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 static inline void mix_block_into(cbox_sample_t **outputs, int oofs, float *src_left, float *src_right)
 124 {
 125     float *dst_left = outputs[oofs];
 126     float *dst_right = outputs[oofs + 1];
 127     for (size_t i = 0; i < CBOX_BLOCK_SIZE; i += 4)
 128     {
 129         float32x2_t l1 = vld1_f32(&src_left[i]);
 130         float32x2_t l2 = vld1_f32(&src_left[i + 2]);
 131         float32x2_t r1 = vld1_f32(&src_right[i]);
 132         float32x2_t r2 = vld1_f32(&src_right[i + 2]);
 133         float32x2_t dl1 = vld1_f32(&dst_left[i]);
 134         float32x2_t dl2 = vld1_f32(&dst_left[i + 2]);
 135         float32x2_t dr1 = vld1_f32(&dst_right[i]);
 136         float32x2_t dr2 = vld1_f32(&dst_right[i + 2]);
 137
 138         l1 = vadd_f32(dl1, l1);
 139         l2 = vadd_f32(dl2, l2);
 140         vst1_f32(&dst_left[i], l1);
 141         vst1_f32(&dst_left[i + 2], l2);
 142         r1 = vadd_f32(dr1, r1);
 143         r2 = vadd_f32(dr2, r2);
 144         vst1_f32(&dst_right[i], r1);
 145         vst1_f32(&dst_right[i + 2], r2);
 146     }
 147 }
 148
 149 #else
 150
 151 static inline void mix_block_into_with_gain(cbox_sample_t **outputs, int oofs, float *src_left, float *src_right, float gain)
 152 {
 153     cbox_sample_t *dst_left = outputs[oofs];
 154     cbox_sample_t *dst_right = outputs[oofs + 1];
 155     for (size_t i = 0; i < CBOX_BLOCK_SIZE; i++)
 156     {
 157         dst_left[i] += gain * src_left[i];
 158         dst_right[i] += gain * src_right[i];
 159     }
 160 }
 161
 162 static inline void mix_block_into(cbox_sample_t **outputs, int oofs, float *src_left, float *src_right)
 163 {
 164     cbox_sample_t *dst_left = outputs[oofs];
 165     cbox_sample_t *dst_right = outputs[oofs + 1];
 166     for (size_t i = 0; i < CBOX_BLOCK_SIZE; i++)
 167     {
 168         dst_left[i] += src_left[i];
 169         dst_right[i] += src_right[i];
 170     }
 171 }
 172
 173 #endif
 174
 175 ////////////////////////////////////////////////////////////////////////////////
 176
 177 void sampler_voice_activate(struct sampler_voice *v, enum sampler_player_type mode)
 178 {
 179     assert(v->gen.mode == spt_inactive);
 180     sampler_voice_unlink(&v->program->module->voices_free, v);
 181     assert(mode != spt_inactive);
 182     assert(v->channel);
 183     v->gen.mode = mode;
 184     sampler_voice_link(&v->channel->voices_running, v);
 185 }
 186
 187 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)
 188 {
 189     struct sampler_module *m = c->module;
 190     sampler_gen_reset(&v->gen);
 191
 192     v->age = 0;
 193     if (l->trigger == stm_release)
 194     {
 195         // time since last 'note on' for that note
 196         v->age = m->current_time - c->prev_note_start_time[note];
 197         double age = v->age *  m->module.srate_inv;
 198         // if attenuation is more than 84dB, ignore the release trigger
 199         if (age * l->rt_decay > 84)
 200             return;
 201     }
 202     uint32_t end = l->eff_waveform->info.frames;
 203     if (l->end != 0)
 204         end = (l->end == -1) ? 0 : l->end;
 205     v->last_waveform = l->eff_waveform;
 206     v->gen.cur_sample_end = end;
 207     if (end > l->eff_waveform->info.frames)
 208         end = l->eff_waveform->info.frames;
 209
 210     assert(!v->current_pipe);
 211     if (end > l->eff_waveform->preloaded_frames)
 212     {
 213         if (l->loop_mode == slm_loop_continuous && l->loop_end < l->eff_waveform->preloaded_frames)
 214         {
 215             // Everything fits in prefetch, because loop ends in prefetch and post-loop part is not being played
 216         }
 217         else
 218         {
 219             uint32_t loop_start = -1, loop_end = end;
 220             // If in loop mode, set the loop over the looped part... unless we're doing sustain-only loop on prefetch area only. Then
 221             // streaming will only cover the release part, and it shouldn't be looped.
 222             if (l->loop_mode == slm_loop_continuous || (l->loop_mode == slm_loop_sustain && l->loop_end >= l->eff_waveform->preloaded_frames))
 223             {
 224                 loop_start = l->loop_start;
 225                 loop_end = l->loop_end;
 226             }
 227             // Those are initial values only, they will be adjusted in process function
 228             v->current_pipe = cbox_prefetch_stack_pop(m->pipe_stack, l->eff_waveform, loop_start, loop_end, l->count);
 229             if (!v->current_pipe)
 230                 g_warning("Prefetch pipe pool exhausted, no streaming playback will be possible");
 231         }
 232     }
 233
 234     v->output_pair_no = l->output % m->output_pairs;
 235     v->serial_no = m->serial_no;
 236
 237     uint32_t pos = l->offset;
 238     pos = l->offset;
 239     if (l->offset_random)
 240         pos += ((uint32_t)(rand() + (rand() << 16))) % l->offset_random;
 241     if (pos >= end)
 242         pos = end;
 243     v->gen.bigpos = ((uint64_t)pos) << 32;
 244
 245     float delay = l->delay;
 246     if (l->delay_random)
 247         delay += rand() * (1.0 / RAND_MAX) * l->delay_random;
 248     if (delay > 0)
 249         v->delay = (int)(delay * m->module.srate);
 250     else
 251         v->delay = 0;
 252     v->gen.loop_overlap = l->loop_overlap;
 253     v->gen.loop_overlap_step = l->loop_overlap > 0 ? 1.0 / l->loop_overlap : 0;
 254     v->gain_fromvel = 1.0 + (l->eff_velcurve[vel] - 1.0) * l->amp_veltrack * 0.01;
 255     v->gain_shift = 0.0;
 256     v->note = note;
 257     v->vel = vel;
 258     v->pitch_shift = 0;
 259     v->released = 0;
 260     v->released_with_sustain = 0;
 261     v->released_with_sostenuto = 0;
 262     v->captured_sostenuto = 0;
 263     v->channel = c;
 264     v->layer = l;
 265     v->program = c->program;
 266     v->amp_env.shape = &l->amp_env_shape;
 267     v->filter_env.shape = &l->filter_env_shape;
 268     v->pitch_env.shape = &l->pitch_env_shape;
 269
 270     v->cutoff_shift = vel * l->fil_veltrack / 127.0 + (note - l->fil_keycenter) * l->fil_keytrack;
 271     v->loop_mode = l->loop_mode;
 272     v->off_by = l->off_by;
 273     int auxes = (m->module.outputs - m->module.aux_offset) / 2;
 274     if (l->effect1bus >= 1 && l->effect1bus < 1 + auxes)
 275         v->send1bus = l->effect1bus;
 276     else
 277         v->send1bus = 0;
 278     if (l->effect2bus >= 1 && l->effect2bus < 1 + auxes)
 279         v->send2bus = l->effect2bus;
 280     else
 281         v->send2bus = 0;
 282     v->send1gain = l->effect1 * 0.01;
 283     v->send2gain = l->effect2 * 0.01;
 284     if (l->group >= 1 && *pexgroupcount < MAX_RELEASED_GROUPS)
 285     {
 286         gboolean found = FALSE;
 287         for (int j = 0; j < *pexgroupcount; j++)
 288         {
 289             if (exgroups[j] == l->group)
 290             {
 291                 found = TRUE;
 292                 break;
 293             }
 294         }
 295         if (!found)
 296         {
 297             exgroups[(*pexgroupcount)++] = l->group;
 298         }
 299     }
 300     lfo_init(&v->amp_lfo, &l->amp_lfo, m->module.srate, m->module.srate_inv);
 301     lfo_init(&v->filter_lfo, &l->filter_lfo, m->module.srate, m->module.srate_inv);
 302     lfo_init(&v->pitch_lfo, &l->pitch_lfo, m->module.srate, m->module.srate_inv);
 303
 304     cbox_biquadf_reset(&v->filter_left);
 305     cbox_biquadf_reset(&v->filter_right);
 306     cbox_biquadf_reset(&v->filter_left2);
 307     cbox_biquadf_reset(&v->filter_right2);
 308
 309     GSList *nif = v->layer->nifs;
 310     while(nif)
 311     {
 312         struct sampler_noteinitfunc *p = nif->data;
 313         p->notefunc(p, v);
 314         nif = nif->next;
 315     }
 316
 317     cbox_envelope_reset(&v->amp_env);
 318     cbox_envelope_reset(&v->filter_env);
 319     cbox_envelope_reset(&v->pitch_env);
 320
 321     v->last_eq_bitmask = 0;
 322
 323     sampler_voice_activate(v, l->eff_waveform->info.channels == 2 ? spt_stereo16 : spt_mono16);
 324
 325     if (v->current_pipe && v->gen.bigpos)
 326         cbox_prefetch_pipe_consumed(v->current_pipe, v->gen.bigpos >> 32);
 327     v->layer_changed = TRUE;
 328 }
 329
 330 void sampler_voice_link(struct sampler_voice **pv, struct sampler_voice *v)
 331 {
 332     v->prev = NULL;
 333     v->next = *pv;
 334     if (*pv)
 335         (*pv)->prev = v;
 336     *pv = v;
 337 }
 338
 339 void sampler_voice_unlink(struct sampler_voice **pv, struct sampler_voice *v)
 340 {
 341     if (*pv == v)
 342         *pv = v->next;
 343     if (v->prev)
 344         v->prev->next = v->next;
 345     if (v->next)
 346         v->next->prev = v->prev;
 347     v->prev = NULL;
 348     v->next = NULL;
 349 }
 350
 351 void sampler_voice_inactivate(struct sampler_voice *v, gboolean expect_active)
 352 {
 353     assert((v->gen.mode != spt_inactive) == expect_active);
 354     sampler_voice_unlink(&v->channel->voices_running, v);
 355     v->gen.mode = spt_inactive;
 356     if (v->current_pipe)
 357     {
 358         cbox_prefetch_stack_push(v->program->module->pipe_stack, v->current_pipe);
 359         v->current_pipe = NULL;
 360     }
 361     v->channel = NULL;
 362     sampler_voice_link(&v->program->module->voices_free, v);
 363 }
 364
 365 void sampler_voice_release(struct sampler_voice *v, gboolean is_polyaft)
 366 {
 367     if ((v->loop_mode == slm_one_shot_chokeable) != is_polyaft)
 368         return;
 369     if (v->delay >= v->age + CBOX_BLOCK_SIZE)
 370     {
 371         v->released = 1;
 372         sampler_voice_inactivate(v, TRUE);
 373     }
 374     else
 375     {
 376         if (v->loop_mode != slm_one_shot && !v->layer->count)
 377         {
 378             v->released = 1;
 379             if (v->loop_mode == slm_loop_sustain && v->current_pipe)
 380             {
 381                 // Break the loop
 382                 v->current_pipe->file_loop_end = v->gen.cur_sample_end;
 383                 v->current_pipe->file_loop_start = -1;
 384             }
 385         }
 386     }
 387 }
 388
 389 void sampler_voice_process(struct sampler_voice *v, struct sampler_module *m, cbox_sample_t **outputs)
 390 {
 391     struct sampler_layer_data *l = v->layer;
 392     assert(v->gen.mode != spt_inactive);
 393
 394     // if it's a DAHD envelope without sustain, consider the note finished
 395     if (__builtin_expect(v->amp_env.cur_stage == 4 && v->amp_env.shape->stages[3].end_value <= 0.f, 0))
 396         cbox_envelope_go_to(&v->amp_env, 15);
 397
 398     struct sampler_channel *c = v->channel;
 399     v->age += CBOX_BLOCK_SIZE;
 400
 401     if (__builtin_expect(v->age < v->delay, 0))
 402         return;
 403
 404     // XXXKF I'm sacrificing sample accuracy for delays for now
 405     v->delay = 0;
 406     const float velscl = v->vel * (1.f / 127.f);
 407     if (__builtin_expect(v->layer_changed, 0))
 408     {
 409         v->last_level = -1;
 410         if (v->last_waveform != v->layer->eff_waveform)
 411         {
 412             v->last_waveform = v->layer->eff_waveform;
 413             if (v->layer->eff_waveform)
 414             {
 415                 v->gen.mode = v->layer->eff_waveform->info.channels == 2 ? spt_stereo16 : spt_mono16;
 416                 v->gen.cur_sample_end = v->layer->eff_waveform->info.frames;
 417             }
 418             else
 419             {
 420                 sampler_voice_inactivate(v, TRUE);
 421                 return;
 422             }
 423         }
 424     #define RECALC_EQ_IF(index) \
 425         if (l->eq_bitmask & (1 << (index - 1))) \
 426         { \
 427             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); \
 428             if (!(v->last_eq_bitmask & (1 << (index - 1)))) \
 429             { \
 430                 cbox_biquadf_reset(&v->eq_left[index-1]); \
 431                 cbox_biquadf_reset(&v->eq_right[index-1]); \
 432             } \
 433         }
 434
 435         RECALC_EQ_IF(1)
 436         RECALC_EQ_IF(2)
 437         RECALC_EQ_IF(3)
 438         v->last_eq_bitmask = l->eq_bitmask;
 439         v->layer_changed = FALSE;
 440     }
 441
 442     float pitch = (v->note - l->pitch_keycenter) * l->pitch_keytrack + l->tune + l->transpose * 100 + v->pitch_shift;
 443     float modsrcs[smsrc_pernote_count];
 444     modsrcs[smsrc_vel - smsrc_pernote_offset] = v->vel * velscl;
 445     modsrcs[smsrc_pitch - smsrc_pernote_offset] = pitch * (1.f / 100.f);
 446     modsrcs[smsrc_polyaft - smsrc_pernote_offset] = 0.f; // XXXKF not supported yet
 447     modsrcs[smsrc_pitchenv - smsrc_pernote_offset] = cbox_envelope_get_next(&v->pitch_env, v->released) * 0.01f;
 448     modsrcs[smsrc_filenv - smsrc_pernote_offset] = cbox_envelope_get_next(&v->filter_env, v->released) * 0.01f;
 449     modsrcs[smsrc_ampenv - smsrc_pernote_offset] = cbox_envelope_get_next(&v->amp_env, v->released) * 0.01f;
 450
 451     modsrcs[smsrc_amplfo - smsrc_pernote_offset] = lfo_run(&v->amp_lfo);
 452     modsrcs[smsrc_fillfo - smsrc_pernote_offset] = lfo_run(&v->filter_lfo);
 453     modsrcs[smsrc_pitchlfo - smsrc_pernote_offset] = lfo_run(&v->pitch_lfo);
 454
 455     if (__builtin_expect(v->amp_env.cur_stage < 0, 0))
 456     {
 457         if (is_tail_finished(v))
 458         {
 459             sampler_voice_inactivate(v, TRUE);
 460             return;
 461         }
 462     }
 463
 464     float moddests[smdestcount];
 465     moddests[smdest_gain] = 0;
 466     moddests[smdest_pitch] = pitch;
 467     moddests[smdest_cutoff] = v->cutoff_shift;
 468     moddests[smdest_resonance] = 0;
 469     GSList *mod = l->modulations;
 470     if (__builtin_expect(l->trigger == stm_release, 0))
 471         moddests[smdest_gain] -= v->age * l->rt_decay * m->module.srate_inv;
 472
 473     if (c->pitchwheel)
 474         moddests[smdest_pitch] += c->pitchwheel * (c->pitchwheel > 0 ? l->bend_up : l->bend_down) >> 13;
 475
 476     static const int modoffset[4] = {0, -1, -1, 1 };
 477     static const int modscale[4] = {1, 1, 2, -2 };
 478     while(mod)
 479     {
 480         struct sampler_modulation *sm = mod->data;
 481         float value = 0.f, value2 = 1.f;
 482         if (sm->src < smsrc_pernote_offset)
 483             value = c->cc[sm->src] * (1.f / 127.f);
 484         else
 485             value = modsrcs[sm->src - smsrc_pernote_offset];
 486         value = modoffset[sm->flags & 3] + value * modscale[sm->flags & 3];
 487
 488         if (sm->src2 != smsrc_none)
 489         {
 490             if (sm->src2 < smsrc_pernote_offset)
 491                 value2 = c->cc[sm->src2] * (1.f / 127.f);
 492             else
 493                 value2 = modsrcs[sm->src2 - smsrc_pernote_offset];
 494
 495             value2 = modoffset[(sm->flags & 12) >> 2] + value2 * modscale[(sm->flags & 12) >> 2];
 496             value *= value2;
 497         }
 498         moddests[sm->dest] += value * sm->amount;
 499
 500         mod = g_slist_next(mod);
 501     }
 502
 503     double maxv = 127 << 7;
 504     double freq = l->eff_freq * cent2factor(moddests[smdest_pitch]) ;
 505     uint64_t freq64 = (uint64_t)(freq * 65536.0 * 65536.0 * m->module.srate_inv);
 506
 507     gboolean playing_sustain_loop = !v->released && v->loop_mode == slm_loop_sustain;
 508     uint32_t loop_start, loop_end;
 509     gboolean bandlimited = FALSE;
 510
 511     if (!v->current_pipe)
 512     {
 513         v->gen.sample_data = v->last_waveform->data;
 514         if (v->last_waveform->levels)
 515         {
 516             gboolean use_cached = v->last_level > 0 && v->last_level < v->last_waveform->level_count
 517                 && freq64 > v->last_level_min_rate && freq64 <= v->last_waveform->levels[v->last_level].max_rate;
 518             if (__builtin_expect(use_cached, 1))
 519             {
 520                 v->gen.sample_data = v->last_waveform->levels[v->last_level].data;
 521                 bandlimited = TRUE;
 522             }
 523             else
 524             {
 525                 for (int i = 0; i < v->last_waveform->level_count; i++)
 526                 {
 527                     if (freq64 <= v->last_waveform->levels[i].max_rate)
 528                     {
 529                         v->last_level = i;
 530                         v->gen.sample_data = v->last_waveform->levels[i].data;
 531                         bandlimited = TRUE;
 532
 533                         break;
 534                     }
 535                     v->last_level_min_rate = v->last_waveform->levels[i].max_rate;
 536                 }
 537             }
 538         }
 539     }
 540
 541     gboolean play_loop = v->layer->loop_end && (v->loop_mode == slm_loop_continuous || playing_sustain_loop) && v->layer->on_cc_number == -1;
 542     loop_start = play_loop ? v->layer->loop_start : (v->layer->count ? 0 : (uint32_t)-1);
 543     loop_end = play_loop ? v->layer->loop_end : v->gen.cur_sample_end;
 544
 545     if (v->current_pipe)
 546     {
 547         v->gen.sample_data = v->gen.loop_count ? v->current_pipe->data : v->last_waveform->data;
 548         v->gen.streaming_buffer = v->current_pipe->data;
 549
 550         v->gen.prefetch_only_loop = (loop_end < v->last_waveform->preloaded_frames);
 551         v->gen.loop_overlap = 0;
 552         if (v->gen.prefetch_only_loop)
 553         {
 554             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)
 555             v->gen.loop_start = loop_start;
 556             v->gen.loop_end = loop_end;
 557             v->gen.streaming_buffer_frames = 0;
 558         }
 559         else
 560         {
 561             v->gen.loop_start = 0;
 562             v->gen.loop_end = v->last_waveform->preloaded_frames;
 563             v->gen.streaming_buffer_frames = v->current_pipe->buffer_loop_end;
 564         }
 565     }
 566     else
 567     {
 568         v->gen.loop_count = v->layer->count;
 569         v->gen.loop_start = loop_start;
 570         v->gen.loop_end = loop_end;
 571
 572         if (!bandlimited)
 573         {
 574             // Use pre-calculated join
 575             v->gen.scratch = loop_start == (uint32_t)-1 ? v->layer->scratch_end : v->layer->scratch_loop;
 576         }
 577         else
 578         {
 579             // The standard waveforms have extra MAX_INTERPOLATION_ORDER of samples from the loop start added past loop_end,
 580             // to avoid wasting time generating the joins in all the practical cases. The slow path covers custom loops
 581             // (i.e. partial loop or no loop) over bandlimited versions of the standard waveforms, and those are probably
 582             // not very useful anyway, as changing the loop removes the guarantee of the waveform being bandlimited and
 583             // may cause looping artifacts or introduce DC offset (e.g. if only a positive part of a sine wave is looped).
 584             if (loop_start == 0 && loop_end == l->eff_waveform->info.frames)
 585                 v->gen.scratch = v->gen.sample_data + l->eff_waveform->info.frames - MAX_INTERPOLATION_ORDER;
 586             else
 587             {
 588                 // Generate the join for the current wave level
 589                 // XXXKF this could be optimised further, by checking if waveform and loops are the same as the last
 590                 // time. However, this code is not likely to be used... ever, so optimising it is not the priority.
 591                 int shift = l->eff_waveform->info.channels == 2 ? 1 : 0;
 592                 uint32_t halfscratch = MAX_INTERPOLATION_ORDER << shift;
 593
 594                 v->gen.scratch = v->gen.scratch_bandlimited;
 595                 memcpy(&v->gen.scratch_bandlimited[0], &v->gen.sample_data[(loop_end - MAX_INTERPOLATION_ORDER) << shift], halfscratch * sizeof(int16_t) );
 596                 if (loop_start != (uint32_t)-1)
 597                     memcpy(v->gen.scratch_bandlimited + halfscratch, &v->gen.sample_data[loop_start << shift], halfscratch * sizeof(int16_t));
 598                 else
 599                     memset(v->gen.scratch_bandlimited + halfscratch, 0, halfscratch * sizeof(int16_t));
 600             }
 601         }
 602     }
 603
 604
 605     v->gen.bigdelta = freq64;
 606     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);
 607     if (moddests[smdest_gain] != 0.f)
 608         gain *= dB2gain(moddests[smdest_gain]);
 609     // http://drealm.info/sfz/plj-sfz.xhtml#amp "The overall gain must remain in the range -144 to 6 decibels."
 610     if (gain > 2.f)
 611         gain = 2.f;
 612     float pan = (l->pan + 100.f) * (1.f / 200.f) + (sampler_channel_addcc(c, 10) * 1.f / maxv - 0.5f) * 2.f;
 613     if (pan < 0.f)
 614         pan = 0.f;
 615     if (pan > 1.f)
 616         pan = 1.f;
 617     v->gen.lgain = gain * (1.f - pan)  / 32768.f;
 618     v->gen.rgain = gain * pan / 32768.f;
 619     struct cbox_biquadf_coeffs *second_filter = &v->filter_coeffs;
 620     gboolean is4p = sampler_layer_data_is_4pole(v->layer);
 621     if (l->cutoff != -1.f)
 622     {
 623         float logcutoff = l->logcutoff + moddests[smdest_cutoff];
 624         if (logcutoff < 0)
 625             logcutoff = 0;
 626         if (logcutoff > 12798)
 627             logcutoff = 12798;
 628         //float resonance = v->resonance*pow(32.0,c->cc[71]/maxv);
 629         float resonance = l->resonance_linearized * dB2gain((is4p ? 0.5 : 1) * moddests[smdest_resonance]);
 630         if (resonance < 0.7f)
 631             resonance = 0.7f;
 632         if (resonance > 32.f)
 633             resonance = 32.f;
 634         switch(l->fil_type)
 635         {
 636         case sft_lp24hybrid:
 637             cbox_biquadf_set_lp_rbj_lookup(&v->filter_coeffs, &m->sincos[(int)logcutoff], resonance * resonance);
 638             cbox_biquadf_set_1plp_lookup(&v->filter_coeffs_extra, &m->sincos[(int)logcutoff], 1);
 639             second_filter = &v->filter_coeffs_extra;
 640             break;
 641
 642         case sft_lp12:
 643         case sft_lp24:
 644             cbox_biquadf_set_lp_rbj_lookup(&v->filter_coeffs, &m->sincos[(int)logcutoff], resonance);
 645             break;
 646         case sft_hp12:
 647         case sft_hp24:
 648             cbox_biquadf_set_hp_rbj_lookup(&v->filter_coeffs, &m->sincos[(int)logcutoff], resonance);
 649             break;
 650         case sft_bp6:
 651         case sft_bp12:
 652             cbox_biquadf_set_bp_rbj_lookup(&v->filter_coeffs, &m->sincos[(int)logcutoff], resonance);
 653             break;
 654         case sft_lp6:
 655         case sft_lp12nr:
 656         case sft_lp24nr:
 657             cbox_biquadf_set_1plp_lookup(&v->filter_coeffs, &m->sincos[(int)logcutoff], l->fil_type != sft_lp6);
 658             break;
 659         case sft_hp6:
 660         case sft_hp12nr:
 661         case sft_hp24nr:
 662             cbox_biquadf_set_1php_lookup(&v->filter_coeffs, &m->sincos[(int)logcutoff], l->fil_type != sft_hp6);
 663             break;
 664         default:
 665             assert(0);
 666         }
 667     }
 668
 669     float left[CBOX_BLOCK_SIZE], right[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, left, right, 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, left, right, (uint32_t)-1);
 689     }
 690
 691     for (int i = samples; i < CBOX_BLOCK_SIZE; i++)
 692         left[i] = right[i] = 0.f;
 693     if (l->cutoff != -1)
 694     {
 695         cbox_biquadf_process(&v->filter_left, &v->filter_coeffs, left);
 696         if (is4p)
 697             cbox_biquadf_process(&v->filter_left2, second_filter, left);
 698         cbox_biquadf_process(&v->filter_right, &v->filter_coeffs, right);
 699         if (is4p)
 700             cbox_biquadf_process(&v->filter_right2, second_filter, right);
 701     }
 702     if (__builtin_expect(l->eq_bitmask, 0))
 703     {
 704         for (int eq = 0; eq < 3; eq++)
 705         {
 706             if (l->eq_bitmask & (1 << eq))
 707             {
 708                 cbox_biquadf_process(&v->eq_left[eq], &v->eq_coeffs[eq], left);
 709                 cbox_biquadf_process(&v->eq_right[eq], &v->eq_coeffs[eq], right);
 710             }
 711         }
 712     }
 713
 714     mix_block_into(outputs, v->output_pair_no * 2, left, right);
 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, left, right, 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, left, right, v->send2gain);
 726         }
 727     }
 728     if (v->gen.mode == spt_inactive)
 729         sampler_voice_inactivate(v, FALSE);
 730 }
 731