-/* Copyright (C) 2021-2022 Harry Godden (hgn) - All Rights Reserved */
+/* Copyright (C) 2021-2023 Harry Godden (hgn) - All Rights Reserved */
#ifndef VG_AUDIO_H
#define VG_AUDIO_H
#include "vg/vg_console.h"
#include "vg/vg_store.h"
#include "vg/vg_profiler.h"
+#include "vg/vg_audio_synth_bird.h"
#include <sys/time.h>
#include <math.h>
#endif
#endif
-#define SFX_MAX_SYSTEMS 32
+#define AUDIO_FRAME_SIZE 512
+#define AUDIO_MIX_FRAME_SIZE 256
+
+#define AUDIO_CHANNELS 32
+#define AUDIO_LFOS 8
+#define AUDIO_FILTERS 16
#define AUDIO_FLAG_LOOP 0x1
-#define AUDIO_FLAG_ONESHOT 0x2
+#define AUDIO_FLAG_NO_DOPPLER 0x2
#define AUDIO_FLAG_SPACIAL_3D 0x4
#define AUDIO_FLAG_AUTO_START 0x8
-#define AUDIO_FLAG_KILL 0x10
-#define FADEOUT_LENGTH 1100
-#define FADEOUT_DIVISOR (1.0f/(float)FADEOUT_LENGTH)
+/* Vorbis will ALWAYS use the maximum amount of channels it can */
+//#define AUDIO_FLAG_MONO 0x100 NOTE: This is the default, so its not used
+//#define AUDIO_FLAG_STEREO 0x200
+//#define AUDIO_FLAG_VORBIS 0x400
+//#define AUDIO_FLAG_BIRD_SYNTH 0x800
-#define AUDIO_DECODE_SIZE (1024*256) /* 256 kb decoding buffers */
+#define AUDIO_FLAG_FORMAT 0x1E00
-enum audio_source_mode
+enum audio_format
{
- k_audio_source_mono,
- k_audio_source_compressed,
+ k_audio_format_mono = 0x000u,
+ k_audio_format_stereo = 0x200u,
+ k_audio_format_vorbis = 0x400u,
+ k_audio_format_none0 = 0x600u,
+ k_audio_format_none1 = 0x800u,
+ k_audio_format_none2 = 0xA00u,
+ k_audio_format_none3 = 0xC00u,
+ k_audio_format_none4 = 0xE00u,
+
+ k_audio_format_bird = 0x1000u,
+ k_audio_format_none5 = 0x1200u,
+ k_audio_format_none6 = 0x1400u,
+ k_audio_format_none7 = 0x1600u,
+ k_audio_format_none8 = 0x1800u,
+ k_audio_format_none9 = 0x1A00u,
+ k_audio_format_none10 = 0x1C00u,
+ k_audio_format_none11 = 0x1E00u,
};
+#define AUDIO_DECODE_SIZE (1024*256) /* 256 kb decoding buffers */
+#define AUDIO_MUTE_VOLUME 0.0f
+#define AUDIO_BASE_VOLUME 1.0f
+
typedef struct audio_clip audio_clip;
-struct audio_clip
-{
- const char *path;
- enum audio_source_mode source_mode;
+typedef struct audio_channel audio_channel;
+typedef struct audio_lfo audio_lfo;
+struct audio_clip{
+ const char *path;
+ u32 flags;
u32 size;
void *data;
};
-typedef struct audio_mix_info audio_mix_info;
-struct audio_mix_info
-{
- audio_clip *source;
- v3f world_position;
-
- float vol, pan;
- u32 flags;
-};
-
-typedef struct audio_player audio_player;
-struct audio_player
-{
- aatree_ptr active_entity; /* non-nil if currently playing */
- audio_mix_info info;
- int enqued, init;
-
- /* Diagnostic */
- const char *name;
-};
-
-typedef struct audio_entity audio_entity;
-struct audio_entity
-{
- audio_player *player;
- audio_mix_info info;
-
- u32 length, cur;
-
- /* Effects */
- u32 fadeout, fadeout_current;
- const char *name;
-};
-
-/*
- * TODO list sunday
- *
- * play again: if already playing, leave in queue while it fadeouts
- * oneshot: create a ghost entity
- *
- */
-
-static struct vg_audio_system
-{
-#if 0
- ma_device miniaudio_device;
- ma_device_config miniaudio_dconfig;
-#endif
+static struct vg_audio_system{
SDL_AudioDeviceID sdl_output_device;
void *audio_pool,
/* synchro */
int sync_locked;
- SDL_mutex *mux_checker,
- *mux_sync;
+ SDL_SpinLock sl_checker,
+ sl_sync;
- /* Audio engine, thread 1 */
- struct active_audio_player
- {
- int active;
- union
- {
- audio_entity ent;
- aatree_pool_node pool_node;
+ struct audio_lfo{
+ u32 time, time_startframe;
+ float sqrt_polynomial_coefficient;
+
+ struct{
+ enum lfo_wave_type{
+ k_lfo_triangle,
+ k_lfo_square,
+ k_lfo_saw,
+ k_lfo_polynomial_bipolar
+ }
+ wave_type;
+
+ u32 period;
+ float polynomial_coefficient;
+ }
+ _, editable_state;
+ u32 editble_state_write_mask;
+ }
+ oscillators[ AUDIO_LFOS ];
+
+ struct audio_channel{
+ int allocated;
+ u32 group;
+
+ char name[32]; /* only editable while allocated == 0 */
+ audio_clip *source; /* ... */
+ u32 flags; /* ... */
+ u32 colour; /* ... */
+
+ /* internal non-readable state
+ * -----------------------------*/
+ u32 cursor, source_length;
+
+ float volume_movement_start,
+ pan_movement_start;
+
+ u32 volume_movement,
+ pan_movement;
+
+ union{
+ struct synth_bird *bird_handle;
+ stb_vorbis *vorbis_handle;
};
-
- stb_vorbis *vorbis_handle;
+
stb_vorbis_alloc vorbis_alloc;
+
+ enum channel_activity{
+ k_channel_activity_reset, /* will advance if allocated==1, to wake */
+ k_channel_activity_wake, /* will advance to either of next two */
+ k_channel_activity_alive,
+ k_channel_activity_end,
+ k_channel_activity_error
+ }
+ activity,
+ readable_activity;
+
+ /*
+ * editable structure, can be modified inside _lock and _unlock
+ * the edit mask tells which to copy into internal _, or to discard
+ * ----------------------------------------------------------------------
+ */
+ struct channel_state{
+ int relinquished;
+
+ float volume, /* current volume */
+ volume_target, /* target volume */
+ pan,
+ pan_target,
+ sampling_rate;
+
+ u32 volume_rate,
+ pan_rate;
+
+ v4f spacial_falloff; /* xyz, range */
+
+ audio_lfo *lfo;
+ float lfo_amount;
+ }
+ _, editable_state;
+ u32 editble_state_write_mask;
}
- active_players[ SFX_MAX_SYSTEMS ];
+ channels[ AUDIO_CHANNELS ];
- aatree active_pool_info; /* note: just using the pool */
- aatree_ptr active_pool_head;
+ int debug_ui, debug_ui_3d, debug_dsp;
- /* System queue, and access from thread 0 */
- audio_entity entity_queue[SFX_MAX_SYSTEMS];
- int queue_len;
- int debug_ui, debug_ui_3d;
+ v3f internal_listener_pos,
+ internal_listener_ears,
+ internal_listener_velocity,
- v3f listener_pos,
- listener_ears;
+ external_listener_pos,
+ external_listener_ears,
+ external_lister_velocity;
- float volume,
- volume_target,
- volume_target_internal,
- volume_console;
+ float internal_global_volume,
+ external_global_volume;
}
-vg_audio = { .volume_console = 1.0f };
+vg_audio = { .external_global_volume = 1.0f };
+
+#include "vg/vg_audio_dsp.h"
static struct vg_profile
_vg_prof_audio_decode = {.mode = k_profile_mode_accum,
.name = "[T2] audio_decode()"},
_vg_prof_audio_mix = {.mode = k_profile_mode_accum,
.name = "[T2] audio_mix()"},
+ _vg_prof_dsp = {.mode = k_profile_mode_accum,
+ .name = "[T2] dsp_process()"},
vg_prof_audio_decode,
- vg_prof_audio_mix;
+ vg_prof_audio_mix,
+ vg_prof_audio_dsp;
/*
* These functions are called from the main thread and used to prevent bad
VG_STATIC int audio_lock_checker_load(void)
{
int value;
- SDL_LockMutex( vg_audio.mux_checker );
+ SDL_AtomicLock( &vg_audio.sl_checker );
value = vg_audio.sync_locked;
- SDL_UnlockMutex( vg_audio.mux_checker );
+ SDL_AtomicUnlock( &vg_audio.sl_checker );
return value;
}
VG_STATIC void audio_lock_checker_store( int value )
{
- SDL_LockMutex( vg_audio.mux_checker );
+ SDL_AtomicLock( &vg_audio.sl_checker );
vg_audio.sync_locked = value;
- SDL_UnlockMutex( vg_audio.mux_checker );
+ SDL_AtomicUnlock( &vg_audio.sl_checker );
}
VG_STATIC void audio_require_lock(void)
VG_STATIC void audio_lock(void)
{
- SDL_LockMutex( vg_audio.mux_sync );
+ SDL_AtomicLock( &vg_audio.sl_sync );
audio_lock_checker_store(1);
}
VG_STATIC void audio_unlock(void)
{
audio_lock_checker_store(0);
- SDL_UnlockMutex( vg_audio.mux_sync );
+ SDL_AtomicUnlock( &vg_audio.sl_sync );
}
VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int frame_count );
VG_STATIC void vg_audio_init(void)
{
- vg_audio.mux_checker = SDL_CreateMutex();
- vg_audio.mux_sync = SDL_CreateMutex();
-
/* TODO: Move here? */
- vg_convar_push( (struct vg_convar){
+ vg_var_push( (struct vg_var){
.name = "debug_audio",
.data = &vg_audio.debug_ui,
- .data_type = k_convar_dtype_i32,
+ .data_type = k_var_dtype_i32,
+ .opt_i32 = { .min=0, .max=1, .clamp=1 },
+ .persistent = 1
+ });
+
+ vg_var_push( (struct vg_var){
+ .name = "debug_dsp",
+ .data = &vg_audio.debug_dsp,
+ .data_type = k_var_dtype_i32,
.opt_i32 = { .min=0, .max=1, .clamp=1 },
.persistent = 1
});
- vg_convar_push( (struct vg_convar){
+ vg_var_push( (struct vg_var){
.name = "volume",
- .data = &vg_audio.volume_console,
- .data_type = k_convar_dtype_f32,
+ .data = &vg_audio.external_global_volume,
+ .data_type = k_var_dtype_f32,
.opt_f32 = { .min=0.0f, .max=2.0f, .clamp=1 },
.persistent = 1
});
VG_MEMORY_SYSTEM );
/* fixed */
- u32 decode_size = AUDIO_DECODE_SIZE * SFX_MAX_SYSTEMS;
+ u32 decode_size = AUDIO_DECODE_SIZE * AUDIO_CHANNELS;
vg_audio.decode_buffer = vg_linear_alloc( vg_mem.rtmemory, decode_size );
- /* setup pool */
- vg_audio.active_pool_info.base = vg_audio.active_players;
- vg_audio.active_pool_info.offset = offsetof(struct active_audio_player,
- pool_node );
- vg_audio.active_pool_info.stride = sizeof(struct active_audio_player);
- vg_audio.active_pool_info.p_cmp = NULL;
- aatree_init_pool( &vg_audio.active_pool_info, SFX_MAX_SYSTEMS );
+ vg_dsp_init();
SDL_AudioSpec spec_desired, spec_got;
spec_desired.callback = audio_mixer_callback;
spec_desired.format = AUDIO_F32;
spec_desired.freq = 44100;
spec_desired.padding = 0;
- spec_desired.samples = 512;
+ spec_desired.samples = AUDIO_FRAME_SIZE;
spec_desired.silence = 0;
spec_desired.size = 0;
spec_desired.userdata = NULL;
vg_audio.sdl_output_device =
- SDL_OpenAudioDevice( NULL, 0, &spec_desired, &spec_got,
- SDL_AUDIO_ALLOW_SAMPLES_CHANGE );
+ SDL_OpenAudioDevice( NULL, 0, &spec_desired, &spec_got,0 );
- if( vg_audio.sdl_output_device )
- {
+ if( vg_audio.sdl_output_device ){
SDL_PauseAudioDevice( vg_audio.sdl_output_device, 0 );
}
- else
- {
+ else{
vg_fatal_exit_loop(
"SDL_OpenAudioDevice failed. Your default audio device must support:\n"
" Frequency: 44100 hz\n"
VG_STATIC void vg_audio_free(void)
{
+ vg_dsp_free();
SDL_CloseAudioDevice( vg_audio.sdl_output_device );
}
* thread 1
*/
-static aatree_ptr audio_alloc_entity_internal(void)
+#define AUDIO_EDIT_VOLUME_SLOPE 0x1
+#define AUDIO_EDIT_VOLUME 0x2
+#define AUDIO_EDIT_LFO_PERIOD 0x4
+#define AUDIO_EDIT_LFO_WAVE 0x8
+#define AUDIO_EDIT_LFO_ATTACHMENT 0x10
+#define AUDIO_EDIT_SPACIAL 0x20
+#define AUDIO_EDIT_OWNERSHIP 0x40
+#define AUDIO_EDIT_SAMPLING_RATE 0x80
+
+static void audio_channel_init( audio_channel *ch, audio_clip *clip, u32 flags )
+{
+ ch->group = 0;
+ ch->source = clip;
+ ch->flags = flags;
+ ch->colour = 0x00333333;
+
+ if( (ch->source->flags & AUDIO_FLAG_FORMAT) == k_audio_format_bird )
+ strcpy( ch->name, "[array]" );
+ else
+ strncpy( ch->name, clip->path, 31 );
+
+ ch->allocated = 1;
+
+ ch->editable_state.relinquished = 0;
+ ch->editable_state.volume = 1.0f;
+ ch->editable_state.volume_target = 1.0f;
+ ch->editable_state.pan = 0.0f;
+ ch->editable_state.pan_target = 0.0f;
+ ch->editable_state.volume_rate = 0;
+ ch->editable_state.pan_rate = 0;
+ v4_copy((v4f){0.0f,0.0f,0.0f,1.0f},ch->editable_state.spacial_falloff);
+ ch->editable_state.lfo = NULL;
+ ch->editable_state.lfo_amount = 0.0f;
+ ch->editable_state.sampling_rate = 1.0f;
+ ch->editble_state_write_mask = 0x00;
+}
+
+static void audio_channel_group( audio_channel *ch, u32 group )
{
- aatree_ptr playerid = aatree_pool_alloc( &vg_audio.active_pool_info,
- &vg_audio.active_pool_head );
+ ch->group = group;
+ ch->colour = ((group * 29986577) & 0x00ffffff) | 0xff000000;
+}
- if( playerid == AATREE_PTR_NIL )
- return AATREE_PTR_NIL;
+static audio_channel *audio_get_first_idle_channel(void)
+{
+ for( int i=0; i<AUDIO_CHANNELS; i++ ){
+ audio_channel *ch = &vg_audio.channels[i];
- struct active_audio_player *aap = &vg_audio.active_players[ playerid ];
- aap->active = 1;
+ if( !ch->allocated ){
+ return ch;
+ }
+ }
- return playerid;
+ return NULL;
}
-VG_STATIC void audio_entity_free_internal( aatree_ptr id )
+static audio_channel *audio_get_group_idle_channel( u32 group, u32 max_count )
{
- struct active_audio_player *aap = &vg_audio.active_players[ id ];
- aap->active = 0;
+ u32 count = 0;
+ audio_channel *dest = NULL;
- /* Notify player that we've finished */
- if( aap->ent.player )
- aap->ent.player->active_entity = AATREE_PTR_NIL;
+ for( int i=0; i<AUDIO_CHANNELS; i++ ){
+ audio_channel *ch = &vg_audio.channels[i];
- /* delete */
- aatree_pool_free( &vg_audio.active_pool_info, id,
- &vg_audio.active_pool_head );
+ if( ch->allocated ){
+ if( ch->group == group ){
+ count ++;
+ }
+ }
+ else{
+ if( !dest )
+ dest = ch;
+ }
+ }
+
+ if( dest && (count < max_count) ){
+ return dest;
+ }
+
+ return NULL;
}
-VG_STATIC void *audio_entity_vorbis_ptr( aatree_ptr entid )
+static audio_channel *audio_get_group_first_active_channel( u32 group )
{
- u8 *buf = (u8*)vg_audio.decode_buffer,
- *loc = &buf[AUDIO_DECODE_SIZE*entid];
+ for( int i=0; i<AUDIO_CHANNELS; i++ ){
+ audio_channel *ch = &vg_audio.channels[i];
+ if( ch->allocated && (ch->group == group) )
+ return ch;
+ }
+ return NULL;
+}
- return (void *)loc;
+static int audio_channel_finished( audio_channel *ch )
+{
+ if( ch->readable_activity == k_channel_activity_end )
+ return 1;
+ else
+ return 0;
}
-VG_STATIC void audio_entity_start( audio_entity *src )
+static audio_channel *audio_relinquish_channel( audio_channel *ch )
{
- aatree_ptr entid = audio_alloc_entity_internal();
- if( entid == AATREE_PTR_NIL )
- return;
+ ch->editable_state.relinquished = 1;
+ ch->editble_state_write_mask |= AUDIO_EDIT_OWNERSHIP;
+ return NULL;
+}
+
+static void audio_channel_slope_volume( audio_channel *ch, float length,
+ float new_volume )
+{
+ ch->editable_state.volume_target = new_volume;
+ ch->editable_state.volume_rate = length * 44100.0f;
+ ch->editble_state_write_mask |= AUDIO_EDIT_VOLUME_SLOPE;
+}
+
+static void audio_channel_set_sampling_rate( audio_channel *ch, float rate )
+{
+ ch->editable_state.sampling_rate = rate;
+ ch->editble_state_write_mask |= AUDIO_EDIT_SAMPLING_RATE;
+}
+
+static void audio_channel_edit_volume( audio_channel *ch,
+ float new_volume, int instant )
+{
+ if( instant ){
+ ch->editable_state.volume = new_volume;
+ ch->editble_state_write_mask |= AUDIO_EDIT_VOLUME;
+ }
+ else{
+ audio_channel_slope_volume( ch, 0.05f, new_volume );
+ }
+}
+
+static audio_channel *audio_channel_fadeout( audio_channel *ch, float length )
+{
+ audio_channel_slope_volume( ch, length, 0.0f );
+ return audio_relinquish_channel( ch );
+}
- audio_entity *ent = &vg_audio.active_players[ entid ].ent;
+static void audio_channel_fadein( audio_channel *ch, float length )
+{
+ audio_channel_edit_volume( ch, 0.0f, 1 );
+ audio_channel_slope_volume( ch, length, 1.0f );
+}
+
+static audio_channel *audio_channel_crossfade( audio_channel *ch,
+ audio_clip *new_clip,
+ float length, u32 flags )
+{
+ u32 cursor = 0;
- ent->info = src->info;
- ent->name = src->info.source->path;
- ent->cur = 0;
- ent->player = src->player;
+ if( ch )
+ ch = audio_channel_fadeout( ch, length );
- ent->fadeout = 0;
- ent->fadeout_current = 0;
+ audio_channel *replacement = audio_get_first_idle_channel();
- /* Notify main player we are dequeud and playing */
- if( src->player )
- {
- src->player->enqued = 0;
- src->player->active_entity = entid;
+ if( replacement ){
+ audio_channel_init( replacement, new_clip, flags );
+ audio_channel_fadein( replacement, length );
}
- if( src->info.source->source_mode == k_audio_source_compressed )
- {
- /* Setup vorbis decoder */
- struct active_audio_player *aap = &vg_audio.active_players[ entid ];
-
- stb_vorbis_alloc alloc = {
- .alloc_buffer = (char *)audio_entity_vorbis_ptr( entid ),
- .alloc_buffer_length_in_bytes = AUDIO_DECODE_SIZE
- };
+ return replacement;
+}
- int err;
- stb_vorbis *decoder = stb_vorbis_open_memory(
- src->info.source->data,
- src->info.source->size, &err, &alloc );
+static void audio_channel_sidechain_lfo( audio_channel *ch, int lfo_id,
+ float amount )
+{
+ ch->editable_state.lfo = &vg_audio.oscillators[ lfo_id ];
+ ch->editable_state.lfo_amount = amount;
+ ch->editble_state_write_mask |= AUDIO_EDIT_LFO_ATTACHMENT;
+}
- if( !decoder )
- {
- vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
- src->info.source->path, err );
+static void audio_channel_set_spacial( audio_channel *ch, v3f co, float range )
+{
+ if( ch->flags & AUDIO_FLAG_SPACIAL_3D ){
+ v3_copy( co, ch->editable_state.spacial_falloff );
- audio_entity_free_internal( entid );
- return;
- }
+ if( range == 0.0f )
+ ch->editable_state.spacial_falloff[3] = 1.0f;
else
- {
- ent->length = stb_vorbis_stream_length_in_samples( decoder );
- }
-
- aap->vorbis_handle = decoder;
+ ch->editable_state.spacial_falloff[3] = 1.0f/range;
+
+ ch->editble_state_write_mask |= AUDIO_EDIT_SPACIAL;
}
- else
- {
- ent->length = src->info.source->size;
+ else{
+ vg_warn( "Tried to set spacialization paramaters for 2D channel (%s)\n",
+ ch->name );
}
}
-/*
- * Read everything from the queue
- */
-VG_STATIC void audio_system_enque(void)
+static int audio_oneshot_3d( audio_clip *clip, v3f position,
+ float range, float volume )
{
- /* Process incoming sound queue */
- audio_lock();
+ audio_channel *ch = audio_get_first_idle_channel();
- vg_audio.volume_target_internal = vg_audio.volume_target;
-
- int wr = 0;
- for( int i=0; i<vg_audio.queue_len; i++ )
- {
- audio_entity *src = &vg_audio.entity_queue[ i ];
-
- if( src->player )
- {
- /* Start new */
- if( src->player->active_entity == AATREE_PTR_NIL )
- {
- audio_entity_start( src );
- }
- else
- {
- /* Otherwise try start fadeout but dont remove from queue */
-
- aatree_ptr entid = src->player->active_entity;
- audio_entity *ent = &vg_audio.active_players[ entid ].ent;
- if( !ent->fadeout )
- {
- ent->fadeout = FADEOUT_LENGTH;
- ent->fadeout_current = FADEOUT_LENGTH;
- }
+ if( ch ){
+ audio_channel_init( ch, clip, AUDIO_FLAG_SPACIAL_3D );
+ audio_channel_set_spacial( ch, position, range );
+ audio_channel_edit_volume( ch, volume, 1 );
+ ch = audio_relinquish_channel( ch );
- vg_audio.entity_queue[ wr ++ ] = *src;
- }
- }
- else
- {
- audio_entity_start( src );
- }
+ return 1;
}
-
- vg_audio.queue_len = wr;
-
- /* Localize others memory */
- for( int i=0; i<SFX_MAX_SYSTEMS; i ++ )
- {
- struct active_audio_player *aap = &vg_audio.active_players[i];
- if( !aap->active )
- continue;
+ else
+ return 0;
+}
- if( aap->ent.player )
- {
- /* Only copy information in whilst not requeing */
- if( aap->ent.player->enqued == 0 )
- {
- aap->ent.info = aap->ent.player->info;
+static int audio_oneshot( audio_clip *clip, float volume, float pan )
+{
+ audio_channel *ch = audio_get_first_idle_channel();
- if( (aap->ent.info.flags & AUDIO_FLAG_KILL) && !aap->ent.fadeout )
- {
- aap->ent.fadeout = FADEOUT_LENGTH;
- aap->ent.fadeout_current = FADEOUT_LENGTH;
- }
- }
- }
+ if( ch ){
+ audio_channel_init( ch, clip, 0x00 );
+ audio_channel_edit_volume( ch, volume, 1 );
+ ch = audio_relinquish_channel( ch );
+
+ return 1;
}
-
- audio_unlock();
+ else
+ return 0;
}
-/*
- * Redistribute sound systems
- */
-VG_STATIC void audio_system_cleanup(void)
+static void audio_set_lfo_wave( int id, enum lfo_wave_type type,
+ float coefficient )
{
- audio_lock();
+ audio_lfo *lfo = &vg_audio.oscillators[ id ];
+ lfo->editable_state.polynomial_coefficient = coefficient;
+ lfo->editable_state.wave_type = type;
- for( int i=0; i<SFX_MAX_SYSTEMS; i ++ )
- {
- struct active_audio_player *aap = &vg_audio.active_players[i];
- if( aap->active )
- {
- audio_entity *src = &aap->ent;
- if( src->cur < src->length || (src->info.flags & AUDIO_FLAG_LOOP ))
- {
- /* Good to keep */
- }
- else
- {
- audio_entity_free_internal( i );
- }
- }
- }
+ lfo->editble_state_write_mask |= AUDIO_EDIT_LFO_WAVE;
+}
- audio_unlock();
+static void audio_set_lfo_frequency( int id, float freq )
+{
+ audio_lfo *lfo = &vg_audio.oscillators[ id ];
+ lfo->editable_state.period = 44100.0f / freq;
+ lfo->editble_state_write_mask |= AUDIO_EDIT_LFO_PERIOD;
}
+
/*
- * Get effective volume and pan from this entity
+ * Committers
+ * -----------------------------------------------------------------------------
*/
-VG_STATIC void audio_entity_spacialize( audio_entity *ent,
- float *vol, float *pan )
+static int audio_channel_load_source( audio_channel *ch )
{
- if( ent->info.vol < 0.01f )
- {
- *vol = ent->info.vol;
- *pan = 0.0f;
- return;
- }
+ u32 format = ch->source->flags & AUDIO_FLAG_FORMAT;
- if( !vg_validf(vg_audio.listener_pos[0]) ||
- !vg_validf(vg_audio.listener_pos[1]) ||
- !vg_validf(vg_audio.listener_pos[2]) ||
- !vg_validf(ent->info.world_position[0]) ||
- !vg_validf(ent->info.world_position[1]) ||
- !vg_validf(ent->info.world_position[2]) )
- {
- vg_error( "NaN listener/world position (%s)\n", ent->name );
- *vol = 0.0f;
- *pan = 0.0f;
- return;
- }
+ if( format == k_audio_format_vorbis ){
+ /* Setup vorbis decoder */
+ u32 index = ch - vg_audio.channels;
- v3f delta;
- v3_sub( ent->info.world_position, vg_audio.listener_pos, delta );
+ u8 *buf = (u8*)vg_audio.decode_buffer,
+ *loc = &buf[AUDIO_DECODE_SIZE*index];
- float dist2 = v3_length2( delta );
+ stb_vorbis_alloc alloc = {
+ .alloc_buffer = (char *)loc,
+ .alloc_buffer_length_in_bytes = AUDIO_DECODE_SIZE
+ };
+
+ int err;
+ stb_vorbis *decoder = stb_vorbis_open_memory(
+ ch->source->data,
+ ch->source->size, &err, &alloc );
- if( dist2 < 0.0001f )
- {
- *pan = 0.0f;
- *vol = 1.0f;
+ if( !decoder ){
+ vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
+ ch->source->path, err );
+ return 0;
+ }
+ else{
+ ch->source_length = stb_vorbis_stream_length_in_samples( decoder );
+ ch->vorbis_handle = decoder;
+ }
}
- else
- {
- float dist = sqrtf( dist2 ),
- attn = (dist / ent->info.vol) +1.0f;
+ else if( format == k_audio_format_bird ){
+ u32 index = ch - vg_audio.channels;
- v3_muls( delta, 1.0f/dist, delta );
- *pan = v3_dot( vg_audio.listener_ears, delta );
- *vol = 1.0f/(attn*attn);
+ u8 *buf = (u8*)vg_audio.decode_buffer;
+ struct synth_bird *loc = (void *)&buf[AUDIO_DECODE_SIZE*index];
+
+ memcpy( loc, ch->source->data, ch->source->size );
+ synth_bird_reset( loc );
+
+ ch->bird_handle = loc;
+ ch->source_length = synth_bird_get_length_in_samples( loc );
+ }
+ else if( format == k_audio_format_stereo ){
+ ch->source_length = ch->source->size / 2;
+ }
+ else{
+ ch->source_length = ch->source->size;
}
+
+ return 1;
}
VG_STATIC void audio_decode_uncompressed_mono( i16 *src, u32 count, float *dst )
{
- for( u32 i=0; i<count; i++ )
- {
+ for( u32 i=0; i<count; i++ ){
dst[ i*2 + 0 ] = ((float)src[i]) * (1.0f/32767.0f);
dst[ i*2 + 1 ] = ((float)src[i]) * (1.0f/32767.0f);
}
int n = 0,
c = VG_MIN( 1, f->channels - 1 );
- while( n < len )
- {
+ while( n < len ) {
int k = f->channel_buffer_end - f->channel_buffer_start;
if( n+k >= len )
k = len - n;
- for( int j=0; j < k; ++j )
- {
+ for( int j=0; j < k; ++j ) {
*buffer++ = f->channel_buffers[ 0 ][f->channel_buffer_start+j];
*buffer++ = f->channel_buffers[ c ][f->channel_buffer_start+j];
}
int n = 0,
c = VG_MIN( 1, f->channels - 1 );
- while( n < len )
- {
+ while( n < len ) {
int k = f->channel_buffer_end - f->channel_buffer_start;
if( n+k >= len )
k = len - n;
- for( int j=0; j < k; ++j )
- {
+ for( int j=0; j < k; ++j ) {
float sl = f->channel_buffers[ 0 ][f->channel_buffer_start+j],
sr = f->channel_buffers[ c ][f->channel_buffer_start+j];
return n;
}
-VG_STATIC void audio_entity_get_samples( aatree_ptr id, u32 count, float *buf )
+static inline float audio_lfo_pull_sample( audio_lfo *lfo )
{
- vg_profile_begin( &_vg_prof_audio_decode );
+ lfo->time ++;
+
+ if( lfo->time >= lfo->_.period )
+ lfo->time = 0;
+
+ float t = lfo->time;
+ t /= (float)lfo->_.period;
+
+ if( lfo->_.wave_type == k_lfo_polynomial_bipolar ){
+ /*
+ * #
+ * # #
+ * # #
+ * # #
+ * ### # ###
+ * ## #
+ * # #
+ * # #
+ * ##
+ */
+
+ t *= 2.0f;
+ t -= 1.0f;
+
+ return (( 2.0f * lfo->sqrt_polynomial_coefficient * t ) /
+ /* --------------------------------------- */
+ ( 1.0f + lfo->_.polynomial_coefficient * t*t )
+
+ ) * (1.0f-fabsf(t));
+ }
+ else{
+ return 0.0f;
+ }
+}
- struct active_audio_player *aap = &vg_audio.active_players[id];
- audio_entity *ent = &aap->ent;
+static void audio_channel_get_samples( audio_channel *ch,
+ u32 count, float *buf )
+{
+ vg_profile_begin( &_vg_prof_audio_decode );
u32 remaining = count;
- u32 cursor = ent->cur;
u32 buffer_pos = 0;
- while( remaining )
- {
- u32 samples_this_run = VG_MIN( remaining, ent->length - cursor );
+ u32 format = ch->source->flags & AUDIO_FLAG_FORMAT;
+
+ while( remaining ){
+ u32 samples_this_run = VG_MIN(remaining, ch->source_length - ch->cursor);
remaining -= samples_this_run;
float *dst = &buf[ buffer_pos * 2 ];
-
- int source_mode = ent->info.source->source_mode;
- if( source_mode == k_audio_source_mono )
- {
- i16 *src_buffer = ent->info.source->data,
- *src = &src_buffer[cursor];
-
- audio_decode_uncompressed_mono( src, samples_this_run, dst );
+ if( format == k_audio_format_stereo ){
+ for( int i=0;i<samples_this_run; i++ ){
+ dst[i*2+0] = 0.0f;
+ dst[i*2+1] = 0.0f;
+ }
}
- else if( source_mode == k_audio_source_compressed )
- {
+ else if( format == k_audio_format_vorbis ){
int read_samples = stb_vorbis_get_samples_float_interleaved_stereo(
- aap->vorbis_handle,
+ ch->vorbis_handle,
dst,
- samples_this_run );
+ samples_this_run );
- if( read_samples != samples_this_run )
- {
- vg_warn( "Invalid samples read (%s)\n", ent->info.source->path );
+ if( read_samples != samples_this_run ){
+ vg_warn( "Invalid samples read (%s)\n", ch->source->path );
+
+ for( int i=0; i<samples_this_run; i++ ){
+ dst[i*2+0] = 0.0f;
+ dst[i*2+1] = 0.0f;
+ }
}
}
+ else if( format == k_audio_format_bird ){
+ synth_bird_generate_samples( ch->bird_handle, dst, samples_this_run );
+ }
+ else{
+ i16 *src_buffer = ch->source->data,
+ *src = &src_buffer[ch->cursor];
+
+ audio_decode_uncompressed_mono( src, samples_this_run, dst );
+ }
- cursor += samples_this_run;
+ ch->cursor += samples_this_run;
buffer_pos += samples_this_run;
- if( (ent->info.flags & AUDIO_FLAG_LOOP) && remaining )
- {
- if( source_mode == k_audio_source_compressed )
- {
- stb_vorbis_seek_start( aap->vorbis_handle );
- }
+ if( (ch->flags & AUDIO_FLAG_LOOP) && remaining ){
+ if( format == k_audio_format_vorbis )
+ stb_vorbis_seek_start( ch->vorbis_handle );
+ else if( format == k_audio_format_bird )
+ synth_bird_reset( ch->bird_handle );
- cursor = 0;
+ ch->cursor = 0;
continue;
}
else
break;
}
- while( remaining )
- {
+ while( remaining ){
buf[ buffer_pos*2 + 0 ] = 0.0f;
buf[ buffer_pos*2 + 1 ] = 0.0f;
buffer_pos ++;
remaining --;
}
- ent->cur = cursor;
vg_profile_end( &_vg_prof_audio_decode );
}
-VG_STATIC void audio_entity_mix( aatree_ptr id, float *buffer,
- u32 frame_count )
+static void audio_channel_mix( audio_channel *ch, float *buffer )
{
- audio_entity *ent = &vg_audio.active_players[id].ent;
+ float framevol_l = vg_audio.internal_global_volume,
+ framevol_r = vg_audio.internal_global_volume;
- u32 cursor = ent->cur, buffer_pos = 0;
- float *pcf = alloca( frame_count * 2 * sizeof(float) );
-
- u32 frames_write = frame_count;
- float fadeout_divisor = 1.0f / (float)ent->fadeout;
+ float frame_samplerate = ch->_.sampling_rate;
- float vol = ent->info.vol,
- pan = ent->info.pan;
+ if( ch->flags & AUDIO_FLAG_SPACIAL_3D ){
+ v3f delta;
+ v3_sub( ch->_.spacial_falloff, vg_audio.internal_listener_pos, delta );
- audio_entity_get_samples( id, frame_count, pcf );
+ float dist = v3_length( delta ),
+ vol = vg_maxf( 0.0f, 1.0f - ch->_.spacial_falloff[3]*dist );
+
+ if( dist <= 0.01f ){
+
+ }
+ else{
+ v3_muls( delta, 1.0f/dist, delta );
+ float pan = v3_dot( vg_audio.internal_listener_ears, delta );
+ vol = powf( vol, 5.0f );
+
+ framevol_l *= (vol * 0.5f) * (1.0f - pan);
+ framevol_r *= (vol * 0.5f) * (1.0f + pan);
+
+ if( !(ch->source->flags & AUDIO_FLAG_NO_DOPPLER) ){
+ const float vs = 323.0f;
+
+ float dv = v3_dot(delta,vg_audio.internal_listener_velocity);
+ float doppler = (vs+dv)/vs;
+ doppler = vg_clampf( doppler, 0.6f, 1.4f );
+
+ if( fabsf(doppler-1.0f) > 0.01f )
+ frame_samplerate *= doppler;
+ }
+ }
+
+ if( !vg_validf( framevol_l ) ) vg_fatal_exit_loop( "NaN left channel" );
+ if( !vg_validf( framevol_r ) ) vg_fatal_exit_loop( "NaN right channel" );
+ if( !vg_validf( frame_samplerate ) )
+ vg_fatal_exit_loop( "NaN sample rate" );
+ }
+
+ u32 buffer_length = AUDIO_MIX_FRAME_SIZE;
+ if( frame_samplerate != 1.0f ){
+ float l = ceilf( (float)(AUDIO_MIX_FRAME_SIZE) * frame_samplerate );
+ buffer_length = l+1;
+ }
+
+ float pcf[ AUDIO_MIX_FRAME_SIZE * 2 * 2 ];
+
+ audio_channel_get_samples( ch, buffer_length, pcf );
vg_profile_begin( &_vg_prof_audio_mix );
- if( ent->info.flags & AUDIO_FLAG_SPACIAL_3D )
- audio_entity_spacialize( ent, &vol, &pan );
+ float volume_movement = ch->volume_movement;
+ float const fvolume_rate = vg_maxf( 1.0f, ch->_.volume_rate );
+ const float inv_volume_rate = 1.0f/fvolume_rate;
- for( u32 j=0; j<frame_count; j++ )
- {
- float frame_vol = vol * vg_audio.volume;
- if( ent->fadeout )
- {
- /* Force this system to be removed now */
- if( ent->fadeout_current == 0 )
- {
- ent->info.flags = 0x00;
- ent->cur = ent->length;
- break;
- }
+ float volume = ch->_.volume;
+ const float volume_start = ch->volume_movement_start;
+ const float volume_target = ch->_.volume_target;
- frame_vol *= (float)ent->fadeout_current * fadeout_divisor;
- ent->fadeout_current --;
- }
+ for( u32 j=0; j<AUDIO_MIX_FRAME_SIZE; j++ ){
+ volume_movement += 1.0f;
+ float movement_t = volume_movement * inv_volume_rate;
+ movement_t = vg_minf( movement_t, 1.0f );
+ volume = vg_lerpf( volume_start, volume_target, movement_t );
+
+ float vol_norm = volume * volume;
- float sl = 1.0f-pan,
- sr = 1.0f+pan;
+ if( ch->_.lfo )
+ vol_norm *= 1.0f + audio_lfo_pull_sample(ch->_.lfo) * ch->_.lfo_amount;
- buffer[ buffer_pos*2+0 ] += pcf[ buffer_pos*2+0 ] * frame_vol * sl;
- buffer[ buffer_pos*2+1 ] += pcf[ buffer_pos*2+1 ] * frame_vol * sr;
+ float vol_l = vol_norm * framevol_l,
+ vol_r = vol_norm * framevol_r,
+ sample_l,
+ sample_r;
- buffer_pos ++;
+ if( frame_samplerate != 1.0f ){
+ /* absolutely garbage resampling, but it will do
+ */
+
+ float sample_index = frame_samplerate * (float)j;
+ float t = vg_fractf( sample_index );
+
+ u32 i0 = floorf( sample_index ),
+ i1 = i0+1;
+
+ sample_l = pcf[ i0*2+0 ]*(1.0f-t) + pcf[ i1*2+0 ]*t;
+ sample_r = pcf[ i0*2+1 ]*(1.0f-t) + pcf[ i1*2+1 ]*t;
+ }
+ else{
+ sample_l = pcf[ j*2+0 ];
+ sample_r = pcf[ j*2+1 ];
+ }
+
+ buffer[ j*2+0 ] += sample_l * vol_l;
+ buffer[ j*2+1 ] += sample_r * vol_r;
}
+ ch->volume_movement += AUDIO_MIX_FRAME_SIZE;
+ ch->volume_movement = VG_MIN( ch->volume_movement, ch->_.volume_rate );
+ ch->_.volume = volume;
+
vg_profile_end( &_vg_prof_audio_mix );
}
VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count )
{
- audio_system_enque();
+ /*
+ * Copy data and move edit flags to commit flags
+ * ------------------------------------------------------------- */
+ audio_lock();
+
+ v3_copy( vg_audio.external_listener_pos, vg_audio.internal_listener_pos );
+ v3_copy( vg_audio.external_listener_ears, vg_audio.internal_listener_ears );
+ v3_copy( vg_audio.external_lister_velocity,
+ vg_audio.internal_listener_velocity );
+ vg_audio.internal_global_volume = vg_audio.external_global_volume;
+
+ for( int i=0; i<AUDIO_CHANNELS; i++ ){
+ audio_channel *ch = &vg_audio.channels[i];
+
+ if( !ch->allocated )
+ continue;
+
+ if( ch->activity == k_channel_activity_alive ){
+ if( (ch->cursor >= ch->source_length) &&
+ !(ch->flags & AUDIO_FLAG_LOOP) )
+ {
+ ch->activity = k_channel_activity_end;
+ }
+ }
+
+ /* process relinquishments */
+ if( (ch->activity != k_channel_activity_reset) && ch->_.relinquished ){
+ if( (ch->activity == k_channel_activity_end)
+ || (ch->_.volume == 0.0f)
+ || (ch->activity == k_channel_activity_error) )
+ {
+ ch->_.relinquished = 0;
+ ch->allocated = 0;
+ ch->activity = k_channel_activity_reset;
+ continue;
+ }
+ }
+
+ /* process new channels */
+ if( ch->activity == k_channel_activity_reset ){
+ ch->_ = ch->editable_state;
+ ch->cursor = 0;
+ ch->source_length = 0;
+ ch->activity = k_channel_activity_wake;
+ }
+
+ if( ch->editble_state_write_mask & AUDIO_EDIT_OWNERSHIP )
+ ch->_.relinquished = ch->editable_state.relinquished;
+ else
+ ch->editable_state.relinquished = ch->_.relinquished;
+
+
+ if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME ){
+ ch->_.volume = ch->editable_state.volume;
+ ch->_.volume_target = ch->editable_state.volume;
+ }
+ else{
+ ch->editable_state.volume = ch->_.volume;
+ }
+
+
+ if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME_SLOPE ){
+ ch->volume_movement_start = ch->_.volume;
+ ch->volume_movement = 0;
+
+ ch->_.volume_target = ch->editable_state.volume_target;
+ ch->_.volume_rate = ch->editable_state.volume_rate;
+ }
+ else{
+ ch->editable_state.volume_target = ch->_.volume_target;
+ ch->editable_state.volume_rate = ch->_.volume_rate;
+ }
+
+
+ if( ch->editble_state_write_mask & AUDIO_EDIT_SAMPLING_RATE )
+ ch->_.sampling_rate = ch->editable_state.sampling_rate;
+ else
+ ch->editable_state.sampling_rate = ch->_.sampling_rate;
+
+
+ if( ch->editble_state_write_mask & AUDIO_EDIT_LFO_ATTACHMENT ){
+ ch->_.lfo = ch->editable_state.lfo;
+ ch->_.lfo_amount = ch->editable_state.lfo_amount;
+ }
+ else{
+ ch->editable_state.lfo = ch->_.lfo;
+ ch->editable_state.lfo_amount = ch->_.lfo_amount;
+ }
+
+
+ if( ch->editble_state_write_mask & AUDIO_EDIT_SPACIAL )
+ v4_copy( ch->editable_state.spacial_falloff,ch->_.spacial_falloff );
+ else
+ v4_copy( ch->_.spacial_falloff,ch->editable_state.spacial_falloff );
+
+
+ /* currently readonly, i guess */
+ ch->editable_state.pan_target = ch->_.pan_target;
+ ch->editable_state.pan = ch->_.pan;
+ ch->editble_state_write_mask = 0x00;
+ }
+
+ for( int i=0; i<AUDIO_LFOS; i++ ){
+ audio_lfo *lfo = &vg_audio.oscillators[ i ];
+
+ if( lfo->editble_state_write_mask & AUDIO_EDIT_LFO_WAVE ){
+ lfo->_.wave_type = lfo->editable_state.wave_type;
+
+ if( lfo->_.wave_type == k_lfo_polynomial_bipolar ){
+ lfo->_.polynomial_coefficient =
+ lfo->editable_state.polynomial_coefficient;
+ lfo->sqrt_polynomial_coefficient =
+ sqrtf(lfo->_.polynomial_coefficient);
+ }
+ }
+
+ if( lfo->editble_state_write_mask & AUDIO_EDIT_LFO_PERIOD ){
+ if( lfo->_.period ){
+ float t = lfo->time;
+ t/= (float)lfo->_.period;
+
+ lfo->_.period = lfo->editable_state.period;
+ lfo->time = lfo->_.period * t;
+ }
+ else{
+ lfo->time = 0;
+ lfo->_.period = lfo->editable_state.period;
+ }
+ }
+
+ lfo->editble_state_write_mask = 0x00;
+ }
+
+ dsp_update_tunings();
+ audio_unlock();
+
+ /*
+ * Process spawns
+ * ------------------------------------------------------------- */
+ for( int i=0; i<AUDIO_CHANNELS; i++ ){
+ audio_channel *ch = &vg_audio.channels[i];
+
+ if( ch->activity == k_channel_activity_wake ){
+ if( audio_channel_load_source( ch ) )
+ ch->activity = k_channel_activity_alive;
+ else
+ ch->activity = k_channel_activity_error;
+ }
+ }
+ /*
+ * Mix everything
+ * -------------------------------------------------------- */
int frame_count = byte_count/(2*sizeof(float));
/* Clear buffer */
for( int i=0; i<frame_count*2; i ++ )
pOut32F[i] = 0.0f;
- float start_vol = vg_audio.volume;
+ for( int i=0; i<AUDIO_LFOS; i++ ){
+ audio_lfo *lfo = &vg_audio.oscillators[i];
+ lfo->time_startframe = lfo->time;
+ }
+
+ for( int i=0; i<AUDIO_CHANNELS; i ++ ){
+ audio_channel *ch = &vg_audio.channels[i];
+
+ if( ch->activity == k_channel_activity_alive ){
+ if( ch->_.lfo )
+ ch->_.lfo->time = ch->_.lfo->time_startframe;
- /* Mix all sounds */
- for( int i=0; i<SFX_MAX_SYSTEMS; i ++ )
- {
- struct active_audio_player *aap = &vg_audio.active_players[i];
- vg_audio.volume = start_vol;
+ u32 remaining = frame_count,
+ subpos = 0;
- if( aap->active )
- audio_entity_mix( i, pOut32F, frame_count );
+ while( remaining ){
+ audio_channel_mix( ch, pOut32F+subpos );
+ remaining -= AUDIO_MIX_FRAME_SIZE;
+ subpos += AUDIO_MIX_FRAME_SIZE*2;
+ }
+ }
}
- float vol_diff = vg_audio.volume_target_internal - vg_audio.volume,
- vol_rate = 1.0f / (44100.0f*0.25f),
- vol_chg = frame_count * vol_rate;
-
- if( vol_chg > fabsf( vol_diff ) )
- vg_audio.volume = vg_audio.volume_target_internal;
- else
- vg_audio.volume += vg_signf( vol_diff ) * vol_chg;
+ vg_profile_begin( &_vg_prof_dsp );
+
+ for( int i=0; i<frame_count; i++ )
+ vg_dsp_process( pOut32F + i*2, pOut32F + i*2 );
+
+ vg_profile_end( &_vg_prof_dsp );
- /* redistribute */
- audio_system_cleanup();
-
audio_lock();
+ for( int i=0; i<AUDIO_CHANNELS; i ++ ){
+ audio_channel *ch = &vg_audio.channels[i];
+ ch->readable_activity = ch->activity;
+ }
+
+ /* Profiling information
+ * ----------------------------------------------- */
vg_profile_increment( &_vg_prof_audio_decode );
vg_profile_increment( &_vg_prof_audio_mix );
+ vg_profile_increment( &_vg_prof_dsp );
vg_prof_audio_mix = _vg_prof_audio_mix;
vg_prof_audio_decode = _vg_prof_audio_decode;
+ vg_prof_audio_dsp = _vg_prof_dsp;
vg_audio.samples_last = frame_count;
+
+ if( vg_audio.debug_dsp ){
+ vg_dsp_update_texture();
+ }
+
audio_unlock();
}
if( lin_alloc == NULL )
lin_alloc = vg_audio.audio_pool;
- if( clip->source_mode == k_audio_source_mono )
- {
+ /* load in directly */
+ u32 format = clip->flags & AUDIO_FLAG_FORMAT;
+
+ /* TODO: This contains audio_lock() and unlock, but i don't know why
+ * can probably remove them. Low priority to check this */
+
+ /* TODO: packed files for vorbis etc, should take from data if its not not
+ * NULL when we get the clip
+ */
+
+ if( format == k_audio_format_vorbis ){
+ if( !clip->path ){
+ vg_fatal_exit_loop( "No path specified, embeded vorbis unsupported" );
+ }
+
+ audio_lock();
+ clip->data = vg_file_read( lin_alloc, clip->path, &clip->size );
+ audio_unlock();
+
+ if( !clip->data )
+ vg_fatal_exit_loop( "Audio failed to load" );
+
+ float mb = (float)(clip->size) / (1024.0f*1024.0f);
+ vg_info( "Loaded audio clip '%s' (%.1fmb)\n", clip->path, mb );
+ }
+ else if( format == k_audio_format_stereo ){
+ vg_fatal_exit_loop( "Unsupported format (Stereo uncompressed)" );
+ }
+ else if( format == k_audio_format_bird ){
+ if( !clip->data ){
+ vg_fatal_exit_loop( "No data, external birdsynth unsupported" );
+ }
+
+ u32 total_size = clip->size + sizeof(struct synth_bird);
+ total_size -= sizeof(struct synth_bird_settings);
+ total_size = vg_align8( total_size );
+
+ if( total_size > AUDIO_DECODE_SIZE )
+ vg_fatal_exit_loop( "Bird coding too long\n" );
+
+ struct synth_bird *bird = vg_linear_alloc( lin_alloc, total_size );
+ memcpy( &bird->settings, clip->data, clip->size );
+
+ clip->data = bird;
+ clip->size = total_size;
+
+ vg_info( "Loaded bird synthesis pattern (%u bytes)\n", total_size );
+ }
+ else{
+ if( !clip->path ){
+ vg_fatal_exit_loop( "No path specified, embeded mono unsupported" );
+ }
+
vg_linear_clear( vg_mem.scratch );
u32 fsize;
stb_vorbis *decoder = stb_vorbis_open_memory(
filedata, fsize, &err, &alloc );
- if( !decoder )
- {
+ if( !decoder ){
vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
clip->path, err );
vg_fatal_exit_loop( "Vorbis decode error" );
data_size = length_samples * sizeof(i16);
audio_lock();
- clip->data = vg_linear_alloc( lin_alloc, data_size );
+ clip->data = vg_linear_alloc( lin_alloc, vg_align8(data_size) );
clip->size = length_samples;
audio_unlock();
vg_info( "Loaded audio clip '%s' (%.1fmb) %u samples\n", clip->path, mb,
length_samples );
}
-
- /* load in directly */
- else if( clip->source_mode == k_audio_source_compressed )
- {
- audio_lock();
- clip->data = vg_file_read( lin_alloc, clip->path, &clip->size );
- audio_unlock();
-
- if( !clip->data )
- vg_fatal_exit_loop( "Audio failed to load" );
-
- float mb = (float)(clip->size) / (1024.0f*1024.0f);
- vg_info( "Loaded audio clip '%s' (%.1fmb)\n", clip->path, mb );
- }
}
VG_STATIC void audio_clip_loadn( audio_clip *arr, int count, void *lin_alloc )
audio_clip_load( &arr[i], lin_alloc );
}
-/* Mark change to be uploaded through queue system */
-VG_STATIC void audio_player_commit( audio_player *sys )
-{
- audio_require_lock();
-
- if( vg_audio.queue_len >= vg_list_size( vg_audio.entity_queue ) )
- {
- vg_warn( "Audio commit queue full\n" );
- return;
- }
-
- if( sys->enqued )
- {
- vg_warn( "[2] Audio commit spamming; already enqued (%s)\n", sys->name );
- return;
- }
-
- sys->enqued = 1;
- audio_entity *ent = &vg_audio.entity_queue[ vg_audio.queue_len ++ ];
- ent->info = sys->info;
- ent->player = sys;
-}
-
-VG_STATIC void audio_require_init( audio_player *player )
-{
- if( player->init )
- return;
-
- audio_unlock();
- vg_fatal_exit_loop( "Must init audio player before playing! \n" );
-}
-
VG_STATIC void audio_require_clip_loaded( audio_clip *clip )
{
if( clip->data && clip->size )
vg_fatal_exit_loop( "Must load audio clip before playing! \n" );
}
-/* Play a clip using player. If its already playing something, it will
- * fadeout quickly and start the next sound */
-VG_STATIC void audio_player_playclip( audio_player *player, audio_clip *clip )
-{
- audio_require_lock();
- audio_require_init( player );
- audio_require_clip_loaded( clip );
-
- if( player->info.flags & AUDIO_FLAG_KILL )
- {
- vg_error( "Can't start audio clip on player that is/has disconnected" );
- return;
- }
-
- if( player->enqued )
- {
- vg_warn( "[1] Audio commit spamming; already enqued (%s)\n",
- player->name );
- return;
- }
-
- player->info.source = clip;
- audio_player_commit( player );
-}
-
-VG_STATIC void audio_play_oneshot( audio_clip *clip, float volume )
-{
- audio_require_lock();
- audio_require_clip_loaded( clip );
-
- if( vg_audio.queue_len >= vg_list_size( vg_audio.entity_queue ) )
- {
- vg_warn( "Audio commit queue full\n" );
- return;
- }
-
- audio_entity *ent = &vg_audio.entity_queue[ vg_audio.queue_len ++ ];
-
- ent->info.flags = AUDIO_FLAG_ONESHOT;
- ent->info.pan = 0.0f;
- ent->info.source = clip;
- ent->info.vol = volume;
- ent->player = NULL;
-}
-
-VG_STATIC void audio_player_init( audio_player *player )
-{
- player->active_entity = AATREE_PTR_NIL;
- player->init = 1;
-}
-
-/*
- * Effects
- */
-
-/*
- * Safety enforced Get/set attributes
- */
-
-VG_STATIC int audio_player_is_playing( audio_player *sys )
-{
- audio_require_lock();
-
- if( sys->active_entity != AATREE_PTR_NIL )
- return 1;
- else
- return 0;
-}
-
-VG_STATIC void audio_player_set_position( audio_player *sys, v3f pos )
-{
- audio_require_lock();
- v3_copy( pos, sys->info.world_position );
-}
-
-VG_STATIC void audio_player_set_vol( audio_player *sys, float vol )
-{
- audio_require_lock();
-
- if( !vg_validf(vol) )
- {
- vg_warn( "NaN volume (%s)\n", sys->name );
- vol = 0.0f;
- }
-
- if( (vol < 0.0f) || (vol > 100.0f) )
- {
- vg_warn( "Invalid volume (%s: %f)\n", sys->name, vol );
- vol = 0.0f;
- }
-
- sys->info.vol = vol;
-}
-
-VG_STATIC float audio_player_get_vol( audio_player *sys )
-{
- audio_require_lock();
- return sys->info.vol;
-}
-
-VG_STATIC void audio_player_set_pan( audio_player *sys, float pan )
-{
- audio_require_lock();
- sys->info.pan = pan;
-}
-
-VG_STATIC float audio_player_get_pan( audio_player *sys )
-{
- audio_require_lock();
- return sys->info.pan;
-}
-
-VG_STATIC void audio_player_set_flags( audio_player *sys, u32 flags )
-{
- audio_require_lock();
- sys->info.flags = flags;
-}
-
-VG_STATIC u32 audio_player_get_flags( audio_player *sys )
-{
- audio_require_lock();
- return sys->info.flags;
-}
-
-VG_STATIC void audio_set_master_vol( float vol )
-{
- audio_require_lock();
- vg_audio.volume_target = vol;
-}
-
-VG_STATIC void audio_push_console_vol(void)
-{
- audio_lock();
- audio_set_master_vol( vg_audio.volume_console );
- audio_unlock();
-}
-
/*
* Debugging
*/
if( !vg_audio.debug_ui )
return;
- /* Get data */
- struct sound_info
- {
- const char *name;
- u32 cursor, flags, length;
- v3f pos;
- float vol;
- }
- infos[ SFX_MAX_SYSTEMS ];
- int num_systems = 0;
-
audio_lock();
-
- for( int i=0; i<SFX_MAX_SYSTEMS; i ++ )
- {
- struct active_audio_player *aap = &vg_audio.active_players[i];
- if( !aap->active )
- continue;
+ glBindTexture( GL_TEXTURE_2D, vg_dsp.view_texture );
+ glTexSubImage2D( GL_TEXTURE_2D, 0, 0, 0, 256, 256,
+ GL_RGBA, GL_UNSIGNED_BYTE,
+ vg_dsp.view_texture_buffer );
- audio_entity *ent = &aap->ent;
- struct sound_info *snd = &infos[ num_systems ++ ];
-
- snd->name = ent->name;
- snd->cursor = ent->cur;
- snd->flags = ent->info.flags;
- snd->length = ent->length;
- snd->vol = ent->info.vol*100.0f;
- v3_copy( ent->info.world_position, snd->pos );
- }
+ /*
+ * Profiler
+ * -----------------------------------------------------------------------
+ */
float budget = ((double)vg_audio.samples_last / 44100.0) * 1000.0;
vg_profile_drawn( (struct vg_profile *[]){ &vg_prof_audio_decode,
- &vg_prof_audio_mix }, 2,
+ &vg_prof_audio_mix,
+ &vg_prof_audio_dsp}, 3,
budget, (ui_rect){ 4, VG_PROFILE_SAMPLE_COUNT*2 + 8,
- 250, 0 }, 3 );
+ 512, 0 }, 3 );
- audio_unlock();
char perf[128];
/* Draw UI */
- vg_uictx.cursor[0] = 258;
- vg_uictx.cursor[1] = VG_PROFILE_SAMPLE_COUNT*2+8+24+12;
+ vg_uictx.cursor[0] = 512 + 8;
+ vg_uictx.cursor[1] = VG_PROFILE_SAMPLE_COUNT*2+8+24+12+12;
vg_uictx.cursor[2] = 150;
vg_uictx.cursor[3] = 12;
+
+ if( vg_audio.debug_dsp ){
+ ui_rect view_thing = { 4, vg.window_y-512-4, 512, 512 };
+ ui_push_image( view_thing, vg_dsp.view_texture );
+ }
float mb1 = 1024.0f*1024.0f,
usage = vg_linear_get_cur( vg_audio.audio_pool ) / mb1,
ui_text( vg_uictx.cursor, perf, 1, 0 );
vg_uictx.cursor[1] += 20;
- ui_rect overlap_buffer[ SFX_MAX_SYSTEMS ];
+ ui_rect overlap_buffer[ AUDIO_CHANNELS ];
u32 overlap_length = 0;
/* Draw audio stack */
- for( int i=0; i<num_systems; i ++ )
- {
- struct sound_info *inf = &infos[i];
+ for( int i=0; i<AUDIO_CHANNELS; i ++ ){
+ audio_channel *ch = &vg_audio.channels[i];
- vg_uictx.cursor[2] = 200;
+ vg_uictx.cursor[2] = 400;
vg_uictx.cursor[3] = 18;
- u32 alpha = 0xa0000000;
-
ui_new_node();
- {
- ui_fill_rect( vg_uictx.cursor, 0x00333333|alpha );
-
- ui_px baseline = vg_uictx.cursor[0],
- w = 200,
- c = baseline + ((float)inf->cursor / (float)inf->length) * w;
-
- /* cursor */
- vg_uictx.cursor[2] = 2;
- vg_uictx.cursor[0] = c;
- ui_fill_rect( vg_uictx.cursor, 0xffffffff );
-
- vg_uictx.cursor[0] = baseline + 2;
- vg_uictx.cursor[1] += 2;
- snprintf( perf, 127, "%s %.1f%%", infos[i].name, infos[i].vol );
- ui_text( vg_uictx.cursor, perf, 1, 0 );
-
- if( inf->flags & AUDIO_FLAG_SPACIAL_3D )
- {
- v4f wpos;
- v3_copy( inf->pos, wpos );
- wpos[3] = 1.0f;
- m4x4_mulv( mtx_pv, wpos, wpos );
- if( wpos[3] <= 0.0f )
- goto projected_behind;
+ if( !ch->allocated ){
+ ui_fill_rect( vg_uictx.cursor, 0x50333333 );
+
+ ui_end_down();
+ vg_uictx.cursor[1] += 1;
+ continue;
+ }
+
+ const char *formats[] =
+ {
+ " mono ",
+ " stereo ",
+ " vorbis ",
+ " none0 ",
+ " none1 ",
+ " none2 ",
+ " none3 ",
+ " none4 ",
+ "synth:bird",
+ " none5 ",
+ " none6 ",
+ " none7 ",
+ " none8 ",
+ " none9 ",
+ " none10 ",
+ " none11 ",
+ };
+
+ const char *activties[] =
+ {
+ "reset",
+ "wake ",
+ "alive",
+ "end ",
+ "error"
+ };
+
+ u32 format_index = (ch->source->flags & AUDIO_FLAG_FORMAT)>>9;
+
+ snprintf( perf, 127, "%02d %c%c%cD %s [%s] %4.2fv'%s'",
+ i,
+ (ch->editable_state.relinquished)? 'r': '_',
+ 0? 'r': '_',
+ 0? '3': '2',
+ formats[format_index],
+ activties[ch->readable_activity],
+ ch->editable_state.volume,
+ ch->name );
+ ui_fill_rect( vg_uictx.cursor, 0xa0000000 | ch->colour );
+
+ vg_uictx.cursor[0] += 2;
+ vg_uictx.cursor[1] += 2;
+ ui_text( vg_uictx.cursor, perf, 1, 0 );
+
+ ui_end_down();
+ vg_uictx.cursor[1] += 1;
+
+ if( AUDIO_FLAG_SPACIAL_3D ){
+ v4f wpos;
+ v3_copy( ch->editable_state.spacial_falloff, wpos );
+
+ wpos[3] = 1.0f;
+ m4x4_mulv( mtx_pv, wpos, wpos );
+
+ if( wpos[3] > 0.0f ){
v2_muls( wpos, (1.0f/wpos[3]) * 0.5f, wpos );
v2_add( wpos, (v2f){ 0.5f, 0.5f }, wpos );
ui_rect wr;
- wr[0] = wpos[0] * vg.window_x;
- wr[1] = (1.0f-wpos[1]) * vg.window_y;
+ wr[0] = vg_clampf(wpos[0] * vg.window_x, -32000.0f,32000.0f);
+ wr[1] = vg_clampf((1.0f-wpos[1]) * vg.window_y,-32000.0f,32000.0f);
wr[2] = 100;
wr[3] = 17;
- for( int j=0; j<12; j++ )
- {
+ for( int j=0; j<12; j++ ){
int collide = 0;
- for( int k=0; k<overlap_length; k++ )
- {
+ for( int k=0; k<overlap_length; k++ ){
ui_px *wk = overlap_buffer[k];
if( ((wr[0] <= wk[0]+wk[2]) && (wr[0]+wr[2] >= wk[0])) &&
((wr[1] <= wk[1]+wk[3]) && (wr[1]+wr[3] >= wk[1])) )
ui_rect_copy( wr, overlap_buffer[ overlap_length ++ ] );
}
- }
-
-projected_behind:
-
- ui_end_down();
- vg_uictx.cursor[1] += 1;
+ }
}
+
+ audio_unlock();
}
#endif /* VG_AUDIO_H */