X-Git-Url: https://harrygodden.com/git/?a=blobdiff_plain;f=vg_audio.h;h=39f494ffa0b843e501c28aaab94db4cfa7dd443e;hb=76d234b7dc5e6500e8a54009b367e7620f11ef97;hp=a0d6b6b2beafc26912c6700a324c2e8581bd95ad;hpb=3b3420de0251e489082814e60dcc45e5e8842224;p=vg.git diff --git a/vg_audio.h b/vg_audio.h index a0d6b6b..39f494f 100644 --- a/vg_audio.h +++ b/vg_audio.h @@ -1,4 +1,4 @@ -/* Copyright (C) 2021-2022 Harry Godden (hgn) - All Rights Reserved */ +/* Copyright (C) 2021-2023 Harry Godden (hgn) - All Rights Reserved */ #ifndef VG_AUDIO_H #define VG_AUDIO_H @@ -10,13 +10,10 @@ #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 -#include +#include "vg/vg_audio_synth_bird.h" #ifdef __GNUC__ #ifndef __clang__ @@ -40,15 +37,38 @@ #endif #endif +#define AUDIO_FRAME_SIZE 512 +#define AUDIO_MIX_FRAME_SIZE 256 + #define AUDIO_CHANNELS 32 #define AUDIO_LFOS 8 +#define AUDIO_FILTERS 16 #define AUDIO_FLAG_LOOP 0x1 -#define AUDIO_FLAG_SPACIAL_3D 0x2 +#define AUDIO_FLAG_NO_DOPPLER 0x2 +#define AUDIO_FLAG_SPACIAL_3D 0x4 +#define AUDIO_FLAG_AUTO_START 0x8 +#define AUDIO_FLAG_FORMAT 0x1E00 -/* 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 +enum audio_format +{ + k_audio_format_mono = 0x000u, + k_audio_format_stereo = 0x200u, + k_audio_format_vorbis = 0x400u, + k_audio_format_none0 = 0x600u, + k_audio_format_none1 = 0x800u, + k_audio_format_none2 = 0xA00u, + k_audio_format_none3 = 0xC00u, + k_audio_format_none4 = 0xE00u, + + k_audio_format_bird = 0x1000u, + k_audio_format_none5 = 0x1200u, + k_audio_format_none6 = 0x1400u, + k_audio_format_none7 = 0x1600u, + k_audio_format_none8 = 0x1800u, + k_audio_format_none9 = 0x1A00u, + k_audio_format_none10 = 0x1C00u, + k_audio_format_none11 = 0x1E00u, +}; #define AUDIO_DECODE_SIZE (1024*256) /* 256 kb decoding buffers */ #define AUDIO_MUTE_VOLUME 0.0f @@ -58,17 +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, @@ -78,18 +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, @@ -105,12 +127,15 @@ 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; /* ... */ + u32 colour; /* ... */ /* internal non-readable state * -----------------------------*/ @@ -122,25 +147,29 @@ static struct vg_audio_system u32 volume_movement, pan_movement; - stb_vorbis *vorbis_handle; + 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, + k_channel_activity_end, k_channel_activity_error } - activity; + activity, + readable_activity; /* * editable structure, can be modified inside _lock and _unlock * the edit mask tells which to copy into internal _, or to discard * ---------------------------------------------------------------------- */ - struct channel_state - { + struct channel_state{ int relinquished; float volume, /* current volume */ @@ -162,49 +191,55 @@ 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; + 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" 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_dsp = {.mode = k_profile_mode_accum, + .name = "[T2] dsp_process()"}, vg_prof_audio_decode, - vg_prof_audio_mix; + vg_prof_audio_mix, + vg_prof_audio_dsp; /* * 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; @@ -213,43 +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, @@ -259,40 +281,38 @@ VG_STATIC void vg_audio_init(void) u32 decode_size = AUDIO_DECODE_SIZE * AUDIO_CHANNELS; vg_audio.decode_buffer = vg_linear_alloc( vg_mem.rtmemory, decode_size ); + vg_dsp_init(); + SDL_AudioSpec spec_desired, spec_got; spec_desired.callback = audio_mixer_callback; spec_desired.channels = 2; spec_desired.format = AUDIO_F32; spec_desired.freq = 44100; spec_desired.padding = 0; - spec_desired.samples = 512; + spec_desired.samples = AUDIO_FRAME_SIZE; spec_desired.silence = 0; spec_desired.size = 0; spec_desired.userdata = NULL; vg_audio.sdl_output_device = - SDL_OpenAudioDevice( NULL, 0, &spec_desired, &spec_got, - SDL_AUDIO_ALLOW_SAMPLES_CHANGE ); + 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 ); } @@ -309,32 +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 ) +{ + ch->group = 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; +} + +static void audio_channel_group( audio_channel *ch, u16 group ) +{ + ch->group = group; + ch->colour = (((u32)group * 29986577) & 0x00ffffff) | 0xff000000; +} + +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 ) - { - ch->source = clip; - ch->flags = flags; - strcpy( ch->name, clip->path ); - - 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 ){ return ch; } } @@ -342,6 +382,50 @@ 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 ) + return 1; + else + return 0; +} + static audio_channel *audio_relinquish_channel( audio_channel *ch ) { ch->editable_state.relinquished = 1; @@ -366,13 +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 ) - { - ch->editable_state.volume = 0.0f; + 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 ); } } @@ -398,10 +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; } @@ -416,14 +500,17 @@ 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 ); - ch->editable_state.spacial_falloff[3] = 1.0f/range; + + if( range == 0.0f ) + ch->editable_state.spacial_falloff[3] = 1.0f; + else + ch->editable_state.spacial_falloff[3] = 1.0f/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 ); } @@ -432,10 +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 ); @@ -448,10 +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 ); @@ -478,14 +565,16 @@ static void audio_set_lfo_frequency( int id, float freq ) lfo->editble_state_write_mask |= AUDIO_EDIT_LFO_PERIOD; } + /* * Committers * ----------------------------------------------------------------------------- */ static int audio_channel_load_source( audio_channel *ch ) { - if( ch->source->flags & AUDIO_FLAG_VORBIS ) - { + u32 format = ch->source->flags & AUDIO_FLAG_FORMAT; + + if( format == k_audio_format_vorbis ){ /* Setup vorbis decoder */ u32 index = ch - vg_audio.channels; @@ -502,34 +591,41 @@ 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( ch->source->flags & AUDIO_FLAG_STEREO ) - { + else if( format == k_audio_format_bird ){ + u32 index = ch - vg_audio.channels; + + u8 *buf = (u8*)vg_audio.decode_buffer; + struct synth_bird *loc = (void *)&buf[AUDIO_DECODE_SIZE*index]; + + memcpy( loc, ch->source->data, ch->source->size ); + synth_bird_reset( loc ); + + ch->bird_handle = loc; + ch->source_length = synth_bird_get_length_in_samples( loc ); + } + 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]; } @@ -574,21 +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]; @@ -609,7 +701,7 @@ stb_vorbis_get_samples_i16_downmixed( stb_vorbis *f, i16 *buffer, int len ) return n; } -static float audio_lfo_pull_sample( audio_lfo *lfo ) +static inline float audio_lfo_pull_sample( audio_lfo *lfo ) { lfo->time ++; @@ -619,8 +711,7 @@ static 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 ){ /* * # * # # @@ -642,8 +733,7 @@ static float audio_lfo_pull_sample( audio_lfo *lfo ) ) * (1.0f-fabsf(t)); } - else - { + else{ return 0.0f; } } @@ -656,41 +746,39 @@ static void audio_channel_get_samples( audio_channel *ch, u32 remaining = count; u32 buffer_pos = 0; - while( remaining ) - { - u32 samples_this_run = VG_MIN( remaining, ch->source_length -ch->cursor ); + u32 format = ch->source->flags & AUDIO_FLAG_FORMAT; + + 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( ch->source->flags & AUDIO_FLAG_STEREO ) - { - for( int i=0;isource->flags & AUDIO_FLAG_VORBIS ) - { + else if( format == k_audio_format_vorbis ){ int read_samples = stb_vorbis_get_samples_float_interleaved_stereo( ch->vorbis_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{ i16 *src_buffer = ch->source->data, *src = &src_buffer[ch->cursor]; @@ -700,10 +788,11 @@ 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->source->flags & AUDIO_FLAG_VORBIS ) + 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 ) + synth_bird_reset( ch->bird_handle ); ch->cursor = 0; continue; @@ -712,8 +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 ++; @@ -724,88 +812,105 @@ static void audio_channel_get_samples( audio_channel *ch, vg_profile_end( &_vg_prof_audio_decode ); } -static void audio_channel_mix( audio_channel *ch, - float *buffer, u32 frame_count ) +static void audio_channel_mix( audio_channel *ch, float *buffer ) { - u32 frames_write = frame_count; - - u32 buffer_length = frame_count; - if( ch->_.sampling_rate != 1.0f ) - { - buffer_length = ceilf( (float)frame_count * ch->_.sampling_rate ) + 1; - } + float framevol_l = vg_audio.internal_global_volume, + framevol_r = vg_audio.internal_global_volume; - float *pcf = alloca( buffer_length * 2 * sizeof(float) ); + float frame_samplerate = ch->_.sampling_rate; - audio_channel_get_samples( ch, buffer_length, pcf ); - vg_profile_begin( &_vg_prof_audio_mix ); - - if( ch->_.lfo ) - ch->_.lfo->time = ch->_.lfo->time_startframe; + if( ch->flags & AUDIO_FLAG_SPACIAL_3D ){ + v3f delta; + v3_sub( ch->_.spacial_falloff, vg_audio.internal_listener_pos, delta ); - float framevol_l = 1.0f, - framevol_r = 1.0f; + float dist = v3_length( delta ), + vol = vg_maxf( 0.0f, 1.0f - ch->_.spacial_falloff[3]*dist ); - 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(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 ); + 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; + } + } - framevol_l = 0.0f; - framevol_r = 0.0f; + 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] + ); } + } - v3f delta; - v3_sub( ch->_.spacial_falloff, vg_audio.listener_pos, delta ); + u32 buffer_length = AUDIO_MIX_FRAME_SIZE; + if( frame_samplerate != 1.0f ){ + float l = ceilf( (float)(AUDIO_MIX_FRAME_SIZE) * frame_samplerate ); + buffer_length = l+1; + } - float dist = v3_length( delta ), - vol = vg_maxf( 0.0f, 1.0f - ch->_.spacial_falloff[3]*dist ); + float pcf[ AUDIO_MIX_FRAME_SIZE * 2 * 2 ]; - v3_muls( delta, 1.0f/dist, delta ); - float pan = v3_dot( vg_audio.listener_ears, delta ); - vol = powf( vol, 5.0f ); + audio_channel_get_samples( ch, buffer_length, pcf ); - framevol_l *= (vol * 0.5f) * (1.0f - pan); - framevol_r *= (vol * 0.5f) * (1.0f + pan); - } + vg_profile_begin( &_vg_prof_audio_mix ); - for( u32 j=0; jvolume_movement < ch->_.volume_rate ) - { - ch->volume_movement ++; + float volume_movement = ch->volume_movement; + float const fvolume_rate = vg_maxf( 1.0f, ch->_.volume_rate ); + const float inv_volume_rate = 1.0f/fvolume_rate; - float movement_t = ch->volume_movement; - movement_t /= (float)ch->_.volume_rate; + float volume = ch->_.volume; + const float volume_start = ch->volume_movement_start; + const float volume_target = ch->_.volume_target; - ch->_.volume = vg_lerpf( ch->volume_movement_start, - ch->_.volume_target, - movement_t ); - } + for( u32 j=0; j_.volume * ch->_.volume; + float vol_norm = volume * volume; if( ch->_.lfo ) - vol_norm *= 1.0f + audio_lfo_pull_sample( ch->_.lfo ) - * ch->_.lfo_amount; + vol_norm *= 1.0f + audio_lfo_pull_sample(ch->_.lfo) * ch->_.lfo_amount; - float vol_l = vol_norm * framevol_l, - vol_r = vol_norm * framevol_r, + float vol_l = vol_norm * framevol_l, + vol_r = vol_norm * framevol_r, sample_l, sample_r; - if( ch->_.sampling_rate != 1.0f ) - { + if( frame_samplerate != 1.0f ){ /* absolutely garbage resampling, but it will do */ - float sample_index = ch->_.sampling_rate * (float)j; + float sample_index = frame_samplerate * (float)j; float t = vg_fractf( sample_index ); u32 i0 = floorf( sample_index ), @@ -814,8 +919,7 @@ static void audio_channel_mix( audio_channel *ch, sample_l = pcf[ i0*2+0 ]*(1.0f-t) + pcf[ i1*2+0 ]*t; sample_r = pcf[ i0*2+1 ]*(1.0f-t) + pcf[ i1*2+1 ]*t; } - else - { + else{ sample_l = pcf[ j*2+0 ]; sample_r = pcf[ j*2+1 ]; } @@ -824,26 +928,43 @@ static void audio_channel_mix( audio_channel *ch, buffer[ j*2+1 ] += sample_r * vol_r; } + ch->volume_movement += AUDIO_MIX_FRAME_SIZE; + ch->volume_movement = VG_MIN( ch->volume_movement, ch->_.volume_rate ); + ch->_.volume = volume; + vg_profile_end( &_vg_prof_audio_mix ); } -VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count ) +static void audio_mixer_callback( void *user, u8 *stream, int byte_count ) { /* * Copy data and move edit flags to commit flags * ------------------------------------------------------------- */ audio_lock(); - for( int i=0; iallocated ) continue; + if( ch->activity == k_channel_activity_alive ){ + if( (ch->cursor >= ch->source_length) && + !(ch->flags & AUDIO_FLAG_LOOP) ) + { + ch->activity = k_channel_activity_end; + } + } + /* process relinquishments */ - if( (ch->activity != k_channel_activity_reset) && ch->_.relinquished ) - { - if( (ch->cursor >= ch->source_length && !(ch->flags & AUDIO_FLAG_LOOP)) + 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) ) { @@ -855,8 +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; @@ -869,22 +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; - else + ch->_.volume_target = ch->editable_state.volume; + } + 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; } @@ -896,13 +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; } @@ -920,16 +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 = @@ -937,18 +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; } @@ -957,18 +1070,16 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count ) lfo->editble_state_write_mask = 0x00; } - + dsp_update_tunings(); audio_unlock(); /* * 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 @@ -986,62 +1097,120 @@ 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 ) - audio_channel_mix( ch, pOut32F, frame_count ); + 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 ){ + audio_channel_mix( ch, pOut32F+subpos ); + remaining -= AUDIO_MIX_FRAME_SIZE; + subpos += AUDIO_MIX_FRAME_SIZE*2; + } + } } - /* - * Relinquishing conditions - * ------------------------------------------------------------------ - */ + vg_profile_begin( &_vg_prof_dsp ); + + for( int i=0; ireadable_activity = ch->activity; + } + /* Profiling information * ----------------------------------------------- */ vg_profile_increment( &_vg_prof_audio_decode ); vg_profile_increment( &_vg_prof_audio_mix ); + vg_profile_increment( &_vg_prof_dsp ); + vg_prof_audio_mix = _vg_prof_audio_mix; vg_prof_audio_decode = _vg_prof_audio_decode; + vg_prof_audio_dsp = _vg_prof_dsp; + vg_audio.samples_last = frame_count; + 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; - /* load in directly */ - if( clip->flags & AUDIO_FLAG_VORBIS ) - { + u32 format = clip->flags & AUDIO_FLAG_FORMAT; + + /* TODO: This contains audio_lock() and unlock, but i don't know why + * can probably remove them. Low priority to check this */ + + /* 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( clip->flags & AUDIO_FLAG_STEREO ) - { - vg_fatal_exit_loop( "Unsupported format (Stereo uncompressed)" ); + else if( format == k_audio_format_stereo ){ + vg_fatal_error( "Unsupported format (Stereo uncompressed)" ); } - else - { + 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( total_size > AUDIO_DECODE_SIZE ) + vg_fatal_error( "Bird coding too long\n" ); + + struct synth_bird *bird = vg_linear_alloc( lin_alloc, total_size ); + memcpy( &bird->settings, clip->data, clip->size ); + + clip->data = bird; + clip->size = total_size; + + vg_info( "Loaded bird synthesis pattern (%u bytes)\n", total_size ); + } + else{ + if( !clip->path ){ + vg_fatal_error( "No path specified, embeded mono unsupported" ); + } + vg_linear_clear( vg_mem.scratch ); u32 fsize; @@ -1056,11 +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 */ @@ -1076,40 +1244,47 @@ 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; audio_lock(); + glBindTexture( GL_TEXTURE_2D, vg_dsp.view_texture ); + glTexSubImage2D( GL_TEXTURE_2D, 0, 0, 0, 256, 256, + GL_RGBA, GL_UNSIGNED_BYTE, + vg_dsp.view_texture_buffer ); + /* * Profiler * ----------------------------------------------------------------------- @@ -1117,117 +1292,107 @@ VG_STATIC void audio_debug_ui( m4x4f mtx_pv ) 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, + &vg_prof_audio_mix, + &vg_prof_audio_dsp}, 3, budget, (ui_rect){ 4, VG_PROFILE_SAMPLE_COUNT*2 + 8, - 250, 0 }, 3 ); + 512, 0 }, 3 ); 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 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_rect row; + ui_split( window, k_ui_axis_h, 18, 1, row, window ); - ui_end_down(); - vg_uictx.cursor[1] += 1; + if( !ch->allocated ){ + ui_fill( row, 0x50333333 ); continue; } const char *formats[] = { - "------", - "Mono ", - "Stereo", - "Vorbis" + " mono ", + " stereo ", + " vorbis ", + " none0 ", + " none1 ", + " none2 ", + " none3 ", + " none4 ", + "synth:bird", + " none5 ", + " none6 ", + " none7 ", + " none8 ", + " none9 ", + " none10 ", + " none11 ", }; - int format_index = 0; + const char *activties[] = + { + "reset", + "wake ", + "alive", + "end ", + "error" + }; - if( ch->source->flags & AUDIO_FLAG_STEREO ) - format_index = 2; - else if( ch->source->flags & AUDIO_FLAG_VORBIS ) - format_index = 3; - else - format_index = 1; + u32 format_index = (ch->source->flags & AUDIO_FLAG_FORMAT)>>9; - snprintf( perf, 127, "%02d %c%c%cD %s %4.2fv'%s'", + snprintf( perf, 127, "%02d[%#04x.%#06x]%c%c%cD %s [%s] %4.2fv'%s'", i, - (ch->editable_state.relinquished)? 'r': ' ', - 0? 'r': ' ', + ch->world_id, ch->group, + (ch->editable_state.relinquished)? 'r': '_', + 0? 'r': '_', 0? '3': '2', formats[format_index], + activties[ch->readable_activity], ch->editable_state.volume, ch->name ); - if( format_index == 0 ) - { - ui_fill_rect( vg_uictx.cursor, 0xa00000ff ); - } - else - { - ui_fill_rect( vg_uictx.cursor, 0xa0333333 ); - } - - 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])) ) @@ -1243,9 +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 ++ ] ); } } }