From 468a2c11d6c83e0e71e6ac87844e52cc4d49ccbc Mon Sep 17 00:00:00 2001 From: hgn Date: Sat, 1 Apr 2023 03:52:14 +0100 Subject: [PATCH] unfinished work --- vg_audio.h | 279 +++++++++++++++++++++++------------------- vg_audio_synth_bird.h | 17 ++- vg_binstr.h | 6 +- 3 files changed, 159 insertions(+), 143 deletions(-) diff --git a/vg_audio.h b/vg_audio.h index bff400f..8f68f82 100644 --- a/vg_audio.h +++ b/vg_audio.h @@ -48,6 +48,7 @@ #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 @@ -88,17 +89,14 @@ typedef struct audio_clip audio_clip; 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, @@ -111,15 +109,12 @@ static struct vg_audio_system 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, @@ -135,9 +130,10 @@ static struct vg_audio_system } 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; /* ... */ @@ -153,16 +149,14 @@ static struct vg_audio_system 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, @@ -177,8 +171,7 @@ static struct vg_audio_system * the edit mask tells which to copy into internal _, or to discard * ---------------------------------------------------------------------- */ - struct channel_state - { + struct channel_state{ int relinquished; float volume, /* current volume */ @@ -318,12 +311,10 @@ VG_STATIC void vg_audio_init(void) 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" @@ -354,37 +345,40 @@ VG_STATIC void vg_audio_free(void) #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; igroup = 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; iallocated ){ return ch; } } @@ -392,6 +386,42 @@ static audio_channel *audio_request_channel( audio_clip *clip, u32 flags ) 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; iallocated ){ + if( ch->group == group ){ + count ++; + } + } + else{ + if( !dest ) + dest = ch; + } + } + + if( dest && (count < max_count) ){ + return dest; + } + + return NULL; +} + +static audio_channel *audio_get_group_first_active_channel( u32 group ) +{ + for( int i=0; iallocated && (ch->group == group) ) + return ch; + } + return NULL; +} + static int audio_channel_finished( audio_channel *ch ) { if( ch->readable_activity == k_channel_activity_end ) @@ -424,13 +454,11 @@ static void audio_channel_set_sampling_rate( audio_channel *ch, float rate ) 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 ); } } @@ -456,10 +484,12 @@ static audio_channel *audio_channel_crossfade( audio_channel *ch, 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; } @@ -493,9 +523,10 @@ static void audio_channel_set_spacial( audio_channel *ch, v3f co, float range ) 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 ){ + 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 ); @@ -508,9 +539,10 @@ static int audio_oneshot_3d( audio_clip *clip, v3f position, 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 ){ + audio_channel_init( ch, clip, 0x00 ); audio_channel_edit_volume( ch, volume, 1 ); ch = audio_relinquish_channel( ch ); @@ -546,8 +578,7 @@ static int audio_channel_load_source( audio_channel *ch ) { 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; @@ -564,20 +595,17 @@ static int audio_channel_load_source( audio_channel *ch ) 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; @@ -589,12 +617,10 @@ static int audio_channel_load_source( audio_channel *ch ) 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; } @@ -603,8 +629,7 @@ static int audio_channel_load_source( audio_channel *ch ) VG_STATIC void audio_decode_uncompressed_mono( i16 *src, u32 count, float *dst ) { - for( u32 i=0; ichannels - 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]; } @@ -818,7 +841,7 @@ static void audio_channel_mix( audio_channel *ch, float *buffer ) framevol_l *= (vol * 0.5f) * (1.0f - pan); framevol_r *= (vol * 0.5f) * (1.0f + pan); - 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 ); @@ -881,8 +904,7 @@ static void audio_channel_mix( audio_channel *ch, float *buffer ) sample_l, sample_r; - if( frame_samplerate != 1.0f ) - { + if( frame_samplerate != 1.0f ){ /* absolutely garbage resampling, but it will do */ @@ -895,8 +917,7 @@ static void audio_channel_mix( audio_channel *ch, float *buffer ) 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 ]; } @@ -1015,8 +1036,7 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count ) 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 = @@ -1047,12 +1067,10 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count ) /* * Process spawns * ------------------------------------------------------------- */ - for( int i=0; iactivity == k_channel_activity_wake ) - { + if( ch->activity == k_channel_activity_wake ){ if( audio_channel_load_source( ch ) ) ch->activity = k_channel_activity_alive; else @@ -1070,26 +1088,22 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count ) for( int i=0; itime_startframe = lfo->time; } - for( int i=0; iactivity == 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; @@ -1106,8 +1120,7 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count ) audio_lock(); - for( int i=0; ireadable_activity = ch->activity; } @@ -1124,8 +1137,7 @@ VG_STATIC void audio_mixer_callback( void *user, u8 *stream, int byte_count ) vg_audio.samples_last = frame_count; - if( vg_audio.debug_ui ) - { + if( vg_audio.debug_ui ){ vg_dsp_update_texture(); } @@ -1143,8 +1155,15 @@ VG_STATIC void audio_clip_load( audio_clip *clip, void *lin_alloc ) /* 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(); @@ -1155,25 +1174,34 @@ VG_STATIC void audio_clip_load( audio_clip *clip, void *lin_alloc ) 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" ); + } + + u32 total_size = clip->size + sizeof(struct synth_bird); + total_size -= sizeof(struct synth_bird_settings); + total_size = vg_align8( total_size ); - if( size > AUDIO_DECODE_SIZE ) - vg_fatal_exit_loop( "Bird code too long\n" ); + if( total_size > AUDIO_DECODE_SIZE ) + vg_fatal_exit_loop( "Bird coding too long\n" ); - clip->size = size; - clip->data = vg_linear_alloc( lin_alloc, size ); + struct synth_bird *bird = vg_linear_alloc( lin_alloc, total_size ); + memcpy( &bird->settings, clip->data, clip->size ); - synth_bird_load( clip->data, clip->path, len ); + 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; @@ -1188,8 +1216,7 @@ VG_STATIC void audio_clip_load( audio_clip *clip, void *lin_alloc ) 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" ); @@ -1285,8 +1312,7 @@ VG_STATIC void audio_debug_ui( m4x4f mtx_pv ) u32 overlap_length = 0; /* Draw audio stack */ - for( int i=0; iallocated ) - { + if( !ch->allocated ){ ui_fill_rect( vg_uictx.cursor, 0x50333333 ); ui_end_down(); @@ -1353,16 +1378,14 @@ VG_STATIC void audio_debug_ui( m4x4f mtx_pv ) 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 ); @@ -1372,11 +1395,9 @@ VG_STATIC void audio_debug_ui( m4x4f mtx_pv ) 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= wk[0])) && ((wr[1] <= wk[1]+wk[3]) && (wr[1]+wr[3] >= wk[1])) ) diff --git a/vg_audio_synth_bird.h b/vg_audio_synth_bird.h index 62c30e6..a439354 100644 --- a/vg_audio_synth_bird.h +++ b/vg_audio_synth_bird.h @@ -8,8 +8,7 @@ #define BIRD_SAMPLE_RATE 44100 #endif -struct synth_bird_signature -{ +struct synth_bird_signature{ float length, pause, /* timings in seconds */ x0,x1,x2,x3, /* polynomial coefficients for the fundemental */ v0,v1,v2,v3; /* volume of each oscillator */ @@ -18,10 +17,8 @@ struct synth_bird_signature float fm; /* LFO modulation depth (+/- hz) */ }; -struct synth_bird -{ - struct - { +struct synth_bird{ + struct{ int osc_main[4], osc_lfo; float volume[4]; @@ -39,15 +36,13 @@ struct synth_bird } rt; - struct synth_bird_settings - { + struct synth_bird_settings{ int tones[4][2]; /* fraction of the fundemental tone for each oscillator */ float adsr_rise, /* rise/fall in seconds */ adsr_fall; - enum bird_lfo_wave - { + enum bird_lfo_wave{ k_bird_lfo_sine_approx, k_bird_lfo_bipolar_poly } @@ -260,6 +255,7 @@ static void synth_bird_save( struct synth_bird *bird, void *txt ) vg_bin_str( src, txt, synth_bird_save_size( bird ) ); } +#if 0 static void synth_bird_load( struct synth_bird *bird, const char *txt, u32 length ) { @@ -273,6 +269,7 @@ static u32 synth_bird_memory_requirement( u32 string_length ) return (string_length/2) + sizeof(struct synth_bird) - sizeof(struct synth_bird_settings); } +#endif #ifdef SYNTH_BIRD_STDLIB #include "stdlib.h" diff --git a/vg_binstr.h b/vg_binstr.h index da41ce0..d6bfb83 100644 --- a/vg_binstr.h +++ b/vg_binstr.h @@ -12,8 +12,7 @@ static void vg_str_bin( const void *txt, void *bin, int size ) const u8 *src = txt; u8 *dst = bin; - for( u32 i=0; i>4u) & 0xf); } -- 2.25.1