audio pt. 2

[vg.git] / src / vg / vg_audio.h

/* Copyright (C) 2021-2022 Harry Godden (hgn) - All Rights Reserved */

#ifndef VG_AUDIO_H
#define VG_AUDIO_H

#define MA_NO_GENERATION
#define MA_NO_DECODING
#define MA_NO_ENCODING
#include "dr_soft/miniaudio.h"
#include "vg/vg.h"
#include "vg/vg_stdint.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 <time.h>

#define STB_VORBIS_MAX_CHANNELS 2
#include "stb/stb_vorbis.h"

#define SFX_MAX_SYSTEMS       32
#define AUDIO_FLAG_LOOP       0x1
#define AUDIO_FLAG_ONESHOT    0x2
#define AUDIO_FLAG_SPACIAL_3D 0x4

#define FADEOUT_LENGTH        1100
#define FADEOUT_DIVISOR       (1.0f/(float)FADEOUT_LENGTH)

#define AUDIO_DECODE_SIZE     (1024*256)  /* 256 kb decoding buffers */

enum audio_source_mode
{
   k_audio_source_mono,
   k_audio_source_compressed,
};

typedef struct audio_clip audio_clip;
struct audio_clip
{
   const char *path;
   enum audio_source_mode source_mode;

   /* result */
   void *data;
   u32   len;  /* decompressed: sample count,
                  compressed:   file size      */
};

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
{
   ma_device         miniaudio_device;
   ma_device_config  miniaudio_dconfig;

   void             *mem, *decode_mem;
   u32               mem_current,
                     mem_total;

   /* synchro */
   int               sync_locked;
   MUTEX_TYPE        mutex_checker;
   MUTEX_TYPE        mutex_sync;

   /* Audio engine, thread 1 */
   struct active_audio_player
   {
      int active;
      union
      {
         audio_entity     ent;
         aatree_pool_node pool_node;
      };
      
      stb_vorbis *vorbis_handle;
      stb_vorbis_alloc vorbis_alloc;
   }
   active_players[ SFX_MAX_SYSTEMS ];

   aatree            active_pool_info;    /* note: just using the pool */
   aatree_ptr        active_pool_head;

   /* System queue, and access from thread 0 */
   audio_entity      entity_queue[SFX_MAX_SYSTEMS];
   int               queue_len;

   char              performance_info[128],
                     performance_sub0[64],
                     performance_sub1[64];

   int               debug_ui;

   v3f               listener_pos,
                     listener_ears;

   double            perf_ms_decode,
                     perf_ms_mix;

   u32               perf_measurements;
}
vg_audio;

static void *audio_alloc( u32 size )
{
   u32 new_current = vg_audio.mem_current + size;
   if( new_current > vg_audio.mem_total )
   {
      vg_error( "audio pool over budget!\n" );
      free( vg_audio.mem );
      return NULL;
   }

   void *ptr = vg_audio.mem + vg_audio.mem_current;
   vg_audio.mem_current = new_current;

   return ptr;
}


/* 
 * These functions are called from the main thread and used to prevent bad 
 * access. TODO: They should be no-ops in release builds.
 */
static int audio_lock_checker_load(void)
{
   int value;
   MUTEX_LOCK( vg_audio.mutex_checker );
   value = vg_audio.sync_locked;
   MUTEX_UNLOCK( vg_audio.mutex_checker );
   return value;
}

static void audio_lock_checker_store( int value )
{
   MUTEX_LOCK( vg_audio.mutex_checker );
   vg_audio.sync_locked = value;
   MUTEX_UNLOCK( vg_audio.mutex_checker );
}

static void audio_require_lock(void)
{
   if( audio_lock_checker_load() )
      return;

   vg_exiterr( "Modifying sound effects systems requires locking\n" );
}

static void audio_lock(void)
{
   MUTEX_LOCK( vg_audio.mutex_sync );
   audio_lock_checker_store(1);
}

static void audio_unlock(void)
{
   audio_lock_checker_store(0);
   MUTEX_UNLOCK( vg_audio.mutex_sync );
}


static void audio_mixer_callback( ma_device *pDevice, void *pOutBuf,
                                    const void *pInput, ma_uint32 frameCount );

static int vg_audio_init(void)
{
   vg_convar_push( (struct vg_convar){
      .name = "debug_audio",
      .data = &vg_audio.debug_ui,
      .data_type = k_convar_dtype_i32,
      .opt_i32 = { .min=0, .max=1, .clamp=1 },
      .persistent = 1
   });

   u32 decode_region    = AUDIO_DECODE_SIZE * SFX_MAX_SYSTEMS;
   vg_audio.mem_total   = 1024*1024*32;
   vg_audio.mem_current = 0;
   vg_audio.mem         = malloc( vg_audio.mem_total + decode_region );
   vg_audio.decode_mem  = &((u8 *)vg_audio.mem)[vg_audio.mem_total];

   /* 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 );

   ma_device_config *dconf  = &vg_audio.miniaudio_dconfig;
   ma_device        *device = &vg_audio.miniaudio_device;

    *dconf = ma_device_config_init( ma_device_type_playback );
   dconf->playback.format    = ma_format_f32;
   dconf->playback.channels  = 2;
   dconf->sampleRate         = 44100;
   dconf->dataCallback       = audio_mixer_callback;
   dconf->periodSizeInFrames = 441;

   dconf->pUserData = NULL;

   vg_info( "Starting audio engine\n" );
   
   if( ma_device_init( NULL, dconf, device) != MA_SUCCESS )
   {
      vg_error( "ma_device failed to initialize" );
      return 0;
   } 
   else 
   {   
      if( ma_device_start( device ) != MA_SUCCESS )
      {
         ma_device_uninit( device );
         vg_error( "ma_device failed to start" );
         return 0;
      }
   }

   return 1;
}

static void vg_audio_free(void)
{
   ma_device        *device = &vg_audio.miniaudio_device;
   ma_device_uninit( device );

   free( vg_audio.mem );
}

/* 
 * thread 1
 */

static aatree_ptr audio_alloc_entity_internal(void)
{
   aatree_ptr playerid = aatree_pool_alloc( &vg_audio.active_pool_info, 
                                            &vg_audio.active_pool_head );

   if( playerid == AATREE_PTR_NIL )
      return AATREE_PTR_NIL;

   struct active_audio_player *aap = &vg_audio.active_players[ playerid ];
   aap->active = 1;

   return playerid;
}

static void audio_entity_free_internal( aatree_ptr id )
{
   struct active_audio_player *aap = &vg_audio.active_players[ id ];
   aap->active = 0;

   /* Notify player that we've finished */
   if( aap->ent.player )
      aap->ent.player->active_entity = AATREE_PTR_NIL;

   /* delete */
   aatree_pool_free( &vg_audio.active_pool_info, id,
                     &vg_audio.active_pool_head );
}

static void *audio_entity_vorbis_ptr( aatree_ptr entid )
{
   u8 *buf = (u8*)vg_audio.decode_mem,
      *loc = &buf[AUDIO_DECODE_SIZE*entid];

   return (void *)loc;
}

static void audio_entity_start( audio_entity *src )
{
   aatree_ptr entid = audio_alloc_entity_internal();
   if( entid == AATREE_PTR_NIL )
      return;

   audio_entity *ent = &vg_audio.active_players[ entid ].ent;

   ent->info = src->info;
   ent->name = src->info.source->path;
   ent->cur  = 0;
   ent->player = src->player;

   ent->fadeout = 0;
   ent->fadeout_current = 0;

   /* Notify main player we are dequeud and playing */
   if( src->player )
   {
      src->player->enqued = 0;
      src->player->active_entity = entid;
   }

   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
      };

      int err;
      stb_vorbis *decoder = stb_vorbis_open_memory( 
            src->info.source->data, src->info.source->len, &err, &alloc );

      if( !decoder )
      {
         vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n", 
                     src->info.source->path, err );

         audio_entity_free_internal( entid );
         return;
      }
      else
      {
         ent->length = stb_vorbis_stream_length_in_samples( decoder );
      }
      
      aap->vorbis_handle = decoder;
   }
   else
   {
      ent->length = src->info.source->len;
   }
}

/*
 * Read everything from the queue 
 */
static void audio_system_enque(void)
{
   /* Process incoming sound queue */
   audio_lock();
   
   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;
            }

            vg_audio.entity_queue[ wr ++ ] = *src;
         }
      }
      else
      {
         audio_entity_start( src );
      }
   }
   
   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;

      if( aap->ent.player )
      {
         /* Only copy information in whilst not requeing */
         if( aap->ent.player->enqued == 0 )
         {
            aap->ent.info = aap->ent.player->info;
         }
      }
   }
   
   audio_unlock();
}

/* 
 * Redistribute sound systems 
 */
static void audio_system_cleanup(void)
{
   audio_lock();

   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 );
         }
      }
   }

   audio_unlock();
}

/* 
 * Get effective volume and pan from this entity
 */
static void audio_entity_spacialize( audio_entity *ent, float *vol, float *pan )
{
   v3f delta;
   v3_sub( ent->info.world_position, vg_audio.listener_pos, delta );

   float dist = v3_length( delta ),
         attn = (dist / ent->info.vol) +1.0f;

   v3_muls( delta, 1.0f/dist, delta );

   *pan = v3_dot( vg_audio.listener_ears, delta );
   *vol = 1.0f/(attn*attn);
}

static void audio_decode_uncompressed_mono( float *src, u32 count, float *dst )
{
   for( u32 i=0; i<count; i++ )
   {
      dst[ i*2 + 0 ] = src[i];
      dst[ i*2 + 1 ] = src[i];
   }
}

/* 
 * adapted from stb_vorbis.h, since the original does not handle mono->stereo
 */
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 ) 
   {
      int k = f->channel_buffer_end - f->channel_buffer_start;

      if( n+k >= len ) 
         k = len - n;

      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];
      }

      n += k;
      f->channel_buffer_start += k;

      if( n == len )
         break;

      if( !stb_vorbis_get_frame_float( f, NULL, NULL ))
         break;
   }

   return n;
}

static void audio_entity_get_samples( aatree_ptr id, u32 count, float *buf )
{
   struct timespec time_start, time_end;
   clock_gettime( CLOCK_REALTIME, &time_start );

   struct active_audio_player *aap = &vg_audio.active_players[id];
   audio_entity *ent = &aap->ent;

   u32 remaining = count;
   u32 cursor = ent->cur;
   u32 buffer_pos = 0;

   while( remaining )
   {
      u32 samples_this_run = VG_MIN( remaining, ent->length - 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 )
      {
         float *src = &((float *)ent->info.source->data)[ cursor ];
         audio_decode_uncompressed_mono( src, samples_this_run, dst );
      }
      else if( source_mode == k_audio_source_compressed )
      {
         int read_samples = stb_vorbis_get_samples_float_interleaved_stereo( 
               aap->vorbis_handle,
               dst,
               samples_this_run  );

         if( read_samples != samples_this_run )
         {
            vg_warn( "Invalid samples read (%s)\n", ent->info.source->path );
         }
      }

      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 );
         }

         cursor = 0;
         continue;
      }
      else
         break;
   }

   while( remaining )
   {
      buf[ buffer_pos*2 + 0 ] = 0.0f;
      buf[ buffer_pos*2 + 1 ] = 0.0f;
      buffer_pos ++;

      remaining --;
   }

   ent->cur = cursor;

   clock_gettime( CLOCK_REALTIME, &time_end );
   vg_audio.perf_ms_decode += vg_time_diff( time_start, time_end );
}

static void audio_entity_mix( aatree_ptr id, float *buffer, 
                              u32 frame_count )
{
   audio_entity *ent = &vg_audio.active_players[id].ent;

   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 vol = ent->info.vol,
         pan = ent->info.pan;

   audio_entity_get_samples( id, frame_count, pcf );

   struct timespec time_start, time_end;
   clock_gettime( CLOCK_REALTIME, &time_start );

   if( ent->info.flags & AUDIO_FLAG_SPACIAL_3D )
      audio_entity_spacialize( ent, &vol, &pan );

   for( u32 j=0; j<frame_count; j++ )
   {
      float frame_vol = vol;

      if( ent->fadeout )
      {
         /* Force this system to be removed now */
         if( ent->fadeout_current == 0 )
         {
            ent->info.flags = 0x00;
            ent->cur = ent->length;
            break;
         }

         frame_vol *= (float)ent->fadeout_current * fadeout_divisor;
         ent->fadeout_current --;
      }

      float sl = 1.0f-pan,
            sr = 1.0f+pan;

      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;
      
      buffer_pos ++;
   }

   clock_gettime( CLOCK_REALTIME, &time_end );
   vg_audio.perf_ms_mix += vg_time_diff( time_start, time_end );
}

/*
 * callback from miniaudio.h interface
 */
static void audio_mixer_callback( ma_device *pDevice, void *pOutBuf, 
                                  const void *pInput, ma_uint32 frame_count )
{
   struct timespec time_start, time_end;
   clock_gettime( CLOCK_REALTIME, &time_start );

   audio_system_enque();
   
   /* Clear buffer */
   float *pOut32F = (float *)pOutBuf;
   for( int i=0; i<frame_count*2; i ++ )
      pOut32F[i] = 0.0f;

   /* Mix all sounds */
   for( int i=0; i<SFX_MAX_SYSTEMS; i ++ )
   {
      struct active_audio_player *aap = &vg_audio.active_players[i];

      if( aap->active )
         audio_entity_mix( i, pOut32F, frame_count );
   }
   
#if 0
   vg_sleep_ms( 20 );
#endif

   /* redistribute */
   audio_system_cleanup();

   /* TODO: what the hell is this? */
   (void)pInput;
   
   /*
    * Debug information
    */
   clock_gettime( CLOCK_REALTIME, &time_end );

   double elapsed = vg_time_diff( time_start, time_end ),
          budget  = ((double)frame_count / 44100.0) * 1000.0,
          percent = (elapsed/budget) * 100.0;

   snprintf( vg_audio.performance_info, 127, 
                  "%.2fms/%.2fms (%.1f%%) (%u frames)",
                  elapsed, budget, percent, frame_count );

   vg_audio.perf_measurements ++;
   if( vg_audio.perf_measurements >= 30 )
   {
      double ms_decode = vg_audio.perf_ms_decode * (1.0/30.0),
             ms_mix    = vg_audio.perf_ms_mix    * (1.0/30.0);
      
      snprintf( vg_audio.performance_sub0, 63, "Decode %.2fms", ms_decode );
      snprintf( vg_audio.performance_sub1, 63, "mix    %.2fms", ms_mix );

      vg_audio.perf_ms_decode = 0.0;
      vg_audio.perf_ms_mix = 0.0;

      vg_audio.perf_measurements = 0;
   }
}

/* Decompress entire vorbis stream into buffer */
static float *audio_decompress_vorbis( const unsigned char *data, int len, 
                                       int channels, u32 *samples )
{
   int err;
   stb_vorbis *pv = stb_vorbis_open_memory( data, len, &err, NULL );
   
   if( !pv )
   {
      vg_error( "stb_vorbis_open_memory() failed with error code: %i\n", err );
      return NULL;
   }
      
   u32 length_samples = stb_vorbis_stream_length_in_samples( pv );

   vg_info( "decompress_vorbis: %u samples (%.2fs), %.1fkb\n",
               length_samples, 
               (float)length_samples / (44100.0f*(float)channels),
               (float)(length_samples*4*channels) / 1024.0f );
   
   float *buffer = audio_alloc( length_samples * channels * sizeof(float) );
   if( !buffer )
   {
      stb_vorbis_close( pv );
      vg_exit();
   }
   
   int read_samples = stb_vorbis_get_samples_float_interleaved( 
                           pv, channels, buffer, length_samples * channels );

   if( read_samples != length_samples )
   {
      vg_warn( "| warning: sample count mismatch. Expected %u got %i\n", 
                  length_samples, read_samples );
      length_samples = read_samples;
   }
   
   stb_vorbis_close( pv );
   *samples = length_samples;
   return buffer;
}

static int audio_clip_load( audio_clip *clip )
{
   /* Load and decompress */
   i64 file_len;
   void *filedata = vg_asset_read_s( clip->path, &file_len );

   if( !filedata )
   {
      vg_error( "OGG load failed (%s)\n", clip->path );
      return 0;
   }

   if( clip->source_mode == k_audio_source_mono )
   {
      u32 samples;
      float *sound = audio_decompress_vorbis( filedata, file_len, 1, &samples );
      clip->data = sound;
      clip->len = samples;

      float seconds = (float)samples / 44100.0f,
            mb      = (float)(samples*4) / (1024.0f*1024.0f);

      vg_info( "Loaded audio clip[mono] '%s' (%.1fs, %.1fmb)\n", 
                                 clip->path, seconds, mb );
   }
   else if( clip->source_mode == k_audio_source_compressed )
   {
      void *data = audio_alloc( file_len );
      memcpy( data, filedata, file_len );

      clip->data = data;
      clip->len = file_len;

      float mb = (float)(file_len) / (1024.0f*1024.0f);
      vg_info( "Loaded audio clip[compressed] '%s' (%.1fmb)\n", 
                                          clip->path, mb );
   }
   else
   {
      /* ... */

      clip->data = NULL;
      clip->len = 0;

      vg_error( "Unkown source mode (%u)\n", clip->source_mode );
      return 0;
   }

   return 1;
}

static void audio_clip_loadn( audio_clip *arr, int count )
{
   for( int i=0; i<count; i++ )
      audio_clip_load( &arr[i] );
}

/* Mark change to be uploaded through queue system */
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( "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;
}

static void audio_require_init( audio_player *player )
{
   if( player->init )
      return;

   char err[128];
   snprintf( err, 127, "Must init before playing! (%s)\n", player->name );
   vg_exiterr( err );
}

/* Play a clip using player. If its already playing something, it will 
 * fadeout quickly and start the next sound */
static void audio_player_playclip( audio_player *player, audio_clip *clip )
{
   audio_require_lock();
   audio_require_init( player );

   player->info.source = clip;
   audio_player_commit( player );
}

#if 0
static void audio_player_playoneshot( audio_player *player, audio_clip *clip )
{
   audio_require_lock();
   audio_require_init( player );
}
#endif

static void audio_play_oneshot( audio_clip *clip, float volume )
{
   audio_require_lock();

   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;
}

static void audio_player_init( audio_player *player )
{
   player->active_entity = AATREE_PTR_NIL;
   player->init = 1;
}

/*
 * Effects
 */

/*
 * Safety enforced Get/set attributes
 */

static void audio_player_set_position( audio_player *sys, v3f pos )
{
   audio_require_lock();
   v3_copy( pos, sys->info.world_position );
}

static void audio_player_set_vol( audio_player *sys, float vol )
{
   audio_require_lock();
   sys->info.vol = vol;
}

static float audio_player_get_vol( audio_player *sys )
{
   audio_require_lock();
   return sys->info.vol;
}

static void audio_player_set_pan( audio_player *sys, float pan )
{
   audio_require_lock();
   sys->info.pan = pan;
}

static float audio_player_get_pan( audio_player *sys )
{
   audio_require_lock();
   return sys->info.pan;
}

static void audio_player_set_flags( audio_player *sys, u32 flags )
{
   audio_require_lock();
   sys->info.flags = flags;
}

static u32 audio_player_get_flags( audio_player *sys )
{
   audio_require_lock();
   return sys->info.flags;
}


/* 
 * Debugging
 */

static void audio_debug_ui( m4x4f mtx_pv )
{
   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;

   char perf[128],
        psub0[64],
        psub1[64];
        
   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;

      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 );
        }
   strcpy( perf, vg_audio.performance_info );
   strcpy( psub0, vg_audio.performance_sub0 );
   strcpy( psub1, vg_audio.performance_sub1 );

   audio_unlock();

   /* Draw UI */
   ui_global_ctx.cursor[0] = 10;
   ui_global_ctx.cursor[1] = 10;
   ui_global_ctx.cursor[2] = 150;
   ui_global_ctx.cursor[3] = 12;

   ui_text( &ui_global_ctx, ui_global_ctx.cursor, perf, 1, 0 );
   ui_global_ctx.cursor[1] += 20;

   ui_text( &ui_global_ctx, ui_global_ctx.cursor, psub0, 1, 0 );
   ui_global_ctx.cursor[1] += 20;

   ui_text( &ui_global_ctx, ui_global_ctx.cursor, psub1, 1, 0 );
   ui_global_ctx.cursor[1] += 20;
   
   float usage = (float)vg_audio.mem_current / (1024.0f*1024.0f),
         total = (float)vg_audio.mem_total   / (1024.0f*1024.0f),
         percent = (usage/total) * 100.0f;
   snprintf( perf, 127, "Mem: %.1f/%.1fmb (%.1f%%)\n", usage, total, percent );

   ui_text( &ui_global_ctx, ui_global_ctx.cursor, perf, 1, 0 );
   ui_global_ctx.cursor[1] += 20;

   ui_rect overlap_buffer[ SFX_MAX_SYSTEMS ];
   u32 overlap_length = 0;

        /* Draw audio stack */
        for( int i=0; i<num_systems; i ++ )
        {
      struct sound_info *inf = &infos[i];

                ui_global_ctx.cursor[2] = 200;
                ui_global_ctx.cursor[3] = 18;
                
                u32 alpha = 0xa0000000;

                ui_new_node( &ui_global_ctx );
                {               
                        ui_fill_rect( &ui_global_ctx, ui_global_ctx.cursor, 0x00333333|alpha );

                        ui_px baseline = ui_global_ctx.cursor[0],
               w  = 200,
               c  = baseline + ((float)inf->cursor / (float)inf->length) * w;
                        
                        /* cursor */
                        ui_global_ctx.cursor[2] = 2;
                        ui_global_ctx.cursor[0] = c;
                        ui_fill_rect( &ui_global_ctx, ui_global_ctx.cursor, 0xffffffff );
                        
                        ui_global_ctx.cursor[0] = baseline + 2;
                        ui_global_ctx.cursor[1] += 2;
         snprintf( perf, 127, "%s %.1f%%", infos[i].name, infos[i].vol );
                        ui_text( &ui_global_ctx, ui_global_ctx.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;

            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[3] = 17;
            
            for( int j=0; j<12; j++ )
            {
               int collide = 0;
               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])) )
                  {
                     collide = 1;
                     break;
                  }
               }

               if( !collide )
                  break;
               else
                  wr[1] += 18;
            }

            ui_text( &ui_global_ctx, wr, perf, 1, 0 );

            ui_rect_copy( wr, overlap_buffer[ overlap_length ++ ] );
         }
                }

projected_behind:

                ui_end_down( &ui_global_ctx );
                ui_global_ctx.cursor[1] += 1;
        }
}

#endif /* VG_AUDIO_H */