-// Ogg Vorbis audio decoder - v1.19 - public domain
+// Ogg Vorbis audio decoder - v1.22 - public domain
// http://nothings.org/stb_vorbis/
//
// Original version written by Sean Barrett in 2007.
// Bernhard Wodo Evan Balster github:alxprd
// Tom Beaumont Ingo Leitgeb Nicolas Guillemot
// Phillip Bennefall Rohit Thiago Goulart
-// github:manxorist saga musix github:infatum
+// github:manxorist Saga Musix github:infatum
// Timur Gagiev Maxwell Koo Peter Waller
-// github:audinowho Dougall Johnson
+// github:audinowho Dougall Johnson David Reid
+// github:Clownacy Pedro J. Estebanez Remi Verschelde
+// AnthoFoxo github:morlat Gabriel Ravier
//
// Partial history:
+// 1.22 - 2021-07-11 - various small fixes
+// 1.21 - 2021-07-02 - fix bug for files with no comments
+// 1.20 - 2020-07-11 - several small fixes
// 1.19 - 2020-02-05 - warnings
// 1.18 - 2020-02-02 - fix seek bugs; parse header comments; misc warnings etc.
// 1.17 - 2019-07-08 - fix CVE-2019-13217..CVE-2019-13223 (by ForAllSecure)
// channel. In other words, (*output)[0][0] contains the first sample from
// the first channel, and (*output)[1][0] contains the first sample from
// the second channel.
+//
+// *output points into stb_vorbis's internal output buffer storage; these
+// buffers are owned by stb_vorbis and application code should not free
+// them or modify their contents. They are transient and will be overwritten
+// once you ask for more data to get decoded, so be sure to grab any data
+// you need before then.
extern void stb_vorbis_flush_pushdata(stb_vorbis *f);
// inform stb_vorbis that your next datablock will not be contiguous with
#if defined(_MSC_VER) || defined(__MINGW32__)
#include <malloc.h>
#endif
- #if defined(__linux__) || defined(__linux) || defined(__EMSCRIPTEN__)
+ #if defined(__linux__) || defined(__linux) || defined(__sun__) || defined(__EMSCRIPTEN__) || defined(__NEWLIB__)
#include <alloca.h>
#endif
#else // STB_VORBIS_NO_CRT
#undef __forceinline
#endif
#define __forceinline
- #ifdef alloca
- #undef alloca
- #endif
+ #ifndef alloca
#define alloca __builtin_alloca
+ #endif
#elif !defined(_MSC_VER)
#if __GNUC__
#define __forceinline inline
typedef float codetype;
+#ifdef _MSC_VER
+#define STBV_NOTUSED(v) (void)(v)
+#else
+#define STBV_NOTUSED(v) (void)sizeof(v)
+#endif
+
// @NOTE
//
// Some arrays below are tagged "//varies", which means it's actually
static void setup_temp_free(vorb *f, void *p, int sz)
{
if (f->alloc.alloc_buffer) {
- f->temp_offset += (sz+3)&~3;
+ f->temp_offset += (sz+7)&~7;
return;
}
free(p);
uint32 sign = x & 0x80000000;
uint32 exp = (x & 0x7fe00000) >> 21;
double res = sign ? -(double)mantissa : (double)mantissa;
- return (float) ldexp((float)res, exp-788);
+ return (float) ldexp((float)res, (int)exp-788);
}
// find the first entry
for (k=0; k < n; ++k) if (len[k] < NO_CODE) break;
if (k == n) { assert(c->sorted_entries == 0); return TRUE; }
+ assert(len[k] < 32); // no error return required, code reading lens checks this
// add to the list
add_entry(c, 0, k, m++, len[k], values);
// add all available leaves
uint32 res;
int z = len[i], y;
if (z == NO_CODE) continue;
+ assert(z < 32); // no error return required, code reading lens checks this
// find lowest available leaf (should always be earliest,
// which is what the specification calls for)
// note that this property, and the fact we can never have
while (z > 0 && !available[z]) --z;
if (z == 0) { return FALSE; }
res = available[z];
- assert(z >= 0 && z < 32);
available[z] = 0;
add_entry(c, bit_reverse(res), i, m++, len[i], values);
// propagate availability up the tree
if (z != len[i]) {
- assert(len[i] >= 0 && len[i] < 32);
for (y=len[i]; y > z; --y) {
assert(available[y] == 0);
available[y] = res + (1 << (32-y));
f->valid_bits += 8;
}
}
- if (f->valid_bits < 0) return 0;
+
+ assert(f->valid_bits >= n);
z = f->acc & ((1 << n)-1);
f->acc >>= n;
f->valid_bits -= n;
while (z > base) {
float k00,k11;
-
- k00 = z[-0] - z[-8];
- k11 = z[-1] - z[-9];
- z[-0] = z[-0] + z[-8];
- z[-1] = z[-1] + z[-9];
- z[-8] = k00;
- z[-9] = k11 ;
-
- k00 = z[ -2] - z[-10];
- k11 = z[ -3] - z[-11];
- z[ -2] = z[ -2] + z[-10];
- z[ -3] = z[ -3] + z[-11];
- z[-10] = (k00+k11) * A2;
- z[-11] = (k11-k00) * A2;
-
- k00 = z[-12] - z[ -4]; // reverse to avoid a unary negation
+ float l00,l11;
+
+ k00 = z[-0] - z[ -8];
+ k11 = z[-1] - z[ -9];
+ l00 = z[-2] - z[-10];
+ l11 = z[-3] - z[-11];
+ z[ -0] = z[-0] + z[ -8];
+ z[ -1] = z[-1] + z[ -9];
+ z[ -2] = z[-2] + z[-10];
+ z[ -3] = z[-3] + z[-11];
+ z[ -8] = k00;
+ z[ -9] = k11;
+ z[-10] = (l00+l11) * A2;
+ z[-11] = (l11-l00) * A2;
+
+ k00 = z[ -4] - z[-12];
k11 = z[ -5] - z[-13];
+ l00 = z[ -6] - z[-14];
+ l11 = z[ -7] - z[-15];
z[ -4] = z[ -4] + z[-12];
z[ -5] = z[ -5] + z[-13];
- z[-12] = k11;
- z[-13] = k00;
-
- k00 = z[-14] - z[ -6]; // reverse to avoid a unary negation
- k11 = z[ -7] - z[-15];
z[ -6] = z[ -6] + z[-14];
z[ -7] = z[ -7] + z[-15];
- z[-14] = (k00+k11) * A2;
- z[-15] = (k00-k11) * A2;
+ z[-12] = k11;
+ z[-13] = -k00;
+ z[-14] = (l11-l00) * A2;
+ z[-15] = (l00+l11) * -A2;
iter_54(z);
iter_54(z-8);
for (q=1; q < g->values; ++q) {
j = g->sorted_order[q];
#ifndef STB_VORBIS_NO_DEFER_FLOOR
+ STBV_NOTUSED(step2_flag);
if (finalY[j] >= 0)
#else
if (step2_flag[j])
// WINDOWING
+ STBV_NOTUSED(left_end);
n = f->blocksize[m->blockflag];
map = &f->mapping[m->mapping];
// this isn't to spec, but spec would require us to read ahead
// and decode the size of all current frames--could be done,
// but presumably it's not a commonly used feature
- f->current_loc = -n2; // start of first frame is positioned for discard
+ f->current_loc = 0u - n2; // start of first frame is positioned for discard (NB this is an intentional unsigned overflow/wrap-around)
// we might have to discard samples "from" the next frame too,
// if we're lapping a large block then a small at the start?
f->discard_samples_deferred = n - right_end;
//file vendor
len = get32_packet(f);
f->vendor = (char*)setup_malloc(f, sizeof(char) * (len+1));
+ if (f->vendor == NULL) return error(f, VORBIS_outofmem);
for(i=0; i < len; ++i) {
f->vendor[i] = get8_packet(f);
}
f->vendor[len] = (char)'\0';
//user comments
f->comment_list_length = get32_packet(f);
- f->comment_list = (char**)setup_malloc(f, sizeof(char*) * (f->comment_list_length));
+ f->comment_list = NULL;
+ if (f->comment_list_length > 0)
+ {
+ f->comment_list = (char**) setup_malloc(f, sizeof(char*) * (f->comment_list_length));
+ if (f->comment_list == NULL) return error(f, VORBIS_outofmem);
+ }
for(i=0; i < f->comment_list_length; ++i) {
len = get32_packet(f);
f->comment_list[i] = (char*)setup_malloc(f, sizeof(char) * (len+1));
+ if (f->comment_list[i] == NULL) return error(f, VORBIS_outofmem);
for(j=0; j < len; ++j) {
f->comment_list[i][j] = get8_packet(f);
unsigned int div=1;
for (k=0; k < c->dimensions; ++k) {
int off = (z / div) % c->lookup_values;
- float val = mults[off];
- val = mults[off]*c->delta_value + c->minimum_value + last;
+ float val = mults[off]*c->delta_value + c->minimum_value + last;
c->multiplicands[j*c->dimensions + k] = val;
if (c->sequence_p)
last = val;
if (g->class_masterbooks[j] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
}
for (k=0; k < 1 << g->class_subclasses[j]; ++k) {
- g->subclass_books[j][k] = get_bits(f,8)-1;
+ g->subclass_books[j][k] = (int16)get_bits(f,8)-1;
if (g->subclass_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
}
}
memset(p, 0, sizeof(*p)); // NULL out all malloc'd pointers to start
if (z) {
p->alloc = *z;
- p->alloc.alloc_buffer_length_in_bytes = (p->alloc.alloc_buffer_length_in_bytes+3) & ~3;
+ p->alloc.alloc_buffer_length_in_bytes &= ~7;
p->temp_offset = p->alloc.alloc_buffer_length_in_bytes;
}
p->eof = 0;
*error = VORBIS_need_more_data;
else
*error = p.error;
+ vorbis_deinit(&p);
return NULL;
}
f = vorbis_alloc(&p);
header[i] = get8(f);
if (f->eof) return 0;
if (header[4] != 0) goto invalid;
- goal = header[22] + (header[23] << 8) + (header[24]<<16) + (header[25]<<24);
+ goal = header[22] + (header[23] << 8) + (header[24]<<16) + ((uint32)header[25]<<24);
for (i=22; i < 26; ++i)
header[i] = 0;
crc = 0;
// set. whoops!
break;
}
- previous_safe = last_page_loc+1;
+ //previous_safe = last_page_loc+1; // NOTE: not used after this point, but note for debugging
last_page_loc = stb_vorbis_get_file_offset(f);
}
stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len, int *error, const stb_vorbis_alloc *alloc)
{
stb_vorbis *f, p;
- if (data == NULL) return NULL;
+ if (!data) {
+ if (error) *error = VORBIS_unexpected_eof;
+ return NULL;
+ }
vorbis_init(&p, alloc);
p.stream = (uint8 *) data;
p.stream_end = (uint8 *) data + len;
static void compute_samples(int mask, short *output, int num_c, float **data, int d_offset, int len)
{
- #define BUFFER_SIZE 32
- float buffer[BUFFER_SIZE];
- int i,j,o,n = BUFFER_SIZE;
+ #define STB_BUFFER_SIZE 32
+ float buffer[STB_BUFFER_SIZE];
+ int i,j,o,n = STB_BUFFER_SIZE;
check_endianness();
- for (o = 0; o < len; o += BUFFER_SIZE) {
+ for (o = 0; o < len; o += STB_BUFFER_SIZE) {
memset(buffer, 0, sizeof(buffer));
if (o + n > len) n = len - o;
for (j=0; j < num_c; ++j) {
output[o+i] = v;
}
}
+ #undef STB_BUFFER_SIZE
}
static void compute_stereo_samples(short *output, int num_c, float **data, int d_offset, int len)
{
- #define BUFFER_SIZE 32
- float buffer[BUFFER_SIZE];
- int i,j,o,n = BUFFER_SIZE >> 1;
+ #define STB_BUFFER_SIZE 32
+ float buffer[STB_BUFFER_SIZE];
+ int i,j,o,n = STB_BUFFER_SIZE >> 1;
// o is the offset in the source data
check_endianness();
- for (o = 0; o < len; o += BUFFER_SIZE >> 1) {
+ for (o = 0; o < len; o += STB_BUFFER_SIZE >> 1) {
// o2 is the offset in the output data
int o2 = o << 1;
memset(buffer, 0, sizeof(buffer));
output[o2+i] = v;
}
}
+ #undef STB_BUFFER_SIZE
}
static void convert_samples_short(int buf_c, short **buffer, int b_offset, int data_c, float **data, int d_offset, int samples)
float **outputs;
int len = num_shorts / channels;
int n=0;
- int z = f->channels;
- if (z > channels) z = channels;
while (n < len) {
int k = f->channel_buffer_end - f->channel_buffer_start;
if (n+k >= len) k = len - n;
{
float **outputs;
int n=0;
- int z = f->channels;
- if (z > channels) z = channels;
while (n < len) {
int k = f->channel_buffer_end - f->channel_buffer_start;
if (n+k >= len) k = len - n;
-/* Copyright (C) 2021 Harry Godden (hgn) - All Rights Reserved */
+/* Copyright (C) 2021-2022 Harry Godden (hgn) - All Rights Reserved */
+
+#ifndef VG_AUDIO_H
+#define VG_AUDIO_H
#define MA_NO_GENERATION
#define MA_NO_DECODING
#define MA_NO_ENCODING
#include "dr_soft/miniaudio.h"
+#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 <time.h>
#define STB_VORBIS_MAX_CHANNELS 2
#include "stb/stb_vorbis.h"
-#define SFX_MAX_SYSTEMS 32
-#define SFX_FLAG_STEREO 0x2
-#define SFX_FLAG_REPEAT 0x4
-#define SFX_FLAG_PERSISTENT 0x8
-#define FADEOUT_LENGTH 4410
-#define FADEOUT_DIVISOR (1.f/(float)FADEOUT_LENGTH)
+#define SFX_MAX_SYSTEMS 32
+#define AUDIO_FLAG_LOOP 0x1
+#define AUDIO_FLAG_ONESHOT 0x2
+#define AUDIO_FLAG_SPACIAL_3D 0x4
-typedef struct sfx_vol_control sfx_vol_control;
-typedef struct sfx_system sfx_system;
+#define FADEOUT_LENGTH 4410
+#define FADEOUT_DIVISOR (1.0f/(float)FADEOUT_LENGTH)
-struct sfx_vol_control
+#define AUDIO_DECODE_SIZE (1024*256) /* 256 kb decoding buffers */
+
+enum audio_source_mode
{
- float val;
- const char *name;
+ k_audio_source_mono,
+ k_audio_source_mono_compressed,
+ k_audio_source_stereo_compressed
};
-struct sfx_system
+typedef struct audio_clip audio_clip;
+struct audio_clip
{
- sfx_system *persisitent_source;
- int in_queue;
+ const char *path;
+ enum audio_source_mode source_mode;
- /* Source buffer start */
- float *source, *replacement;
-
- u32 clip_start, clip_end, buffer_length;
-
- /* Modifiers */
- sfx_vol_control *vol_src;
- float vol, cvol, pan;
+ /* result */
+ void *data;
+ u32 len; /* decompressed: sample count,
+ compressed: file size */
+};
- u32 delay;
-
- /* Info */
- u32 ch, end, cur;
- u32 flags;
-
- /* Effects */
- u32 fadeout, fadeout_current;
-
- /* Diagnostic */
- const char *name;
+typedef struct audio_mix_info audio_mix_info;
+struct audio_mix_info
+{
+ audio_clip *source;
+ v3f world_position;
+
+ float vol, pan;
+ u32 flags;
};
-/* Set of up to 8 sound effects packed into one */
-typedef struct sfx_set sfx_set;
-struct sfx_set
+typedef struct audio_player audio_player;
+struct audio_player
{
- float *main;
- char *sources;
-
- u32 segments[32]; /* from->to,from->to ... */
- u32 numsegments;
- u32 ch;
- u32 flags;
+ aatree_ptr active_entity; /* non-nil if currently playing */
+ audio_mix_info info;
+ int enqued;
+
+ /* Diagnostic */
+ const char *name;
};
-ma_device g_aud_device;
-ma_device_config g_aud_dconfig;
+typedef struct audio_entity audio_entity;
+struct audio_entity
+{
+ audio_player *player;
+ audio_mix_info info;
+
+ u32 length, cur;
+
+ /* Effects */
+ u32 fadeout, fadeout_current;
+ const char *name;
+};
-/*
- * Thread 1 - audio engine ( spawned from miniaudio.h )
- * ======================================================
+/*
+ * TODO list sunday
+ *
+ * play again: if already playing, leave in queue while it fadeouts
+ * oneshot: create a ghost entity
+ *
*/
-sfx_system sfx_sys[SFX_MAX_SYSTEMS];
-int sfx_sys_len = 0;
+
+static struct vg_audio_system
+{
+ ma_device miniaudio_device;
+ ma_device_config miniaudio_dconfig;
+
+ void *mem, *decode_mem;
+ u32 mem_current,
+ mem_total;
+
+ /* synchro */
+ int sync_locked;
+ MUTEX_TYPE mutex_checker;
+ MUTEX_TYPE mutex_sync;
+
+ /* Audio engine, thread 1 */
+ struct active_audio_player
+ {
+ int active;
+ union
+ {
+ audio_entity ent;
+ aatree_pool_node pool_node;
+ };
+
+ stb_vorbis *vorbis_handle;
+ stb_vorbis_alloc vorbis_alloc;
+ }
+ active_players[ SFX_MAX_SYSTEMS ];
+
+ aatree active_pool_info; /* note: just using the pool */
+ aatree_ptr active_pool_head;
+
+ /* System queue, and access from thread 0 */
+ audio_entity entity_queue[SFX_MAX_SYSTEMS];
+ int queue_len;
+
+ char performance_info[128];
+ int debug_ui;
+
+ v3f listener_pos,
+ listener_ears;
+}
+vg_audio;
+
+static void *audio_alloc( u32 size )
+{
+ u32 new_current = vg_audio.mem_current + size;
+ if( new_current > vg_audio.mem_total )
+ {
+ vg_error( "audio pool over budget!\n" );
+ free( vg_audio.mem );
+ return NULL;
+ }
+
+ void *ptr = vg_audio.mem + vg_audio.mem_current;
+ vg_audio.mem_current = new_current;
+
+ return ptr;
+}
+
/*
- * Thread 0 - Critical transfer section
- * ======================================================
+ * These functions are called from the main thread and used to prevent bad
+ * access. TODO: They should be no-ops in release builds.
*/
-MUTEX_TYPE sfx_mux_t01; /* Resources share: 0 & 1 */
+static int audio_lock_checker_load(void)
+{
+ int value;
+ MUTEX_LOCK( vg_audio.mutex_checker );
+ value = vg_audio.sync_locked;
+ MUTEX_UNLOCK( vg_audio.mutex_checker );
+ return value;
+}
-sfx_system *sfx_q[SFX_MAX_SYSTEMS]; /* Stuff changed */
-int sfx_q_len = 0;
+static void audio_lock_checker_store( int value )
+{
+ MUTEX_LOCK( vg_audio.mutex_checker );
+ vg_audio.sync_locked = value;
+ MUTEX_UNLOCK( vg_audio.mutex_checker );
+}
-float g_master_volume = 1.f;
+static void audio_require_lock(void)
+{
+ if( audio_lock_checker_load() )
+ return;
-/* Decompress entire vorbis stream into buffer */
-static float *sfx_vorbis_stream( const unsigned char *data, int len,
- int channels, u32 *samples )
+ vg_exiterr( "Modifying sound effects systems requires locking\n" );
+}
+
+static void audio_lock(void)
{
- int err;
- stb_vorbis *pv = stb_vorbis_open_memory( data, len, &err, NULL );
-
- if( !pv )
- {
- vg_error( "stb_vorbis_open_memory() failed with error code: %i\n", err );
- return NULL;
- }
-
- u32 length_samples = stb_vorbis_stream_length_in_samples( pv );
- float *buffer = (float *)malloc( length_samples * channels * sizeof(float));
-
- if( !buffer )
- {
- stb_vorbis_close( pv );
- vg_error( "out of memory while allocating sound resource\n" );
- return NULL;
- }
-
- int read_samples = stb_vorbis_get_samples_float_interleaved(
- pv, channels, buffer, length_samples * channels );
+ MUTEX_LOCK( vg_audio.mutex_sync );
+ audio_lock_checker_store(1);
+}
- if( read_samples != length_samples )
- {
- vg_warn( "| warning: sample count mismatch. Expected %u got %i\n",
- length_samples, read_samples );
- length_samples = read_samples;
- }
-
- stb_vorbis_close( pv );
- *samples = length_samples;
- return buffer;
+static void audio_unlock(void)
+{
+ audio_lock_checker_store(0);
+ MUTEX_UNLOCK( vg_audio.mutex_sync );
}
-static float *sfx_vorbis( const char *strFileName, int channels, u32 *samples )
+
+static void audio_mixer_callback( ma_device *pDevice, void *pOutBuf,
+ const void *pInput, ma_uint32 frameCount );
+
+static int vg_audio_init(void)
{
- i64 len;
- void *filedata = vg_asset_read_s( strFileName, &len );
-
- if( filedata )
- {
- float *wav = sfx_vorbis_stream( filedata, len, channels, samples );
- free( filedata );
- return wav;
- }
- else
- {
- vg_error( "OGG load failed\n" );
- return NULL;
- }
+ vg_convar_push( (struct vg_convar){
+ .name = "debug_audio",
+ .data = &vg_audio.debug_ui,
+ .data_type = k_convar_dtype_i32,
+ .opt_i32 = { .min=0, .max=1, .clamp=1 },
+ .persistent = 1
+ });
+
+ u32 decode_region = AUDIO_DECODE_SIZE * SFX_MAX_SYSTEMS;
+ vg_audio.mem_total = 1024*1024*32;
+ vg_audio.mem_current = 0;
+ vg_audio.mem = malloc( vg_audio.mem_total + decode_region );
+ vg_audio.decode_mem = &((u8 *)vg_audio.mem)[vg_audio.mem_total];
+
+ /* setup pool */
+ vg_audio.active_pool_info.base = vg_audio.active_players;
+ vg_audio.active_pool_info.offset = offsetof(struct active_audio_player,
+ pool_node );
+ vg_audio.active_pool_info.stride = sizeof(struct active_audio_player);
+ vg_audio.active_pool_info.p_cmp = NULL;
+ aatree_init_pool( &vg_audio.active_pool_info, SFX_MAX_SYSTEMS );
+
+ ma_device_config *dconf = &vg_audio.miniaudio_dconfig;
+ ma_device *device = &vg_audio.miniaudio_device;
+
+ *dconf = ma_device_config_init( ma_device_type_playback );
+ dconf->playback.format = ma_format_f32;
+ dconf->playback.channels = 2;
+ dconf->sampleRate = 44100;
+ dconf->dataCallback = audio_mixer_callback;
+
+ dconf->pUserData = NULL;
+
+ vg_info( "Starting audio engine\n" );
+
+ if( ma_device_init( NULL, dconf, device) != MA_SUCCESS )
+ {
+ vg_error( "ma_device failed to initialize" );
+ return 0;
+ }
+ else
+ {
+ if( ma_device_start( device ) != MA_SUCCESS )
+ {
+ ma_device_uninit( device );
+ vg_error( "ma_device failed to start" );
+ return 0;
+ }
+ }
+
+ return 1;
+}
+
+static void vg_audio_free(void)
+{
+ ma_device *device = &vg_audio.miniaudio_device;
+ ma_device_uninit( device );
+
+ free( vg_audio.mem );
}
/*
- * thread 0 / client code
+ * thread 1
*/
-static int sfx_begin_edit( sfx_system *sys )
+
+static aatree_ptr audio_alloc_entity_internal(void)
{
- MUTEX_LOCK( sfx_mux_t01 );
-
- if( sfx_q_len >= SFX_MAX_SYSTEMS && !sys->in_queue )
- {
- MUTEX_UNLOCK( sfx_mux_t01 );
- vg_warn( "Warning: No free space in sound queue\n" );
- return 0;
- }
-
- return 1;
+ aatree_ptr playerid = aatree_pool_alloc( &vg_audio.active_pool_info,
+ &vg_audio.active_pool_head );
+
+ if( playerid == AATREE_PTR_NIL )
+ return AATREE_PTR_NIL;
+
+ struct active_audio_player *aap = &vg_audio.active_players[ playerid ];
+ aap->active = 1;
+
+ return playerid;
}
-static void sfx_end_edit( sfx_system *sys )
+static void audio_entity_free_internal( aatree_ptr id )
{
- MUTEX_UNLOCK( sfx_mux_t01 );
+ struct active_audio_player *aap = &vg_audio.active_players[ id ];
+ aap->active = 0;
+
+ /* Notify player that we've finished */
+ if( aap->ent.player )
+ aap->ent.player->active_entity = AATREE_PTR_NIL;
+
+ /* delete */
+ aatree_pool_free( &vg_audio.active_pool_info, id,
+ &vg_audio.active_pool_head );
}
-/* Mark change to be uploaded to queue system */
-static int sfx_push( sfx_system *sys )
+static void *audio_entity_vorbis_ptr( aatree_ptr entid )
{
- if( !sys->in_queue )
- {
- sfx_q[ sfx_q_len ++ ] = sys;
- sys->in_queue = 1;
- }
-
- MUTEX_UNLOCK( sfx_mux_t01 );
-
- return 1;
+ u8 *buf = (u8*)vg_audio.decode_mem,
+ *loc = &buf[AUDIO_DECODE_SIZE*entid];
+
+ return (void *)loc;
}
-/* Edit a volume float, has to be function wrapped because of mutex */
-static void sfx_vol_fset( sfx_vol_control *src, float to )
+static void audio_entity_start( audio_entity *src )
{
- MUTEX_LOCK( sfx_mux_t01 );
+ aatree_ptr entid = audio_alloc_entity_internal();
+ if( entid == AATREE_PTR_NIL )
+ return;
+
+ audio_entity *ent = &vg_audio.active_players[ entid ].ent;
+
+ ent->info = src->info;
+ ent->name = "todo";
+ ent->cur = 0;
+ ent->player = src->player;
+
+ ent->fadeout = 0;
+ ent->fadeout_current = 0;
+
+ /* Notify main player we are dequeud and playing */
+ if( src->player )
+ {
+ src->player->enqued = 0;
+ src->player->active_entity = entid;
+ }
+
+ if( src->info.source->source_mode == k_audio_source_mono_compressed ||
+ src->info.source->source_mode == k_audio_source_stereo_compressed )
+ {
+ /* Setup vorbis decoder */
+ struct active_audio_player *aap = &vg_audio.active_players[ entid ];
+
+ stb_vorbis_alloc alloc = {
+ .alloc_buffer = (char *)audio_entity_vorbis_ptr( entid ),
+ .alloc_buffer_length_in_bytes = AUDIO_DECODE_SIZE
+ };
+
+ int err;
+ stb_vorbis *decoder = stb_vorbis_open_memory(
+ src->info.source->data, src->info.source->len, &err, &alloc );
+
+ if( !decoder )
+ {
+ vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
+ src->info.source->path, err );
+
+ audio_entity_free_internal( entid );
+ return;
+ }
+ else
+ {
+ ent->length = stb_vorbis_stream_length_in_samples( decoder );
+ }
+
+ aap->vorbis_handle = decoder;
+ }
+ else
+ {
+ ent->length = src->info.source->len;
+ }
+}
- src->val = to;
+/*
+ * Read everything from the queue
+ */
+static void audio_system_enque(void)
+{
+ /* Process incoming sound queue */
+ audio_lock();
+
+ int wr = 0;
+ for( int i=0; i<vg_audio.queue_len; i++ )
+ {
+ audio_entity *src = &vg_audio.entity_queue[ i ];
+
+ if( src->player )
+ {
+ /* Start new */
+ if( src->player->active_entity == AATREE_PTR_NIL )
+ {
+ audio_entity_start( src );
+ }
+ else
+ {
+ /* Otherwise try start fadeout but dont remove from queue */
+
+ aatree_ptr entid = src->player->active_entity;
+ audio_entity *ent = &vg_audio.active_players[ entid ].ent;
+ if( !ent->fadeout )
+ {
+ ent->fadeout = 1;
+ ent->fadeout_current = FADEOUT_LENGTH;
+ }
+
+ vg_audio.entity_queue[ wr ++ ] = *src;
+ }
+ }
+ else
+ {
+ audio_entity_start( src );
+ }
+ }
+
+ vg_audio.queue_len = wr;
+
+ /* Localize others memory */
+ for( int i=0; i<SFX_MAX_SYSTEMS; i ++ )
+ {
+ struct active_audio_player *aap = &vg_audio.active_players[i];
+ if( !aap->active )
+ continue;
+
+ if( aap->ent.player )
+ {
+ /* Only copy information in whilst not requeing */
+ if( aap->ent.player->enqued == 0 )
+ {
+ aap->ent.info = aap->ent.player->info;
+ }
+ }
+ }
+
+ audio_unlock();
+}
- MUTEX_UNLOCK( sfx_mux_t01 );
+/*
+ * Redistribute sound systems
+ */
+static void audio_system_cleanup(void)
+{
+ audio_lock();
+
+ for( int i=0; i<SFX_MAX_SYSTEMS; i ++ )
+ {
+ struct active_audio_player *aap = &vg_audio.active_players[i];
+ if( aap->active )
+ {
+ audio_entity *src = &aap->ent;
+ if( src->cur < src->length || (src->info.flags & AUDIO_FLAG_LOOP ))
+ {
+ /* Good to keep */
+ }
+ else
+ {
+ audio_entity_free_internal( i );
+ }
+ }
+ }
+
+ audio_unlock();
}
-/* thread-safe get volume value */
-static float sfx_vol_fget( sfx_vol_control *src )
+/*
+ * Get effective volume and pan from this entity
+ */
+static void audio_entity_spacialize( audio_entity *ent, float *vol, float *pan )
{
- float val;
-
- MUTEX_LOCK( sfx_mux_t01 );
-
- val = src->val;
-
- MUTEX_UNLOCK( sfx_mux_t01 );
-
- return val;
+ v3f delta;
+ v3_sub( ent->info.world_position, vg_audio.listener_pos, delta );
+
+ float dist = v3_length( delta ),
+ attn = (dist / ent->info.vol) +1.0f;
+
+ v3_muls( delta, 1.0f/dist, delta );
+
+ *pan = v3_dot( vg_audio.listener_ears, delta );
+ *vol = 1.0f/(attn*attn);
}
-/* thread-safe set master volume */
-static void sfx_set_master( float to )
+static void audio_decode_uncompressed_mono( float *src, u32 count, float *dst )
{
- MUTEX_LOCK( sfx_mux_t01 );
-
- g_master_volume = to;
-
- MUTEX_UNLOCK( sfx_mux_t01 );
+ for( u32 i=0; i<count; i++ )
+ {
+ dst[ i*2 + 0 ] = src[i];
+ dst[ i*2 + 1 ] = src[i];
+ }
}
-/* thread-safe get master volume */
-static float sfx_get_master(void)
+static void audio_entity_get_samples( aatree_ptr id, u32 count, float *buf )
{
- float val;
+ struct active_audio_player *aap = &vg_audio.active_players[id];
+ audio_entity *ent = &aap->ent;
+
+ u32 remaining = count;
+ u32 cursor = ent->cur;
+ u32 buffer_pos = 0;
+
+ while( remaining )
+ {
+ u32 samples_this_run = VG_MIN( remaining, ent->length - cursor );
+ remaining -= samples_this_run;
+
+ float *dst = &buf[ buffer_pos * 2 ];
+
+ if( ent->info.source->source_mode == k_audio_source_mono )
+ {
+ float *src = &((float *)ent->info.source->data)[ cursor ];
+ audio_decode_uncompressed_mono( src, samples_this_run, dst );
+ }
+ else if( ent->info.source->source_mode == k_audio_source_mono_compressed )
+ {
+ int read_samples = stb_vorbis_get_samples_float_interleaved(
+ aap->vorbis_handle,
+ 2,
+ dst,
+ samples_this_run * 2 );
+
+ if( read_samples != samples_this_run )
+ {
+ vg_warn( "Invalid samples read (%s)\n", ent->info.source->path );
+ }
+ }
+
+ cursor += samples_this_run;
+ buffer_pos += samples_this_run;
+
+ if( (ent->info.flags & AUDIO_FLAG_LOOP) && remaining )
+ {
+ if( ent->info.source->source_mode == k_audio_source_mono_compressed ||
+ ent->info.source->source_mode == k_audio_source_stereo_compressed )
+ {
+ stb_vorbis_seek_start( aap->vorbis_handle );
+ }
+
+ cursor = 0;
+ continue;
+ }
+ else
+ break;
+ }
+
+ while( remaining )
+ {
+ buf[ buffer_pos*2 + 0 ] = 0.0f;
+ buf[ buffer_pos*2 + 1 ] = 0.0f;
+ buffer_pos ++;
+
+ remaining --;
+ }
+
+ ent->cur = cursor;
+}
- MUTEX_LOCK( sfx_mux_t01 );
-
- val = g_master_volume;
-
- MUTEX_UNLOCK( sfx_mux_t01 );
-
- return val;
+static void audio_entity_mix( aatree_ptr id, float *buffer,
+ u32 frame_count )
+{
+ audio_entity *ent = &vg_audio.active_players[id].ent;
+
+ u32 cursor = ent->cur, buffer_pos = 0;
+ float *pcf = alloca( frame_count * 2 * sizeof(float) );
+
+ u32 frames_write = frame_count;
+ float fadeout_divisor = 1.0f / (float)ent->fadeout;
+
+ float vol = ent->info.vol,
+ pan = ent->info.pan;
+
+ audio_entity_get_samples( id, frame_count, pcf );
+
+ if( ent->info.flags & AUDIO_FLAG_SPACIAL_3D )
+ audio_entity_spacialize( ent, &vol, &pan );
+
+ for( u32 j=0; j<frame_count; j++ )
+ {
+ float frame_vol = vol;
+
+ if( ent->fadeout )
+ {
+ /* Force this system to be removed now */
+ if( ent->fadeout_current == 0 )
+ {
+ ent->info.flags = 0x00;
+ ent->cur = ent->length;
+ break;
+ }
+
+ frame_vol *= (float)ent->fadeout_current * fadeout_divisor;
+ ent->fadeout_current --;
+ }
+
+ float sl = 1.0f-pan,
+ sr = 1.0f+pan;
+
+ buffer[ buffer_pos*2+0 ] += pcf[ buffer_pos*2+0 ] * frame_vol * sl;
+ buffer[ buffer_pos*2+1 ] += pcf[ buffer_pos*2+1 ] * frame_vol * sr;
+
+ buffer_pos ++;
+ }
}
-void audio_mixer_callback( ma_device *pDevice, void *pOutBuf,
- const void *pInput, ma_uint32 frameCount );
+static void vg_sleep_ms( long msec )
+{
+ struct timespec ts;
-static void vg_audio_init(void)
+ ts.tv_sec = msec / 1000;
+ ts.tv_nsec = (msec % 1000) * 1000000;
+ nanosleep( &ts, &ts );
+}
+
+/*
+ * callback from miniaudio.h interface
+ */
+static void audio_mixer_callback( ma_device *pDevice, void *pOutBuf,
+ const void *pInput, ma_uint32 frame_count )
{
- g_aud_dconfig = ma_device_config_init( ma_device_type_playback );
- g_aud_dconfig.playback.format = ma_format_f32;
- g_aud_dconfig.playback.channels = 2;
- g_aud_dconfig.sampleRate = 44100;
- g_aud_dconfig.dataCallback = audio_mixer_callback;
-
- g_aud_dconfig.pUserData = NULL;
-
- vg_info( "Starting audio engine\n" );
-
- if( ma_device_init( NULL, &g_aud_dconfig, &g_aud_device ) != MA_SUCCESS )
- {
- vg_exiterr( "ma_device failed to initialize" );
- }
- else
- {
- if( ma_device_start( &g_aud_device ) != MA_SUCCESS )
- {
- ma_device_uninit( &g_aud_device );
- vg_exiterr( "ma_device failed to start" );
- }
- }
+ struct timespec time_start, time_end;
+ clock_gettime( CLOCK_REALTIME, &time_start );
+
+ audio_system_enque();
+
+ /* Clear buffer */
+ float *pOut32F = (float *)pOutBuf;
+ for( int i=0; i<frame_count*2; i ++ )
+ pOut32F[i] = 0.0f;
+
+ /* Mix all sounds */
+ for( int i=0; i<SFX_MAX_SYSTEMS; i ++ )
+ {
+ struct active_audio_player *aap = &vg_audio.active_players[i];
+
+ if( aap->active )
+ audio_entity_mix( i, pOut32F, frame_count );
+ }
+
+#if 0
+ vg_sleep_ms( 20 );
+#endif
+
+ /* redistribute */
+ audio_system_cleanup();
+
+ /* TODO: what the hell is this? */
+ (void)pInput;
+
+ /*
+ * Debug information
+ */
+ clock_gettime( CLOCK_REALTIME, &time_end );
+
+ double elapsed = 1000.0*time_end.tv_sec + 1e-6*time_end.tv_nsec
+ - (1000.0*time_start.tv_sec + 1e-6*time_start.tv_nsec),
+ budget = ((double)frame_count / 44100.0) * 1000.0,
+ percent = (elapsed/budget) * 100.0;
+
+ snprintf( vg_audio.performance_info, 127,
+ "%.1fms/%.1fms (%.1f%%) (%u frames)",
+ elapsed, budget, percent, frame_count );
}
-static void vg_audio_free(void)
+/* Decompress entire vorbis stream into buffer */
+static float *audio_decompress_vorbis( const unsigned char *data, int len,
+ int channels, u32 *samples )
{
- ma_device_uninit( &g_aud_device );
+ int err;
+ stb_vorbis *pv = stb_vorbis_open_memory( data, len, &err, NULL );
+
+ if( !pv )
+ {
+ vg_error( "stb_vorbis_open_memory() failed with error code: %i\n", err );
+ return NULL;
+ }
+
+ u32 length_samples = stb_vorbis_stream_length_in_samples( pv );
+
+ vg_info( "decompress_vorbis: %u samples (%.2fs), %.1fkb\n",
+ length_samples,
+ (float)length_samples / (44100.0f*(float)channels),
+ (float)(length_samples*4*channels) / 1024.0f );
+
+ float *buffer = audio_alloc( length_samples * channels * sizeof(float) );
+ if( !buffer )
+ {
+ stb_vorbis_close( pv );
+ vg_exit();
+ }
+
+ int read_samples = stb_vorbis_get_samples_float_interleaved(
+ pv, channels, buffer, length_samples * channels );
+
+ if( read_samples != length_samples )
+ {
+ vg_warn( "| warning: sample count mismatch. Expected %u got %i\n",
+ length_samples, read_samples );
+ length_samples = read_samples;
+ }
+
+ stb_vorbis_close( pv );
+ *samples = length_samples;
+ return buffer;
}
-/*
- * thread 1
+static int audio_clip_load( audio_clip *clip )
+{
+ /* Load and decompress */
+ i64 file_len;
+ void *filedata = vg_asset_read_s( clip->path, &file_len );
+
+ if( !filedata )
+ {
+ vg_error( "OGG load failed (%s)\n", clip->path );
+ return 0;
+ }
+
+ if( clip->source_mode == k_audio_source_mono )
+ {
+ u32 samples;
+ float *sound = audio_decompress_vorbis( filedata, file_len, 1, &samples );
+ clip->data = sound;
+ clip->len = samples;
+
+ float seconds = (float)samples / 44100.0f,
+ mb = (float)(samples*4) / (1024.0f*1024.0f);
+
+ vg_info( "Loaded audio clip[mono] '%s' (%.1fs, %.1fmb)\n",
+ clip->path, seconds, mb );
+ }
+ else if( clip->source_mode == k_audio_source_mono_compressed )
+ {
+ void *data = audio_alloc( file_len );
+ memcpy( data, filedata, file_len );
+
+ clip->data = data;
+ clip->len = file_len;
+
+ float mb = (float)(file_len) / (1024.0f*1024.0f);
+ vg_info( "Loaded audio clip[mono_compressed] '%s' (%.1fmb)\n",
+ clip->path, mb );
+ }
+ else if( clip->source_mode == k_audio_source_stereo_compressed )
+ {
+ /* ... */
+
+ clip->data = NULL;
+ clip->len = 0;
+
+ vg_error( "Source mode (%u) currently unsupported\n", clip->source_mode );
+ return 0;
+ }
+ else
+ {
+ /* ... */
+
+ clip->data = NULL;
+ clip->len = 0;
+
+ vg_error( "Unkown source mode (%u)\n", clip->source_mode );
+ return 0;
+ }
+
+ return 1;
+}
+
+static void audio_clip_loadn( audio_clip *arr, int count )
+{
+ for( int i=0; i<count; i++ )
+ audio_clip_load( &arr[i] );
+}
+
+#if 0
+/*
+ * Client code
*/
+static void audio_pack_play( audio_pack *source, audio_player *sys, int id )
+{
+ audio_require_lock();
+
+ sys->fadeout = 0;
+ sys->fadeout_current = 0;
+ sys->source = source->data;
+ sys->cur = source->segments[ id*2 + 0 ];
+ sys->end = source->segments[ id*2 + 1 ];
+ sys->ch = source->ch;
+ sys->source_mode = source->source_mode;
+
+ /* for diagnostics */
+ sys->clip_start = sys->cur;
+ sys->clip_end = sys->end;
+ sys->buffer_length = source->segments[ (source->numsegments-1)*2 + 1 ];
+ sys->is_playing = 1;
+
+ audio_player_push( sys );
+}
-static sfx_system *sfx_alloc(void)
+#endif
+
+/* Mark change to be uploaded through queue system */
+static void audio_player_commit( audio_player *sys )
{
- if( sfx_sys_len >= SFX_MAX_SYSTEMS )
- return NULL;
-
- /*
- * A conditional is done against this in localization step,
- * Needs to be initialized.
- */
- sfx_sys[ sfx_sys_len ].source = NULL;
-
- return sfx_sys + (sfx_sys_len++);
+ audio_require_lock();
+
+ if( vg_audio.queue_len >= vg_list_size( vg_audio.entity_queue ) )
+ {
+ vg_warn( "Audio commit queue full\n" );
+ return;
+ }
+
+ if( sys->enqued )
+ {
+ vg_warn( "Audio commit spamming; already enqued (%s)\n", sys->name );
+ return;
+ }
+
+ sys->enqued = 1;
+ audio_entity *ent = &vg_audio.entity_queue[ vg_audio.queue_len ++ ];
+ ent->info = sys->info;
+ ent->player = sys;
+ sys->active_entity = AATREE_PTR_NIL;
}
-/* Fetch samples into pcf */
-static void audio_mixer_getsamples( float *pcf, float *source, u32 cur, u32 ch )
+/* Play a clip using player. If its already playing something, it will
+ * fadeout quickly and start the next sound */
+static void audio_player_playclip( audio_player *player, audio_clip *clip )
{
- if( ch == 2 )
- {
- pcf[0] = source[ cur*2+0 ];
- pcf[1] = source[ cur*2+1 ];
- }
- else
- {
- pcf[0] = source[ cur ];
- pcf[1] = source[ cur ];
- }
+ audio_require_lock();
+
+ player->info.source = clip;
+ audio_player_commit( player );
+}
+
+static void audio_player_playoneshot( audio_player *player, audio_clip *clip )
+{
+
}
/*
- * callback from miniaudio.h interface
+ * Effects
*/
-void audio_mixer_callback( ma_device *pDevice, void *pOutBuf,
- const void *pInput, ma_uint32 frameCount )
+
+/*
+ * Safety enforced Get/set attributes
+ */
+
+static void audio_player_set_position( audio_player *sys, v3f pos )
{
- /* Process incoming sound queue */
- MUTEX_LOCK( sfx_mux_t01 );
-
- while( sfx_q_len --> 0 )
- {
- sfx_system *src = sfx_q[sfx_q_len];
- sfx_system *clone;
-
- src->in_queue = 0;
-
- clone = sfx_alloc();
- *clone = *src;
-
- /* Links need to exist on persistent sounds */
- clone->persisitent_source = src->flags & SFX_FLAG_PERSISTENT? src: NULL;
- }
-
- sfx_q_len = 0;
-
- /* Volume modifiers */
- for( int i = 0; i < sfx_sys_len; i ++ )
- {
- sfx_system *sys = sfx_sys + i;
-
- /* Apply persistent volume if linked */
- if( sys->flags & SFX_FLAG_PERSISTENT )
- {
- sys->vol = sys->persisitent_source->vol * g_master_volume;
- sys->pan = sys->persisitent_source->pan;
-
- /* Fadeout effect ( + remove ) */
- if( sys->persisitent_source->fadeout )
- {
- sys->fadeout_current = sys->persisitent_source->fadeout_current;
- sys->fadeout = sys->persisitent_source->fadeout;
-
- sys->persisitent_source = NULL;
- sys->flags &= ~SFX_FLAG_PERSISTENT;
- }
- }
-
- /* Apply volume slider if it has one linked */
- if( sys->vol_src )
- sys->cvol = sys->vol * sys->vol_src->val;
- else
- sys->cvol = sys->vol;
- }
-
- MUTEX_UNLOCK( sfx_mux_t01 );
-
- /* Clear buffer */
- float *pOut32F = (float *)pOutBuf;
- for( int i = 0; i < frameCount * 2; i ++ ){
- pOut32F[i] = 0.f;
- }
+ audio_require_lock();
+ v3_copy( pos, sys->info.world_position );
+}
- for( int i = 0; i < sfx_sys_len; i ++ )
- {
- sfx_system *sys = sfx_sys + i;
-
- u32 cursor = sys->cur, buffer_pos = 0;
- float pcf[2] = { 0.f, 0.0f };
-
- u32 frames_write = frameCount;
- float fadeout_divisor = 1.0f / (float)sys->fadeout;
-
- while( frames_write )
- {
- u32 samples_this_run = VG_MIN( frames_write, sys->end - cursor );
-
- if( sys->fadeout )
- {
- /* Force this system to be removed now */
- if( sys->fadeout_current == 0 )
- {
- sys->flags &= 0x00000000;
- sys->cur = sys->end;
- break;
- }
-
- samples_this_run = VG_MIN( samples_this_run, sys->fadeout_current );
- }
-
- for( u32 j=0; j<samples_this_run; j++ )
- {
- audio_mixer_getsamples( pcf, sys->source, cursor, sys->ch );
-
- float vol = sys->cvol;
-
- if( sys->fadeout )
- {
- vol *= (float)sys->fadeout_current * fadeout_divisor;
- sys->fadeout_current --;
- }
-
- if( buffer_pos >= frameCount )
- {
- break;
- }
-
- float sl = 1.0f-sys->pan,
- sr = 1.0f+sys->pan;
-
- pOut32F[ buffer_pos*2+0 ] += pcf[0] * vol * sl;
- pOut32F[ buffer_pos*2+1 ] += pcf[1] * vol * sr;
-
- cursor ++;
- buffer_pos ++;
- }
-
- frames_write -= samples_this_run;
-
- if( sys->flags & SFX_FLAG_REPEAT )
- {
- if( frames_write )
- {
- cursor = sys->clip_start;
- continue;
- }
- }
-
- sys->cur = cursor;
- break;
- }
- }
+static void audio_player_set_vol( audio_player *sys, float vol )
+{
+ audio_require_lock();
+ sys->info.vol = vol;
+}
- /* Redistribute sound systems */
- MUTEX_LOCK( sfx_mux_t01 );
+static float audio_player_get_vol( audio_player *sys )
+{
+ audio_require_lock();
+ return sys->info.vol;
+}
- u32 idx = 0, wr = 0;
- while( idx != sfx_sys_len )
- {
- sfx_system *src = sfx_sys + idx;
-
- /* Keep only if cursor is before end or repeating */
- if( src->cur < src->end || (src->flags & SFX_FLAG_REPEAT) )
- {
- sfx_sys[ wr ++ ] = sfx_sys[ idx ];
- }
-
- idx ++ ;
- }
- sfx_sys_len = wr;
-
- MUTEX_UNLOCK( sfx_mux_t01 );
-
- (void)pInput;
+static void audio_player_set_pan( audio_player *sys, float pan )
+{
+ audio_require_lock();
+ sys->info.pan = pan;
}
-/*
- * Load strings into sfx_set's memory
- * String layout: "sounda.ogg\0soundb.ogg\0soundc.ogg\0\0"
- */
-static void sfx_set_strings( sfx_set *dest, char *strSources,
- u32 flags, int bAsync )
+static float audio_player_get_pan( audio_player *sys )
{
- dest->ch = (flags & SFX_FLAG_STEREO)? 2: 1;
-
- dest->main = NULL;
- dest->numsegments = 0;
- char *source = strSources;
-
- u32 total = 0;
- int len;
- while( (len = strlen( source )) )
- {
- u32 samples;
- float *sound = sfx_vorbis( source, dest->ch, &samples );
-
- if( !sound )
- {
- free( dest->main );
- dest->numsegments = 0;
- return;
- }
-
- total += samples;
-
- float *nbuf = realloc( dest->main, total * dest->ch * sizeof(float) );
-
- if( nbuf )
- {
- dest->main = nbuf;
- memcpy( dest->main + (total-samples)*dest->ch,
- sound, samples*dest->ch*sizeof(float) );
- free( sound );
-
- dest->segments[ dest->numsegments*2+0 ] = total-samples;
- dest->segments[ dest->numsegments*2+1 ] = total;
- }
- else
- {
- vg_error( "realloc() failed\n" );
- free( sound );
- return;
- }
-
- source += len +1;
- dest->numsegments ++;
- }
+ audio_require_lock();
+ return sys->info.pan;
}
-static void sfx_set_init( sfx_set *dest, char *sources )
+static void audio_player_set_flags( audio_player *sys, u32 flags )
{
- if( !sources )
- sfx_set_strings( dest, dest->sources, dest->flags, 0 );
- else
- sfx_set_strings( dest, sources, dest->flags, 0 );
+ audio_require_lock();
+ sys->info.flags = flags;
}
-static void sfx_set_play( sfx_set *source, sfx_system *sys, int id )
+static u32 audio_player_get_flags( audio_player *sys )
{
- if( sfx_begin_edit( sys ) )
- {
- sys->fadeout = 0;
- sys->fadeout_current = 0;
- sys->source = source->main;
- sys->cur = source->segments[ id*2 + 0 ];
- sys->end = source->segments[ id*2 + 1 ];
- sys->ch = source->ch;
-
- /* for diagnostics */
- sys->clip_start = sys->cur;
- sys->clip_end = sys->end;
- sys->buffer_length = source->segments[ (source->numsegments-1)*2 + 1 ];
-
- sfx_push( sys );
- }
+ audio_require_lock();
+ return sys->info.flags;
}
-/* Pick a random sound from the buffer and play it into system */
-static void sfx_set_playrnd( sfx_set *source, sfx_system *sys,
- int min_id, int max_id )
+
+/*
+ * Debugging
+ */
+
+static void audio_debug_ui(void)
{
- if( !source->numsegments )
- return;
+ if( !vg_audio.debug_ui )
+ return;
- if( max_id > source->numsegments )
+ /* Get data */
+ struct sound_info
{
- vg_error( "Max ID out of range for playrnd\n" );
- return;
+ const char *name;
+ u32 cursor, flags, length;
+ float vol;
}
+ infos[ SFX_MAX_SYSTEMS ];
+ int num_systems = 0;
- int pick = (rand() % (max_id-min_id)) + min_id;
+ char perf[128];
- sfx_set_play( source, sys, pick );
-}
+ audio_lock();
+
+ for( int i=0; i<SFX_MAX_SYSTEMS; i ++ )
+ {
+ struct active_audio_player *aap = &vg_audio.active_players[i];
-static void sfx_system_fadeout( sfx_system *sys, u32 length_samples )
-{
- if( sfx_begin_edit( sys ) )
+ if( !aap->active )
+ continue;
+
+ audio_entity *ent = &aap->ent;
+ struct sound_info *snd = &infos[ num_systems ++ ];
+
+ snd->name = ent->name;
+ snd->cursor = ent->cur;
+ snd->flags = ent->info.flags;
+ snd->length = ent->length;
+ snd->vol = ent->info.vol*100.0f;
+ }
+ strcpy( perf, vg_audio.performance_info );
+ audio_unlock();
+
+ /* Draw UI */
+ ui_global_ctx.cursor[0] = 10;
+ ui_global_ctx.cursor[1] = 10;
+ ui_global_ctx.cursor[2] = 150;
+ ui_global_ctx.cursor[3] = 12;
+ ui_text( &ui_global_ctx, ui_global_ctx.cursor, perf, 1, 0 );
+
+ float usage = (float)vg_audio.mem_current / (1024.0f*1024.0f),
+ total = (float)vg_audio.mem_total / (1024.0f*1024.0f),
+ percent = (usage/total) * 100.0f;
+ snprintf( perf, 127, "Mem: %.1f/%.1fmb (%.1f%%)\n", usage, total, percent );
+
+ ui_global_ctx.cursor[1] += 20;
+ ui_text( &ui_global_ctx, ui_global_ctx.cursor, perf, 1, 0 );
+
+ ui_global_ctx.cursor[1] += 20;
+
+ /* Draw audio stack */
+ for( int i=0; i<num_systems; i ++ )
{
- sys->fadeout_current = length_samples;
- sys->fadeout = length_samples;
+ struct sound_info *inf = &infos[i];
+
+ ui_global_ctx.cursor[2] = 150;
+ ui_global_ctx.cursor[3] = 12;
- sfx_end_edit( sys );
+ u32 alpha = 0xa0000000;
+
+ ui_new_node( &ui_global_ctx );
+ {
+ ui_fill_rect( &ui_global_ctx, ui_global_ctx.cursor, 0x00333333|alpha );
+
+ ui_px baseline = ui_global_ctx.cursor[0],
+ w = 150,
+ c = baseline + ((float)inf->cursor / (float)inf->length) * w;
+
+ /* cursor */
+ ui_global_ctx.cursor[2] = 2;
+ ui_global_ctx.cursor[0] = c;
+ ui_fill_rect( &ui_global_ctx, ui_global_ctx.cursor, 0xffffffff );
+
+ ui_global_ctx.cursor[0] = baseline + 2;
+ ui_global_ctx.cursor[1] += 2;
+ snprintf( perf, 127, "%s %.1f%%", infos[i].name, infos[i].vol );
+ ui_text( &ui_global_ctx, ui_global_ctx.cursor, perf, 1, 0 );
+ }
+
+ ui_end_down( &ui_global_ctx );
+ ui_global_ctx.cursor[1] += 1;
}
}
-static void sfx_set_free( sfx_set *set )
-{
- free( set->main );
-}
+#endif /* VG_AUDIO_H */
projects / vg.git / commitdiff