-/* 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_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
#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
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;
};
};
-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;
u32 samples_last;
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;
float internal_global_volume,
external_global_volume;
}
-static vg_audio = { .external_global_volume = 1.0f, .dsp_enabled = 1 };
-
-#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.
- */
-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;
-}
-
-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 );
-}
-
-static void audio_require_lock(void)
-{
- if( audio_lock_checker_load() )
- return;
-
- vg_error( "Modifying sound effects systems requires locking\n" );
- abort();
-}
-
-static void audio_lock(void)
-{
- SDL_AtomicLock( &vg_audio.sl_sync );
- audio_lock_checker_store(1);
-}
-
-static void audio_unlock(void)
-{
- audio_lock_checker_store(0);
- SDL_AtomicUnlock( &vg_audio.sl_sync );
-}
-static void audio_mixer_callback( void *user, u8 *stream, int frame_count );
-
-static void vg_audio_device_init(void){
- 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( vg_audio.device_choice.buffer, 0,
- &spec_desired, &spec_got,0 );
-
- vg_info( "Start audio device (%u, F32, %u) @%s\n",
- spec_desired.freq,
- AUDIO_FRAME_SIZE,
- vg_audio.device_choice.buffer );
-
- if( vg_audio.sdl_output_device ){
- SDL_PauseAudioDevice( vg_audio.sdl_output_device, 0 );
- vg_success( "Unpaused device %d.\n", vg_audio.sdl_output_device );
- }
- else{
- vg_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" );
- }
-}
-
-static void vg_audio_register(void){
- 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 );
- vg_console_reg_var( "vg_audio_device", &vg_audio.device_choice,
- k_var_dtype_str, VG_VAR_PERSISTENT );
- vg_console_reg_var( "vg_dsp", &vg_audio.dsp_enabled,
- k_var_dtype_i32, VG_VAR_PERSISTENT );
-}
-
-static void vg_audio_init(void){
- /* 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();
- vg_audio_device_init();
-}
-
-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 ){
- audio_require_lock();
- 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 if( (ch->source->flags & AUDIO_FLAG_FORMAT) == k_audio_format_gen )
- strcpy( ch->name, "[program]" );
- 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 )
-{
- audio_require_lock();
- ch->group = group;
- ch->colour = (((u32)group * 29986577) & 0x00ffffff) | 0xff000000;
-}
-
-static void audio_channel_world( audio_channel *ch, u8 world_id )
-{
- audio_require_lock();
- ch->world_id = world_id;
-}
-
-static audio_channel *audio_get_first_idle_channel(void)
-{
- audio_require_lock();
- for( int i=0; i<AUDIO_CHANNELS; i++ ){
- audio_channel *ch = &vg_audio.channels[i];
-
- if( !ch->allocated ){
- return ch;
- }
- }
-
- return NULL;
-}
-
-static audio_channel *audio_get_group_idle_channel( u16 group, u32 max_count )
-{
- audio_require_lock();
- u32 count = 0;
- audio_channel *dest = NULL;
-
- for( int i=0; i<AUDIO_CHANNELS; i++ ){
- audio_channel *ch = &vg_audio.channels[i];
-
- if( ch->allocated ){
- 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 )
-{
- audio_require_lock();
- for( int i=0; i<AUDIO_CHANNELS; i++ ){
- audio_channel *ch = &vg_audio.channels[i];
- if( ch->allocated && (ch->group == group) )
- return ch;
- }
- return NULL;
-}
-
-static int audio_channel_finished( audio_channel *ch )
-{
- audio_require_lock();
- if( ch->readable_activity == k_channel_activity_end )
- return 1;
- else
- return 0;
-}
-
-static audio_channel *audio_relinquish_channel( audio_channel *ch )
-{
- audio_require_lock();
- 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 )
-{
- audio_require_lock();
- 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 )
-{
- audio_require_lock();
- 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 )
-{
- audio_require_lock();
- 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_require_lock();
- audio_channel_slope_volume( ch, length, 0.0f );
- return audio_relinquish_channel( ch );
-}
-
-static void audio_channel_fadein( audio_channel *ch, float length )
-{
- audio_require_lock();
- 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 )
-{
- audio_require_lock();
- 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 )
-{
- audio_require_lock();
- 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 )
-{
- audio_require_lock();
- 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_require_lock();
- 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_require_lock();
- 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_require_lock();
- 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_require_lock();
- 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 if( format == k_audio_format_gen ){
- ch->source_length = 0xffffffff;
- }
- else{
- ch->source_length = ch->source->size;
- }
-
- return 1;
-}
-
-static void audio_decode_uncompressed_mono( i16 *src, u32 count, float *dst )
-{
- for( u32 i=0; i<count; i++ ){
- dst[ i*2 + 0 ] = ((float)src[i]) * (1.0f/32767.0f);
- dst[ i*2 + 1 ] = ((float)src[i]) * (1.0f/32767.0f);
- }
-}
-
-/*
- * adapted from stb_vorbis.h, since the original does not handle mono->stereo
- */
-static int
-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!
- */
-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;i<samples_this_run; i++ ){
- dst[i*2+0] = 0.0f;
- dst[i*2+1] = 0.0f;
- }
- }
- 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 ){
- vg_warn( "Invalid samples read (%s)\n", ch->source->path );
-
- for( int i=0; i<samples_this_run; i++ ){
- dst[i*2+0] = 0.0f;
- dst[i*2+1] = 0.0f;
- }
- }
- }
- else if( format == k_audio_format_bird ){
- synth_bird_generate_samples( ch->bird_handle, dst, samples_this_run );
- }
- else if( format == k_audio_format_gen ){
- void (*fn)( void *data, f32 *buf, u32 count ) = ch->source->func;
- fn( ch->source->data, 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_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 ){
- 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<AUDIO_MIX_FRAME_SIZE; j++ ){
- volume_movement += 1.0f;
- float movement_t = volume_movement * inv_volume_rate;
- movement_t = vg_minf( movement_t, 1.0f );
- volume = vg_lerpf( volume_start, volume_target, movement_t );
-
- float vol_norm = volume * volume;
-
- if( ch->_.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 );
-}
-
-static void audio_mixer_callback( void *user, u8 *stream, int byte_count ){
- /*
- * Copy data and move edit flags to commit flags
- * ------------------------------------------------------------- */
- audio_lock();
- int use_dsp = vg_audio.dsp_enabled;
-
- 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; i<AUDIO_CHANNELS; i++ ){
- audio_channel *ch = &vg_audio.channels[i];
-
- if( !ch->allocated )
- 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; i<AUDIO_LFOS; i++ ){
- audio_lfo *lfo = &vg_audio.oscillators[ i ];
-
- 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 ){
- 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; i<AUDIO_CHANNELS; i++ ){
- audio_channel *ch = &vg_audio.channels[i];
-
- if( ch->activity == 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; i<frame_count*2; i ++ )
- pOut32F[i] = 0.0f;
-
- for( int i=0; i<AUDIO_LFOS; i++ ){
- audio_lfo *lfo = &vg_audio.oscillators[i];
- lfo->time_startframe = lfo->time;
- }
-
- for( int i=0; i<AUDIO_CHANNELS; i ++ ){
- audio_channel *ch = &vg_audio.channels[i];
-
- 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;
- }
- }
- }
-
- if( use_dsp ){
- vg_profile_begin( &_vg_prof_dsp );
- for( int i=0; i<frame_count; i++ )
- vg_dsp_process( pOut32F + i*2, pOut32F + i*2 );
- vg_profile_end( &_vg_prof_dsp );
- }
-
- audio_lock();
-
- for( int i=0; i<AUDIO_CHANNELS; i ++ ){
- audio_channel *ch = &vg_audio.channels[i];
- ch->readable_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();
-}
-
-static void audio_clip_load( audio_clip *clip, void *lin_alloc )
-{
- if( lin_alloc == NULL )
- lin_alloc = vg_audio.audio_pool;
-
-#ifdef VG_AUDIO_FORCE_COMPRESSED
-
- if( (clip->flags & AUDIO_FLAG_FORMAT) != k_audio_format_bird ){
- clip->flags &= ~AUDIO_FLAG_FORMAT;
- clip->flags |= k_audio_format_vorbis;
- }
-
-#endif
-
- /* 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" );
-
-#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
- }
-}
-
-static void audio_clip_loadn( audio_clip *arr, int count, void *lin_alloc )
-{
- for( int i=0; i<count; i++ )
- audio_clip_load( &arr[i], lin_alloc );
-}
-
-static void audio_require_clip_loaded( audio_clip *clip )
-{
- if( clip->data && clip->size )
- return;
-
- audio_unlock();
- vg_fatal_error( "Must load audio clip before playing! \n" );
-}
-
-/*
- * Debugging
- */
-
-static void audio_debug_ui(
+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
- 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;
- 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 }, 0, 0 );
-
-
- 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; i<AUDIO_CHANNELS; i ++ ){
- audio_channel *ch = &vg_audio.channels[i];
-
- ui_rect row;
- ui_split( window, k_ui_axis_h, 18, 1, row, window );
-
- if( !ch->allocated ){
- 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 );
-
-#ifdef VG_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 ){
- 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<overlap_length; k++ ){
- ui_px *wk = overlap_buffer[k];
- if( ((wr[0] <= wk[0]+wk[2]) && (wr[0]+wr[2] >= wk[0])) &&
- ((wr[1] <= wk[1]+wk[3]) && (wr[1]+wr[3] >= wk[1])) )
- {
- collide = 1;
- break;
- }
- }
-
- if( !collide )
- break;
- else
- wr[1] += 18;
- }
-
- ui_text( wr, perf, 1, k_ui_align_middle_left, 0 );
- rect_copy( wr, overlap_buffer[ overlap_length ++ ] );
- }
- }
-#endif
- }
-
- audio_unlock();
-}
-
-#endif /* VG_AUDIO_H */
+ mtx_pv );
projects / vg.git / blobdiff