-/* 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_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 <sys/time.h>
-#include <math.h>
+#include "vg/vg_audio_synth_bird.h"
#ifdef __GNUC__
#ifndef __clang__
#endif
#endif
+#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_SPACIAL_3D 0x2
+#define AUDIO_FLAG_NO_DOPPLER 0x2
+#define AUDIO_FLAG_SPACIAL_3D 0x4
+#define AUDIO_FLAG_AUTO_START 0x8
+#define AUDIO_FLAG_FORMAT 0x1E00
-/* 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
+enum audio_format
+{
+ 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
typedef struct audio_channel audio_channel;
typedef struct audio_lfo audio_lfo;
-struct audio_clip
-{
- const char *path;
- u32 flags;
+struct audio_clip{
+ 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,
/* synchro */
int sync_locked;
- SDL_mutex *mux_checker,
- *mux_sync;
+ SDL_SpinLock sl_checker,
+ sl_sync;
- struct audio_lfo
- {
+ struct audio_lfo{
u32 time, time_startframe;
float sqrt_polynomial_coefficient;
- struct
- {
- enum lfo_wave_type
- {
+ struct{
+ enum lfo_wave_type{
k_lfo_triangle,
k_lfo_square,
k_lfo_saw,
}
oscillators[ AUDIO_LFOS ];
- struct audio_channel
- {
+ struct audio_channel{
int allocated;
+ u16 group;
+ u8 world_id;
+
char name[32]; /* only editable while allocated == 0 */
audio_clip *source; /* ... */
u32 flags; /* ... */
+ u32 colour; /* ... */
/* internal non-readable state
* -----------------------------*/
u32 volume_movement,
pan_movement;
- stb_vorbis *vorbis_handle;
+ union{
+ struct synth_bird *bird_handle;
+ stb_vorbis *vorbis_handle;
+ };
+
stb_vorbis_alloc vorbis_alloc;
- enum channel_activity
- {
+ 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;
+ 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
- {
+ struct channel_state{
int relinquished;
float volume, /* current volume */
volume_target, /* target volume */
pan,
- pan_target;
+ pan_target,
+ sampling_rate;
u32 volume_rate,
pan_rate;
}
channels[ AUDIO_CHANNELS ];
- /* System queue, and access from thread 0 */
- int debug_ui, debug_ui_3d;
+ int debug_ui, debug_ui_3d, debug_dsp;
- v3f listener_pos,
- listener_ears;
+ v3f internal_listener_pos,
+ internal_listener_ears,
+ internal_listener_velocity,
- float volume,
- volume_target,
- volume_target_internal,
- volume_console;
+ external_listener_pos,
+ external_listener_ears,
+ external_lister_velocity;
+
+ float internal_global_volume,
+ external_global_volume;
}
-vg_audio = { .volume_console = 1.0f };
+static 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
* 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_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 )
+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)
+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_LockMutex( vg_audio.mux_sync );
+ 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_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)
+static void audio_mixer_callback( void *user, u8 *stream, int frame_count );
+static void vg_audio_init(void)
{
- vg_audio.mux_checker = SDL_CreateMutex();
- vg_audio.mux_sync = SDL_CreateMutex();
-
/* TODO: Move here? */
- vg_var_push( (struct vg_var){
- .name = "debug_audio",
- .data = &vg_audio.debug_ui,
- .data_type = k_var_dtype_i32,
- .opt_i32 = { .min=0, .max=1, .clamp=1 },
- .persistent = 1
- });
-
- vg_var_push( (struct vg_var){
- .name = "volume",
- .data = &vg_audio.volume_console,
- .data_type = k_var_dtype_f32,
- .opt_f32 = { .min=0.0f, .max=2.0f, .clamp=1 },
- .persistent = 1
- });
+ vg_console_reg_var( "debug_audio", &vg_audio.debug_ui,
+ k_var_dtype_i32, VG_VAR_CHEAT );
+ vg_console_reg_var( "debug_dsp", &vg_audio.debug_dsp,
+ k_var_dtype_i32, VG_VAR_CHEAT );
+ vg_console_reg_var( "volume", &vg_audio.external_global_volume,
+ k_var_dtype_f32, VG_VAR_PERSISTENT );
/* allocate memory */
-
/* 32mb fixed */
vg_audio.audio_pool =
vg_create_linear_allocator( vg_mem.rtmemory, 1024*1024*32,
u32 decode_size = AUDIO_DECODE_SIZE * AUDIO_CHANNELS;
vg_audio.decode_buffer = vg_linear_alloc( vg_mem.rtmemory, decode_size );
+ vg_dsp_init();
+
SDL_AudioSpec spec_desired, spec_got;
spec_desired.callback = audio_mixer_callback;
spec_desired.channels = 2;
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
- {
- vg_fatal_exit_loop(
+ else{
+ vg_fatal_error(
"SDL_OpenAudioDevice failed. Your default audio device must support:\n"
" Frequency: 44100 hz\n"
" Buffer size: 512\n"
" Channels: 2\n"
" Format: s16 or f32\n" );
}
-
- vg_success( "Ready\n" );
}
-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_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->world_id = 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
+ vg_strncpy( clip->path, ch->name, 32, k_strncpy_always_add_null );
+
+ 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, u16 group )
+{
+ ch->group = group;
+ ch->colour = (((u32)group * 29986577) & 0x00ffffff) | 0xff000000;
+}
-static audio_channel *audio_request_channel( audio_clip *clip, u32 flags )
+static void audio_channel_world( audio_channel *ch, u8 world_id )
{
- for( int i=0; i<AUDIO_CHANNELS; i++ )
- {
+ ch->world_id = world_id;
+}
+
+static audio_channel *audio_get_first_idle_channel(void)
+{
+ for( int i=0; i<AUDIO_CHANNELS; i++ ){
audio_channel *ch = &vg_audio.channels[i];
- if( !ch->allocated )
- {
- ch->source = clip;
- ch->flags = flags;
- strcpy( ch->name, clip->path );
-
- 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->editble_state_write_mask = 0x00;
+ if( !ch->allocated ){
return ch;
}
}
return NULL;
}
+static audio_channel *audio_get_group_idle_channel( u16 group, u32 max_count )
+{
+ u32 count = 0;
+ audio_channel *dest = NULL;
+
+ for( int i=0; i<AUDIO_CHANNELS; i++ ){
+ audio_channel *ch = &vg_audio.channels[i];
+
+ if( ch->allocated ){
+ if( ch->group == group ){
+ count ++;
+ }
+ }
+ else{
+ if( !dest )
+ dest = ch;
+ }
+ }
+
+ if( dest && (count < max_count) ){
+ return dest;
+ }
+
+ return NULL;
+}
+
+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];
+ if( ch->allocated && (ch->group == group) )
+ return ch;
+ }
+ return NULL;
+}
+
+static int audio_channel_finished( audio_channel *ch )
+{
+ if( ch->readable_activity == k_channel_activity_end )
+ return 1;
+ else
+ return 0;
+}
+
static audio_channel *audio_relinquish_channel( audio_channel *ch )
{
ch->editable_state.relinquished = 1;
return NULL;
}
-static audio_channel *audio_channel_slope_volume( audio_channel *ch,
- float length,
- float new_volume )
+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;
+}
- return ch;
+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 audio_channel *audio_channel_edit_volume( audio_channel *ch,
- float new_volume, int instant )
+static void audio_channel_edit_volume( audio_channel *ch,
+ float new_volume, int instant )
{
- if( instant )
- {
- ch->editable_state.volume = 0.0f;
+ if( instant ){
+ ch->editable_state.volume = new_volume;
ch->editble_state_write_mask |= AUDIO_EDIT_VOLUME;
- return ch;
}
- else
- {
- return audio_channel_slope_volume( ch, 0.05f, new_volume );
+ else{
+ audio_channel_slope_volume( ch, 0.05f, new_volume );
}
}
static audio_channel *audio_channel_fadeout( audio_channel *ch, float length )
{
- ch = audio_channel_slope_volume( ch, length, 0.0f );
- ch = audio_relinquish_channel( ch );
-
- return ch;
+ audio_channel_slope_volume( ch, length, 0.0f );
+ return audio_relinquish_channel( ch );
}
-static audio_channel *audio_channel_fadein( audio_channel *ch, float length )
+static void audio_channel_fadein( audio_channel *ch, float length )
{
- ch = audio_channel_edit_volume( ch, 0.0f, 1 );
- ch = audio_channel_slope_volume( ch, length, 1.0f );
- return ch;
+ 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,
u32 cursor = 0;
if( ch )
- {
ch = audio_channel_fadeout( ch, length );
- }
- audio_channel *replacement = audio_request_channel( new_clip, flags );
+ audio_channel *replacement = audio_get_first_idle_channel();
- if( replacement )
- {
- replacement = audio_channel_fadein( replacement, length );
+ if( replacement ){
+ audio_channel_init( replacement, new_clip, flags );
+ audio_channel_fadein( replacement, length );
}
return replacement;
}
-static audio_channel *audio_channel_sidechain_lfo( audio_channel *ch,
- int lfo_id, float amount )
+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;
-
- return ch;
}
-static audio_channel *audio_channel_set_spacial( audio_channel *ch,
- v3f co, float range )
+static void audio_channel_set_spacial( audio_channel *ch, v3f co, float range )
{
- if( ch->flags & AUDIO_FLAG_SPACIAL_3D )
- {
+ if( ch->flags & AUDIO_FLAG_SPACIAL_3D ){
v3_copy( co, ch->editable_state.spacial_falloff );
- ch->editable_state.spacial_falloff[3] = 1.0f/range;
+
+ if( range == 0.0f )
+ ch->editable_state.spacial_falloff[3] = 1.0f;
+ else
+ ch->editable_state.spacial_falloff[3] = 1.0f/range;
+
ch->editble_state_write_mask |= AUDIO_EDIT_SPACIAL;
}
- else
- {
+ else{
vg_warn( "Tried to set spacialization paramaters for 2D channel (%s)\n",
ch->name );
}
-
- return ch;
}
-static audio_channel *audio_oneshot_3d( audio_clip *clip, v3f position,
- float range, float volume )
+static int audio_oneshot_3d( audio_clip *clip, v3f position,
+ float range, float volume )
{
- audio_channel *ch = audio_request_channel( clip, AUDIO_FLAG_SPACIAL_3D );
+ audio_channel *ch = audio_get_first_idle_channel();
- if( ch )
- {
- ch = audio_channel_set_spacial( ch, position, range );
- ch = audio_channel_edit_volume( ch, volume, 1 );
+ 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 );
- }
- return ch;
+ return 1;
+ }
+ else
+ return 0;
}
-static audio_channel *audio_oneshot( audio_clip *clip, float volume, float pan )
+static int audio_oneshot( audio_clip *clip, float volume, float pan )
{
- audio_channel *ch = audio_request_channel( clip, 0x00 );
+ audio_channel *ch = audio_get_first_idle_channel();
- if( ch )
- {
- ch = audio_channel_edit_volume( ch, volume, 1 );
+ if( ch ){
+ audio_channel_init( ch, clip, 0x00 );
+ audio_channel_edit_volume( ch, volume, 1 );
ch = audio_relinquish_channel( ch );
- }
- return ch;
+ return 1;
+ }
+ else
+ return 0;
}
static void audio_set_lfo_wave( int id, enum lfo_wave_type type,
lfo->editble_state_write_mask |= AUDIO_EDIT_LFO_PERIOD;
}
+
/*
* Committers
* -----------------------------------------------------------------------------
*/
static int audio_channel_load_source( audio_channel *ch )
{
- if( ch->source->flags & AUDIO_FLAG_VORBIS )
- {
+ u32 format = ch->source->flags & AUDIO_FLAG_FORMAT;
+
+ if( format == k_audio_format_vorbis ){
/* Setup vorbis decoder */
u32 index = ch - vg_audio.channels;
ch->source->data,
ch->source->size, &err, &alloc );
- if( !decoder )
- {
+ if( !decoder ){
vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
ch->source->path, err );
return 0;
}
- else
- {
+ else{
ch->source_length = stb_vorbis_stream_length_in_samples( decoder );
ch->vorbis_handle = decoder;
}
}
- else if( ch->source->flags & AUDIO_FLAG_STEREO )
- {
+ else if( format == k_audio_format_bird ){
+ u32 index = ch - vg_audio.channels;
+
+ 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
- {
+ 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++ )
- {
+ 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);
}
/*
* 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 )
{
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];
}
/*
* ........ 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,
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;
}
-static float audio_lfo_pull_sample( audio_lfo *lfo )
+static inline float audio_lfo_pull_sample( audio_lfo *lfo )
{
lfo->time ++;
float t = lfo->time;
t /= (float)lfo->_.period;
- if( lfo->_.wave_type == k_lfo_polynomial_bipolar )
- {
+ if( lfo->_.wave_type == k_lfo_polynomial_bipolar ){
/*
* #
* # #
) * (1.0f-fabsf(t));
}
- else
- {
+ else{
return 0.0f;
}
}
u32 remaining = count;
u32 buffer_pos = 0;
- while( remaining )
- {
- u32 samples_this_run = VG_MIN( remaining, ch->source_length -ch->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 ];
- if( ch->source->flags & AUDIO_FLAG_STEREO )
- {
- for( int i=0;i<samples_this_run; i++ )
- {
+ 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( ch->source->flags & AUDIO_FLAG_VORBIS )
- {
+ else if( format == k_audio_format_vorbis ){
int read_samples = stb_vorbis_get_samples_float_interleaved_stereo(
ch->vorbis_handle,
dst,
samples_this_run );
- if( read_samples != samples_this_run )
- {
+ 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++ )
- {
+ for( int i=0; i<samples_this_run; i++ ){
dst[i*2+0] = 0.0f;
dst[i*2+1] = 0.0f;
}
}
}
- else
- {
+ 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];
ch->cursor += samples_this_run;
buffer_pos += samples_this_run;
- if( (ch->flags & AUDIO_FLAG_LOOP) && remaining )
- {
- if( ch->source->flags & AUDIO_FLAG_VORBIS )
+ 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 );
ch->cursor = 0;
continue;
break;
}
- while( remaining )
- {
+ while( remaining ){
buf[ buffer_pos*2 + 0 ] = 0.0f;
buf[ buffer_pos*2 + 1 ] = 0.0f;
buffer_pos ++;
vg_profile_end( &_vg_prof_audio_decode );
}
-static void audio_channel_mix( audio_channel *ch,
- float *buffer, u32 frame_count )
+static void audio_channel_mix( audio_channel *ch, float *buffer )
{
- u32 buffer_pos = 0;
- float *pcf = alloca( frame_count * 2 * sizeof(float) );
- u32 frames_write = frame_count;
+ float framevol_l = vg_audio.internal_global_volume,
+ framevol_r = vg_audio.internal_global_volume;
- audio_channel_get_samples( ch, frame_count, pcf );
- vg_profile_begin( &_vg_prof_audio_mix );
+ float frame_samplerate = ch->_.sampling_rate;
- if( ch->_.lfo )
- ch->_.lfo->time = ch->_.lfo->time_startframe;
+ if( ch->flags & AUDIO_FLAG_SPACIAL_3D ){
+ v3f delta;
+ v3_sub( ch->_.spacial_falloff, vg_audio.internal_listener_pos, delta );
- float framevol_l = 1.0f,
- framevol_r = 1.0f;
+ float dist = v3_length( delta ),
+ vol = vg_maxf( 0.0f, 1.0f - ch->_.spacial_falloff[3]*dist );
- if( ch->flags & AUDIO_FLAG_SPACIAL_3D )
- {
- if( !vg_validf(vg_audio.listener_pos[0]) ||
- !vg_validf(vg_audio.listener_pos[1]) ||
- !vg_validf(vg_audio.listener_pos[2]) ||
- !vg_validf(ch->_.spacial_falloff[0]) ||
- !vg_validf(ch->_.spacial_falloff[1]) ||
- !vg_validf(ch->_.spacial_falloff[2]) )
- {
- vg_error( "NaN listener/world position (%s)\n", ch->name );
+ 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;
+ }
+ }
- framevol_l = 0.0f;
- framevol_r = 0.0f;
+ 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]
+ );
}
+ }
- v3f delta;
- v3_sub( ch->_.spacial_falloff, vg_audio.listener_pos, delta );
+ 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 dist = v3_length( delta ),
- vol = vg_maxf( 0.0f, 1.0f - ch->_.spacial_falloff[3]*dist );
+ float pcf[ AUDIO_MIX_FRAME_SIZE * 2 * 2 ];
- v3_muls( delta, 1.0f/dist, delta );
- float pan = v3_dot( vg_audio.listener_ears, delta );
- vol = powf( vol, 5.0f );
+ audio_channel_get_samples( ch, buffer_length, pcf );
- framevol_l *= (vol * 0.5f) * (1.0f - pan);
- framevol_r *= (vol * 0.5f) * (1.0f + pan);
- }
+ vg_profile_begin( &_vg_prof_audio_mix );
- for( u32 j=0; j<frame_count; j++ )
- {
- if( ch->volume_movement < ch->_.volume_rate )
- {
- ch->volume_movement ++;
+ 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;
- float movement_t = ch->volume_movement;
- movement_t /= (float)ch->_.volume_rate;
+ float volume = ch->_.volume;
+ const float volume_start = ch->volume_movement_start;
+ const float volume_target = ch->_.volume_target;
- ch->_.volume = vg_lerpf( ch->volume_movement_start,
- ch->_.volume_target,
- movement_t );
- }
+ 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 = ch->_.volume * ch->_.volume;
+ float vol_norm = volume * volume;
if( ch->_.lfo )
- vol_norm *= 1.0f + audio_lfo_pull_sample( ch->_.lfo )
- * ch->_.lfo_amount;
-
- float vol_l = vol_norm * framevol_l,
- vol_r = vol_norm * framevol_r;
+ vol_norm *= 1.0f + audio_lfo_pull_sample(ch->_.lfo) * ch->_.lfo_amount;
- buffer[ buffer_pos*2+0 ] += pcf[ buffer_pos*2+0 ] * vol_l;
- buffer[ buffer_pos*2+1 ] += pcf[ buffer_pos*2+1 ] * vol_r;
+ 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 )
+static void audio_mixer_callback( void *user, u8 *stream, int byte_count )
{
/*
* Copy data and move edit flags to commit flags
* ------------------------------------------------------------- */
audio_lock();
- for( int i=0; i<AUDIO_CHANNELS; i++ )
- {
+
+ 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->cursor >= ch->source_length && !(ch->flags & AUDIO_FLAG_LOOP))
+ 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) )
{
}
/* process new channels */
- if( ch->activity == k_channel_activity_reset )
- {
+ if( ch->activity == k_channel_activity_reset ){
ch->_ = ch->editable_state;
ch->cursor = 0;
ch->source_length = 0;
ch->editable_state.relinquished = ch->_.relinquished;
- if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME )
+ if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME ){
ch->_.volume = ch->editable_state.volume;
- else
+ ch->_.volume_target = ch->editable_state.volume;
+ }
+ else{
ch->editable_state.volume = ch->_.volume;
+ }
- if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME_SLOPE )
- {
+ 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
- {
+ 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_LFO_ATTACHMENT )
- {
+
+ 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
- {
+ else{
ch->editable_state.lfo = ch->_.lfo;
ch->editable_state.lfo_amount = ch->_.lfo_amount;
}
ch->editble_state_write_mask = 0x00;
}
- for( int i=0; i<AUDIO_LFOS; i++ )
- {
+ 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 )
- {
+ 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 )
- {
+ if( lfo->_.wave_type == k_lfo_polynomial_bipolar ){
lfo->_.polynomial_coefficient =
lfo->editable_state.polynomial_coefficient;
lfo->sqrt_polynomial_coefficient =
}
}
- if( lfo->editble_state_write_mask & AUDIO_EDIT_LFO_PERIOD )
- {
- if( lfo->_.period )
- {
+ 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
- {
+ 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++ )
- {
+ for( int i=0; i<AUDIO_CHANNELS; i++ ){
audio_channel *ch = &vg_audio.channels[i];
- if( ch->activity == k_channel_activity_wake )
- {
+ if( ch->activity == k_channel_activity_wake ){
if( audio_channel_load_source( ch ) )
ch->activity = k_channel_activity_alive;
else
for( int i=0; i<frame_count*2; i ++ )
pOut32F[i] = 0.0f;
- for( int i=0; i<AUDIO_LFOS; i++ )
- {
+ 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 ++ )
- {
+ for( int i=0; i<AUDIO_CHANNELS; i ++ ){
audio_channel *ch = &vg_audio.channels[i];
- if( ch->activity == k_channel_activity_alive )
- audio_channel_mix( ch, pOut32F, frame_count );
+ if( ch->activity == k_channel_activity_alive ){
+ if( ch->_.lfo )
+ ch->_.lfo->time = ch->_.lfo->time_startframe;
+
+ u32 remaining = frame_count,
+ subpos = 0;
+
+ while( remaining ){
+ audio_channel_mix( ch, pOut32F+subpos );
+ remaining -= AUDIO_MIX_FRAME_SIZE;
+ subpos += AUDIO_MIX_FRAME_SIZE*2;
+ }
+ }
}
- /*
- * Relinquishing conditions
- * ------------------------------------------------------------------
- */
+ 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 );
+
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();
}
-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;
-
/* load in directly */
- if( clip->flags & AUDIO_FLAG_VORBIS )
- {
+ 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_error( "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" );
+ vg_fatal_error( "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( clip->flags & AUDIO_FLAG_STEREO )
- {
- vg_fatal_exit_loop( "Unsupported format (Stereo uncompressed)" );
+ else if( format == k_audio_format_stereo ){
+ vg_fatal_error( "Unsupported format (Stereo uncompressed)" );
}
- else
- {
+ else if( format == k_audio_format_bird ){
+ if( !clip->data ){
+ vg_fatal_error( "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_error( "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_error( "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" );
+ vg_fatal_error( "Vorbis decode error" );
}
/* only mono is supported in uncompressed */
decoder, clip->data, length_samples );
if( read_samples != length_samples )
- vg_fatal_exit_loop( "Decode error" );
+ 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;
audio_unlock();
- vg_fatal_exit_loop( "Must load audio clip before playing! \n" );
+ vg_fatal_error( "Must load audio clip before playing! \n" );
}
/*
* Debugging
*/
-VG_STATIC void audio_debug_ui( m4x4f mtx_pv )
+static void audio_debug_ui( m4x4f mtx_pv )
{
if( !vg_audio.debug_ui )
return;
audio_lock();
+ 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 );
+
/*
* 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 );
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[2] = 150;
- vg_uictx.cursor[3] = 12;
-
- 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 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_image( view_thing, vg_dsp.view_texture );
+ }
ui_rect overlap_buffer[ AUDIO_CHANNELS ];
u32 overlap_length = 0;
/* Draw audio stack */
- for( int i=0; i<AUDIO_CHANNELS; i ++ )
- {
+ 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();
-
- if( !ch->allocated )
- {
- ui_fill_rect( vg_uictx.cursor, 0x50333333 );
+ ui_rect row;
+ ui_split( window, k_ui_axis_h, 18, 1, row, window );
- ui_end_down();
- vg_uictx.cursor[1] += 1;
+ if( !ch->allocated ){
+ ui_fill( row, 0x50333333 );
continue;
}
const char *formats[] =
{
- "------",
- "Mono ",
- "Stereo",
- "Vorbis"
+ " mono ",
+ " stereo ",
+ " vorbis ",
+ " none0 ",
+ " none1 ",
+ " none2 ",
+ " none3 ",
+ " none4 ",
+ "synth:bird",
+ " none5 ",
+ " none6 ",
+ " none7 ",
+ " none8 ",
+ " none9 ",
+ " none10 ",
+ " none11 ",
};
- int format_index = 0;
+ const char *activties[] =
+ {
+ "reset",
+ "wake ",
+ "alive",
+ "end ",
+ "error"
+ };
- if( ch->source->flags & AUDIO_FLAG_STEREO )
- format_index = 2;
- else if( ch->source->flags & AUDIO_FLAG_VORBIS )
- format_index = 3;
- else
- format_index = 1;
+ u32 format_index = (ch->source->flags & AUDIO_FLAG_FORMAT)>>9;
- snprintf( perf, 127, "%02d %c%c%cD %s %4.2fv'%s'",
+ snprintf( perf, 127, "%02d[%#04x.%#06x]%c%c%cD %s [%s] %4.2fv'%s'",
i,
- (ch->editable_state.relinquished)? 'r': ' ',
- 0? 'r': ' ',
+ ch->world_id, ch->group,
+ (ch->editable_state.relinquished)? 'r': '_',
+ 0? 'r': '_',
0? '3': '2',
formats[format_index],
+ activties[ch->readable_activity],
ch->editable_state.volume,
ch->name );
- if( format_index == 0 )
- {
- ui_fill_rect( vg_uictx.cursor, 0xa00000ff );
- }
- else
- {
- ui_fill_rect( vg_uictx.cursor, 0xa0333333 );
- }
-
- 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 )
- {
+ 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 )
- {
+ 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[2] = 100;
+ 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] = 1000;
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])) )
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 ++ ] );
}
}
}
projects / vg.git / blobdiff