#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__
- #pragma GCC push_options
- #pragma GCC optimize ("O3")
+ #ifndef __clang__
+ #pragma GCC push_options
+ #pragma GCC optimize ("O3")
+ #pragma GCC diagnostic push
+ #pragma GCC diagnostic ignored "-Wdeprecated-declarations"
+ #endif
#endif
-#pragma GCC diagnostic push
-#pragma GCC diagnostic ignored "-Wdeprecated-declarations"
-
#define STB_VORBIS_MAX_CHANNELS 2
#include "stb/stb_vorbis.h"
-#pragma GCC diagnostic pop
-
#ifdef __GNUC__
- #pragma GCC pop_options
+ #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 1100
#define FADEOUT_DIVISOR (1.0f/(float)FADEOUT_LENGTH)
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;
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 */
vg_prof_audio_decode,
vg_prof_audio_mix;
-static void *audio_alloc( u32 size )
-{
- u32 new_current = vg_audio.mem_current + size;
- if( new_current > vg_audio.mem_total )
- vg_fatal_exit_loop( "Audio pool ran out of memory" );
-
- void *ptr = vg_audio.mem + vg_audio.mem_current;
- vg_audio.mem_current = new_current;
-
- return ptr;
-}
-
-
/*
* 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;
vg_mutex_lock( &vg_audio.mux_checker );
return value;
}
-static void audio_lock_checker_store( int 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 );
}
-static void audio_require_lock(void)
+VG_STATIC void audio_require_lock(void)
{
if( audio_lock_checker_load() )
return;
abort();
}
-static void audio_lock(void)
+VG_STATIC void audio_lock(void)
{
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);
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 void 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
});
- vg_convar_push( (struct vg_convar){
- .name = "debug_audio_clips",
- .data = &vg_audio.debug_ui_3d,
- .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 );
- u32 decode_region = AUDIO_DECODE_SIZE * SFX_MAX_SYSTEMS;
- vg_audio.mem_total = 1024*1024*32;
- vg_audio.mem_current = 0;
- vg_audio.mem = vg_alloc( vg_audio.mem_total + decode_region );
- vg_audio.decode_mem = &((u8 *)vg_audio.mem)[vg_audio.mem_total];
+ /* 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_success( "Ready\n" );
}
-static void vg_audio_free(void * nothing)
+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
}
/*
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 )
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();
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 )
{
}
}
-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
*/
-static int
+VG_STATIC int
stb_vorbis_get_samples_float_interleaved_stereo( stb_vorbis *f, float *buffer,
int len )
{
return n;
}
-static void audio_entity_get_samples( aatree_ptr id, u32 count, float *buf )
+/*
+ * ........ 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;
+}
+
+VG_STATIC void audio_entity_get_samples( aatree_ptr id, u32 count, float *buf )
{
vg_profile_begin( &_vg_prof_audio_decode );
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( source_mode == k_audio_source_compressed )
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;
/*
* 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;
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_error( "Failed to allocated memory for audio\n" );
- return NULL;
- }
-
- 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;
- }
+ int err;
+ stb_vorbis *decoder = stb_vorbis_open_memory(
+ filedata, fsize, &err, &alloc );
- if( clip->source_mode == k_audio_source_mono )
- {
- u32 samples = 0;
- float *sound = audio_decompress_vorbis( filedata, file_len, 1, &samples );
- clip->data = sound;
- clip->len = samples;
+ if( !decoder )
+ {
+ vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
+ clip->path, err );
+ vg_fatal_exit_loop( "Vorbis decode error" );
+ }
- float seconds = (float)samples / 44100.0f,
- mb = (float)(samples*4) / (1024.0f*1024.0f);
+ /* only mono is supported in uncompressed */
+ u32 length_samples = stb_vorbis_stream_length_in_samples( decoder ),
+ data_size = length_samples * sizeof(i16);
- vg_info( "Loaded audio clip[mono] '%s' (%.1fs, %.1fmb)\n",
- clip->path, seconds, mb );
- }
- else if( clip->source_mode == k_audio_source_compressed )
- {
- void *data = audio_alloc( file_len );
- memcpy( data, filedata, file_len );
+ 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 );
- clip->data = data;
- clip->len = file_len;
+ if( read_samples != length_samples )
+ vg_fatal_exit_loop( "Decode error" );
- float mb = (float)(file_len) / (1024.0f*1024.0f);
- vg_info( "Loaded audio clip[compressed] '%s' (%.1fmb)\n",
- clip->path, mb );
+ 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] );
+ audio_clip_load( &arr[i], lin_alloc );
}
/* 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;
}
ent->player = sys;
}
-static void audio_require_init( audio_player *player )
+VG_STATIC void audio_require_init( audio_player *player )
{
if( player->init )
return;
vg_fatal_exit_loop( "Must init audio player before playing! \n" );
}
-static void audio_require_clip_loaded( audio_clip *clip )
+VG_STATIC void audio_require_clip_loaded( audio_clip *clip )
{
- if( clip->data )
+ 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 );
}
#if 0
-static void audio_player_playoneshot( audio_player *player, audio_clip *clip )
+VG_STATIC void audio_player_playoneshot( audio_player *player, audio_clip *clip )
{
audio_require_lock();
audio_require_init( player );
}
#endif
-static void audio_play_oneshot( audio_clip *clip, float volume )
+VG_STATIC void audio_play_oneshot( audio_clip *clip, float volume )
{
audio_require_lock();
audio_require_clip_loaded( clip );
ent->player = NULL;
}
-static void audio_player_init( audio_player *player )
+VG_STATIC void audio_player_init( audio_player *player )
{
player->active_entity = AATREE_PTR_NIL;
player->init = 1;
* Safety enforced Get/set attributes
*/
-static int audio_player_is_playing( audio_player *sys )
+VG_STATIC int audio_player_is_playing( audio_player *sys )
{
audio_require_lock();
return 0;
}
-static void audio_player_set_position( audio_player *sys, v3f pos )
+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( m4x4f mtx_pv )
+VG_STATIC void audio_debug_ui( m4x4f mtx_pv )
{
if( !vg_audio.debug_ui )
return;
char perf[128];
/* Draw UI */
- ui_global_ctx.cursor[0] = 258;
- ui_global_ctx.cursor[1] = VG_PROFILE_SAMPLE_COUNT*2+8+24+12;
- ui_global_ctx.cursor[2] = 150;
- ui_global_ctx.cursor[3] = 12;
+ 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_text( &ui_global_ctx, ui_global_ctx.cursor, perf, 1, 0 );
- ui_global_ctx.cursor[1] += 20;
+ ui_text( vg_uictx.cursor, perf, 1, 0 );
+ vg_uictx.cursor[1] += 20;
ui_rect overlap_buffer[ SFX_MAX_SYSTEMS ];
u32 overlap_length = 0;
- if( !vg_audio.debug_ui_3d )
- return;
-
/* Draw audio stack */
for( int i=0; i<num_systems; i ++ )
{
struct sound_info *inf = &infos[i];
- ui_global_ctx.cursor[2] = 200;
- ui_global_ctx.cursor[3] = 18;
+ 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],
+ 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 )
{
wr[1] += 18;
}
- ui_text( &ui_global_ctx, wr, perf, 1, 0 );
+ ui_text( wr, perf, 1, 0 );
ui_rect_copy( wr, overlap_buffer[ overlap_length ++ ] );
}
projected_behind:
- ui_end_down( &ui_global_ctx );
- ui_global_ctx.cursor[1] += 1;
+ ui_end_down();
+ vg_uictx.cursor[1] += 1;
}
}