#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_console.h"
#include "vg/vg_store.h"
-#include <time.h>
+#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 "stb/stb_vorbis.h"
+#ifdef __GNUC__
+ #ifndef __clang__
+ #pragma GCC pop_options
+ #pragma GCC diagnostic pop
+ #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
-#define FADEOUT_LENGTH 4410
+#define FADEOUT_LENGTH 1100
#define FADEOUT_DIVISOR (1.0f/(float)FADEOUT_LENGTH)
#define AUDIO_DECODE_SIZE (1024*256) /* 256 kb decoding buffers */
enum audio_source_mode
{
k_audio_source_mono,
- k_audio_source_mono_compressed,
- k_audio_source_stereo_compressed
+ k_audio_source_compressed,
};
typedef struct audio_clip audio_clip;
const char *path;
enum audio_source_mode source_mode;
- /* result */
+ u32 size;
void *data;
- u32 len; /* decompressed: sample count,
- compressed: file size */
};
typedef struct audio_mix_info audio_mix_info;
{
aatree_ptr active_entity; /* non-nil if currently playing */
audio_mix_info info;
- int enqued;
+ int enqued, init;
/* Diagnostic */
const char *name;
ma_device miniaudio_device;
ma_device_config miniaudio_dconfig;
- void *mem, *decode_mem;
- u32 mem_current,
- mem_total;
+ void *audio_pool,
+ *decode_buffer;
+ u32 samples_last;
/* synchro */
int sync_locked;
- MUTEX_TYPE mutex_checker;
- MUTEX_TYPE mutex_sync;
+
+ vg_mutex mux_checker,
+ mux_sync;
/* Audio engine, thread 1 */
struct active_audio_player
/* System queue, and access from thread 0 */
audio_entity entity_queue[SFX_MAX_SYSTEMS];
int queue_len;
-
- char performance_info[128];
- int debug_ui;
+ int debug_ui, debug_ui_3d;
v3f listener_pos,
listener_ears;
}
vg_audio;
-static void *audio_alloc( u32 size )
-{
- u32 new_current = vg_audio.mem_current + size;
- if( new_current > vg_audio.mem_total )
- {
- vg_error( "audio pool over budget!\n" );
- free( vg_audio.mem );
- return NULL;
- }
-
- void *ptr = vg_audio.mem + vg_audio.mem_current;
- vg_audio.mem_current = new_current;
-
- return ptr;
-}
-
+static struct vg_profile
+ _vg_prof_audio_decode = {.mode = k_profile_mode_accum,
+ .name = "[T2] audio_decode()"},
+ _vg_prof_audio_mix = {.mode = k_profile_mode_accum,
+ .name = "[T2] audio_mix()"},
+ vg_prof_audio_decode,
+ vg_prof_audio_mix;
/*
* These functions are called from the main thread and used to prevent bad
* access. TODO: They should be no-ops in release builds.
*/
-static int audio_lock_checker_load(void)
+VG_STATIC int audio_lock_checker_load(void)
{
int value;
- MUTEX_LOCK( vg_audio.mutex_checker );
+ vg_mutex_lock( &vg_audio.mux_checker );
value = vg_audio.sync_locked;
- MUTEX_UNLOCK( vg_audio.mutex_checker );
+ vg_mutex_unlock( &vg_audio.mux_checker );
return value;
}
-static void audio_lock_checker_store( int value )
+VG_STATIC void audio_lock_checker_store( int value )
{
- MUTEX_LOCK( vg_audio.mutex_checker );
+ vg_mutex_lock( &vg_audio.mux_checker );
vg_audio.sync_locked = value;
- MUTEX_UNLOCK( vg_audio.mutex_checker );
+ vg_mutex_unlock( &vg_audio.mux_checker );
}
-static void audio_require_lock(void)
+VG_STATIC void audio_require_lock(void)
{
if( audio_lock_checker_load() )
return;
- vg_exiterr( "Modifying sound effects systems requires locking\n" );
+ vg_error( "Modifying sound effects systems requires locking\n" );
+ abort();
}
-static void audio_lock(void)
+VG_STATIC void audio_lock(void)
{
- MUTEX_LOCK( vg_audio.mutex_sync );
+ vg_mutex_lock( &vg_audio.mux_sync );
audio_lock_checker_store(1);
}
-static void audio_unlock(void)
+VG_STATIC void audio_unlock(void)
{
audio_lock_checker_store(0);
- MUTEX_UNLOCK( vg_audio.mutex_sync );
+ vg_mutex_unlock( &vg_audio.mux_sync );
}
-static void audio_mixer_callback( ma_device *pDevice, void *pOutBuf,
+VG_STATIC void audio_mixer_callback( ma_device *pDevice, void *pOutBuf,
const void *pInput, ma_uint32 frameCount );
-static int vg_audio_init(void)
+VG_STATIC void vg_audio_init(void)
{
+ 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,
.persistent = 1
});
- u32 decode_region = AUDIO_DECODE_SIZE * SFX_MAX_SYSTEMS;
- vg_audio.mem_total = 1024*1024*32;
- vg_audio.mem_current = 0;
- vg_audio.mem = malloc( vg_audio.mem_total + decode_region );
- vg_audio.decode_mem = &((u8 *)vg_audio.mem)[vg_audio.mem_total];
+ /* 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;
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 = 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;
if( ma_device_init( NULL, dconf, device) != MA_SUCCESS )
{
- vg_error( "ma_device failed to initialize" );
- return 0;
+ vg_fatal_exit_loop( "(audio) ma_device failed to initialize" );
}
else
{
if( ma_device_start( device ) != MA_SUCCESS )
{
ma_device_uninit( device );
- vg_error( "ma_device failed to start" );
- return 0;
+ vg_fatal_exit_loop( "(audio) ma_device failed to start" );
}
}
-
- return 1;
+
+ vg_success( "Ready\n" );
}
-static void vg_audio_free(void)
+VG_STATIC void vg_audio_free(void * nothing)
{
ma_device *device = &vg_audio.miniaudio_device;
ma_device_uninit( device );
- free( vg_audio.mem );
+#if 0
+ vg_free( vg_audio.mem );
+ vg_audio.mem = NULL;
+#endif
}
/*
return playerid;
}
-static void audio_entity_free_internal( aatree_ptr id )
+VG_STATIC void audio_entity_free_internal( aatree_ptr id )
{
struct active_audio_player *aap = &vg_audio.active_players[ id ];
aap->active = 0;
&vg_audio.active_pool_head );
}
-static void *audio_entity_vorbis_ptr( aatree_ptr entid )
+VG_STATIC void *audio_entity_vorbis_ptr( aatree_ptr entid )
{
- u8 *buf = (u8*)vg_audio.decode_mem,
+ u8 *buf = (u8*)vg_audio.decode_buffer,
*loc = &buf[AUDIO_DECODE_SIZE*entid];
return (void *)loc;
}
-static void audio_entity_start( audio_entity *src )
+VG_STATIC void audio_entity_start( audio_entity *src )
{
aatree_ptr entid = audio_alloc_entity_internal();
if( entid == AATREE_PTR_NIL )
audio_entity *ent = &vg_audio.active_players[ entid ].ent;
ent->info = src->info;
- ent->name = "todo";
+ ent->name = src->info.source->path;
ent->cur = 0;
ent->player = src->player;
src->player->active_entity = entid;
}
- if( src->info.source->source_mode == k_audio_source_mono_compressed ||
- src->info.source->source_mode == k_audio_source_stereo_compressed )
+ if( src->info.source->source_mode == k_audio_source_compressed )
{
/* Setup vorbis decoder */
struct active_audio_player *aap = &vg_audio.active_players[ entid ];
int err;
stb_vorbis *decoder = stb_vorbis_open_memory(
- src->info.source->data, src->info.source->len, &err, &alloc );
+ src->info.source->data,
+ src->info.source->size, &err, &alloc );
if( !decoder )
{
}
else
{
- ent->length = src->info.source->len;
+ ent->length = src->info.source->size;
}
}
/*
* Read everything from the queue
*/
-static void audio_system_enque(void)
+VG_STATIC void audio_system_enque(void)
{
/* Process incoming sound queue */
audio_lock();
audio_entity *ent = &vg_audio.active_players[ entid ].ent;
if( !ent->fadeout )
{
- ent->fadeout = 1;
+ ent->fadeout = FADEOUT_LENGTH;
ent->fadeout_current = FADEOUT_LENGTH;
}
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;
+ }
}
}
}
/*
* Redistribute sound systems
*/
-static void audio_system_cleanup(void)
+VG_STATIC void audio_system_cleanup(void)
{
audio_lock();
/*
* Get effective volume and pan from this entity
*/
-static void audio_entity_spacialize( audio_entity *ent, float *vol, float *pan )
+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 dist = v3_length( delta ),
- attn = (dist / ent->info.vol) +1.0f;
+ float dist2 = v3_length2( delta );
- v3_muls( delta, 1.0f/dist, delta );
+ if( dist2 < 0.0001f )
+ {
+ *pan = 0.0f;
+ *vol = 1.0f;
+ }
+ else
+ {
+ float dist = sqrtf( dist2 ),
+ attn = (dist / ent->info.vol) +1.0f;
- *pan = v3_dot( vg_audio.listener_ears, delta );
- *vol = 1.0f/(attn*attn);
+ v3_muls( delta, 1.0f/dist, delta );
+ *pan = v3_dot( vg_audio.listener_ears, delta );
+ *vol = 1.0f/(attn*attn);
+ }
}
-static void audio_decode_uncompressed_mono( float *src, u32 count, float *dst )
+VG_STATIC void audio_decode_uncompressed_mono( i16 *src, u32 count, float *dst )
{
for( u32 i=0; i<count; i++ )
{
- dst[ i*2 + 0 ] = src[i];
- dst[ i*2 + 1 ] = src[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 void audio_entity_get_samples( aatree_ptr id, u32 count, float *buf )
+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;
remaining -= samples_this_run;
float *dst = &buf[ buffer_pos * 2 ];
+
+ int source_mode = ent->info.source->source_mode;
- if( ent->info.source->source_mode == k_audio_source_mono )
+ if( source_mode == k_audio_source_mono )
{
- float *src = &((float *)ent->info.source->data)[ cursor ];
+ i16 *src_buffer = ent->info.source->data,
+ *src = &src_buffer[cursor];
+
audio_decode_uncompressed_mono( src, samples_this_run, dst );
}
- else if( ent->info.source->source_mode == k_audio_source_mono_compressed )
+ else if( source_mode == k_audio_source_compressed )
{
- int read_samples = stb_vorbis_get_samples_float_interleaved(
+ int read_samples = stb_vorbis_get_samples_float_interleaved_stereo(
aap->vorbis_handle,
- 2,
dst,
- samples_this_run * 2 );
+ samples_this_run );
if( read_samples != samples_this_run )
{
if( (ent->info.flags & AUDIO_FLAG_LOOP) && remaining )
{
- if( ent->info.source->source_mode == k_audio_source_mono_compressed ||
- ent->info.source->source_mode == k_audio_source_stereo_compressed )
+ if( source_mode == k_audio_source_compressed )
{
stb_vorbis_seek_start( aap->vorbis_handle );
}
}
ent->cur = cursor;
+ vg_profile_end( &_vg_prof_audio_decode );
}
-static void audio_entity_mix( aatree_ptr id, float *buffer,
- u32 frame_count )
+VG_STATIC void audio_entity_mix( aatree_ptr id, float *buffer,
+ u32 frame_count )
{
audio_entity *ent = &vg_audio.active_players[id].ent;
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 );
buffer_pos ++;
}
-}
-static void vg_sleep_ms( long msec )
-{
- struct timespec ts;
-
- ts.tv_sec = msec / 1000;
- ts.tv_nsec = (msec % 1000) * 1000000;
- nanosleep( &ts, &ts );
+ vg_profile_end( &_vg_prof_audio_mix );
}
/*
* callback from miniaudio.h interface
*/
-static void audio_mixer_callback( ma_device *pDevice, void *pOutBuf,
+VG_STATIC void audio_mixer_callback( ma_device *pDevice, void *pOutBuf,
const void *pInput, ma_uint32 frame_count )
{
struct timespec time_start, time_end;
struct active_audio_player *aap = &vg_audio.active_players[i];
if( aap->active )
+ {
audio_entity_mix( i, pOut32F, frame_count );
+ }
}
-
-#if 0
- vg_sleep_ms( 20 );
-#endif
/* redistribute */
audio_system_cleanup();
/* TODO: what the hell is this? */
(void)pInput;
- /*
- * Debug information
- */
- clock_gettime( CLOCK_REALTIME, &time_end );
- double elapsed = 1000.0*time_end.tv_sec + 1e-6*time_end.tv_nsec
- - (1000.0*time_start.tv_sec + 1e-6*time_start.tv_nsec),
- budget = ((double)frame_count / 44100.0) * 1000.0,
- percent = (elapsed/budget) * 100.0;
+ 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;
- snprintf( vg_audio.performance_info, 127,
- "%.1fms/%.1fms (%.1f%%) (%u frames)",
- elapsed, budget, percent, frame_count );
+ vg_audio.samples_last = frame_count;
+ audio_unlock();
}
-/* Decompress entire vorbis stream into buffer */
-static float *audio_decompress_vorbis( const unsigned char *data, int len,
- int channels, u32 *samples )
+VG_STATIC void audio_clip_load( audio_clip *clip, void *lin_alloc )
{
- 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 );
+ if( lin_alloc == NULL )
+ lin_alloc = vg_audio.audio_pool;
- vg_info( "decompress_vorbis: %u samples (%.2fs), %.1fkb\n",
- length_samples,
- (float)length_samples / (44100.0f*(float)channels),
- (float)(length_samples*4*channels) / 1024.0f );
-
- float *buffer = audio_alloc( length_samples * channels * sizeof(float) );
- if( !buffer )
+ if( clip->source_mode == k_audio_source_mono )
{
- stb_vorbis_close( pv );
- vg_exit();
- }
-
- int read_samples = stb_vorbis_get_samples_float_interleaved(
- pv, channels, buffer, length_samples * channels );
+ vg_linear_clear( vg_mem.scratch );
+ u32 fsize;
- 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;
-}
+ stb_vorbis_alloc alloc = {
+ .alloc_buffer = vg_linear_alloc( vg_mem.scratch, AUDIO_DECODE_SIZE ),
+ .alloc_buffer_length_in_bytes = AUDIO_DECODE_SIZE
+ };
-static int audio_clip_load( audio_clip *clip )
-{
- /* Load and decompress */
- i64 file_len;
- void *filedata = vg_asset_read_s( clip->path, &file_len );
+ void *filedata = vg_file_read( vg_mem.scratch, clip->path, &fsize );
- if( !filedata )
- {
- vg_error( "OGG load failed (%s)\n", clip->path );
- return 0;
- }
-
- if( clip->source_mode == k_audio_source_mono )
- {
- u32 samples;
- float *sound = audio_decompress_vorbis( filedata, file_len, 1, &samples );
- clip->data = sound;
- clip->len = samples;
+ int err;
+ stb_vorbis *decoder = stb_vorbis_open_memory(
+ filedata, fsize, &err, &alloc );
- float seconds = (float)samples / 44100.0f,
- mb = (float)(samples*4) / (1024.0f*1024.0f);
+ if( !decoder )
+ {
+ vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
+ clip->path, err );
+ vg_fatal_exit_loop( "Vorbis decode error" );
+ }
- vg_info( "Loaded audio clip[mono] '%s' (%.1fs, %.1fmb)\n",
- clip->path, seconds, mb );
- }
- else if( clip->source_mode == k_audio_source_mono_compressed )
- {
- void *data = audio_alloc( file_len );
- memcpy( data, filedata, file_len );
+ /* only mono is supported in uncompressed */
+ u32 length_samples = stb_vorbis_stream_length_in_samples( decoder ),
+ data_size = length_samples * sizeof(i16);
- clip->data = data;
- clip->len = file_len;
+ audio_lock();
+ clip->data = vg_linear_alloc( lin_alloc, data_size );
+ clip->size = length_samples;
+ audio_unlock();
- float mb = (float)(file_len) / (1024.0f*1024.0f);
- vg_info( "Loaded audio clip[mono_compressed] '%s' (%.1fmb)\n",
- clip->path, mb );
- }
- else if( clip->source_mode == k_audio_source_stereo_compressed )
- {
- /* ... */
+ int read_samples = stb_vorbis_get_samples_i16_downmixed(
+ decoder, clip->data, length_samples );
- clip->data = NULL;
- clip->len = 0;
+ if( read_samples != length_samples )
+ vg_fatal_exit_loop( "Decode error" );
- vg_error( "Source mode (%u) currently unsupported\n", clip->source_mode );
- return 0;
+ float mb = (float)(data_size) / (1024.0f*1024.0f);
+ vg_info( "Loaded audio clip '%s' (%.1fmb) %u samples\n", clip->path, mb,
+ length_samples );
}
- else
+
+ /* 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();
- clip->data = NULL;
- clip->len = 0;
+ if( !clip->data )
+ vg_fatal_exit_loop( "Audio failed to load" );
- vg_error( "Unkown source mode (%u)\n", clip->source_mode );
- return 0;
+ float mb = (float)(clip->size) / (1024.0f*1024.0f);
+ vg_info( "Loaded audio clip '%s' (%.1fmb)\n", clip->path, mb );
}
-
- return 1;
}
-static void audio_clip_loadn( audio_clip *arr, int count )
+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] );
-}
-
-#if 0
-/*
- * Client code
- */
-static void audio_pack_play( audio_pack *source, audio_player *sys, int id )
-{
- audio_require_lock();
-
- sys->fadeout = 0;
- sys->fadeout_current = 0;
- sys->source = source->data;
- sys->cur = source->segments[ id*2 + 0 ];
- sys->end = source->segments[ id*2 + 1 ];
- sys->ch = source->ch;
- sys->source_mode = source->source_mode;
-
- /* for diagnostics */
- sys->clip_start = sys->cur;
- sys->clip_end = sys->end;
- sys->buffer_length = source->segments[ (source->numsegments-1)*2 + 1 ];
- sys->is_playing = 1;
-
- audio_player_push( sys );
+ audio_clip_load( &arr[i], lin_alloc );
}
-#endif
-
/* Mark change to be uploaded through queue system */
-static void audio_player_commit( audio_player *sys )
+VG_STATIC void audio_player_commit( audio_player *sys )
{
audio_require_lock();
if( sys->enqued )
{
- vg_warn( "Audio commit spamming; already enqued (%s)\n", sys->name );
+ vg_warn( "[2] Audio commit spamming; already enqued (%s)\n", sys->name );
return;
}
audio_entity *ent = &vg_audio.entity_queue[ vg_audio.queue_len ++ ];
ent->info = sys->info;
ent->player = sys;
- sys->active_entity = AATREE_PTR_NIL;
+}
+
+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 */
-static void audio_player_playclip( audio_player *player, audio_clip *clip )
+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 );
}
-static void audio_player_playoneshot( audio_player *player, audio_clip *clip )
+#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;
}
/*
* Safety enforced Get/set attributes
*/
-static void audio_player_set_position( audio_player *sys, v3f pos )
+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 );
}
-static void audio_player_set_vol( audio_player *sys, float vol )
+VG_STATIC void audio_player_set_vol( audio_player *sys, float vol )
{
audio_require_lock();
sys->info.vol = vol;
}
-static float audio_player_get_vol( audio_player *sys )
+VG_STATIC float audio_player_get_vol( audio_player *sys )
{
audio_require_lock();
return sys->info.vol;
}
-static void audio_player_set_pan( audio_player *sys, float pan )
+VG_STATIC void audio_player_set_pan( audio_player *sys, float pan )
{
audio_require_lock();
sys->info.pan = pan;
}
-static float audio_player_get_pan( audio_player *sys )
+VG_STATIC float audio_player_get_pan( audio_player *sys )
{
audio_require_lock();
return sys->info.pan;
}
-static void audio_player_set_flags( audio_player *sys, u32 flags )
+VG_STATIC void audio_player_set_flags( audio_player *sys, u32 flags )
{
audio_require_lock();
sys->info.flags = flags;
}
-static u32 audio_player_get_flags( audio_player *sys )
+VG_STATIC u32 audio_player_get_flags( audio_player *sys )
{
audio_require_lock();
return sys->info.flags;
* Debugging
*/
-static void audio_debug_ui(void)
+VG_STATIC void audio_debug_ui( m4x4f mtx_pv )
{
if( !vg_audio.debug_ui )
return;
{
const char *name;
u32 cursor, flags, length;
+ v3f pos;
float vol;
}
infos[ SFX_MAX_SYSTEMS ];
int num_systems = 0;
- char perf[128];
-
audio_lock();
for( int i=0; i<SFX_MAX_SYSTEMS; i ++ )
snd->flags = ent->info.flags;
snd->length = ent->length;
snd->vol = ent->info.vol*100.0f;
+ v3_copy( ent->info.world_position, snd->pos );
}
- strcpy( perf, vg_audio.performance_info );
+
+ 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 */
- ui_global_ctx.cursor[0] = 10;
- ui_global_ctx.cursor[1] = 10;
- ui_global_ctx.cursor[2] = 150;
- ui_global_ctx.cursor[3] = 12;
- ui_text( &ui_global_ctx, ui_global_ctx.cursor, perf, 1, 0 );
+ 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 usage = (float)vg_audio.mem_current / (1024.0f*1024.0f),
- total = (float)vg_audio.mem_total / (1024.0f*1024.0f),
- percent = (usage/total) * 100.0f;
+ 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_global_ctx.cursor[1] += 20;
- ui_text( &ui_global_ctx, ui_global_ctx.cursor, perf, 1, 0 );
+ ui_text( vg_uictx.cursor, perf, 1, 0 );
+ vg_uictx.cursor[1] += 20;
- ui_global_ctx.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];
- ui_global_ctx.cursor[2] = 150;
- ui_global_ctx.cursor[3] = 12;
+ vg_uictx.cursor[2] = 200;
+ vg_uictx.cursor[3] = 18;
u32 alpha = 0xa0000000;
- ui_new_node( &ui_global_ctx );
+ ui_new_node();
{
- ui_fill_rect( &ui_global_ctx, ui_global_ctx.cursor, 0x00333333|alpha );
+ ui_fill_rect( vg_uictx.cursor, 0x00333333|alpha );
- ui_px baseline = ui_global_ctx.cursor[0],
- w = 150,
+ ui_px baseline = vg_uictx.cursor[0],
+ w = 200,
c = baseline + ((float)inf->cursor / (float)inf->length) * w;
/* cursor */
- ui_global_ctx.cursor[2] = 2;
- ui_global_ctx.cursor[0] = c;
- ui_fill_rect( &ui_global_ctx, ui_global_ctx.cursor, 0xffffffff );
+ vg_uictx.cursor[2] = 2;
+ vg_uictx.cursor[0] = c;
+ ui_fill_rect( vg_uictx.cursor, 0xffffffff );
- ui_global_ctx.cursor[0] = baseline + 2;
- ui_global_ctx.cursor[1] += 2;
+ vg_uictx.cursor[0] = baseline + 2;
+ vg_uictx.cursor[1] += 2;
snprintf( perf, 127, "%s %.1f%%", infos[i].name, infos[i].vol );
- ui_text( &ui_global_ctx, ui_global_ctx.cursor, perf, 1, 0 );
+ 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 ++ ] );
+ }
}
- ui_end_down( &ui_global_ctx );
- ui_global_ctx.cursor[1] += 1;
+projected_behind:
+
+ ui_end_down();
+ vg_uictx.cursor[1] += 1;
}
}
projects / vg.git / blobdiff