bird_reset( loc
);
604 ch
->bird_handle
= loc
;
605 ch
->source_length
= synth_bird_get_length_in_samples( loc
);
607 else if( format
== k_audio_format_stereo
){
608 ch
->source_length
= ch
->source
->size
/ 2;
611 ch
->source_length
= ch
->source
->size
;
617 VG_STATIC
void audio_decode_uncompressed_mono( i16
*src
, u32 count
, float *dst
)
619 for( u32 i
=0; i
<count
; i
++ ){
620 dst
[ i
*2 + 0 ] = ((float)src
[i
]) * (1.0f
/32767.0f
);
621 dst
[ i
*2 + 1 ] = ((float)src
[i
]) * (1.0f
/32767.0f
);
626 * adapted from stb_vorbis.h, since the original does not handle mono->stereo
629 stb_vorbis_get_samples_float_interleaved_stereo( stb_vorbis
*f
, float *buffer
,
633 c
= VG_MIN( 1, f
->channels
- 1 );
636 int k
= f
->channel_buffer_end
- f
->channel_buffer_start
;
641 for( int j
=0; j
< k
; ++j
) {
642 *buffer
++ = f
->channel_buffers
[ 0 ][f
->channel_buffer_start
+j
];
643 *buffer
++ = f
->channel_buffers
[ c
][f
->channel_buffer_start
+j
];
647 f
->channel_buffer_start
+= k
;
652 if( !stb_vorbis_get_frame_float( f
, NULL
, NULL
))
660 * ........ more wrecked code sorry!
663 stb_vorbis_get_samples_i16_downmixed( stb_vorbis
*f
, i16
*buffer
, int len
)
666 c
= VG_MIN( 1, f
->channels
- 1 );
669 int k
= f
->channel_buffer_end
- f
->channel_buffer_start
;
674 for( int j
=0; j
< k
; ++j
) {
675 float sl
= f
->channel_buffers
[ 0 ][f
->channel_buffer_start
+j
],
676 sr
= f
->channel_buffers
[ c
][f
->channel_buffer_start
+j
];
678 *buffer
++ = vg_clampf( 0.5f
*(sl
+sr
), -1.0f
, 1.0f
) * 32767.0f
;
679 //*buffer++ = vg_clampf( sr, -1.0f, 1.0f ) * 32767.0f;
683 f
->channel_buffer_start
+= k
;
688 if( !stb_vorbis_get_frame_float( f
, NULL
, NULL
))
695 static inline float audio_lfo_pull_sample( audio_lfo
*lfo
)
699 if( lfo
->time
>= lfo
->_
.period
)
703 t
/= (float)lfo
->_
.period
;
705 if( lfo
->_
.wave_type
== k_lfo_polynomial_bipolar
){
721 return (( 2.0f
* lfo
->sqrt_polynomial_coefficient
* t
) /
722 /* --------------------------------------- */
723 ( 1.0f
+ lfo
->_
.polynomial_coefficient
* t
*t
)
732 static void audio_channel_get_samples( audio_channel
*ch
,
733 u32 count
, float *buf
)
735 vg_profile_begin( &_vg_prof_audio_decode
);
737 u32 remaining
= count
;
740 u32 format
= ch
->source
->flags
& AUDIO_FLAG_FORMAT
;
743 u32 samples_this_run
= VG_MIN(remaining
, ch
->source_length
- ch
->cursor
);
744 remaining
-= samples_this_run
;
746 float *dst
= &buf
[ buffer_pos
* 2 ];
748 if( format
== k_audio_format_stereo
){
749 for( int i
=0;i
<samples_this_run
; i
++ ){
754 else if( format
== k_audio_format_vorbis
){
755 int read_samples
= stb_vorbis_get_samples_float_interleaved_stereo(
760 if( read_samples
!= samples_this_run
){
761 vg_warn( "Invalid samples read (%s)\n", ch
->source
->path
);
763 for( int i
=0; i
<samples_this_run
; i
++ ){
769 else if( format
== k_audio_format_bird
){
770 synth_bird_generate_samples( ch
->bird_handle
, dst
, samples_this_run
);
773 i16
*src_buffer
= ch
->source
->data
,
774 *src
= &src_buffer
[ch
->cursor
];
776 audio_decode_uncompressed_mono( src
, samples_this_run
, dst
);
779 ch
->cursor
+= samples_this_run
;
780 buffer_pos
+= samples_this_run
;
782 if( (ch
->flags
& AUDIO_FLAG_LOOP
) && remaining
){
783 if( format
== k_audio_format_vorbis
)
784 stb_vorbis_seek_start( ch
->vorbis_handle
);
785 else if( format
== k_audio_format_bird
)
786 synth_bird_reset( ch
->bird_handle
);
796 buf
[ buffer_pos
*2 + 0 ] = 0.0f
;
797 buf
[ buffer_pos
*2 + 1 ] = 0.0f
;
803 vg_profile_end( &_vg_prof_audio_decode
);
806 static void audio_channel_mix( audio_channel
*ch
, float *buffer
)
808 float framevol_l
= vg_audio
.internal_global_volume
,
809 framevol_r
= vg_audio
.internal_global_volume
;
811 float frame_samplerate
= ch
->_
.sampling_rate
;
813 if( ch
->flags
& AUDIO_FLAG_SPACIAL_3D
){
815 v3_sub( ch
->_
.spacial_falloff
, vg_audio
.internal_listener_pos
, delta
);
817 float dist
= v3_length( delta
),
818 vol
= vg_maxf( 0.0f
, 1.0f
- ch
->_
.spacial_falloff
[3]*dist
);
824 v3_muls( delta
, 1.0f
/dist
, delta
);
825 float pan
= v3_dot( vg_audio
.internal_listener_ears
, delta
);
826 vol
= powf( vol
, 5.0f
);
828 framevol_l
*= (vol
* 0.5f
) * (1.0f
- pan
);
829 framevol_r
*= (vol
* 0.5f
) * (1.0f
+ pan
);
831 if( !(ch
->source
->flags
& AUDIO_FLAG_NO_DOPPLER
) ){
832 const float vs
= 323.0f
;
834 float dv
= v3_dot(delta
,vg_audio
.internal_listener_velocity
);
835 float doppler
= (vs
+dv
)/vs
;
836 doppler
= vg_clampf( doppler
, 0.6f
, 1.4f
);
838 if( fabsf(doppler
-1.0f
) > 0.01f
)
839 frame_samplerate
*= doppler
;
843 if( !vg_validf( framevol_l
) ) vg_fatal_exit_loop( "NaN left channel" );
844 if( !vg_validf( framevol_r
) ) vg_fatal_exit_loop( "NaN right channel" );
845 if( !vg_validf( frame_samplerate
) )
846 vg_fatal_exit_loop( "NaN sample rate" );
849 u32 buffer_length
= AUDIO_MIX_FRAME_SIZE
;
850 if( frame_samplerate
!= 1.0f
){
851 float l
= ceilf( (float)(AUDIO_MIX_FRAME_SIZE
) * frame_samplerate
);
855 float pcf
[ AUDIO_MIX_FRAME_SIZE
* 2 * 2 ];
857 audio_channel_get_samples( ch
, buffer_length
, pcf
);
859 vg_profile_begin( &_vg_prof_audio_mix
);
861 float volume_movement
= ch
->volume_movement
;
862 float const fvolume_rate
= vg_maxf( 1.0f
, ch
->_
.volume_rate
);
863 const float inv_volume_rate
= 1.0f
/fvolume_rate
;
865 float volume
= ch
->_
.volume
;
866 const float volume_start
= ch
->volume_movement_start
;
867 const float volume_target
= ch
->_
.volume_target
;
869 for( u32 j
=0; j
<AUDIO_MIX_FRAME_SIZE
; j
++ ){
870 volume_movement
+= 1.0f
;
871 float movement_t
= volume_movement
* inv_volume_rate
;
872 movement_t
= vg_minf( movement_t
, 1.0f
);
873 volume
= vg_lerpf( volume_start
, volume_target
, movement_t
);
875 float vol_norm
= volume
* volume
;
878 vol_norm
*= 1.0f
+ audio_lfo_pull_sample(ch
->_
.lfo
) * ch
->_
.lfo_amount
;
880 float vol_l
= vol_norm
* framevol_l
,
881 vol_r
= vol_norm
* framevol_r
,
885 if( frame_samplerate
!= 1.0f
){
886 /* absolutely garbage resampling, but it will do
889 float sample_index
= frame_samplerate
* (float)j
;
890 float t
= vg_fractf( sample_index
);
892 u32 i0
= floorf( sample_index
),
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
;
899 sample_l
= pcf
[ j
*2+0 ];
900 sample_r
= pcf
[ j
*2+1 ];
903 buffer
[ j
*2+0 ] += sample_l
* vol_l
;
904 buffer
[ j
*2+1 ] += sample_r
* vol_r
;
907 ch
->volume_movement
+= AUDIO_MIX_FRAME_SIZE
;
908 ch
->volume_movement
= VG_MIN( ch
->volume_movement
, ch
->_
.volume_rate
);
909 ch
->_
.volume
= volume
;
911 vg_profile_end( &_vg_prof_audio_mix
);
914 VG_STATIC
void audio_mixer_callback( void *user
, u8
*stream
, int byte_count
)
917 * Copy data and move edit flags to commit flags
918 * ------------------------------------------------------------- */
921 v3_copy( vg_audio
.external_listener_pos
, vg_audio
.internal_listener_pos
);
922 v3_copy( vg_audio
.external_listener_ears
, vg_audio
.internal_listener_ears
);
923 v3_copy( vg_audio
.external_lister_velocity
,
924 vg_audio
.internal_listener_velocity
);
925 vg_audio
.internal_global_volume
= vg_audio
.external_global_volume
;
927 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
928 audio_channel
*ch
= &vg_audio
.channels
[i
];
933 if( ch
->activity
== k_channel_activity_alive
){
934 if( (ch
->cursor
>= ch
->source_length
) &&
935 !(ch
->flags
& AUDIO_FLAG_LOOP
) )
937 ch
->activity
= k_channel_activity_end
;
941 /* process relinquishments */
942 if( (ch
->activity
!= k_channel_activity_reset
) && ch
->_
.relinquished
){
943 if( (ch
->activity
== k_channel_activity_end
)
944 || (ch
->_
.volume
== 0.0f
)
945 || (ch
->activity
== k_channel_activity_error
) )
947 ch
->_
.relinquished
= 0;
949 ch
->activity
= k_channel_activity_reset
;
954 /* process new channels */
955 if( ch
->activity
== k_channel_activity_reset
){
956 ch
->_
= ch
->editable_state
;
958 ch
->source_length
= 0;
959 ch
->activity
= k_channel_activity_wake
;
962 if( ch
->editble_state_write_mask
& AUDIO_EDIT_OWNERSHIP
)
963 ch
->_
.relinquished
= ch
->editable_state
.relinquished
;
965 ch
->editable_state
.relinquished
= ch
->_
.relinquished
;
968 if( ch
->editble_state_write_mask
& AUDIO_EDIT_VOLUME
){
969 ch
->_
.volume
= ch
->editable_state
.volume
;
970 ch
->_
.volume_target
= ch
->editable_state
.volume
;
973 ch
->editable_state
.volume
= ch
->_
.volume
;
977 if( ch
->editble_state_write_mask
& AUDIO_EDIT_VOLUME_SLOPE
){
978 ch
->volume_movement_start
= ch
->_
.volume
;
979 ch
->volume_movement
= 0;
981 ch
->_
.volume_target
= ch
->editable_state
.volume_target
;
982 ch
->_
.volume_rate
= ch
->editable_state
.volume_rate
;
985 ch
->editable_state
.volume_target
= ch
->_
.volume_target
;
986 ch
->editable_state
.volume_rate
= ch
->_
.volume_rate
;
990 if( ch
->editble_state_write_mask
& AUDIO_EDIT_SAMPLING_RATE
)
991 ch
->_
.sampling_rate
= ch
->editable_state
.sampling_rate
;
993 ch
->editable_state
.sampling_rate
= ch
->_
.sampling_rate
;
996 if( ch
->editble_state_write_mask
& AUDIO_EDIT_LFO_ATTACHMENT
){
997 ch
->_
.lfo
= ch
->editable_state
.lfo
;
998 ch
->_
.lfo_amount
= ch
->editable_state
.lfo_amount
;
1001 ch
->editable_state
.lfo
= ch
->_
.lfo
;
1002 ch
->editable_state
.lfo_amount
= ch
->_
.lfo_amount
;
1006 if( ch
->editble_state_write_mask
& AUDIO_EDIT_SPACIAL
)
1007 v4_copy( ch
->editable_state
.spacial_falloff
,ch
->_
.spacial_falloff
);
1009 v4_copy( ch
->_
.spacial_falloff
,ch
->editable_state
.spacial_falloff
);
1012 /* currently readonly, i guess */
1013 ch
->editable_state
.pan_target
= ch
->_
.pan_target
;
1014 ch
->editable_state
.pan
= ch
->_
.pan
;
1015 ch
->editble_state_write_mask
= 0x00;
1018 for( int i
=0; i
<AUDIO_LFOS
; i
++ ){
1019 audio_lfo
*lfo
= &vg_audio
.oscillators
[ i
];
1021 if( lfo
->editble_state_write_mask
& AUDIO_EDIT_LFO_WAVE
){
1022 lfo
->_
.wave_type
= lfo
->editable_state
.wave_type
;
1024 if( lfo
->_
.wave_type
== k_lfo_polynomial_bipolar
){
1025 lfo
->_
.polynomial_coefficient
=
1026 lfo
->editable_state
.polynomial_coefficient
;
1027 lfo
->sqrt_polynomial_coefficient
=
1028 sqrtf(lfo
->_
.polynomial_coefficient
);
1032 if( lfo
->editble_state_write_mask
& AUDIO_EDIT_LFO_PERIOD
){
1033 if( lfo
->_
.period
){
1034 float t
= lfo
->time
;
1035 t
/= (float)lfo
->_
.period
;
1037 lfo
->_
.period
= lfo
->editable_state
.period
;
1038 lfo
->time
= lfo
->_
.period
* t
;
1042 lfo
->_
.period
= lfo
->editable_state
.period
;
1046 lfo
->editble_state_write_mask
= 0x00;
1049 dsp_update_tunings();
1054 * ------------------------------------------------------------- */
1055 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1056 audio_channel
*ch
= &vg_audio
.channels
[i
];
1058 if( ch
->activity
== k_channel_activity_wake
){
1059 if( audio_channel_load_source( ch
) )
1060 ch
->activity
= k_channel_activity_alive
;
1062 ch
->activity
= k_channel_activity_error
;
1068 * -------------------------------------------------------- */
1069 int frame_count
= byte_count
/(2*sizeof(float));
1072 float *pOut32F
= (float *)stream
;
1073 for( int i
=0; i
<frame_count
*2; i
++ )
1076 for( int i
=0; i
<AUDIO_LFOS
; i
++ ){
1077 audio_lfo
*lfo
= &vg_audio
.oscillators
[i
];
1078 lfo
->time_startframe
= lfo
->time
;
1081 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1082 audio_channel
*ch
= &vg_audio
.channels
[i
];
1084 if( ch
->activity
== k_channel_activity_alive
){
1086 ch
->_
.lfo
->time
= ch
->_
.lfo
->time_startframe
;
1088 u32 remaining
= frame_count
,
1092 audio_channel_mix( ch
, pOut32F
+subpos
);
1093 remaining
-= AUDIO_MIX_FRAME_SIZE
;
1094 subpos
+= AUDIO_MIX_FRAME_SIZE
*2;
1099 vg_profile_begin( &_vg_prof_dsp
);
1101 for( int i
=0; i
<frame_count
; i
++ )
1102 vg_dsp_process( pOut32F
+ i
*2, pOut32F
+ i
*2 );
1104 vg_profile_end( &_vg_prof_dsp
);
1108 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1109 audio_channel
*ch
= &vg_audio
.channels
[i
];
1110 ch
->readable_activity
= ch
->activity
;
1113 /* Profiling information
1114 * ----------------------------------------------- */
1115 vg_profile_increment( &_vg_prof_audio_decode
);
1116 vg_profile_increment( &_vg_prof_audio_mix
);
1117 vg_profile_increment( &_vg_prof_dsp
);
1119 vg_prof_audio_mix
= _vg_prof_audio_mix
;
1120 vg_prof_audio_decode
= _vg_prof_audio_decode
;
1121 vg_prof_audio_dsp
= _vg_prof_dsp
;
1123 vg_audio
.samples_last
= frame_count
;
1125 if( vg_audio
.debug_dsp
){
1126 vg_dsp_update_texture();
1132 VG_STATIC
void audio_clip_load( audio_clip
*clip
, void *lin_alloc
)
1134 if( lin_alloc
== NULL
)
1135 lin_alloc
= vg_audio
.audio_pool
;
1137 /* load in directly */
1138 u32 format
= clip
->flags
& AUDIO_FLAG_FORMAT
;
1140 /* TODO: This contains audio_lock() and unlock, but i don't know why
1141 * can probably remove them. Low priority to check this */
1143 /* TODO: packed files for vorbis etc, should take from data if its not not
1144 * NULL when we get the clip
1147 if( format
== k_audio_format_vorbis
){
1149 vg_fatal_exit_loop( "No path specified, embeded vorbis unsupported" );
1153 clip
->data
= vg_file_read( lin_alloc
, clip
->path
, &clip
->size
);
1157 vg_fatal_exit_loop( "Audio failed to load" );
1159 float mb
= (float)(clip
->size
) / (1024.0f
*1024.0f
);
1160 vg_info( "Loaded audio clip '%s' (%.1fmb)\n", clip
->path
, mb
);
1162 else if( format
== k_audio_format_stereo
){
1163 vg_fatal_exit_loop( "Unsupported format (Stereo uncompressed)" );
1165 else if( format
== k_audio_format_bird
){
1167 vg_fatal_exit_loop( "No data, external birdsynth unsupported" );
1170 u32 total_size
= clip
->size
+ sizeof(struct synth_bird
);
1171 total_size
-= sizeof(struct synth_bird_settings
);
1172 total_size
= vg_align8( total_size
);
1174 if( total_size
> AUDIO_DECODE_SIZE
)
1175 vg_fatal_exit_loop( "Bird coding too long\n" );
1177 struct synth_bird
*bird
= vg_linear_alloc( lin_alloc
, total_size
);
1178 memcpy( &bird
->settings
, clip
->data
, clip
->size
);
1181 clip
->size
= total_size
;
1183 vg_info( "Loaded bird synthesis pattern (%u bytes)\n", total_size
);
1187 vg_fatal_exit_loop( "No path specified, embeded mono unsupported" );
1190 vg_linear_clear( vg_mem
.scratch
);
1193 stb_vorbis_alloc alloc
= {
1194 .alloc_buffer
= vg_linear_alloc( vg_mem
.scratch
, AUDIO_DECODE_SIZE
),
1195 .alloc_buffer_length_in_bytes
= AUDIO_DECODE_SIZE
1198 void *filedata
= vg_file_read( vg_mem
.scratch
, clip
->path
, &fsize
);
1201 stb_vorbis
*decoder
= stb_vorbis_open_memory(
1202 filedata
, fsize
, &err
, &alloc
);
1205 vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
1207 vg_fatal_exit_loop( "Vorbis decode error" );
1210 /* only mono is supported in uncompressed */
1211 u32 length_samples
= stb_vorbis_stream_length_in_samples( decoder
),
1212 data_size
= length_samples
* sizeof(i16
);
1215 clip
->data
= vg_linear_alloc( lin_alloc
, vg_align8(data_size
) );
1216 clip
->size
= length_samples
;
1219 int read_samples
= stb_vorbis_get_samples_i16_downmixed(
1220 decoder
, clip
->data
, length_samples
);
1222 if( read_samples
!= length_samples
)
1223 vg_fatal_exit_loop( "Decode error" );
1225 float mb
= (float)(data_size
) / (1024.0f
*1024.0f
);
1226 vg_info( "Loaded audio clip '%s' (%.1fmb) %u samples\n", clip
->path
, mb
,
1231 VG_STATIC
void audio_clip_loadn( audio_clip
*arr
, int count
, void *lin_alloc
)
1233 for( int i
=0; i
<count
; i
++ )
1234 audio_clip_load( &arr
[i
], lin_alloc
);
1237 VG_STATIC
void audio_require_clip_loaded( audio_clip
*clip
)
1239 if( clip
->data
&& clip
->size
)
1243 vg_fatal_exit_loop( "Must load audio clip before playing! \n" );
1250 VG_STATIC
void audio_debug_ui( m4x4f mtx_pv
)
1252 if( !vg_audio
.debug_ui
)
1257 glBindTexture( GL_TEXTURE_2D
, vg_dsp
.view_texture
);
1258 glTexSubImage2D( GL_TEXTURE_2D
, 0, 0, 0, 256, 256,
1259 GL_RGBA
, GL_UNSIGNED_BYTE
,
1260 vg_dsp
.view_texture_buffer
);
1264 * -----------------------------------------------------------------------
1267 float budget
= ((double)vg_audio
.samples_last
/ 44100.0) * 1000.0;
1268 vg_profile_drawn( (struct vg_profile
*[]){ &vg_prof_audio_decode
,
1270 &vg_prof_audio_dsp
}, 3,
1271 budget
, (ui_rect
){ 4, VG_PROFILE_SAMPLE_COUNT
*2 + 8,
1278 vg_uictx
.cursor
[0] = 512 + 8;
1279 vg_uictx
.cursor
[1] = VG_PROFILE_SAMPLE_COUNT
*2+8+24+12+12;
1280 vg_uictx
.cursor
[2] = 150;
1281 vg_uictx
.cursor
[3] = 12;
1283 if( vg_audio
.debug_dsp
){
1284 ui_rect view_thing
= { 4, vg
.window_y
-512-4, 512, 512 };
1285 ui_push_image( view_thing
, vg_dsp
.view_texture
);
1288 float mb1
= 1024.0f
*1024.0f
,
1289 usage
= vg_linear_get_cur( vg_audio
.audio_pool
) / mb1
,
1290 total
= vg_linear_get_capacity( vg_audio
.audio_pool
) / mb1
,
1291 percent
= (usage
/total
) * 100.0f
;
1293 snprintf( perf
, 127, "Mem: %.1f/%.1fmb (%.1f%%)\n", usage
, total
, percent
);
1295 ui_text( vg_uictx
.cursor
, perf
, 1, 0 );
1296 vg_uictx
.cursor
[1] += 20;
1298 ui_rect overlap_buffer
[ AUDIO_CHANNELS
];
1299 u32 overlap_length
= 0;
1301 /* Draw audio stack */
1302 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1303 audio_channel
*ch
= &vg_audio
.channels
[i
];
1305 vg_uictx
.cursor
[2] = 400;
1306 vg_uictx
.cursor
[3] = 18;
1310 if( !ch
->allocated
){
1311 ui_fill_rect( vg_uictx
.cursor
, 0x50333333 );
1314 vg_uictx
.cursor
[1] += 1;
1318 const char *formats
[] =
1338 const char *activties
[] =
1347 u32 format_index
= (ch
->source
->flags
& AUDIO_FLAG_FORMAT
)>>9;
1349 snprintf( perf
, 127, "%02d %c%c%cD %s [%s] %4.2fv'%s'",
1351 (ch
->editable_state
.relinquished
)? 'r': '_',
1354 formats
[format_index
],
1355 activties
[ch
->readable_activity
],
1356 ch
->editable_state
.volume
,
1359 ui_fill_rect( vg_uictx
.cursor
, 0xa0000000 | ch
->colour
);
1361 vg_uictx
.cursor
[0] += 2;
1362 vg_uictx
.cursor
[1] += 2;
1363 ui_text( vg_uictx
.cursor
, perf
, 1, 0 );
1366 vg_uictx
.cursor
[1] += 1;
1368 if( AUDIO_FLAG_SPACIAL_3D
){
1370 v3_copy( ch
->editable_state
.spacial_falloff
, wpos
);
1373 m4x4_mulv( mtx_pv
, wpos
, wpos
);
1375 if( wpos
[3] > 0.0f
){
1376 v2_muls( wpos
, (1.0f
/wpos
[3]) * 0.5f
, wpos
);
1377 v2_add( wpos
, (v2f
){ 0.5f
, 0.5f
}, wpos
);
1380 wr
[0] = vg_clampf(wpos
[0] * vg
.window_x
, -32000.0f
,32000.0f
);
1381 wr
[1] = vg_clampf((1.0f
-wpos
[1]) * vg
.window_y
,-32000.0f
,32000.0f
);
1385 for( int j
=0; j
<12; j
++ ){
1387 for( int k
=0; k
<overlap_length
; k
++ ){
1388 ui_px
*wk
= overlap_buffer
[k
];
1389 if( ((wr
[0] <= wk
[0]+wk
[2]) && (wr
[0]+wr
[2] >= wk
[0])) &&
1390 ((wr
[1] <= wk
[1]+wk
[3]) && (wr
[1]+wr
[3] >= wk
[1])) )
1403 ui_text( wr
, perf
, 1, 0 );
1405 ui_rect_copy( wr
, overlap_buffer
[ overlap_length
++ ] );
1413 #endif /* VG_AUDIO_H */