getting stuff working on windows again
[vg.git] / vg_audio.h
index 07c4be09b2d61361c383ba695dfeff70bc674f76..d28dd8ddec19338b1f229dcf8f76e177251965cb 100644 (file)
@@ -1,4 +1,4 @@
-/* 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
@@ -14,9 +14,7 @@
 #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 SFX_MAX_SYSTEMS       32
+#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_ONESHOT    0x2
+#define AUDIO_FLAG_NO_DOPPLER 0x2
 #define AUDIO_FLAG_SPACIAL_3D 0x4
 #define AUDIO_FLAG_AUTO_START 0x8
-#define AUDIO_FLAG_KILL       0x10
-
-#define FADEOUT_LENGTH        1100
-#define FADEOUT_DIVISOR       (1.0f/(float)FADEOUT_LENGTH)
-
-#define AUDIO_DECODE_SIZE     (1024*256)  /* 256 kb decoding buffers */
+#define AUDIO_FLAG_FORMAT     0x1E00
 
-enum audio_source_mode
+enum audio_format
 {
-   k_audio_source_mono,
-   k_audio_source_compressed,
+   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
+#define AUDIO_BASE_VOLUME     1.0f
+
 typedef struct audio_clip audio_clip;
-struct audio_clip
-{
-   const char *path;
-   enum audio_source_mode source_mode;
+typedef struct audio_channel audio_channel;
+typedef struct audio_lfo audio_lfo;
 
+struct audio_clip{
+   const char *path;
+   u32 flags;
    u32 size;
    void *data;
 };
 
-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
-{
-#if 0
-   ma_device         miniaudio_device;
-   ma_device_config  miniaudio_dconfig;
-#endif
+static struct vg_audio_system{
    SDL_AudioDeviceID sdl_output_device;
 
    void             *audio_pool, 
@@ -125,49 +96,121 @@ static struct vg_audio_system
    /* synchro */
    int               sync_locked;
 
-   SDL_mutex        *mux_checker,
-                    *mux_sync;
+   SDL_SpinLock     sl_checker,
+                    sl_sync;
 
-   /* Audio engine, thread 1 */
-   struct active_audio_player
-   {
-      int active;
-      union
-      {
-         audio_entity     ent;
-         aatree_pool_node pool_node;
+   struct audio_lfo{
+      u32 time, time_startframe;
+      float sqrt_polynomial_coefficient;
+
+      struct{
+         enum lfo_wave_type{
+            k_lfo_triangle,
+            k_lfo_square,
+            k_lfo_saw,
+            k_lfo_polynomial_bipolar
+         }
+         wave_type;
+
+         u32   period;
+         float polynomial_coefficient;
+      }
+      _, editable_state;
+      u32 editble_state_write_mask;
+   }
+   oscillators[ AUDIO_LFOS ];
+
+   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 cursor, source_length;
+
+      float volume_movement_start,
+            pan_movement_start;
+
+      u32 volume_movement,
+          pan_movement;
+
+      union{
+         struct synth_bird *bird_handle;
+         stb_vorbis *vorbis_handle;
       };
-      
-      stb_vorbis *vorbis_handle;
+
       stb_vorbis_alloc vorbis_alloc;
+
+      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,
+      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{
+         int   relinquished;
+
+         float volume,          /* current volume */
+               volume_target,   /* target volume */
+               pan,
+               pan_target,
+               sampling_rate;
+
+         u32   volume_rate,
+               pan_rate;
+
+         v4f   spacial_falloff; /* xyz, range */
+         
+         audio_lfo *lfo;
+         float      lfo_amount;
+      }
+      _, editable_state;
+      u32 editble_state_write_mask;
    }
-   active_players[ SFX_MAX_SYSTEMS ];
+   channels[ AUDIO_CHANNELS ];
 
-   aatree            active_pool_info;    /* note: just using the pool */
-   aatree_ptr        active_pool_head;
+   int               debug_ui, debug_ui_3d, debug_dsp;
 
-   /* System queue, and access from thread 0 */
-   audio_entity      entity_queue[SFX_MAX_SYSTEMS];
-   int               queue_len;
-   int               debug_ui, debug_ui_3d;
+   v3f               internal_listener_pos,
+                     internal_listener_ears,
+                     internal_listener_velocity,
 
-   v3f               listener_pos,
-                     listener_ears;
+                     external_listener_pos,
+                     external_listener_ears,
+                     external_lister_velocity;
 
-   float             volume,
-                     volume_target,
-                     volume_target_internal,
-                     volume_console;
+   float             internal_global_volume,
+                     external_global_volume;
 }
-vg_audio = { .volume_console = 1.0f };
+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 
@@ -176,17 +219,17 @@ static struct vg_profile
 VG_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 )
 {
-   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)
@@ -200,57 +243,38 @@ VG_STATIC void audio_require_lock(void)
 
 VG_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)
 {
    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)
 {
-   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, 
                                   VG_MEMORY_SYSTEM );
 
    /* fixed */
-   u32 decode_size = AUDIO_DECODE_SIZE * SFX_MAX_SYSTEMS;
+   u32 decode_size = AUDIO_DECODE_SIZE * AUDIO_CHANNELS;
    vg_audio.decode_buffer = vg_linear_alloc( vg_mem.rtmemory, decode_size );
 
-   /* 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 );
+   vg_dsp_init();
 
    SDL_AudioSpec spec_desired, spec_got;
    spec_desired.callback = audio_mixer_callback;
@@ -258,34 +282,30 @@ VG_STATIC void vg_audio_init(void)
    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)
 {
+   vg_dsp_free();
    SDL_CloseAudioDevice( vg_audio.sdl_output_device );
 }
 
@@ -293,250 +313,312 @@ VG_STATIC void vg_audio_free(void)
  * thread 1
  */
 
-static aatree_ptr audio_alloc_entity_internal(void)
+#define AUDIO_EDIT_VOLUME_SLOPE   0x1
+#define AUDIO_EDIT_VOLUME         0x2
+#define AUDIO_EDIT_LFO_PERIOD     0x4
+#define AUDIO_EDIT_LFO_WAVE       0x8
+#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 )
 {
-   aatree_ptr playerid = aatree_pool_alloc( &vg_audio.active_pool_info, 
-                                            &vg_audio.active_pool_head );
+   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;
+}
 
-   if( playerid == AATREE_PTR_NIL )
-      return AATREE_PTR_NIL;
+static void audio_channel_group( audio_channel *ch, u16 group )
+{
+   ch->group = group;
+   ch->colour = (((u32)group * 29986577) & 0x00ffffff) | 0xff000000;
+}
+
+static void audio_channel_world( audio_channel *ch, u8 world_id )
+{
+   ch->world_id = world_id;
+}
 
-   struct active_audio_player *aap = &vg_audio.active_players[ playerid ];
-   aap->active = 1;
+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 ){
+         return ch;
+      }
+   }
 
-   return playerid;
+   return NULL;
 }
 
-VG_STATIC void audio_entity_free_internal( aatree_ptr id )
+static audio_channel *audio_get_group_idle_channel( u16 group, u32 max_count )
 {
-   struct active_audio_player *aap = &vg_audio.active_players[ id ];
-   aap->active = 0;
+   u32 count = 0;
+   audio_channel *dest = NULL;
 
-   /* Notify player that we've finished */
-   if( aap->ent.player )
-      aap->ent.player->active_entity = AATREE_PTR_NIL;
+   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;
+   }
 
-   /* delete */
-   aatree_pool_free( &vg_audio.active_pool_info, id,
-                     &vg_audio.active_pool_head );
+   return NULL;
 }
 
-VG_STATIC void *audio_entity_vorbis_ptr( aatree_ptr entid )
+static audio_channel *audio_get_group_first_active_channel( u16 group )
 {
-   u8 *buf = (u8*)vg_audio.decode_buffer,
-      *loc = &buf[AUDIO_DECODE_SIZE*entid];
+   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;
+}
 
-   return (void *)loc;
+static int audio_channel_finished( audio_channel *ch )
+{
+   if( ch->readable_activity == k_channel_activity_end )
+      return 1;
+   else
+      return 0;
 }
 
-VG_STATIC void audio_entity_start( audio_entity *src )
+static audio_channel *audio_relinquish_channel( audio_channel *ch )
 {
-   aatree_ptr entid = audio_alloc_entity_internal();
-   if( entid == AATREE_PTR_NIL )
-      return;
+   ch->editable_state.relinquished = 1;
+   ch->editble_state_write_mask |= AUDIO_EDIT_OWNERSHIP;
+   return NULL;
+}
 
-   audio_entity *ent = &vg_audio.active_players[ entid ].ent;
+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;
+}
 
-   ent->info = src->info;
-   ent->name = src->info.source->path;
-   ent->cur  = 0;
-   ent->player = src->player;
+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;
+}
 
-   ent->fadeout = 0;
-   ent->fadeout_current = 0;
+static void audio_channel_edit_volume( audio_channel *ch,
+                                       float new_volume, int instant )
+{
+   if( instant ){
+      ch->editable_state.volume = new_volume;
+      ch->editble_state_write_mask |= AUDIO_EDIT_VOLUME;
+   }
+   else{
+      audio_channel_slope_volume( ch, 0.05f, new_volume );
+   }
+}
+
+static audio_channel *audio_channel_fadeout( audio_channel *ch, float length )
+{
+   audio_channel_slope_volume( ch, length, 0.0f );
+   return audio_relinquish_channel( ch );
+}
+
+static void audio_channel_fadein( audio_channel *ch, float length )
+{
+   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, 
+                                               audio_clip *new_clip,
+                                               float length, u32 flags )
+{
+   u32 cursor = 0;
+
+   if( ch )
+      ch = audio_channel_fadeout( ch, length );
 
-   /* Notify main player we are dequeud and playing */
-   if( src->player )
-   {
-      src->player->enqued = 0;
-      src->player->active_entity = entid;
+   audio_channel *replacement = audio_get_first_idle_channel();
+
+   if( replacement ){
+      audio_channel_init( replacement, new_clip, flags );
+      audio_channel_fadein( replacement, length );
    }
 
-   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
-      };
+   return replacement;
+}
 
-      int err;
-      stb_vorbis *decoder = stb_vorbis_open_memory( 
-            src->info.source->data,
-            src->info.source->size, &err, &alloc );
+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;
+}
 
-      if( !decoder )
-      {
-         vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n", 
-                     src->info.source->path, err );
+static void audio_channel_set_spacial( audio_channel *ch, v3f co, float range )
+{
+   if( ch->flags & AUDIO_FLAG_SPACIAL_3D ){
+      v3_copy( co, ch->editable_state.spacial_falloff );
 
-         audio_entity_free_internal( entid );
-         return;
-      }
+      if( range == 0.0f )
+         ch->editable_state.spacial_falloff[3] = 1.0f;
       else
-      {
-         ent->length = stb_vorbis_stream_length_in_samples( decoder );
-      }
-      
-      aap->vorbis_handle = decoder;
+         ch->editable_state.spacial_falloff[3] = 1.0f/range;
+
+      ch->editble_state_write_mask |= AUDIO_EDIT_SPACIAL;
    }
-   else
-   {
-      ent->length = src->info.source->size;
+   else{
+      vg_warn( "Tried to set spacialization paramaters for 2D channel (%s)\n",
+               ch->name );
    }
 }
 
-/*
- * Read everything from the queue 
- */
-VG_STATIC void audio_system_enque(void)
+static int audio_oneshot_3d( audio_clip *clip, v3f position, 
+                             float range, float volume )
 {
-   /* Process incoming sound queue */
-   audio_lock();
+   audio_channel *ch = audio_get_first_idle_channel();
 
-   vg_audio.volume_target_internal = vg_audio.volume_target;
-   
-   int wr = 0;
-   for( int i=0; i<vg_audio.queue_len; i++ )
-   {
-      audio_entity *src = &vg_audio.entity_queue[ i ];
+   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 );
 
-      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 );
-      }
+      return 1;
    }
-   
-   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;
+   else
+      return 0;
+}
 
-      if( aap->ent.player )
-      {
-         /* Only copy information in whilst not requeing */
-         if( aap->ent.player->enqued == 0 )
-         {
-            aap->ent.info = aap->ent.player->info;
+static int audio_oneshot( audio_clip *clip, float volume, float pan )
+{
+   audio_channel *ch = audio_get_first_idle_channel();
 
-            if( (aap->ent.info.flags & AUDIO_FLAG_KILL) && !aap->ent.fadeout )
-            {
-               aap->ent.fadeout = FADEOUT_LENGTH;
-               aap->ent.fadeout_current = FADEOUT_LENGTH;
-            }
-         }
-      }
+   if( ch ){
+      audio_channel_init( ch, clip, 0x00 );
+      audio_channel_edit_volume( ch, volume, 1 );
+      ch = audio_relinquish_channel( ch );
+
+      return 1;
    }
-   
-   audio_unlock();
+   else
+      return 0;
 }
 
-/* 
- * Redistribute sound systems 
- */
-VG_STATIC void audio_system_cleanup(void)
+static void audio_set_lfo_wave( int id, enum lfo_wave_type type, 
+                                float coefficient )
 {
-   audio_lock();
+   audio_lfo *lfo = &vg_audio.oscillators[ id ];
+   lfo->editable_state.polynomial_coefficient = coefficient;
+   lfo->editable_state.wave_type = type;
 
-   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 );
-         }
-      }
-   }
+   lfo->editble_state_write_mask |= AUDIO_EDIT_LFO_WAVE;
+}
 
-   audio_unlock();
+static void audio_set_lfo_frequency( int id, float freq )
+{
+   audio_lfo *lfo = &vg_audio.oscillators[ id ];
+   lfo->editable_state.period = 44100.0f / freq;
+   lfo->editble_state_write_mask |= AUDIO_EDIT_LFO_PERIOD;
 }
 
+
 /* 
- * Get effective volume and pan from this entity
+ * Committers
+ * -----------------------------------------------------------------------------
  */
-VG_STATIC void audio_entity_spacialize( audio_entity *ent, 
-                                        float *vol, float *pan )
+static int audio_channel_load_source( audio_channel *ch )
 {
-   if( ent->info.vol < 0.01f )
-   {
-      *vol = ent->info.vol;
-      *pan = 0.0f;
-      return;
-   }
+   u32 format = ch->source->flags & AUDIO_FLAG_FORMAT;
 
-   if( !vg_validf(vg_audio.listener_pos[0]) ||
-       !vg_validf(vg_audio.listener_pos[1]) ||
-       !vg_validf(vg_audio.listener_pos[2]) ||
-       !vg_validf(ent->info.world_position[0]) ||
-       !vg_validf(ent->info.world_position[1]) ||
-       !vg_validf(ent->info.world_position[2]) )
-   {
-      vg_error( "NaN listener/world position (%s)\n", ent->name );
-      *vol = 0.0f;
-      *pan = 0.0f;
-      return;
-   }
+   if( format == k_audio_format_vorbis ){
+      /* Setup vorbis decoder */
+      u32 index = ch - vg_audio.channels;
 
-   v3f delta;
-   v3_sub( ent->info.world_position, vg_audio.listener_pos, delta );
+      u8 *buf = (u8*)vg_audio.decode_buffer,
+         *loc = &buf[AUDIO_DECODE_SIZE*index];
 
-   float dist2 = v3_length2( delta );
+      stb_vorbis_alloc alloc = {
+         .alloc_buffer = (char *)loc,
+         .alloc_buffer_length_in_bytes = AUDIO_DECODE_SIZE
+      };
+
+      int err;
+      stb_vorbis *decoder = stb_vorbis_open_memory( 
+            ch->source->data,
+            ch->source->size, &err, &alloc );
 
-   if( dist2 < 0.0001f )
-   {
-      *pan = 0.0f;
-      *vol = 1.0f;
+      if( !decoder ){
+         vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n", 
+                     ch->source->path, err );
+         return 0;
+      }
+      else{
+         ch->source_length = stb_vorbis_stream_length_in_samples( decoder );
+         ch->vorbis_handle = decoder;
+      }
    }
-   else
-   {
-      float dist = sqrtf( dist2 ),
-            attn = (dist / ent->info.vol) +1.0f;
+   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];
 
-      v3_muls( delta, 1.0f/dist, delta );
-      *pan = v3_dot( vg_audio.listener_ears, delta );
-      *vol = 1.0f/(attn*attn);
+      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{
+      ch->source_length = ch->source->size;
+   }
+
+   return 1;
 }
 
 VG_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);
    }
@@ -552,15 +634,13 @@ stb_vorbis_get_samples_float_interleaved_stereo( stb_vorbis *f, float *buffer,
    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];
       }
@@ -587,15 +667,13 @@ 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];
 
@@ -616,65 +694,107 @@ stb_vorbis_get_samples_i16_downmixed( stb_vorbis *f, i16 *buffer, int len )
    return n;
 }
 
-VG_STATIC void audio_entity_get_samples( aatree_ptr id, u32 count, float *buf )
+static inline float audio_lfo_pull_sample( audio_lfo *lfo )
 {
-   vg_profile_begin( &_vg_prof_audio_decode );
+   lfo->time ++;
+
+   if( lfo->time >= lfo->_.period )
+      lfo->time = 0;
+
+   float t  = lfo->time;
+         t /= (float)lfo->_.period;
+
+   if( lfo->_.wave_type == k_lfo_polynomial_bipolar ){
+      /*
+       *           #
+       *          # #
+       *          # #
+       *          #  #
+       * ###     #    ###
+       *    ##   #
+       *      #  #
+       *       # #
+       *       ##
+       */           
+
+      t *= 2.0f;
+      t -= 1.0f;
+
+      return (( 2.0f * lfo->sqrt_polynomial_coefficient * t ) /
+              /* --------------------------------------- */
+               ( 1.0f + lfo->_.polynomial_coefficient * t*t )
+              
+             ) * (1.0f-fabsf(t));
+   }
+   else{
+      return 0.0f;
+   }
+}
 
-   struct active_audio_player *aap = &vg_audio.active_players[id];
-   audio_entity *ent = &aap->ent;
+static void audio_channel_get_samples( audio_channel *ch, 
+                                       u32 count, float *buf )
+{
+   vg_profile_begin( &_vg_prof_audio_decode );
 
    u32 remaining = count;
-   u32 cursor = ent->cur;
    u32 buffer_pos = 0;
 
-   while( remaining )
-   {
-      u32 samples_this_run = VG_MIN( remaining, ent->length - 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 ]; 
-
-      int source_mode = ent->info.source->source_mode;
       
-      if( source_mode == k_audio_source_mono )
-      {
-         i16 *src_buffer = ent->info.source->data,
-             *src = &src_buffer[cursor];
-
-         audio_decode_uncompressed_mono( src, samples_this_run, dst );
+      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( source_mode == k_audio_source_compressed )
-      {
+      else if( format == k_audio_format_vorbis ){
          int read_samples = stb_vorbis_get_samples_float_interleaved_stereo( 
-               aap->vorbis_handle,
+               ch->vorbis_handle,
                dst,
-               samples_this_run  );
+               samples_this_run );
 
-         if( read_samples != samples_this_run )
-         {
-            vg_warn( "Invalid samples read (%s)\n", ent->info.source->path );
+         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++ ){
+               dst[i*2+0] = 0.0f;
+               dst[i*2+1] = 0.0f;
+            }
          }
       }
+      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];
+
+         audio_decode_uncompressed_mono( src, samples_this_run, dst );
+      }
 
-      cursor += samples_this_run;
+      ch->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 );
-         }
+      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 );
 
-         cursor = 0;
+         ch->cursor = 0;
          continue;
       }
       else
          break;
    }
 
-   while( remaining )
-   {
+   while( remaining ){
       buf[ buffer_pos*2 + 0 ] = 0.0f;
       buf[ buffer_pos*2 + 1 ] = 0.0f;
       buffer_pos ++;
@@ -682,64 +802,272 @@ VG_STATIC void audio_entity_get_samples( aatree_ptr id, u32 count, float *buf )
       remaining --;
    }
 
-   ent->cur = cursor;
    vg_profile_end( &_vg_prof_audio_decode );
 }
 
-VG_STATIC void audio_entity_mix( aatree_ptr id, float *buffer, 
-                                u32 frame_count )
+static void audio_channel_mix( audio_channel *ch, float *buffer )
 {
-   audio_entity *ent = &vg_audio.active_players[id].ent;
+   float framevol_l = vg_audio.internal_global_volume,
+         framevol_r = vg_audio.internal_global_volume;
 
-   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 frame_samplerate = ch->_.sampling_rate;
+
+   if( ch->flags & AUDIO_FLAG_SPACIAL_3D ){
+      v3f delta;
+      v3_sub( ch->_.spacial_falloff, vg_audio.internal_listener_pos, delta );
+
+      float dist = v3_length( delta ),
+            vol  = vg_maxf( 0.0f, 1.0f - ch->_.spacial_falloff[3]*dist );
+
+      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;
+         }
+      }
+
+      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" );
+   }
+
+   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 vol = ent->info.vol,
-         pan = ent->info.pan;
+   float pcf[ AUDIO_MIX_FRAME_SIZE * 2 * 2 ];
 
-   audio_entity_get_samples( id, frame_count, pcf );
+   audio_channel_get_samples( ch, buffer_length, pcf );
 
    vg_profile_begin( &_vg_prof_audio_mix );
 
-   if( ent->info.flags & AUDIO_FLAG_SPACIAL_3D )
-      audio_entity_spacialize( ent, &vol, &pan );
+   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;
 
-   for( u32 j=0; j<frame_count; j++ )
-   {
-      float frame_vol = vol * vg_audio.volume;
-      if( ent->fadeout )
-      {
-         /* Force this system to be removed now */
-         if( ent->fadeout_current == 0 )
-         {
-            ent->info.flags = 0x00;
-            ent->cur = ent->length;
-            break;
-         }
+   float volume = ch->_.volume;
+   const float volume_start  = ch->volume_movement_start;
+   const float volume_target = ch->_.volume_target;
 
-         frame_vol *= (float)ent->fadeout_current * fadeout_divisor;
-         ent->fadeout_current --;
-      }
+   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 = volume * volume;
 
-      float sl = 1.0f-pan,
-            sr = 1.0f+pan;
+      if( ch->_.lfo )
+         vol_norm *= 1.0f + audio_lfo_pull_sample(ch->_.lfo) * ch->_.lfo_amount;
 
-      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;
+      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 )
 {
-   audio_system_enque();
+   /*
+    * Copy data and move edit flags to commit flags
+    * ------------------------------------------------------------- */
+   audio_lock();
+   
+   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->activity == k_channel_activity_end)
+            || (ch->_.volume == 0.0f)
+            || (ch->activity == k_channel_activity_error) )
+         {
+            ch->_.relinquished = 0;
+            ch->allocated = 0;
+            ch->activity = k_channel_activity_reset;
+            continue;
+         }
+      }
+
+      /* process new channels */
+      if( ch->activity == k_channel_activity_reset ){
+         ch->_ = ch->editable_state;
+         ch->cursor = 0;
+         ch->source_length = 0;
+         ch->activity = k_channel_activity_wake;
+      }
+
+      if( ch->editble_state_write_mask & AUDIO_EDIT_OWNERSHIP )
+         ch->_.relinquished = ch->editable_state.relinquished;
+      else
+         ch->editable_state.relinquished = ch->_.relinquished;
+
+
+      if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME ){
+         ch->_.volume = ch->editable_state.volume;
+         ch->_.volume_target = ch->editable_state.volume;
+      }
+      else{
+         ch->editable_state.volume = ch->_.volume;
+      }
+      
+
+      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{
+         ch->editable_state.volume_target = ch->_.volume_target;
+         ch->editable_state.volume_rate   = ch->_.volume_rate;
+      }
+
+
+      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{
+         ch->editable_state.lfo        = ch->_.lfo;
+         ch->editable_state.lfo_amount = ch->_.lfo_amount;
+      }
+
+
+      if( ch->editble_state_write_mask & AUDIO_EDIT_SPACIAL )
+         v4_copy( ch->editable_state.spacial_falloff,ch->_.spacial_falloff );
+      else
+         v4_copy( ch->_.spacial_falloff,ch->editable_state.spacial_falloff );
+
+
+      /* currently readonly, i guess */
+      ch->editable_state.pan_target = ch->_.pan_target;
+      ch->editable_state.pan        = ch->_.pan;
+      ch->editble_state_write_mask  = 0x00;
+   }
+
+   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 ){
+         lfo->_.wave_type = lfo->editable_state.wave_type;
+
+         if( lfo->_.wave_type == k_lfo_polynomial_bipolar ){
+            lfo->_.polynomial_coefficient = 
+               lfo->editable_state.polynomial_coefficient;
+            lfo->sqrt_polynomial_coefficient = 
+               sqrtf(lfo->_.polynomial_coefficient);
+         }
+      }
+
+      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{
+            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++ ){
+      audio_channel *ch = &vg_audio.channels[i];
+
+      if( ch->activity == k_channel_activity_wake ){
+         if( audio_channel_load_source( ch ) )
+            ch->activity = k_channel_activity_alive;
+         else
+            ch->activity = k_channel_activity_error;
+      }
+   }
+
+   /*
+    * Mix everything 
+    * -------------------------------------------------------- */
    int frame_count = byte_count/(2*sizeof(float));
    
    /* Clear buffer */
@@ -747,39 +1075,59 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count )
    for( int i=0; i<frame_count*2; i ++ )
       pOut32F[i] = 0.0f;
 
-   float start_vol = vg_audio.volume;
+   for( int i=0; i<AUDIO_LFOS; i++ ){
+      audio_lfo *lfo = &vg_audio.oscillators[i];
+      lfo->time_startframe = lfo->time;
+   }
 
-   /* Mix all sounds */
-   for( int i=0; i<SFX_MAX_SYSTEMS; i ++ )
-   {
-      struct active_audio_player *aap = &vg_audio.active_players[i];
-      vg_audio.volume = start_vol;
+   for( int i=0; i<AUDIO_CHANNELS; i ++ ){
+      audio_channel *ch = &vg_audio.channels[i];
 
-      if( aap->active )
-         audio_entity_mix( i, 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;
+         }
+      }
    }
 
-   float vol_diff = vg_audio.volume_target_internal - vg_audio.volume,
-         vol_rate = 1.0f / (44100.0f*0.25f),
-         vol_chg  = frame_count * vol_rate;
-   
-   if( vol_chg > fabsf( vol_diff ) )
-      vg_audio.volume = vg_audio.volume_target_internal;
-   else
-      vg_audio.volume += vg_signf( vol_diff ) * vol_chg;
+   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 );
 
-   /* redistribute */
-   audio_system_cleanup();
-   
    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();
 }
 
@@ -788,8 +1136,59 @@ VG_STATIC void audio_clip_load( audio_clip *clip, void *lin_alloc )
    if( lin_alloc == NULL )
       lin_alloc = vg_audio.audio_pool;
 
-   if( clip->source_mode == k_audio_source_mono )
-   {
+   /* load in directly */
+   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_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( format == k_audio_format_stereo ){
+      vg_fatal_error( "Unsupported format (Stereo uncompressed)" );
+   }
+   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;
 
@@ -804,11 +1203,10 @@ VG_STATIC void audio_clip_load( audio_clip *clip, void *lin_alloc )
       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 */
@@ -824,26 +1222,12 @@ VG_STATIC void audio_clip_load( audio_clip *clip, void *lin_alloc )
                               decoder, clip->data, length_samples );
 
       if( read_samples != length_samples )
-         vg_fatal_exit_loop( "Decode error" );
+         vg_fatal_error( "Decode error" );
 
       float mb = (float)(data_size) / (1024.0f*1024.0f);
       vg_info( "Loaded audio clip '%s' (%.1fmb) %u samples\n", clip->path, mb,
                length_samples );
    }
-
-   /* load in directly */
-   else if( clip->source_mode == k_audio_source_compressed )
-   {
-      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" );
-
-      float mb = (float)(clip->size) / (1024.0f*1024.0f);
-      vg_info( "Loaded audio clip '%s' (%.1fmb)\n", clip->path, mb );
-   }
 }
 
 VG_STATIC void audio_clip_loadn( audio_clip *arr, int count, void *lin_alloc )
@@ -852,182 +1236,13 @@ VG_STATIC void audio_clip_loadn( audio_clip *arr, int count, void *lin_alloc )
       audio_clip_load( &arr[i], lin_alloc );
 }
 
-/* Mark change to be uploaded through queue system */
-VG_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( "[2] 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;
-}
-
-VG_STATIC void audio_require_init( audio_player *player )
-{
-   if( player->init )
-      return;
-
-   audio_unlock();
-   vg_fatal_exit_loop( "Must init audio player before playing! \n" );
-}
-
 VG_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" );
-}
-
-/* Play a clip using player. If its already playing something, it will 
- * fadeout quickly and start the next sound */
-VG_STATIC void audio_player_playclip( audio_player *player, audio_clip *clip )
-{
-   audio_require_lock();
-   audio_require_init( player );
-   audio_require_clip_loaded( clip );
-
-   if( player->info.flags & AUDIO_FLAG_KILL )
-   {
-      vg_error( "Can't start audio clip on player that is/has disconnected" );
-      return;
-   }
-
-   if( player->enqued )
-   {
-      vg_warn( "[1] Audio commit spamming; already enqued (%s)\n", 
-               player->name );
-      return;
-   }
-
-   player->info.source = clip;
-   audio_player_commit( player );
-}
-
-VG_STATIC void audio_play_oneshot( audio_clip *clip, float volume )
-{
-   audio_require_lock();
-   audio_require_clip_loaded( clip );
-
-   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;
-}
-
-VG_STATIC void audio_player_init( audio_player *player )
-{
-   player->active_entity = AATREE_PTR_NIL;
-   player->init = 1;
-}
-
-/*
- * Effects
- */
-
-/*
- * Safety enforced Get/set attributes
- */
-
-VG_STATIC int audio_player_is_playing( audio_player *sys )
-{
-   audio_require_lock();
-
-   if( sys->active_entity != AATREE_PTR_NIL )
-      return 1;
-   else 
-      return 0;
-}
-
-VG_STATIC void audio_player_set_position( audio_player *sys, v3f pos )
-{
-   audio_require_lock();
-   v3_copy( pos, sys->info.world_position );
-}
-
-VG_STATIC void audio_player_set_vol( audio_player *sys, float vol )
-{
-   audio_require_lock();
-
-   if( !vg_validf(vol) )
-   {
-      vg_warn( "NaN volume (%s)\n", sys->name );
-      vol = 0.0f;
-   }
-
-   if( (vol < 0.0f) || (vol > 100.0f) )
-   {
-      vg_warn( "Invalid volume (%s: %f)\n", sys->name, vol );
-      vol = 0.0f;
-   }
-
-   sys->info.vol = vol;
-}
-
-VG_STATIC float audio_player_get_vol( audio_player *sys )
-{
-   audio_require_lock();
-   return sys->info.vol;
-}
-
-VG_STATIC void audio_player_set_pan( audio_player *sys, float pan )
-{
-   audio_require_lock();
-   sys->info.pan = pan;
-}
-
-VG_STATIC float audio_player_get_pan( audio_player *sys )
-{
-   audio_require_lock();
-   return sys->info.pan;
-}
-
-VG_STATIC void audio_player_set_flags( audio_player *sys, u32 flags )
-{
-   audio_require_lock();
-   sys->info.flags = flags;
-}
-
-VG_STATIC u32 audio_player_get_flags( audio_player *sys )
-{
-   audio_require_lock();
-   return sys->info.flags;
-}
-
-VG_STATIC void audio_set_master_vol( float vol )
-{
-   audio_require_lock();
-   vg_audio.volume_target = vol;
-}
-
-VG_STATIC void audio_push_console_vol(void)
-{
-   audio_lock();
-   audio_set_master_vol( vg_audio.volume_console );
-   audio_unlock();
+   vg_fatal_error( "Must load audio clip before playing! \n" );
 }
 
 /* 
@@ -1039,118 +1254,123 @@ VG_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;
-
    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;
+   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 );
 
-      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 );
-       }
+   /* 
+    * Profiler
+    * -----------------------------------------------------------------------
+    */
 
    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 }, 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 );
+#endif
 
-   audio_unlock();
 
    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[ SFX_MAX_SYSTEMS ];
+   ui_rect overlap_buffer[ AUDIO_CHANNELS ];
    u32 overlap_length = 0;
 
        /* Draw audio stack */
-       for( int i=0; i<num_systems; i ++ )
-       {
-      struct sound_info *inf = &infos[i];
-
-               vg_uictx.cursor[2] = 200;
-               vg_uictx.cursor[3] = 18;
-               
-               u32 alpha = 0xa0000000;
-
-               ui_new_node();
-               {               
-                       ui_fill_rect( vg_uictx.cursor, 0x00333333|alpha );
-
-                       ui_px baseline = vg_uictx.cursor[0],
-               w  = 200,
-               c  = baseline + ((float)inf->cursor / (float)inf->length) * w;
-                       
-                       /* cursor */
-                       vg_uictx.cursor[2] = 2;
-                       vg_uictx.cursor[0] = c;
-                       ui_fill_rect( vg_uictx.cursor, 0xffffffff );
-                       
-                       vg_uictx.cursor[0] = baseline + 2;
-                       vg_uictx.cursor[1] += 2;
-         snprintf( perf, 127, "%s %.1f%%", infos[i].name, infos[i].vol );
-                       ui_text( vg_uictx.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 );
+       for( int i=0; i<AUDIO_CHANNELS; i ++ ){
+      audio_channel *ch = &vg_audio.channels[i];
+
+      ui_rect row;
+      ui_split( window, k_ui_axis_h, 18, 1, row, window );
+
+      if( !ch->allocated ){
+         ui_fill( row, 0x50333333 );
+         continue;
+      }
 
-            if( wpos[3] <= 0.0f )
-               goto projected_behind;
+      const char *formats[] =
+      {
+         "   mono   ",
+         "  stereo  ", 
+         "  vorbis  ",
+         "   none0  ",
+         "   none1  ",
+         "   none2  ",
+         "   none3  ",
+         "   none4  ",
+         "synth:bird",
+         "   none5  ",
+         "   none6  ",
+         "   none7  ",
+         "   none8  ",
+         "   none9  ",
+         "  none10  ",
+         "  none11  ",
+      };
+
+      const char *activties[] =
+      {
+         "reset",
+         "wake ",
+         "alive",
+         "end  ",
+         "error"
+      };
+
+      u32 format_index = (ch->source->flags & AUDIO_FLAG_FORMAT)>>9;
+
+      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',
+               formats[format_index],
+               activties[ch->readable_activity],
+               ch->editable_state.volume,
+               ch->name );
+
+      ui_fill( row, 0xa0000000 | ch->colour );
+      ui_text( row, perf, 1, k_ui_align_middle_left, 0 );
+      
+      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 ){
             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])) )
@@ -1166,17 +1386,13 @@ VG_STATIC void audio_debug_ui( m4x4f mtx_pv )
                   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 ++ ] );
          }
-               }
-
-projected_behind:
-
-               ui_end_down();
-               vg_uictx.cursor[1] += 1;
+      }
        }
+
+   audio_unlock();
 }
 
 #endif /* VG_AUDIO_H */