src/modules_filter.cpp

   1 /* Calf DSP plugin pack
   2  * Equalization related plugins
   3  *
   4  * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
   5  *
   6  * This program is free software; you can redistribute it and/or
   7  * modify it under the terms of the GNU Lesser General Public
   8  * License as published by the Free Software Foundation; either
   9  * version 2 of the License, or (at your option) any later version.
  10  *
  11  * This program is distributed in the hope that it will be useful,
  12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  14  * Lesser General Public License for more details.
  15  *
  16  * You should have received a copy of the GNU Lesser General
  17  * Public License along with this program; if not, write to the
  18  * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
  19  * Boston, MA  02110-1301  USA
  20  */
  21 #include <limits.h>
  22 #include <memory.h>
  23 #include <calf/giface.h>
  24 #include <calf/modules_filter.h>
  25
  26 using namespace dsp;
  27 using namespace calf_plugins;
  28
  29 #define SET_IF_CONNECTED(name) if (params[AM::param_##name] != NULL) *params[AM::param_##name] = name;
  30
  31 /**********************************************************************
  32  * EQUALIZER N BAND by Markus Schmidt and Krzysztof Foltman
  33 **********************************************************************/
  34
  35 inline void diff_ms(float &left, float &right) {
  36     float tmp = (left + right) / 2;
  37     right = left - right;
  38     left = tmp;
  39 }
  40 inline void undiff_ms(float &left, float &right) {
  41     float tmp = left + right / 2;
  42     right = left - right / 2;
  43     left = tmp;
  44 }
  45
  46 template<class BaseClass, bool has_lphp>
  47 equalizerNband_audio_module<BaseClass, has_lphp>::equalizerNband_audio_module()
  48 {
  49     is_active = false;
  50     srate = 0;
  51     last_generation = 0;
  52     hp_freq_old = lp_freq_old = 0;
  53     hs_freq_old = ls_freq_old = 0;
  54     hs_level_old = ls_level_old = 0;
  55     keep_gliding = 0;
  56     last_peak = 0;
  57     indiv_old = -1;
  58     analyzer_old = false;
  59     for (int i = 0; i < AM::PeakBands; i++)
  60     {
  61         p_freq_old[i] = 0;
  62         p_level_old[i] = 0;
  63         p_q_old[i] = 0;
  64     }
  65     for (int i = 0; i < graph_param_count; i++)
  66         old_params_for_graph[i] = -1;
  67     redraw_graph = true;
  68 }
  69
  70 template<class BaseClass, bool has_lphp>
  71 void equalizerNband_audio_module<BaseClass, has_lphp>::activate()
  72 {
  73     is_active = true;
  74     // set all filters
  75     params_changed();
  76 }
  77
  78 template<class BaseClass, bool has_lphp>
  79 void equalizerNband_audio_module<BaseClass, has_lphp>::deactivate()
  80 {
  81     is_active = false;
  82 }
  83
  84 static inline void copy_lphp(biquad_d2 filters[3][2])
  85 {
  86     for (int i = 0; i < 3; i++)
  87         for (int j = 0; j < 2; j++)
  88             if (i || j)
  89                 filters[i][j].copy_coeffs(filters[0][0]);
  90 }
  91
  92 static inline double glide(double value, double target, int &keep_gliding)
  93 {
  94     if (target == value)
  95         return value;
  96     keep_gliding = 1;
  97     if (target > value)
  98         return std::min(target, (value + 0.1) * 1.003);
  99     else
 100         return std::max(target, (value / 1.003) - 0.1);
 101 }
 102
 103 template<class BaseClass, bool has_lphp>
 104 void equalizerNband_audio_module<BaseClass, has_lphp>::params_changed()
 105 {
 106     keep_gliding = 0;
 107     // set the params of all filters
 108
 109     // lp/hp first (if available)
 110     if (has_lphp)
 111     {
 112         hp_mode = (CalfEqMode)(int)*params[AM::param_hp_mode];
 113         lp_mode = (CalfEqMode)(int)*params[AM::param_lp_mode];
 114
 115         float hpfreq = *params[AM::param_hp_freq], lpfreq = *params[AM::param_lp_freq];
 116
 117         if(hpfreq != hp_freq_old) {
 118             hpfreq = glide(hp_freq_old, hpfreq, keep_gliding);
 119             hp[0][0].set_hp_rbj(hpfreq, 0.707, (float)srate, 1.0);
 120             copy_lphp(hp);
 121             hp_freq_old = hpfreq;
 122         }
 123         if(lpfreq != lp_freq_old) {
 124             lpfreq = glide(lp_freq_old, lpfreq, keep_gliding);
 125             lp[0][0].set_lp_rbj(lpfreq, 0.707, (float)srate, 1.0);
 126             copy_lphp(lp);
 127             lp_freq_old = lpfreq;
 128         }
 129     }
 130
 131     // then shelves
 132     float hsfreq = *params[AM::param_hs_freq], hslevel = *params[AM::param_hs_level];
 133     float lsfreq = *params[AM::param_ls_freq], lslevel = *params[AM::param_ls_level];
 134
 135     if(lsfreq != ls_freq_old or lslevel != ls_level_old) {
 136         lsfreq = glide(ls_freq_old, lsfreq, keep_gliding);
 137         lsL.set_lowshelf_rbj(lsfreq, 0.707, lslevel, (float)srate);
 138         lsR.copy_coeffs(lsL);
 139         ls_level_old = lslevel;
 140         ls_freq_old = lsfreq;
 141     }
 142     if(hsfreq != hs_freq_old or hslevel != hs_level_old) {
 143         hsfreq = glide(hs_freq_old, hsfreq, keep_gliding);
 144         hsL.set_highshelf_rbj(hsfreq, 0.707, hslevel, (float)srate);
 145         hsR.copy_coeffs(hsL);
 146         hs_level_old = hslevel;
 147         hs_freq_old = hsfreq;
 148     }
 149     for (int i = 0; i < AM::PeakBands; i++)
 150     {
 151         int offset = i * params_per_band;
 152         float freq = *params[AM::param_p1_freq + offset];
 153         float level = *params[AM::param_p1_level + offset];
 154         float q = *params[AM::param_p1_q + offset];
 155         if(freq != p_freq_old[i] or level != p_level_old[i] or q != p_q_old[i]) {
 156             freq = glide(p_freq_old[i], freq, keep_gliding);
 157             pL[i].set_peakeq_rbj(freq, q, level, (float)srate);
 158             pR[i].copy_coeffs(pL[i]);
 159             p_freq_old[i] = freq;
 160             p_level_old[i] = level;
 161             p_q_old[i] = q;
 162         }
 163     }
 164     if (*params[AM::param_individuals] != indiv_old) {
 165         indiv_old = *params[AM::param_individuals];
 166         redraw_graph = true;
 167     }
 168
 169     // check if any important parameter for redrawing the graph changed
 170     for (int i = 0; i < graph_param_count; i++) {
 171         if (*params[AM::first_graph_param + i] != old_params_for_graph[i])
 172             redraw_graph = true;
 173         old_params_for_graph[i] = *params[AM::first_graph_param + i];
 174     }
 175
 176     _analyzer.set_params(
 177         256, 1, 6, 0, 1,
 178         *params[AM::param_analyzer_mode] + (*params[AM::param_analyzer_mode] >= 3 ? 5 : 1),
 179         0, 0, 15, 2, 0, 0
 180     );
 181
 182     if ((bool)*params[AM::param_analyzer_active] != analyzer_old) {
 183         redraw_graph = true;
 184         analyzer_old = (bool)*params[AM::param_analyzer_active];
 185     }
 186 }
 187
 188 template<class BaseClass, bool has_lphp>
 189 inline void equalizerNband_audio_module<BaseClass, has_lphp>::process_hplp(float &left, float &right)
 190 {
 191     if (!has_lphp)
 192         return;
 193     int active = *params[AM::param_lp_active];
 194     if (active > 0.f)
 195     {
 196         if (active > 3) diff_ms(left, right);
 197         switch(lp_mode)
 198         {
 199             case MODE12DB:
 200                 if (active == 1 or active == 2 or active == 4)
 201                     left = lp[0][0].process(left);
 202                 if (active == 1 or active == 3 or active == 5)
 203                     right = lp[0][1].process(right);
 204                 break;
 205             case MODE24DB:
 206                 if (active == 1 or active == 2 or active == 4)
 207                     left = lp[1][0].process(lp[0][0].process(left));
 208                 if (active == 1 or active == 3 or active == 5)
 209                     right = lp[1][1].process(lp[0][1].process(right));
 210                 break;
 211             case MODE36DB:
 212                 if (active == 1 or active == 2 or active == 4)
 213                     left = lp[2][0].process(lp[1][0].process(lp[0][0].process(left)));
 214                 if (active == 1 or active == 3 or active == 5)
 215                     right = lp[2][1].process(lp[1][1].process(lp[0][1].process(right)));
 216                 break;
 217         }
 218         if (active > 3) undiff_ms(left, right);
 219     }
 220     active = *params[AM::param_hp_active];
 221     if (active > 0.f)
 222     {
 223         if (active > 3) diff_ms(left, right);
 224         switch(hp_mode)
 225         {
 226             case MODE12DB:
 227                 if (active == 1 or active == 2 or active == 4)
 228                     left = hp[0][0].process(left);
 229                 if (active == 1 or active == 3 or active == 5)
 230                     right = hp[0][1].process(right);
 231                 break;
 232             case MODE24DB:
 233                 if (active == 1 or active == 2 or active == 4)
 234                     left = hp[1][0].process(hp[0][0].process(left));
 235                 if (active == 1 or active == 3 or active == 5)
 236                     right = hp[1][1].process(hp[0][1].process(right));
 237                 break;
 238             case MODE36DB:
 239                 if (active == 1 or active == 2 or active == 4)
 240                     left = hp[2][0].process(hp[1][0].process(hp[0][0].process(left)));
 241                 if (active == 1 or active == 3 or active == 5)
 242                     right = hp[2][1].process(hp[1][1].process(hp[0][1].process(right)));
 243                 break;
 244         }
 245         if (active > 3) undiff_ms(left, right);
 246     }
 247 }
 248
 249 template<class BaseClass, bool has_lphp>
 250 uint32_t equalizerNband_audio_module<BaseClass, has_lphp>::process(uint32_t offset, uint32_t numsamples, uint32_t inputs_mask, uint32_t outputs_mask)
 251 {
 252     bool bypass = *params[AM::param_bypass] > 0.f;
 253     if (keep_gliding && !bypass)
 254     {
 255         // ensure that if params have changed, the params_changed method is
 256         // called every 8 samples to interpolate filter parameters
 257         while(numsamples > 8 && keep_gliding)
 258         {
 259             params_changed();
 260             outputs_mask |= process(offset, 8, inputs_mask, outputs_mask);
 261             offset += 8;
 262             numsamples -= 8;
 263         }
 264         if (keep_gliding)
 265             params_changed();
 266     }
 267     numsamples += offset;
 268     if(bypass) {
 269         // everything bypassed
 270         while(offset < numsamples) {
 271             outs[0][offset] = ins[0][offset];
 272             outs[1][offset] = ins[1][offset];
 273             float values[] = {0, 0, 0, 0};
 274             meters.process(values);
 275             _analyzer.process(0, 0);
 276             ++offset;
 277         }
 278     } else {
 279         // process
 280         while(offset < numsamples) {
 281             // cycle through samples
 282             float outL = 0.f;
 283             float outR = 0.f;
 284             float inL = ins[0][offset];
 285             float inR = ins[1][offset];
 286             // in level
 287             inR *= *params[AM::param_level_in];
 288             inL *= *params[AM::param_level_in];
 289
 290             float procL = inL;
 291             float procR = inR;
 292
 293             // all filters in chain
 294             process_hplp(procL, procR);
 295
 296             int active = *params[AM::param_ls_active];
 297             if (active > 3) diff_ms(procL, procR);
 298             if (active == 1 or active == 2 or active == 4)
 299                 procL = lsL.process(procL);
 300             if (active == 1 or active == 3 or active == 5)
 301                 procR = lsR.process(procR);
 302             if (active > 3) undiff_ms(procL, procR);
 303
 304             active = *params[AM::param_hs_active];
 305             if (active > 3) diff_ms(procL, procR);
 306             if (active == 1 or active == 2 or active == 4)
 307                 procL = hsL.process(procL);
 308             if (active == 1 or active == 3 or active == 5)
 309                 procR = hsR.process(procR);
 310             if (active > 3) undiff_ms(procL, procR);
 311
 312             for (int i = 0; i < AM::PeakBands; i++)
 313             {
 314                 int offset = i * params_per_band;
 315                 int active = *params[AM::param_p1_active + offset];
 316                 if (active > 3) diff_ms(procL, procR);
 317                 if (active == 1 or active == 2 or active == 4)
 318                     procL = pL[i].process(procL);
 319                 if (active == 1 or active == 3 or active == 5)
 320                     procR = pR[i].process(procR);
 321                 if (active > 3) undiff_ms(procL, procR);
 322             }
 323
 324             outL = procL * *params[AM::param_level_out];
 325             outR = procR * *params[AM::param_level_out];
 326
 327             // analyzer
 328             _analyzer.process((inL + inR) / 2.f, (outL + outR) / 2.f);
 329
 330             // send to output
 331             outs[0][offset] = outL;
 332             outs[1][offset] = outR;
 333
 334             float values[] = {inL, inR, outL, outR};
 335             meters.process(values);
 336
 337             // next sample
 338             ++offset;
 339         } // cycle trough samples
 340         // clean up
 341         for(int i = 0; i < 3; ++i) {
 342             hp[i][0].sanitize();
 343             hp[i][1].sanitize();
 344             lp[i][0].sanitize();
 345             lp[i][1].sanitize();
 346         }
 347         lsL.sanitize();
 348         hsR.sanitize();
 349         for(int i = 0; i < AM::PeakBands; ++i) {
 350             pL[i].sanitize();
 351             pR[i].sanitize();
 352         }
 353     }
 354     meters.fall(numsamples);
 355     return outputs_mask;
 356 }
 357
 358 static inline float adjusted_lphp_gain(const float *const *params, int param_active, int param_mode, const biquad_d2 &filter, float freq, float srate)
 359 {
 360     if(*params[param_active] > 0.f) {
 361         float gain = filter.freq_gain(freq, srate);
 362         switch((int)*params[param_mode]) {
 363             case MODE12DB:
 364                 return gain;
 365             case MODE24DB:
 366                 return gain * gain;
 367             case MODE36DB:
 368                 return gain * gain * gain;
 369         }
 370     }
 371     return 1;
 372 }
 373
 374 template<class BaseClass, bool has_lphp>
 375 bool equalizerNband_audio_module<BaseClass, has_lphp>::get_graph(int index, int subindex, int phase, float *data, int points, cairo_iface *context, int *mode) const
 376 {
 377     if (phase and *params[AM::param_analyzer_active]) {
 378         bool r = _analyzer.get_graph(subindex, phase, data, points, context, mode);
 379         if (*params[AM::param_analyzer_mode] == 2) {
 380             set_channel_color(context, subindex ? 0 : 1, 0.15);
 381         } else {
 382             context->set_source_rgba(0,0,0,0.1);
 383         }
 384         return r;
 385     } else if (phase and !*params[AM::param_analyzer_active]) {
 386         redraw_graph = false;
 387         return false;
 388     } else {
 389         int max = PeakBands + 2 + (has_lphp ? 2 : 0);
 390
 391         if (!is_active
 392         or (subindex and !*params[AM::param_individuals])
 393         or (subindex > max and *params[AM::param_individuals])) {
 394             redraw_graph = false;
 395             return false;
 396         }
 397
 398         // first graph is the overall frequency response graph
 399         if (!subindex)
 400             return ::get_graph(*this, subindex, data, points, 128 * *params[AM::param_zoom], 0);
 401
 402         // get out if max band is reached
 403         if (last_peak >= max) {
 404             last_peak = 0;
 405             redraw_graph = false;
 406             return false;
 407         }
 408
 409         // get the next filter to draw a curve for and leave out inactive
 410         // filters
 411         while (last_peak < PeakBands and !*params[AM::param_p1_active + last_peak * params_per_band])
 412             last_peak ++;
 413         if (last_peak == PeakBands and !*params[AM::param_ls_active])
 414             last_peak ++;
 415         if (last_peak == PeakBands + 1 and !*params[AM::param_hs_active])
 416             last_peak ++;
 417         if (has_lphp and last_peak == PeakBands + 2 and !*params[AM::param_hp_active])
 418             last_peak ++;
 419         if (has_lphp and last_peak == PeakBands + 3 and !*params[AM::param_lp_active])
 420             last_peak ++;
 421
 422         // get out if max band is reached
 423         if (last_peak >= max) { // and !*params[param_analyzer_active]) {
 424             last_peak = 0;
 425             redraw_graph = false;
 426             return false;
 427         }
 428          //else if *params[param_analyzer_active]) {
 429             //bool goon = _analyzer.get_graph(subindex, phase, data, points, context, mode);
 430             //if (!goon)
 431                 //last_peak = 0;
 432             //return goon;
 433         //}
 434
 435         // draw the individual curve of the actual filter
 436         for (int i = 0; i < points; i++) {
 437             double freq = 20.0 * pow (20000.0 / 20.0, i * 1.0 / points);
 438             if (last_peak < PeakBands) {
 439                 data[i] = pL[last_peak].freq_gain(freq, (float)srate);
 440             } else if (last_peak == PeakBands) {
 441                 data[i] = lsL.freq_gain(freq, (float)srate);
 442             } else if (last_peak == PeakBands + 1) {
 443                 data[i] = hsL.freq_gain(freq, (float)srate);
 444             } else if (last_peak == PeakBands + 2 and has_lphp) {
 445                 data[i] = adjusted_lphp_gain(params, AM::param_hp_active, AM::param_hp_mode, hp[0][0], freq, (float)srate);
 446             } else if (last_peak == PeakBands + 3 and has_lphp) {
 447                 data[i] = adjusted_lphp_gain(params, AM::param_lp_active, AM::param_lp_mode, lp[0][0], freq, (float)srate);
 448             }
 449             data[i] = dB_grid(data[i], 128 * *params[AM::param_zoom], 0);
 450         }
 451
 452         last_peak ++;
 453         *mode = 4;
 454         context->set_source_rgba(0,0,0,0.075);
 455         return true;
 456     }
 457 }
 458 template<class BaseClass, bool has_lphp>
 459 bool equalizerNband_audio_module<BaseClass, has_lphp>::get_layers(int index, int generation, unsigned int &layers) const
 460 {
 461     redraw_graph = redraw_graph || !generation;
 462     layers = *params[AM::param_analyzer_active] ? LG_REALTIME_GRAPH : 0;
 463     layers |= (generation ? LG_NONE : LG_CACHE_GRID) | (redraw_graph ? LG_CACHE_GRAPH : LG_NONE);
 464     redraw_graph |= (bool)*params[AM::param_analyzer_active];
 465     return redraw_graph or !generation;
 466 }
 467
 468 template<class BaseClass, bool has_lphp>
 469 bool equalizerNband_audio_module<BaseClass, has_lphp>::get_gridline(int index, int subindex, int phase, float &pos, bool &vertical, std::string &legend, cairo_iface *context) const
 470 {
 471     if (!is_active or phase)
 472         return false;
 473     return get_freq_gridline(subindex, pos, vertical, legend, context, true, 128 * *params[AM::param_zoom], 0);
 474 }
 475
 476 template<class BaseClass, bool has_lphp>
 477 float equalizerNband_audio_module<BaseClass, has_lphp>::freq_gain(int index, double freq) const
 478 {
 479     float ret = 1.f;
 480     if (has_lphp)
 481     {
 482         ret *= adjusted_lphp_gain(params, AM::param_hp_active, AM::param_hp_mode, hp[0][0], freq, (float)srate);
 483         ret *= adjusted_lphp_gain(params, AM::param_lp_active, AM::param_lp_mode, lp[0][0], freq, (float)srate);
 484     }
 485     ret *= (*params[AM::param_ls_active] > 0.f) ? lsL.freq_gain(freq, (float)srate) : 1;
 486     ret *= (*params[AM::param_hs_active] > 0.f) ? hsL.freq_gain(freq, (float)srate) : 1;
 487     for (int i = 0; i < PeakBands; i++)
 488         ret *= (*params[AM::param_p1_active + i * params_per_band] > 0.f) ? pL[i].freq_gain(freq, (float)srate) : 1;
 489     return ret;
 490 }
 491
 492 template class equalizerNband_audio_module<equalizer5band_metadata, false>;
 493 template class equalizerNband_audio_module<equalizer8band_metadata, true>;
 494 template class equalizerNband_audio_module<equalizer12band_metadata, true>;
 495
 496
 497 /**********************************************************************
 498  * FILTERKLAVIER by Hans Baier
 499 **********************************************************************/
 500
 501 filterclavier_audio_module::filterclavier_audio_module()
 502 : filter_module_with_inertia<biquad_filter_module, filterclavier_metadata>(ins, outs, params)
 503 , min_gain(1.0)
 504 , max_gain(32.0)
 505 , last_note(-1)
 506 , last_velocity(-1)
 507 {
 508 }
 509
 510 void filterclavier_audio_module::params_changed()
 511 {
 512     inertia_filter_module::inertia_cutoff.set_inertia(
 513         note_to_hz(last_note + *params[par_transpose], *params[par_detune]));
 514
 515     float min_resonance = param_props[par_max_resonance].min;
 516      inertia_filter_module::inertia_resonance.set_inertia(
 517              (float(last_velocity) / 127.0)
 518              // 0.001: see below
 519              * (*params[par_max_resonance] - min_resonance + 0.001)
 520              + min_resonance);
 521
 522     adjust_gain_according_to_filter_mode(last_velocity);
 523
 524     inertia_filter_module::calculate_filter();
 525     redraw_graph = true;
 526 }
 527
 528 void filterclavier_audio_module::activate()
 529 {
 530     inertia_filter_module::activate();
 531 }
 532
 533 void filterclavier_audio_module::set_sample_rate(uint32_t sr)
 534 {
 535     inertia_filter_module::set_sample_rate(sr);
 536 }
 537
 538 void filterclavier_audio_module::deactivate()
 539 {
 540     inertia_filter_module::deactivate();
 541 }
 542
 543
 544 void filterclavier_audio_module::note_on(int channel, int note, int vel)
 545 {
 546     last_note     = note;
 547     last_velocity = vel;
 548     inertia_filter_module::inertia_cutoff.set_inertia(
 549             note_to_hz(note + *params[par_transpose], *params[par_detune]));
 550
 551     float min_resonance = param_props[par_max_resonance].min;
 552     inertia_filter_module::inertia_resonance.set_inertia(
 553             (float(vel) / 127.0)
 554             // 0.001: if the difference is equal to zero (which happens
 555             // when the max_resonance knom is at minimum position
 556             // then the filter gain doesnt seem to snap to zero on most note offs
 557             * (*params[par_max_resonance] - min_resonance + 0.001)
 558             + min_resonance);
 559
 560     adjust_gain_according_to_filter_mode(vel);
 561
 562     inertia_filter_module::calculate_filter();
 563     redraw_graph = true;
 564 }
 565
 566 void filterclavier_audio_module::note_off(int channel, int note, int vel)
 567 {
 568     if (note == last_note) {
 569         inertia_filter_module::inertia_resonance.set_inertia(param_props[par_max_resonance].min);
 570         inertia_filter_module::inertia_gain.set_inertia(min_gain);
 571         inertia_filter_module::calculate_filter();
 572         last_velocity = 0;
 573         redraw_graph = true;
 574     }
 575 }
 576
 577 void filterclavier_audio_module::adjust_gain_according_to_filter_mode(int velocity)
 578 {
 579     int   mode = dsp::fastf2i_drm(*params[par_mode]);
 580
 581     // for bandpasses: boost gain for velocities > 0
 582     if ( (mode_6db_bp <= mode) && (mode <= mode_18db_bp) ) {
 583         // gain for velocity 0:   1.0
 584         // gain for velocity 127: 32.0
 585         float mode_max_gain = max_gain;
 586         // max_gain is right for mode_6db_bp
 587         if (mode == mode_12db_bp)
 588             mode_max_gain /= 6.0;
 589         if (mode == mode_18db_bp)
 590             mode_max_gain /= 10.5;
 591
 592         inertia_filter_module::inertia_gain.set_now(
 593                 (float(velocity) / 127.0) * (mode_max_gain - min_gain) + min_gain);
 594     } else {
 595         inertia_filter_module::inertia_gain.set_now(min_gain);
 596     }
 597 }
 598
 599 /**********************************************************************
 600  * EMPHASIS by Damien Zammit
 601 **********************************************************************/
 602
 603 emphasis_audio_module::emphasis_audio_module()
 604 {
 605     is_active = false;
 606     srate = 0;
 607     redraw_graph = true;
 608     mode = -1;
 609     type = -1;
 610 }
 611
 612 void emphasis_audio_module::activate()
 613 {
 614     is_active = true;
 615     // set all filters
 616     params_changed();
 617 }
 618
 619 void emphasis_audio_module::deactivate()
 620 {
 621     is_active = false;
 622 }
 623
 624 void emphasis_audio_module::params_changed()
 625 {
 626     if (mode != *params[param_mode] or type != *params[param_type] or bypass != *params[param_bypass])
 627         redraw_graph = true;
 628     mode   = *params[param_mode];
 629     type   = *params[param_type];
 630     bypass = *params[param_bypass];
 631     riaacurvL.set(srate, mode, type);
 632     riaacurvR.set(srate, mode, type);
 633 }
 634
 635 uint32_t emphasis_audio_module::process(uint32_t offset, uint32_t numsamples, uint32_t inputs_mask, uint32_t outputs_mask)
 636 {
 637     uint32_t orig_numsamples = numsamples;
 638     bool bypass = *params[param_bypass] > 0.f;
 639     if (!bypass)
 640     {
 641         // ensure that if params have changed, the params_changed method is
 642         // called every 8 samples to interpolate filter parameters
 643         while(numsamples > 8)
 644         {
 645             params_changed();
 646             outputs_mask |= process(offset, 8, inputs_mask, outputs_mask);
 647             offset += 8;
 648             numsamples -= 8;
 649         }
 650     }
 651     numsamples += offset;
 652     if(bypass) {
 653         // everything bypassed
 654         while(offset < numsamples) {
 655             outs[0][offset] = ins[0][offset];
 656             outs[1][offset] = ins[1][offset];
 657             float values[] = {0, 0, 0, 0};
 658             meters.process(values);
 659             ++offset;
 660         }
 661     } else {
 662         // process
 663         while(offset < numsamples) {
 664             // cycle through samples
 665             float outL = 0.f;
 666             float outR = 0.f;
 667             float inL = ins[0][offset];
 668             float inR = ins[1][offset];
 669             // in level
 670             inR *= *params[param_level_in];
 671             inL *= *params[param_level_in];
 672
 673             float procL = inL;
 674             float procR = inR;
 675
 676             procL = riaacurvL.process(procL);
 677             procR = riaacurvR.process(procR);
 678
 679             outL = procL * *params[param_level_out];
 680             outR = procR * *params[param_level_out];
 681
 682             // send to output
 683             outs[0][offset] = outL;
 684             outs[1][offset] = outR;
 685
 686             float values[] = {inL, inR, outL, outR};
 687             meters.process(values);
 688
 689             // next sample
 690             ++offset;
 691         } // cycle trough samples
 692         // clean up
 693         riaacurvL.sanitize();
 694         riaacurvR.sanitize();
 695     }
 696     meters.fall(orig_numsamples);
 697     return outputs_mask;
 698 }
 699 bool emphasis_audio_module::get_graph(int index, int subindex, int phase, float *data, int points, cairo_iface *context, int *mode) const
 700 {
 701     if (phase or subindex)
 702         return false;
 703     if (bypass)
 704         context->set_source_rgba(0.15, 0.2, 0.0, 0.3);
 705     return ::get_graph(*this, subindex, data, points, 32, 0);
 706 }
 707 bool emphasis_audio_module::get_gridline(int index, int subindex, int phase, float &pos, bool &vertical, std::string &legend, cairo_iface *context) const
 708 {
 709     if (phase)
 710         return false;
 711     return get_freq_gridline(subindex, pos, vertical, legend, context, true, 32, 0);
 712 }
 713
 714 /**********************************************************************
 715  * CROSSOVER N BAND by Markus Schmidt
 716 **********************************************************************/
 717
 718 template<class XoverBaseClass>
 719 xover_audio_module<XoverBaseClass>::xover_audio_module()
 720 {
 721     is_active = false;
 722     srate = 0;
 723     redraw_graph = true;
 724     crossover.init(AM::channels, AM::bands, 44100);
 725 }
 726 template<class XoverBaseClass>
 727 xover_audio_module<XoverBaseClass>::~xover_audio_module()
 728 {
 729     free(buffer);
 730 }
 731 template<class XoverBaseClass>
 732 void xover_audio_module<XoverBaseClass>::activate()
 733 {
 734     is_active = true;
 735     params_changed();
 736 }
 737
 738 template<class XoverBaseClass>
 739 void xover_audio_module<XoverBaseClass>::deactivate()
 740 {
 741     is_active = false;
 742 }
 743 template<class XoverBaseClass>
 744 void xover_audio_module<XoverBaseClass>::set_sample_rate(uint32_t sr)
 745 {
 746     srate = sr;
 747     // set srate of crossover
 748     crossover.set_sample_rate(srate);
 749     // rebuild buffer
 750     buffer_size = (int)(srate / 10 * AM::channels * AM::bands + AM::channels * AM::bands); // buffer size attack rate multiplied by channels and bands
 751     buffer = (float*) calloc(buffer_size, sizeof(float));
 752     pos = 0;
 753     int amount = AM::bands * AM::channels + AM::channels;
 754     int meter[amount];
 755     int clip[amount];
 756     for(int b = 0; b < AM::bands; b++) {
 757         for (int c = 0; c < AM::channels; c++) {
 758             meter[b * AM::channels + c] = AM::param_meter_01 + b * params_per_band + c;
 759             clip[b * AM::channels + c] = -1;
 760         }
 761     }
 762     for (int c = 0; c < AM::channels; c++) {
 763         meter[c + AM::bands * AM::channels] = AM::param_meter_0 + c;
 764         clip[c + AM::bands * AM::channels] = -1;
 765     }
 766     meters.init(params, meter, clip, amount, srate);
 767 }
 768 template<class XoverBaseClass>
 769 void xover_audio_module<XoverBaseClass>::params_changed()
 770 {
 771     int mode = *params[AM::param_mode];
 772     crossover.set_mode(mode);
 773     for (int i = 0; i < AM::bands - 1; i++) {
 774         crossover.set_filter(i,  *params[AM::param_freq0 + i]);
 775     }
 776     for (int i = 0; i < AM::bands; i++) {
 777         int offset = i * params_per_band;
 778         crossover.set_level(i, *params[AM::param_level1 + offset]);
 779         crossover.set_active(i, *params[AM::param_active1 + offset] > 0.5);
 780     }
 781     redraw_graph = true;
 782 }
 783
 784 template<class XoverBaseClass>
 785 uint32_t xover_audio_module<XoverBaseClass>::process(uint32_t offset, uint32_t numsamples, uint32_t inputs_mask, uint32_t outputs_mask)
 786 {
 787     unsigned int targ = numsamples + offset;
 788     float xval;
 789     float values[AM::bands * AM::channels + AM::channels];
 790     while(offset < targ) {
 791         // cycle through samples
 792
 793         // level
 794         for (int c = 0; c < AM::channels; c++) {
 795             in[c] = ins[c][offset] * *params[AM::param_level];
 796         }
 797         crossover.process(in);
 798
 799         for (int b = 0; b < AM::bands; b++) {
 800             int nbuf = 0;
 801             int off = b * params_per_band;
 802             // calc position in delay buffer
 803             if (*params[AM::param_delay1 + off]) {
 804                 nbuf = srate * (fabs(*params[AM::param_delay1 + off]) / 1000.f) * AM::bands * AM::channels;
 805                 nbuf -= nbuf % (AM::bands * AM::channels);
 806             }
 807             for (int c = 0; c < AM::channels; c++) {
 808                 // define a pointer between 0 and channels * bands
 809                 int ptr = b * AM::channels + c;
 810
 811                 // get output from crossover module if active
 812                 xval = *params[AM::param_active1 + off] > 0.5 ? crossover.get_value(c, b) : 0.f;
 813
 814                 // fill delay buffer
 815                 buffer[pos + ptr] = xval;
 816
 817                 // get value from delay buffer if neccessary
 818                 if (*params[AM::param_delay1 + off])
 819                     xval = buffer[(pos - (int)nbuf + ptr + buffer_size) % buffer_size];
 820
 821                 // set value with phase to output
 822                 outs[ptr][offset] = *params[AM::param_phase1 + off] > 0.5 ? xval * -1 : xval;
 823
 824                 // band meters
 825                 values[b * AM::channels + c] = outs[ptr][offset];
 826             }
 827         }
 828         // in meters
 829         for (int c = 0; c < AM::channels; c++) {
 830             values[c + AM::bands * AM::channels] = ins[c][offset];
 831         }
 832         meters.process(values);
 833         // next sample
 834         ++offset;
 835         // delay buffer pos forward
 836         pos = (pos + AM::channels * AM::bands) % buffer_size;
 837
 838     } // cycle trough samples
 839     meters.fall(numsamples);
 840     return outputs_mask;
 841 }
 842
 843 template<class XoverBaseClass>
 844 bool xover_audio_module<XoverBaseClass>::get_graph(int index, int subindex, int phase, float *data, int points, cairo_iface *context, int *mode) const
 845 {
 846     return crossover.get_graph(subindex, phase, data, points, context, mode);
 847 }
 848 template<class XoverBaseClass>
 849 bool xover_audio_module<XoverBaseClass>::get_layers(int index, int generation, unsigned int &layers) const
 850 {
 851     return crossover.get_layers(index, generation, layers);
 852 }
 853
 854 template class xover_audio_module<xover2_metadata>;
 855 template class xover_audio_module<xover3_metadata>;
 856 template class xover_audio_module<xover4_metadata>;
 857
 858
 859 /**********************************************************************
 860  * Vocoder by Markus Schmidt and Christian Holschuh
 861 **********************************************************************/
 862
 863 vocoder_audio_module::vocoder_audio_module()
 864 {
 865     is_active = false;
 866     srate     = 0;
 867     attack    = 0;
 868     release   = 0;
 869     fcoeff    = 0;
 870     bands     = 0;
 871     bands_old = -1;
 872     order     = 0;
 873     order_old = -1;
 874     fcoeff    = log10(20.f);
 875     log2_     = log(2);
 876     memset(env_mods, 0, 32 * 2 * sizeof(double));
 877 }
 878
 879 void vocoder_audio_module::activate()
 880 {
 881     is_active = true;
 882 }
 883
 884 void vocoder_audio_module::deactivate()
 885 {
 886     is_active = false;
 887 }
 888
 889 void vocoder_audio_module::params_changed()
 890 {
 891     attack  = exp(log(0.01)/( *params[param_attack]  * srate * 0.001));
 892     release = exp(log(0.01)/( *params[param_release] * srate * 0.001));
 893
 894     bands = *params[param_bands];
 895     order = round((bands - 4) / 28.f * (maxorder / (*params[param_hiq] ? 1 : 2) - 1) + 1);
 896     if (bands != bands_old or order != order_old) {
 897         bands_old = bands;
 898         order_old = order;
 899         for (int i = 0; i < bands; i++) {
 900             // set all actually used filters
 901             detector[0][0][i].set_bp_rbj(pow(10, fcoeff + (0.5f + (float)i) * 3.f / (float)bands), pow(1, 1.0 / order), (double)srate);
 902             for (int j = 0; j < order; j++) {
 903                 if (j)
 904                     detector[0][j][i].copy_coeffs(detector[0][0][i]);
 905                 detector[1][j][i].copy_coeffs(detector[0][0][i]);
 906                 modulator[0][j][i].copy_coeffs(detector[0][0][i]);
 907                 modulator[1][j][i].copy_coeffs(detector[0][0][i]);
 908             }
 909         }
 910     }
 911     set_leds();
 912     _analyzer.set_params(256, 1, 6, 0, 1, 0, 0, 0, 15, 2, 0, 0);
 913     redraw_graph = true;
 914 }
 915
 916 int vocoder_audio_module::get_solo() const {
 917     for (int i = 0; i < bands; i++)
 918         if (*params[param_solo0 + i * band_params])
 919             return 1;
 920     return 0;
 921 }
 922 void vocoder_audio_module::set_leds() {
 923     int solo = get_solo();
 924     for (int i = 0; i < 32; i++) {
 925         float l = 0;
 926         if (i < bands)
 927             l = (!*params[param_solo0 + i * band_params] and solo) ? 1 : 0.5;
 928         if (*params[param_active0 + i * band_params] != l)
 929             redraw_graph = true;
 930         *params[param_active0 + i * band_params] = l;
 931     }
 932 }
 933
 934 uint32_t vocoder_audio_module::process(uint32_t offset, uint32_t numsamples, uint32_t inputs_mask, uint32_t outputs_mask)
 935 {
 936     uint32_t orig_numsamples = numsamples;
 937     bool bypass = *params[param_bypass] > 0.f;
 938     int solo = get_solo();
 939     numsamples += offset;
 940     float led[32] = {0};
 941     if(bypass) {
 942         // everything bypassed
 943         while(offset < numsamples) {
 944             outs[0][offset] = ins[0][offset];
 945             outs[1][offset] = ins[1][offset];
 946             float values[] = {0, 0, 0, 0, 0, 0};
 947             meters.process(values);
 948             ++offset;
 949         }
 950     } else {
 951         // process
 952         while(offset < numsamples) {
 953             // cycle through samples
 954             double outL = 0;
 955             double outR = 0;
 956             double pL   = 0;
 957             double pR   = 0;
 958
 959             // carrier with level
 960             double cL = ins[0][offset] * *params[param_carrier_in];
 961             double cR = ins[1][offset] * *params[param_carrier_in];
 962
 963             // modulator with level
 964             double mL = ins[2][offset] * *params[param_mod_in];
 965             double mR = ins[3][offset] * *params[param_mod_in];
 966
 967             // noise generator
 968             double nL = (float)rand() / (float)RAND_MAX;
 969             double nR = (float)rand() / (float)RAND_MAX;
 970
 971             for (int i = 0; i < bands; i++) {
 972                 double mL_ = mL;
 973                 double mR_ = mR;
 974                 double cL_ = cL + nL * *params[param_noise0 + i * band_params];
 975                 double cR_ = cR + nR * *params[param_noise0 + i * band_params];
 976
 977                 if ((solo and *params[param_solo0 + i * band_params]) or !solo) {
 978                     for (int j = 0; j < order; j++) {
 979                         // filter modulator
 980                         if (*params[param_link] > 0.5) {
 981                             mL_ = detector[0][j][i].process(std::max(mL_, mR_));
 982                             mR_ = mL_;
 983                         } else {
 984                             mL_ = detector[0][j][i].process(mL_);
 985                             mR_ = detector[1][j][i].process(mR_);
 986                         }
 987                         // filter carrier with noise
 988                         cL_ = modulator[0][j][i].process(cL_);
 989                         cR_ = modulator[1][j][i].process(cR_);
 990                     }
 991                     // level by envelope with levelling
 992                     cL_ *= env_mods[0][i] * order * 4;
 993                     cR_ *= env_mods[1][i] * order * 4;
 994
 995                     // add band volume setting
 996                     cL_ *= *params[param_volume0 + i * band_params];
 997                     cR_ *= *params[param_volume0 + i * band_params];
 998
 999                     // add filtered modulator
1000                     cL_ += mL_ * *params[param_mod0 + i * band_params];
1001                     cR_ += mR_ * *params[param_mod0 + i * band_params];
1002
1003                     // Balance
1004                     cL_ *= (*params[param_pan0 + i * band_params] > 0
1005                          ? -*params[param_pan0 + i * band_params] + 1 : 1);
1006                     cR_ *= (*params[param_pan0 + i * band_params] < 0
1007                           ? *params[param_pan0 + i * band_params] + 1 : 1);
1008
1009                     // add to outputs with proc level
1010                     pL += cL_ * *params[param_proc];
1011                     pR += cR_ * *params[param_proc];
1012                 }
1013                 // LED
1014                 if (*params[param_detectors] > 0.5)
1015                     if (env_mods[0][i] + env_mods[1][i] > led[i])
1016                         led[i] = env_mods[0][i] + env_mods[1][i];
1017
1018                 // advance envelopes
1019                 env_mods[0][i] = (fabs(mL_) > env_mods[0][i] ? attack : release) * (env_mods[0][i] - fabs(mL_)) + fabs(mL_);
1020                 env_mods[1][i] = (fabs(mR_) > env_mods[1][i] ? attack : release) * (env_mods[1][i] - fabs(mR_)) + fabs(mR_);
1021             }
1022
1023             outL = pL;
1024             outR = pR;
1025
1026             // dry carrier
1027             outL += cL * *params[param_carrier];
1028             outR += cR * *params[param_carrier];
1029
1030             // dry modulator
1031             outL += mL * *params[param_mod];
1032             outR += mR * *params[param_mod];
1033
1034             // analyzer
1035             switch ((int)*params[param_analyzer]) {
1036                 case 0:
1037                 default:
1038                     break;
1039                 case 1:
1040                     _analyzer.process((float)cL, (float)cR);
1041                     break;
1042                 case 2:
1043                     _analyzer.process((float)mL, (float)mR);
1044                     break;
1045                 case 3:
1046                     _analyzer.process((float)pL, (float)pR);
1047                     break;
1048                 case 4:
1049                     _analyzer.process((float)outL, (float)outR);
1050                     break;
1051             }
1052
1053             // out level
1054             outL *= *params[param_out];
1055             outR *= *params[param_out];
1056
1057             // send to outputs
1058             outs[0][offset] = outL;
1059             outs[1][offset] = outR;
1060
1061             // meters
1062             float values[] = {(float)cL, (float)cR, (float)mL, (float)mR, (float)outL, (float)outR};
1063             meters.process(values);
1064
1065             // next sample
1066             ++offset;
1067         } // cycle trough samples
1068         // clean up
1069         for (int i = 0; i < bands; i++) {
1070             for (int j = 0; j < order; j++) {
1071                 detector[0][j][i].sanitize();
1072                 detector[1][j][i].sanitize();
1073                 modulator[0][j][i].sanitize();
1074                 modulator[1][j][i].sanitize();
1075             }
1076         }
1077     }
1078
1079     // LED
1080     for (int i = 0; i < 32; i++) {
1081         float val = 0;
1082         if (*params[param_detectors] > 0.5)
1083             val = std::max(0.0, 1 + log((led[i] / 2) * order) / log2_ / 10);
1084         *params[param_level0 + i * band_params] = val;
1085     }
1086     meters.fall(orig_numsamples);
1087     return outputs_mask;
1088 }
1089 bool vocoder_audio_module::get_graph(int index, int subindex, int phase, float *data, int points, cairo_iface *context, int *mode) const
1090 {
1091     if (phase and *params[param_analyzer]) {
1092         if (subindex) {
1093             return false;
1094         }
1095         bool r = _analyzer.get_graph(subindex, phase, data, points, context, mode);
1096         context->set_source_rgba(0,0,0,0.25);
1097         return r;
1098     } else if (phase) {
1099         return false;
1100     } else {
1101         // quit
1102         if (subindex >= bands) {
1103             redraw_graph = false;
1104             return false;
1105         }
1106         int solo = get_solo();
1107         if (solo and !*params[param_solo0 + subindex * band_params])
1108             context->set_source_rgba(0,0,0,0.15);
1109         context->set_line_width(0.99);
1110         int drawn = 0;
1111         double fq = pow(10, fcoeff + (0.5f + (float)subindex) * 3.f / (float)bands);
1112         for (int i = 0; i < points; i++) {
1113             double freq = 20.0 * pow (20000.0 / 20.0, i * 1.0 / points);
1114             float level = 1;
1115             for (int j = 0; j < order; j++)
1116                 level *= detector[0][0][subindex].freq_gain(freq, srate);
1117             level *= *params[param_volume0 + subindex * band_params];
1118             data[i] = dB_grid(level, 256, 0.4);
1119             if (!drawn and freq > fq) {
1120                 drawn = 1;
1121                 char str[32];
1122                 sprintf(str, "%d", subindex + 1);
1123                 draw_cairo_label(context, str, i, context->size_y * (1 - (data[i] + 1) / 2.f), 0, 0, 0.5);
1124             }
1125         }
1126     }
1127     return true;
1128 }
1129 bool vocoder_audio_module::get_layers(int index, int generation, unsigned int &layers) const
1130 {
1131     redraw_graph = redraw_graph || !generation;
1132     layers = *params[param_analyzer] ? LG_REALTIME_GRAPH : 0;
1133     layers |= (generation ? LG_NONE : LG_CACHE_GRID) | (redraw_graph ? LG_CACHE_GRAPH : LG_NONE);
1134     redraw_graph |= (bool)*params[param_analyzer];
1135     return redraw_graph or !generation;
1136 }