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
55 /* Vorbis will ALWAYS use the maximum amount of channels it can */
56 //#define AUDIO_FLAG_MONO 0x100 NOTE: This is the default, so its not used
57 //#define AUDIO_FLAG_STEREO 0x200
58 //#define AUDIO_FLAG_VORBIS 0x400
59 //#define AUDIO_FLAG_BIRD_SYNTH 0x800
61 #define AUDIO_FLAG_FORMAT 0x1E00
65 k_audio_format_mono
= 0x000u
,
66 k_audio_format_stereo
= 0x200u
,
67 k_audio_format_vorbis
= 0x400u
,
68 k_audio_format_none0
= 0x600u
,
69 k_audio_format_none1
= 0x800u
,
70 k_audio_format_none2
= 0xA00u
,
71 k_audio_format_none3
= 0xC00u
,
72 k_audio_format_none4
= 0xE00u
,
74 k_audio_format_bird
= 0x1000u
,
75 k_audio_format_none5
= 0x1200u
,
76 k_audio_format_none6
= 0x1400u
,
77 k_audio_format_none7
= 0x1600u
,
78 k_audio_format_none8
= 0x1800u
,
79 k_audio_format_none9
= 0x1A00u
,
80 k_audio_format_none10
= 0x1C00u
,
81 k_audio_format_none11
= 0x1E00u
,
84 #define AUDIO_DECODE_SIZE (1024*256) /* 256 kb decoding buffers */
85 #define AUDIO_MUTE_VOLUME 0.0f
86 #define AUDIO_BASE_VOLUME 1.0f
88 typedef struct audio_clip audio_clip
;
89 typedef struct audio_channel audio_channel
;
90 typedef struct audio_lfo audio_lfo
;
99 static struct vg_audio_system
{
100 SDL_AudioDeviceID sdl_output_device
;
109 SDL_SpinLock sl_checker
,
113 u32 time
, time_startframe
;
114 float sqrt_polynomial_coefficient
;
121 k_lfo_polynomial_bipolar
126 float polynomial_coefficient
;
129 u32 editble_state_write_mask
;
131 oscillators
[ AUDIO_LFOS
];
133 struct audio_channel
{
137 char name
[32]; /* only editable while allocated == 0 */
138 audio_clip
*source
; /* ... */
140 u32 colour
; /* ... */
142 /* internal non-readable state
143 * -----------------------------*/
144 u32 cursor
, source_length
;
146 float volume_movement_start
,
153 struct synth_bird
*bird_handle
;
154 stb_vorbis
*vorbis_handle
;
157 stb_vorbis_alloc vorbis_alloc
;
159 enum channel_activity
{
160 k_channel_activity_reset
, /* will advance if allocated==1, to wake */
161 k_channel_activity_wake
, /* will advance to either of next two */
162 k_channel_activity_alive
,
163 k_channel_activity_end
,
164 k_channel_activity_error
170 * editable structure, can be modified inside _lock and _unlock
171 * the edit mask tells which to copy into internal _, or to discard
172 * ----------------------------------------------------------------------
174 struct channel_state
{
177 float volume
, /* current volume */
178 volume_target
, /* target volume */
186 v4f spacial_falloff
; /* xyz, range */
192 u32 editble_state_write_mask
;
194 channels
[ AUDIO_CHANNELS
];
196 int debug_ui
, debug_ui_3d
, debug_dsp
;
198 v3f internal_listener_pos
,
199 internal_listener_ears
,
200 internal_listener_velocity
,
202 external_listener_pos
,
203 external_listener_ears
,
204 external_lister_velocity
;
206 float internal_global_volume
,
207 external_global_volume
;
209 vg_audio
= { .external_global_volume
= 1.0f
};
211 #include "vg/vg_audio_dsp.h"
213 static struct vg_profile
214 _vg_prof_audio_decode
= {.mode
= k_profile_mode_accum
,
215 .name
= "[T2] audio_decode()"},
216 _vg_prof_audio_mix
= {.mode
= k_profile_mode_accum
,
217 .name
= "[T2] audio_mix()"},
218 _vg_prof_dsp
= {.mode
= k_profile_mode_accum
,
219 .name
= "[T2] dsp_process()"},
220 vg_prof_audio_decode
,
225 * These functions are called from the main thread and used to prevent bad
226 * access. TODO: They should be no-ops in release builds.
228 VG_STATIC
int audio_lock_checker_load(void)
231 SDL_AtomicLock( &vg_audio
.sl_checker
);
232 value
= vg_audio
.sync_locked
;
233 SDL_AtomicUnlock( &vg_audio
.sl_checker
);
237 VG_STATIC
void audio_lock_checker_store( int value
)
239 SDL_AtomicLock( &vg_audio
.sl_checker
);
240 vg_audio
.sync_locked
= value
;
241 SDL_AtomicUnlock( &vg_audio
.sl_checker
);
244 VG_STATIC
void audio_require_lock(void)
246 if( audio_lock_checker_load() )
249 vg_error( "Modifying sound effects systems requires locking\n" );
253 VG_STATIC
void audio_lock(void)
255 SDL_AtomicLock( &vg_audio
.sl_sync
);
256 audio_lock_checker_store(1);
259 VG_STATIC
void audio_unlock(void)
261 audio_lock_checker_store(0);
262 SDL_AtomicUnlock( &vg_audio
.sl_sync
);
265 VG_STATIC
void audio_mixer_callback( void *user
, u8
*stream
, int frame_count
);
266 VG_STATIC
void vg_audio_init(void)
268 /* TODO: Move here? */
269 vg_var_push( (struct vg_var
){
270 .name
= "debug_audio",
271 .data
= &vg_audio
.debug_ui
,
272 .data_type
= k_var_dtype_i32
,
273 .opt_i32
= { .min
=0, .max
=1, .clamp
=1 },
277 vg_var_push( (struct vg_var
){
279 .data
= &vg_audio
.debug_dsp
,
280 .data_type
= k_var_dtype_i32
,
281 .opt_i32
= { .min
=0, .max
=1, .clamp
=1 },
285 vg_var_push( (struct vg_var
){
287 .data
= &vg_audio
.external_global_volume
,
288 .data_type
= k_var_dtype_f32
,
289 .opt_f32
= { .min
=0.0f
, .max
=2.0f
, .clamp
=1 },
293 /* allocate memory */
296 vg_audio
.audio_pool
=
297 vg_create_linear_allocator( vg_mem
.rtmemory
, 1024*1024*32,
301 u32 decode_size
= AUDIO_DECODE_SIZE
* AUDIO_CHANNELS
;
302 vg_audio
.decode_buffer
= vg_linear_alloc( vg_mem
.rtmemory
, decode_size
);
306 SDL_AudioSpec spec_desired
, spec_got
;
307 spec_desired
.callback
= audio_mixer_callback
;
308 spec_desired
.channels
= 2;
309 spec_desired
.format
= AUDIO_F32
;
310 spec_desired
.freq
= 44100;
311 spec_desired
.padding
= 0;
312 spec_desired
.samples
= AUDIO_FRAME_SIZE
;
313 spec_desired
.silence
= 0;
314 spec_desired
.size
= 0;
315 spec_desired
.userdata
= NULL
;
317 vg_audio
.sdl_output_device
=
318 SDL_OpenAudioDevice( NULL
, 0, &spec_desired
, &spec_got
,0 );
320 if( vg_audio
.sdl_output_device
){
321 SDL_PauseAudioDevice( vg_audio
.sdl_output_device
, 0 );
325 "SDL_OpenAudioDevice failed. Your default audio device must support:\n"
326 " Frequency: 44100 hz\n"
327 " Buffer size: 512\n"
329 " Format: s16 or f32\n" );
332 vg_success( "Ready\n" );
335 VG_STATIC
void vg_audio_free(void)
338 SDL_CloseAudioDevice( vg_audio
.sdl_output_device
);
345 #define AUDIO_EDIT_VOLUME_SLOPE 0x1
346 #define AUDIO_EDIT_VOLUME 0x2
347 #define AUDIO_EDIT_LFO_PERIOD 0x4
348 #define AUDIO_EDIT_LFO_WAVE 0x8
349 #define AUDIO_EDIT_LFO_ATTACHMENT 0x10
350 #define AUDIO_EDIT_SPACIAL 0x20
351 #define AUDIO_EDIT_OWNERSHIP 0x40
352 #define AUDIO_EDIT_SAMPLING_RATE 0x80
354 static void audio_channel_init( audio_channel
*ch
, audio_clip
*clip
, u32 flags
)
359 ch
->colour
= 0x00333333;
361 if( (ch
->source
->flags
& AUDIO_FLAG_FORMAT
) == k_audio_format_bird
)
362 strcpy( ch
->name
, "[array]" );
364 strncpy( ch
->name
, clip
->path
, 31 );
368 ch
->editable_state
.relinquished
= 0;
369 ch
->editable_state
.volume
= 1.0f
;
370 ch
->editable_state
.volume_target
= 1.0f
;
371 ch
->editable_state
.pan
= 0.0f
;
372 ch
->editable_state
.pan_target
= 0.0f
;
373 ch
->editable_state
.volume_rate
= 0;
374 ch
->editable_state
.pan_rate
= 0;
375 v4_copy((v4f
){0.0f
,0.0f
,0.0f
,1.0f
},ch
->editable_state
.spacial_falloff
);
376 ch
->editable_state
.lfo
= NULL
;
377 ch
->editable_state
.lfo_amount
= 0.0f
;
378 ch
->editable_state
.sampling_rate
= 1.0f
;
379 ch
->editble_state_write_mask
= 0x00;
382 static void audio_channel_group( audio_channel
*ch
, u32 group
)
385 ch
->colour
= ((group
* 29986577) & 0x00ffffff) | 0xff000000;
388 static audio_channel
*audio_get_first_idle_channel(void)
390 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
391 audio_channel
*ch
= &vg_audio
.channels
[i
];
393 if( !ch
->allocated
){
401 static audio_channel
*audio_get_group_idle_channel( u32 group
, u32 max_count
)
404 audio_channel
*dest
= NULL
;
406 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
407 audio_channel
*ch
= &vg_audio
.channels
[i
];
410 if( ch
->group
== group
){
420 if( dest
&& (count
< max_count
) ){
427 static audio_channel
*audio_get_group_first_active_channel( u32 group
)
429 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
430 audio_channel
*ch
= &vg_audio
.channels
[i
];
431 if( ch
->allocated
&& (ch
->group
== group
) )
437 static int audio_channel_finished( audio_channel
*ch
)
439 if( ch
->readable_activity
== k_channel_activity_end
)
445 static audio_channel
*audio_relinquish_channel( audio_channel
*ch
)
447 ch
->editable_state
.relinquished
= 1;
448 ch
->editble_state_write_mask
|= AUDIO_EDIT_OWNERSHIP
;
452 static void audio_channel_slope_volume( audio_channel
*ch
, float length
,
455 ch
->editable_state
.volume_target
= new_volume
;
456 ch
->editable_state
.volume_rate
= length
* 44100.0f
;
457 ch
->editble_state_write_mask
|= AUDIO_EDIT_VOLUME_SLOPE
;
460 static void audio_channel_set_sampling_rate( audio_channel
*ch
, float rate
)
462 ch
->editable_state
.sampling_rate
= rate
;
463 ch
->editble_state_write_mask
|= AUDIO_EDIT_SAMPLING_RATE
;
466 static void audio_channel_edit_volume( audio_channel
*ch
,
467 float new_volume
, int instant
)
470 ch
->editable_state
.volume
= new_volume
;
471 ch
->editble_state_write_mask
|= AUDIO_EDIT_VOLUME
;
474 audio_channel_slope_volume( ch
, 0.05f
, new_volume
);
478 static audio_channel
*audio_channel_fadeout( audio_channel
*ch
, float length
)
480 audio_channel_slope_volume( ch
, length
, 0.0f
);
481 return audio_relinquish_channel( ch
);
484 static void audio_channel_fadein( audio_channel
*ch
, float length
)
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
)
497 ch
= audio_channel_fadeout( ch
, length
);
499 audio_channel
*replacement
= audio_get_first_idle_channel();
502 audio_channel_init( replacement
, new_clip
, flags
);
503 audio_channel_fadein( replacement
, length
);
509 static void audio_channel_sidechain_lfo( audio_channel
*ch
, int lfo_id
,
512 ch
->editable_state
.lfo
= &vg_audio
.oscillators
[ lfo_id
];
513 ch
->editable_state
.lfo_amount
= amount
;
514 ch
->editble_state_write_mask
|= AUDIO_EDIT_LFO_ATTACHMENT
;
517 static void audio_channel_set_spacial( audio_channel
*ch
, v3f co
, float range
)
519 if( ch
->flags
& AUDIO_FLAG_SPACIAL_3D
){
520 v3_copy( co
, ch
->editable_state
.spacial_falloff
);
523 ch
->editable_state
.spacial_falloff
[3] = 1.0f
;
525 ch
->editable_state
.spacial_falloff
[3] = 1.0f
/range
;
527 ch
->editble_state_write_mask
|= AUDIO_EDIT_SPACIAL
;
530 vg_warn( "Tried to set spacialization paramaters for 2D channel (%s)\n",
535 static int audio_oneshot_3d( audio_clip
*clip
, v3f position
,
536 float range
, float volume
)
538 audio_channel
*ch
= audio_get_first_idle_channel();
541 audio_channel_init( ch
, clip
, AUDIO_FLAG_SPACIAL_3D
);
542 audio_channel_set_spacial( ch
, position
, range
);
543 audio_channel_edit_volume( ch
, volume
, 1 );
544 ch
= audio_relinquish_channel( ch
);
552 static int audio_oneshot( audio_clip
*clip
, float volume
, float pan
)
554 audio_channel
*ch
= audio_get_first_idle_channel();
557 audio_channel_init( ch
, clip
, 0x00 );
558 audio_channel_edit_volume( ch
, volume
, 1 );
559 ch
= audio_relinquish_channel( ch
);
567 static void audio_set_lfo_wave( int id
, enum lfo_wave_type type
,
570 audio_lfo
*lfo
= &vg_audio
.oscillators
[ id
];
571 lfo
->editable_state
.polynomial_coefficient
= coefficient
;
572 lfo
->editable_state
.wave_type
= type
;
574 lfo
->editble_state_write_mask
|= AUDIO_EDIT_LFO_WAVE
;
577 static void audio_set_lfo_frequency( int id
, float freq
)
579 audio_lfo
*lfo
= &vg_audio
.oscillators
[ id
];
580 lfo
->editable_state
.period
= 44100.0f
/ freq
;
581 lfo
->editble_state_write_mask
|= AUDIO_EDIT_LFO_PERIOD
;
587 * -----------------------------------------------------------------------------
589 static int audio_channel_load_source( audio_channel
*ch
)
591 u32 format
= ch
->source
->flags
& AUDIO_FLAG_FORMAT
;
593 if( format
== k_audio_format_vorbis
){
594 /* Setup vorbis decoder */
595 u32 index
= ch
- vg_audio
.channels
;
597 u8
*buf
= (u8
*)vg_audio
.decode_buffer
,
598 *loc
= &buf
[AUDIO_DECODE_SIZE
*index
];
600 stb_vorbis_alloc alloc
= {
601 .alloc_buffer
= (char *)loc
,
602 .alloc_buffer_length_in_bytes
= AUDIO_DECODE_SIZE
606 stb_vorbis
*decoder
= stb_vorbis_open_memory(
608 ch
->source
->size
, &err
, &alloc
);
611 vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
612 ch
->source
->path
, err
);
616 ch
->source_length
= stb_vorbis_stream_length_in_samples( decoder
);
617 ch
->vorbis_handle
= decoder
;
620 else if( format
== k_audio_format_bird
){
621 u32 index
= ch
- vg_audio
.channels
;
623 u8
*buf
= (u8
*)vg_audio
.decode_buffer
;
624 struct synth_bird
*loc
= (void *)&buf
[AUDIO_DECODE_SIZE
*index
];
626 memcpy( loc
, ch
->source
->data
, ch
->source
->size
);
627 synth_bird_reset( loc
);
629 ch
->bird_handle
= loc
;
630 ch
->source_length
= synth_bird_get_length_in_samples( loc
);
632 else if( format
== k_audio_format_stereo
){
633 ch
->source_length
= ch
->source
->size
/ 2;
636 ch
->source_length
= ch
->source
->size
;
642 VG_STATIC
void audio_decode_uncompressed_mono( i16
*src
, u32 count
, float *dst
)
644 for( u32 i
=0; i
<count
; i
++ ){
645 dst
[ i
*2 + 0 ] = ((float)src
[i
]) * (1.0f
/32767.0f
);
646 dst
[ i
*2 + 1 ] = ((float)src
[i
]) * (1.0f
/32767.0f
);
651 * adapted from stb_vorbis.h, since the original does not handle mono->stereo
654 stb_vorbis_get_samples_float_interleaved_stereo( stb_vorbis
*f
, float *buffer
,
658 c
= VG_MIN( 1, f
->channels
- 1 );
661 int k
= f
->channel_buffer_end
- f
->channel_buffer_start
;
666 for( int j
=0; j
< k
; ++j
) {
667 *buffer
++ = f
->channel_buffers
[ 0 ][f
->channel_buffer_start
+j
];
668 *buffer
++ = f
->channel_buffers
[ c
][f
->channel_buffer_start
+j
];
672 f
->channel_buffer_start
+= k
;
677 if( !stb_vorbis_get_frame_float( f
, NULL
, NULL
))
685 * ........ more wrecked code sorry!
688 stb_vorbis_get_samples_i16_downmixed( stb_vorbis
*f
, i16
*buffer
, int len
)
691 c
= VG_MIN( 1, f
->channels
- 1 );
694 int k
= f
->channel_buffer_end
- f
->channel_buffer_start
;
699 for( int j
=0; j
< k
; ++j
) {
700 float sl
= f
->channel_buffers
[ 0 ][f
->channel_buffer_start
+j
],
701 sr
= f
->channel_buffers
[ c
][f
->channel_buffer_start
+j
];
703 *buffer
++ = vg_clampf( 0.5f
*(sl
+sr
), -1.0f
, 1.0f
) * 32767.0f
;
704 //*buffer++ = vg_clampf( sr, -1.0f, 1.0f ) * 32767.0f;
708 f
->channel_buffer_start
+= k
;
713 if( !stb_vorbis_get_frame_float( f
, NULL
, NULL
))
720 static inline float audio_lfo_pull_sample( audio_lfo
*lfo
)
724 if( lfo
->time
>= lfo
->_
.period
)
728 t
/= (float)lfo
->_
.period
;
730 if( lfo
->_
.wave_type
== k_lfo_polynomial_bipolar
){
746 return (( 2.0f
* lfo
->sqrt_polynomial_coefficient
* t
) /
747 /* --------------------------------------- */
748 ( 1.0f
+ lfo
->_
.polynomial_coefficient
* t
*t
)
757 static void audio_channel_get_samples( audio_channel
*ch
,
758 u32 count
, float *buf
)
760 vg_profile_begin( &_vg_prof_audio_decode
);
762 u32 remaining
= count
;
765 u32 format
= ch
->source
->flags
& AUDIO_FLAG_FORMAT
;
768 u32 samples_this_run
= VG_MIN(remaining
, ch
->source_length
- ch
->cursor
);
769 remaining
-= samples_this_run
;
771 float *dst
= &buf
[ buffer_pos
* 2 ];
773 if( format
== k_audio_format_stereo
){
774 for( int i
=0;i
<samples_this_run
; i
++ ){
779 else if( format
== k_audio_format_vorbis
){
780 int read_samples
= stb_vorbis_get_samples_float_interleaved_stereo(
785 if( read_samples
!= samples_this_run
){
786 vg_warn( "Invalid samples read (%s)\n", ch
->source
->path
);
788 for( int i
=0; i
<samples_this_run
; i
++ ){
794 else if( format
== k_audio_format_bird
){
795 synth_bird_generate_samples( ch
->bird_handle
, dst
, samples_this_run
);
798 i16
*src_buffer
= ch
->source
->data
,
799 *src
= &src_buffer
[ch
->cursor
];
801 audio_decode_uncompressed_mono( src
, samples_this_run
, dst
);
804 ch
->cursor
+= samples_this_run
;
805 buffer_pos
+= samples_this_run
;
807 if( (ch
->flags
& AUDIO_FLAG_LOOP
) && remaining
){
808 if( format
== k_audio_format_vorbis
)
809 stb_vorbis_seek_start( ch
->vorbis_handle
);
810 else if( format
== k_audio_format_bird
)
811 synth_bird_reset( ch
->bird_handle
);
821 buf
[ buffer_pos
*2 + 0 ] = 0.0f
;
822 buf
[ buffer_pos
*2 + 1 ] = 0.0f
;
828 vg_profile_end( &_vg_prof_audio_decode
);
831 static void audio_channel_mix( audio_channel
*ch
, float *buffer
)
833 float framevol_l
= vg_audio
.internal_global_volume
,
834 framevol_r
= vg_audio
.internal_global_volume
;
836 float frame_samplerate
= ch
->_
.sampling_rate
;
838 if( ch
->flags
& AUDIO_FLAG_SPACIAL_3D
){
840 v3_sub( ch
->_
.spacial_falloff
, vg_audio
.internal_listener_pos
, delta
);
842 float dist
= v3_length( delta
),
843 vol
= vg_maxf( 0.0f
, 1.0f
- ch
->_
.spacial_falloff
[3]*dist
);
849 v3_muls( delta
, 1.0f
/dist
, delta
);
850 float pan
= v3_dot( vg_audio
.internal_listener_ears
, delta
);
851 vol
= powf( vol
, 5.0f
);
853 framevol_l
*= (vol
* 0.5f
) * (1.0f
- pan
);
854 framevol_r
*= (vol
* 0.5f
) * (1.0f
+ pan
);
856 if( !(ch
->source
->flags
& AUDIO_FLAG_NO_DOPPLER
) ){
857 const float vs
= 323.0f
;
859 float dv
= v3_dot(delta
,vg_audio
.internal_listener_velocity
);
860 float doppler
= (vs
+dv
)/vs
;
861 doppler
= vg_clampf( doppler
, 0.6f
, 1.4f
);
863 if( fabsf(doppler
-1.0f
) > 0.01f
)
864 frame_samplerate
*= doppler
;
868 if( !vg_validf( framevol_l
) ) vg_fatal_exit_loop( "NaN left channel" );
869 if( !vg_validf( framevol_r
) ) vg_fatal_exit_loop( "NaN right channel" );
870 if( !vg_validf( frame_samplerate
) )
871 vg_fatal_exit_loop( "NaN sample rate" );
874 u32 buffer_length
= AUDIO_MIX_FRAME_SIZE
;
875 if( frame_samplerate
!= 1.0f
){
876 float l
= ceilf( (float)(AUDIO_MIX_FRAME_SIZE
) * frame_samplerate
);
880 float pcf
[ AUDIO_MIX_FRAME_SIZE
* 2 * 2 ];
882 audio_channel_get_samples( ch
, buffer_length
, pcf
);
884 vg_profile_begin( &_vg_prof_audio_mix
);
886 float volume_movement
= ch
->volume_movement
;
887 float const fvolume_rate
= vg_maxf( 1.0f
, ch
->_
.volume_rate
);
888 const float inv_volume_rate
= 1.0f
/fvolume_rate
;
890 float volume
= ch
->_
.volume
;
891 const float volume_start
= ch
->volume_movement_start
;
892 const float volume_target
= ch
->_
.volume_target
;
894 for( u32 j
=0; j
<AUDIO_MIX_FRAME_SIZE
; j
++ ){
895 volume_movement
+= 1.0f
;
896 float movement_t
= volume_movement
* inv_volume_rate
;
897 movement_t
= vg_minf( movement_t
, 1.0f
);
898 volume
= vg_lerpf( volume_start
, volume_target
, movement_t
);
900 float vol_norm
= volume
* volume
;
903 vol_norm
*= 1.0f
+ audio_lfo_pull_sample(ch
->_
.lfo
) * ch
->_
.lfo_amount
;
905 float vol_l
= vol_norm
* framevol_l
,
906 vol_r
= vol_norm
* framevol_r
,
910 if( frame_samplerate
!= 1.0f
){
911 /* absolutely garbage resampling, but it will do
914 float sample_index
= frame_samplerate
* (float)j
;
915 float t
= vg_fractf( sample_index
);
917 u32 i0
= floorf( sample_index
),
920 sample_l
= pcf
[ i0
*2+0 ]*(1.0f
-t
) + pcf
[ i1
*2+0 ]*t
;
921 sample_r
= pcf
[ i0
*2+1 ]*(1.0f
-t
) + pcf
[ i1
*2+1 ]*t
;
924 sample_l
= pcf
[ j
*2+0 ];
925 sample_r
= pcf
[ j
*2+1 ];
928 buffer
[ j
*2+0 ] += sample_l
* vol_l
;
929 buffer
[ j
*2+1 ] += sample_r
* vol_r
;
932 ch
->volume_movement
+= AUDIO_MIX_FRAME_SIZE
;
933 ch
->volume_movement
= VG_MIN( ch
->volume_movement
, ch
->_
.volume_rate
);
934 ch
->_
.volume
= volume
;
936 vg_profile_end( &_vg_prof_audio_mix
);
939 VG_STATIC
void audio_mixer_callback( void *user
, u8
*stream
, int byte_count
)
942 * Copy data and move edit flags to commit flags
943 * ------------------------------------------------------------- */
946 v3_copy( vg_audio
.external_listener_pos
, vg_audio
.internal_listener_pos
);
947 v3_copy( vg_audio
.external_listener_ears
, vg_audio
.internal_listener_ears
);
948 v3_copy( vg_audio
.external_lister_velocity
,
949 vg_audio
.internal_listener_velocity
);
950 vg_audio
.internal_global_volume
= vg_audio
.external_global_volume
;
952 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
953 audio_channel
*ch
= &vg_audio
.channels
[i
];
958 if( ch
->activity
== k_channel_activity_alive
){
959 if( (ch
->cursor
>= ch
->source_length
) &&
960 !(ch
->flags
& AUDIO_FLAG_LOOP
) )
962 ch
->activity
= k_channel_activity_end
;
966 /* process relinquishments */
967 if( (ch
->activity
!= k_channel_activity_reset
) && ch
->_
.relinquished
){
968 if( (ch
->activity
== k_channel_activity_end
)
969 || (ch
->_
.volume
== 0.0f
)
970 || (ch
->activity
== k_channel_activity_error
) )
972 ch
->_
.relinquished
= 0;
974 ch
->activity
= k_channel_activity_reset
;
979 /* process new channels */
980 if( ch
->activity
== k_channel_activity_reset
){
981 ch
->_
= ch
->editable_state
;
983 ch
->source_length
= 0;
984 ch
->activity
= k_channel_activity_wake
;
987 if( ch
->editble_state_write_mask
& AUDIO_EDIT_OWNERSHIP
)
988 ch
->_
.relinquished
= ch
->editable_state
.relinquished
;
990 ch
->editable_state
.relinquished
= ch
->_
.relinquished
;
993 if( ch
->editble_state_write_mask
& AUDIO_EDIT_VOLUME
){
994 ch
->_
.volume
= ch
->editable_state
.volume
;
995 ch
->_
.volume_target
= ch
->editable_state
.volume
;
998 ch
->editable_state
.volume
= ch
->_
.volume
;
1002 if( ch
->editble_state_write_mask
& AUDIO_EDIT_VOLUME_SLOPE
){
1003 ch
->volume_movement_start
= ch
->_
.volume
;
1004 ch
->volume_movement
= 0;
1006 ch
->_
.volume_target
= ch
->editable_state
.volume_target
;
1007 ch
->_
.volume_rate
= ch
->editable_state
.volume_rate
;
1010 ch
->editable_state
.volume_target
= ch
->_
.volume_target
;
1011 ch
->editable_state
.volume_rate
= ch
->_
.volume_rate
;
1015 if( ch
->editble_state_write_mask
& AUDIO_EDIT_SAMPLING_RATE
)
1016 ch
->_
.sampling_rate
= ch
->editable_state
.sampling_rate
;
1018 ch
->editable_state
.sampling_rate
= ch
->_
.sampling_rate
;
1021 if( ch
->editble_state_write_mask
& AUDIO_EDIT_LFO_ATTACHMENT
){
1022 ch
->_
.lfo
= ch
->editable_state
.lfo
;
1023 ch
->_
.lfo_amount
= ch
->editable_state
.lfo_amount
;
1026 ch
->editable_state
.lfo
= ch
->_
.lfo
;
1027 ch
->editable_state
.lfo_amount
= ch
->_
.lfo_amount
;
1031 if( ch
->editble_state_write_mask
& AUDIO_EDIT_SPACIAL
)
1032 v4_copy( ch
->editable_state
.spacial_falloff
,ch
->_
.spacial_falloff
);
1034 v4_copy( ch
->_
.spacial_falloff
,ch
->editable_state
.spacial_falloff
);
1037 /* currently readonly, i guess */
1038 ch
->editable_state
.pan_target
= ch
->_
.pan_target
;
1039 ch
->editable_state
.pan
= ch
->_
.pan
;
1040 ch
->editble_state_write_mask
= 0x00;
1043 for( int i
=0; i
<AUDIO_LFOS
; i
++ ){
1044 audio_lfo
*lfo
= &vg_audio
.oscillators
[ i
];
1046 if( lfo
->editble_state_write_mask
& AUDIO_EDIT_LFO_WAVE
){
1047 lfo
->_
.wave_type
= lfo
->editable_state
.wave_type
;
1049 if( lfo
->_
.wave_type
== k_lfo_polynomial_bipolar
){
1050 lfo
->_
.polynomial_coefficient
=
1051 lfo
->editable_state
.polynomial_coefficient
;
1052 lfo
->sqrt_polynomial_coefficient
=
1053 sqrtf(lfo
->_
.polynomial_coefficient
);
1057 if( lfo
->editble_state_write_mask
& AUDIO_EDIT_LFO_PERIOD
){
1058 if( lfo
->_
.period
){
1059 float t
= lfo
->time
;
1060 t
/= (float)lfo
->_
.period
;
1062 lfo
->_
.period
= lfo
->editable_state
.period
;
1063 lfo
->time
= lfo
->_
.period
* t
;
1067 lfo
->_
.period
= lfo
->editable_state
.period
;
1071 lfo
->editble_state_write_mask
= 0x00;
1074 dsp_update_tunings();
1079 * ------------------------------------------------------------- */
1080 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1081 audio_channel
*ch
= &vg_audio
.channels
[i
];
1083 if( ch
->activity
== k_channel_activity_wake
){
1084 if( audio_channel_load_source( ch
) )
1085 ch
->activity
= k_channel_activity_alive
;
1087 ch
->activity
= k_channel_activity_error
;
1093 * -------------------------------------------------------- */
1094 int frame_count
= byte_count
/(2*sizeof(float));
1097 float *pOut32F
= (float *)stream
;
1098 for( int i
=0; i
<frame_count
*2; i
++ )
1101 for( int i
=0; i
<AUDIO_LFOS
; i
++ ){
1102 audio_lfo
*lfo
= &vg_audio
.oscillators
[i
];
1103 lfo
->time_startframe
= lfo
->time
;
1106 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1107 audio_channel
*ch
= &vg_audio
.channels
[i
];
1109 if( ch
->activity
== k_channel_activity_alive
){
1111 ch
->_
.lfo
->time
= ch
->_
.lfo
->time_startframe
;
1113 u32 remaining
= frame_count
,
1117 audio_channel_mix( ch
, pOut32F
+subpos
);
1118 remaining
-= AUDIO_MIX_FRAME_SIZE
;
1119 subpos
+= AUDIO_MIX_FRAME_SIZE
*2;
1124 vg_profile_begin( &_vg_prof_dsp
);
1126 for( int i
=0; i
<frame_count
; i
++ )
1127 vg_dsp_process( pOut32F
+ i
*2, pOut32F
+ i
*2 );
1129 vg_profile_end( &_vg_prof_dsp
);
1133 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1134 audio_channel
*ch
= &vg_audio
.channels
[i
];
1135 ch
->readable_activity
= ch
->activity
;
1138 /* Profiling information
1139 * ----------------------------------------------- */
1140 vg_profile_increment( &_vg_prof_audio_decode
);
1141 vg_profile_increment( &_vg_prof_audio_mix
);
1142 vg_profile_increment( &_vg_prof_dsp
);
1144 vg_prof_audio_mix
= _vg_prof_audio_mix
;
1145 vg_prof_audio_decode
= _vg_prof_audio_decode
;
1146 vg_prof_audio_dsp
= _vg_prof_dsp
;
1148 vg_audio
.samples_last
= frame_count
;
1150 if( vg_audio
.debug_dsp
){
1151 vg_dsp_update_texture();
1157 VG_STATIC
void audio_clip_load( audio_clip
*clip
, void *lin_alloc
)
1159 if( lin_alloc
== NULL
)
1160 lin_alloc
= vg_audio
.audio_pool
;
1162 /* load in directly */
1163 u32 format
= clip
->flags
& AUDIO_FLAG_FORMAT
;
1165 /* TODO: This contains audio_lock() and unlock, but i don't know why
1166 * can probably remove them. Low priority to check this */
1168 /* TODO: packed files for vorbis etc, should take from data if its not not
1169 * NULL when we get the clip
1172 if( format
== k_audio_format_vorbis
){
1174 vg_fatal_exit_loop( "No path specified, embeded vorbis unsupported" );
1178 clip
->data
= vg_file_read( lin_alloc
, clip
->path
, &clip
->size
);
1182 vg_fatal_exit_loop( "Audio failed to load" );
1184 float mb
= (float)(clip
->size
) / (1024.0f
*1024.0f
);
1185 vg_info( "Loaded audio clip '%s' (%.1fmb)\n", clip
->path
, mb
);
1187 else if( format
== k_audio_format_stereo
){
1188 vg_fatal_exit_loop( "Unsupported format (Stereo uncompressed)" );
1190 else if( format
== k_audio_format_bird
){
1192 vg_fatal_exit_loop( "No data, external birdsynth unsupported" );
1195 u32 total_size
= clip
->size
+ sizeof(struct synth_bird
);
1196 total_size
-= sizeof(struct synth_bird_settings
);
1197 total_size
= vg_align8( total_size
);
1199 if( total_size
> AUDIO_DECODE_SIZE
)
1200 vg_fatal_exit_loop( "Bird coding too long\n" );
1202 struct synth_bird
*bird
= vg_linear_alloc( lin_alloc
, total_size
);
1203 memcpy( &bird
->settings
, clip
->data
, clip
->size
);
1206 clip
->size
= total_size
;
1208 vg_info( "Loaded bird synthesis pattern (%u bytes)\n", total_size
);
1212 vg_fatal_exit_loop( "No path specified, embeded mono unsupported" );
1215 vg_linear_clear( vg_mem
.scratch
);
1218 stb_vorbis_alloc alloc
= {
1219 .alloc_buffer
= vg_linear_alloc( vg_mem
.scratch
, AUDIO_DECODE_SIZE
),
1220 .alloc_buffer_length_in_bytes
= AUDIO_DECODE_SIZE
1223 void *filedata
= vg_file_read( vg_mem
.scratch
, clip
->path
, &fsize
);
1226 stb_vorbis
*decoder
= stb_vorbis_open_memory(
1227 filedata
, fsize
, &err
, &alloc
);
1230 vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
1232 vg_fatal_exit_loop( "Vorbis decode error" );
1235 /* only mono is supported in uncompressed */
1236 u32 length_samples
= stb_vorbis_stream_length_in_samples( decoder
),
1237 data_size
= length_samples
* sizeof(i16
);
1240 clip
->data
= vg_linear_alloc( lin_alloc
, vg_align8(data_size
) );
1241 clip
->size
= length_samples
;
1244 int read_samples
= stb_vorbis_get_samples_i16_downmixed(
1245 decoder
, clip
->data
, length_samples
);
1247 if( read_samples
!= length_samples
)
1248 vg_fatal_exit_loop( "Decode error" );
1250 float mb
= (float)(data_size
) / (1024.0f
*1024.0f
);
1251 vg_info( "Loaded audio clip '%s' (%.1fmb) %u samples\n", clip
->path
, mb
,
1256 VG_STATIC
void audio_clip_loadn( audio_clip
*arr
, int count
, void *lin_alloc
)
1258 for( int i
=0; i
<count
; i
++ )
1259 audio_clip_load( &arr
[i
], lin_alloc
);
1262 VG_STATIC
void audio_require_clip_loaded( audio_clip
*clip
)
1264 if( clip
->data
&& clip
->size
)
1268 vg_fatal_exit_loop( "Must load audio clip before playing! \n" );
1275 VG_STATIC
void audio_debug_ui( m4x4f mtx_pv
)
1277 if( !vg_audio
.debug_ui
)
1282 glBindTexture( GL_TEXTURE_2D
, vg_dsp
.view_texture
);
1283 glTexSubImage2D( GL_TEXTURE_2D
, 0, 0, 0, 256, 256,
1284 GL_RGBA
, GL_UNSIGNED_BYTE
,
1285 vg_dsp
.view_texture_buffer
);
1289 * -----------------------------------------------------------------------
1292 float budget
= ((double)vg_audio
.samples_last
/ 44100.0) * 1000.0;
1293 vg_profile_drawn( (struct vg_profile
*[]){ &vg_prof_audio_decode
,
1295 &vg_prof_audio_dsp
}, 3,
1296 budget
, (ui_rect
){ 4, VG_PROFILE_SAMPLE_COUNT
*2 + 8,
1303 vg_uictx
.cursor
[0] = 512 + 8;
1304 vg_uictx
.cursor
[1] = VG_PROFILE_SAMPLE_COUNT
*2+8+24+12+12;
1305 vg_uictx
.cursor
[2] = 150;
1306 vg_uictx
.cursor
[3] = 12;
1308 if( vg_audio
.debug_dsp
){
1309 ui_rect view_thing
= { 4, vg
.window_y
-512-4, 512, 512 };
1310 ui_push_image( view_thing
, vg_dsp
.view_texture
);
1313 float mb1
= 1024.0f
*1024.0f
,
1314 usage
= vg_linear_get_cur( vg_audio
.audio_pool
) / mb1
,
1315 total
= vg_linear_get_capacity( vg_audio
.audio_pool
) / mb1
,
1316 percent
= (usage
/total
) * 100.0f
;
1318 snprintf( perf
, 127, "Mem: %.1f/%.1fmb (%.1f%%)\n", usage
, total
, percent
);
1320 ui_text( vg_uictx
.cursor
, perf
, 1, 0 );
1321 vg_uictx
.cursor
[1] += 20;
1323 ui_rect overlap_buffer
[ AUDIO_CHANNELS
];
1324 u32 overlap_length
= 0;
1326 /* Draw audio stack */
1327 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1328 audio_channel
*ch
= &vg_audio
.channels
[i
];
1330 vg_uictx
.cursor
[2] = 400;
1331 vg_uictx
.cursor
[3] = 18;
1335 if( !ch
->allocated
){
1336 ui_fill_rect( vg_uictx
.cursor
, 0x50333333 );
1339 vg_uictx
.cursor
[1] += 1;
1343 const char *formats
[] =
1363 const char *activties
[] =
1372 u32 format_index
= (ch
->source
->flags
& AUDIO_FLAG_FORMAT
)>>9;
1374 snprintf( perf
, 127, "%02d %c%c%cD %s [%s] %4.2fv'%s'",
1376 (ch
->editable_state
.relinquished
)? 'r': '_',
1379 formats
[format_index
],
1380 activties
[ch
->readable_activity
],
1381 ch
->editable_state
.volume
,
1384 ui_fill_rect( vg_uictx
.cursor
, 0xa0000000 | ch
->colour
);
1386 vg_uictx
.cursor
[0] += 2;
1387 vg_uictx
.cursor
[1] += 2;
1388 ui_text( vg_uictx
.cursor
, perf
, 1, 0 );
1391 vg_uictx
.cursor
[1] += 1;
1393 if( AUDIO_FLAG_SPACIAL_3D
){
1395 v3_copy( ch
->editable_state
.spacial_falloff
, wpos
);
1398 m4x4_mulv( mtx_pv
, wpos
, wpos
);
1400 if( wpos
[3] > 0.0f
){
1401 v2_muls( wpos
, (1.0f
/wpos
[3]) * 0.5f
, wpos
);
1402 v2_add( wpos
, (v2f
){ 0.5f
, 0.5f
}, wpos
);
1405 wr
[0] = vg_clampf(wpos
[0] * vg
.window_x
, -32000.0f
,32000.0f
);
1406 wr
[1] = vg_clampf((1.0f
-wpos
[1]) * vg
.window_y
,-32000.0f
,32000.0f
);
1410 for( int j
=0; j
<12; j
++ ){
1412 for( int k
=0; k
<overlap_length
; k
++ ){
1413 ui_px
*wk
= overlap_buffer
[k
];
1414 if( ((wr
[0] <= wk
[0]+wk
[2]) && (wr
[0]+wr
[2] >= wk
[0])) &&
1415 ((wr
[1] <= wk
[1]+wk
[3]) && (wr
[1]+wr
[3] >= wk
[1])) )
1428 ui_text( wr
, perf
, 1, 0 );
1430 ui_rect_copy( wr
, overlap_buffer
[ overlap_length
++ ] );
1438 #endif /* VG_AUDIO_H */