X-Git-Url: https://harrygodden.com/git/?p=vg.git;a=blobdiff_plain;f=vg_audio.h;h=fc09d103aa5338167100acdd0738679d656e3aba;hp=4d203bf9c1aafae550af5dc158bac615b108ba7c;hb=HEAD;hpb=4f4aba081a05f72ad1bbc7844ca4060cd1f43fe6 diff --git a/vg_audio.h b/vg_audio.h index 4d203bf..ba10348 100644 --- a/vg_audio.h +++ b/vg_audio.h @@ -1,42 +1,11 @@ -/* Copyright (C) 2021-2023 Harry Godden (hgn) - All Rights Reserved */ +/* Copyright (C) 2021-2024 Harry Godden (hgn) - All Rights Reserved */ -#ifndef VG_AUDIO_H -#define VG_AUDIO_H +#pragma once -#define VG_GAME - -#include "vg/vg.h" -#include "vg/vg_stdint.h" -#include "vg/vg_platform.h" -#include "vg/vg_io.h" -#include "vg/vg_m.h" -#include "vg/vg_ui.h" -#include "vg/vg_console.h" -#include "vg/vg_store.h" -#include "vg/vg_profiler.h" -#include "vg/vg_audio_synth_bird.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 "submodules/stb/stb_vorbis.c" -#undef L -#undef R -#undef C - -#ifdef __GNUC__ - #ifndef __clang__ - #pragma GCC pop_options - #pragma GCC diagnostic pop - #endif -#endif +#include "vg_platform.h" +#include "vg_engine.h" +#include "vg_string.h" +#include "vg_vorbis.h" #define AUDIO_FRAME_SIZE 512 #define AUDIO_MIX_FRAME_SIZE 256 @@ -48,6 +17,7 @@ #define AUDIO_FLAG_NO_DOPPLER 0x2 #define AUDIO_FLAG_SPACIAL_3D 0x4 #define AUDIO_FLAG_AUTO_START 0x8 +#define AUDIO_FLAG_NO_DSP 0x10 #define AUDIO_FLAG_FORMAT 0x1E00 enum audio_format @@ -62,7 +32,7 @@ enum audio_format k_audio_format_none4 = 0xE00u, k_audio_format_bird = 0x1000u, - k_audio_format_none5 = 0x1200u, + k_audio_format_gen = 0x1200u, k_audio_format_none6 = 0x1400u, k_audio_format_none7 = 0x1600u, k_audio_format_none8 = 0x1800u, @@ -79,10 +49,12 @@ typedef struct audio_clip audio_clip; typedef struct audio_channel audio_channel; typedef struct audio_lfo audio_lfo; -struct audio_clip{ +struct audio_clip +{ union { /* TODO oof.. */ u64 _p64_; const char *path; + void *func; }; u32 flags; @@ -94,8 +66,12 @@ struct audio_clip{ }; }; -static struct vg_audio_system{ +struct vg_audio_system +{ SDL_AudioDeviceID sdl_output_device; + vg_str device_choice; /* buffer is null? use default from OS */ + + bool always_keep_compressed; void *audio_pool, *decode_buffer; @@ -149,9 +125,10 @@ static struct vg_audio_system{ pan_movement; union{ - struct synth_bird *bird_handle; - stb_vorbis *vorbis_handle; - }; + struct synth_bird *bird; + stb_vorbis *vorbis; + } + handle; stb_vorbis_alloc vorbis_alloc; @@ -192,7 +169,7 @@ static struct vg_audio_system{ } channels[ AUDIO_CHANNELS ]; - int debug_ui, debug_ui_3d, debug_dsp; + int debug_ui, debug_ui_3d, debug_dsp, dsp_enabled; v3f internal_listener_pos, internal_listener_ears, @@ -205,1202 +182,48 @@ static struct vg_audio_system{ float internal_global_volume, external_global_volume; } -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_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) -{ - int value; - SDL_AtomicLock( &vg_audio.sl_checker ); - value = vg_audio.sync_locked; - SDL_AtomicUnlock( &vg_audio.sl_checker ); - return value; -} - -VG_STATIC void audio_lock_checker_store( int value ) -{ - SDL_AtomicLock( &vg_audio.sl_checker ); - vg_audio.sync_locked = value; - SDL_AtomicUnlock( &vg_audio.sl_checker ); -} - -VG_STATIC void audio_require_lock(void) -{ - if( audio_lock_checker_load() ) - return; - - vg_error( "Modifying sound effects systems requires locking\n" ); - abort(); -} - -VG_STATIC void audio_lock(void) -{ - SDL_AtomicLock( &vg_audio.sl_sync ); - audio_lock_checker_store(1); -} - -VG_STATIC void audio_unlock(void) -{ - audio_lock_checker_store(0); - 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) -{ - /* TODO: Move here? */ - 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, - VG_MEMORY_SYSTEM ); - - /* fixed */ - 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 = 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,0 ); - - if( vg_audio.sdl_output_device ){ - SDL_PauseAudioDevice( vg_audio.sdl_output_device, 0 ); - } - 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_STATIC void vg_audio_free(void) -{ - vg_dsp_free(); - SDL_CloseAudioDevice( vg_audio.sdl_output_device ); -} - -/* - * thread 1 - */ - -#define AUDIO_EDIT_VOLUME_SLOPE 0x1 -#define AUDIO_EDIT_VOLUME 0x2 -#define AUDIO_EDIT_LFO_PERIOD 0x4 -#define AUDIO_EDIT_LFO_WAVE 0x8 -#define AUDIO_EDIT_LFO_ATTACHMENT 0x10 -#define AUDIO_EDIT_SPACIAL 0x20 -#define AUDIO_EDIT_OWNERSHIP 0x40 -#define AUDIO_EDIT_SAMPLING_RATE 0x80 - -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 ){ - return ch; - } - } - - 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; - ch->editble_state_write_mask |= AUDIO_EDIT_OWNERSHIP; - return NULL; -} - -static void audio_channel_slope_volume( audio_channel *ch, float length, - float new_volume ) -{ - ch->editable_state.volume_target = new_volume; - ch->editable_state.volume_rate = length * 44100.0f; - ch->editble_state_write_mask |= AUDIO_EDIT_VOLUME_SLOPE; -} - -static void audio_channel_set_sampling_rate( audio_channel *ch, float rate ) -{ - ch->editable_state.sampling_rate = rate; - ch->editble_state_write_mask |= AUDIO_EDIT_SAMPLING_RATE; -} - -static void audio_channel_edit_volume( audio_channel *ch, - float new_volume, int instant ) -{ - if( instant ){ - ch->editable_state.volume = new_volume; - ch->editble_state_write_mask |= AUDIO_EDIT_VOLUME; - } - else{ - audio_channel_slope_volume( ch, 0.05f, new_volume ); - } -} - -static audio_channel *audio_channel_fadeout( audio_channel *ch, float length ) -{ - audio_channel_slope_volume( ch, length, 0.0f ); - return audio_relinquish_channel( ch ); -} - -static void audio_channel_fadein( audio_channel *ch, float length ) -{ - audio_channel_edit_volume( ch, 0.0f, 1 ); - audio_channel_slope_volume( ch, length, 1.0f ); -} - -static audio_channel *audio_channel_crossfade( audio_channel *ch, - audio_clip *new_clip, - float length, u32 flags ) -{ - u32 cursor = 0; - - if( ch ) - ch = audio_channel_fadeout( ch, length ); - - audio_channel *replacement = audio_get_first_idle_channel(); - - if( replacement ){ - audio_channel_init( replacement, new_clip, flags ); - audio_channel_fadein( replacement, length ); - } - - return replacement; -} - -static void audio_channel_sidechain_lfo( audio_channel *ch, int lfo_id, - float amount ) -{ - ch->editable_state.lfo = &vg_audio.oscillators[ lfo_id ]; - ch->editable_state.lfo_amount = amount; - ch->editble_state_write_mask |= AUDIO_EDIT_LFO_ATTACHMENT; -} - -static void audio_channel_set_spacial( audio_channel *ch, v3f co, float range ) -{ - if( ch->flags & AUDIO_FLAG_SPACIAL_3D ){ - v3_copy( co, ch->editable_state.spacial_falloff ); - - 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{ - vg_warn( "Tried to set spacialization paramaters for 2D channel (%s)\n", - ch->name ); - } -} - -static int audio_oneshot_3d( audio_clip *clip, v3f position, - float range, float volume ) -{ - audio_channel *ch = audio_get_first_idle_channel(); - - 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 ); - - return 1; - } - else - return 0; -} - -static int audio_oneshot( audio_clip *clip, float volume, float pan ) -{ - audio_channel *ch = audio_get_first_idle_channel(); - - if( ch ){ - audio_channel_init( ch, clip, 0x00 ); - audio_channel_edit_volume( ch, volume, 1 ); - ch = audio_relinquish_channel( ch ); - - return 1; - } - else - return 0; -} - -static void audio_set_lfo_wave( int id, enum lfo_wave_type type, - float coefficient ) -{ - audio_lfo *lfo = &vg_audio.oscillators[ id ]; - lfo->editable_state.polynomial_coefficient = coefficient; - lfo->editable_state.wave_type = type; - - lfo->editble_state_write_mask |= AUDIO_EDIT_LFO_WAVE; -} - -static void audio_set_lfo_frequency( int id, float freq ) -{ - audio_lfo *lfo = &vg_audio.oscillators[ id ]; - lfo->editable_state.period = 44100.0f / freq; - lfo->editble_state_write_mask |= AUDIO_EDIT_LFO_PERIOD; -} - - -/* - * Committers - * ----------------------------------------------------------------------------- - */ -static int audio_channel_load_source( audio_channel *ch ) -{ - u32 format = ch->source->flags & AUDIO_FLAG_FORMAT; - - if( format == k_audio_format_vorbis ){ - /* Setup vorbis decoder */ - u32 index = ch - vg_audio.channels; - - u8 *buf = (u8*)vg_audio.decode_buffer, - *loc = &buf[AUDIO_DECODE_SIZE*index]; - - stb_vorbis_alloc alloc = { - .alloc_buffer = (char *)loc, - .alloc_buffer_length_in_bytes = AUDIO_DECODE_SIZE - }; - - int err; - stb_vorbis *decoder = stb_vorbis_open_memory( - ch->source->data, - ch->source->size, &err, &alloc ); - - if( !decoder ){ - vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n", - ch->source->path, err ); - return 0; - } - else{ - ch->source_length = stb_vorbis_stream_length_in_samples( decoder ); - ch->vorbis_handle = decoder; - } - } - 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{ - ch->source_length = ch->source->size; - } - - return 1; -} - -VG_STATIC void audio_decode_uncompressed_mono( i16 *src, u32 count, float *dst ) -{ - for( u32 i=0; istereo - */ -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 inline float audio_lfo_pull_sample( audio_lfo *lfo ) -{ - lfo->time ++; - - if( lfo->time >= lfo->_.period ) - lfo->time = 0; - - float t = lfo->time; - t /= (float)lfo->_.period; - - if( lfo->_.wave_type == k_lfo_polynomial_bipolar ){ - /* - * # - * # # - * # # - * # # - * ### # ### - * ## # - * # # - * # # - * ## - */ - - t *= 2.0f; - t -= 1.0f; - - return (( 2.0f * lfo->sqrt_polynomial_coefficient * t ) / - /* --------------------------------------- */ - ( 1.0f + lfo->_.polynomial_coefficient * t*t ) - - ) * (1.0f-fabsf(t)); - } - else{ - return 0.0f; - } -} - -static void audio_channel_get_samples( audio_channel *ch, - u32 count, float *buf ) -{ - vg_profile_begin( &_vg_prof_audio_decode ); - - u32 remaining = count; - u32 buffer_pos = 0; - - 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( format == k_audio_format_stereo ){ - for( int i=0;ivorbis_handle, - dst, - 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]; - - audio_decode_uncompressed_mono( src, samples_this_run, dst ); - } - - ch->cursor += samples_this_run; - buffer_pos += samples_this_run; - - 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; - } - else - break; - } - - while( remaining ){ - buf[ buffer_pos*2 + 0 ] = 0.0f; - buf[ buffer_pos*2 + 1 ] = 0.0f; - buffer_pos ++; - - remaining --; - } - - vg_profile_end( &_vg_prof_audio_decode ); -} - -static void audio_channel_mix( audio_channel *ch, float *buffer ) -{ - 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 ){ - v3f 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 ); - - 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; - } - } - - if( !vg_validf( framevol_l ) ) vg_fatal_error( "NaN left channel" ); - if( !vg_validf( framevol_r ) ) vg_fatal_error( "NaN right channel" ); - if( !vg_validf( frame_samplerate ) ) - vg_fatal_error( "NaN sample rate" ); - } - - 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 pcf[ AUDIO_MIX_FRAME_SIZE * 2 * 2 ]; - - audio_channel_get_samples( ch, buffer_length, pcf ); - - vg_profile_begin( &_vg_prof_audio_mix ); - - 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 volume = ch->_.volume; - const float volume_start = ch->volume_movement_start; - const float volume_target = ch->_.volume_target; - - for( u32 j=0; j_.lfo ) - 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, - sample_l, - sample_r; - - if( frame_samplerate != 1.0f ){ - /* absolutely garbage resampling, but it will do - */ - - float sample_index = frame_samplerate * (float)j; - float t = vg_fractf( sample_index ); - - u32 i0 = floorf( sample_index ), - i1 = i0+1; - - 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{ - sample_l = pcf[ j*2+0 ]; - sample_r = pcf[ j*2+1 ]; - } - - buffer[ j*2+0 ] += sample_l * vol_l; - 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 ) -{ - /* - * Copy data and move edit flags to commit flags - * ------------------------------------------------------------- */ - audio_lock(); - - v3_copy( vg_audio.external_listener_pos, vg_audio.internal_listener_pos ); - v3_copy( vg_audio.external_listener_ears, vg_audio.internal_listener_ears ); - v3_copy( vg_audio.external_lister_velocity, - vg_audio.internal_listener_velocity ); - vg_audio.internal_global_volume = vg_audio.external_global_volume; - - 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->activity == k_channel_activity_end) - || (ch->_.volume == 0.0f) - || (ch->activity == k_channel_activity_error) ) - { - ch->_.relinquished = 0; - ch->allocated = 0; - ch->activity = k_channel_activity_reset; - continue; - } - } - - /* process new channels */ - if( ch->activity == k_channel_activity_reset ){ - ch->_ = ch->editable_state; - ch->cursor = 0; - ch->source_length = 0; - ch->activity = k_channel_activity_wake; - } - - if( ch->editble_state_write_mask & AUDIO_EDIT_OWNERSHIP ) - ch->_.relinquished = ch->editable_state.relinquished; - else - ch->editable_state.relinquished = ch->_.relinquished; - - - if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME ){ - ch->_.volume = ch->editable_state.volume; - ch->_.volume_target = ch->editable_state.volume; - } - else{ - ch->editable_state.volume = ch->_.volume; - } - - - 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{ - ch->editable_state.volume_target = ch->_.volume_target; - ch->editable_state.volume_rate = ch->_.volume_rate; - } - - - if( ch->editble_state_write_mask & AUDIO_EDIT_SAMPLING_RATE ) - ch->_.sampling_rate = ch->editable_state.sampling_rate; - else - ch->editable_state.sampling_rate = ch->_.sampling_rate; - - - 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{ - ch->editable_state.lfo = ch->_.lfo; - ch->editable_state.lfo_amount = ch->_.lfo_amount; - } - - - if( ch->editble_state_write_mask & AUDIO_EDIT_SPACIAL ) - v4_copy( ch->editable_state.spacial_falloff,ch->_.spacial_falloff ); - else - v4_copy( ch->_.spacial_falloff,ch->editable_state.spacial_falloff ); - - - /* currently readonly, i guess */ - ch->editable_state.pan_target = ch->_.pan_target; - ch->editable_state.pan = ch->_.pan; - ch->editble_state_write_mask = 0x00; - } - - for( int i=0; ieditble_state_write_mask & AUDIO_EDIT_LFO_WAVE ){ - lfo->_.wave_type = lfo->editable_state.wave_type; - - if( lfo->_.wave_type == k_lfo_polynomial_bipolar ){ - lfo->_.polynomial_coefficient = - lfo->editable_state.polynomial_coefficient; - lfo->sqrt_polynomial_coefficient = - sqrtf(lfo->_.polynomial_coefficient); - } - } - - 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{ - lfo->time = 0; - lfo->_.period = lfo->editable_state.period; - } - } - - lfo->editble_state_write_mask = 0x00; - } - - dsp_update_tunings(); - audio_unlock(); - - /* - * Process spawns - * ------------------------------------------------------------- */ - for( int i=0; iactivity == k_channel_activity_wake ){ - if( audio_channel_load_source( ch ) ) - ch->activity = k_channel_activity_alive; - else - ch->activity = k_channel_activity_error; - } - } - - /* - * Mix everything - * -------------------------------------------------------- */ - int frame_count = byte_count/(2*sizeof(float)); - - /* Clear buffer */ - float *pOut32F = (float *)stream; - for( int i=0; itime_startframe = lfo->time; - } - - for( int i=0; iactivity == 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; - } - } - } - - 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 ) -{ - if( lin_alloc == NULL ) - lin_alloc = vg_audio.audio_pool; - - /* load in directly */ - 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_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_error( "Unsupported format (Stereo uncompressed)" ); - } - 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; - - stb_vorbis_alloc alloc = { - .alloc_buffer = vg_linear_alloc( vg_mem.scratch, AUDIO_DECODE_SIZE ), - .alloc_buffer_length_in_bytes = AUDIO_DECODE_SIZE - }; - - void *filedata = vg_file_read( vg_mem.scratch, clip->path, &fsize ); - - int err; - stb_vorbis *decoder = stb_vorbis_open_memory( - filedata, fsize, &err, &alloc ); - - if( !decoder ){ - vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n", - clip->path, err ); - vg_fatal_error( "Vorbis decode error" ); - } - - /* only mono is supported in uncompressed */ - u32 length_samples = stb_vorbis_stream_length_in_samples( decoder ), - data_size = length_samples * sizeof(i16); - - audio_lock(); - clip->data = vg_linear_alloc( lin_alloc, vg_align8(data_size) ); - clip->size = length_samples; - audio_unlock(); - - int read_samples = stb_vorbis_get_samples_i16_downmixed( - decoder, clip->data, length_samples ); - - if( read_samples != length_samples ) - vg_fatal_error( "Decode error" ); - - float mb = (float)(data_size) / (1024.0f*1024.0f); - vg_info( "Loaded audio clip '%s' (%.1fmb) %u samples\n", clip->path, mb, - length_samples ); - } -} - -VG_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_error( "Must load audio clip before playing! \n" ); -} - -/* - * Debugging - */ - -VG_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 - * ----------------------------------------------------------------------- - */ - - float budget = ((double)vg_audio.samples_last / 44100.0) * 1000.0; -#if 0 - vg_profile_drawn( (struct vg_profile *[]){ &vg_prof_audio_decode, - &vg_prof_audio_mix, - &vg_prof_audio_dsp}, 3, - budget, (ui_rect){ 4, VG_PROFILE_SAMPLE_COUNT*2 + 8, - 512, 0 }, 3 ); +extern vg_audio; + +void audio_clip_load( audio_clip *clip, void *lin_alloc ); +void audio_clip_loadn( audio_clip *arr, int count, void *lin_alloc ); + +void vg_audio_register(void); +void vg_audio_device_init(void); +void vg_audio_init(void); +void vg_audio_free(void); + +void audio_lock(void); +void audio_unlock(void); + +void audio_channel_init( audio_channel *ch, audio_clip *clip, u32 flags ); +void audio_channel_group( audio_channel *ch, u16 group ); +void audio_channel_world( audio_channel *ch, u8 world_id ); +audio_channel *audio_get_first_idle_channel(void); +audio_channel *audio_get_group_idle_channel( u16 group, u32 max_count ); +audio_channel *audio_get_group_first_active_channel( u16 group ); +int audio_channel_finished( audio_channel *ch ); +audio_channel *audio_relinquish_channel( audio_channel *ch ); +void audio_channel_slope_volume( audio_channel *ch, f32 length, f32 new_vol ); +void audio_channel_set_sampling_rate( audio_channel *ch, float rate ); +void audio_channel_edit_volume( audio_channel *ch, f32 new_vol, int instant ); +audio_channel *audio_channel_fadeout( audio_channel *ch, float length ); +void audio_channel_fadein( audio_channel *ch, float length ); +audio_channel *audio_channel_crossfade( audio_channel *ch, + audio_clip *new_clip, + float length, u32 flags ); +void audio_channel_sidechain_lfo( audio_channel *ch, int lfo_id, f32 amount ); +void audio_channel_set_spacial( audio_channel *ch, v3f co, float range ); +int audio_oneshot_3d( audio_clip *clip, v3f position, f32 range, f32 volume ); +int audio_oneshot( audio_clip *clip, f32 volume, f32 pan ); +void audio_set_lfo_wave( int id, enum lfo_wave_type type, f32 coefficient ); +void audio_set_lfo_frequency( int id, float freq ); +int audio_channel_load_source( audio_channel *ch ); + +void audio_debug_ui( + +#ifdef VG_3D + m4x4f +#else + m3x3f #endif - - - char perf[128]; - - /* Draw UI */ - 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( row, 0x50333333 ); - continue; - } - - const char *formats[] = - { - " mono ", - " stereo ", - " vorbis ", - " none0 ", - " none1 ", - " none2 ", - " none3 ", - " none4 ", - "synth:bird", - " none5 ", - " none6 ", - " none7 ", - " none8 ", - " none9 ", - " none10 ", - " none11 ", - }; - - const char *activties[] = - { - "reset", - "wake ", - "alive", - "end ", - "error" - }; - - u32 format_index = (ch->source->flags & AUDIO_FLAG_FORMAT)>>9; - - 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', - formats[format_index], - activties[ch->readable_activity], - ch->editable_state.volume, - ch->name ); - - ui_fill( row, 0xa0000000 | ch->colour ); - ui_text( row, perf, 1, k_ui_align_middle_left, 0 ); - - 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 ){ - v2_muls( wpos, (1.0f/wpos[3]) * 0.5f, wpos ); - v2_add( wpos, (v2f){ 0.5f, 0.5f }, wpos ); - - ui_rect wr; - 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++ ){ - int collide = 0; - for( int k=0; k= 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, k_ui_align_middle_left, 0 ); - rect_copy( wr, overlap_buffer[ overlap_length ++ ] ); - } - } - } - - audio_unlock(); -} - -#endif /* VG_AUDIO_H */ + mtx_pv );