X-Git-Url: https://harrygodden.com/git/?a=blobdiff_plain;f=vg_audio.h;h=39f494ffa0b843e501c28aaab94db4cfa7dd443e;hb=76d234b7dc5e6500e8a54009b367e7620f11ef97;hp=5b65ccbd0dc040bf03be9559cb8f8b5ec2e5e4d7;hpb=28ec4a17033cc45198adab23ddf66ccd7a230db8;p=vg.git diff --git a/vg_audio.h b/vg_audio.h index 5b65ccb..39f494f 100644 --- a/vg_audio.h +++ b/vg_audio.h @@ -10,15 +10,11 @@ #include "vg/vg_platform.h" #include "vg/vg_io.h" #include "vg/vg_m.h" -#include "vg/vg_ui.h" #include "vg/vg_console.h" #include "vg/vg_store.h" #include "vg/vg_profiler.h" #include "vg/vg_audio_synth_bird.h" -#include -#include - #ifdef __GNUC__ #ifndef __clang__ #pragma GCC push_options @@ -48,15 +44,9 @@ #define AUDIO_LFOS 8 #define AUDIO_FILTERS 16 #define AUDIO_FLAG_LOOP 0x1 +#define AUDIO_FLAG_NO_DOPPLER 0x2 #define AUDIO_FLAG_SPACIAL_3D 0x4 #define AUDIO_FLAG_AUTO_START 0x8 - -/* Vorbis will ALWAYS use the maximum amount of channels it can */ -//#define AUDIO_FLAG_MONO 0x100 NOTE: This is the default, so its not used -//#define AUDIO_FLAG_STEREO 0x200 -//#define AUDIO_FLAG_VORBIS 0x400 -//#define AUDIO_FLAG_BIRD_SYNTH 0x800 - #define AUDIO_FLAG_FORMAT 0x1E00 enum audio_format @@ -88,17 +78,22 @@ typedef struct audio_clip audio_clip; typedef struct audio_channel audio_channel; typedef struct audio_lfo audio_lfo; -struct audio_clip -{ - const char *path; - u32 flags; +struct audio_clip{ + union { /* TODO oof.. */ + u64 _p64_; + const char *path; + }; + u32 flags; u32 size; - void *data; + + union{ + u64 _p64; + void *data; + }; }; -static struct vg_audio_system -{ +struct vg_audio_system{ SDL_AudioDeviceID sdl_output_device; void *audio_pool, @@ -108,18 +103,15 @@ static struct vg_audio_system /* synchro */ int sync_locked; - SDL_mutex *mux_checker, - *mux_sync; + SDL_SpinLock sl_checker, + sl_sync; - struct audio_lfo - { + struct audio_lfo{ u32 time, time_startframe; float sqrt_polynomial_coefficient; - struct - { - enum lfo_wave_type - { + struct{ + enum lfo_wave_type{ k_lfo_triangle, k_lfo_square, k_lfo_saw, @@ -135,9 +127,11 @@ static struct vg_audio_system } oscillators[ AUDIO_LFOS ]; - struct audio_channel - { + struct audio_channel{ int allocated; + u16 group; + u8 world_id; + char name[32]; /* only editable while allocated == 0 */ audio_clip *source; /* ... */ u32 flags; /* ... */ @@ -153,16 +147,14 @@ static struct vg_audio_system u32 volume_movement, pan_movement; - union - { + union{ struct synth_bird *bird_handle; stb_vorbis *vorbis_handle; }; stb_vorbis_alloc vorbis_alloc; - enum channel_activity - { + 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, @@ -177,8 +169,7 @@ static struct vg_audio_system * the edit mask tells which to copy into internal _, or to discard * ---------------------------------------------------------------------- */ - struct channel_state - { + struct channel_state{ int relinquished; float volume, /* current volume */ @@ -200,19 +191,20 @@ static struct vg_audio_system } channels[ AUDIO_CHANNELS ]; - /* System queue, and access from thread 0 */ - int debug_ui, debug_ui_3d; + int debug_ui, debug_ui_3d, debug_dsp; - v3f listener_pos, - listener_ears, - listener_velocity; + v3f internal_listener_pos, + internal_listener_ears, + internal_listener_velocity, - float volume, - volume_target, - volume_target_internal, - volume_console; + external_listener_pos, + external_listener_ears, + external_lister_velocity; + + float internal_global_volume, + external_global_volume; } -vg_audio = { .volume_console = 1.0f }; +static vg_audio = { .external_global_volume = 1.0f }; #include "vg/vg_audio_dsp.h" @@ -231,23 +223,23 @@ static struct vg_profile * 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) +static int audio_lock_checker_load(void) { int value; - SDL_LockMutex( vg_audio.mux_checker ); + SDL_AtomicLock( &vg_audio.sl_checker ); value = vg_audio.sync_locked; - SDL_UnlockMutex( vg_audio.mux_checker ); + SDL_AtomicUnlock( &vg_audio.sl_checker ); return value; } -VG_STATIC void audio_lock_checker_store( int value ) +static void audio_lock_checker_store( int value ) { - SDL_LockMutex( vg_audio.mux_checker ); + SDL_AtomicLock( &vg_audio.sl_checker ); vg_audio.sync_locked = value; - SDL_UnlockMutex( vg_audio.mux_checker ); + SDL_AtomicUnlock( &vg_audio.sl_checker ); } -VG_STATIC void audio_require_lock(void) +static void audio_require_lock(void) { if( audio_lock_checker_load() ) return; @@ -256,43 +248,30 @@ VG_STATIC void audio_require_lock(void) abort(); } -VG_STATIC void audio_lock(void) +static void audio_lock(void) { - SDL_LockMutex( vg_audio.mux_sync ); + SDL_AtomicLock( &vg_audio.sl_sync ); audio_lock_checker_store(1); } -VG_STATIC void audio_unlock(void) +static void audio_unlock(void) { audio_lock_checker_store(0); - SDL_UnlockMutex( vg_audio.mux_sync ); + SDL_AtomicUnlock( &vg_audio.sl_sync ); } -VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int frame_count ); -VG_STATIC void vg_audio_init(void) +static void audio_mixer_callback( void *user, u8 *stream, int frame_count ); +static void vg_audio_init(void) { - vg_audio.mux_checker = SDL_CreateMutex(); - vg_audio.mux_sync = SDL_CreateMutex(); - /* TODO: Move here? */ - vg_var_push( (struct vg_var){ - .name = "debug_audio", - .data = &vg_audio.debug_ui, - .data_type = k_var_dtype_i32, - .opt_i32 = { .min=0, .max=1, .clamp=1 }, - .persistent = 1 - }); - - vg_var_push( (struct vg_var){ - .name = "volume", - .data = &vg_audio.volume_console, - .data_type = k_var_dtype_f32, - .opt_f32 = { .min=0.0f, .max=2.0f, .clamp=1 }, - .persistent = 1 - }); + vg_console_reg_var( "debug_audio", &vg_audio.debug_ui, + k_var_dtype_i32, VG_VAR_CHEAT ); + vg_console_reg_var( "debug_dsp", &vg_audio.debug_dsp, + k_var_dtype_i32, VG_VAR_CHEAT ); + vg_console_reg_var( "volume", &vg_audio.external_global_volume, + k_var_dtype_f32, VG_VAR_PERSISTENT ); /* allocate memory */ - /* 32mb fixed */ vg_audio.audio_pool = vg_create_linear_allocator( vg_mem.rtmemory, 1024*1024*32, @@ -318,24 +297,20 @@ VG_STATIC void vg_audio_init(void) vg_audio.sdl_output_device = SDL_OpenAudioDevice( NULL, 0, &spec_desired, &spec_got,0 ); - if( vg_audio.sdl_output_device ) - { + if( vg_audio.sdl_output_device ){ SDL_PauseAudioDevice( vg_audio.sdl_output_device, 0 ); } - else - { - vg_fatal_exit_loop( + else{ + vg_fatal_error( "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) +static void vg_audio_free(void) { vg_dsp_free(); SDL_CloseAudioDevice( vg_audio.sdl_output_device ); @@ -354,37 +329,52 @@ VG_STATIC void vg_audio_free(void) #define AUDIO_EDIT_OWNERSHIP 0x40 #define AUDIO_EDIT_SAMPLING_RATE 0x80 -static audio_channel *audio_request_channel( audio_clip *clip, u32 flags ) +static void audio_channel_init( audio_channel *ch, audio_clip *clip, u32 flags ) { - for( int i=0; igroup = 0; + ch->world_id = 0; + ch->source = clip; + ch->flags = flags; + ch->colour = 0x00333333; + + if( (ch->source->flags & AUDIO_FLAG_FORMAT) == k_audio_format_bird ) + strcpy( ch->name, "[array]" ); + else + vg_strncpy( clip->path, ch->name, 32, k_strncpy_always_add_null ); + + ch->allocated = 1; + + ch->editable_state.relinquished = 0; + ch->editable_state.volume = 1.0f; + ch->editable_state.volume_target = 1.0f; + ch->editable_state.pan = 0.0f; + ch->editable_state.pan_target = 0.0f; + ch->editable_state.volume_rate = 0; + ch->editable_state.pan_rate = 0; + v4_copy((v4f){0.0f,0.0f,0.0f,1.0f},ch->editable_state.spacial_falloff); + ch->editable_state.lfo = NULL; + ch->editable_state.lfo_amount = 0.0f; + ch->editable_state.sampling_rate = 1.0f; + ch->editble_state_write_mask = 0x00; +} - if( !ch->allocated ) - { - ch->source = clip; - ch->flags = flags; - ch->colour = 0x00333333; +static void audio_channel_group( audio_channel *ch, u16 group ) +{ + ch->group = group; + ch->colour = (((u32)group * 29986577) & 0x00ffffff) | 0xff000000; +} - if( (ch->source->flags & AUDIO_FLAG_FORMAT) == k_audio_format_bird ) - strcpy( ch->name, "[array]" ); - else - strncpy( ch->name, clip->path, 31 ); - - ch->allocated = 1; - - ch->editable_state.relinquished = 0; - ch->editable_state.volume = 1.0f; - ch->editable_state.volume_target = 1.0f; - ch->editable_state.pan = 0.0f; - ch->editable_state.pan_target = 0.0f; - ch->editable_state.volume_rate = 0; - ch->editable_state.pan_rate = 0; - v4_copy((v4f){0.0f,0.0f,0.0f,1.0f},ch->editable_state.spacial_falloff); - ch->editable_state.lfo = NULL; - ch->editable_state.lfo_amount = 0.0f; - ch->editable_state.sampling_rate = 1.0f; - ch->editble_state_write_mask = 0x00; +static void audio_channel_world( audio_channel *ch, u8 world_id ) +{ + ch->world_id = world_id; +} + +static audio_channel *audio_get_first_idle_channel(void) +{ + for( int i=0; iallocated ){ return ch; } } @@ -392,6 +382,42 @@ static audio_channel *audio_request_channel( audio_clip *clip, u32 flags ) return NULL; } +static audio_channel *audio_get_group_idle_channel( u16 group, u32 max_count ) +{ + u32 count = 0; + audio_channel *dest = NULL; + + for( int i=0; iallocated ){ + if( ch->group == group ){ + count ++; + } + } + else{ + if( !dest ) + dest = ch; + } + } + + if( dest && (count < max_count) ){ + return dest; + } + + return NULL; +} + +static audio_channel *audio_get_group_first_active_channel( u16 group ) +{ + for( int i=0; iallocated && (ch->group == group) ) + return ch; + } + return NULL; +} + static int audio_channel_finished( audio_channel *ch ) { if( ch->readable_activity == k_channel_activity_end ) @@ -424,13 +450,11 @@ static void audio_channel_set_sampling_rate( audio_channel *ch, float rate ) static void audio_channel_edit_volume( audio_channel *ch, float new_volume, int instant ) { - if( instant ) - { + if( instant ){ ch->editable_state.volume = new_volume; ch->editble_state_write_mask |= AUDIO_EDIT_VOLUME; } - else - { + else{ audio_channel_slope_volume( ch, 0.05f, new_volume ); } } @@ -456,10 +480,12 @@ static audio_channel *audio_channel_crossfade( audio_channel *ch, if( ch ) ch = audio_channel_fadeout( ch, length ); - audio_channel *replacement = audio_request_channel( new_clip, flags ); + audio_channel *replacement = audio_get_first_idle_channel(); - if( replacement ) + if( replacement ){ + audio_channel_init( replacement, new_clip, flags ); audio_channel_fadein( replacement, length ); + } return replacement; } @@ -474,8 +500,7 @@ static void audio_channel_sidechain_lfo( audio_channel *ch, int lfo_id, static void audio_channel_set_spacial( audio_channel *ch, v3f co, float range ) { - if( ch->flags & AUDIO_FLAG_SPACIAL_3D ) - { + if( ch->flags & AUDIO_FLAG_SPACIAL_3D ){ v3_copy( co, ch->editable_state.spacial_falloff ); if( range == 0.0f ) @@ -485,8 +510,7 @@ static void audio_channel_set_spacial( audio_channel *ch, v3f co, float range ) ch->editble_state_write_mask |= AUDIO_EDIT_SPACIAL; } - else - { + else{ vg_warn( "Tried to set spacialization paramaters for 2D channel (%s)\n", ch->name ); } @@ -495,10 +519,10 @@ static void audio_channel_set_spacial( audio_channel *ch, v3f co, float range ) static int audio_oneshot_3d( audio_clip *clip, v3f position, float range, float volume ) { - audio_channel *ch = audio_request_channel( clip, AUDIO_FLAG_SPACIAL_3D ); + audio_channel *ch = audio_get_first_idle_channel(); - if( ch ) - { + if( ch ){ + audio_channel_init( ch, clip, AUDIO_FLAG_SPACIAL_3D ); audio_channel_set_spacial( ch, position, range ); audio_channel_edit_volume( ch, volume, 1 ); ch = audio_relinquish_channel( ch ); @@ -511,10 +535,10 @@ static int audio_oneshot_3d( audio_clip *clip, v3f position, static int audio_oneshot( audio_clip *clip, float volume, float pan ) { - audio_channel *ch = audio_request_channel( clip, 0x00 ); + audio_channel *ch = audio_get_first_idle_channel(); - if( ch ) - { + if( ch ){ + audio_channel_init( ch, clip, 0x00 ); audio_channel_edit_volume( ch, volume, 1 ); ch = audio_relinquish_channel( ch ); @@ -542,7 +566,6 @@ static void audio_set_lfo_frequency( int id, float freq ) } - /* * Committers * ----------------------------------------------------------------------------- @@ -551,8 +574,7 @@ static int audio_channel_load_source( audio_channel *ch ) { u32 format = ch->source->flags & AUDIO_FLAG_FORMAT; - if( format == k_audio_format_vorbis ) - { + if( format == k_audio_format_vorbis ){ /* Setup vorbis decoder */ u32 index = ch - vg_audio.channels; @@ -569,20 +591,17 @@ static int audio_channel_load_source( audio_channel *ch ) ch->source->data, ch->source->size, &err, &alloc ); - if( !decoder ) - { + if( !decoder ){ vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n", ch->source->path, err ); return 0; } - else - { + else{ ch->source_length = stb_vorbis_stream_length_in_samples( decoder ); ch->vorbis_handle = decoder; } } - else if( format == k_audio_format_bird ) - { + else if( format == k_audio_format_bird ){ u32 index = ch - vg_audio.channels; u8 *buf = (u8*)vg_audio.decode_buffer; @@ -594,22 +613,19 @@ static int audio_channel_load_source( audio_channel *ch ) ch->bird_handle = loc; ch->source_length = synth_bird_get_length_in_samples( loc ); } - else if( format == k_audio_format_stereo ) - { + else if( format == k_audio_format_stereo ){ ch->source_length = ch->source->size / 2; } - else - { + else{ ch->source_length = ch->source->size; } return 1; } -VG_STATIC void audio_decode_uncompressed_mono( i16 *src, u32 count, float *dst ) +static void audio_decode_uncompressed_mono( i16 *src, u32 count, float *dst ) { - for( u32 i=0; istereo */ -VG_STATIC int +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 ) - { + 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 ) - { + 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]; } @@ -654,21 +668,19 @@ stb_vorbis_get_samples_float_interleaved_stereo( stb_vorbis *f, float *buffer, /* * ........ more wrecked code sorry! */ -VG_STATIC int +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 ) - { + 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 ) - { + 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]; @@ -699,8 +711,7 @@ static inline float audio_lfo_pull_sample( audio_lfo *lfo ) float t = lfo->time; t /= (float)lfo->_.period; - if( lfo->_.wave_type == k_lfo_polynomial_bipolar ) - { + if( lfo->_.wave_type == k_lfo_polynomial_bipolar ){ /* * # * # # @@ -722,8 +733,7 @@ static inline float audio_lfo_pull_sample( audio_lfo *lfo ) ) * (1.0f-fabsf(t)); } - else - { + else{ return 0.0f; } } @@ -738,45 +748,37 @@ static void audio_channel_get_samples( audio_channel *ch, u32 format = ch->source->flags & AUDIO_FLAG_FORMAT; - while( remaining ) - { + while( remaining ){ u32 samples_this_run = VG_MIN(remaining, ch->source_length - ch->cursor); remaining -= samples_this_run; float *dst = &buf[ buffer_pos * 2 ]; - if( format == k_audio_format_stereo ) - { - for( int i=0;ivorbis_handle, dst, samples_this_run ); - if( read_samples != samples_this_run ) - { + if( read_samples != samples_this_run ){ vg_warn( "Invalid samples read (%s)\n", ch->source->path ); - for( int i=0; ibird_handle, dst, samples_this_run ); } - else - { + else{ i16 *src_buffer = ch->source->data, *src = &src_buffer[ch->cursor]; @@ -786,8 +788,7 @@ static void audio_channel_get_samples( audio_channel *ch, ch->cursor += samples_this_run; buffer_pos += samples_this_run; - if( (ch->flags & AUDIO_FLAG_LOOP) && remaining ) - { + if( (ch->flags & AUDIO_FLAG_LOOP) && remaining ){ if( format == k_audio_format_vorbis ) stb_vorbis_seek_start( ch->vorbis_handle ); else if( format == k_audio_format_bird ) @@ -800,8 +801,7 @@ static void audio_channel_get_samples( audio_channel *ch, break; } - while( remaining ) - { + while( remaining ){ buf[ buffer_pos*2 + 0 ] = 0.0f; buf[ buffer_pos*2 + 1 ] = 0.0f; buffer_pos ++; @@ -814,55 +814,64 @@ static void audio_channel_get_samples( audio_channel *ch, static void audio_channel_mix( audio_channel *ch, float *buffer ) { - float framevol_l = 1.0f, - framevol_r = 1.0f; + float framevol_l = vg_audio.internal_global_volume, + framevol_r = vg_audio.internal_global_volume; float frame_samplerate = ch->_.sampling_rate; - if( ch->flags & AUDIO_FLAG_SPACIAL_3D ) - { - if( !vg_validf(vg_audio.listener_pos[0]) || - !vg_validf(vg_audio.listener_pos[1]) || - !vg_validf(vg_audio.listener_pos[2]) || - !vg_validf(vg_audio.listener_velocity[0]) || - !vg_validf(vg_audio.listener_velocity[1]) || - !vg_validf(vg_audio.listener_velocity[2]) || - !vg_validf(ch->_.spacial_falloff[0]) || - !vg_validf(ch->_.spacial_falloff[1]) || - !vg_validf(ch->_.spacial_falloff[2]) ) - { - vg_error( "NaN listener/world position (%s)\n", ch->name ); - - framevol_l = 0.0f; - framevol_r = 0.0f; - } - + if( ch->flags & AUDIO_FLAG_SPACIAL_3D ){ v3f delta; - v3_sub( ch->_.spacial_falloff, vg_audio.listener_pos, delta ); + v3_sub( ch->_.spacial_falloff, vg_audio.internal_listener_pos, delta ); float dist = v3_length( delta ), vol = vg_maxf( 0.0f, 1.0f - ch->_.spacial_falloff[3]*dist ); - v3_muls( delta, 1.0f/dist, delta ); - float pan = v3_dot( vg_audio.listener_ears, delta ); - vol = powf( vol, 5.0f ); - - framevol_l *= (vol * 0.5f) * (1.0f - pan); - framevol_r *= (vol * 0.5f) * (1.0f + pan); - - v3_muls( delta, 1.0f/dist, delta ); + if( dist <= 0.01f ){ + + } + else{ + v3_muls( delta, 1.0f/dist, delta ); + float pan = v3_dot( vg_audio.internal_listener_ears, delta ); + vol = powf( vol, 5.0f ); + + framevol_l *= (vol * 0.5f) * (1.0f - pan); + framevol_r *= (vol * 0.5f) * (1.0f + pan); + + if( !(ch->source->flags & AUDIO_FLAG_NO_DOPPLER) ){ + const float vs = 323.0f; + + float dv = v3_dot(delta,vg_audio.internal_listener_velocity); + float doppler = (vs+dv)/vs; + doppler = vg_clampf( doppler, 0.6f, 1.4f ); + + if( fabsf(doppler-1.0f) > 0.01f ) + frame_samplerate *= doppler; + } + } - const float vs = 100.0f; - float doppler = (vs+v3_dot(delta,vg_audio.listener_velocity))/vs; - doppler = vg_clampf( doppler, 0.6f, 1.4f ); - - if( fabsf(doppler-1.0f) > 0.01f ) - frame_samplerate *= doppler; + if( !vg_validf( framevol_l ) || + !vg_validf( framevol_r ) || + !vg_validf( frame_samplerate ) ){ + vg_fatal_error( "Invalid sampling conditions.\n" + "This crash is to protect your ears.\n" + " channel: %p (%s)\n" + " sample_rate: %f\n" + " volume: L%f R%f\n" + " listener: %.2f %.2f %.2f [%.2f %.2f %.2f]\n", + ch, ch->name, frame_samplerate, + framevol_l, framevol_r, + vg_audio.internal_listener_pos[0], + vg_audio.internal_listener_pos[1], + vg_audio.internal_listener_pos[2], + vg_audio.internal_listener_ears[0], + vg_audio.internal_listener_ears[1], + vg_audio.internal_listener_ears[2] + ); + } } u32 buffer_length = AUDIO_MIX_FRAME_SIZE; - if( frame_samplerate != 1.0f ) - { + if( frame_samplerate != 1.0f ){ float l = ceilf( (float)(AUDIO_MIX_FRAME_SIZE) * frame_samplerate ); buffer_length = l+1; } @@ -881,24 +890,10 @@ static void audio_channel_mix( audio_channel *ch, float *buffer ) const float volume_start = ch->volume_movement_start; const float volume_target = ch->_.volume_target; - for( u32 j=0; jallocated ) continue; - if( ch->activity == k_channel_activity_alive ) - { + if( ch->activity == k_channel_activity_alive ){ if( (ch->cursor >= ch->source_length) && !(ch->flags & AUDIO_FLAG_LOOP) ) { @@ -965,8 +963,7 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count ) } /* process relinquishments */ - if( (ch->activity != k_channel_activity_reset) && ch->_.relinquished ) - { + if( (ch->activity != k_channel_activity_reset) && ch->_.relinquished ){ if( (ch->activity == k_channel_activity_end) || (ch->_.volume == 0.0f) || (ch->activity == k_channel_activity_error) ) @@ -979,8 +976,7 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count ) } /* process new channels */ - if( ch->activity == k_channel_activity_reset ) - { + if( ch->activity == k_channel_activity_reset ){ ch->_ = ch->editable_state; ch->cursor = 0; ch->source_length = 0; @@ -993,27 +989,23 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count ) ch->editable_state.relinquished = ch->_.relinquished; - if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME ) - { + if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME ){ ch->_.volume = ch->editable_state.volume; ch->_.volume_target = ch->editable_state.volume; } - else - { + else{ ch->editable_state.volume = ch->_.volume; } - if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME_SLOPE ) - { + if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME_SLOPE ){ ch->volume_movement_start = ch->_.volume; ch->volume_movement = 0; ch->_.volume_target = ch->editable_state.volume_target; ch->_.volume_rate = ch->editable_state.volume_rate; } - else - { + else{ ch->editable_state.volume_target = ch->_.volume_target; ch->editable_state.volume_rate = ch->_.volume_rate; } @@ -1025,13 +1017,11 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count ) ch->editable_state.sampling_rate = ch->_.sampling_rate; - if( ch->editble_state_write_mask & AUDIO_EDIT_LFO_ATTACHMENT ) - { + if( ch->editble_state_write_mask & AUDIO_EDIT_LFO_ATTACHMENT ){ ch->_.lfo = ch->editable_state.lfo; ch->_.lfo_amount = ch->editable_state.lfo_amount; } - else - { + else{ ch->editable_state.lfo = ch->_.lfo; ch->editable_state.lfo_amount = ch->_.lfo_amount; } @@ -1049,16 +1039,13 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count ) ch->editble_state_write_mask = 0x00; } - for( int i=0; ieditble_state_write_mask & AUDIO_EDIT_LFO_WAVE ) - { + if( lfo->editble_state_write_mask & AUDIO_EDIT_LFO_WAVE ){ lfo->_.wave_type = lfo->editable_state.wave_type; - if( lfo->_.wave_type == k_lfo_polynomial_bipolar ) - { + if( lfo->_.wave_type == k_lfo_polynomial_bipolar ){ lfo->_.polynomial_coefficient = lfo->editable_state.polynomial_coefficient; lfo->sqrt_polynomial_coefficient = @@ -1066,18 +1053,15 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count ) } } - if( lfo->editble_state_write_mask & AUDIO_EDIT_LFO_PERIOD ) - { - if( lfo->_.period ) - { + if( lfo->editble_state_write_mask & AUDIO_EDIT_LFO_PERIOD ){ + if( lfo->_.period ){ float t = lfo->time; t/= (float)lfo->_.period; lfo->_.period = lfo->editable_state.period; lfo->time = lfo->_.period * t; } - else - { + else{ lfo->time = 0; lfo->_.period = lfo->editable_state.period; } @@ -1092,12 +1076,10 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count ) /* * Process spawns * ------------------------------------------------------------- */ - for( int i=0; iactivity == k_channel_activity_wake ) - { + if( ch->activity == k_channel_activity_wake ){ if( audio_channel_load_source( ch ) ) ch->activity = k_channel_activity_alive; else @@ -1115,26 +1097,22 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count ) for( int i=0; itime_startframe = lfo->time; } - for( int i=0; iactivity == k_channel_activity_alive ) - { + if( ch->activity == k_channel_activity_alive ){ if( ch->_.lfo ) ch->_.lfo->time = ch->_.lfo->time_startframe; u32 remaining = frame_count, subpos = 0; - while( remaining ) - { + while( remaining ){ audio_channel_mix( ch, pOut32F+subpos ); remaining -= AUDIO_MIX_FRAME_SIZE; subpos += AUDIO_MIX_FRAME_SIZE*2; @@ -1151,8 +1129,7 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count ) audio_lock(); - for( int i=0; ireadable_activity = ch->activity; } @@ -1169,15 +1146,14 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count ) vg_audio.samples_last = frame_count; - if( vg_audio.debug_ui ) - { + if( vg_audio.debug_dsp ){ vg_dsp_update_texture(); } audio_unlock(); } -VG_STATIC void audio_clip_load( audio_clip *clip, void *lin_alloc ) +static void audio_clip_load( audio_clip *clip, void *lin_alloc ) { if( lin_alloc == NULL ) lin_alloc = vg_audio.audio_pool; @@ -1188,37 +1164,53 @@ VG_STATIC void audio_clip_load( audio_clip *clip, void *lin_alloc ) /* TODO: This contains audio_lock() and unlock, but i don't know why * can probably remove them. Low priority to check this */ - if( format == k_audio_format_vorbis ) - { + /* TODO: packed files for vorbis etc, should take from data if its not not + * NULL when we get the clip + */ + + if( format == k_audio_format_vorbis ){ + if( !clip->path ){ + vg_fatal_error( "No path specified, embeded vorbis unsupported" ); + } + audio_lock(); clip->data = vg_file_read( lin_alloc, clip->path, &clip->size ); audio_unlock(); if( !clip->data ) - vg_fatal_exit_loop( "Audio failed to load" ); + vg_fatal_error( "Audio failed to load" ); float mb = (float)(clip->size) / (1024.0f*1024.0f); vg_info( "Loaded audio clip '%s' (%.1fmb)\n", clip->path, mb ); } - else if( format == k_audio_format_stereo ) - { - vg_fatal_exit_loop( "Unsupported format (Stereo uncompressed)" ); + else if( format == k_audio_format_stereo ){ + vg_fatal_error( "Unsupported format (Stereo uncompressed)" ); } - else if( format == k_audio_format_bird ) - { - u32 len = strlen( clip->path ), - size = synth_bird_memory_requirement( len ); + else if( format == k_audio_format_bird ){ + if( !clip->data ){ + vg_fatal_error( "No data, external birdsynth unsupported" ); + } + + u32 total_size = clip->size + sizeof(struct synth_bird); + total_size -= sizeof(struct synth_bird_settings); + total_size = vg_align8( total_size ); - if( size > AUDIO_DECODE_SIZE ) - vg_fatal_exit_loop( "Bird code too long\n" ); + if( total_size > AUDIO_DECODE_SIZE ) + vg_fatal_error( "Bird coding too long\n" ); - clip->size = size; - clip->data = vg_linear_alloc( lin_alloc, size ); + struct synth_bird *bird = vg_linear_alloc( lin_alloc, total_size ); + memcpy( &bird->settings, clip->data, clip->size ); - synth_bird_load( clip->data, clip->path, len ); + clip->data = bird; + clip->size = total_size; + + vg_info( "Loaded bird synthesis pattern (%u bytes)\n", total_size ); } - else - { + else{ + if( !clip->path ){ + vg_fatal_error( "No path specified, embeded mono unsupported" ); + } + vg_linear_clear( vg_mem.scratch ); u32 fsize; @@ -1233,11 +1225,10 @@ VG_STATIC void audio_clip_load( audio_clip *clip, void *lin_alloc ) stb_vorbis *decoder = stb_vorbis_open_memory( filedata, fsize, &err, &alloc ); - if( !decoder ) - { + if( !decoder ){ vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n", clip->path, err ); - vg_fatal_exit_loop( "Vorbis decode error" ); + vg_fatal_error( "Vorbis decode error" ); } /* only mono is supported in uncompressed */ @@ -1253,34 +1244,36 @@ VG_STATIC void audio_clip_load( audio_clip *clip, void *lin_alloc ) decoder, clip->data, length_samples ); if( read_samples != length_samples ) - vg_fatal_exit_loop( "Decode error" ); + vg_fatal_error( "Decode error" ); +#if 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 ); +#endif } } -VG_STATIC void audio_clip_loadn( audio_clip *arr, int count, void *lin_alloc ) +static void audio_clip_loadn( audio_clip *arr, int count, void *lin_alloc ) { for( int i=0; idata && clip->size ) return; audio_unlock(); - vg_fatal_exit_loop( "Must load audio clip before playing! \n" ); + vg_fatal_error( "Must load audio clip before playing! \n" ); } /* * Debugging */ -VG_STATIC void audio_debug_ui( m4x4f mtx_pv ) +static void audio_debug_ui( m4x4f mtx_pv ) { if( !vg_audio.debug_ui ) return; @@ -1308,43 +1301,30 @@ VG_STATIC void audio_debug_ui( m4x4f mtx_pv ) char perf[128]; /* Draw UI */ - vg_uictx.cursor[0] = 512 + 8; - vg_uictx.cursor[1] = VG_PROFILE_SAMPLE_COUNT*2+8+24+12+12; - vg_uictx.cursor[2] = 150; - vg_uictx.cursor[3] = 12; - - ui_rect view_thing = { 4, vg.window_y-512-4, 512, 512 }; - ui_push_image( view_thing, vg_dsp.view_texture ); - - float mb1 = 1024.0f*1024.0f, - usage = vg_linear_get_cur( vg_audio.audio_pool ) / mb1, - total = vg_linear_get_capacity( vg_audio.audio_pool ) / mb1, - percent = (usage/total) * 100.0f; - - snprintf( perf, 127, "Mem: %.1f/%.1fmb (%.1f%%)\n", usage, total, percent ); - - ui_text( vg_uictx.cursor, perf, 1, 0 ); - vg_uictx.cursor[1] += 20; + ui_rect window = { + 0, + 0, + 800, + AUDIO_CHANNELS * 18 + }; + + if( vg_audio.debug_dsp ){ + ui_rect view_thing = { 4, vg.window_y-512-4, 512, 512 }; + ui_image( view_thing, vg_dsp.view_texture ); + } ui_rect overlap_buffer[ AUDIO_CHANNELS ]; u32 overlap_length = 0; /* Draw audio stack */ - for( int i=0; iallocated ) - { - ui_fill_rect( vg_uictx.cursor, 0x50333333 ); - - ui_end_down(); - vg_uictx.cursor[1] += 1; + if( !ch->allocated ){ + ui_fill( row, 0x50333333 ); continue; } @@ -1379,8 +1359,9 @@ VG_STATIC void audio_debug_ui( m4x4f mtx_pv ) u32 format_index = (ch->source->flags & AUDIO_FLAG_FORMAT)>>9; - snprintf( perf, 127, "%02d %c%c%cD %s [%s] %4.2fv'%s'", + snprintf( perf, 127, "%02d[%#04x.%#06x]%c%c%cD %s [%s] %4.2fv'%s'", i, + ch->world_id, ch->group, (ch->editable_state.relinquished)? 'r': '_', 0? 'r': '_', 0? '3': '2', @@ -1389,39 +1370,29 @@ VG_STATIC void audio_debug_ui( m4x4f mtx_pv ) ch->editable_state.volume, ch->name ); - ui_fill_rect( vg_uictx.cursor, 0xa0000000 | ch->colour ); - - vg_uictx.cursor[0] += 2; - vg_uictx.cursor[1] += 2; - ui_text( vg_uictx.cursor, perf, 1, 0 ); - - ui_end_down(); - vg_uictx.cursor[1] += 1; + ui_fill( row, 0xa0000000 | ch->colour ); + ui_text( row, perf, 1, k_ui_align_middle_left, 0 ); - if( AUDIO_FLAG_SPACIAL_3D ) - { + if( AUDIO_FLAG_SPACIAL_3D ){ v4f wpos; v3_copy( ch->editable_state.spacial_falloff, wpos ); wpos[3] = 1.0f; m4x4_mulv( mtx_pv, wpos, wpos ); - if( wpos[3] > 0.0f ) - { + if( wpos[3] > 0.0f ){ v2_muls( wpos, (1.0f/wpos[3]) * 0.5f, wpos ); v2_add( wpos, (v2f){ 0.5f, 0.5f }, wpos ); ui_rect wr; - wr[0] = wpos[0] * vg.window_x; - wr[1] = (1.0f-wpos[1]) * vg.window_y; - wr[2] = 100; + wr[0] = vg_clampf(wpos[0] * vg.window_x, -32000.0f,32000.0f); + wr[1] = vg_clampf((1.0f-wpos[1]) * vg.window_y,-32000.0f,32000.0f); + wr[2] = 1000; wr[3] = 17; - for( int j=0; j<12; j++ ) - { + for( int j=0; j<12; j++ ){ int collide = 0; - for( int k=0; k= wk[0])) && ((wr[1] <= wk[1]+wk[3]) && (wr[1]+wr[3] >= wk[1])) ) @@ -1437,9 +1408,8 @@ VG_STATIC void audio_debug_ui( m4x4f mtx_pv ) wr[1] += 18; } - ui_text( wr, perf, 1, 0 ); - - ui_rect_copy( wr, overlap_buffer[ overlap_length ++ ] ); + ui_text( wr, perf, 1, k_ui_align_middle_left, 0 ); + rect_copy( wr, overlap_buffer[ overlap_length ++ ] ); } } }