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.h"
  20 #include "config-api.h"
  21 #include "dspmath.h"
  22 #include "errors.h"
  23 #include "midi.h"
  24 #include "module.h"
  25 #include "rt.h"
  26 #include "sampler.h"
  27 #include "sampler_impl.h"
  28 #include "sfzloader.h"
  29 #include "stm.h"
  30 #include <assert.h>
  31 #include <errno.h>
  32 #include <glib.h>
  33 #include <math.h>
  34 #include <memory.h>
  35 #include <sndfile.h>
  36 #include <stdio.h>
  37 #include <stdlib.h>
  38
  39
  40 static void lfo_init(struct sampler_lfo *lfo, struct sampler_lfo_params *lfop, int srate, double srate_inv)
  41 {
  42     lfo->phase = 0;
  43     lfo->age = 0;
  44     lfo->delta = (uint32_t)(lfop->freq * 65536.0 * 65536.0 * CBOX_BLOCK_SIZE * srate_inv);
  45     lfo->delay = (uint32_t)(lfop->delay * srate);
  46     lfo->fade = (uint32_t)(lfop->fade * srate);
  47 }
  48
  49 static inline float lfo_run(struct sampler_lfo *lfo)
  50 {
  51     if (lfo->age < lfo->delay)
  52     {
  53         lfo->age += CBOX_BLOCK_SIZE;
  54         return 0.f;
  55     }
  56
  57     const int FRAC_BITS = 32 - 11;
  58     lfo->phase += lfo->delta;
  59     uint32_t iphase = lfo->phase >> FRAC_BITS;
  60     float frac = (lfo->phase & ((1 << FRAC_BITS) - 1)) * (1.0 / (1 << FRAC_BITS));
  61
  62     float v = sampler_sine_wave[iphase] + (sampler_sine_wave[iphase + 1] - sampler_sine_wave[iphase]) * frac;
  63     if (lfo->fade && lfo->age < lfo->delay + lfo->fade)
  64     {
  65         v *= (lfo->age - lfo->delay) * 1.0 / lfo->fade;
  66         lfo->age += CBOX_BLOCK_SIZE;
  67     }
  68
  69     return v;
  70 }
  71
  72 static gboolean is_tail_finished(struct sampler_voice *v)
  73 {
  74     if (v->layer->cutoff == -1)
  75         return TRUE;
  76     double eps = 1.0 / 65536.0;
  77     if (cbox_biquadf_is_audible(&v->filter_left, eps))
  78         return FALSE;
  79     if (cbox_biquadf_is_audible(&v->filter_right, eps))
  80         return FALSE;
  81     if (sampler_layer_data_is_4pole(v->layer))
  82     {
  83         if (cbox_biquadf_is_audible(&v->filter_left2, eps))
  84             return FALSE;
  85         if (cbox_biquadf_is_audible(&v->filter_right2, eps))
  86             return FALSE;
  87     }
  88
  89     return TRUE;
  90 }
  91
  92 #if USE_NEON
  93
  94 #include <arm_neon.h>
  95
  96 static inline void mix_block_into_with_gain(cbox_sample_t **outputs, int oofs, float *src_left, float *src_right, float gain)
  97 {
  98     float *dst_left = outputs[oofs];
  99     float *dst_right = outputs[oofs + 1];
 100     float32x2_t gain2 = {gain, gain};
 101     for (size_t i = 0; i < CBOX_BLOCK_SIZE; i += 4)
 102     {
 103         float32x2_t l1 = vld1_f32(&src_left[i]);
 104         float32x2_t l2 = vld1_f32(&src_left[i + 2]);
 105         float32x2_t r1 = vld1_f32(&src_right[i]);
 106         float32x2_t r2 = vld1_f32(&src_right[i + 2]);
 107         float32x2_t dl1 = vld1_f32(&dst_left[i]);
 108         float32x2_t dl2 = vld1_f32(&dst_left[i + 2]);
 109         float32x2_t dr1 = vld1_f32(&dst_right[i]);
 110         float32x2_t dr2 = vld1_f32(&dst_right[i + 2]);
 111
 112         l1 = vmla_f32(dl1, l1, gain2);
 113         l2 = vmla_f32(dl2, l2, gain2);
 114         vst1_f32(&dst_left[i], l1);
 115         vst1_f32(&dst_left[i + 2], l2);
 116         r1 = vmla_f32(dr1, r1, gain2);
 117         r2 = vmla_f32(dr2, r2, gain2);
 118         vst1_f32(&dst_right[i], r1);
 119         vst1_f32(&dst_right[i + 2], r2);
 120     }
 121 }
 122
 123 #else
 124
 125 static inline void mix_block_into_with_gain(cbox_sample_t **outputs, int oofs, float *src_left, float *src_right, float gain)
 126 {
 127     cbox_sample_t *dst_left = outputs[oofs];
 128     cbox_sample_t *dst_right = outputs[oofs + 1];
 129     for (size_t i = 0; i < CBOX_BLOCK_SIZE; i++)
 130     {
 131         dst_left[i] += gain * src_left[i];
 132         dst_right[i] += gain * src_right[i];
 133     }
 134 }
 135
 136 #endif
 137
 138 ////////////////////////////////////////////////////////////////////////////////
 139
 140 void sampler_voice_activate(struct sampler_voice *v, enum sampler_player_type mode)
 141 {
 142     assert(v->gen.mode == spt_inactive);
 143     sampler_voice_unlink(&v->program->module->voices_free, v);
 144     assert(mode != spt_inactive);
 145     assert(v->channel);
 146     v->gen.mode = mode;
 147     sampler_voice_link(&v->channel->voices_running, v);
 148 }
 149
 150 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)
 151 {
 152     struct sampler_module *m = c->module;
 153     sampler_gen_reset(&v->gen);
 154
 155     v->age = 0;
 156     if (l->trigger == stm_release)
 157     {
 158         // time since last 'note on' for that note
 159         v->age = m->current_time - c->prev_note_start_time[note];
 160         double age = v->age *  m->module.srate_inv;
 161         // if attenuation is more than 84dB, ignore the release trigger
 162         if (age * l->rt_decay > 84)
 163             return;
 164     }
 165     uint32_t end = l->eff_waveform->info.frames;
 166     if (l->end != 0)
 167         end = (l->end == -1) ? 0 : l->end;
 168     v->last_waveform = l->eff_waveform;
 169     v->gen.cur_sample_end = end;
 170     if (end > l->eff_waveform->info.frames)
 171         end = l->eff_waveform->info.frames;
 172
 173     assert(!v->current_pipe);
 174     if (end > l->eff_waveform->preloaded_frames)
 175     {
 176         if (l->loop_mode == slm_loop_continuous && l->loop_end < l->eff_waveform->preloaded_frames)
 177         {
 178             // Everything fits in prefetch, because loop ends in prefetch and post-loop part is not being played
 179         }
 180         else
 181         {
 182             uint32_t loop_start = -1, loop_end = end;
 183             // If in loop mode, set the loop over the looped part... unless we're doing sustain-only loop on prefetch area only. Then
 184             // streaming will only cover the release part, and it shouldn't be looped.
 185             if (l->loop_mode == slm_loop_continuous || (l->loop_mode == slm_loop_sustain && l->loop_end >= l->eff_waveform->preloaded_frames))
 186             {
 187                 loop_start = l->loop_start;
 188                 loop_end = l->loop_end;
 189             }
 190             // Those are initial values only, they will be adjusted in process function
 191             v->current_pipe = cbox_prefetch_stack_pop(m->pipe_stack, l->eff_waveform, loop_start, loop_end, l->count);
 192             if (!v->current_pipe)
 193                 g_warning("Prefetch pipe pool exhausted, no streaming playback will be possible");
 194         }
 195     }
 196
 197     v->output_pair_no = l->output % m->output_pairs;
 198     v->serial_no = m->serial_no;
 199
 200     uint32_t pos = l->offset;
 201     pos = l->offset;
 202     if (l->offset_random)
 203         pos += ((uint32_t)(rand() + (rand() << 16))) % l->offset_random;
 204     if (pos >= end)
 205         pos = end;
 206     v->gen.bigpos = ((uint64_t)pos) << 32;
 207
 208     float delay = l->delay;
 209     if (l->delay_random)
 210         delay += rand() * (1.0 / RAND_MAX) * l->delay_random;
 211     if (delay > 0)
 212         v->delay = (int)(delay * m->module.srate);
 213     else
 214         v->delay = 0;
 215     v->gen.loop_overlap = l->loop_overlap;
 216     v->gen.loop_overlap_step = l->loop_overlap > 0 ? 1.0 / l->loop_overlap : 0;
 217     v->gain_fromvel = 1.0 + (l->eff_velcurve[vel] - 1.0) * l->amp_veltrack * 0.01;
 218     v->gain_shift = 0.0;
 219     v->note = note;
 220     v->vel = vel;
 221     v->pitch_shift = 0;
 222     v->released = 0;
 223     v->released_with_sustain = 0;
 224     v->released_with_sostenuto = 0;
 225     v->captured_sostenuto = 0;
 226     v->channel = c;
 227     v->layer = l;
 228     v->program = c->program;
 229     v->amp_env.shape = &l->amp_env_shape;
 230     v->filter_env.shape = &l->filter_env_shape;
 231     v->pitch_env.shape = &l->pitch_env_shape;
 232
 233     v->cutoff_shift = vel * l->fil_veltrack / 127.0 + (note - l->fil_keycenter) * l->fil_keytrack;
 234     v->loop_mode = l->loop_mode;
 235     v->off_by = l->off_by;
 236     int auxes = (m->module.outputs - m->module.aux_offset) / 2;
 237     if (l->effect1bus >= 1 && l->effect1bus < 1 + auxes)
 238         v->send1bus = l->effect1bus;
 239     else
 240         v->send1bus = 0;
 241     if (l->effect2bus >= 1 && l->effect2bus < 1 + auxes)
 242         v->send2bus = l->effect2bus;
 243     else
 244         v->send2bus = 0;
 245     v->send1gain = l->effect1 * 0.01;
 246     v->send2gain = l->effect2 * 0.01;
 247     if (l->group >= 1 && *pexgroupcount < MAX_RELEASED_GROUPS)
 248     {
 249         gboolean found = FALSE;
 250         for (int j = 0; j < *pexgroupcount; j++)
 251         {
 252             if (exgroups[j] == l->group)
 253             {
 254                 found = TRUE;
 255                 break;
 256             }
 257         }
 258         if (!found)
 259         {
 260             exgroups[(*pexgroupcount)++] = l->group;
 261         }
 262     }
 263     lfo_init(&v->amp_lfo, &l->amp_lfo, m->module.srate, m->module.srate_inv);
 264     lfo_init(&v->filter_lfo, &l->filter_lfo, m->module.srate, m->module.srate_inv);
 265     lfo_init(&v->pitch_lfo, &l->pitch_lfo, m->module.srate, m->module.srate_inv);
 266
 267     cbox_biquadf_reset(&v->filter_left);
 268     cbox_biquadf_reset(&v->filter_right);
 269     cbox_biquadf_reset(&v->filter_left2);
 270     cbox_biquadf_reset(&v->filter_right2);
 271
 272     GSList *nif = v->layer->nifs;
 273     while(nif)
 274     {
 275         struct sampler_noteinitfunc *p = nif->data;
 276         p->notefunc(p, v);
 277         nif = nif->next;
 278     }
 279
 280     cbox_envelope_reset(&v->amp_env);
 281     cbox_envelope_reset(&v->filter_env);
 282     cbox_envelope_reset(&v->pitch_env);
 283
 284     sampler_voice_activate(v, l->eff_waveform->info.channels == 2 ? spt_stereo16 : spt_mono16);
 285
 286     if (v->current_pipe && v->gen.bigpos)
 287         cbox_prefetch_pipe_consumed(v->current_pipe, v->gen.bigpos >> 32);
 288 }
 289
 290 void sampler_voice_link(struct sampler_voice **pv, struct sampler_voice *v)
 291 {
 292     v->prev = NULL;
 293     v->next = *pv;
 294     if (*pv)
 295         (*pv)->prev = v;
 296     *pv = v;
 297 }
 298
 299 void sampler_voice_unlink(struct sampler_voice **pv, struct sampler_voice *v)
 300 {
 301     if (*pv == v)
 302         *pv = v->next;
 303     if (v->prev)
 304         v->prev->next = v->next;
 305     if (v->next)
 306         v->next->prev = v->prev;
 307     v->prev = NULL;
 308     v->next = NULL;
 309 }
 310
 311 void sampler_voice_inactivate(struct sampler_voice *v, gboolean expect_active)
 312 {
 313     assert((v->gen.mode != spt_inactive) == expect_active);
 314     sampler_voice_unlink(&v->channel->voices_running, v);
 315     v->gen.mode = spt_inactive;
 316     if (v->current_pipe)
 317     {
 318         cbox_prefetch_stack_push(v->program->module->pipe_stack, v->current_pipe);
 319         v->current_pipe = NULL;
 320     }
 321     v->channel = NULL;
 322     sampler_voice_link(&v->program->module->voices_free, v);
 323 }
 324
 325 void sampler_voice_release(struct sampler_voice *v, gboolean is_polyaft)
 326 {
 327     if ((v->loop_mode == slm_one_shot_chokeable) != is_polyaft)
 328         return;
 329     if (v->delay >= v->age + CBOX_BLOCK_SIZE)
 330     {
 331         v->released = 1;
 332         sampler_voice_inactivate(v, TRUE);
 333     }
 334     else
 335     {
 336         if (v->loop_mode != slm_one_shot && !v->layer->count)
 337         {
 338             v->released = 1;
 339             if (v->loop_mode == slm_loop_sustain && v->current_pipe)
 340             {
 341                 // Break the loop
 342                 v->current_pipe->file_loop_end = v->gen.cur_sample_end;
 343                 v->current_pipe->file_loop_start = -1;
 344             }
 345         }
 346     }
 347 }
 348
 349 void sampler_voice_process(struct sampler_voice *v, struct sampler_module *m, cbox_sample_t **outputs)
 350 {
 351     struct sampler_layer_data *l = v->layer;
 352     assert(v->gen.mode != spt_inactive);
 353
 354     // if it's a DAHD envelope without sustain, consider the note finished
 355     if (v->amp_env.cur_stage == 4 && v->amp_env.shape->stages[3].end_value <= 0.f)
 356         cbox_envelope_go_to(&v->amp_env, 15);
 357
 358     struct sampler_channel *c = v->channel;
 359     v->age += CBOX_BLOCK_SIZE;
 360
 361     if (v->age < v->delay)
 362         return;
 363
 364     if (v->last_waveform != v->layer->eff_waveform)
 365     {
 366         v->last_waveform = v->layer->eff_waveform;
 367         if (v->layer->eff_waveform)
 368         {
 369             v->gen.mode = v->layer->eff_waveform->info.channels == 2 ? spt_stereo16 : spt_mono16;
 370             v->gen.cur_sample_end = v->layer->eff_waveform->info.frames;
 371         }
 372         else
 373         {
 374             sampler_voice_inactivate(v, TRUE);
 375             return;
 376         }
 377     }
 378     // XXXKF I'm sacrificing sample accuracy for delays for now
 379     v->delay = 0;
 380
 381     float pitch = (v->note - l->pitch_keycenter) * l->pitch_keytrack + l->tune + l->transpose * 100 + v->pitch_shift;
 382     float modsrcs[smsrc_pernote_count];
 383     modsrcs[smsrc_vel - smsrc_pernote_offset] = v->vel * (1.f / 127.f);
 384     modsrcs[smsrc_pitch - smsrc_pernote_offset] = pitch * (1.f / 100.f);
 385     modsrcs[smsrc_polyaft - smsrc_pernote_offset] = 0.f; // XXXKF not supported yet
 386     modsrcs[smsrc_pitchenv - smsrc_pernote_offset] = cbox_envelope_get_next(&v->pitch_env, v->released) * 0.01f;
 387     modsrcs[smsrc_filenv - smsrc_pernote_offset] = cbox_envelope_get_next(&v->filter_env, v->released) * 0.01f;
 388     modsrcs[smsrc_ampenv - smsrc_pernote_offset] = cbox_envelope_get_next(&v->amp_env, v->released) * 0.01f;
 389
 390     modsrcs[smsrc_amplfo - smsrc_pernote_offset] = lfo_run(&v->amp_lfo);
 391     modsrcs[smsrc_fillfo - smsrc_pernote_offset] = lfo_run(&v->filter_lfo);
 392     modsrcs[smsrc_pitchlfo - smsrc_pernote_offset] = lfo_run(&v->pitch_lfo);
 393
 394     if (v->amp_env.cur_stage < 0)
 395     {
 396         if (is_tail_finished(v))
 397         {
 398             sampler_voice_inactivate(v, TRUE);
 399             return;
 400         }
 401     }
 402
 403     float moddests[smdestcount];
 404     moddests[smdest_gain] = 0;
 405     moddests[smdest_pitch] = pitch;
 406     moddests[smdest_cutoff] = v->cutoff_shift;
 407     moddests[smdest_resonance] = 0;
 408     GSList *mod = l->modulations;
 409     if (l->trigger == stm_release)
 410         moddests[smdest_gain] -= v->age * l->rt_decay * m->module.srate_inv;
 411
 412     if (c->pitchwheel)
 413         moddests[smdest_pitch] += c->pitchwheel * (c->pitchwheel > 0 ? l->bend_up : l->bend_down) >> 13;
 414
 415     static const int modoffset[4] = {0, -1, -1, 1 };
 416     static const int modscale[4] = {1, 1, 2, -2 };
 417     while(mod)
 418     {
 419         struct sampler_modulation *sm = mod->data;
 420         float value = 0.f, value2 = 1.f;
 421         if (sm->src < smsrc_pernote_offset)
 422             value = c->cc[sm->src] * (1.f / 127.f);
 423         else
 424             value = modsrcs[sm->src - smsrc_pernote_offset];
 425         value = modoffset[sm->flags & 3] + value * modscale[sm->flags & 3];
 426
 427         if (sm->src2 != smsrc_none)
 428         {
 429             if (sm->src2 < smsrc_pernote_offset)
 430                 value2 = c->cc[sm->src2] * (1.f / 127.f);
 431             else
 432                 value2 = modsrcs[sm->src2 - smsrc_pernote_offset];
 433
 434             value2 = modoffset[(sm->flags & 12) >> 2] + value2 * modscale[(sm->flags & 12) >> 2];
 435             value *= value2;
 436         }
 437         moddests[sm->dest] += value * sm->amount;
 438
 439         mod = g_slist_next(mod);
 440     }
 441
 442     double maxv = 127 << 7;
 443     double freq = l->eff_freq * cent2factor(moddests[smdest_pitch]) ;
 444     uint64_t freq64 = (uint64_t)(freq * 65536.0 * 65536.0 * m->module.srate_inv);
 445
 446     gboolean playing_sustain_loop = !v->released && v->loop_mode == slm_loop_sustain;
 447     uint32_t loop_start, loop_end;
 448     gboolean bandlimited = FALSE;
 449
 450     if (!v->current_pipe)
 451     {
 452         v->gen.sample_data = v->last_waveform->data;
 453         if (v->last_waveform->levels)
 454         {
 455             // XXXKF: optimise later by caching last lookup value
 456             // XXXKF: optimise later by using binary search
 457             for (int i = 0; i < v->last_waveform->level_count; i++)
 458             {
 459                 if (freq64 <= v->last_waveform->levels[i].max_rate)
 460                 {
 461                     v->gen.sample_data = v->last_waveform->levels[i].data;
 462                     bandlimited = TRUE;
 463
 464                     break;
 465                 }
 466             }
 467         }
 468     }
 469
 470     gboolean play_loop = v->layer->loop_end && (v->loop_mode == slm_loop_continuous || playing_sustain_loop) && v->layer->on_cc_number == -1;
 471     loop_start = play_loop ? v->layer->loop_start : (v->layer->count ? 0 : (uint32_t)-1);
 472     loop_end = play_loop ? v->layer->loop_end : v->gen.cur_sample_end;
 473
 474     if (v->current_pipe)
 475     {
 476         v->gen.sample_data = v->gen.loop_count ? v->current_pipe->data : v->last_waveform->data;
 477         v->gen.streaming_buffer = v->current_pipe->data;
 478
 479         v->gen.prefetch_only_loop = (loop_end < v->last_waveform->preloaded_frames);
 480         v->gen.loop_overlap = 0;
 481         if (v->gen.prefetch_only_loop)
 482         {
 483             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)
 484             v->gen.loop_start = loop_start;
 485             v->gen.loop_end = loop_end;
 486             v->gen.streaming_buffer_frames = 0;
 487         }
 488         else
 489         {
 490             v->gen.loop_start = 0;
 491             v->gen.loop_end = v->last_waveform->preloaded_frames;
 492             v->gen.streaming_buffer_frames = v->current_pipe->buffer_loop_end;
 493         }
 494     }
 495     else
 496     {
 497         v->gen.loop_count = v->layer->count;
 498         v->gen.loop_start = loop_start;
 499         v->gen.loop_end = loop_end;
 500
 501         if (!bandlimited)
 502         {
 503             // Use pre-calculated join
 504             v->gen.scratch = loop_start == (uint32_t)-1 ? v->layer->scratch_end : v->layer->scratch_loop;
 505         }
 506         else
 507         {
 508             // The standard waveforms have extra MAX_INTERPOLATION_ORDER of samples from the loop start added past loop_end,
 509             // to avoid wasting time generating the joins in all the practical cases. The slow path covers custom loops
 510             // (i.e. partial loop or no loop) over bandlimited versions of the standard waveforms, and those are probably
 511             // not very useful anyway, as changing the loop removes the guarantee of the waveform being bandlimited and
 512             // may cause looping artifacts or introduce DC offset (e.g. if only a positive part of a sine wave is looped).
 513             if (loop_start == 0 && loop_end == l->eff_waveform->info.frames)
 514                 v->gen.scratch = v->gen.sample_data + l->eff_waveform->info.frames - MAX_INTERPOLATION_ORDER;
 515             else
 516             {
 517                 // Generate the join for the current wave level
 518                 // XXXKF this could be optimised further, by checking if waveform and loops are the same as the last
 519                 // time. However, this code is
 520                 int shift = l->eff_waveform->info.channels == 2 ? 1 : 0;
 521                 uint32_t halfscratch = MAX_INTERPOLATION_ORDER << shift;
 522
 523                 v->gen.scratch = v->gen.scratch_bandlimited;
 524                 memcpy(&v->gen.scratch_bandlimited[0], &v->gen.sample_data[(loop_end - MAX_INTERPOLATION_ORDER) << shift], halfscratch * sizeof(int16_t) );
 525                 if (loop_start != (uint32_t)-1)
 526                     memcpy(v->gen.scratch_bandlimited + halfscratch, &v->gen.sample_data[loop_start << shift], halfscratch * sizeof(int16_t));
 527                 else
 528                     memset(v->gen.scratch_bandlimited + halfscratch, 0, halfscratch * sizeof(int16_t));
 529             }
 530         }
 531     }
 532
 533
 534     v->gen.bigdelta = freq64;
 535     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);
 536     if (moddests[smdest_gain] != 0.f)
 537         gain *= dB2gain(moddests[smdest_gain]);
 538     // http://drealm.info/sfz/plj-sfz.xhtml#amp "The overall gain must remain in the range -144 to 6 decibels."
 539     if (gain > 2.f)
 540         gain = 2.f;
 541     float pan = (l->pan + 100.f) * (1.f / 200.f) + (sampler_channel_addcc(c, 10) * 1.f / maxv - 0.5f) * 2.f;
 542     if (pan < 0.f)
 543         pan = 0.f;
 544     if (pan > 1.f)
 545         pan = 1.f;
 546     v->gen.lgain = gain * (1.f - pan)  / 32768.f;
 547     v->gen.rgain = gain * pan / 32768.f;
 548     gboolean is4p = sampler_layer_data_is_4pole(v->layer);
 549     if (l->cutoff != -1.f)
 550     {
 551         float logcutoff = l->logcutoff + moddests[smdest_cutoff];
 552         if (logcutoff < 0)
 553             logcutoff = 0;
 554         if (logcutoff > 12798)
 555             logcutoff = 12798;
 556         //float resonance = v->resonance*pow(32.0,c->cc[71]/maxv);
 557         float resonance = l->resonance_linearized * dB2gain((is4p ? 0.5 : 1) * moddests[smdest_resonance]);
 558         if (resonance < 0.7f)
 559             resonance = 0.7f;
 560         if (resonance > 32.f)
 561             resonance = 32.f;
 562         switch(l->fil_type)
 563         {
 564         case sft_lp12:
 565         case sft_lp24:
 566         {
 567             cbox_biquadf_set_lp_rbj_lookup(&v->filter_coeffs, &m->sincos[(int)logcutoff], resonance);
 568             break;
 569         }
 570         case sft_hp12:
 571         case sft_hp24:
 572             cbox_biquadf_set_hp_rbj_lookup(&v->filter_coeffs, &m->sincos[(int)logcutoff], resonance);
 573             break;
 574         case sft_bp6:
 575         case sft_bp12:
 576             cbox_biquadf_set_bp_rbj_lookup(&v->filter_coeffs, &m->sincos[(int)logcutoff], resonance);
 577             break;
 578         case sft_lp6:
 579         case sft_lp12nr:
 580         case sft_lp24nr:
 581             cbox_biquadf_set_1plp_lookup(&v->filter_coeffs, &m->sincos[(int)logcutoff], l->fil_type != sft_lp6);
 582             break;
 583         case sft_hp6:
 584         case sft_hp12nr:
 585         case sft_hp24nr:
 586             cbox_biquadf_set_1php_lookup(&v->filter_coeffs, &m->sincos[(int)logcutoff], l->fil_type != sft_hp6);
 587             break;
 588         default:
 589             assert(0);
 590         }
 591     }
 592
 593     float left[CBOX_BLOCK_SIZE], right[CBOX_BLOCK_SIZE];
 594
 595     uint32_t samples = 0;
 596
 597
 598     if (v->current_pipe)
 599     {
 600         uint32_t limit = cbox_prefetch_pipe_get_remaining(v->current_pipe) - 4;
 601         if (!limit)
 602             v->gen.mode = spt_inactive;
 603         else
 604         {
 605             samples = sampler_gen_sample_playback(&v->gen, left, right, limit);
 606             cbox_prefetch_pipe_consumed(v->current_pipe, v->gen.consumed);
 607             v->gen.consumed = 0;
 608         }
 609     }
 610     else
 611     {
 612         samples = sampler_gen_sample_playback(&v->gen, left, right, (uint32_t)-1);
 613     }
 614
 615     for (int i = samples; i < CBOX_BLOCK_SIZE; i++)
 616         left[i] = right[i] = 0.f;
 617     if (l->cutoff != -1)
 618     {
 619         cbox_biquadf_process(&v->filter_left, &v->filter_coeffs, left);
 620         if (is4p)
 621             cbox_biquadf_process(&v->filter_left2, &v->filter_coeffs, left);
 622         cbox_biquadf_process(&v->filter_right, &v->filter_coeffs, right);
 623         if (is4p)
 624             cbox_biquadf_process(&v->filter_right2, &v->filter_coeffs, right);
 625     }
 626     mix_block_into_with_gain(outputs, v->output_pair_no * 2, left, right, 1.f);
 627     if ((v->send1bus > 0 && v->send1gain != 0) || (v->send2bus > 0 && v->send2gain != 0))
 628     {
 629         if (v->send1bus > 0 && v->send1gain != 0)
 630         {
 631             int oofs = m->module.aux_offset + (v->send1bus - 1) * 2;
 632             mix_block_into_with_gain(outputs, oofs, left, right, v->send1gain);
 633         }
 634         if (v->send2bus > 0 && v->send2gain != 0)
 635         {
 636             int oofs = m->module.aux_offset + (v->send2bus - 1) * 2;
 637             mix_block_into_with_gain(outputs, oofs, left, right, v->send2gain);
 638         }
 639     }
 640     if (v->gen.mode == spt_inactive)
 641         sampler_voice_inactivate(v, FALSE);
 642 }
 643