#define AUDIO_LFOS 8
#define AUDIO_FILTERS 16
#define AUDIO_FLAG_LOOP 0x1
+#define AUDIO_FLAG_NO_DOPPLER 0x2
#define AUDIO_FLAG_SPACIAL_3D 0x4
#define AUDIO_FLAG_AUTO_START 0x8
typedef struct audio_channel audio_channel;
typedef struct audio_lfo audio_lfo;
-struct audio_clip
-{
+struct audio_clip{
const char *path;
u32 flags;
-
u32 size;
void *data;
};
-static struct vg_audio_system
-{
+static struct vg_audio_system{
SDL_AudioDeviceID sdl_output_device;
void *audio_pool,
SDL_mutex *mux_checker,
*mux_sync;
- struct audio_lfo
- {
+ struct audio_lfo{
u32 time, time_startframe;
float sqrt_polynomial_coefficient;
- struct
- {
- enum lfo_wave_type
- {
+ struct{
+ enum lfo_wave_type{
k_lfo_triangle,
k_lfo_square,
k_lfo_saw,
}
oscillators[ AUDIO_LFOS ];
- struct audio_channel
- {
+ struct audio_channel{
int allocated;
+ u32 group;
+
char name[32]; /* only editable while allocated == 0 */
audio_clip *source; /* ... */
u32 flags; /* ... */
u32 volume_movement,
pan_movement;
- union
- {
+ union{
struct synth_bird *bird_handle;
stb_vorbis *vorbis_handle;
};
stb_vorbis_alloc vorbis_alloc;
- enum channel_activity
- {
+ enum channel_activity{
k_channel_activity_reset, /* will advance if allocated==1, to wake */
k_channel_activity_wake, /* will advance to either of next two */
k_channel_activity_alive,
* the edit mask tells which to copy into internal _, or to discard
* ----------------------------------------------------------------------
*/
- struct channel_state
- {
+ struct channel_state{
int relinquished;
float volume, /* current volume */
vg_audio.sdl_output_device =
SDL_OpenAudioDevice( NULL, 0, &spec_desired, &spec_got,0 );
- if( vg_audio.sdl_output_device )
- {
+ if( vg_audio.sdl_output_device ){
SDL_PauseAudioDevice( vg_audio.sdl_output_device, 0 );
}
- else
- {
+ else{
vg_fatal_exit_loop(
"SDL_OpenAudioDevice failed. Your default audio device must support:\n"
" Frequency: 44100 hz\n"
#define AUDIO_EDIT_OWNERSHIP 0x40
#define AUDIO_EDIT_SAMPLING_RATE 0x80
-static audio_channel *audio_request_channel( audio_clip *clip, u32 flags )
+static void audio_channel_init( audio_channel *ch, audio_clip *clip, u32 flags )
{
- for( int i=0; i<AUDIO_CHANNELS; i++ )
- {
- audio_channel *ch = &vg_audio.channels[i];
+ ch->group = 0;
+ ch->source = clip;
+ ch->flags = flags;
+ ch->colour = 0x00333333;
- if( !ch->allocated )
- {
- 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
+ strncpy( ch->name, clip->path, 31 );
+
+ ch->allocated = 1;
+
+ ch->editable_state.relinquished = 0;
+ ch->editable_state.volume = 1.0f;
+ ch->editable_state.volume_target = 1.0f;
+ ch->editable_state.pan = 0.0f;
+ ch->editable_state.pan_target = 0.0f;
+ ch->editable_state.volume_rate = 0;
+ ch->editable_state.pan_rate = 0;
+ v4_copy((v4f){0.0f,0.0f,0.0f,1.0f},ch->editable_state.spacial_falloff);
+ ch->editable_state.lfo = NULL;
+ ch->editable_state.lfo_amount = 0.0f;
+ ch->editable_state.sampling_rate = 1.0f;
+ ch->editble_state_write_mask = 0x00;
+}
- if( (ch->source->flags & AUDIO_FLAG_FORMAT) == k_audio_format_bird )
- strcpy( ch->name, "[array]" );
- else
- strncpy( ch->name, clip->path, 31 );
-
- ch->allocated = 1;
-
- ch->editable_state.relinquished = 0;
- ch->editable_state.volume = 1.0f;
- ch->editable_state.volume_target = 1.0f;
- ch->editable_state.pan = 0.0f;
- ch->editable_state.pan_target = 0.0f;
- ch->editable_state.volume_rate = 0;
- ch->editable_state.pan_rate = 0;
- v4_copy((v4f){0.0f,0.0f,0.0f,1.0f},ch->editable_state.spacial_falloff);
- ch->editable_state.lfo = NULL;
- ch->editable_state.lfo_amount = 0.0f;
- ch->editable_state.sampling_rate = 1.0f;
- ch->editble_state_write_mask = 0x00;
+static audio_channel *audio_get_first_idle_channel(void)
+{
+ 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( u32 group, u32 max_count )
+{
+ u32 count = 0;
+ audio_channel *dest;
+
+ 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( u32 group )
+{
+ 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 )
{
if( ch->readable_activity == k_channel_activity_end )
static void audio_channel_edit_volume( audio_channel *ch,
float new_volume, int instant )
{
- if( instant )
- {
+ if( instant ){
ch->editable_state.volume = new_volume;
ch->editble_state_write_mask |= AUDIO_EDIT_VOLUME;
}
- else
- {
+ else{
audio_channel_slope_volume( ch, 0.05f, new_volume );
}
}
if( ch )
ch = audio_channel_fadeout( ch, length );
- audio_channel *replacement = audio_request_channel( new_clip, flags );
+ audio_channel *replacement = audio_get_first_idle_channel();
- if( replacement )
+ if( replacement ){
+ audio_channel_init( replacement, new_clip, flags );
audio_channel_fadein( replacement, length );
+ }
return replacement;
}
static void audio_channel_set_spacial( audio_channel *ch, v3f co, float range )
{
- if( ch->flags & AUDIO_FLAG_SPACIAL_3D )
- {
+ if( ch->flags & AUDIO_FLAG_SPACIAL_3D ){
v3_copy( co, ch->editable_state.spacial_falloff );
if( range == 0.0f )
ch->editble_state_write_mask |= AUDIO_EDIT_SPACIAL;
}
- else
- {
+ 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_request_channel( clip, AUDIO_FLAG_SPACIAL_3D );
+ audio_channel *ch = audio_get_first_idle_channel();
- if( ch )
- {
+ if( ch ){
+ audio_channel_init( ch, clip, AUDIO_FLAG_SPACIAL_3D );
audio_channel_set_spacial( ch, position, range );
audio_channel_edit_volume( ch, volume, 1 );
ch = audio_relinquish_channel( ch );
static int audio_oneshot( audio_clip *clip, float volume, float pan )
{
- audio_channel *ch = audio_request_channel( clip, 0x00 );
+ audio_channel *ch = audio_get_first_idle_channel();
- if( ch )
- {
+ if( ch ){
+ audio_channel_init( ch, clip, 0x00 );
audio_channel_edit_volume( ch, volume, 1 );
ch = audio_relinquish_channel( ch );
}
-
/*
* Committers
* -----------------------------------------------------------------------------
{
u32 format = ch->source->flags & AUDIO_FLAG_FORMAT;
- if( format == k_audio_format_vorbis )
- {
+ if( format == k_audio_format_vorbis ){
/* Setup vorbis decoder */
u32 index = ch - vg_audio.channels;
ch->source->data,
ch->source->size, &err, &alloc );
- if( !decoder )
- {
+ if( !decoder ){
vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
ch->source->path, err );
return 0;
}
- else
- {
+ else{
ch->source_length = stb_vorbis_stream_length_in_samples( decoder );
ch->vorbis_handle = decoder;
}
}
- else if( format == k_audio_format_bird )
- {
+ else if( format == k_audio_format_bird ){
u32 index = ch - vg_audio.channels;
u8 *buf = (u8*)vg_audio.decode_buffer;
ch->bird_handle = loc;
ch->source_length = synth_bird_get_length_in_samples( loc );
}
- else if( format == k_audio_format_stereo )
- {
+ else if( format == k_audio_format_stereo ){
ch->source_length = ch->source->size / 2;
}
- else
- {
+ else{
ch->source_length = ch->source->size;
}
VG_STATIC void audio_decode_uncompressed_mono( i16 *src, u32 count, float *dst )
{
- for( u32 i=0; i<count; i++ )
- {
+ 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);
}
int n = 0,
c = VG_MIN( 1, f->channels - 1 );
- while( n < len )
- {
+ while( n < len ) {
int k = f->channel_buffer_end - f->channel_buffer_start;
if( n+k >= len )
k = len - n;
- for( int j=0; j < k; ++j )
- {
+ for( int j=0; j < k; ++j ) {
*buffer++ = f->channel_buffers[ 0 ][f->channel_buffer_start+j];
*buffer++ = f->channel_buffers[ c ][f->channel_buffer_start+j];
}
int n = 0,
c = VG_MIN( 1, f->channels - 1 );
- while( n < len )
- {
+ while( n < len ) {
int k = f->channel_buffer_end - f->channel_buffer_start;
if( n+k >= len )
k = len - n;
- for( int j=0; j < k; ++j )
- {
+ for( int j=0; j < k; ++j ) {
float sl = f->channel_buffers[ 0 ][f->channel_buffer_start+j],
sr = f->channel_buffers[ c ][f->channel_buffer_start+j];
float t = lfo->time;
t /= (float)lfo->_.period;
- if( lfo->_.wave_type == k_lfo_polynomial_bipolar )
- {
+ if( lfo->_.wave_type == k_lfo_polynomial_bipolar ){
/*
* #
* # #
) * (1.0f-fabsf(t));
}
- else
- {
+ else{
return 0.0f;
}
}
u32 format = ch->source->flags & AUDIO_FLAG_FORMAT;
- while( remaining )
- {
+ while( remaining ){
u32 samples_this_run = VG_MIN(remaining, ch->source_length - ch->cursor);
remaining -= samples_this_run;
float *dst = &buf[ buffer_pos * 2 ];
- if( format == k_audio_format_stereo )
- {
- for( int i=0;i<samples_this_run; i++ )
- {
+ 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 )
- {
+ else if( format == k_audio_format_vorbis ){
int read_samples = stb_vorbis_get_samples_float_interleaved_stereo(
ch->vorbis_handle,
dst,
samples_this_run );
- if( read_samples != samples_this_run )
- {
+ if( read_samples != samples_this_run ){
vg_warn( "Invalid samples read (%s)\n", ch->source->path );
- for( int i=0; i<samples_this_run; i++ )
- {
+ 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 )
- {
+ else if( format == k_audio_format_bird ){
synth_bird_generate_samples( ch->bird_handle, dst, samples_this_run );
}
- else
- {
+ else{
i16 *src_buffer = ch->source->data,
*src = &src_buffer[ch->cursor];
ch->cursor += samples_this_run;
buffer_pos += samples_this_run;
- if( (ch->flags & AUDIO_FLAG_LOOP) && remaining )
- {
+ if( (ch->flags & AUDIO_FLAG_LOOP) && remaining ){
if( format == k_audio_format_vorbis )
stb_vorbis_seek_start( ch->vorbis_handle );
else if( format == k_audio_format_bird )
break;
}
- while( remaining )
- {
+ while( remaining ){
buf[ buffer_pos*2 + 0 ] = 0.0f;
buf[ buffer_pos*2 + 1 ] = 0.0f;
buffer_pos ++;
float frame_samplerate = ch->_.sampling_rate;
- if( ch->flags & AUDIO_FLAG_SPACIAL_3D )
- {
- if( !vg_validf(vg_audio.listener_pos[0]) ||
- !vg_validf(vg_audio.listener_pos[1]) ||
- !vg_validf(vg_audio.listener_pos[2]) ||
- !vg_validf(vg_audio.listener_velocity[0]) ||
- !vg_validf(vg_audio.listener_velocity[1]) ||
- !vg_validf(vg_audio.listener_velocity[2]) ||
- !vg_validf(ch->_.spacial_falloff[0]) ||
- !vg_validf(ch->_.spacial_falloff[1]) ||
- !vg_validf(ch->_.spacial_falloff[2]) )
- {
- vg_error( "NaN listener/world position (%s)\n", ch->name );
+ if( ch->flags & AUDIO_FLAG_SPACIAL_3D ){
+ v3f delta;
+ v3_sub( ch->_.spacial_falloff, vg_audio.listener_pos, delta );
+
+ float dist = v3_length( delta ),
+ vol = vg_maxf( 0.0f, 1.0f - ch->_.spacial_falloff[3]*dist );
- framevol_l = 0.0f;
- framevol_r = 0.0f;
+ if( dist <= 0.01f ){
+
}
else{
- v3f delta;
- v3_sub( ch->_.spacial_falloff, vg_audio.listener_pos, delta );
-
- float dist = v3_length( delta ),
- vol = vg_maxf( 0.0f, 1.0f - ch->_.spacial_falloff[3]*dist );
-
v3_muls( delta, 1.0f/dist, delta );
- float pan = v3_dot( vg_audio.listener_ears, delta );
+ float pan = v3_dot( vg_audio.listener_ears, delta );
vol = powf( vol, 5.0f );
framevol_l *= (vol * 0.5f) * (1.0f - pan);
framevol_r *= (vol * 0.5f) * (1.0f + pan);
- v3_muls( delta, 1.0f/dist, delta );
-
- const float vs = 100.0f;
+ const float vs = 323.0f;
float doppler = (vs+v3_dot(delta,vg_audio.listener_velocity))/vs;
doppler = vg_clampf( doppler, 0.6f, 1.4f );
if( fabsf(doppler-1.0f) > 0.01f )
frame_samplerate *= doppler;
}
+
+ if( !vg_validf( framevol_l ) ) vg_fatal_exit_loop( "NaN left channel" );
+ if( !vg_validf( framevol_r ) ) vg_fatal_exit_loop( "NaN right channel" );
+ if( !vg_validf( frame_samplerate ) )
+ vg_fatal_exit_loop( "NaN sample rate" );
}
u32 buffer_length = AUDIO_MIX_FRAME_SIZE;
- if( frame_samplerate != 1.0f )
- {
+ if( frame_samplerate != 1.0f ){
float l = ceilf( (float)(AUDIO_MIX_FRAME_SIZE) * frame_samplerate );
buffer_length = l+1;
}
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++ )
- {
+ for( u32 j=0; j<AUDIO_MIX_FRAME_SIZE; j++ ){
/*
* there is some REALLY weird behaviour with minss,
* i cannot begin to guess what the cause is, but the bahaviour when
sample_l,
sample_r;
- if( frame_samplerate != 1.0f )
- {
+ if( frame_samplerate != 1.0f ){
/* absolutely garbage resampling, but it will do
*/
sample_l = pcf[ i0*2+0 ]*(1.0f-t) + pcf[ i1*2+0 ]*t;
sample_r = pcf[ i0*2+1 ]*(1.0f-t) + pcf[ i1*2+1 ]*t;
}
- else
- {
+ else{
sample_l = pcf[ j*2+0 ];
sample_r = pcf[ j*2+1 ];
}
* Copy data and move edit flags to commit flags
* ------------------------------------------------------------- */
audio_lock();
- for( int i=0; i<AUDIO_CHANNELS; i++ )
- {
+ 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->activity == k_channel_activity_alive ){
if( (ch->cursor >= ch->source_length) &&
!(ch->flags & AUDIO_FLAG_LOOP) )
{
}
/* process relinquishments */
- if( (ch->activity != k_channel_activity_reset) && ch->_.relinquished )
- {
+ if( (ch->activity != k_channel_activity_reset) && ch->_.relinquished ){
if( (ch->activity == k_channel_activity_end)
|| (ch->_.volume == 0.0f)
|| (ch->activity == k_channel_activity_error) )
}
/* process new channels */
- if( ch->activity == k_channel_activity_reset )
- {
+ if( ch->activity == k_channel_activity_reset ){
ch->_ = ch->editable_state;
ch->cursor = 0;
ch->source_length = 0;
ch->editable_state.relinquished = ch->_.relinquished;
- if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME )
- {
+ if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME ){
ch->_.volume = ch->editable_state.volume;
ch->_.volume_target = ch->editable_state.volume;
}
- else
- {
+ else{
ch->editable_state.volume = ch->_.volume;
}
- if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME_SLOPE )
- {
+ if( ch->editble_state_write_mask & AUDIO_EDIT_VOLUME_SLOPE ){
ch->volume_movement_start = ch->_.volume;
ch->volume_movement = 0;
ch->_.volume_target = ch->editable_state.volume_target;
ch->_.volume_rate = ch->editable_state.volume_rate;
}
- else
- {
+ else{
ch->editable_state.volume_target = ch->_.volume_target;
ch->editable_state.volume_rate = ch->_.volume_rate;
}
ch->editable_state.sampling_rate = ch->_.sampling_rate;
- if( ch->editble_state_write_mask & AUDIO_EDIT_LFO_ATTACHMENT )
- {
+ if( ch->editble_state_write_mask & AUDIO_EDIT_LFO_ATTACHMENT ){
ch->_.lfo = ch->editable_state.lfo;
ch->_.lfo_amount = ch->editable_state.lfo_amount;
}
- else
- {
+ else{
ch->editable_state.lfo = ch->_.lfo;
ch->editable_state.lfo_amount = ch->_.lfo_amount;
}
ch->editble_state_write_mask = 0x00;
}
- for( int i=0; i<AUDIO_LFOS; i++ )
- {
+ 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 )
- {
+ if( lfo->editble_state_write_mask & AUDIO_EDIT_LFO_WAVE ){
lfo->_.wave_type = lfo->editable_state.wave_type;
- if( lfo->_.wave_type == k_lfo_polynomial_bipolar )
- {
+ if( lfo->_.wave_type == k_lfo_polynomial_bipolar ){
lfo->_.polynomial_coefficient =
lfo->editable_state.polynomial_coefficient;
lfo->sqrt_polynomial_coefficient =
}
}
- if( lfo->editble_state_write_mask & AUDIO_EDIT_LFO_PERIOD )
- {
- if( lfo->_.period )
- {
+ if( lfo->editble_state_write_mask & AUDIO_EDIT_LFO_PERIOD ){
+ if( lfo->_.period ){
float t = lfo->time;
t/= (float)lfo->_.period;
lfo->_.period = lfo->editable_state.period;
lfo->time = lfo->_.period * t;
}
- else
- {
+ else{
lfo->time = 0;
lfo->_.period = lfo->editable_state.period;
}
/*
* Process spawns
* ------------------------------------------------------------- */
- for( int i=0; i<AUDIO_CHANNELS; i++ )
- {
+ for( int i=0; i<AUDIO_CHANNELS; i++ ){
audio_channel *ch = &vg_audio.channels[i];
- if( ch->activity == k_channel_activity_wake )
- {
+ if( ch->activity == k_channel_activity_wake ){
if( audio_channel_load_source( ch ) )
ch->activity = k_channel_activity_alive;
else
for( int i=0; i<frame_count*2; i ++ )
pOut32F[i] = 0.0f;
- for( int i=0; i<AUDIO_LFOS; i++ )
- {
+ 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 ++ )
- {
+ for( int i=0; i<AUDIO_CHANNELS; i ++ ){
audio_channel *ch = &vg_audio.channels[i];
- if( ch->activity == k_channel_activity_alive )
- {
+ if( ch->activity == k_channel_activity_alive ){
if( ch->_.lfo )
ch->_.lfo->time = ch->_.lfo->time_startframe;
u32 remaining = frame_count,
subpos = 0;
- while( remaining )
- {
+ while( remaining ){
audio_channel_mix( ch, pOut32F+subpos );
remaining -= AUDIO_MIX_FRAME_SIZE;
subpos += AUDIO_MIX_FRAME_SIZE*2;
audio_lock();
- for( int i=0; i<AUDIO_CHANNELS; i ++ )
- {
+ for( int i=0; i<AUDIO_CHANNELS; i ++ ){
audio_channel *ch = &vg_audio.channels[i];
ch->readable_activity = ch->activity;
}
vg_audio.samples_last = frame_count;
- if( vg_audio.debug_ui )
- {
+ if( vg_audio.debug_ui ){
vg_dsp_update_texture();
}
/* TODO: This contains audio_lock() and unlock, but i don't know why
* can probably remove them. Low priority to check this */
- if( format == k_audio_format_vorbis )
- {
+ /* TODO: packed files for vorbis etc, should take from data if its not not
+ * NULL when we get the clip
+ */
+
+ if( format == k_audio_format_vorbis ){
+ if( !clip->path ){
+ vg_fatal_exit_loop( "No path specified, embeded vorbis unsupported" );
+ }
+
audio_lock();
clip->data = vg_file_read( lin_alloc, clip->path, &clip->size );
audio_unlock();
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 )
- {
+ else if( format == k_audio_format_stereo ){
vg_fatal_exit_loop( "Unsupported format (Stereo uncompressed)" );
}
- else if( format == k_audio_format_bird )
- {
- u32 len = strlen( clip->path ),
- size = synth_bird_memory_requirement( len );
+ else if( format == k_audio_format_bird ){
+ if( !clip->data ){
+ vg_fatal_exit_loop( "No data, external birdsynth unsupported" );
+ }
- if( size > AUDIO_DECODE_SIZE )
- vg_fatal_exit_loop( "Bird code too long\n" );
+ u32 total_size = clip->size + sizeof(struct synth_bird);
+ total_size -= sizeof(struct synth_bird_settings);
+ total_size = vg_align8( total_size );
- clip->size = size;
- clip->data = vg_linear_alloc( lin_alloc, size );
+ if( total_size > AUDIO_DECODE_SIZE )
+ vg_fatal_exit_loop( "Bird coding too long\n" );
- synth_bird_load( clip->data, clip->path, len );
+ 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
- {
+ else{
+ if( !clip->path ){
+ vg_fatal_exit_loop( "No path specified, embeded mono unsupported" );
+ }
+
vg_linear_clear( vg_mem.scratch );
u32 fsize;
stb_vorbis *decoder = stb_vorbis_open_memory(
filedata, fsize, &err, &alloc );
- if( !decoder )
- {
+ if( !decoder ){
vg_error( "stb_vorbis_open_memory failed on '%s' (%d)\n",
clip->path, err );
vg_fatal_exit_loop( "Vorbis decode error" );
u32 overlap_length = 0;
/* Draw audio stack */
- for( int i=0; i<AUDIO_CHANNELS; i ++ )
- {
+ for( int i=0; i<AUDIO_CHANNELS; i ++ ){
audio_channel *ch = &vg_audio.channels[i];
vg_uictx.cursor[2] = 400;
ui_new_node();
- if( !ch->allocated )
- {
+ if( !ch->allocated ){
ui_fill_rect( vg_uictx.cursor, 0x50333333 );
ui_end_down();
ui_end_down();
vg_uictx.cursor[1] += 1;
- if( AUDIO_FLAG_SPACIAL_3D )
- {
+ if( AUDIO_FLAG_SPACIAL_3D ){
v4f wpos;
v3_copy( ch->editable_state.spacial_falloff, wpos );
wpos[3] = 1.0f;
m4x4_mulv( mtx_pv, wpos, wpos );
- if( wpos[3] > 0.0f )
- {
+ if( wpos[3] > 0.0f ){
v2_muls( wpos, (1.0f/wpos[3]) * 0.5f, wpos );
v2_add( wpos, (v2f){ 0.5f, 0.5f }, wpos );
wr[2] = 100;
wr[3] = 17;
- for( int j=0; j<12; j++ )
- {
+ for( int j=0; j<12; j++ ){
int collide = 0;
- for( int k=0; k<overlap_length; k++ )
- {
+ 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])) )
projects / vg.git / blobdiff