sampler_voice.c

   1 /*
   2 Calf Box, an open source musical instrument.
   3 Copyright (C) 2010-2013 Krzysztof Foltman
   4
   5 This program is free software: you can redistribute it and/or modify
   6 it under the terms of the GNU General Public License as published by
   7 the Free Software Foundation, either version 3 of the License, or
   8 (at your option) any later version.
   9
  10 This program is distributed in the hope that it will be useful,
  11 but WITHOUT ANY WARRANTY; without even the implied warranty of
  12 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  13 GNU General Public License for more details.
  14
  15 You should have received a copy of the GNU General Public License
  16 along with this program.  If not, see <http://www.gnu.org/licenses/>.
  17 */
  18
  19 #include "config-api.h"
  20 #include "dspmath.h"
  21 #include "errors.h"
  22 #include "midi.h"
  23 #include "module.h"
  24 #include "rt.h"
  25 #include "sampler.h"
  26 #include "sampler_impl.h"
  27 #include "sfzloader.h"
  28 #include "stm.h"
  29 #include <assert.h>
  30 #include <errno.h>
  31 #include <glib.h>
  32 #include <math.h>
  33 #include <memory.h>
  34 #include <sndfile.h>
  35 #include <stdio.h>
  36 #include <stdlib.h>
  37
  38
  39 static void lfo_init(struct sampler_lfo *lfo, struct sampler_lfo_params *lfop, int srate, double srate_inv)
  40 {
  41     lfo->phase = 0;
  42     lfo->age = 0;
  43     lfo->delta = (uint32_t)(lfop->freq * 65536.0 * 65536.0 * CBOX_BLOCK_SIZE * srate_inv);
  44     lfo->delay = (uint32_t)(lfop->delay * srate);
  45     lfo->fade = (uint32_t)(lfop->fade * srate);
  46 }
  47
  48 static inline float lfo_run(struct sampler_lfo *lfo)
  49 {
  50     if (lfo->age < lfo->delay)
  51     {
  52         lfo->age += CBOX_BLOCK_SIZE;
  53         return 0.f;
  54     }
  55
  56     const int FRAC_BITS = 32 - 11;
  57     lfo->phase += lfo->delta;
  58     uint32_t iphase = lfo->phase >> FRAC_BITS;
  59     float frac = (lfo->phase & ((1 << FRAC_BITS) - 1)) * (1.0 / (1 << FRAC_BITS));
  60
  61     float v = sampler_sine_wave[iphase] + (sampler_sine_wave[iphase + 1] - sampler_sine_wave[iphase]) * frac;
  62     if (lfo->fade && lfo->age < lfo->delay + lfo->fade)
  63     {
  64         v *= (lfo->age - lfo->delay) * 1.0 / lfo->fade;
  65         lfo->age += CBOX_BLOCK_SIZE;
  66     }
  67
  68     return v;
  69 }
  70
  71 static gboolean is_4pole(struct sampler_layer_data *v)
  72 {
  73     if (v->cutoff == -1)
  74         return FALSE;
  75     return v->fil_type == sft_lp24 || v->fil_type == sft_hp24 || v->fil_type == sft_bp12;
  76 }
  77
  78 static gboolean is_tail_finished(struct sampler_voice *v)
  79 {
  80     if (v->layer->cutoff == -1)
  81         return TRUE;
  82     double eps = 1.0 / 65536.0;
  83     if (cbox_biquadf_is_audible(&v->filter_left, eps))
  84         return FALSE;
  85     if (cbox_biquadf_is_audible(&v->filter_right, eps))
  86         return FALSE;
  87     if (is_4pole(v->layer))
  88     {
  89         if (cbox_biquadf_is_audible(&v->filter_left2, eps))
  90             return FALSE;
  91         if (cbox_biquadf_is_audible(&v->filter_right2, eps))
  92             return FALSE;
  93     }
  94
  95     return TRUE;
  96 }
  97
  98 static inline void mix_block_into_with_gain(cbox_sample_t **outputs, int oofs, float *src_left, float *src_right, float gain)
  99 {
 100     cbox_sample_t *dst_left = outputs[oofs];
 101     cbox_sample_t *dst_right = outputs[oofs + 1];
 102     for (size_t i = 0; i < CBOX_BLOCK_SIZE; i++)
 103     {
 104         dst_left[i] += gain * src_left[i];
 105         dst_right[i] += gain * src_right[i];
 106     }
 107 }
 108
 109 ////////////////////////////////////////////////////////////////////////////////
 110
 111 void sampler_voice_activate(struct sampler_voice *v, enum sampler_player_type mode)
 112 {
 113     assert(v->gen.mode == spt_inactive);
 114     sampler_voice_unlink(&v->program->module->voices_free, v);
 115     assert(mode != spt_inactive);
 116     assert(v->channel);
 117     v->gen.mode = mode;
 118     sampler_voice_link(&v->channel->voices_running, v);
 119 }
 120
 121 void sampler_voice_start(struct sampler_voice *v, struct sampler_channel *c, struct sampler_layer_data *l, int note, int vel, int *exgroups, int *pexgroupcount)
 122 {
 123     struct sampler_module *m = c->module;
 124     sampler_gen_reset(&v->gen);
 125
 126     v->age = 0;
 127     if (l->trigger == stm_release)
 128     {
 129         // time since last 'note on' for that note
 130         v->age = m->current_time - c->prev_note_start_time[note];
 131         double age = v->age *  m->module.srate_inv;
 132         // if attenuation is more than 84dB, ignore the release trigger
 133         if (age * l->rt_decay > 84)
 134             return;
 135     }
 136     uint32_t end = l->eff_waveform->info.frames;
 137     if (l->end != 0)
 138         end = (l->end == -1) ? 0 : l->end;
 139     v->last_waveform = l->eff_waveform;
 140     v->gen.cur_sample_end = end;
 141     if (end > l->eff_waveform->info.frames)
 142         end = l->eff_waveform->info.frames;
 143
 144     assert(!v->current_pipe);
 145     if (end > l->eff_waveform->preloaded_frames)
 146     {
 147         if (l->loop_mode == slm_loop_continuous && l->loop_end < l->eff_waveform->preloaded_frames)
 148         {
 149             // Everything fits in prefetch, because loop ends in prefetch and post-loop part is not being played
 150         }
 151         else
 152         {
 153             uint32_t loop_start = -1, loop_end = end;
 154             // If in loop mode, set the loop over the looped part... unless we're doing sustain-only loop on prefetch area only. Then
 155             // streaming will only cover the release part, and it shouldn't be looped.
 156             if (l->loop_mode == slm_loop_continuous || (l->loop_mode == slm_loop_sustain && l->loop_end >= l->eff_waveform->preloaded_frames))
 157             {
 158                 loop_start = l->loop_start;
 159                 loop_end = l->loop_end;
 160             }
 161             // Those are initial values only, they will be adjusted in process function
 162             v->current_pipe = cbox_prefetch_stack_pop(m->pipe_stack, l->eff_waveform, loop_start, loop_end, l->count);
 163             if (!v->current_pipe)
 164                 g_warning("Prefetch pipe pool exhausted, no streaming playback will be possible");
 165         }
 166     }
 167
 168     v->output_pair_no = l->output % m->output_pairs;
 169     v->serial_no = m->serial_no;
 170
 171     uint32_t pos = l->offset;
 172     pos = l->offset;
 173     if (l->offset_random)
 174         pos += ((uint32_t)(rand() + (rand() << 16))) % l->offset_random;
 175     if (pos >= end)
 176         pos = end;
 177     v->gen.bigpos = ((uint64_t)pos) << 32;
 178
 179     float delay = l->delay;
 180     if (l->delay_random)
 181         delay += rand() * (1.0 / RAND_MAX) * l->delay_random;
 182     if (delay > 0)
 183         v->delay = (int)(delay * m->module.srate);
 184     else
 185         v->delay = 0;
 186     v->gen.loop_overlap = l->loop_overlap;
 187     v->gen.loop_overlap_step = l->loop_overlap > 0 ? 1.0 / l->loop_overlap : 0;
 188     v->gain_fromvel = 1.0 + (l->eff_velcurve[vel] - 1.0) * l->amp_veltrack * 0.01;
 189     v->gain_shift = 0.0;
 190     v->note = note;
 191     v->vel = vel;
 192     v->pitch_shift = 0;
 193     v->released = 0;
 194     v->released_with_sustain = 0;
 195     v->released_with_sostenuto = 0;
 196     v->captured_sostenuto = 0;
 197     v->channel = c;
 198     v->layer = l;
 199     v->program = c->program;
 200     v->amp_env.shape = &l->amp_env_shape;
 201     v->filter_env.shape = &l->filter_env_shape;
 202     v->pitch_env.shape = &l->pitch_env_shape;
 203
 204     v->cutoff_shift = vel * l->fil_veltrack / 127.0 + (note - l->fil_keycenter) * l->fil_keytrack;
 205     v->loop_mode = l->loop_mode;
 206     v->off_by = l->off_by;
 207     int auxes = (m->module.outputs - m->module.aux_offset) / 2;
 208     if (l->effect1bus >= 1 && l->effect1bus < 1 + auxes)
 209         v->send1bus = l->effect1bus;
 210     else
 211         v->send1bus = 0;
 212     if (l->effect2bus >= 1 && l->effect2bus < 1 + auxes)
 213         v->send2bus = l->effect2bus;
 214     else
 215         v->send2bus = 0;
 216     v->send1gain = l->effect1 * 0.01;
 217     v->send2gain = l->effect2 * 0.01;
 218     if (l->group >= 1 && *pexgroupcount < MAX_RELEASED_GROUPS)
 219     {
 220         gboolean found = FALSE;
 221         for (int j = 0; j < *pexgroupcount; j++)
 222         {
 223             if (exgroups[j] == l->group)
 224             {
 225                 found = TRUE;
 226                 break;
 227             }
 228         }
 229         if (!found)
 230         {
 231             exgroups[(*pexgroupcount)++] = l->group;
 232         }
 233     }
 234     lfo_init(&v->amp_lfo, &l->amp_lfo, m->module.srate, m->module.srate_inv);
 235     lfo_init(&v->filter_lfo, &l->filter_lfo, m->module.srate, m->module.srate_inv);
 236     lfo_init(&v->pitch_lfo, &l->pitch_lfo, m->module.srate, m->module.srate_inv);
 237
 238     cbox_biquadf_reset(&v->filter_left);
 239     cbox_biquadf_reset(&v->filter_right);
 240     cbox_biquadf_reset(&v->filter_left2);
 241     cbox_biquadf_reset(&v->filter_right2);
 242
 243     GSList *nif = v->layer->nifs;
 244     while(nif)
 245     {
 246         struct sampler_noteinitfunc *p = nif->data;
 247         p->notefunc(p, v);
 248         nif = nif->next;
 249     }
 250
 251     cbox_envelope_reset(&v->amp_env);
 252     cbox_envelope_reset(&v->filter_env);
 253     cbox_envelope_reset(&v->pitch_env);
 254
 255     sampler_voice_activate(v, l->eff_waveform->info.channels == 2 ? spt_stereo16 : spt_mono16);
 256
 257     if (v->current_pipe && v->gen.bigpos)
 258         cbox_prefetch_pipe_consumed(v->current_pipe, v->gen.bigpos >> 32);
 259 }
 260
 261 void sampler_voice_link(struct sampler_voice **pv, struct sampler_voice *v)
 262 {
 263     v->prev = NULL;
 264     v->next = *pv;
 265     if (*pv)
 266         (*pv)->prev = v;
 267     *pv = v;
 268 }
 269
 270 void sampler_voice_unlink(struct sampler_voice **pv, struct sampler_voice *v)
 271 {
 272     if (*pv == v)
 273         *pv = v->next;
 274     if (v->prev)
 275         v->prev->next = v->next;
 276     if (v->next)
 277         v->next->prev = v->prev;
 278     v->prev = NULL;
 279     v->next = NULL;
 280 }
 281
 282 void sampler_voice_inactivate(struct sampler_voice *v, gboolean expect_active)
 283 {
 284     assert((v->gen.mode != spt_inactive) == expect_active);
 285     sampler_voice_unlink(&v->channel->voices_running, v);
 286     v->gen.mode = spt_inactive;
 287     if (v->current_pipe)
 288     {
 289         cbox_prefetch_stack_push(v->program->module->pipe_stack, v->current_pipe);
 290         v->current_pipe = NULL;
 291     }
 292     v->channel = NULL;
 293     sampler_voice_link(&v->program->module->voices_free, v);
 294 }
 295
 296 void sampler_voice_release(struct sampler_voice *v, gboolean is_polyaft)
 297 {
 298     if ((v->loop_mode == slm_one_shot_chokeable) != is_polyaft)
 299         return;
 300     if (v->delay >= v->age + CBOX_BLOCK_SIZE)
 301     {
 302         v->released = 1;
 303         sampler_voice_inactivate(v, TRUE);
 304     }
 305     else
 306     {
 307         if (v->loop_mode != slm_one_shot && !v->layer->count)
 308         {
 309             v->released = 1;
 310             if (v->loop_mode == slm_loop_sustain && v->current_pipe)
 311             {
 312                 // Break the loop
 313                 v->current_pipe->file_loop_end = v->gen.cur_sample_end;
 314                 v->current_pipe->file_loop_start = -1;
 315             }
 316         }
 317     }
 318 }
 319
 320 void sampler_voice_process(struct sampler_voice *v, struct sampler_module *m, cbox_sample_t **outputs)
 321 {
 322     struct sampler_layer_data *l = v->layer;
 323     assert(v->gen.mode != spt_inactive);
 324
 325     // if it's a DAHD envelope without sustain, consider the note finished
 326     if (v->amp_env.cur_stage == 4 && v->amp_env.shape->stages[3].end_value <= 0.f)
 327         cbox_envelope_go_to(&v->amp_env, 15);
 328
 329     struct sampler_channel *c = v->channel;
 330     v->age += CBOX_BLOCK_SIZE;
 331
 332     if (v->age < v->delay)
 333         return;
 334
 335     if (v->last_waveform != v->layer->eff_waveform)
 336     {
 337         v->last_waveform = v->layer->eff_waveform;
 338         if (v->layer->eff_waveform)
 339         {
 340             v->gen.mode = v->layer->eff_waveform->info.channels == 2 ? spt_stereo16 : spt_mono16;
 341             v->gen.cur_sample_end = v->layer->eff_waveform->info.frames;
 342         }
 343         else
 344         {
 345             sampler_voice_inactivate(v, TRUE);
 346             return;
 347         }
 348     }
 349     // XXXKF I'm sacrificing sample accuracy for delays for now
 350     v->delay = 0;
 351
 352     float pitch = (v->note - l->pitch_keycenter) * l->pitch_keytrack + l->tune + l->transpose * 100 + v->pitch_shift;
 353     float modsrcs[smsrc_pernote_count];
 354     modsrcs[smsrc_vel - smsrc_pernote_offset] = v->vel * (1.f / 127.f);
 355     modsrcs[smsrc_pitch - smsrc_pernote_offset] = pitch * (1.f / 100.f);
 356     modsrcs[smsrc_polyaft - smsrc_pernote_offset] = 0.f; // XXXKF not supported yet
 357     modsrcs[smsrc_pitchenv - smsrc_pernote_offset] = cbox_envelope_get_next(&v->pitch_env, v->released) * 0.01f;
 358     modsrcs[smsrc_filenv - smsrc_pernote_offset] = cbox_envelope_get_next(&v->filter_env, v->released) * 0.01f;
 359     modsrcs[smsrc_ampenv - smsrc_pernote_offset] = cbox_envelope_get_next(&v->amp_env, v->released) * 0.01f;
 360
 361     modsrcs[smsrc_amplfo - smsrc_pernote_offset] = lfo_run(&v->amp_lfo);
 362     modsrcs[smsrc_fillfo - smsrc_pernote_offset] = lfo_run(&v->filter_lfo);
 363     modsrcs[smsrc_pitchlfo - smsrc_pernote_offset] = lfo_run(&v->pitch_lfo);
 364
 365     if (v->amp_env.cur_stage < 0)
 366     {
 367         if (is_tail_finished(v))
 368         {
 369             sampler_voice_inactivate(v, TRUE);
 370             return;
 371         }
 372     }
 373
 374     float moddests[smdestcount];
 375     moddests[smdest_gain] = 0;
 376     moddests[smdest_pitch] = pitch;
 377     moddests[smdest_cutoff] = v->cutoff_shift;
 378     moddests[smdest_resonance] = 0;
 379     GSList *mod = l->modulations;
 380     if (l->trigger == stm_release)
 381         moddests[smdest_gain] -= v->age * l->rt_decay * m->module.srate_inv;
 382
 383     if (c->pitchwheel)
 384         moddests[smdest_pitch] += c->pitchwheel * (c->pitchwheel > 0 ? l->bend_up : l->bend_down) >> 13;
 385
 386     static const int modoffset[4] = {0, -1, -1, 1 };
 387     static const int modscale[4] = {1, 1, 2, -2 };
 388     while(mod)
 389     {
 390         struct sampler_modulation *sm = mod->data;
 391         float value = 0.f, value2 = 1.f;
 392         if (sm->src < smsrc_pernote_offset)
 393             value = c->cc[sm->src] * (1.f / 127.f);
 394         else
 395             value = modsrcs[sm->src - smsrc_pernote_offset];
 396         value = modoffset[sm->flags & 3] + value * modscale[sm->flags & 3];
 397
 398         if (sm->src2 != smsrc_none)
 399         {
 400             if (sm->src2 < smsrc_pernote_offset)
 401                 value2 = c->cc[sm->src2] * (1.f / 127.f);
 402             else
 403                 value2 = modsrcs[sm->src2 - smsrc_pernote_offset];
 404
 405             value2 = modoffset[(sm->flags & 12) >> 2] + value2 * modscale[(sm->flags & 12) >> 2];
 406             value *= value2;
 407         }
 408         moddests[sm->dest] += value * sm->amount;
 409
 410         mod = g_slist_next(mod);
 411     }
 412
 413     double maxv = 127 << 7;
 414     double freq = l->eff_freq * cent2factor(moddests[smdest_pitch]) ;
 415     uint64_t freq64 = (uint64_t)(freq * 65536.0 * 65536.0 * m->module.srate_inv);
 416
 417     gboolean playing_sustain_loop = !v->released && v->loop_mode == slm_loop_sustain;
 418     uint32_t loop_start, loop_end;
 419     gboolean bandlimited = FALSE;
 420
 421     if (!v->current_pipe)
 422     {
 423         v->gen.sample_data = v->last_waveform->data;
 424         if (v->last_waveform->levels)
 425         {
 426             // XXXKF: optimise later by caching last lookup value
 427             // XXXKF: optimise later by using binary search
 428             for (int i = 0; i < v->last_waveform->level_count; i++)
 429             {
 430                 if (freq64 <= v->last_waveform->levels[i].max_rate)
 431                 {
 432                     v->gen.sample_data = v->last_waveform->levels[i].data;
 433                     bandlimited = TRUE;
 434
 435                     break;
 436                 }
 437             }
 438         }
 439     }
 440
 441     gboolean play_loop = v->layer->loop_end && (v->loop_mode == slm_loop_continuous || playing_sustain_loop) && v->layer->on_cc_number == -1;
 442     loop_start = play_loop ? v->layer->loop_start : (v->layer->count ? 0 : (uint32_t)-1);
 443     loop_end = play_loop ? v->layer->loop_end : v->gen.cur_sample_end;
 444
 445     if (v->current_pipe)
 446     {
 447         v->gen.sample_data = v->gen.loop_count ? v->current_pipe->data : v->last_waveform->data;
 448         v->gen.streaming_buffer = v->current_pipe->data;
 449
 450         v->gen.prefetch_only_loop = (loop_end < v->last_waveform->preloaded_frames);
 451         v->gen.loop_overlap = 0;
 452         if (v->gen.prefetch_only_loop)
 453         {
 454             assert(!v->gen.in_streaming_buffer); // XXXKF this won't hold true when loops are edited while sound is being played (but that's not supported yet anyway)
 455             v->gen.loop_start = loop_start;
 456             v->gen.loop_end = loop_end;
 457             v->gen.streaming_buffer_frames = 0;
 458         }
 459         else
 460         {
 461             v->gen.loop_start = 0;
 462             v->gen.loop_end = v->last_waveform->preloaded_frames;
 463             v->gen.streaming_buffer_frames = v->current_pipe->buffer_loop_end;
 464         }
 465     }
 466     else
 467     {
 468         v->gen.loop_count = v->layer->count;
 469         v->gen.loop_start = loop_start;
 470         v->gen.loop_end = loop_end;
 471
 472         if (!bandlimited)
 473         {
 474             // Use pre-calculated join
 475             v->gen.scratch = loop_start == (uint32_t)-1 ? v->layer->scratch_end : v->layer->scratch_loop;
 476         }
 477         else
 478         {
 479             // The standard waveforms have extra MAX_INTERPOLATION_ORDER of samples from the loop start added past loop_end,
 480             // to avoid wasting time generating the joins in all the practical cases. The slow path covers custom loops
 481             // (i.e. partial loop or no loop) over bandlimited versions of the standard waveforms, and those are probably
 482             // not very useful anyway, as changing the loop removes the guarantee of the waveform being bandlimited and
 483             // may cause looping artifacts or introduce DC offset (e.g. if only a positive part of a sine wave is looped).
 484             if (loop_start == 0 && loop_end == l->eff_waveform->info.frames)
 485                 v->gen.scratch = v->gen.sample_data + l->eff_waveform->info.frames - MAX_INTERPOLATION_ORDER;
 486             else
 487             {
 488                 // Generate the join for the current wave level
 489                 // XXXKF this could be optimised further, by checking if waveform and loops are the same as the last
 490                 // time. However, this code is
 491                 int shift = l->eff_waveform->info.channels == 2 ? 1 : 0;
 492                 uint32_t halfscratch = MAX_INTERPOLATION_ORDER << shift;
 493
 494                 v->gen.scratch = v->gen.scratch_bandlimited;
 495                 memcpy(&v->gen.scratch_bandlimited[0], &v->gen.sample_data[(loop_end - MAX_INTERPOLATION_ORDER) << shift], halfscratch * sizeof(int16_t) );
 496                 if (loop_start != (uint32_t)-1)
 497                     memcpy(v->gen.scratch_bandlimited + halfscratch, &v->gen.sample_data[loop_start << shift], halfscratch * sizeof(int16_t));
 498                 else
 499                     memset(v->gen.scratch_bandlimited + halfscratch, 0, halfscratch * sizeof(int16_t));
 500             }
 501         }
 502     }
 503
 504
 505     v->gen.bigdelta = freq64;
 506     float gain = modsrcs[smsrc_ampenv - smsrc_pernote_offset] * l->volume_linearized * v->gain_fromvel * sampler_channel_addcc(c, 7) * sampler_channel_addcc(c, 11) / (maxv * maxv);
 507     if (moddests[smdest_gain] != 0.f)
 508         gain *= dB2gain(moddests[smdest_gain]);
 509     // http://drealm.info/sfz/plj-sfz.xhtml#amp "The overall gain must remain in the range -144 to 6 decibels."
 510     if (gain > 2.f)
 511         gain = 2.f;
 512     float pan = (l->pan + 100.f) * (1.f / 200.f) + (sampler_channel_addcc(c, 10) * 1.f / maxv - 0.5f) * 2.f;
 513     if (pan < 0.f)
 514         pan = 0.f;
 515     if (pan > 1.f)
 516         pan = 1.f;
 517     v->gen.lgain = gain * (1.f - pan)  / 32768.f;
 518     v->gen.rgain = gain * pan / 32768.f;
 519     if (l->cutoff != -1.f)
 520     {
 521         float cutoff = l->cutoff * cent2factor(moddests[smdest_cutoff]);
 522         if (cutoff < 20.f)
 523             cutoff = 20.f;
 524         if (cutoff > m->module.srate * 0.45f)
 525             cutoff = m->module.srate * 0.45f;
 526         //float resonance = v->resonance*pow(32.0,c->cc[71]/maxv);
 527         float resonance = l->resonance_linearized * dB2gain(moddests[smdest_resonance]);
 528         if (resonance < 0.7f)
 529             resonance = 0.7f;
 530         if (resonance > 32.f)
 531             resonance = 32.f;
 532         // XXXKF this is found experimentally and probably far off from correct formula
 533         if (is_4pole(v->layer))
 534             resonance = sqrtf(resonance / 0.707f) * 0.5f;
 535         switch(l->fil_type)
 536         {
 537         case sft_lp12:
 538         case sft_lp24:
 539             cbox_biquadf_set_lp_rbj(&v->filter_coeffs, cutoff, resonance, m->module.srate);
 540             break;
 541         case sft_hp12:
 542         case sft_hp24:
 543             cbox_biquadf_set_hp_rbj(&v->filter_coeffs, cutoff, resonance, m->module.srate);
 544             break;
 545         case sft_bp6:
 546         case sft_bp12:
 547             cbox_biquadf_set_bp_rbj(&v->filter_coeffs, cutoff, resonance, m->module.srate);
 548             break;
 549         case sft_lp6:
 550             cbox_biquadf_set_1plp(&v->filter_coeffs, cutoff, m->module.srate);
 551             break;
 552         case sft_hp6:
 553             cbox_biquadf_set_1php(&v->filter_coeffs, cutoff, m->module.srate);
 554             break;
 555         default:
 556             assert(0);
 557         }
 558     }
 559
 560     float left[CBOX_BLOCK_SIZE], right[CBOX_BLOCK_SIZE];
 561
 562     uint32_t samples = 0;
 563
 564
 565     if (v->current_pipe)
 566     {
 567         uint32_t limit = cbox_prefetch_pipe_get_remaining(v->current_pipe) - 4;
 568         if (!limit)
 569             v->gen.mode = spt_inactive;
 570         else
 571         {
 572             samples = sampler_gen_sample_playback(&v->gen, left, right, limit);
 573             cbox_prefetch_pipe_consumed(v->current_pipe, v->gen.consumed);
 574             v->gen.consumed = 0;
 575         }
 576     }
 577     else
 578     {
 579         samples = sampler_gen_sample_playback(&v->gen, left, right, (uint32_t)-1);
 580     }
 581
 582     for (int i = samples; i < CBOX_BLOCK_SIZE; i++)
 583         left[i] = right[i] = 0.f;
 584     if (l->cutoff != -1)
 585     {
 586         cbox_biquadf_process(&v->filter_left, &v->filter_coeffs, left);
 587         if (is_4pole(v->layer))
 588             cbox_biquadf_process(&v->filter_left2, &v->filter_coeffs, left);
 589         cbox_biquadf_process(&v->filter_right, &v->filter_coeffs, right);
 590         if (is_4pole(v->layer))
 591             cbox_biquadf_process(&v->filter_right2, &v->filter_coeffs, right);
 592     }
 593     mix_block_into_with_gain(outputs, v->output_pair_no * 2, left, right, 1.f);
 594     if ((v->send1bus > 0 && v->send1gain != 0) || (v->send2bus > 0 && v->send2gain != 0))
 595     {
 596         if (v->send1bus > 0 && v->send1gain != 0)
 597         {
 598             int oofs = m->module.aux_offset + (v->send1bus - 1) * 2;
 599             mix_block_into_with_gain(outputs, oofs, left, right, v->send1gain);
 600         }
 601         if (v->send2bus > 0 && v->send2gain != 0)
 602         {
 603             int oofs = m->module.aux_offset + (v->send2bus - 1) * 2;
 604             mix_block_into_with_gain(outputs, oofs, left, right, v->send2gain);
 605         }
 606     }
 607     if (v->gen.mode == spt_inactive)
 608         sampler_voice_inactivate(v, FALSE);
 609 }
 610