getting stuff working on windows again
[vg.git] / vg_audio.h
index bff400f340b345eed97f81e31c16f0aeffa7245d..d28dd8ddec19338b1f229dcf8f76e177251965cb 100644 (file)
@@ -16,9 +16,6 @@
 #include "vg/vg_profiler.h"
 #include "vg/vg_audio_synth_bird.h"
 
-#include <sys/time.h>
-#include <math.h>
-
 #ifdef __GNUC__
   #ifndef __clang__
     #pragma GCC push_options
 #define AUDIO_LFOS            8
 #define AUDIO_FILTERS         16
 #define AUDIO_FLAG_LOOP       0x1
+#define AUDIO_FLAG_NO_DOPPLER 0x2
 #define AUDIO_FLAG_SPACIAL_3D 0x4
 #define AUDIO_FLAG_AUTO_START 0x8
-
-/* 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
-//#define AUDIO_FLAG_BIRD_SYNTH 0x800
-
 #define AUDIO_FLAG_FORMAT     0x1E00
 
 enum audio_format
@@ -88,17 +79,14 @@ typedef struct audio_clip audio_clip;
 typedef struct audio_channel audio_channel;
 typedef struct audio_lfo audio_lfo;
 
-struct audio_clip
-{
+struct audio_clip{
    const char *path;
    u32 flags;
-
    u32 size;
    void *data;
 };
 
-static struct vg_audio_system
-{
+static struct vg_audio_system{
    SDL_AudioDeviceID sdl_output_device;
 
    void             *audio_pool, 
@@ -108,18 +96,15 @@ static struct vg_audio_system
    /* 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,
@@ -135,9 +120,11 @@ static struct vg_audio_system
    }
    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;           /* ... */
@@ -153,16 +140,14 @@ static struct vg_audio_system
       u32 volume_movement,
           pan_movement;
 
-      union
-      {
+      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,
@@ -177,8 +162,7 @@ static struct vg_audio_system
        * the edit mask tells which to copy into internal _, or to discard
        * ----------------------------------------------------------------------
        */
-      struct channel_state
-      {
+      struct channel_state{
          int   relinquished;
 
          float volume,          /* current volume */
@@ -200,19 +184,20 @@ static struct vg_audio_system
    }
    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               internal_listener_pos,
+                     internal_listener_ears,
+                     internal_listener_velocity,
 
-   v3f               listener_pos,
-                     listener_ears,
-                     listener_velocity;
+                     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"
 
@@ -234,17 +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)
@@ -258,41 +243,28 @@ 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, 
@@ -318,21 +290,17 @@ VG_STATIC void vg_audio_init(void)
    vg_audio.sdl_output_device = 
       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)
@@ -354,37 +322,52 @@ VG_STATIC void vg_audio_free(void)
 #define AUDIO_EDIT_OWNERSHIP      0x40
 #define AUDIO_EDIT_SAMPLING_RATE  0x80
 
-static audio_channel *audio_request_channel( audio_clip *clip, u32 flags )
+static void audio_channel_init( audio_channel *ch, audio_clip *clip, u32 flags )
 {
-   for( int i=0; i<AUDIO_CHANNELS; i++ )
-   {
-      audio_channel *ch = &vg_audio.channels[i];
+   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( !ch->allocated )
-      {
-         ch->source = clip;
-         ch->flags = flags;
-         ch->colour = 0x00333333;
+static void audio_channel_group( audio_channel *ch, u16 group )
+{
+   ch->group = group;
+   ch->colour = (((u32)group * 29986577) & 0x00ffffff) | 0xff000000;
+}
 
-         if( (ch->source->flags & AUDIO_FLAG_FORMAT) == k_audio_format_bird )
-            strcpy( ch->name, "[array]" );
-         else
-            strncpy( ch->name, clip->path, 31 );
-
-         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_world( audio_channel *ch, u8 world_id )
+{
+   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 ){
          return ch;
       }
    }
@@ -392,6 +375,42 @@ static audio_channel *audio_request_channel( audio_clip *clip, u32 flags )
    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 )
@@ -424,13 +443,11 @@ static void audio_channel_set_sampling_rate( audio_channel *ch, float rate )
 static void audio_channel_edit_volume( audio_channel *ch,
                                        float new_volume, int instant )
 {
-   if( instant )
-   {
+   if( instant ){
       ch->editable_state.volume = new_volume;
       ch->editble_state_write_mask |= AUDIO_EDIT_VOLUME;
    }
-   else
-   {
+   else{
       audio_channel_slope_volume( ch, 0.05f, new_volume );
    }
 }
@@ -456,10 +473,12 @@ static audio_channel *audio_channel_crossfade( audio_channel *ch,
    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 )
+   if( replacement ){
+      audio_channel_init( replacement, new_clip, flags );
       audio_channel_fadein( replacement, length );
+   }
 
    return replacement;
 }
@@ -493,9 +512,10 @@ static void audio_channel_set_spacial( audio_channel *ch, v3f co, float range )
 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 ){
+      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 );
@@ -508,9 +528,10 @@ static int audio_oneshot_3d( audio_clip *clip, v3f position,
 
 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 ){
+      audio_channel_init( ch, clip, 0x00 );
       audio_channel_edit_volume( ch, volume, 1 );
       ch = audio_relinquish_channel( ch );
 
@@ -546,8 +567,7 @@ static int audio_channel_load_source( audio_channel *ch )
 {
    u32 format = ch->source->flags & AUDIO_FLAG_FORMAT;
 
-   if( format == k_audio_format_vorbis )
-   {
+   if( format == k_audio_format_vorbis ){
       /* Setup vorbis decoder */
       u32 index = ch - vg_audio.channels;
 
@@ -564,20 +584,17 @@ static int audio_channel_load_source( audio_channel *ch )
             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( format == k_audio_format_bird )
-   {
+   else if( format == k_audio_format_bird ){
       u32 index = ch - vg_audio.channels;
 
       u8 *buf = (u8*)vg_audio.decode_buffer;
@@ -589,12 +606,10 @@ static int audio_channel_load_source( audio_channel *ch )
       ch->bird_handle = loc;
       ch->source_length = synth_bird_get_length_in_samples( loc );
    }
-   else if( format == k_audio_format_stereo )
-   {
+   else if( format == k_audio_format_stereo ){
       ch->source_length = ch->source->size / 2;
    }
-   else
-   {
+   else{
       ch->source_length = ch->source->size;
    }
 
@@ -603,8 +618,7 @@ static int audio_channel_load_source( audio_channel *ch )
 
 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);
    }
@@ -620,15 +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];
       }
@@ -795,14 +807,14 @@ static void audio_channel_get_samples( audio_channel *ch,
 
 static void audio_channel_mix( audio_channel *ch, float *buffer )
 {
-   float framevol_l = 1.0f,
-         framevol_r = 1.0f;
+   float framevol_l = vg_audio.internal_global_volume,
+         framevol_r = vg_audio.internal_global_volume;
 
    float frame_samplerate = ch->_.sampling_rate;
 
    if( ch->flags & AUDIO_FLAG_SPACIAL_3D ){
       v3f delta;
-      v3_sub( ch->_.spacial_falloff, vg_audio.listener_pos, 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 );
@@ -812,24 +824,28 @@ static void audio_channel_mix( audio_channel *ch, float *buffer )
       }
       else{
          v3_muls( delta, 1.0f/dist, delta );
-         float pan = v3_dot( vg_audio.listener_ears, 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);
 
-         const float vs = 100.0f;
-         float doppler = (vs+v3_dot(delta,vg_audio.listener_velocity))/vs;
-               doppler = vg_clampf( doppler, 0.6f, 1.4f );
-               
-         if( fabsf(doppler-1.0f) > 0.01f )
-            frame_samplerate *= doppler;
+         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_exit_loop( "NaN left channel" );
-      if( !vg_validf( framevol_r ) ) vg_fatal_exit_loop( "NaN right channel" );
+      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_exit_loop( "NaN sample rate" );
+         vg_fatal_error( "NaN sample rate" );
    }
 
    u32 buffer_length = AUDIO_MIX_FRAME_SIZE;
@@ -853,22 +869,9 @@ static void audio_channel_mix( audio_channel *ch, float *buffer )
    const float volume_target = ch->_.volume_target;
 
    for( u32 j=0; j<AUDIO_MIX_FRAME_SIZE; j++ ){
-      /*
-       * there is some REALLY weird behaviour with minss,
-       * i cannot begin to guess what the cause is, but the bahaviour when
-       * the second argument is not 1.0 would seemingly tripple or up to 
-       * eight times this routine.
-       *
-       * the times it would happen are when moving from empty space into areas
-       * with geometry. in the bvh for skate rift.
-       *
-       * it should be completely unrelated to this, but somehow -- it is
-       * effecting the speed of minss. and severely at that too.
-       **/
-
       volume_movement += 1.0f;
       float movement_t = volume_movement * inv_volume_rate;
-            movement_t = vg_minf( volume_movement, 1.0f );
+            movement_t = vg_minf( movement_t, 1.0f );
       volume           = vg_lerpf( volume_start, volume_target, movement_t );
 
       float vol_norm = volume * volume;
@@ -881,8 +884,7 @@ static void audio_channel_mix( audio_channel *ch, float *buffer )
             sample_l,
             sample_r;
       
-      if( frame_samplerate != 1.0f )
-      {
+      if( frame_samplerate != 1.0f ){
          /* absolutely garbage resampling, but it will do
           */
 
@@ -895,8 +897,7 @@ static void audio_channel_mix( audio_channel *ch, float *buffer )
          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
-      {
+      else{
          sample_l = pcf[ j*2+0 ];
          sample_r = pcf[ j*2+1 ];
       }
@@ -918,6 +919,13 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count )
     * 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];
 
@@ -1015,8 +1023,7 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count )
       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 = 
@@ -1047,12 +1054,10 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count )
    /*
     * 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
@@ -1070,26 +1075,22 @@ 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;
 
-   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 )
-      {
+      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 )
-         {
+         while( remaining ){
             audio_channel_mix( ch, pOut32F+subpos );
             remaining -= AUDIO_MIX_FRAME_SIZE;
             subpos += AUDIO_MIX_FRAME_SIZE*2;
@@ -1106,8 +1107,7 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count )
 
    audio_lock();
 
-   for( int i=0; i<AUDIO_CHANNELS; i ++ )
-   {
+   for( int i=0; i<AUDIO_CHANNELS; i ++ ){
       audio_channel *ch = &vg_audio.channels[i];
       ch->readable_activity = ch->activity;
    }
@@ -1124,8 +1124,7 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count )
 
    vg_audio.samples_last = frame_count;
 
-   if( vg_audio.debug_ui )
-   {
+   if( vg_audio.debug_dsp ){
       vg_dsp_update_texture();
    }
 
@@ -1143,37 +1142,53 @@ VG_STATIC void audio_clip_load( audio_clip *clip, void *lin_alloc )
    /* TODO: This contains audio_lock() and unlock, but i don't know why
     *       can probably remove them. Low priority to check this */
 
-   if( format == k_audio_format_vorbis )
-   {
+   /* 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( format == k_audio_format_stereo )
-   {
-      vg_fatal_exit_loop( "Unsupported format (Stereo uncompressed)" );
+   else if( format == k_audio_format_stereo ){
+      vg_fatal_error( "Unsupported format (Stereo uncompressed)" );
    }
-   else if( format == k_audio_format_bird )
-   {
-      u32 len = strlen( clip->path ),
-          size = synth_bird_memory_requirement( len );
+   else if( format == k_audio_format_bird ){
+      if( !clip->data ){
+         vg_fatal_error( "No data, external birdsynth unsupported" );
+      }
 
-      if( size > AUDIO_DECODE_SIZE )
-         vg_fatal_exit_loop( "Bird code too long\n" );
+      u32 total_size  = clip->size + sizeof(struct synth_bird);
+          total_size -= sizeof(struct synth_bird_settings);
+          total_size  = vg_align8( total_size );
 
-      clip->size = size;
-      clip->data = vg_linear_alloc( lin_alloc, size );
+      if( total_size > AUDIO_DECODE_SIZE )
+         vg_fatal_error( "Bird coding too long\n" );
 
-      synth_bird_load( clip->data, clip->path, len );
+      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
-   {
+   else{
+      if( !clip->path ){
+         vg_fatal_error( "No path specified, embeded mono unsupported" );
+      }
+
       vg_linear_clear( vg_mem.scratch );
       u32 fsize;
 
@@ -1188,11 +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 */
@@ -1208,7 +1222,7 @@ 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,
@@ -1228,7 +1242,7 @@ VG_STATIC void audio_require_clip_loaded( audio_clip *clip )
       return;
 
    audio_unlock();
-   vg_fatal_exit_loop( "Must load audio clip before playing! \n" );
+   vg_fatal_error( "Must load audio clip before playing! \n" );
 }
 
 /* 
@@ -1253,53 +1267,42 @@ VG_STATIC void audio_debug_ui( m4x4f mtx_pv )
     */
 
    float budget = ((double)vg_audio.samples_last / 44100.0) * 1000.0;
+#if 0
    vg_profile_drawn( (struct vg_profile *[]){ &vg_prof_audio_decode,
                                               &vg_prof_audio_mix,
                                               &vg_prof_audio_dsp}, 3, 
                      budget, (ui_rect){ 4, VG_PROFILE_SAMPLE_COUNT*2 + 8,
                                         512, 0 }, 3 );
+#endif
 
 
    char perf[128];
        
    /* Draw UI */
-   vg_uictx.cursor[0] = 512 + 8;
-   vg_uictx.cursor[1] = VG_PROFILE_SAMPLE_COUNT*2+8+24+12+12;
-   vg_uictx.cursor[2] = 150;
-   vg_uictx.cursor[3] = 12;
-
-   ui_rect view_thing = { 4, vg.window_y-512-4, 512, 512 };
-   ui_push_image( view_thing, vg_dsp.view_texture );
-   
-   float mb1      = 1024.0f*1024.0f,
-         usage    = vg_linear_get_cur( vg_audio.audio_pool )      / mb1,
-         total    = vg_linear_get_capacity( vg_audio.audio_pool ) / mb1,
-         percent  = (usage/total) * 100.0f;
-
-   snprintf( perf, 127, "Mem: %.1f/%.1fmb (%.1f%%)\n", usage, total, percent );
-
-   ui_text( vg_uictx.cursor, perf, 1, 0 );
-   vg_uictx.cursor[1] += 20;
+   ui_rect 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;
       }
 
@@ -1334,8 +1337,9 @@ VG_STATIC void audio_debug_ui( m4x4f mtx_pv )
 
       u32 format_index = (ch->source->flags & AUDIO_FLAG_FORMAT)>>9;
 
-      snprintf( perf, 127, "%02d %c%c%cD %s [%s] %4.2fv'%s'", 
+      snprintf( perf, 127, "%02d[%#04x.%#06x]%c%c%cD %s [%s] %4.2fv'%s'", 
                i,
+               ch->world_id, ch->group,
                (ch->editable_state.relinquished)? 'r': '_',
                0?                                 'r': '_',
                0?                                 '3': '2',
@@ -1344,39 +1348,29 @@ VG_STATIC void audio_debug_ui( m4x4f mtx_pv )
                ch->editable_state.volume,
                ch->name );
 
-      ui_fill_rect( vg_uictx.cursor, 0xa0000000 | ch->colour );
-
-      vg_uictx.cursor[0] += 2;
-      vg_uictx.cursor[1] += 2;
-      ui_text( vg_uictx.cursor, perf, 1, 0 );
-
-               ui_end_down();
-               vg_uictx.cursor[1] += 1;
+      ui_fill( row, 0xa0000000 | ch->colour );
+      ui_text( row, perf, 1, k_ui_align_middle_left, 0 );
       
-      if( AUDIO_FLAG_SPACIAL_3D )
-      {
+      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])) )
@@ -1392,9 +1386,8 @@ 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 ++ ] );
          }
       }
        }