--- /dev/null
+#include "vg_audio.h"
+#include "vg_audio_dsp.h"
+#include "vg_platform.h"
+#include "vg_io.h"
+#include "vg_m.h"
+#include "vg_console.h"
+#include "vg_profiler.h"
+#include "vg_audio_synth_bird.h"
+#include "vg_vorbis.h"
+#include <string.h>
+
+struct vg_audio_system vg_audio =
+{
+ .external_global_volume = 1.0f,
+ .dsp_enabled = 1
+};
+
+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();
+}
+
+void audio_lock(void)
+{
+ SDL_AtomicLock( &vg_audio.sl_sync );
+ audio_lock_checker_store(1);
+}
+
+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 );
+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" );
+ }
+}
+
+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 );
+}
+
+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();
+}
+
+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
+
+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;
+}
+
+void audio_channel_group( audio_channel *ch, u16 group )
+{
+ audio_require_lock();
+ ch->group = group;
+ ch->colour = (((u32)group * 29986577) & 0x00ffffff) | 0xff000000;
+}
+
+void audio_channel_world( audio_channel *ch, u8 world_id )
+{
+ audio_require_lock();
+ ch->world_id = world_id;
+}
+
+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;
+}
+
+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;
+}
+
+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;
+}
+
+int audio_channel_finished( audio_channel *ch )
+{
+ audio_require_lock();
+ if( ch->readable_activity == k_channel_activity_end )
+ return 1;
+ else
+ return 0;
+}
+
+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;
+}
+
+void audio_channel_slope_volume( audio_channel *ch, f32 length, f32 new_vol )
+{
+ audio_require_lock();
+ ch->editable_state.volume_target = new_vol;
+ ch->editable_state.volume_rate = length * 44100.0f;
+ ch->editble_state_write_mask |= AUDIO_EDIT_VOLUME_SLOPE;
+}
+
+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;
+}
+
+void audio_channel_edit_volume( audio_channel *ch, f32 new_vol, int instant )
+{
+ audio_require_lock();
+ if( instant ){
+ ch->editable_state.volume = new_vol;
+ ch->editble_state_write_mask |= AUDIO_EDIT_VOLUME;
+ }
+ else{
+ audio_channel_slope_volume( ch, 0.05f, new_vol );
+ }
+}
+
+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 );
+}
+
+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 );
+}
+
+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;
+}
+
+void audio_channel_sidechain_lfo( audio_channel *ch, int lfo_id, f32 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;
+}
+
+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 );
+ }
+}
+
+int audio_oneshot_3d( audio_clip *clip, v3f position, f32 range, f32 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;
+}
+
+int audio_oneshot( audio_clip *clip, f32 volume, f32 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;
+}
+
+void audio_set_lfo_wave( int id, enum lfo_wave_type type, f32 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;
+}
+
+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
+ * -----------------------------------------------------------------------------
+ */
+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->handle.vorbis = 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->handle.bird = 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);
+ }
+}
+
+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->handle.vorbis,
+ 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->handle.bird, 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->handle.vorbis );
+ else if( format == k_audio_format_bird )
+ synth_bird_reset( ch->handle.bird );
+
+ 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();
+}
+
+void audio_clip_load( audio_clip *clip, void *lin_alloc )
+{
+ if( lin_alloc == NULL )
+ lin_alloc = vg_audio.audio_pool;
+
+ if( vg_audio.always_keep_compressed )
+ {
+ if( (clip->flags & AUDIO_FLAG_FORMAT) != k_audio_format_bird ){
+ clip->flags &= ~AUDIO_FLAG_FORMAT;
+ clip->flags |= k_audio_format_vorbis;
+ }
+ }
+
+ /* 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
+ }
+}
+
+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
+ */
+
+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();
+}
projects / vg.git / blobdiff