#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"
#include "vg/vg_audio_synth_bird.h"
-#include <sys/time.h>
-#include <math.h>
-
#ifdef __GNUC__
#ifndef __clang__
#pragma GCC push_options
typedef struct audio_lfo audio_lfo;
struct audio_clip{
- const char *path;
+ union { /* TODO oof.. */
+ u64 _p64_;
+ const char *path;
+ };
+
u32 flags;
u32 size;
- void *data;
+
+ union{
+ u64 _p64;
+ void *data;
+ };
};
-static struct vg_audio_system{
+struct vg_audio_system{
SDL_AudioDeviceID sdl_output_device;
void *audio_pool,
struct audio_channel{
int allocated;
- u32 group;
+ u16 group;
+ u8 world_id;
char name[32]; /* only editable while allocated == 0 */
audio_clip *source; /* ... */
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 )
{
ch->group = 0;
+ ch->world_id = 0;
ch->source = clip;
ch->flags = flags;
ch->colour = 0x00333333;
ch->editble_state_write_mask = 0x00;
}
-static void audio_channel_group( audio_channel *ch, u32 group )
+static void audio_channel_group( audio_channel *ch, u16 group )
{
ch->group = group;
- ch->colour = ((group * 29986577) & 0x00ffffff) | 0xff000000;
+ ch->colour = (((u32)group * 29986577) & 0x00ffffff) | 0xff000000;
+}
+
+static void audio_channel_world( audio_channel *ch, u8 world_id )
+{
+ ch->world_id = world_id;
}
static audio_channel *audio_get_first_idle_channel(void)
return NULL;
}
-static audio_channel *audio_get_group_idle_channel( u32 group, u32 max_count )
+static audio_channel *audio_get_group_idle_channel( u16 group, u32 max_count )
{
u32 count = 0;
audio_channel *dest = NULL;
return NULL;
}
-static audio_channel *audio_get_group_first_active_channel( u32 group )
+static audio_channel *audio_get_group_first_active_channel( u16 group )
{
for( int i=0; i<AUDIO_CHANNELS; i++ ){
audio_channel *ch = &vg_audio.channels[i];
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;
* -----------------------------------------------------------------------
*/
-#if 0
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,
char perf[128];
/* Draw UI */
- 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;
+ ui_rect window = {
+ 0,
+ 0,
+ 800,
+ AUDIO_CHANNELS * 18
+ };
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 );
+ ui_image( view_thing, vg_dsp.view_texture );
}
-
- float mb1 = 1024.0f*1024.0f,
- usage = vg_linear_get_cur( vg_audio.audio_pool ) / mb1,
- total = vg_linear_get_capacity( vg_audio.audio_pool ) / mb1,
- percent = (usage/total) * 100.0f;
-
- snprintf( perf, 127, "Mem: %.1f/%.1fmb (%.1f%%)\n", usage, total, percent );
-
- ui_text( vg_uictx.cursor, perf, 1, 0 );
- vg_uictx.cursor[1] += 20;
ui_rect overlap_buffer[ AUDIO_CHANNELS ];
u32 overlap_length = 0;
for( int i=0; i<AUDIO_CHANNELS; i ++ ){
audio_channel *ch = &vg_audio.channels[i];
- vg_uictx.cursor[2] = 400;
- vg_uictx.cursor[3] = 18;
-
- ui_new_node();
+ ui_rect row;
+ ui_split( window, k_ui_axis_h, 18, 1, row, window );
if( !ch->allocated ){
- ui_fill_rect( vg_uictx.cursor, 0x50333333 );
-
- ui_end_down();
- vg_uictx.cursor[1] += 1;
+ ui_fill( row, 0x50333333 );
continue;
}
u32 format_index = (ch->source->flags & AUDIO_FLAG_FORMAT)>>9;
- snprintf( perf, 127, "%02d %c%c%cD %s [%s] %4.2fv'%s'",
+ snprintf( perf, 127, "%02d[%#04x.%#06x]%c%c%cD %s [%s] %4.2fv'%s'",
i,
+ ch->world_id, ch->group,
(ch->editable_state.relinquished)? 'r': '_',
0? 'r': '_',
0? '3': '2',
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;
+ ui_fill( row, 0xa0000000 | ch->colour );
+ ui_text( row, perf, 1, k_ui_align_middle_left, 0 );
if( AUDIO_FLAG_SPACIAL_3D ){
v4f wpos;
ui_rect wr;
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[2] = 1000;
wr[3] = 17;
for( int j=0; j<12; j++ ){
wr[1] += 18;
}
- ui_text( wr, perf, 1, 0 );
-
- ui_rect_copy( wr, overlap_buffer[ overlap_length ++ ] );
+ ui_text( wr, perf, 1, k_ui_align_middle_left, 0 );
+ rect_copy( wr, overlap_buffer[ overlap_length ++ ] );
}
}
}
-#endif
audio_unlock();
}
projects / vg.git / blobdiff