k_audio_format_none4 = 0xE00u,
k_audio_format_bird = 0x1000u,
- k_audio_format_none5 = 0x1200u,
+ k_audio_format_gen = 0x1200u,
k_audio_format_none6 = 0x1400u,
k_audio_format_none7 = 0x1600u,
k_audio_format_none8 = 0x1800u,
union { /* TODO oof.. */
u64 _p64_;
const char *path;
+ void *func;
};
u32 flags;
* These functions are called from the main thread and used to prevent bad
* access. TODO: They should be no-ops in release builds.
*/
-VG_STATIC int audio_lock_checker_load(void)
+static int audio_lock_checker_load(void)
{
int value;
SDL_AtomicLock( &vg_audio.sl_checker );
return value;
}
-VG_STATIC void audio_lock_checker_store( int value )
+static void audio_lock_checker_store( int value )
{
SDL_AtomicLock( &vg_audio.sl_checker );
vg_audio.sync_locked = value;
SDL_AtomicUnlock( &vg_audio.sl_checker );
}
-VG_STATIC void audio_require_lock(void)
+static void audio_require_lock(void)
{
if( audio_lock_checker_load() )
return;
abort();
}
-VG_STATIC void audio_lock(void)
+static void audio_lock(void)
{
SDL_AtomicLock( &vg_audio.sl_sync );
audio_lock_checker_store(1);
}
-VG_STATIC void audio_unlock(void)
+static void audio_unlock(void)
{
audio_lock_checker_store(0);
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)
+static void audio_mixer_callback( void *user, u8 *stream, int frame_count );
+static void vg_audio_init(void)
{
/* TODO: Move here? */
vg_console_reg_var( "debug_audio", &vg_audio.debug_ui,
}
}
-VG_STATIC void vg_audio_free(void)
+static void vg_audio_free(void)
{
vg_dsp_free();
SDL_CloseAudioDevice( vg_audio.sdl_output_device );
#define AUDIO_EDIT_OWNERSHIP 0x40
#define AUDIO_EDIT_SAMPLING_RATE 0x80
-static void audio_channel_init( audio_channel *ch, audio_clip *clip, u32 flags )
-{
+static void audio_channel_init( audio_channel *ch, audio_clip *clip,
+ u32 flags ){
+ audio_require_lock();
ch->group = 0;
ch->world_id = 0;
ch->source = clip;
if( (ch->source->flags & AUDIO_FLAG_FORMAT) == k_audio_format_bird )
strcpy( ch->name, "[array]" );
+ else if( (ch->source->flags & AUDIO_FLAG_FORMAT) == k_audio_format_gen )
+ strcpy( ch->name, "[program]" );
else
vg_strncpy( clip->path, ch->name, 32, k_strncpy_always_add_null );
static void audio_channel_group( audio_channel *ch, u16 group )
{
+ audio_require_lock();
ch->group = group;
ch->colour = (((u32)group * 29986577) & 0x00ffffff) | 0xff000000;
}
static void audio_channel_world( audio_channel *ch, u8 world_id )
{
+ audio_require_lock();
ch->world_id = world_id;
}
static audio_channel *audio_get_first_idle_channel(void)
{
+ audio_require_lock();
for( int i=0; i<AUDIO_CHANNELS; i++ ){
audio_channel *ch = &vg_audio.channels[i];
static audio_channel *audio_get_group_idle_channel( u16 group, u32 max_count )
{
+ audio_require_lock();
u32 count = 0;
audio_channel *dest = NULL;
static audio_channel *audio_get_group_first_active_channel( u16 group )
{
+ audio_require_lock();
for( int i=0; i<AUDIO_CHANNELS; i++ ){
audio_channel *ch = &vg_audio.channels[i];
if( ch->allocated && (ch->group == group) )
static int audio_channel_finished( audio_channel *ch )
{
+ audio_require_lock();
if( ch->readable_activity == k_channel_activity_end )
return 1;
else
static audio_channel *audio_relinquish_channel( audio_channel *ch )
{
+ audio_require_lock();
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 )
{
+ audio_require_lock();
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 )
{
+ audio_require_lock();
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 )
{
+ audio_require_lock();
if( instant ){
ch->editable_state.volume = new_volume;
ch->editble_state_write_mask |= AUDIO_EDIT_VOLUME;
static audio_channel *audio_channel_fadeout( audio_channel *ch, float length )
{
+ audio_require_lock();
audio_channel_slope_volume( ch, length, 0.0f );
return audio_relinquish_channel( ch );
}
static void audio_channel_fadein( audio_channel *ch, float length )
{
+ audio_require_lock();
audio_channel_edit_volume( ch, 0.0f, 1 );
audio_channel_slope_volume( ch, length, 1.0f );
}
audio_clip *new_clip,
float length, u32 flags )
{
+ audio_require_lock();
u32 cursor = 0;
if( ch )
static void audio_channel_sidechain_lfo( audio_channel *ch, int lfo_id,
float amount )
{
+ audio_require_lock();
ch->editable_state.lfo = &vg_audio.oscillators[ lfo_id ];
ch->editable_state.lfo_amount = amount;
ch->editble_state_write_mask |= AUDIO_EDIT_LFO_ATTACHMENT;
static void audio_channel_set_spacial( audio_channel *ch, v3f co, float range )
{
+ audio_require_lock();
if( ch->flags & AUDIO_FLAG_SPACIAL_3D ){
v3_copy( co, ch->editable_state.spacial_falloff );
static int audio_oneshot_3d( audio_clip *clip, v3f position,
float range, float volume )
{
+ audio_require_lock();
audio_channel *ch = audio_get_first_idle_channel();
if( ch ){
static int audio_oneshot( audio_clip *clip, float volume, float pan )
{
+ audio_require_lock();
audio_channel *ch = audio_get_first_idle_channel();
if( ch ){
static void audio_set_lfo_wave( int id, enum lfo_wave_type type,
float coefficient )
{
+ audio_require_lock();
audio_lfo *lfo = &vg_audio.oscillators[ id ];
lfo->editable_state.polynomial_coefficient = coefficient;
lfo->editable_state.wave_type = type;
static void audio_set_lfo_frequency( int id, float freq )
{
+ audio_require_lock();
audio_lfo *lfo = &vg_audio.oscillators[ id ];
lfo->editable_state.period = 44100.0f / freq;
lfo->editble_state_write_mask |= AUDIO_EDIT_LFO_PERIOD;
else if( format == k_audio_format_stereo ){
ch->source_length = ch->source->size / 2;
}
+ else if( format == k_audio_format_gen ){
+ ch->source_length = 0xffffffff;
+ }
else{
ch->source_length = ch->source->size;
}
return 1;
}
-VG_STATIC void audio_decode_uncompressed_mono( i16 *src, u32 count, float *dst )
+static void audio_decode_uncompressed_mono( i16 *src, u32 count, float *dst )
{
for( u32 i=0; i<count; i++ ){
dst[ i*2 + 0 ] = ((float)src[i]) * (1.0f/32767.0f);
/*
* adapted from stb_vorbis.h, since the original does not handle mono->stereo
*/
-VG_STATIC int
+static int
stb_vorbis_get_samples_float_interleaved_stereo( stb_vorbis *f, float *buffer,
int len )
{
/*
* ........ more wrecked code sorry!
*/
-VG_STATIC int
+static int
stb_vorbis_get_samples_i16_downmixed( stb_vorbis *f, i16 *buffer, int len )
{
int n = 0,
else if( format == k_audio_format_bird ){
synth_bird_generate_samples( ch->bird_handle, dst, samples_this_run );
}
+ else if( format == k_audio_format_gen ){
+ void (*fn)( void *data, f32 *buf, u32 count ) = ch->source->func;
+ fn( ch->source->data, dst, samples_this_run );
+ }
else{
i16 *src_buffer = ch->source->data,
*src = &src_buffer[ch->cursor];
vg_profile_end( &_vg_prof_audio_mix );
}
-VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count )
+static void audio_mixer_callback( void *user, u8 *stream, int byte_count )
{
/*
* Copy data and move edit flags to commit flags
audio_unlock();
}
-VG_STATIC void audio_clip_load( audio_clip *clip, void *lin_alloc )
+static void audio_clip_load( audio_clip *clip, void *lin_alloc )
{
if( lin_alloc == NULL )
lin_alloc = vg_audio.audio_pool;
+#ifdef VG_AUDIO_FORCE_COMPRESSED
+
+ if( (clip->flags & AUDIO_FLAG_FORMAT) != k_audio_format_bird ){
+ clip->flags &= ~AUDIO_FLAG_FORMAT;
+ clip->flags |= k_audio_format_vorbis;
+ }
+
+#endif
+
/* load in directly */
u32 format = clip->flags & AUDIO_FLAG_FORMAT;
}
}
-VG_STATIC void audio_clip_loadn( audio_clip *arr, int count, void *lin_alloc )
+static void audio_clip_loadn( audio_clip *arr, int count, void *lin_alloc )
{
for( int i=0; i<count; i++ )
audio_clip_load( &arr[i], lin_alloc );
}
-VG_STATIC void audio_require_clip_loaded( audio_clip *clip )
+static void audio_require_clip_loaded( audio_clip *clip )
{
if( clip->data && clip->size )
return;
* Debugging
*/
-VG_STATIC void audio_debug_ui( m4x4f mtx_pv )
+static void audio_debug_ui( m4x4f mtx_pv )
{
if( !vg_audio.debug_ui )
return;
projects / vg.git / blobdiff