adwadwa

[vg.git] / vg_audio.h

/* Copyright (C) 2021-2023 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 "vg/vg_audio_synth_bird.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_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_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
{
   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;
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_SpinLock     sl_checker,
                    sl_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;
      u32 group;

      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_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;
   }
   channels[ AUDIO_CHANNELS ];

   int               debug_ui, debug_ui_3d;

   v3f               internal_listener_pos,
                     internal_listener_ears,
                     internal_listener_velocity,

                     external_listener_pos,
                     external_listener_ears,
                     external_lister_velocity;

   float             internal_global_volume,
                     external_global_volume;
}
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_dsp;

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

VG_STATIC void audio_lock_checker_store( int value )
{
   SDL_AtomicLock( &vg_audio.sl_checker );
   vg_audio.sync_locked = value;
   SDL_AtomicUnlock( &vg_audio.sl_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_AtomicLock( &vg_audio.sl_sync );
   audio_lock_checker_store(1);
}

VG_STATIC void audio_unlock(void)
{
   audio_lock_checker_store(0);
   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)
{
   /* 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.external_global_volume,
      .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 );

   vg_dsp_init();

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

   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)
{
   vg_dsp_free();
   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 void audio_channel_init( audio_channel *ch, audio_clip *clip, u32 flags )
{
   ch->group = 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
      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_group( audio_channel *ch, u32 group )
{
   ch->group = group;
   ch->colour = ((group * 29986577) & 0x00ffffff) | 0xff000000;
}

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 NULL;
}

static audio_channel *audio_get_group_idle_channel( u32 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( u32 group )
{
   for( int i=0; i<AUDIO_CHANNELS; i++ ){
      audio_channel *ch = &vg_audio.channels[i];
      if( ch->allocated && (ch->group == group) )
         return ch;
   }
   return NULL;
}

static int audio_channel_finished( audio_channel *ch )
{
   if( ch->readable_activity == k_channel_activity_end )
      return 1;
   else
      return 0;
}

static audio_channel *audio_relinquish_channel( audio_channel *ch )
{
   ch->editable_state.relinquished = 1;
   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 = 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 );

   audio_channel *replacement = audio_get_first_idle_channel();

   if( replacement ){
      audio_channel_init( replacement, new_clip, flags );
      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 );

      if( range == 0.0f )
         ch->editable_state.spacial_falloff[3] = 1.0f;
      else
         ch->editable_state.spacial_falloff[3] = 1.0f/range;

      ch->editble_state_write_mask |= AUDIO_EDIT_SPACIAL;
   }
   else{
      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_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 );

      return 1;
   }
   else
      return 0;
}

static int audio_oneshot( audio_clip *clip, float volume, float pan )
{
   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 );

      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 )
{
   u32 format = ch->source->flags & AUDIO_FLAG_FORMAT;

   if( format == k_audio_format_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( format == k_audio_format_bird ){
      u32 index = ch - vg_audio.channels;

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

      memcpy( loc, ch->source->data, ch->source->size );
      synth_bird_reset( loc );

      ch->bird_handle = loc;
      ch->source_length = synth_bird_get_length_in_samples( loc );
   }
   else if( format == k_audio_format_stereo ){
      ch->source_length = ch->source->size / 2;
   }
   else{
      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 inline 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;

   u32 format = ch->source->flags & AUDIO_FLAG_FORMAT;

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

      float *dst = &buf[ buffer_pos * 2 ]; 
      
      if( 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( format == k_audio_format_vorbis ){
         int read_samples = stb_vorbis_get_samples_float_interleaved_stereo( 
               ch->vorbis_handle,
               dst,
               samples_this_run );

         if( read_samples != samples_this_run ){
            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 );
      }

      ch->cursor += samples_this_run;
      buffer_pos += samples_this_run;
      
      if( (ch->flags & AUDIO_FLAG_LOOP) && remaining ){
         if( format == k_audio_format_vorbis )
            stb_vorbis_seek_start( ch->vorbis_handle );
         else if( format == k_audio_format_bird )
            synth_bird_reset( ch->bird_handle );

         ch->cursor = 0;
         continue;
      }
      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 )
{
   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.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_exit_loop( "NaN left channel" );
      if( !vg_validf( framevol_r ) ) vg_fatal_exit_loop( "NaN right channel" );
      if( !vg_validf( frame_samplerate ) ) 
         vg_fatal_exit_loop( "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 pcf[ AUDIO_MIX_FRAME_SIZE * 2 * 2 ];

   audio_channel_get_samples( ch, buffer_length, pcf );

   vg_profile_begin( &_vg_prof_audio_mix );

   float volume_movement = ch->volume_movement;
   float const fvolume_rate = vg_maxf( 1.0f, ch->_.volume_rate );
   const float inv_volume_rate = 1.0f/fvolume_rate;

   float volume = ch->_.volume;
   const float volume_start  = ch->volume_movement_start;
   const float volume_target = ch->_.volume_target;

   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;

      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( 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 )
{
   /*
    * 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 */
   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 ){
         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;
         }
      }
   }

   vg_profile_begin( &_vg_prof_dsp );

   for( int i=0; i<frame_count; i++ )
      vg_dsp_process( pOut32F + i*2, pOut32F + i*2 );

   vg_profile_end( &_vg_prof_dsp );

   audio_lock();

   for( int i=0; i<AUDIO_CHANNELS; i ++ ){
      audio_channel *ch = &vg_audio.channels[i];
      ch->readable_activity = ch->activity;
   }

   /* Profiling information 
    * ----------------------------------------------- */
   vg_profile_increment( &_vg_prof_audio_decode );
   vg_profile_increment( &_vg_prof_audio_mix );
   vg_profile_increment( &_vg_prof_dsp );

   vg_prof_audio_mix = _vg_prof_audio_mix;
   vg_prof_audio_decode = _vg_prof_audio_decode;
   vg_prof_audio_dsp = _vg_prof_dsp;

   vg_audio.samples_last = frame_count;

   if( vg_audio.debug_ui ){
      vg_dsp_update_texture();
   }

   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 */
   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_exit_loop( "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" );

      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_bird ){
      if( !clip->data ){
         vg_fatal_exit_loop( "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_exit_loop( "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_exit_loop( "No path specified, embeded mono unsupported" );
      }

      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();

   glBindTexture( GL_TEXTURE_2D, vg_dsp.view_texture );
   glTexSubImage2D( GL_TEXTURE_2D, 0, 0, 0, 256, 256, 
                     GL_RGBA, GL_UNSIGNED_BYTE,
                     vg_dsp.view_texture_buffer );

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

   float budget = ((double)vg_audio.samples_last / 44100.0) * 1000.0;
   vg_profile_drawn( (struct vg_profile *[]){ &vg_prof_audio_decode,
                                              &vg_prof_audio_mix,
                                              &vg_prof_audio_dsp}, 3, 
                     budget, (ui_rect){ 4, VG_PROFILE_SAMPLE_COUNT*2 + 8,
                                        512, 0 }, 3 );


   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 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  ",
         "   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 %c%c%cD %s [%s] %4.2fv'%s'", 
               i,
               (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_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;
      
      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] = 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] = 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 */