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_leftright, 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 += 2)
 102     {
 103         float32x2_t lr1 = vld1_f32(&src_leftright[2 * i]);
 104         float32x2_t lr2 = vld1_f32(&src_leftright[2 * i + 2]);
 105         float32x2x2_t lr12 = vtrn_f32(lr1, lr2);
 106         float32x2_t dl1 = vld1_f32(&dst_left[i]);
 107         float32x2_t dr1 = vld1_f32(&dst_right[i]);
 108
 109         float32x2_t l1 = vmla_f32(dl1, lr12.val[0], gain2);
 110         vst1_f32(&dst_left[i], l1);
 111         float32x2_t r1 = vmla_f32(dr1, lr12.val[1], gain2);
 112         vst1_f32(&dst_right[i], r1);
 113     }
 114 }
 115
 116 static inline void mix_block_into(cbox_sample_t **outputs, int oofs, float *src_leftright)
 117 {
 118     float *dst_left = outputs[oofs];
 119     float *dst_right = outputs[oofs + 1];
 120     for (size_t i = 0; i < CBOX_BLOCK_SIZE; i += 2)
 121     {
 122         float32x2_t lr1 = vld1_f32(&src_leftright[2 * i]);
 123         float32x2_t lr2 = vld1_f32(&src_leftright[2 * i + 2]);
 124         float32x2x2_t lr12 = vtrn_f32(lr1, lr2);
 125         float32x2_t dl1 = vld1_f32(&dst_left[i]);
 126         float32x2_t dr1 = vld1_f32(&dst_right[i]);
 127
 128         float32x2_t l1 = vadd_f32(dl1, lr12.val[0]);
 129         vst1_f32(&dst_left[i], l1);
 130         float32x2_t r1 = vadd_f32(dr1, lr12.val[1]);
 131         vst1_f32(&dst_right[i], r1);
 132     }
 133 }
 134
 135 #else
 136
 137 static inline void mix_block_into_with_gain(cbox_sample_t **outputs, int oofs, float *src_leftright, float gain)
 138 {
 139     cbox_sample_t *dst_left = outputs[oofs];
 140     cbox_sample_t *dst_right = outputs[oofs + 1];
 141     for (size_t i = 0; i < CBOX_BLOCK_SIZE; i++)
 142     {
 143         dst_left[i] += gain * src_leftright[2 * i];
 144         dst_right[i] += gain * src_leftright[2 * i + 1];
 145     }
 146 }
 147
 148 static inline void mix_block_into(cbox_sample_t **outputs, int oofs, float *src_leftright)
 149 {
 150     cbox_sample_t *dst_left = outputs[oofs];
 151     cbox_sample_t *dst_right = outputs[oofs + 1];
 152     for (size_t i = 0; i < CBOX_BLOCK_SIZE; i++)
 153     {
 154         dst_left[i] += src_leftright[2 * i];
 155         dst_right[i] += src_leftright[2 * i + 1];
 156     }
 157 }
 158
 159 #endif
 160
 161 ////////////////////////////////////////////////////////////////////////////////
 162
 163 void sampler_voice_activate(struct sampler_voice *v, enum sampler_player_type mode)
 164 {
 165     assert(v->gen.mode == spt_inactive);
 166     sampler_voice_unlink(&v->program->module->voices_free, v);
 167     assert(mode != spt_inactive);
 168     assert(v->channel);
 169     v->gen.mode = mode;
 170     sampler_voice_link(&v->channel->voices_running, v);
 171 }
 172
 173 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)
 174 {
 175     struct sampler_module *m = c->module;
 176     sampler_gen_reset(&v->gen);
 177
 178     v->age = 0;
 179     if (l->trigger == stm_release)
 180     {
 181         // time since last 'note on' for that note
 182         v->age = m->current_time - c->prev_note_start_time[note];
 183         double age = v->age *  m->module.srate_inv;
 184         // if attenuation is more than 84dB, ignore the release trigger
 185         if (age * l->rt_decay > 84)
 186             return;
 187     }
 188     uint32_t end = l->eff_waveform->info.frames;
 189     if (l->end != 0)
 190         end = (l->end == -1) ? 0 : l->end;
 191     v->last_waveform = l->eff_waveform;
 192     v->gen.cur_sample_end = end;
 193     if (end > l->eff_waveform->info.frames)
 194         end = l->eff_waveform->info.frames;
 195
 196     assert(!v->current_pipe);
 197     if (end > l->eff_waveform->preloaded_frames)
 198     {
 199         if (l->loop_mode == slm_loop_continuous && l->loop_end < l->eff_waveform->preloaded_frames)
 200         {
 201             // Everything fits in prefetch, because loop ends in prefetch and post-loop part is not being played
 202         }
 203         else
 204         {
 205             uint32_t loop_start = -1, loop_end = end;
 206             // If in loop mode, set the loop over the looped part... unless we're doing sustain-only loop on prefetch area only. Then
 207             // streaming will only cover the release part, and it shouldn't be looped.
 208             if (l->loop_mode == slm_loop_continuous || (l->loop_mode == slm_loop_sustain && l->loop_end >= l->eff_waveform->preloaded_frames))
 209             {
 210                 loop_start = l->loop_start;
 211                 loop_end = l->loop_end;
 212             }
 213             // Those are initial values only, they will be adjusted in process function
 214             v->current_pipe = cbox_prefetch_stack_pop(m->pipe_stack, l->eff_waveform, loop_start, loop_end, l->count);
 215             if (!v->current_pipe)
 216                 g_warning("Prefetch pipe pool exhausted, no streaming playback will be possible");
 217         }
 218     }
 219
 220     v->output_pair_no = l->output % m->output_pairs;
 221     v->serial_no = m->serial_no;
 222
 223     uint32_t pos = l->offset;
 224     pos = l->offset;
 225     if (l->offset_random)
 226         pos += ((uint32_t)(rand() + (rand() << 16))) % l->offset_random;
 227     if (pos >= end)
 228         pos = end;
 229     v->gen.bigpos = ((uint64_t)pos) << 32;
 230
 231     float delay = l->delay;
 232     if (l->delay_random)
 233         delay += rand() * (1.0 / RAND_MAX) * l->delay_random;
 234     if (delay > 0)
 235         v->delay = (int)(delay * m->module.srate);
 236     else
 237         v->delay = 0;
 238     v->gen.loop_overlap = l->loop_overlap;
 239     v->gen.loop_overlap_step = l->loop_overlap > 0 ? 1.0 / l->loop_overlap : 0;
 240     v->gain_fromvel = 1.0 + (l->eff_velcurve[vel] - 1.0) * l->amp_veltrack * 0.01;
 241     v->gain_shift = 0.0;
 242     v->note = note;
 243     v->vel = vel;
 244     v->pitch_shift = 0;
 245     v->released = 0;
 246     v->released_with_sustain = 0;
 247     v->released_with_sostenuto = 0;
 248     v->captured_sostenuto = 0;
 249     v->channel = c;
 250     v->layer = l;
 251     v->program = c->program;
 252     v->amp_env.shape = &l->amp_env_shape;
 253     v->filter_env.shape = &l->filter_env_shape;
 254     v->pitch_env.shape = &l->pitch_env_shape;
 255
 256     v->cutoff_shift = vel * l->fil_veltrack / 127.0 + (note - l->fil_keycenter) * l->fil_keytrack;
 257     v->loop_mode = l->loop_mode;
 258     v->off_by = l->off_by;
 259     int auxes = (m->module.outputs - m->module.aux_offset) / 2;
 260     if (l->effect1bus >= 1 && l->effect1bus < 1 + auxes)
 261         v->send1bus = l->effect1bus;
 262     else
 263         v->send1bus = 0;
 264     if (l->effect2bus >= 1 && l->effect2bus < 1 + auxes)
 265         v->send2bus = l->effect2bus;
 266     else
 267         v->send2bus = 0;
 268     v->send1gain = l->effect1 * 0.01;
 269     v->send2gain = l->effect2 * 0.01;
 270     if (l->group >= 1 && *pexgroupcount < MAX_RELEASED_GROUPS)
 271     {
 272         gboolean found = FALSE;
 273         for (int j = 0; j < *pexgroupcount; j++)
 274         {
 275             if (exgroups[j] == l->group)
 276             {
 277                 found = TRUE;
 278                 break;
 279             }
 280         }
 281         if (!found)
 282         {
 283             exgroups[(*pexgroupcount)++] = l->group;
 284         }
 285     }
 286     lfo_init(&v->amp_lfo, &l->amp_lfo, m->module.srate, m->module.srate_inv);
 287     lfo_init(&v->filter_lfo, &l->filter_lfo, m->module.srate, m->module.srate_inv);
 288     lfo_init(&v->pitch_lfo, &l->pitch_lfo, m->module.srate, m->module.srate_inv);
 289
 290     cbox_biquadf_reset(&v->filter_left);
 291     cbox_biquadf_reset(&v->filter_right);
 292     cbox_biquadf_reset(&v->filter_left2);
 293     cbox_biquadf_reset(&v->filter_right2);
 294     cbox_onepolef_reset(&v->onepole_left);
 295     cbox_onepolef_reset(&v->onepole_right);
 296     // set gain later (it's a less expensive operation)
 297     if (l->tonectl_freq != 0)
 298         cbox_onepolef_set_highshelf_tonectl(&v->onepole_coeffs, l->tonectl_freq * M_PI * m->module.srate_inv, 1.0);
 299
 300     GSList *nif = v->layer->nifs;
 301     while(nif)
 302     {
 303         struct sampler_noteinitfunc *p = nif->data;
 304         p->notefunc(p, v);
 305         nif = nif->next;
 306     }
 307
 308     cbox_envelope_reset(&v->amp_env);
 309     cbox_envelope_reset(&v->filter_env);
 310     cbox_envelope_reset(&v->pitch_env);
 311
 312     v->last_eq_bitmask = 0;
 313
 314     sampler_voice_activate(v, l->eff_waveform->info.channels == 2 ? spt_stereo16 : spt_mono16);
 315
 316     if (v->current_pipe && v->gen.bigpos)
 317         cbox_prefetch_pipe_consumed(v->current_pipe, v->gen.bigpos >> 32);
 318     v->layer_changed = TRUE;
 319 }
 320
 321 void sampler_voice_link(struct sampler_voice **pv, struct sampler_voice *v)
 322 {
 323     v->prev = NULL;
 324     v->next = *pv;
 325     if (*pv)
 326         (*pv)->prev = v;
 327     *pv = v;
 328 }
 329
 330 void sampler_voice_unlink(struct sampler_voice **pv, struct sampler_voice *v)
 331 {
 332     if (*pv == v)
 333         *pv = v->next;
 334     if (v->prev)
 335         v->prev->next = v->next;
 336     if (v->next)
 337         v->next->prev = v->prev;
 338     v->prev = NULL;
 339     v->next = NULL;
 340 }
 341
 342 void sampler_voice_inactivate(struct sampler_voice *v, gboolean expect_active)
 343 {
 344     assert((v->gen.mode != spt_inactive) == expect_active);
 345     sampler_voice_unlink(&v->channel->voices_running, v);
 346     v->gen.mode = spt_inactive;
 347     if (v->current_pipe)
 348     {
 349         cbox_prefetch_stack_push(v->program->module->pipe_stack, v->current_pipe);
 350         v->current_pipe = NULL;
 351     }
 352     v->channel = NULL;
 353     sampler_voice_link(&v->program->module->voices_free, v);
 354 }
 355
 356 void sampler_voice_release(struct sampler_voice *v, gboolean is_polyaft)
 357 {
 358     if ((v->loop_mode == slm_one_shot_chokeable) != is_polyaft)
 359         return;
 360     if (v->delay >= v->age + CBOX_BLOCK_SIZE)
 361     {
 362         v->released = 1;
 363         sampler_voice_inactivate(v, TRUE);
 364     }
 365     else
 366     {
 367         if (v->loop_mode != slm_one_shot && !v->layer->count)
 368         {
 369             v->released = 1;
 370             if (v->loop_mode == slm_loop_sustain && v->current_pipe)
 371             {
 372                 // Break the loop
 373                 v->current_pipe->file_loop_end = v->gen.cur_sample_end;
 374                 v->current_pipe->file_loop_start = -1;
 375             }
 376         }
 377     }
 378 }
 379
 380 void sampler_voice_process(struct sampler_voice *v, struct sampler_module *m, cbox_sample_t **outputs)
 381 {
 382     struct sampler_layer_data *l = v->layer;
 383     assert(v->gen.mode != spt_inactive);
 384
 385     // if it's a DAHD envelope without sustain, consider the note finished
 386     if (__builtin_expect(v->amp_env.cur_stage == 4 && v->amp_env.shape->stages[3].end_value <= 0.f, 0))
 387         cbox_envelope_go_to(&v->amp_env, 15);
 388
 389     struct sampler_channel *c = v->channel;
 390     v->age += CBOX_BLOCK_SIZE;
 391
 392     if (__builtin_expect(v->age < v->delay, 0))
 393         return;
 394
 395     // XXXKF I'm sacrificing sample accuracy for delays for now
 396     v->delay = 0;
 397     const float velscl = v->vel * (1.f / 127.f);
 398     if (__builtin_expect(v->layer_changed, 0))
 399     {
 400         v->last_level = -1;
 401         if (v->last_waveform != v->layer->eff_waveform)
 402         {
 403             v->last_waveform = v->layer->eff_waveform;
 404             if (v->layer->eff_waveform)
 405             {
 406                 v->gen.mode = v->layer->eff_waveform->info.channels == 2 ? spt_stereo16 : spt_mono16;
 407                 v->gen.cur_sample_end = v->layer->eff_waveform->info.frames;
 408             }
 409             else
 410             {
 411                 sampler_voice_inactivate(v, TRUE);
 412                 return;
 413             }
 414         }
 415     #define RECALC_EQ_IF(index) \
 416         if (l->eq_bitmask & (1 << (index - 1))) \
 417         { \
 418             cbox_biquadf_set_peakeq_rbj_scaled(&v->eq_coeffs[index - 1], l->eq##index.effective_freq + velscl * l->eq##index.vel2freq, 1.0 / l->eq##index.bw, dB2gain(0.5 * (l->eq##index.gain + velscl * l->eq##index.vel2gain)), m->module.srate); \
 419             if (!(v->last_eq_bitmask & (1 << (index - 1)))) \
 420             { \
 421                 cbox_biquadf_reset(&v->eq_left[index-1]); \
 422                 cbox_biquadf_reset(&v->eq_right[index-1]); \
 423             } \
 424         }
 425
 426         RECALC_EQ_IF(1)
 427         RECALC_EQ_IF(2)
 428         RECALC_EQ_IF(3)
 429         v->last_eq_bitmask = l->eq_bitmask;
 430         v->layer_changed = FALSE;
 431     }
 432
 433     float pitch = (v->note - l->pitch_keycenter) * l->pitch_keytrack + l->tune + l->transpose * 100 + v->pitch_shift;
 434     float modsrcs[smsrc_pernote_count];
 435     modsrcs[smsrc_vel - smsrc_pernote_offset] = v->vel * velscl;
 436     modsrcs[smsrc_pitch - smsrc_pernote_offset] = pitch * (1.f / 100.f);
 437     modsrcs[smsrc_polyaft - smsrc_pernote_offset] = 0.f; // XXXKF not supported yet
 438     modsrcs[smsrc_pitchenv - smsrc_pernote_offset] = cbox_envelope_get_next(&v->pitch_env, v->released) * 0.01f;
 439     modsrcs[smsrc_filenv - smsrc_pernote_offset] = cbox_envelope_get_next(&v->filter_env, v->released) * 0.01f;
 440     modsrcs[smsrc_ampenv - smsrc_pernote_offset] = cbox_envelope_get_next(&v->amp_env, v->released) * 0.01f;
 441
 442     modsrcs[smsrc_amplfo - smsrc_pernote_offset] = lfo_run(&v->amp_lfo);
 443     modsrcs[smsrc_fillfo - smsrc_pernote_offset] = lfo_run(&v->filter_lfo);
 444     modsrcs[smsrc_pitchlfo - smsrc_pernote_offset] = lfo_run(&v->pitch_lfo);
 445
 446     if (__builtin_expect(v->amp_env.cur_stage < 0, 0))
 447     {
 448         if (__builtin_expect(is_tail_finished(v), 0))
 449         {
 450             sampler_voice_inactivate(v, TRUE);
 451             return;
 452         }
 453     }
 454
 455     float moddests[smdestcount];
 456     moddests[smdest_gain] = 0;
 457     moddests[smdest_pitch] = pitch;
 458     moddests[smdest_cutoff] = v->cutoff_shift;
 459     moddests[smdest_resonance] = 0;
 460     moddests[smdest_tonectl] = 0;
 461     GSList *mod = l->modulations;
 462     if (__builtin_expect(l->trigger == stm_release, 0))
 463         moddests[smdest_gain] -= v->age * l->rt_decay * m->module.srate_inv;
 464
 465     if (c->pitchwheel)
 466         moddests[smdest_pitch] += c->pitchwheel * (c->pitchwheel > 0 ? l->bend_up : l->bend_down) >> 13;
 467
 468     static const int modoffset[4] = {0, -1, -1, 1 };
 469     static const int modscale[4] = {1, 1, 2, -2 };
 470     while(mod)
 471     {
 472         struct sampler_modulation *sm = mod->data;
 473         float value = 0.f, value2 = 1.f;
 474         if (sm->src < smsrc_pernote_offset)
 475             value = c->cc[sm->src] * (1.f / 127.f);
 476         else
 477             value = modsrcs[sm->src - smsrc_pernote_offset];
 478         value = modoffset[sm->flags & 3] + value * modscale[sm->flags & 3];
 479
 480         if (sm->src2 != smsrc_none)
 481         {
 482             if (sm->src2 < smsrc_pernote_offset)
 483                 value2 = c->cc[sm->src2] * (1.f / 127.f);
 484             else
 485                 value2 = modsrcs[sm->src2 - smsrc_pernote_offset];
 486
 487             value2 = modoffset[(sm->flags & 12) >> 2] + value2 * modscale[(sm->flags & 12) >> 2];
 488             value *= value2;
 489         }
 490         moddests[sm->dest] += value * sm->amount;
 491
 492         mod = g_slist_next(mod);
 493     }
 494
 495     double maxv = 127 << 7;
 496     double freq = l->eff_freq * cent2factor(moddests[smdest_pitch]) ;
 497     uint64_t freq64 = (uint64_t)(freq * 65536.0 * 65536.0 * m->module.srate_inv);
 498
 499     gboolean playing_sustain_loop = !v->released && v->loop_mode == slm_loop_sustain;
 500     uint32_t loop_start, loop_end;
 501     gboolean bandlimited = FALSE;
 502
 503     if (!v->current_pipe)
 504     {
 505         v->gen.sample_data = v->last_waveform->data;
 506         if (v->last_waveform->levels)
 507         {
 508             gboolean use_cached = v->last_level > 0 && v->last_level < v->last_waveform->level_count
 509                 && freq64 > v->last_level_min_rate && freq64 <= v->last_waveform->levels[v->last_level].max_rate;
 510             if (__builtin_expect(use_cached, 1))
 511             {
 512                 v->gen.sample_data = v->last_waveform->levels[v->last_level].data;
 513                 bandlimited = TRUE;
 514             }
 515             else
 516             {
 517                 for (int i = 0; i < v->last_waveform->level_count; i++)
 518                 {
 519                     if (freq64 <= v->last_waveform->levels[i].max_rate)
 520                     {
 521                         v->last_level = i;
 522                         v->gen.sample_data = v->last_waveform->levels[i].data;
 523                         bandlimited = TRUE;
 524
 525                         break;
 526                     }
 527                     v->last_level_min_rate = v->last_waveform->levels[i].max_rate;
 528                 }
 529             }
 530         }
 531     }
 532
 533     gboolean play_loop = v->layer->loop_end && (v->loop_mode == slm_loop_continuous || playing_sustain_loop) && v->layer->on_cc_number == -1;
 534     loop_start = play_loop ? v->layer->loop_start : (v->layer->count ? 0 : (uint32_t)-1);
 535     loop_end = play_loop ? v->layer->loop_end : v->gen.cur_sample_end;
 536
 537     if (v->current_pipe)
 538     {
 539         v->gen.sample_data = v->gen.loop_count ? v->current_pipe->data : v->last_waveform->data;
 540         v->gen.streaming_buffer = v->current_pipe->data;
 541
 542         v->gen.prefetch_only_loop = (loop_end < v->last_waveform->preloaded_frames);
 543         v->gen.loop_overlap = 0;
 544         if (v->gen.prefetch_only_loop)
 545         {
 546             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)
 547             v->gen.loop_start = loop_start;
 548             v->gen.loop_end = loop_end;
 549             v->gen.streaming_buffer_frames = 0;
 550         }
 551         else
 552         {
 553             v->gen.loop_start = 0;
 554             v->gen.loop_end = v->last_waveform->preloaded_frames;
 555             v->gen.streaming_buffer_frames = v->current_pipe->buffer_loop_end;
 556         }
 557     }
 558     else
 559     {
 560         v->gen.loop_count = v->layer->count;
 561         v->gen.loop_start = loop_start;
 562         v->gen.loop_end = loop_end;
 563
 564         if (!bandlimited)
 565         {
 566             // Use pre-calculated join
 567             v->gen.scratch = loop_start == (uint32_t)-1 ? v->layer->scratch_end : v->layer->scratch_loop;
 568         }
 569         else
 570         {
 571             // The standard waveforms have extra MAX_INTERPOLATION_ORDER of samples from the loop start added past loop_end,
 572             // to avoid wasting time generating the joins in all the practical cases. The slow path covers custom loops
 573             // (i.e. partial loop or no loop) over bandlimited versions of the standard waveforms, and those are probably
 574             // not very useful anyway, as changing the loop removes the guarantee of the waveform being bandlimited and
 575             // may cause looping artifacts or introduce DC offset (e.g. if only a positive part of a sine wave is looped).
 576             if (loop_start == 0 && loop_end == l->eff_waveform->info.frames)
 577                 v->gen.scratch = v->gen.sample_data + l->eff_waveform->info.frames - MAX_INTERPOLATION_ORDER;
 578             else
 579             {
 580                 // Generate the join for the current wave level
 581                 // XXXKF this could be optimised further, by checking if waveform and loops are the same as the last
 582                 // time. However, this code is not likely to be used... ever, so optimising it is not the priority.
 583                 int shift = l->eff_waveform->info.channels == 2 ? 1 : 0;
 584                 uint32_t halfscratch = MAX_INTERPOLATION_ORDER << shift;
 585
 586                 v->gen.scratch = v->gen.scratch_bandlimited;
 587                 memcpy(&v->gen.scratch_bandlimited[0], &v->gen.sample_data[(loop_end - MAX_INTERPOLATION_ORDER) << shift], halfscratch * sizeof(int16_t) );
 588                 if (loop_start != (uint32_t)-1)
 589                     memcpy(v->gen.scratch_bandlimited + halfscratch, &v->gen.sample_data[loop_start << shift], halfscratch * sizeof(int16_t));
 590                 else
 591                     memset(v->gen.scratch_bandlimited + halfscratch, 0, halfscratch * sizeof(int16_t));
 592             }
 593         }
 594     }
 595
 596
 597     v->gen.bigdelta = freq64;
 598     float gain = modsrcs[smsrc_ampenv - smsrc_pernote_offset] * l->volume_linearized * v->gain_fromvel * c->channel_volume_cc * sampler_channel_addcc(c, 11) / (maxv * maxv);
 599     if (moddests[smdest_gain] != 0.f)
 600         gain *= dB2gain(moddests[smdest_gain]);
 601     // http://drealm.info/sfz/plj-sfz.xhtml#amp "The overall gain must remain in the range -144 to 6 decibels."
 602     if (gain > 2.f)
 603         gain = 2.f;
 604     float pan = (l->pan + 100.f) * (1.f / 200.f) + (c->channel_pan_cc * 1.f / maxv - 0.5f) * 2.f;
 605     if (pan < 0.f)
 606         pan = 0.f;
 607     if (pan > 1.f)
 608         pan = 1.f;
 609     v->gen.lgain = gain * (1.f - pan)  / 32768.f;
 610     v->gen.rgain = gain * pan / 32768.f;
 611     struct cbox_biquadf_coeffs *second_filter = &v->filter_coeffs;
 612     gboolean is4p = sampler_layer_data_is_4pole(v->layer);
 613     if (l->cutoff != -1.f)
 614     {
 615         float logcutoff = l->logcutoff + moddests[smdest_cutoff];
 616         if (logcutoff < 0)
 617             logcutoff = 0;
 618         if (logcutoff > 12798)
 619             logcutoff = 12798;
 620         //float resonance = v->resonance*pow(32.0,c->cc[71]/maxv);
 621         float resonance = l->resonance_linearized * dB2gain((is4p ? 0.5 : 1) * moddests[smdest_resonance]);
 622         if (resonance < 0.7f)
 623             resonance = 0.7f;
 624         if (resonance > 32.f)
 625             resonance = 32.f;
 626         switch(l->fil_type)
 627         {
 628         case sft_lp24hybrid:
 629             cbox_biquadf_set_lp_rbj_lookup(&v->filter_coeffs, &m->sincos[(int)logcutoff], resonance * resonance);
 630             cbox_biquadf_set_1plp_lookup(&v->filter_coeffs_extra, &m->sincos[(int)logcutoff], 1);
 631             second_filter = &v->filter_coeffs_extra;
 632             break;
 633
 634         case sft_lp12:
 635         case sft_lp24:
 636             cbox_biquadf_set_lp_rbj_lookup(&v->filter_coeffs, &m->sincos[(int)logcutoff], resonance);
 637             break;
 638         case sft_hp12:
 639         case sft_hp24:
 640             cbox_biquadf_set_hp_rbj_lookup(&v->filter_coeffs, &m->sincos[(int)logcutoff], resonance);
 641             break;
 642         case sft_bp6:
 643         case sft_bp12:
 644             cbox_biquadf_set_bp_rbj_lookup(&v->filter_coeffs, &m->sincos[(int)logcutoff], resonance);
 645             break;
 646         case sft_lp6:
 647         case sft_lp12nr:
 648         case sft_lp24nr:
 649             cbox_biquadf_set_1plp_lookup(&v->filter_coeffs, &m->sincos[(int)logcutoff], l->fil_type != sft_lp6);
 650             break;
 651         case sft_hp6:
 652         case sft_hp12nr:
 653         case sft_hp24nr:
 654             cbox_biquadf_set_1php_lookup(&v->filter_coeffs, &m->sincos[(int)logcutoff], l->fil_type != sft_hp6);
 655             break;
 656         default:
 657             assert(0);
 658         }
 659     }
 660     if (__builtin_expect(l->tonectl_freq != 0, 0))
 661     {
 662         float ctl = l->tonectl + moddests[smdest_tonectl];
 663         if (fabs(ctl) > 0.0001f)
 664             cbox_onepolef_set_highshelf_setgain(&v->onepole_coeffs, dB2gain(ctl));
 665         else
 666             cbox_onepolef_set_highshelf_setgain(&v->onepole_coeffs, 1.0);
 667     }
 668
 669     float leftright[2 * CBOX_BLOCK_SIZE];
 670
 671     uint32_t samples = 0;
 672
 673
 674     if (v->current_pipe)
 675     {
 676         uint32_t limit = cbox_prefetch_pipe_get_remaining(v->current_pipe) - 4;
 677         if (!limit)
 678             v->gen.mode = spt_inactive;
 679         else
 680         {
 681             samples = sampler_gen_sample_playback(&v->gen, leftright, limit);
 682             cbox_prefetch_pipe_consumed(v->current_pipe, v->gen.consumed);
 683             v->gen.consumed = 0;
 684         }
 685     }
 686     else
 687     {
 688         samples = sampler_gen_sample_playback(&v->gen, leftright, (uint32_t)-1);
 689     }
 690
 691     for (int i = 2 * samples; i < 2 * CBOX_BLOCK_SIZE; i++)
 692         leftright[i] = 0.f;
 693     if (l->cutoff != -1)
 694     {
 695         cbox_biquadf_process_stereo(&v->filter_left, &v->filter_right, &v->filter_coeffs, leftright);
 696         if (is4p)
 697             cbox_biquadf_process_stereo(&v->filter_left, &v->filter_right, second_filter, leftright);
 698     }
 699     if (__builtin_expect(l->tonectl_freq != 0, 0))
 700     {
 701         cbox_onepolef_process_stereo(&v->onepole_left, &v->onepole_right, &v->onepole_coeffs, leftright);
 702     }
 703     if (__builtin_expect(l->eq_bitmask, 0))
 704     {
 705         for (int eq = 0; eq < 3; eq++)
 706         {
 707             if (l->eq_bitmask & (1 << eq))
 708             {
 709                 cbox_biquadf_process_stereo(&v->eq_left[eq], &v->eq_right[eq], &v->eq_coeffs[eq], leftright);
 710             }
 711         }
 712     }
 713
 714     mix_block_into(outputs, v->output_pair_no * 2, leftright);
 715     if (__builtin_expect((v->send1bus > 0 && v->send1gain != 0) || (v->send2bus > 0 && v->send2gain != 0), 0))
 716     {
 717         if (v->send1bus > 0 && v->send1gain != 0)
 718         {
 719             int oofs = m->module.aux_offset + (v->send1bus - 1) * 2;
 720             mix_block_into_with_gain(outputs, oofs, leftright, v->send1gain);
 721         }
 722         if (v->send2bus > 0 && v->send2gain != 0)
 723         {
 724             int oofs = m->module.aux_offset + (v->send2bus - 1) * 2;
 725             mix_block_into_with_gain(outputs, oofs, leftright, v->send2gain);
 726         }
 727     }
 728     if (v->gen.mode == spt_inactive)
 729         sampler_voice_inactivate(v, FALSE);
 730 }
 731