#include "vg/vg_platform.h"
#include "vg/vg_io.h"
#include "vg/vg_m.h"
-#include "vg/vg_ui.h"
#include "vg/vg_console.h"
#include "vg/vg_store.h"
#include "vg/vg_profiler.h"
};
};
-static struct vg_audio_system{
+struct vg_audio_system{
SDL_AudioDeviceID sdl_output_device;
void *audio_pool,
float internal_global_volume,
external_global_volume;
}
-vg_audio = { .external_global_volume = 1.0f };
+static vg_audio = { .external_global_volume = 1.0f };
#include "vg/vg_audio_dsp.h"
* 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 );
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;
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;
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,
}
}
- if( !vg_validf( framevol_l ) ) vg_fatal_error( "NaN left channel" );
- if( !vg_validf( framevol_r ) ) vg_fatal_error( "NaN right channel" );
- if( !vg_validf( frame_samplerate ) )
- vg_fatal_error( "NaN sample rate" );
+ if( !vg_validf( framevol_l ) ||
+ !vg_validf( framevol_r ) ||
+ !vg_validf( frame_samplerate ) ){
+ vg_fatal_error( "Invalid sampling conditions.\n"
+ "This crash is to protect your ears.\n"
+ " channel: %p (%s)\n"
+ " sample_rate: %f\n"
+ " volume: L%f R%f\n"
+ " listener: %.2f %.2f %.2f [%.2f %.2f %.2f]\n",
+ ch, ch->name, frame_samplerate,
+ framevol_l, framevol_r,
+ vg_audio.internal_listener_pos[0],
+ vg_audio.internal_listener_pos[1],
+ vg_audio.internal_listener_pos[2],
+ vg_audio.internal_listener_ears[0],
+ vg_audio.internal_listener_ears[1],
+ vg_audio.internal_listener_ears[2]
+ );
+ }
}
u32 buffer_length = AUDIO_MIX_FRAME_SIZE;
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;
if( read_samples != length_samples )
vg_fatal_error( "Decode error" );
+#if 0
float mb = (float)(data_size) / (1024.0f*1024.0f);
vg_info( "Loaded audio clip '%s' (%.1fmb) %u samples\n", clip->path, mb,
length_samples );
+#endif
}
}
-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;
*/
float budget = ((double)vg_audio.samples_last / 44100.0) * 1000.0;
-#if 0
vg_profile_drawn( (struct vg_profile *[]){ &vg_prof_audio_decode,
&vg_prof_audio_mix,
&vg_prof_audio_dsp}, 3,
budget, (ui_rect){ 4, VG_PROFILE_SAMPLE_COUNT*2 + 8,
512, 0 }, 3 );
-#endif
char perf[128];
projects / vg.git / blobdiff