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_mutex
*mux_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 /* System queue, and access from thread 0 */
197 int debug_ui
, debug_ui_3d
;
205 volume_target_internal
,
208 vg_audio
= { .volume_console
= 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_LockMutex( vg_audio
.mux_checker
);
231 value
= vg_audio
.sync_locked
;
232 SDL_UnlockMutex( vg_audio
.mux_checker
);
236 VG_STATIC
void audio_lock_checker_store( int value
)
238 SDL_LockMutex( vg_audio
.mux_checker
);
239 vg_audio
.sync_locked
= value
;
240 SDL_UnlockMutex( vg_audio
.mux_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_LockMutex( vg_audio
.mux_sync
);
255 audio_lock_checker_store(1);
258 VG_STATIC
void audio_unlock(void)
260 audio_lock_checker_store(0);
261 SDL_UnlockMutex( vg_audio
.mux_sync
);
264 VG_STATIC
void audio_mixer_callback( void *user
, u8
*stream
, int frame_count
);
265 VG_STATIC
void vg_audio_init(void)
267 vg_audio
.mux_checker
= SDL_CreateMutex();
268 vg_audio
.mux_sync
= SDL_CreateMutex();
270 /* TODO: Move here? */
271 vg_var_push( (struct vg_var
){
272 .name
= "debug_audio",
273 .data
= &vg_audio
.debug_ui
,
274 .data_type
= k_var_dtype_i32
,
275 .opt_i32
= { .min
=0, .max
=1, .clamp
=1 },
279 vg_var_push( (struct vg_var
){
281 .data
= &vg_audio
.volume_console
,
282 .data_type
= k_var_dtype_f32
,
283 .opt_f32
= { .min
=0.0f
, .max
=2.0f
, .clamp
=1 },
287 /* allocate memory */
290 vg_audio
.audio_pool
=
291 vg_create_linear_allocator( vg_mem
.rtmemory
, 1024*1024*32,
295 u32 decode_size
= AUDIO_DECODE_SIZE
* AUDIO_CHANNELS
;
296 vg_audio
.decode_buffer
= vg_linear_alloc( vg_mem
.rtmemory
, decode_size
);
300 SDL_AudioSpec spec_desired
, spec_got
;
301 spec_desired
.callback
= audio_mixer_callback
;
302 spec_desired
.channels
= 2;
303 spec_desired
.format
= AUDIO_F32
;
304 spec_desired
.freq
= 44100;
305 spec_desired
.padding
= 0;
306 spec_desired
.samples
= AUDIO_FRAME_SIZE
;
307 spec_desired
.silence
= 0;
308 spec_desired
.size
= 0;
309 spec_desired
.userdata
= NULL
;
311 vg_audio
.sdl_output_device
=
312 SDL_OpenAudioDevice( NULL
, 0, &spec_desired
, &spec_got
,0 );
314 if( vg_audio
.sdl_output_device
){
315 SDL_PauseAudioDevice( vg_audio
.sdl_output_device
, 0 );
319 "SDL_OpenAudioDevice failed. Your default audio device must support:\n"
320 " Frequency: 44100 hz\n"
321 " Buffer size: 512\n"
323 " Format: s16 or f32\n" );
326 vg_success( "Ready\n" );
329 VG_STATIC
void vg_audio_free(void)
332 SDL_CloseAudioDevice( vg_audio
.sdl_output_device
);
339 #define AUDIO_EDIT_VOLUME_SLOPE 0x1
340 #define AUDIO_EDIT_VOLUME 0x2
341 #define AUDIO_EDIT_LFO_PERIOD 0x4
342 #define AUDIO_EDIT_LFO_WAVE 0x8
343 #define AUDIO_EDIT_LFO_ATTACHMENT 0x10
344 #define AUDIO_EDIT_SPACIAL 0x20
345 #define AUDIO_EDIT_OWNERSHIP 0x40
346 #define AUDIO_EDIT_SAMPLING_RATE 0x80
348 static void audio_channel_init( audio_channel
*ch
, audio_clip
*clip
, u32 flags
)
353 ch
->colour
= 0x00333333;
355 if( (ch
->source
->flags
& AUDIO_FLAG_FORMAT
) == k_audio_format_bird
)
356 strcpy( ch
->name
, "[array]" );
358 strncpy( ch
->name
, clip
->path
, 31 );
362 ch
->editable_state
.relinquished
= 0;
363 ch
->editable_state
.volume
= 1.0f
;
364 ch
->editable_state
.volume_target
= 1.0f
;
365 ch
->editable_state
.pan
= 0.0f
;
366 ch
->editable_state
.pan_target
= 0.0f
;
367 ch
->editable_state
.volume_rate
= 0;
368 ch
->editable_state
.pan_rate
= 0;
369 v4_copy((v4f
){0.0f
,0.0f
,0.0f
,1.0f
},ch
->editable_state
.spacial_falloff
);
370 ch
->editable_state
.lfo
= NULL
;
371 ch
->editable_state
.lfo_amount
= 0.0f
;
372 ch
->editable_state
.sampling_rate
= 1.0f
;
373 ch
->editble_state_write_mask
= 0x00;
376 static audio_channel
*audio_get_first_idle_channel(void)
378 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
379 audio_channel
*ch
= &vg_audio
.channels
[i
];
381 if( !ch
->allocated
){
389 static audio_channel
*audio_get_group_idle_channel( u32 group
, u32 max_count
)
394 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
395 audio_channel
*ch
= &vg_audio
.channels
[i
];
398 if( ch
->group
== group
){
408 if( dest
&& (count
< max_count
) ){
415 static audio_channel
*audio_get_group_first_active_channel( u32 group
)
417 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
418 audio_channel
*ch
= &vg_audio
.channels
[i
];
419 if( ch
->allocated
&& (ch
->group
== group
) )
425 static int audio_channel_finished( audio_channel
*ch
)
427 if( ch
->readable_activity
== k_channel_activity_end
)
433 static audio_channel
*audio_relinquish_channel( audio_channel
*ch
)
435 ch
->editable_state
.relinquished
= 1;
436 ch
->editble_state_write_mask
|= AUDIO_EDIT_OWNERSHIP
;
440 static void audio_channel_slope_volume( audio_channel
*ch
, float length
,
443 ch
->editable_state
.volume_target
= new_volume
;
444 ch
->editable_state
.volume_rate
= length
* 44100.0f
;
445 ch
->editble_state_write_mask
|= AUDIO_EDIT_VOLUME_SLOPE
;
448 static void audio_channel_set_sampling_rate( audio_channel
*ch
, float rate
)
450 ch
->editable_state
.sampling_rate
= rate
;
451 ch
->editble_state_write_mask
|= AUDIO_EDIT_SAMPLING_RATE
;
454 static void audio_channel_edit_volume( audio_channel
*ch
,
455 float new_volume
, int instant
)
458 ch
->editable_state
.volume
= new_volume
;
459 ch
->editble_state_write_mask
|= AUDIO_EDIT_VOLUME
;
462 audio_channel_slope_volume( ch
, 0.05f
, new_volume
);
466 static audio_channel
*audio_channel_fadeout( audio_channel
*ch
, float length
)
468 audio_channel_slope_volume( ch
, length
, 0.0f
);
469 return audio_relinquish_channel( ch
);
472 static void audio_channel_fadein( audio_channel
*ch
, float length
)
474 audio_channel_edit_volume( ch
, 0.0f
, 1 );
475 audio_channel_slope_volume( ch
, length
, 1.0f
);
478 static audio_channel
*audio_channel_crossfade( audio_channel
*ch
,
479 audio_clip
*new_clip
,
480 float length
, u32 flags
)
485 ch
= audio_channel_fadeout( ch
, length
);
487 audio_channel
*replacement
= audio_get_first_idle_channel();
490 audio_channel_init( replacement
, new_clip
, flags
);
491 audio_channel_fadein( replacement
, length
);
497 static void audio_channel_sidechain_lfo( audio_channel
*ch
, int lfo_id
,
500 ch
->editable_state
.lfo
= &vg_audio
.oscillators
[ lfo_id
];
501 ch
->editable_state
.lfo_amount
= amount
;
502 ch
->editble_state_write_mask
|= AUDIO_EDIT_LFO_ATTACHMENT
;
505 static void audio_channel_set_spacial( audio_channel
*ch
, v3f co
, float range
)
507 if( ch
->flags
& AUDIO_FLAG_SPACIAL_3D
){
508 v3_copy( co
, ch
->editable_state
.spacial_falloff
);
511 ch
->editable_state
.spacial_falloff
[3] = 1.0f
;
513 ch
->editable_state
.spacial_falloff
[3] = 1.0f
/range
;
515 ch
->editble_state_write_mask
|= AUDIO_EDIT_SPACIAL
;
518 vg_warn( "Tried to set spacialization paramaters for 2D channel (%s)\n",
523 static int audio_oneshot_3d( audio_clip
*clip
, v3f position
,
524 float range
, float volume
)
526 audio_channel
*ch
= audio_get_first_idle_channel();
529 audio_channel_init( ch
, clip
, AUDIO_FLAG_SPACIAL_3D
);
530 audio_channel_set_spacial( ch
, position
, range
);
531 audio_channel_edit_volume( ch
, volume
, 1 );
532 ch
= audio_relinquish_channel( ch
);
540 static int audio_oneshot( audio_clip
*clip
, float volume
, float pan
)
542 audio_channel
*ch
= audio_get_first_idle_channel();
545 audio_channel_init( ch
, clip
, 0x00 );
546 audio_channel_edit_volume( ch
, volume
, 1 );
547 ch
= audio_relinquish_channel( ch
);
555 static void audio_set_lfo_wave( int id
, enum lfo_wave_type type
,
558 audio_lfo
*lfo
= &vg_audio
.oscillators
[ id
];
559 lfo
->editable_state
.polynomial_coefficient
= coefficient
;
560 lfo
->editable_state
.wave_type
= type
;
562 lfo
->editble_state_write_mask
|= AUDIO_EDIT_LFO_WAVE
;
565 static void audio_set_lfo_frequency( int id
, float freq
)
567 audio_lfo
*lfo
= &vg_audio
.oscillators
[ id
];
568 lfo
->editable_state
.period
= 44100.0f
/ freq
;
569 lfo
->editble_state_write_mask
|= AUDIO_EDIT_LFO_PERIOD
;
575 * -----------------------------------------------------------------------------
577 static int audio_channel_load_source( audio_channel
*ch
)
579 u32 format
= ch
->source
->flags
& AUDIO_FLAG_FORMAT
;
581 if( format
== k_audio_format_vorbis
){
582 /* Setup vorbis decoder */
583 u32 index
= ch
- vg_audio
.channels
;
585 u8
*buf
= (u8
*)vg_audio
.decode_buffer
,
586 *loc
= &buf
[AUDIO_DECODE_SIZE
*index
];
588 stb_vorbis_alloc alloc
= {
589 .alloc_buffer
= (char *)loc
,
590 .alloc_buffer_length_in_bytes
= AUDIO_DECODE_SIZE
594 stb_vorbis
*decoder
= stb_vorbis_open_memory(
596 ch
->source
->size
, &err
, &alloc
);
599 vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
600 ch
->source
->path
, err
);
604 ch
->source_length
= stb_vorbis_stream_length_in_samples( decoder
);
605 ch
->vorbis_handle
= decoder
;
608 else if( format
== k_audio_format_bird
){
609 u32 index
= ch
- vg_audio
.channels
;
611 u8
*buf
= (u8
*)vg_audio
.decode_buffer
;
612 struct synth_bird
*loc
= (void *)&buf
[AUDIO_DECODE_SIZE
*index
];
614 memcpy( loc
, ch
->source
->data
, ch
->source
->size
);
615 synth_bird_reset( loc
);
617 ch
->bird_handle
= loc
;
618 ch
->source_length
= synth_bird_get_length_in_samples( loc
);
620 else if( format
== k_audio_format_stereo
){
621 ch
->source_length
= ch
->source
->size
/ 2;
624 ch
->source_length
= ch
->source
->size
;
630 VG_STATIC
void audio_decode_uncompressed_mono( i16
*src
, u32 count
, float *dst
)
632 for( u32 i
=0; i
<count
; i
++ ){
633 dst
[ i
*2 + 0 ] = ((float)src
[i
]) * (1.0f
/32767.0f
);
634 dst
[ i
*2 + 1 ] = ((float)src
[i
]) * (1.0f
/32767.0f
);
639 * adapted from stb_vorbis.h, since the original does not handle mono->stereo
642 stb_vorbis_get_samples_float_interleaved_stereo( stb_vorbis
*f
, float *buffer
,
646 c
= VG_MIN( 1, f
->channels
- 1 );
649 int k
= f
->channel_buffer_end
- f
->channel_buffer_start
;
654 for( int j
=0; j
< k
; ++j
) {
655 *buffer
++ = f
->channel_buffers
[ 0 ][f
->channel_buffer_start
+j
];
656 *buffer
++ = f
->channel_buffers
[ c
][f
->channel_buffer_start
+j
];
660 f
->channel_buffer_start
+= k
;
665 if( !stb_vorbis_get_frame_float( f
, NULL
, NULL
))
673 * ........ more wrecked code sorry!
676 stb_vorbis_get_samples_i16_downmixed( stb_vorbis
*f
, i16
*buffer
, int len
)
679 c
= VG_MIN( 1, f
->channels
- 1 );
682 int k
= f
->channel_buffer_end
- f
->channel_buffer_start
;
687 for( int j
=0; j
< k
; ++j
) {
688 float sl
= f
->channel_buffers
[ 0 ][f
->channel_buffer_start
+j
],
689 sr
= f
->channel_buffers
[ c
][f
->channel_buffer_start
+j
];
691 *buffer
++ = vg_clampf( 0.5f
*(sl
+sr
), -1.0f
, 1.0f
) * 32767.0f
;
692 //*buffer++ = vg_clampf( sr, -1.0f, 1.0f ) * 32767.0f;
696 f
->channel_buffer_start
+= k
;
701 if( !stb_vorbis_get_frame_float( f
, NULL
, NULL
))
708 static inline float audio_lfo_pull_sample( audio_lfo
*lfo
)
712 if( lfo
->time
>= lfo
->_
.period
)
716 t
/= (float)lfo
->_
.period
;
718 if( lfo
->_
.wave_type
== k_lfo_polynomial_bipolar
){
734 return (( 2.0f
* lfo
->sqrt_polynomial_coefficient
* t
) /
735 /* --------------------------------------- */
736 ( 1.0f
+ lfo
->_
.polynomial_coefficient
* t
*t
)
745 static void audio_channel_get_samples( audio_channel
*ch
,
746 u32 count
, float *buf
)
748 vg_profile_begin( &_vg_prof_audio_decode
);
750 u32 remaining
= count
;
753 u32 format
= ch
->source
->flags
& AUDIO_FLAG_FORMAT
;
756 u32 samples_this_run
= VG_MIN(remaining
, ch
->source_length
- ch
->cursor
);
757 remaining
-= samples_this_run
;
759 float *dst
= &buf
[ buffer_pos
* 2 ];
761 if( format
== k_audio_format_stereo
){
762 for( int i
=0;i
<samples_this_run
; i
++ ){
767 else if( format
== k_audio_format_vorbis
){
768 int read_samples
= stb_vorbis_get_samples_float_interleaved_stereo(
773 if( read_samples
!= samples_this_run
){
774 vg_warn( "Invalid samples read (%s)\n", ch
->source
->path
);
776 for( int i
=0; i
<samples_this_run
; i
++ ){
782 else if( format
== k_audio_format_bird
){
783 synth_bird_generate_samples( ch
->bird_handle
, dst
, samples_this_run
);
786 i16
*src_buffer
= ch
->source
->data
,
787 *src
= &src_buffer
[ch
->cursor
];
789 audio_decode_uncompressed_mono( src
, samples_this_run
, dst
);
792 ch
->cursor
+= samples_this_run
;
793 buffer_pos
+= samples_this_run
;
795 if( (ch
->flags
& AUDIO_FLAG_LOOP
) && remaining
){
796 if( format
== k_audio_format_vorbis
)
797 stb_vorbis_seek_start( ch
->vorbis_handle
);
798 else if( format
== k_audio_format_bird
)
799 synth_bird_reset( ch
->bird_handle
);
809 buf
[ buffer_pos
*2 + 0 ] = 0.0f
;
810 buf
[ buffer_pos
*2 + 1 ] = 0.0f
;
816 vg_profile_end( &_vg_prof_audio_decode
);
819 static void audio_channel_mix( audio_channel
*ch
, float *buffer
)
821 float framevol_l
= 1.0f
,
824 float frame_samplerate
= ch
->_
.sampling_rate
;
826 if( ch
->flags
& AUDIO_FLAG_SPACIAL_3D
){
828 v3_sub( ch
->_
.spacial_falloff
, vg_audio
.listener_pos
, delta
);
830 float dist
= v3_length( delta
),
831 vol
= vg_maxf( 0.0f
, 1.0f
- ch
->_
.spacial_falloff
[3]*dist
);
837 v3_muls( delta
, 1.0f
/dist
, delta
);
838 float pan
= v3_dot( vg_audio
.listener_ears
, delta
);
839 vol
= powf( vol
, 5.0f
);
841 framevol_l
*= (vol
* 0.5f
) * (1.0f
- pan
);
842 framevol_r
*= (vol
* 0.5f
) * (1.0f
+ pan
);
844 const float vs
= 323.0f
;
845 float doppler
= (vs
+v3_dot(delta
,vg_audio
.listener_velocity
))/vs
;
846 doppler
= vg_clampf( doppler
, 0.6f
, 1.4f
);
848 if( fabsf(doppler
-1.0f
) > 0.01f
)
849 frame_samplerate
*= doppler
;
852 if( !vg_validf( framevol_l
) ) vg_fatal_exit_loop( "NaN left channel" );
853 if( !vg_validf( framevol_r
) ) vg_fatal_exit_loop( "NaN right channel" );
854 if( !vg_validf( frame_samplerate
) )
855 vg_fatal_exit_loop( "NaN sample rate" );
858 u32 buffer_length
= AUDIO_MIX_FRAME_SIZE
;
859 if( frame_samplerate
!= 1.0f
){
860 float l
= ceilf( (float)(AUDIO_MIX_FRAME_SIZE
) * frame_samplerate
);
864 float pcf
[ AUDIO_MIX_FRAME_SIZE
* 2 * 2 ];
866 audio_channel_get_samples( ch
, buffer_length
, pcf
);
868 vg_profile_begin( &_vg_prof_audio_mix
);
870 float volume_movement
= ch
->volume_movement
;
871 float const fvolume_rate
= vg_maxf( 1.0f
, ch
->_
.volume_rate
);
872 const float inv_volume_rate
= 1.0f
/fvolume_rate
;
874 float volume
= ch
->_
.volume
;
875 const float volume_start
= ch
->volume_movement_start
;
876 const float volume_target
= ch
->_
.volume_target
;
878 for( u32 j
=0; j
<AUDIO_MIX_FRAME_SIZE
; j
++ ){
880 * there is some REALLY weird behaviour with minss,
881 * i cannot begin to guess what the cause is, but the bahaviour when
882 * the second argument is not 1.0 would seemingly tripple or up to
883 * eight times this routine.
885 * the times it would happen are when moving from empty space into areas
886 * with geometry. in the bvh for skate rift.
888 * it should be completely unrelated to this, but somehow -- it is
889 * effecting the speed of minss. and severely at that too.
892 volume_movement
+= 1.0f
;
893 float movement_t
= volume_movement
* inv_volume_rate
;
894 movement_t
= vg_minf( volume_movement
, 1.0f
);
895 volume
= vg_lerpf( volume_start
, volume_target
, movement_t
);
897 float vol_norm
= volume
* volume
;
900 vol_norm
*= 1.0f
+ audio_lfo_pull_sample(ch
->_
.lfo
) * ch
->_
.lfo_amount
;
902 float vol_l
= vol_norm
* framevol_l
,
903 vol_r
= vol_norm
* framevol_r
,
907 if( frame_samplerate
!= 1.0f
){
908 /* absolutely garbage resampling, but it will do
911 float sample_index
= frame_samplerate
* (float)j
;
912 float t
= vg_fractf( sample_index
);
914 u32 i0
= floorf( sample_index
),
917 sample_l
= pcf
[ i0
*2+0 ]*(1.0f
-t
) + pcf
[ i1
*2+0 ]*t
;
918 sample_r
= pcf
[ i0
*2+1 ]*(1.0f
-t
) + pcf
[ i1
*2+1 ]*t
;
921 sample_l
= pcf
[ j
*2+0 ];
922 sample_r
= pcf
[ j
*2+1 ];
925 buffer
[ j
*2+0 ] += sample_l
* vol_l
;
926 buffer
[ j
*2+1 ] += sample_r
* vol_r
;
929 ch
->volume_movement
+= AUDIO_MIX_FRAME_SIZE
;
930 ch
->volume_movement
= VG_MIN( ch
->volume_movement
, ch
->_
.volume_rate
);
931 ch
->_
.volume
= volume
;
933 vg_profile_end( &_vg_prof_audio_mix
);
936 VG_STATIC
void audio_mixer_callback( void *user
, u8
*stream
, int byte_count
)
939 * Copy data and move edit flags to commit flags
940 * ------------------------------------------------------------- */
942 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
943 audio_channel
*ch
= &vg_audio
.channels
[i
];
948 if( ch
->activity
== k_channel_activity_alive
){
949 if( (ch
->cursor
>= ch
->source_length
) &&
950 !(ch
->flags
& AUDIO_FLAG_LOOP
) )
952 ch
->activity
= k_channel_activity_end
;
956 /* process relinquishments */
957 if( (ch
->activity
!= k_channel_activity_reset
) && ch
->_
.relinquished
){
958 if( (ch
->activity
== k_channel_activity_end
)
959 || (ch
->_
.volume
== 0.0f
)
960 || (ch
->activity
== k_channel_activity_error
) )
962 ch
->_
.relinquished
= 0;
964 ch
->activity
= k_channel_activity_reset
;
969 /* process new channels */
970 if( ch
->activity
== k_channel_activity_reset
){
971 ch
->_
= ch
->editable_state
;
973 ch
->source_length
= 0;
974 ch
->activity
= k_channel_activity_wake
;
977 if( ch
->editble_state_write_mask
& AUDIO_EDIT_OWNERSHIP
)
978 ch
->_
.relinquished
= ch
->editable_state
.relinquished
;
980 ch
->editable_state
.relinquished
= ch
->_
.relinquished
;
983 if( ch
->editble_state_write_mask
& AUDIO_EDIT_VOLUME
){
984 ch
->_
.volume
= ch
->editable_state
.volume
;
985 ch
->_
.volume_target
= ch
->editable_state
.volume
;
988 ch
->editable_state
.volume
= ch
->_
.volume
;
992 if( ch
->editble_state_write_mask
& AUDIO_EDIT_VOLUME_SLOPE
){
993 ch
->volume_movement_start
= ch
->_
.volume
;
994 ch
->volume_movement
= 0;
996 ch
->_
.volume_target
= ch
->editable_state
.volume_target
;
997 ch
->_
.volume_rate
= ch
->editable_state
.volume_rate
;
1000 ch
->editable_state
.volume_target
= ch
->_
.volume_target
;
1001 ch
->editable_state
.volume_rate
= ch
->_
.volume_rate
;
1005 if( ch
->editble_state_write_mask
& AUDIO_EDIT_SAMPLING_RATE
)
1006 ch
->_
.sampling_rate
= ch
->editable_state
.sampling_rate
;
1008 ch
->editable_state
.sampling_rate
= ch
->_
.sampling_rate
;
1011 if( ch
->editble_state_write_mask
& AUDIO_EDIT_LFO_ATTACHMENT
){
1012 ch
->_
.lfo
= ch
->editable_state
.lfo
;
1013 ch
->_
.lfo_amount
= ch
->editable_state
.lfo_amount
;
1016 ch
->editable_state
.lfo
= ch
->_
.lfo
;
1017 ch
->editable_state
.lfo_amount
= ch
->_
.lfo_amount
;
1021 if( ch
->editble_state_write_mask
& AUDIO_EDIT_SPACIAL
)
1022 v4_copy( ch
->editable_state
.spacial_falloff
,ch
->_
.spacial_falloff
);
1024 v4_copy( ch
->_
.spacial_falloff
,ch
->editable_state
.spacial_falloff
);
1027 /* currently readonly, i guess */
1028 ch
->editable_state
.pan_target
= ch
->_
.pan_target
;
1029 ch
->editable_state
.pan
= ch
->_
.pan
;
1030 ch
->editble_state_write_mask
= 0x00;
1033 for( int i
=0; i
<AUDIO_LFOS
; i
++ ){
1034 audio_lfo
*lfo
= &vg_audio
.oscillators
[ i
];
1036 if( lfo
->editble_state_write_mask
& AUDIO_EDIT_LFO_WAVE
){
1037 lfo
->_
.wave_type
= lfo
->editable_state
.wave_type
;
1039 if( lfo
->_
.wave_type
== k_lfo_polynomial_bipolar
){
1040 lfo
->_
.polynomial_coefficient
=
1041 lfo
->editable_state
.polynomial_coefficient
;
1042 lfo
->sqrt_polynomial_coefficient
=
1043 sqrtf(lfo
->_
.polynomial_coefficient
);
1047 if( lfo
->editble_state_write_mask
& AUDIO_EDIT_LFO_PERIOD
){
1048 if( lfo
->_
.period
){
1049 float t
= lfo
->time
;
1050 t
/= (float)lfo
->_
.period
;
1052 lfo
->_
.period
= lfo
->editable_state
.period
;
1053 lfo
->time
= lfo
->_
.period
* t
;
1057 lfo
->_
.period
= lfo
->editable_state
.period
;
1061 lfo
->editble_state_write_mask
= 0x00;
1064 dsp_update_tunings();
1069 * ------------------------------------------------------------- */
1070 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1071 audio_channel
*ch
= &vg_audio
.channels
[i
];
1073 if( ch
->activity
== k_channel_activity_wake
){
1074 if( audio_channel_load_source( ch
) )
1075 ch
->activity
= k_channel_activity_alive
;
1077 ch
->activity
= k_channel_activity_error
;
1083 * -------------------------------------------------------- */
1084 int frame_count
= byte_count
/(2*sizeof(float));
1087 float *pOut32F
= (float *)stream
;
1088 for( int i
=0; i
<frame_count
*2; i
++ )
1091 for( int i
=0; i
<AUDIO_LFOS
; i
++ ){
1092 audio_lfo
*lfo
= &vg_audio
.oscillators
[i
];
1093 lfo
->time_startframe
= lfo
->time
;
1096 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1097 audio_channel
*ch
= &vg_audio
.channels
[i
];
1099 if( ch
->activity
== k_channel_activity_alive
){
1101 ch
->_
.lfo
->time
= ch
->_
.lfo
->time_startframe
;
1103 u32 remaining
= frame_count
,
1107 audio_channel_mix( ch
, pOut32F
+subpos
);
1108 remaining
-= AUDIO_MIX_FRAME_SIZE
;
1109 subpos
+= AUDIO_MIX_FRAME_SIZE
*2;
1114 vg_profile_begin( &_vg_prof_dsp
);
1116 for( int i
=0; i
<frame_count
; i
++ )
1117 vg_dsp_process( pOut32F
+ i
*2, pOut32F
+ i
*2 );
1119 vg_profile_end( &_vg_prof_dsp
);
1123 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1124 audio_channel
*ch
= &vg_audio
.channels
[i
];
1125 ch
->readable_activity
= ch
->activity
;
1128 /* Profiling information
1129 * ----------------------------------------------- */
1130 vg_profile_increment( &_vg_prof_audio_decode
);
1131 vg_profile_increment( &_vg_prof_audio_mix
);
1132 vg_profile_increment( &_vg_prof_dsp
);
1134 vg_prof_audio_mix
= _vg_prof_audio_mix
;
1135 vg_prof_audio_decode
= _vg_prof_audio_decode
;
1136 vg_prof_audio_dsp
= _vg_prof_dsp
;
1138 vg_audio
.samples_last
= frame_count
;
1140 if( vg_audio
.debug_ui
){
1141 vg_dsp_update_texture();
1147 VG_STATIC
void audio_clip_load( audio_clip
*clip
, void *lin_alloc
)
1149 if( lin_alloc
== NULL
)
1150 lin_alloc
= vg_audio
.audio_pool
;
1152 /* load in directly */
1153 u32 format
= clip
->flags
& AUDIO_FLAG_FORMAT
;
1155 /* TODO: This contains audio_lock() and unlock, but i don't know why
1156 * can probably remove them. Low priority to check this */
1158 /* TODO: packed files for vorbis etc, should take from data if its not not
1159 * NULL when we get the clip
1162 if( format
== k_audio_format_vorbis
){
1164 vg_fatal_exit_loop( "No path specified, embeded vorbis unsupported" );
1168 clip
->data
= vg_file_read( lin_alloc
, clip
->path
, &clip
->size
);
1172 vg_fatal_exit_loop( "Audio failed to load" );
1174 float mb
= (float)(clip
->size
) / (1024.0f
*1024.0f
);
1175 vg_info( "Loaded audio clip '%s' (%.1fmb)\n", clip
->path
, mb
);
1177 else if( format
== k_audio_format_stereo
){
1178 vg_fatal_exit_loop( "Unsupported format (Stereo uncompressed)" );
1180 else if( format
== k_audio_format_bird
){
1182 vg_fatal_exit_loop( "No data, external birdsynth unsupported" );
1185 u32 total_size
= clip
->size
+ sizeof(struct synth_bird
);
1186 total_size
-= sizeof(struct synth_bird_settings
);
1187 total_size
= vg_align8( total_size
);
1189 if( total_size
> AUDIO_DECODE_SIZE
)
1190 vg_fatal_exit_loop( "Bird coding too long\n" );
1192 struct synth_bird
*bird
= vg_linear_alloc( lin_alloc
, total_size
);
1193 memcpy( &bird
->settings
, clip
->data
, clip
->size
);
1196 clip
->size
= total_size
;
1198 vg_info( "Loaded bird synthesis pattern (%u bytes)\n", total_size
);
1202 vg_fatal_exit_loop( "No path specified, embeded mono unsupported" );
1205 vg_linear_clear( vg_mem
.scratch
);
1208 stb_vorbis_alloc alloc
= {
1209 .alloc_buffer
= vg_linear_alloc( vg_mem
.scratch
, AUDIO_DECODE_SIZE
),
1210 .alloc_buffer_length_in_bytes
= AUDIO_DECODE_SIZE
1213 void *filedata
= vg_file_read( vg_mem
.scratch
, clip
->path
, &fsize
);
1216 stb_vorbis
*decoder
= stb_vorbis_open_memory(
1217 filedata
, fsize
, &err
, &alloc
);
1220 vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
1222 vg_fatal_exit_loop( "Vorbis decode error" );
1225 /* only mono is supported in uncompressed */
1226 u32 length_samples
= stb_vorbis_stream_length_in_samples( decoder
),
1227 data_size
= length_samples
* sizeof(i16
);
1230 clip
->data
= vg_linear_alloc( lin_alloc
, vg_align8(data_size
) );
1231 clip
->size
= length_samples
;
1234 int read_samples
= stb_vorbis_get_samples_i16_downmixed(
1235 decoder
, clip
->data
, length_samples
);
1237 if( read_samples
!= length_samples
)
1238 vg_fatal_exit_loop( "Decode error" );
1240 float mb
= (float)(data_size
) / (1024.0f
*1024.0f
);
1241 vg_info( "Loaded audio clip '%s' (%.1fmb) %u samples\n", clip
->path
, mb
,
1246 VG_STATIC
void audio_clip_loadn( audio_clip
*arr
, int count
, void *lin_alloc
)
1248 for( int i
=0; i
<count
; i
++ )
1249 audio_clip_load( &arr
[i
], lin_alloc
);
1252 VG_STATIC
void audio_require_clip_loaded( audio_clip
*clip
)
1254 if( clip
->data
&& clip
->size
)
1258 vg_fatal_exit_loop( "Must load audio clip before playing! \n" );
1265 VG_STATIC
void audio_debug_ui( m4x4f mtx_pv
)
1267 if( !vg_audio
.debug_ui
)
1272 glBindTexture( GL_TEXTURE_2D
, vg_dsp
.view_texture
);
1273 glTexSubImage2D( GL_TEXTURE_2D
, 0, 0, 0, 256, 256,
1274 GL_RGBA
, GL_UNSIGNED_BYTE
,
1275 vg_dsp
.view_texture_buffer
);
1279 * -----------------------------------------------------------------------
1282 float budget
= ((double)vg_audio
.samples_last
/ 44100.0) * 1000.0;
1283 vg_profile_drawn( (struct vg_profile
*[]){ &vg_prof_audio_decode
,
1285 &vg_prof_audio_dsp
}, 3,
1286 budget
, (ui_rect
){ 4, VG_PROFILE_SAMPLE_COUNT
*2 + 8,
1293 vg_uictx
.cursor
[0] = 512 + 8;
1294 vg_uictx
.cursor
[1] = VG_PROFILE_SAMPLE_COUNT
*2+8+24+12+12;
1295 vg_uictx
.cursor
[2] = 150;
1296 vg_uictx
.cursor
[3] = 12;
1298 ui_rect view_thing
= { 4, vg
.window_y
-512-4, 512, 512 };
1299 ui_push_image( view_thing
, vg_dsp
.view_texture
);
1301 float mb1
= 1024.0f
*1024.0f
,
1302 usage
= vg_linear_get_cur( vg_audio
.audio_pool
) / mb1
,
1303 total
= vg_linear_get_capacity( vg_audio
.audio_pool
) / mb1
,
1304 percent
= (usage
/total
) * 100.0f
;
1306 snprintf( perf
, 127, "Mem: %.1f/%.1fmb (%.1f%%)\n", usage
, total
, percent
);
1308 ui_text( vg_uictx
.cursor
, perf
, 1, 0 );
1309 vg_uictx
.cursor
[1] += 20;
1311 ui_rect overlap_buffer
[ AUDIO_CHANNELS
];
1312 u32 overlap_length
= 0;
1314 /* Draw audio stack */
1315 for( int i
=0; i
<AUDIO_CHANNELS
; i
++ ){
1316 audio_channel
*ch
= &vg_audio
.channels
[i
];
1318 vg_uictx
.cursor
[2] = 400;
1319 vg_uictx
.cursor
[3] = 18;
1323 if( !ch
->allocated
){
1324 ui_fill_rect( vg_uictx
.cursor
, 0x50333333 );
1327 vg_uictx
.cursor
[1] += 1;
1331 const char *formats
[] =
1351 const char *activties
[] =
1360 u32 format_index
= (ch
->source
->flags
& AUDIO_FLAG_FORMAT
)>>9;
1362 snprintf( perf
, 127, "%02d %c%c%cD %s [%s] %4.2fv'%s'",
1364 (ch
->editable_state
.relinquished
)? 'r': '_',
1367 formats
[format_index
],
1368 activties
[ch
->readable_activity
],
1369 ch
->editable_state
.volume
,
1372 ui_fill_rect( vg_uictx
.cursor
, 0xa0000000 | ch
->colour
);
1374 vg_uictx
.cursor
[0] += 2;
1375 vg_uictx
.cursor
[1] += 2;
1376 ui_text( vg_uictx
.cursor
, perf
, 1, 0 );
1379 vg_uictx
.cursor
[1] += 1;
1381 if( AUDIO_FLAG_SPACIAL_3D
){
1383 v3_copy( ch
->editable_state
.spacial_falloff
, wpos
);
1386 m4x4_mulv( mtx_pv
, wpos
, wpos
);
1388 if( wpos
[3] > 0.0f
){
1389 v2_muls( wpos
, (1.0f
/wpos
[3]) * 0.5f
, wpos
);
1390 v2_add( wpos
, (v2f
){ 0.5f
, 0.5f
}, wpos
);
1393 wr
[0] = wpos
[0] * vg
.window_x
;
1394 wr
[1] = (1.0f
-wpos
[1]) * vg
.window_y
;
1398 for( int j
=0; j
<12; j
++ ){
1400 for( int k
=0; k
<overlap_length
; k
++ ){
1401 ui_px
*wk
= overlap_buffer
[k
];
1402 if( ((wr
[0] <= wk
[0]+wk
[2]) && (wr
[0]+wr
[2] >= wk
[0])) &&
1403 ((wr
[1] <= wk
[1]+wk
[3]) && (wr
[1]+wr
[3] >= wk
[1])) )
1416 ui_text( wr
, perf
, 1, 0 );
1418 ui_rect_copy( wr
, overlap_buffer
[ overlap_length
++ ] );
1426 #endif /* VG_AUDIO_H */