-/* Copyright (C) 2021-2022 Harry Godden (hgn) - All Rights Reserved */
+/* Copyright (C) 2021-2024 Harry Godden (hgn) - All Rights Reserved */
-#ifndef VG_AUDIO_H
-#define VG_AUDIO_H
+#pragma once
-#define VG_GAME
+#include "vg_platform.h"
+#include "vg_engine.h"
+#include "vg_string.h"
+#include "vg_vorbis.h"
-#include "vg/vg.h"
-#include "vg/vg_stdint.h"
-#include "vg/vg_platform.h"
-#include "vg/vg_io.h"
-#include "vg/vg_m.h"
-#include "vg/vg_ui.h"
-#include "vg/vg_console.h"
-#include "vg/vg_store.h"
-#include "vg/vg_profiler.h"
+#define AUDIO_FRAME_SIZE 512
+#define AUDIO_MIX_FRAME_SIZE 256
-#include <sys/time.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 SFX_MAX_SYSTEMS 32
+#define AUDIO_CHANNELS 32
+#define AUDIO_LFOS 8
+#define AUDIO_FILTERS 16
#define AUDIO_FLAG_LOOP 0x1
-#define AUDIO_FLAG_ONESHOT 0x2
+#define AUDIO_FLAG_NO_DOPPLER 0x2
#define AUDIO_FLAG_SPACIAL_3D 0x4
#define AUDIO_FLAG_AUTO_START 0x8
-#define AUDIO_FLAG_KILL 0x10
-
-#define FADEOUT_LENGTH 1100
-#define FADEOUT_DIVISOR (1.0f/(float)FADEOUT_LENGTH)
+#define AUDIO_FLAG_NO_DSP 0x10
+#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_gen = 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 */
-
-enum audio_source_mode
-{
- k_audio_source_mono,
- k_audio_source_compressed,
-};
+#define AUDIO_MUTE_VOLUME 0.0f
+#define AUDIO_BASE_VOLUME 1.0f
typedef struct audio_clip audio_clip;
-struct audio_clip
-{
- const char *path;
- enum audio_source_mode source_mode;
-
- u32 size;
- void *data;
-};
+typedef struct audio_channel audio_channel;
+typedef struct audio_lfo audio_lfo;
-typedef struct audio_mix_info audio_mix_info;
-struct audio_mix_info
+struct audio_clip
{
- audio_clip *source;
- v3f world_position;
+ union { /* TODO oof.. */
+ u64 _p64_;
+ const char *path;
+ void *func;
+ };
- float vol, pan;
u32 flags;
-};
-
-typedef struct audio_player audio_player;
-struct audio_player
-{
- aatree_ptr active_entity; /* non-nil if currently playing */
- audio_mix_info info;
- int enqued, init;
-
- /* Diagnostic */
- const char *name;
-};
-
-typedef struct audio_entity audio_entity;
-struct audio_entity
-{
- audio_player *player;
- audio_mix_info info;
+ u32 size;
- u32 length, cur;
-
- /* Effects */
- u32 fadeout, fadeout_current;
- const char *name;
+ union{
+ u64 _p64;
+ void *data;
+ };
};
-/*
- * TODO list sunday
- *
- * play again: if already playing, leave in queue while it fadeouts
- * oneshot: create a ghost entity
- *
- */
-
-static struct vg_audio_system
+struct vg_audio_system
{
-#if 0
- ma_device miniaudio_device;
- ma_device_config miniaudio_dconfig;
-#endif
SDL_AudioDeviceID sdl_output_device;
+ vg_str device_choice; /* buffer is null? use default from OS */
+
+ bool always_keep_compressed;
void *audio_pool,
*decode_buffer;
/* synchro */
int sync_locked;
- SDL_mutex *mux_checker,
- *mux_sync;
-
- /* Audio engine, thread 1 */
- struct active_audio_player
- {
- int active;
- union
- {
- audio_entity ent;
- aatree_pool_node pool_node;
- };
-
- stb_vorbis *vorbis_handle;
- stb_vorbis_alloc vorbis_alloc;
- }
- active_players[ SFX_MAX_SYSTEMS ];
-
- aatree active_pool_info; /* note: just using the pool */
- aatree_ptr active_pool_head;
-
- /* System queue, and access from thread 0 */
- audio_entity entity_queue[SFX_MAX_SYSTEMS];
- int queue_len;
- int debug_ui, debug_ui_3d;
-
- v3f listener_pos,
- listener_ears;
-}
-vg_audio;
-
-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;
+ SDL_SpinLock sl_checker,
+ sl_sync;
-/*
- * 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_convar_push( (struct vg_convar){
- .name = "debug_audio",
- .data = &vg_audio.debug_ui,
- .data_type = k_convar_dtype_i32,
- .opt_i32 = { .min=0, .max=1, .clamp=1 },
- .persistent = 1
- });
-
- /* allocate memory */
-
- /* 32mb fixed */
- vg_audio.audio_pool =
- vg_create_linear_allocator( vg_mem.rtmemory, 1024*1024*32,
- VG_MEMORY_SYSTEM );
-
- /* fixed */
- u32 decode_size = AUDIO_DECODE_SIZE * SFX_MAX_SYSTEMS;
- vg_audio.decode_buffer = vg_linear_alloc( vg_mem.rtmemory, decode_size );
-
- /* setup pool */
- vg_audio.active_pool_info.base = vg_audio.active_players;
- vg_audio.active_pool_info.offset = offsetof(struct active_audio_player,
- pool_node );
- vg_audio.active_pool_info.stride = sizeof(struct active_audio_player);
- vg_audio.active_pool_info.p_cmp = NULL;
- aatree_init_pool( &vg_audio.active_pool_info, SFX_MAX_SYSTEMS );
-
- 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
- */
-
-static aatree_ptr audio_alloc_entity_internal(void)
-{
- aatree_ptr playerid = aatree_pool_alloc( &vg_audio.active_pool_info,
- &vg_audio.active_pool_head );
-
- if( playerid == AATREE_PTR_NIL )
- return AATREE_PTR_NIL;
-
- struct active_audio_player *aap = &vg_audio.active_players[ playerid ];
- aap->active = 1;
-
- return playerid;
-}
-
-VG_STATIC void audio_entity_free_internal( aatree_ptr id )
-{
- struct active_audio_player *aap = &vg_audio.active_players[ id ];
- aap->active = 0;
-
- /* Notify player that we've finished */
- if( aap->ent.player )
- aap->ent.player->active_entity = AATREE_PTR_NIL;
-
- /* delete */
- aatree_pool_free( &vg_audio.active_pool_info, id,
- &vg_audio.active_pool_head );
-}
+ struct audio_lfo{
+ u32 time, time_startframe;
+ float sqrt_polynomial_coefficient;
-VG_STATIC void *audio_entity_vorbis_ptr( aatree_ptr entid )
-{
- u8 *buf = (u8*)vg_audio.decode_buffer,
- *loc = &buf[AUDIO_DECODE_SIZE*entid];
-
- return (void *)loc;
-}
-
-VG_STATIC void audio_entity_start( audio_entity *src )
-{
- aatree_ptr entid = audio_alloc_entity_internal();
- if( entid == AATREE_PTR_NIL )
- return;
-
- audio_entity *ent = &vg_audio.active_players[ entid ].ent;
-
- ent->info = src->info;
- ent->name = src->info.source->path;
- ent->cur = 0;
- ent->player = src->player;
-
- ent->fadeout = 0;
- ent->fadeout_current = 0;
-
- /* Notify main player we are dequeud and playing */
- if( src->player )
- {
- src->player->enqued = 0;
- src->player->active_entity = entid;
- }
-
- if( src->info.source->source_mode == k_audio_source_compressed )
- {
- /* Setup vorbis decoder */
- struct active_audio_player *aap = &vg_audio.active_players[ entid ];
-
- stb_vorbis_alloc alloc = {
- .alloc_buffer = (char *)audio_entity_vorbis_ptr( entid ),
- .alloc_buffer_length_in_bytes = AUDIO_DECODE_SIZE
- };
-
- int err;
- stb_vorbis *decoder = stb_vorbis_open_memory(
- src->info.source->data,
- src->info.source->size, &err, &alloc );
-
- if( !decoder )
- {
- vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
- src->info.source->path, err );
-
- audio_entity_free_internal( entid );
- return;
- }
- else
- {
- ent->length = stb_vorbis_stream_length_in_samples( decoder );
- }
-
- aap->vorbis_handle = decoder;
- }
- else
- {
- ent->length = src->info.source->size;
- }
-}
-
-/*
- * Read everything from the queue
- */
-VG_STATIC void audio_system_enque(void)
-{
- /* Process incoming sound queue */
- audio_lock();
-
- int wr = 0;
- for( int i=0; i<vg_audio.queue_len; i++ )
- {
- audio_entity *src = &vg_audio.entity_queue[ i ];
-
- if( src->player )
- {
- /* Start new */
- if( src->player->active_entity == AATREE_PTR_NIL )
- {
- audio_entity_start( src );
- }
- else
- {
- /* Otherwise try start fadeout but dont remove from queue */
-
- aatree_ptr entid = src->player->active_entity;
- audio_entity *ent = &vg_audio.active_players[ entid ].ent;
- if( !ent->fadeout )
- {
- ent->fadeout = FADEOUT_LENGTH;
- ent->fadeout_current = FADEOUT_LENGTH;
- }
-
- vg_audio.entity_queue[ wr ++ ] = *src;
- }
- }
- else
- {
- audio_entity_start( src );
- }
- }
-
- vg_audio.queue_len = wr;
-
- /* Localize others memory */
- for( int i=0; i<SFX_MAX_SYSTEMS; i ++ )
- {
- struct active_audio_player *aap = &vg_audio.active_players[i];
- if( !aap->active )
- continue;
-
- if( aap->ent.player )
- {
- /* Only copy information in whilst not requeing */
- if( aap->ent.player->enqued == 0 )
- {
- aap->ent.info = aap->ent.player->info;
-
- if( (aap->ent.info.flags & AUDIO_FLAG_KILL) && !aap->ent.fadeout )
- {
- aap->ent.fadeout = FADEOUT_LENGTH;
- aap->ent.fadeout_current = FADEOUT_LENGTH;
- }
- }
- }
- }
-
- audio_unlock();
-}
-
-/*
- * Redistribute sound systems
- */
-VG_STATIC void audio_system_cleanup(void)
-{
- audio_lock();
-
- for( int i=0; i<SFX_MAX_SYSTEMS; i ++ )
- {
- struct active_audio_player *aap = &vg_audio.active_players[i];
- if( aap->active )
- {
- audio_entity *src = &aap->ent;
- if( src->cur < src->length || (src->info.flags & AUDIO_FLAG_LOOP ))
- {
- /* Good to keep */
- }
- else
- {
- audio_entity_free_internal( i );
+ struct{
+ enum lfo_wave_type{
+ k_lfo_triangle,
+ k_lfo_square,
+ k_lfo_saw,
+ k_lfo_polynomial_bipolar
}
- }
- }
-
- audio_unlock();
-}
-
-/*
- * Get effective volume and pan from this entity
- */
-VG_STATIC void audio_entity_spacialize( audio_entity *ent, float *vol, float *pan )
-{
- if( ent->info.vol < 0.01f )
- {
- *vol = ent->info.vol;
- *pan = 0.0f;
- return;
- }
-
- v3f delta;
- v3_sub( ent->info.world_position, vg_audio.listener_pos, delta );
-
- float dist2 = v3_length2( delta );
-
- if( dist2 < 0.0001f )
- {
- *pan = 0.0f;
- *vol = 1.0f;
- }
- else
- {
- float dist = sqrtf( dist2 ),
- attn = (dist / ent->info.vol) +1.0f;
-
- v3_muls( delta, 1.0f/dist, delta );
- *pan = v3_dot( vg_audio.listener_ears, delta );
- *vol = 1.0f/(attn*attn);
- }
-}
-
-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;
+ wave_type;
- 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];
+ u32 period;
+ float polynomial_coefficient;
}
-
- n += k;
- f->channel_buffer_start += k;
-
- if( n == len )
- break;
-
- if( !stb_vorbis_get_frame_float( f, NULL, NULL ))
- break;
+ _, editable_state;
+ u32 editble_state_write_mask;
}
+ oscillators[ AUDIO_LFOS ];
- return n;
-}
+ struct audio_channel{
+ int allocated;
+ u16 group;
+ u8 world_id;
-/*
- * ........ 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 );
+ char name[32]; /* only editable while allocated == 0 */
+ audio_clip *source; /* ... */
+ u32 flags; /* ... */
+ u32 colour; /* ... */
- while( n < len )
- {
- int k = f->channel_buffer_end - f->channel_buffer_start;
+ /* internal non-readable state
+ * -----------------------------*/
+ u32 cursor, source_length;
- if( n+k >= len )
- k = len - n;
+ float volume_movement_start,
+ pan_movement_start;
- 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];
+ u32 volume_movement,
+ pan_movement;
- *buffer++ = vg_clampf( 0.5f*(sl+sr), -1.0f, 1.0f ) * 32767.0f;
- //*buffer++ = vg_clampf( sr, -1.0f, 1.0f ) * 32767.0f;
+ union{
+ struct synth_bird *bird;
+ stb_vorbis *vorbis;
}
+ handle;
- n += k;
- f->channel_buffer_start += k;
-
- if( n == len )
- break;
-
- if( !stb_vorbis_get_frame_float( f, NULL, NULL ))
- break;
- }
-
- return n;
-}
-
-VG_STATIC void audio_entity_get_samples( aatree_ptr id, u32 count, float *buf )
-{
- vg_profile_begin( &_vg_prof_audio_decode );
-
- struct active_audio_player *aap = &vg_audio.active_players[id];
- audio_entity *ent = &aap->ent;
-
- u32 remaining = count;
- u32 cursor = ent->cur;
- u32 buffer_pos = 0;
-
- while( remaining )
- {
- u32 samples_this_run = VG_MIN( remaining, ent->length - cursor );
- remaining -= samples_this_run;
-
- float *dst = &buf[ buffer_pos * 2 ];
-
- int source_mode = ent->info.source->source_mode;
-
- if( source_mode == k_audio_source_mono )
- {
- i16 *src_buffer = ent->info.source->data,
- *src = &src_buffer[cursor];
-
- audio_decode_uncompressed_mono( src, samples_this_run, dst );
- }
- else if( source_mode == k_audio_source_compressed )
- {
- int read_samples = stb_vorbis_get_samples_float_interleaved_stereo(
- aap->vorbis_handle,
- dst,
- samples_this_run );
-
- if( read_samples != samples_this_run )
- {
- vg_warn( "Invalid samples read (%s)\n", ent->info.source->path );
- }
- }
-
- cursor += samples_this_run;
- buffer_pos += samples_this_run;
-
- if( (ent->info.flags & AUDIO_FLAG_LOOP) && remaining )
- {
- if( source_mode == k_audio_source_compressed )
- {
- stb_vorbis_seek_start( aap->vorbis_handle );
- }
-
- cursor = 0;
- continue;
- }
- else
- break;
- }
-
- while( remaining )
- {
- buf[ buffer_pos*2 + 0 ] = 0.0f;
- buf[ buffer_pos*2 + 1 ] = 0.0f;
- buffer_pos ++;
-
- remaining --;
- }
-
- ent->cur = cursor;
- vg_profile_end( &_vg_prof_audio_decode );
-}
-
-VG_STATIC void audio_entity_mix( aatree_ptr id, float *buffer,
- u32 frame_count )
-{
- audio_entity *ent = &vg_audio.active_players[id].ent;
-
- u32 cursor = ent->cur, buffer_pos = 0;
- float *pcf = alloca( frame_count * 2 * sizeof(float) );
-
- u32 frames_write = frame_count;
- float fadeout_divisor = 1.0f / (float)ent->fadeout;
-
- float vol = ent->info.vol,
- pan = ent->info.pan;
-
- audio_entity_get_samples( id, frame_count, pcf );
-
- vg_profile_begin( &_vg_prof_audio_mix );
-
- if( ent->info.flags & AUDIO_FLAG_SPACIAL_3D )
- audio_entity_spacialize( ent, &vol, &pan );
-
- for( u32 j=0; j<frame_count; j++ )
- {
- float frame_vol = vol;
-
- if( ent->fadeout )
- {
- /* Force this system to be removed now */
- if( ent->fadeout_current == 0 )
- {
- ent->info.flags = 0x00;
- ent->cur = ent->length;
- break;
- }
-
- frame_vol *= (float)ent->fadeout_current * fadeout_divisor;
- ent->fadeout_current --;
- }
-
- float sl = 1.0f-pan,
- sr = 1.0f+pan;
-
- buffer[ buffer_pos*2+0 ] += pcf[ buffer_pos*2+0 ] * frame_vol * sl;
- buffer[ buffer_pos*2+1 ] += pcf[ buffer_pos*2+1 ] * frame_vol * sr;
-
- buffer_pos ++;
- }
-
- vg_profile_end( &_vg_prof_audio_mix );
-}
-
-VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count )
-{
- audio_system_enque();
-
- 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;
-
- /* Mix all sounds */
- for( int i=0; i<SFX_MAX_SYSTEMS; i ++ )
- {
- struct active_audio_player *aap = &vg_audio.active_players[i];
-
- if( aap->active )
- {
- audio_entity_mix( i, pOut32F, frame_count );
- }
- }
-
- /* redistribute */
- audio_system_cleanup();
-
- audio_lock();
- 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;
-
- if( clip->source_mode == k_audio_source_mono )
- {
- 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 );
+ stb_vorbis_alloc vorbis_alloc;
- if( !decoder )
- {
- vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
- clip->path, err );
- vg_fatal_exit_loop( "Vorbis decode error" );
+ 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
}
-
- /* 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, 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 );
- }
-
- /* load in directly */
- else if( clip->source_mode == k_audio_source_compressed )
- {
- audio_lock();
- clip->data = vg_file_read( lin_alloc, clip->path, &clip->size );
- audio_unlock();
-
- if( !clip->data )
- vg_fatal_exit_loop( "Audio failed to load" );
-
- float mb = (float)(clip->size) / (1024.0f*1024.0f);
- vg_info( "Loaded audio clip '%s' (%.1fmb)\n", clip->path, mb );
- }
-}
-
-VG_STATIC void audio_clip_loadn( audio_clip *arr, int count, void *lin_alloc )
-{
- for( int i=0; i<count; i++ )
- audio_clip_load( &arr[i], lin_alloc );
-}
-
-/* Mark change to be uploaded through queue system */
-VG_STATIC void audio_player_commit( audio_player *sys )
-{
- audio_require_lock();
-
- if( vg_audio.queue_len >= vg_list_size( vg_audio.entity_queue ) )
- {
- vg_warn( "Audio commit queue full\n" );
- return;
- }
-
- if( sys->enqued )
- {
- vg_warn( "[2] Audio commit spamming; already enqued (%s)\n", sys->name );
- return;
- }
-
- sys->enqued = 1;
- audio_entity *ent = &vg_audio.entity_queue[ vg_audio.queue_len ++ ];
- ent->info = sys->info;
- ent->player = sys;
-}
-
-VG_STATIC void audio_require_init( audio_player *player )
-{
- if( player->init )
- return;
-
- audio_unlock();
- vg_fatal_exit_loop( "Must init audio player before playing! \n" );
-}
-
-VG_STATIC void audio_require_clip_loaded( audio_clip *clip )
-{
- if( clip->data && clip->size )
- return;
-
- audio_unlock();
- vg_fatal_exit_loop( "Must load audio clip before playing! \n" );
-}
-
-/* Play a clip using player. If its already playing something, it will
- * fadeout quickly and start the next sound */
-VG_STATIC void audio_player_playclip( audio_player *player, audio_clip *clip )
-{
- audio_require_lock();
- audio_require_init( player );
- audio_require_clip_loaded( clip );
-
- if( player->info.flags & AUDIO_FLAG_KILL )
- {
- vg_error( "Can't start audio clip on player that is/has disconnected" );
- return;
- }
-
- if( player->enqued )
- {
- vg_warn( "[1] Audio commit spamming; already enqued (%s)\n",
- player->name );
- return;
- }
-
- player->info.source = clip;
- audio_player_commit( player );
-}
-
-VG_STATIC void audio_play_oneshot( audio_clip *clip, float volume )
-{
- audio_require_lock();
- audio_require_clip_loaded( clip );
-
- if( vg_audio.queue_len >= vg_list_size( vg_audio.entity_queue ) )
- {
- vg_warn( "Audio commit queue full\n" );
- return;
- }
-
- audio_entity *ent = &vg_audio.entity_queue[ vg_audio.queue_len ++ ];
-
- ent->info.flags = AUDIO_FLAG_ONESHOT;
- ent->info.pan = 0.0f;
- ent->info.source = clip;
- ent->info.vol = volume;
- ent->player = NULL;
-}
-
-VG_STATIC void audio_player_init( audio_player *player )
-{
- player->active_entity = AATREE_PTR_NIL;
- player->init = 1;
-}
-
-/*
- * Effects
- */
-
-/*
- * Safety enforced Get/set attributes
- */
-
-VG_STATIC int audio_player_is_playing( audio_player *sys )
-{
- audio_require_lock();
-
- if( sys->active_entity != AATREE_PTR_NIL )
- return 1;
- else
- return 0;
-}
-
-VG_STATIC void audio_player_set_position( audio_player *sys, v3f pos )
-{
- audio_require_lock();
- v3_copy( pos, sys->info.world_position );
-}
-
-VG_STATIC void audio_player_set_vol( audio_player *sys, float vol )
-{
- audio_require_lock();
- sys->info.vol = vol;
-}
-
-VG_STATIC float audio_player_get_vol( audio_player *sys )
-{
- audio_require_lock();
- return sys->info.vol;
-}
-
-VG_STATIC void audio_player_set_pan( audio_player *sys, float pan )
-{
- audio_require_lock();
- sys->info.pan = pan;
-}
-
-VG_STATIC float audio_player_get_pan( audio_player *sys )
-{
- audio_require_lock();
- return sys->info.pan;
-}
-
-VG_STATIC void audio_player_set_flags( audio_player *sys, u32 flags )
-{
- audio_require_lock();
- sys->info.flags = flags;
-}
-
-VG_STATIC u32 audio_player_get_flags( audio_player *sys )
-{
- audio_require_lock();
- return sys->info.flags;
-}
-
-
-/*
- * Debugging
- */
-
-VG_STATIC void audio_debug_ui( m4x4f mtx_pv )
-{
- if( !vg_audio.debug_ui )
- return;
-
- /* Get data */
- struct sound_info
- {
- const char *name;
- u32 cursor, flags, length;
- v3f pos;
- float vol;
- }
- infos[ SFX_MAX_SYSTEMS ];
- int num_systems = 0;
-
- audio_lock();
-
- for( int i=0; i<SFX_MAX_SYSTEMS; i ++ )
- {
- struct active_audio_player *aap = &vg_audio.active_players[i];
-
- if( !aap->active )
- continue;
-
- audio_entity *ent = &aap->ent;
- struct sound_info *snd = &infos[ num_systems ++ ];
-
- snd->name = ent->name;
- snd->cursor = ent->cur;
- snd->flags = ent->info.flags;
- snd->length = ent->length;
- snd->vol = ent->info.vol*100.0f;
- v3_copy( ent->info.world_position, snd->pos );
- }
-
- 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 );
-
- audio_unlock();
-
- char perf[128];
-
- /* Draw UI */
- vg_uictx.cursor[0] = 258;
- vg_uictx.cursor[1] = VG_PROFILE_SAMPLE_COUNT*2+8+24+12;
- vg_uictx.cursor[2] = 150;
- vg_uictx.cursor[3] = 12;
+ activity,
+ readable_activity;
- 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[ SFX_MAX_SYSTEMS ];
- u32 overlap_length = 0;
-
- /* Draw audio stack */
- for( int i=0; i<num_systems; i ++ )
- {
- struct sound_info *inf = &infos[i];
-
- vg_uictx.cursor[2] = 200;
- vg_uictx.cursor[3] = 18;
-
- u32 alpha = 0xa0000000;
-
- ui_new_node();
- {
- ui_fill_rect( vg_uictx.cursor, 0x00333333|alpha );
-
- ui_px baseline = vg_uictx.cursor[0],
- w = 200,
- c = baseline + ((float)inf->cursor / (float)inf->length) * w;
-
- /* cursor */
- vg_uictx.cursor[2] = 2;
- vg_uictx.cursor[0] = c;
- ui_fill_rect( vg_uictx.cursor, 0xffffffff );
-
- vg_uictx.cursor[0] = baseline + 2;
- vg_uictx.cursor[1] += 2;
- snprintf( perf, 127, "%s %.1f%%", infos[i].name, infos[i].vol );
- ui_text( vg_uictx.cursor, perf, 1, 0 );
+ /*
+ * 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 */
- if( inf->flags & AUDIO_FLAG_SPACIAL_3D )
- {
- v4f wpos;
- v3_copy( inf->pos, wpos );
- wpos[3] = 1.0f;
- m4x4_mulv( mtx_pv, wpos, wpos );
-
- if( wpos[3] < 0.0f )
- goto projected_behind;
-
- v2_muls( wpos, (1.0f/wpos[3]) * 0.5f, wpos );
- v2_add( wpos, (v2f){ 0.5f, 0.5f }, wpos );
-
- ui_rect wr;
- wr[0] = wpos[0] * vg.window_x;
- wr[1] = (1.0f-wpos[1]) * vg.window_y;
- wr[2] = 100;
- wr[3] = 17;
-
- for( int j=0; j<12; j++ )
- {
- int collide = 0;
- for( int k=0; k<overlap_length; k++ )
- {
- ui_px *wk = overlap_buffer[k];
- if( ((wr[0] <= wk[0]+wk[2]) && (wr[0]+wr[2] >= wk[0])) &&
- ((wr[1] <= wk[1]+wk[3]) && (wr[1]+wr[3] >= wk[1])) )
- {
- collide = 1;
- break;
- }
- }
-
- if( !collide )
- break;
- else
- wr[1] += 18;
- }
-
- ui_text( wr, perf, 1, 0 );
-
- ui_rect_copy( wr, overlap_buffer[ overlap_length ++ ] );
- }
- }
-
-projected_behind:
-
- ui_end_down();
- vg_uictx.cursor[1] += 1;
- }
-}
-
-#endif /* VG_AUDIO_H */
+ audio_lfo *lfo;
+ float lfo_amount;
+ }
+ _, editable_state;
+ u32 editble_state_write_mask;
+ }
+ channels[ AUDIO_CHANNELS ];
+
+ int debug_ui, debug_ui_3d, debug_dsp, dsp_enabled;
+
+ 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;
+}
+extern vg_audio;
+
+void audio_clip_load( audio_clip *clip, void *lin_alloc );
+void audio_clip_loadn( audio_clip *arr, int count, void *lin_alloc );
+
+void vg_audio_register(void);
+void vg_audio_device_init(void);
+void vg_audio_init(void);
+void vg_audio_free(void);
+
+void audio_lock(void);
+void audio_unlock(void);
+
+void audio_channel_init( audio_channel *ch, audio_clip *clip, u32 flags );
+void audio_channel_group( audio_channel *ch, u16 group );
+void audio_channel_world( audio_channel *ch, u8 world_id );
+audio_channel *audio_get_first_idle_channel(void);
+audio_channel *audio_get_group_idle_channel( u16 group, u32 max_count );
+audio_channel *audio_get_group_first_active_channel( u16 group );
+int audio_channel_finished( audio_channel *ch );
+audio_channel *audio_relinquish_channel( audio_channel *ch );
+void audio_channel_slope_volume( audio_channel *ch, f32 length, f32 new_vol );
+void audio_channel_set_sampling_rate( audio_channel *ch, float rate );
+void audio_channel_edit_volume( audio_channel *ch, f32 new_vol, int instant );
+audio_channel *audio_channel_fadeout( audio_channel *ch, float length );
+void audio_channel_fadein( audio_channel *ch, float length );
+audio_channel *audio_channel_crossfade( audio_channel *ch,
+ audio_clip *new_clip,
+ float length, u32 flags );
+void audio_channel_sidechain_lfo( audio_channel *ch, int lfo_id, f32 amount );
+void audio_channel_set_spacial( audio_channel *ch, v3f co, float range );
+int audio_oneshot_3d( audio_clip *clip, v3f position, f32 range, f32 volume );
+int audio_oneshot( audio_clip *clip, f32 volume, f32 pan );
+void audio_set_lfo_wave( int id, enum lfo_wave_type type, f32 coefficient );
+void audio_set_lfo_frequency( int id, float freq );
+int audio_channel_load_source( audio_channel *ch );
+
+void audio_debug_ui(
+
+#ifdef VG_3D
+ m4x4f
+#else
+ m3x3f
+#endif
+ mtx_pv );
projects / vg.git / blobdiff