resampling rate

[vg.git] / vg_audio.h

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

#ifndef VG_AUDIO_H
#define VG_AUDIO_H

#define VG_GAME

#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 "vg/vg_profiler.h"

#include <sys/time.h>
#include <math.h>

#ifdef __GNUC__
  #ifndef __clang__
    #pragma GCC push_options
    #pragma GCC optimize ("O3")
    #pragma GCC diagnostic push
    #pragma GCC diagnostic ignored "-Wdeprecated-declarations"
  #endif
#endif

#define STB_VORBIS_MAX_CHANNELS 2
#include "submodules/stb/stb_vorbis.c"
#undef L
#undef R
#undef C

#ifdef __GNUC__
  #ifndef __clang__
    #pragma GCC pop_options
    #pragma GCC diagnostic pop 
  #endif
#endif

#define AUDIO_CHANNELS        32
#define AUDIO_LFOS            8
#define AUDIO_FLAG_LOOP       0x1
#define AUDIO_FLAG_SPACIAL_3D 0x2

/* 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_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;
typedef struct audio_channel audio_channel;
typedef struct audio_lfo audio_lfo;

struct audio_clip
{
   const char *path;
   u32 flags;

   u32 size;
   void *data;
};

static struct vg_audio_system
{
   SDL_AudioDeviceID sdl_output_device;

   void             *audio_pool, 
                    *decode_buffer;
   u32               samples_last;

   /* synchro */
   int               sync_locked;

   SDL_mutex        *mux_checker,
                    *mux_sync;

   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;
      char name[32];       /* only editable while allocated == 0 */
      audio_clip *source;  /* ... */
      u32 flags;           /* ... */

      /* internal non-readable state 
       * -----------------------------*/
      u32 cursor, source_length;

      float volume_movement_start,
            pan_movement_start;

      u32 volume_movement,
          pan_movement;

      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_error
      }
      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;
   }
   channels[ AUDIO_CHANNELS ];

   /* System queue, and access from thread 0 */
   int               debug_ui, debug_ui_3d;

   v3f               listener_pos,
                     listener_ears;

   float             volume,
                     volume_target,
                     volume_target_internal,
                     volume_console;
}
vg_audio = { .volume_console = 1.0f };


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_audio_decode,
   vg_prof_audio_mix;

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

VG_STATIC void audio_lock_checker_store( int value )
{
   SDL_LockMutex( vg_audio.mux_checker );
   vg_audio.sync_locked = value;
   SDL_UnlockMutex( vg_audio.mux_checker );
}

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

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

VG_STATIC void audio_lock(void)
{
   SDL_LockMutex( vg_audio.mux_sync );
   audio_lock_checker_store(1);
}

VG_STATIC void audio_unlock(void)
{
   audio_lock_checker_store(0);
   SDL_UnlockMutex( vg_audio.mux_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
   });

   /* 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 * AUDIO_CHANNELS;
   vg_audio.decode_buffer = vg_linear_alloc( vg_mem.rtmemory, decode_size );

   SDL_AudioSpec spec_desired, spec_got;
   spec_desired.callback = audio_mixer_callback;
   spec_desired.channels = 2;
   spec_desired.format   = AUDIO_F32;
   spec_desired.freq     = 44100;
   spec_desired.padding  = 0;
   spec_desired.samples  = 512;
   spec_desired.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 );

   if( vg_audio.sdl_output_device )
   {
      SDL_PauseAudioDevice( vg_audio.sdl_output_device, 0 );
   }
   else
   {
      vg_fatal_exit_loop( 
         "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)
{
   SDL_CloseAudioDevice( vg_audio.sdl_output_device );
}

/* 
 * thread 1
 */

#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 audio_channel *audio_request_channel( audio_clip *clip, u32 flags )
{
   for( int i=0; i<AUDIO_CHANNELS; i++ )
   {
      audio_channel *ch = &vg_audio.channels[i];

      if( !ch->allocated )
      {
         ch->source = clip;
         ch->flags = flags;
         strcpy( ch->name, clip->path );

         ch->allocated = 1;

         ch->editable_state.relinquished = 0;
         ch->editable_state.volume = 1.0f;
         ch->editable_state.volume_target = 1.0f;
         ch->editable_state.pan = 0.0f;
         ch->editable_state.pan_target = 0.0f;
         ch->editable_state.volume_rate = 0;
         ch->editable_state.pan_rate = 0;
         v4_copy((v4f){0.0f,0.0f,0.0f,1.0f},ch->editable_state.spacial_falloff);
         ch->editable_state.lfo = NULL;
         ch->editable_state.lfo_amount = 0.0f;
         ch->editable_state.sampling_rate = 1.0f;
         ch->editble_state_write_mask = 0x00;
         return ch;
      }
   }

   return NULL;
}

static audio_channel *audio_relinquish_channel( audio_channel *ch )
{
   ch->editable_state.relinquished = 1;
   ch->editble_state_write_mask |= AUDIO_EDIT_OWNERSHIP;
   return NULL;
}

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;
}

static void audio_channel_set_sampling_rate( audio_channel *ch, float rate )
{
   ch->editable_state.sampling_rate = rate;
   ch->editble_state_write_mask |= AUDIO_EDIT_SAMPLING_RATE;
}

static void audio_channel_edit_volume( audio_channel *ch,
                                       float new_volume, int instant )
{
   if( instant )
   {
      ch->editable_state.volume = 0.0f;
      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 );

   audio_channel *replacement = audio_request_channel( new_clip, flags );

   if( replacement )
      audio_channel_fadein( replacement, length );

   return replacement;
}

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;
}

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 );
      ch->editable_state.spacial_falloff[3] = 1.0f/range;
      ch->editble_state_write_mask |= AUDIO_EDIT_SPACIAL;
   }
   else
   {
      vg_warn( "Tried to set spacialization paramaters for 2D channel (%s)\n",
               ch->name );
   }
}

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 );

   if( ch )
   {
      audio_channel_set_spacial( ch, position, range );
      audio_channel_edit_volume( ch, volume, 1 );
      ch = audio_relinquish_channel( ch );

      return 1;
   }
   else
      return 0;
}

static int audio_oneshot( audio_clip *clip, float volume, float pan )
{
   audio_channel *ch = audio_request_channel( clip, 0x00 );

   if( ch )
   {
      audio_channel_edit_volume( ch, volume, 1 );
      ch = audio_relinquish_channel( ch );

      return 1;
   }
   else
      return 0;
}

static void audio_set_lfo_wave( int id, enum lfo_wave_type type, 
                                float coefficient )
{
   audio_lfo *lfo = &vg_audio.oscillators[ id ];
   lfo->editable_state.polynomial_coefficient = coefficient;
   lfo->editable_state.wave_type = type;

   lfo->editble_state_write_mask |= AUDIO_EDIT_LFO_WAVE;
}

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;
}

/* 
 * Committers
 * -----------------------------------------------------------------------------
 */
static int audio_channel_load_source( audio_channel *ch )
{
   if( ch->source->flags & AUDIO_FLAG_VORBIS )
   {
      /* Setup vorbis decoder */
      u32 index = ch - vg_audio.channels;

      u8 *buf = (u8*)vg_audio.decode_buffer,
         *loc = &buf[AUDIO_DECODE_SIZE*index];

      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( !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 if( ch->source->flags & AUDIO_FLAG_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++ )
   {
      dst[ i*2 + 0 ] = ((float)src[i]) * (1.0f/32767.0f);
      dst[ i*2 + 1 ] = ((float)src[i]) * (1.0f/32767.0f);
   }
}

/* 
 * adapted from stb_vorbis.h, since the original does not handle mono->stereo
 */
VG_STATIC int 
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;
}

/* 
 * ........ more wrecked code sorry!
 */
VG_STATIC int 
stb_vorbis_get_samples_i16_downmixed( stb_vorbis *f, i16 *buffer, int len )
{
   int n = 0,
       c = VG_MIN( 1, f->channels - 1 );

   while( n < len ) 
   {
      int k = f->channel_buffer_end - f->channel_buffer_start;

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

      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];

         *buffer++ = vg_clampf( 0.5f*(sl+sr), -1.0f, 1.0f ) * 32767.0f;
         //*buffer++ = vg_clampf( sr, -1.0f, 1.0f ) * 32767.0f;
      }

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

      if( n == len )
         break;

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

   return n;
}

static float audio_lfo_pull_sample( audio_lfo *lfo )
{
   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;
   }
}

static void audio_channel_get_samples( audio_channel *ch, 
                                       u32 count, float *buf )
{
   vg_profile_begin( &_vg_prof_audio_decode );

   u32 remaining = count;
   u32 buffer_pos = 0;

   while( remaining )
   {
      u32 samples_this_run = VG_MIN( remaining, ch->source_length -ch->cursor );
      remaining -= samples_this_run;

      float *dst = &buf[ buffer_pos * 2 ]; 
      
      if( ch->source->flags & AUDIO_FLAG_STEREO )
      {
         for( int i=0;i<samples_this_run; i++ )
         {
            dst[i*2+0] = 0.0f;
            dst[i*2+1] = 0.0f;
         }
      }
      else if( ch->source->flags & AUDIO_FLAG_VORBIS )
      {
         int read_samples = stb_vorbis_get_samples_float_interleaved_stereo( 
               ch->vorbis_handle,
               dst,
               samples_this_run );

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

            for( int i=0; i<samples_this_run; i++ )
            {
               dst[i*2+0] = 0.0f;
               dst[i*2+1] = 0.0f;
            }
         }
      }
      else
      {
         i16 *src_buffer = ch->source->data,
             *src        = &src_buffer[ch->cursor];

         audio_decode_uncompressed_mono( src, samples_this_run, dst );
      }

      ch->cursor += samples_this_run;
      buffer_pos += samples_this_run;
      
      if( (ch->flags & AUDIO_FLAG_LOOP) && remaining )
      {
         if( ch->source->flags & AUDIO_FLAG_VORBIS )
            stb_vorbis_seek_start( ch->vorbis_handle );

         ch->cursor = 0;
         continue;
      }
      else
         break;
   }

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

      remaining --;
   }

   vg_profile_end( &_vg_prof_audio_decode );
}

static void audio_channel_mix( audio_channel *ch, 
                               float *buffer, u32 frame_count )
{
   u32 frames_write = frame_count;

   u32 buffer_length = frame_count;
   if( ch->_.sampling_rate != 1.0f )
   {
      buffer_length = ceilf( (float)frame_count * ch->_.sampling_rate ) + 1;
   }

   float *pcf = alloca( buffer_length * 2 * sizeof(float) );

   audio_channel_get_samples( ch, buffer_length, pcf );
   vg_profile_begin( &_vg_prof_audio_mix );

   if( ch->_.lfo )
      ch->_.lfo->time = ch->_.lfo->time_startframe;

   float framevol_l = 1.0f,
         framevol_r = 1.0f;

   if( ch->flags & AUDIO_FLAG_SPACIAL_3D )
   {
      if( !vg_validf(vg_audio.listener_pos[0]) ||
          !vg_validf(vg_audio.listener_pos[1]) ||
          !vg_validf(vg_audio.listener_pos[2]) ||
          !vg_validf(ch->_.spacial_falloff[0]) ||
          !vg_validf(ch->_.spacial_falloff[1]) ||
          !vg_validf(ch->_.spacial_falloff[2]) )
      {
         vg_error( "NaN listener/world position (%s)\n", ch->name );

         framevol_l = 0.0f;
         framevol_r = 0.0f;
      }

      v3f delta;
      v3_sub( ch->_.spacial_falloff, vg_audio.listener_pos, delta );

      float dist = v3_length( delta ),
            vol  = vg_maxf( 0.0f, 1.0f - ch->_.spacial_falloff[3]*dist );

      v3_muls( delta, 1.0f/dist, delta );
      float pan  = v3_dot( vg_audio.listener_ears, delta );
      vol = powf( vol, 5.0f );

      framevol_l *= (vol * 0.5f) * (1.0f - pan);
      framevol_r *= (vol * 0.5f) * (1.0f + pan);
   }

   for( u32 j=0; j<frame_count; j++ )
   {
      if( ch->volume_movement < ch->_.volume_rate )
      {
         ch->volume_movement ++;

         float movement_t  = ch->volume_movement;
               movement_t /= (float)ch->_.volume_rate;

         ch->_.volume = vg_lerpf( ch->volume_movement_start,
                                  ch->_.volume_target, 
                                  movement_t );
      }

      float vol_norm = ch->_.volume * ch->_.volume;

      if( ch->_.lfo )
         vol_norm *= 1.0f + audio_lfo_pull_sample( ch->_.lfo ) 
                                          * ch->_.lfo_amount;

      float vol_l     = vol_norm * framevol_l,
            vol_r     = vol_norm * framevol_r,
            sample_l,
            sample_r;
      
      if( ch->_.sampling_rate != 1.0f )
      {
         /* absolutely garbage resampling, but it will do
          */

         float sample_index = ch->_.sampling_rate * (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;
   }

   vg_profile_end( &_vg_prof_audio_mix );
}

VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count )
{
   /*
    * Copy data and move edit flags to commit flags
    * ------------------------------------------------------------- */
   audio_lock();
   for( int i=0; i<AUDIO_CHANNELS; i++ )
   {
      audio_channel *ch = &vg_audio.channels[i];

      if( !ch->allocated )
         continue;

      /* process relinquishments */
      if( (ch->activity != k_channel_activity_reset) && ch->_.relinquished )
      {
         if( (ch->cursor >= ch->source_length && !(ch->flags & AUDIO_FLAG_LOOP)) 
            || (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;
      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;
   }


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

   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 ++ )
   {
      audio_channel *ch = &vg_audio.channels[i];

      if( ch->activity == k_channel_activity_alive )
         audio_channel_mix( ch, pOut32F, frame_count );
   }

   /* 
    * Relinquishing conditions
    * ------------------------------------------------------------------
    */
   audio_lock();

   /* Profiling information 
    * ----------------------------------------------- */
   vg_profile_increment( &_vg_prof_audio_decode );
   vg_profile_increment( &_vg_prof_audio_mix );
   vg_prof_audio_mix = _vg_prof_audio_mix;
   vg_prof_audio_decode = _vg_prof_audio_decode;
   vg_audio.samples_last = frame_count;

   audio_unlock();
}

VG_STATIC void audio_clip_load( audio_clip *clip, void *lin_alloc )
{
   if( lin_alloc == NULL )
      lin_alloc = vg_audio.audio_pool;


   /* load in directly */
   if( clip->flags & AUDIO_FLAG_VORBIS )
   {
      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 );
   }
   else if( clip->flags & AUDIO_FLAG_STEREO )
   {
      vg_fatal_exit_loop( "Unsupported format (Stereo uncompressed)" );
   }
   else
   {
      vg_linear_clear( vg_mem.scratch );
      u32 fsize;

      stb_vorbis_alloc alloc = {
         .alloc_buffer = vg_linear_alloc( vg_mem.scratch, AUDIO_DECODE_SIZE ),
         .alloc_buffer_length_in_bytes = AUDIO_DECODE_SIZE
      };

      void *filedata = vg_file_read( vg_mem.scratch, clip->path, &fsize );

      int err;
      stb_vorbis *decoder = stb_vorbis_open_memory( 
                            filedata, fsize, &err, &alloc );

      if( !decoder )
      {
         vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n", 
                     clip->path, err );
         vg_fatal_exit_loop( "Vorbis decode error" );
      }

      /* only mono is supported in uncompressed */
      u32 length_samples = stb_vorbis_stream_length_in_samples( decoder ),
          data_size      = length_samples * sizeof(i16);

      audio_lock();
      clip->data = vg_linear_alloc( lin_alloc, vg_align8(data_size) );
      clip->size = length_samples;
      audio_unlock();

      int read_samples = stb_vorbis_get_samples_i16_downmixed( 
                              decoder, clip->data, length_samples );

      if( read_samples != length_samples )
         vg_fatal_exit_loop( "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 );
   }
}

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

VG_STATIC void audio_require_clip_loaded( audio_clip *clip )
{
   if( clip->data && clip->size )
      return;

   audio_unlock();
   vg_fatal_exit_loop( "Must load audio clip before playing! \n" );
}

/* 
 * Debugging
 */

VG_STATIC void audio_debug_ui( m4x4f mtx_pv )
{
   if( !vg_audio.debug_ui )
      return;

   audio_lock();

   /* 
    * Profiler
    * -----------------------------------------------------------------------
    */

   float budget = ((double)vg_audio.samples_last / 44100.0) * 1000.0;
   vg_profile_drawn( (struct vg_profile *[]){ &vg_prof_audio_decode,
                                              &vg_prof_audio_mix }, 2, 
                     budget, (ui_rect){ 4, VG_PROFILE_SAMPLE_COUNT*2 + 8,
                                        250, 0 }, 3 );


   char perf[128];
        
   /* Draw UI */
   vg_uictx.cursor[0] = 258;
   vg_uictx.cursor[1] = VG_PROFILE_SAMPLE_COUNT*2+8+24+12;
   vg_uictx.cursor[2] = 150;
   vg_uictx.cursor[3] = 12;
   
   float mb1      = 1024.0f*1024.0f,
         usage    = vg_linear_get_cur( vg_audio.audio_pool )      / mb1,
         total    = vg_linear_get_capacity( vg_audio.audio_pool ) / mb1,
         percent  = (usage/total) * 100.0f;

   snprintf( perf, 127, "Mem: %.1f/%.1fmb (%.1f%%)\n", usage, total, percent );

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

   ui_rect overlap_buffer[ AUDIO_CHANNELS ];
   u32 overlap_length = 0;

        /* Draw audio stack */
        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_end_down();
         vg_uictx.cursor[1] += 1;
         continue;
      }

      const char *formats[] =
      {
         "------",
         "Mono  ", 
         "Stereo",
         "Vorbis"
      };

      int format_index = 0;

      if( ch->source->flags & AUDIO_FLAG_STEREO )
         format_index = 2;
      else if( ch->source->flags & AUDIO_FLAG_VORBIS )
         format_index = 3;
      else
         format_index = 1;

      snprintf( perf, 127, "%02d %c%c%cD %s %4.2fv'%s'", 
               i,
               (ch->editable_state.relinquished)? 'r': ' ',
               0?                                 'r': ' ',
               0?                                 '3': '2',
               formats[format_index],
               ch->editable_state.volume,
               ch->name );

      if( format_index == 0 )
      {
         ui_fill_rect( vg_uictx.cursor, 0xa00000ff );
      }
      else
      {
         ui_fill_rect( vg_uictx.cursor, 0xa0333333 );
      }

      vg_uictx.cursor[0] += 2;
      vg_uictx.cursor[1] += 2;
      ui_text( vg_uictx.cursor, perf, 1, 0 );

                ui_end_down();
                vg_uictx.cursor[1] += 1;
      
      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[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( wr, perf, 1, 0 );

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

   audio_unlock();
}

#endif /* VG_AUDIO_H */