-/* Copyright (C) 2021 Harry Godden (hgn) - All Rights Reserved */
+/* Copyright (C) 2021-2022 Harry Godden (hgn) - All Rights Reserved */
+
+#ifndef VG_AUDIO_H
+#define VG_AUDIO_H
#define MA_NO_GENERATION
#define MA_NO_DECODING
#define MA_NO_ENCODING
+#define MA_NO_WAV
+#define MA_NO_FLAC
+#define MA_NO_MP3
+#define MA_NO_ENGINE
+#define MA_NO_NODE_GRAPH
+#define MA_NO_RESOURCE_MANAGER
+
#include "dr_soft/miniaudio.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 <time.h>
+
+#ifdef __GNUC__
+#ifndef __clang__
+ #pragma GCC push_options
+ #pragma GCC optimize ("O3")
+#endif
+#endif
+
+#pragma GCC diagnostic push
+#pragma GCC diagnostic ignored "-Wdeprecated-declarations"
+
#define STB_VORBIS_MAX_CHANNELS 2
#include "stb/stb_vorbis.h"
-#define SFX_MAX_SYSTEMS 32
-#define SFX_FLAG_STEREO 0x2
-#define SFX_FLAG_REPEAT 0x4
-#define SFX_FLAG_PERSISTENT 0x8
-#define FADEOUT_LENGTH 4410
-#define FADEOUT_DIVISOR (1.f/(float)FADEOUT_LENGTH)
+#pragma GCC diagnostic pop
+
+#ifdef __GNUC__
+#ifndef __clang__
+ #pragma GCC pop_options
+#endif
+#endif
+
+#define SFX_MAX_SYSTEMS 32
+#define AUDIO_FLAG_LOOP 0x1
+#define AUDIO_FLAG_ONESHOT 0x2
+#define AUDIO_FLAG_SPACIAL_3D 0x4
+#define AUDIO_FLAG_AUTO_START 0x8
+#define AUDIO_FLAG_KILL 0x10
-typedef struct sfx_vol_control sfx_vol_control;
-typedef struct sfx_system sfx_system;
+#define FADEOUT_LENGTH 1100
+#define FADEOUT_DIVISOR (1.0f/(float)FADEOUT_LENGTH)
-struct sfx_vol_control
+#define AUDIO_DECODE_SIZE (1024*256) /* 256 kb decoding buffers */
+
+enum audio_source_mode
{
- float val;
- const char *name;
+ k_audio_source_mono,
+ k_audio_source_compressed,
};
-struct sfx_system
+typedef struct audio_clip audio_clip;
+struct audio_clip
{
- sfx_system *persisitent_source;
- int in_queue;
+ const char *path;
+ enum audio_source_mode source_mode;
- /* Source buffer start */
- float *source, *replacement;
-
- u32 clip_start, clip_end, buffer_length;
-
- /* Modifiers */
- sfx_vol_control *vol_src;
- float vol, cvol, pan;
+ u32 size;
+ void *data;
+};
- u32 delay;
-
- /* Info */
- u32 ch, end, cur;
- u32 flags;
-
- /* Effects */
- u32 fadeout, fadeout_current;
-
- /* Diagnostic */
- const char *name;
+typedef struct audio_mix_info audio_mix_info;
+struct audio_mix_info
+{
+ audio_clip *source;
+ v3f world_position;
+
+ float vol, pan;
+ u32 flags;
};
-/* Set of up to 8 sound effects packed into one */
-typedef struct sfx_set sfx_set;
-struct sfx_set
+typedef struct audio_player audio_player;
+struct audio_player
{
- float *main;
- char *sources;
-
- u32 segments[32]; /* from->to,from->to ... */
- u32 numsegments;
- u32 ch;
- u32 flags;
+ aatree_ptr active_entity; /* non-nil if currently playing */
+ audio_mix_info info;
+ int enqued, init;
+
+ /* Diagnostic */
+ const char *name;
};
-ma_device g_aud_device;
-ma_device_config g_aud_dconfig;
+typedef struct audio_entity audio_entity;
+struct audio_entity
+{
+ audio_player *player;
+ audio_mix_info info;
-/*
- * Thread 1 - audio engine ( spawned from miniaudio.h )
- * ======================================================
+ u32 length, cur;
+
+ /* Effects */
+ u32 fadeout, fadeout_current;
+ const char *name;
+};
+
+/*
+ * TODO list sunday
+ *
+ * play again: if already playing, leave in queue while it fadeouts
+ * oneshot: create a ghost entity
+ *
*/
-sfx_system sfx_sys[SFX_MAX_SYSTEMS];
-int sfx_sys_len = 0;
+
+static struct vg_audio_system
+{
+ ma_device miniaudio_device;
+ ma_device_config miniaudio_dconfig;
+
+ void *audio_pool,
+ *decode_buffer;
+ u32 samples_last;
+
+ /* synchro */
+ int sync_locked;
+
+ vg_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;
/*
- * Thread 0 - Critical transfer section
- * ======================================================
+ * These functions are called from the main thread and used to prevent bad
+ * access. TODO: They should be no-ops in release builds.
*/
-MUTEX_TYPE sfx_mux_t01; /* Resources share: 0 & 1 */
+VG_STATIC int audio_lock_checker_load(void)
+{
+ int value;
+ vg_mutex_lock( &vg_audio.mux_checker );
+ value = vg_audio.sync_locked;
+ vg_mutex_unlock( &vg_audio.mux_checker );
+ return value;
+}
+
+VG_STATIC void audio_lock_checker_store( int value )
+{
+ vg_mutex_lock( &vg_audio.mux_checker );
+ vg_audio.sync_locked = value;
+ vg_mutex_unlock( &vg_audio.mux_checker );
+}
-sfx_system *sfx_q[SFX_MAX_SYSTEMS]; /* Stuff changed */
-int sfx_q_len = 0;
+VG_STATIC void audio_require_lock(void)
+{
+ if( audio_lock_checker_load() )
+ return;
-float g_master_volume = 1.f;
+ vg_error( "Modifying sound effects systems requires locking\n" );
+ abort();
+}
-/* Decompress entire vorbis stream into buffer */
-static float *sfx_vorbis_stream( const unsigned char *data, int len,
- int channels, u32 *samples )
+VG_STATIC void audio_lock(void)
{
- int err;
- stb_vorbis *pv = stb_vorbis_open_memory( data, len, &err, NULL );
-
- if( !pv )
- {
- vg_error( "stb_vorbis_open_memory() failed with error code: %i\n", err );
- return NULL;
- }
-
- u32 length_samples = stb_vorbis_stream_length_in_samples( pv );
- float *buffer = (float *)malloc( length_samples * channels * sizeof(float));
-
- if( !buffer )
- {
- stb_vorbis_close( pv );
- vg_error( "out of memory while allocating sound resource\n" );
- return NULL;
- }
-
- int read_samples = stb_vorbis_get_samples_float_interleaved(
- pv, channels, buffer, length_samples * channels );
+ vg_mutex_lock( &vg_audio.mux_sync );
+ audio_lock_checker_store(1);
+}
- if( read_samples != length_samples )
- {
- vg_warn( "| warning: sample count mismatch. Expected %u got %i\n",
- length_samples, read_samples );
- length_samples = read_samples;
- }
-
- stb_vorbis_close( pv );
- *samples = length_samples;
- return buffer;
+VG_STATIC void audio_unlock(void)
+{
+ audio_lock_checker_store(0);
+ vg_mutex_unlock( &vg_audio.mux_sync );
}
-static float *sfx_vorbis( const char *strFileName, int channels, u32 *samples )
+
+VG_STATIC void audio_mixer_callback( ma_device *pDevice, void *pOutBuf,
+ const void *pInput, ma_uint32 frameCount );
+
+VG_STATIC void vg_audio_init(void)
{
- i64 len;
- void *filedata = vg_asset_read_s( strFileName, &len );
-
- if( filedata )
- {
- float *wav = sfx_vorbis_stream( filedata, len, channels, samples );
- free( filedata );
- return wav;
- }
- else
- {
- vg_error( "OGG load failed\n" );
- return NULL;
- }
+ vg_mutex_init( &vg_audio.mux_checker );
+ vg_mutex_init( &vg_audio.mux_sync );
+
+ /* 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 );
+
+ ma_device_config *dconf = &vg_audio.miniaudio_dconfig;
+ ma_device *device = &vg_audio.miniaudio_device;
+
+ *dconf = ma_device_config_init( ma_device_type_playback );
+ dconf->playback.format = ma_format_f32;
+ dconf->playback.channels = 2;
+ dconf->sampleRate = 44100;
+ dconf->dataCallback = audio_mixer_callback;
+ dconf->periodSizeInFrames = 441;
+
+ dconf->pUserData = NULL;
+
+ vg_info( "Starting audio engine\n" );
+
+ if( ma_device_init( NULL, dconf, device) != MA_SUCCESS )
+ {
+ vg_fatal_exit_loop( "(audio) ma_device failed to initialize" );
+ }
+ else
+ {
+ if( ma_device_start( device ) != MA_SUCCESS )
+ {
+ ma_device_uninit( device );
+ vg_fatal_exit_loop( "(audio) ma_device failed to start" );
+ }
+ }
+
+ vg_success( "Ready\n" );
+}
+
+VG_STATIC void vg_audio_free(void * nothing)
+{
+ ma_device *device = &vg_audio.miniaudio_device;
+ ma_device_uninit( device );
+
+#if 0
+ vg_free( vg_audio.mem );
+ vg_audio.mem = NULL;
+#endif
}
/*
- * thread 0 / client code
+ * thread 1
*/
-static int sfx_begin_edit( sfx_system *sys )
+
+static aatree_ptr audio_alloc_entity_internal(void)
{
- MUTEX_LOCK( sfx_mux_t01 );
-
- if( sfx_q_len >= SFX_MAX_SYSTEMS && !sys->in_queue )
- {
- MUTEX_UNLOCK( sfx_mux_t01 );
- vg_warn( "Warning: No free space in sound queue\n" );
- return 0;
- }
-
- return 1;
+ 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;
}
-static void sfx_end_edit( sfx_system *sys )
+VG_STATIC void audio_entity_free_internal( aatree_ptr id )
{
- MUTEX_UNLOCK( sfx_mux_t01 );
+ 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 );
}
-/* Mark change to be uploaded to queue system */
-static int sfx_push( sfx_system *sys )
+VG_STATIC void *audio_entity_vorbis_ptr( aatree_ptr entid )
{
- if( !sys->in_queue )
- {
- sfx_q[ sfx_q_len ++ ] = sys;
- sys->in_queue = 1;
- }
-
- MUTEX_UNLOCK( sfx_mux_t01 );
-
- return 1;
+ u8 *buf = (u8*)vg_audio.decode_buffer,
+ *loc = &buf[AUDIO_DECODE_SIZE*entid];
+
+ return (void *)loc;
}
-/* Edit a volume float, has to be function wrapped because of mutex */
-static void sfx_vol_fset( sfx_vol_control *src, float to )
+VG_STATIC void audio_entity_start( audio_entity *src )
{
- MUTEX_LOCK( sfx_mux_t01 );
+ 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
+ };
- src->val = to;
+ int err;
+ stb_vorbis *decoder = stb_vorbis_open_memory(
+ src->info.source->data,
+ src->info.source->size, &err, &alloc );
- MUTEX_UNLOCK( sfx_mux_t01 );
+ 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;
+ }
}
-/* thread-safe get volume value */
-static float sfx_vol_fget( sfx_vol_control *src )
+/*
+ * Read everything from the queue
+ */
+VG_STATIC void audio_system_enque(void)
{
- float val;
-
- MUTEX_LOCK( sfx_mux_t01 );
-
- val = src->val;
-
- MUTEX_UNLOCK( sfx_mux_t01 );
-
- return val;
+ /* 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();
}
-/* thread-safe set master volume */
-static void sfx_set_master( float to )
+/*
+ * Redistribute sound systems
+ */
+VG_STATIC void audio_system_cleanup(void)
{
- MUTEX_LOCK( sfx_mux_t01 );
-
- g_master_volume = to;
-
- MUTEX_UNLOCK( sfx_mux_t01 );
+ 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 );
+ }
+ }
+ }
+
+ audio_unlock();
}
-/* thread-safe get master volume */
-static float sfx_get_master(void)
+/*
+ * Get effective volume and pan from this entity
+ */
+VG_STATIC void audio_entity_spacialize( audio_entity *ent, float *vol, float *pan )
{
- float val;
+ if( ent->info.vol < 0.01f )
+ {
+ *vol = ent->info.vol;
+ *pan = 0.0f;
+ return;
+ }
- MUTEX_LOCK( sfx_mux_t01 );
-
- val = g_master_volume;
-
- MUTEX_UNLOCK( sfx_mux_t01 );
-
- return val;
-}
+ 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;
-void audio_mixer_callback( ma_device *pDevice, void *pOutBuf,
- const void *pInput, ma_uint32 frameCount );
+ v3_muls( delta, 1.0f/dist, delta );
+ *pan = v3_dot( vg_audio.listener_ears, delta );
+ *vol = 1.0f/(attn*attn);
+ }
+}
-static void vg_audio_init(void)
+VG_STATIC void audio_decode_uncompressed_mono( i16 *src, u32 count, float *dst )
{
- g_aud_dconfig = ma_device_config_init( ma_device_type_playback );
- g_aud_dconfig.playback.format = ma_format_f32;
- g_aud_dconfig.playback.channels = 2;
- g_aud_dconfig.sampleRate = 44100;
- g_aud_dconfig.dataCallback = audio_mixer_callback;
-
- g_aud_dconfig.pUserData = NULL;
-
- vg_info( "Starting audio engine\n" );
-
- if( ma_device_init( NULL, &g_aud_dconfig, &g_aud_device ) != MA_SUCCESS )
- {
- vg_exiterr( "ma_device failed to initialize" );
- }
- else
- {
- if( ma_device_start( &g_aud_device ) != MA_SUCCESS )
- {
- ma_device_uninit( &g_aud_device );
- vg_exiterr( "ma_device failed to start" );
- }
- }
+ 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);
+ }
}
-static void vg_audio_free(void)
+/*
+ * 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 )
{
- ma_device_uninit( &g_aud_device );
+ 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;
}
/*
- * thread 1
+ * ........ 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 sfx_system *sfx_alloc(void)
+VG_STATIC void audio_entity_get_samples( aatree_ptr id, u32 count, float *buf )
{
- if( sfx_sys_len >= SFX_MAX_SYSTEMS )
- return NULL;
-
- /*
- * A conditional is done against this in localization step,
- * Needs to be initialized.
- */
- sfx_sys[ sfx_sys_len ].source = NULL;
-
- return sfx_sys + (sfx_sys_len++);
+ 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 );
}
-/* Fetch samples into pcf */
-static void audio_mixer_getsamples( float *pcf, float *source, u32 cur, u32 ch )
+VG_STATIC void audio_entity_mix( aatree_ptr id, float *buffer,
+ u32 frame_count )
{
- if( ch == 2 )
- {
- pcf[0] = source[ cur*2+0 ];
- pcf[1] = source[ cur*2+1 ];
- }
- else
- {
- pcf[0] = source[ cur ];
- pcf[1] = source[ cur ];
- }
+ 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 );
}
/*
* callback from miniaudio.h interface
*/
-void audio_mixer_callback( ma_device *pDevice, void *pOutBuf,
- const void *pInput, ma_uint32 frameCount )
+VG_STATIC void audio_mixer_callback( ma_device *pDevice, void *pOutBuf,
+ const void *pInput, ma_uint32 frame_count )
{
- /* Process incoming sound queue */
- MUTEX_LOCK( sfx_mux_t01 );
-
- while( sfx_q_len --> 0 )
- {
- sfx_system *src = sfx_q[sfx_q_len];
- sfx_system *clone;
-
- src->in_queue = 0;
-
- clone = sfx_alloc();
- *clone = *src;
-
- /* Links need to exist on persistent sounds */
- clone->persisitent_source = src->flags & SFX_FLAG_PERSISTENT? src: NULL;
- }
-
- sfx_q_len = 0;
-
- /* Volume modifiers */
- for( int i = 0; i < sfx_sys_len; i ++ )
- {
- sfx_system *sys = sfx_sys + i;
-
- /* Apply persistent volume if linked */
- if( sys->flags & SFX_FLAG_PERSISTENT )
- {
- sys->vol = sys->persisitent_source->vol * g_master_volume;
- sys->pan = sys->persisitent_source->pan;
-
- /* Fadeout effect ( + remove ) */
- if( sys->persisitent_source->fadeout )
- {
- sys->fadeout_current = sys->persisitent_source->fadeout_current;
- sys->fadeout = sys->persisitent_source->fadeout;
-
- sys->persisitent_source = NULL;
- sys->flags &= ~SFX_FLAG_PERSISTENT;
- }
- }
-
- /* Apply volume slider if it has one linked */
- if( sys->vol_src )
- sys->cvol = sys->vol * sys->vol_src->val;
- else
- sys->cvol = sys->vol;
- }
-
- MUTEX_UNLOCK( sfx_mux_t01 );
-
- /* Clear buffer */
- float *pOut32F = (float *)pOutBuf;
- for( int i = 0; i < frameCount * 2; i ++ ){
- pOut32F[i] = 0.f;
- }
+ struct timespec time_start, time_end;
+ clock_gettime( CLOCK_REALTIME, &time_start );
- for( int i = 0; i < sfx_sys_len; i ++ )
- {
- sfx_system *sys = sfx_sys + i;
-
- u32 cursor = sys->cur, buffer_pos = 0;
- float pcf[2] = { 0.f, 0.0f };
-
- u32 frames_write = frameCount;
- float fadeout_divisor = 1.0f / (float)sys->fadeout;
-
- while( frames_write )
- {
- u32 samples_this_run = VG_MIN( frames_write, sys->end - cursor );
-
- if( sys->fadeout )
- {
- /* Force this system to be removed now */
- if( sys->fadeout_current == 0 )
- {
- sys->flags &= 0x00000000;
- sys->cur = sys->end;
- break;
- }
-
- samples_this_run = VG_MIN( samples_this_run, sys->fadeout_current );
- }
-
- for( u32 j=0; j<samples_this_run; j++ )
- {
- audio_mixer_getsamples( pcf, sys->source, cursor, sys->ch );
-
- float vol = sys->cvol;
-
- if( sys->fadeout )
- {
- vol *= (float)sys->fadeout_current * fadeout_divisor;
- sys->fadeout_current --;
- }
-
- if( buffer_pos >= frameCount )
- {
- break;
- }
-
- float sl = 1.0f-sys->pan,
- sr = 1.0f+sys->pan;
-
- pOut32F[ buffer_pos*2+0 ] += pcf[0] * vol * sl;
- pOut32F[ buffer_pos*2+1 ] += pcf[1] * vol * sr;
-
- cursor ++;
- buffer_pos ++;
- }
-
- frames_write -= samples_this_run;
-
- if( sys->flags & SFX_FLAG_REPEAT )
- {
- if( frames_write )
- {
- cursor = sys->clip_start;
- continue;
- }
- }
-
- sys->cur = cursor;
- break;
- }
- }
+ audio_system_enque();
+
+ /* Clear buffer */
+ float *pOut32F = (float *)pOutBuf;
+ for( int i=0; i<frame_count*2; i ++ )
+ pOut32F[i] = 0.0f;
- /* Redistribute sound systems */
- MUTEX_LOCK( sfx_mux_t01 );
+ /* Mix all sounds */
+ for( int i=0; i<SFX_MAX_SYSTEMS; i ++ )
+ {
+ struct active_audio_player *aap = &vg_audio.active_players[i];
- u32 idx = 0, wr = 0;
- while( idx != sfx_sys_len )
- {
- sfx_system *src = sfx_sys + idx;
-
- /* Keep only if cursor is before end or repeating */
- if( src->cur < src->end || (src->flags & SFX_FLAG_REPEAT) )
- {
- sfx_sys[ wr ++ ] = sfx_sys[ idx ];
- }
-
- idx ++ ;
- }
- sfx_sys_len = wr;
-
- MUTEX_UNLOCK( sfx_mux_t01 );
-
- (void)pInput;
+ if( aap->active )
+ {
+ audio_entity_mix( i, pOut32F, frame_count );
+ }
+ }
+
+ /* redistribute */
+ audio_system_cleanup();
+
+ /* TODO: what the hell is this? */
+ (void)pInput;
+
+
+ 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();
}
-/*
- * Load strings into sfx_set's memory
- * String layout: "sounda.ogg\0soundb.ogg\0soundc.ogg\0\0"
+VG_STATIC void audio_clip_load( audio_clip *clip )
+{
+ 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 );
+
+ 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( vg_audio.audio_pool, 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( vg_audio.audio_pool, 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 )
+{
+ for( int i=0; i<count; i++ )
+ audio_clip_load( &arr[i] );
+}
+
+/* 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;
+
+ 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;
+
+ 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 );
+}
+
+#if 0
+VG_STATIC void audio_player_playoneshot( audio_player *player, audio_clip *clip )
+{
+ audio_require_lock();
+ audio_require_init( player );
+}
+#endif
+
+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
*/
-static void sfx_set_strings( sfx_set *dest, char *strSources,
- u32 flags, int bAsync )
+
+VG_STATIC int audio_player_is_playing( audio_player *sys )
{
- dest->ch = (flags & SFX_FLAG_STEREO)? 2: 1;
-
- dest->main = NULL;
- dest->numsegments = 0;
- char *source = strSources;
-
- u32 total = 0;
- int len;
- while( (len = strlen( source )) )
- {
- u32 samples;
- float *sound = sfx_vorbis( source, dest->ch, &samples );
-
- if( !sound )
- {
- free( dest->main );
- dest->numsegments = 0;
- return;
- }
-
- total += samples;
-
- float *nbuf = realloc( dest->main, total * dest->ch * sizeof(float) );
-
- if( nbuf )
- {
- dest->main = nbuf;
- memcpy( dest->main + (total-samples)*dest->ch,
- sound, samples*dest->ch*sizeof(float) );
- free( sound );
-
- dest->segments[ dest->numsegments*2+0 ] = total-samples;
- dest->segments[ dest->numsegments*2+1 ] = total;
- }
- else
- {
- vg_error( "realloc() failed\n" );
- free( sound );
- return;
- }
-
- source += len +1;
- dest->numsegments ++;
- }
+ audio_require_lock();
+
+ if( sys->active_entity != AATREE_PTR_NIL )
+ return 1;
+ else
+ return 0;
}
-static void sfx_set_init( sfx_set *dest, char *sources )
+VG_STATIC void audio_player_set_position( audio_player *sys, v3f pos )
{
- if( !sources )
- sfx_set_strings( dest, dest->sources, dest->flags, 0 );
- else
- sfx_set_strings( dest, sources, dest->flags, 0 );
+ audio_require_lock();
+ v3_copy( pos, sys->info.world_position );
}
-static void sfx_set_play( sfx_set *source, sfx_system *sys, int id )
+VG_STATIC void audio_player_set_vol( audio_player *sys, float vol )
{
- if( sfx_begin_edit( sys ) )
- {
- sys->fadeout = 0;
- sys->fadeout_current = 0;
- sys->source = source->main;
- sys->cur = source->segments[ id*2 + 0 ];
- sys->end = source->segments[ id*2 + 1 ];
- sys->ch = source->ch;
-
- /* for diagnostics */
- sys->clip_start = sys->cur;
- sys->clip_end = sys->end;
- sys->buffer_length = source->segments[ (source->numsegments-1)*2 + 1 ];
-
- sfx_push( sys );
- }
+ 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;
}
-/* Pick a random sound from the buffer and play it into system */
-static void sfx_set_playrnd( sfx_set *source, sfx_system *sys,
- int min_id, int max_id )
+VG_STATIC float audio_player_get_pan( audio_player *sys )
{
- if( !source->numsegments )
- return;
+ 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
+ */
- if( max_id > source->numsegments )
+VG_STATIC void audio_debug_ui( m4x4f mtx_pv )
+{
+ if( !vg_audio.debug_ui )
+ return;
+
+ /* Get data */
+ struct sound_info
{
- vg_error( "Max ID out of range for playrnd\n" );
- return;
+ const char *name;
+ u32 cursor, flags, length;
+ v3f pos;
+ float vol;
}
+ infos[ SFX_MAX_SYSTEMS ];
+ int num_systems = 0;
- int pick = (rand() % (max_id-min_id)) + min_id;
+ audio_lock();
- sfx_set_play( source, sys, pick );
-}
+ for( int i=0; i<SFX_MAX_SYSTEMS; i ++ )
+ {
+ struct active_audio_player *aap = &vg_audio.active_players[i];
-static void sfx_system_fadeout( sfx_system *sys, u32 length_samples )
-{
- if( sfx_begin_edit( sys ) )
+ 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;
+
+ 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 ++ )
{
- sys->fadeout_current = length_samples;
- sys->fadeout = length_samples;
+ struct sound_info *inf = &infos[i];
+
+ vg_uictx.cursor[2] = 200;
+ vg_uictx.cursor[3] = 18;
- sfx_end_edit( sys );
+ u32 alpha = 0xa0000000;
+
+ ui_new_node();
+ {
+ ui_fill_rect( vg_uictx.cursor, 0x00333333|alpha );
+
+ ui_px baseline = vg_uictx.cursor[0],
+ w = 200,
+ c = baseline + ((float)inf->cursor / (float)inf->length) * w;
+
+ /* cursor */
+ vg_uictx.cursor[2] = 2;
+ vg_uictx.cursor[0] = c;
+ ui_fill_rect( vg_uictx.cursor, 0xffffffff );
+
+ vg_uictx.cursor[0] = baseline + 2;
+ vg_uictx.cursor[1] += 2;
+ snprintf( perf, 127, "%s %.1f%%", infos[i].name, infos[i].vol );
+ ui_text( vg_uictx.cursor, perf, 1, 0 );
+
+ if( inf->flags & AUDIO_FLAG_SPACIAL_3D )
+ {
+ v4f wpos;
+ v3_copy( inf->pos, wpos );
+ wpos[3] = 1.0f;
+ m4x4_mulv( mtx_pv, wpos, wpos );
+
+ 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;
}
}
-static void sfx_set_free( sfx_set *set )
-{
- free( set->main );
-}
+#endif /* VG_AUDIO_H */
projects / vg.git / blobdiff