src/monosynth.cpp

   1 /* Calf DSP Library
   2  * Example audio modules - monosynth
   3  *
   4  * Copyright (C) 2001-2007 Krzysztof Foltman
   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., 59 Temple Place, Suite 330,
  19  * Boston, MA 02111-1307, USA.
  20  */
  21 #include <assert.h>
  22 #include <memory.h>
  23 #include <config.h>
  24 #include <complex>
  25 #if USE_JACK
  26 #include <jack/jack.h>
  27 #endif
  28 #include <calf/giface.h>
  29 #include <calf/modules_synths.h>
  30
  31 using namespace dsp;
  32 using namespace calf_plugins;
  33 using namespace std;
  34
  35 float silence[4097];
  36
  37 void monosynth_audio_module::activate() {
  38     monosynth_audio_module::generate_waves();
  39     running = false;
  40     output_pos = 0;
  41     queue_note_on = -1;
  42     stop_count = 0;
  43     pitchbend = 1.f;
  44     filter.reset();
  45     filter2.reset();
  46     stack.clear();
  47 }
  48
  49 waveform_family<MONOSYNTH_WAVE_BITS> *monosynth_audio_module::waves;
  50
  51 void monosynth_audio_module::generate_waves()
  52 {
  53     float data[1 << MONOSYNTH_WAVE_BITS];
  54     bandlimiter<MONOSYNTH_WAVE_BITS> bl;
  55
  56     if (waves)
  57         return;
  58
  59     static waveform_family<MONOSYNTH_WAVE_BITS> waves_data[wave_count];
  60     waves = waves_data;
  61
  62     enum { S = 1 << MONOSYNTH_WAVE_BITS, HS = S / 2, QS = S / 4, QS3 = 3 * QS };
  63     float iQS = 1.0 / QS;
  64
  65
  66     // yes these waves don't have really perfect 1/x spectrum because of aliasing
  67     // (so what?)
  68     for (int i = 0 ; i < HS; i++)
  69         data[i] = (float)(i * 1.0 / HS),
  70         data[i + HS] = (float)(i * 1.0 / HS - 1.0f);
  71     waves[wave_saw].make(bl, data);
  72
  73     for (int i = 0 ; i < S; i++)
  74         data[i] = (float)(i < HS ? -1.f : 1.f);
  75     waves[wave_sqr].make(bl, data);
  76
  77     for (int i = 0 ; i < S; i++)
  78         data[i] = (float)(i < (64 * S / 2048)? -1.f : 1.f);
  79     waves[wave_pulse].make(bl, data);
  80
  81     // XXXKF sure this is a waste of space, this will be fixed some day by better bandlimiter
  82     for (int i = 0 ; i < S; i++)
  83         data[i] = (float)sin(i * M_PI / HS);
  84     waves[wave_sine].make(bl, data);
  85
  86     for (int i = 0 ; i < QS; i++) {
  87         data[i] = i * iQS,
  88         data[i + QS] = 1 - i * iQS,
  89         data[i + HS] = - i * iQS,
  90         data[i + QS3] = -1 + i * iQS;
  91     }
  92     waves[wave_triangle].make(bl, data);
  93
  94     for (int i = 0, j = 1; i < S; i++) {
  95         data[i] = -1 + j * 1.0 / HS;
  96         if (i == j)
  97             j *= 2;
  98     }
  99     waves[wave_varistep].make(bl, data);
 100
 101     for (int i = 0; i < S; i++) {
 102         data[i] = (min(1.f, (float)(i / 64.f))) * (1.0 - i * 1.0 / S) * (-1 + fmod (i * i / 262144.0, 2.0));
 103     }
 104     waves[wave_skewsaw].make(bl, data);
 105     for (int i = 0; i < S; i++) {
 106         data[i] = (min(1.f, (float)(i / 64.f))) * (1.0 - i * 1.0 / S) * (fmod (i * i / 262144.0, 2.0) < 1.0 ? -1.0 : +1.0);
 107     }
 108     waves[wave_skewsqr].make(bl, data);
 109
 110     for (int i = 0; i < S; i++) {
 111         if (i < QS3) {
 112             float p = i * 1.0 / QS3;
 113             data[i] = sin(M_PI * p * p * p);
 114         } else {
 115             float p = (i - QS3 * 1.0) / QS;
 116             data[i] = -0.5 * sin(3 * M_PI * p * p);
 117         }
 118     }
 119     waves[wave_test1].make(bl, data);
 120     for (int i = 0; i < S; i++) {
 121         data[i] = exp(-i * 1.0 / HS) * sin(i * M_PI / HS) * cos(2 * M_PI * i / HS);
 122     }
 123     normalize_waveform(data, S);
 124     waves[wave_test2].make(bl, data);
 125     for (int i = 0; i < S; i++) {
 126         //int ii = (i < HS) ? i : S - i;
 127         int ii = HS;
 128         data[i] = (ii * 1.0 / HS) * sin(i * 3 * M_PI / HS + 2 * M_PI * sin(M_PI / 4 + i * 4 * M_PI / HS)) * sin(i * 5 * M_PI / HS + 2 * M_PI * sin(M_PI / 8 + i * 6 * M_PI / HS));
 129     }
 130     waves[wave_test3].make(bl, data);
 131     for (int i = 0; i < S; i++) {
 132         data[i] = sin(i * 2 * M_PI / HS + sin(i * 2 * M_PI / HS + 0.5 * M_PI * sin(i * 18 * M_PI / HS)) * sin(i * 1 * M_PI / HS + 0.5 * M_PI * sin(i * 11 * M_PI / HS)));
 133     }
 134     waves[wave_test4].make(bl, data);
 135     for (int i = 0; i < S; i++) {
 136         data[i] = sin(i * 2 * M_PI / HS + 0.2 * M_PI * sin(i * 13 * M_PI / HS) + 0.1 * M_PI * sin(i * 37 * M_PI / HS)) * sin(i * M_PI / HS + 0.2 * M_PI * sin(i * 15 * M_PI / HS));
 137     }
 138     waves[wave_test5].make(bl, data);
 139     for (int i = 0; i < S; i++) {
 140         if (i < HS)
 141             data[i] = sin(i * 2 * M_PI / HS);
 142         else
 143         if (i < 3 * S / 4)
 144             data[i] = sin(i * 4 * M_PI / HS);
 145         else
 146         if (i < 7 * S / 8)
 147             data[i] = sin(i * 8 * M_PI / HS);
 148         else
 149             data[i] = sin(i * 8 * M_PI / HS) * (S - i) / (S / 8);
 150     }
 151     waves[wave_test6].make(bl, data);
 152     for (int i = 0; i < S; i++) {
 153         int j = i >> (MONOSYNTH_WAVE_BITS - 11);
 154         data[i] = (j ^ 0x1D0) * 1.0 / HS - 1;
 155     }
 156     waves[wave_test7].make(bl, data);
 157     for (int i = 0; i < S; i++) {
 158         int j = i >> (MONOSYNTH_WAVE_BITS - 11);
 159         data[i] = -1 + 0.66 * (3 & ((j >> 8) ^ (j >> 10) ^ (j >> 6)));
 160     }
 161     waves[wave_test8].make(bl, data);
 162 }
 163
 164 bool monosynth_audio_module::get_static_graph(int index, int subindex, float value, float *data, int points, cairo_iface *context)
 165 {
 166     monosynth_audio_module::generate_waves();
 167     if (index == par_wave1 || index == par_wave2) {
 168         if (subindex)
 169             return false;
 170         enum { S = 1 << MONOSYNTH_WAVE_BITS };
 171         int wave = dsp::clip(dsp::fastf2i_drm(value), 0, (int)wave_count - 1);
 172
 173         float *waveform = waves[wave].original;
 174         for (int i = 0; i < points; i++)
 175             data[i] = waveform[i * S / points];
 176         return true;
 177     }
 178     return false;
 179 }
 180
 181 bool monosynth_audio_module::get_graph(int index, int subindex, float *data, int points, cairo_iface *context)
 182 {
 183     monosynth_audio_module::generate_waves();
 184     // printf("get_graph %d %p %d wave1=%d wave2=%d\n", index, data, points, wave1, wave2);
 185     if (index == par_filtertype) {
 186         if (!running)
 187             return false;
 188         if (subindex > (is_stereo_filter() ? 1 : 0))
 189             return false;
 190         for (int i = 0; i < points; i++)
 191         {
 192             typedef complex<double> cfloat;
 193             double freq = 20.0 * pow (20000.0 / 20.0, i * 1.0 / points);
 194
 195             dsp::biquad_d1<float> &f = subindex ? filter2 : filter;
 196             float level = f.freq_gain(freq, srate);
 197             if (!is_stereo_filter())
 198                 level *= filter2.freq_gain(freq, srate);
 199             level *= fgain;
 200
 201             data[i] = log(level) / log(1024.0) + 0.5;
 202         }
 203         return true;
 204     }
 205     return get_static_graph(index, subindex, *params[index], data, points, context);
 206 }
 207
 208 void monosynth_audio_module::calculate_buffer_ser()
 209 {
 210     for (uint32_t i = 0; i < step_size; i++)
 211     {
 212         float osc1val = osc1.get();
 213         float osc2val = osc2.get();
 214         float wave = fgain * (osc1val + (osc2val - osc1val) * xfade);
 215         wave = filter.process(wave);
 216         wave = filter2.process(wave);
 217         buffer[i] = wave;
 218         fgain += fgain_delta;
 219     }
 220 }
 221
 222 void monosynth_audio_module::calculate_buffer_single()
 223 {
 224     for (uint32_t i = 0; i < step_size; i++)
 225     {
 226         float osc1val = osc1.get();
 227         float osc2val = osc2.get();
 228         float wave = fgain * (osc1val + (osc2val - osc1val) * xfade);
 229         wave = filter.process(wave);
 230         buffer[i] = wave;
 231         fgain += fgain_delta;
 232     }
 233 }
 234
 235 void monosynth_audio_module::calculate_buffer_stereo()
 236 {
 237     for (uint32_t i = 0; i < step_size; i++)
 238     {
 239         float osc1val = osc1.get();
 240         float osc2val = osc2.get();
 241         float wave1 = osc1val + (osc2val - osc1val) * xfade;
 242         float wave2 = phaseshifter.process_ap(wave1);
 243         buffer[i] = fgain * filter.process(wave1);
 244         buffer2[i] = fgain * filter2.process(wave2);
 245         fgain += fgain_delta;
 246     }
 247 }
 248
 249 void monosynth_audio_module::delayed_note_on()
 250 {
 251     stop_count = 0;
 252     porta_time = 0.f;
 253     start_freq = freq;
 254     target_freq = freq = 440 * pow(2.0, (queue_note_on - 69) / 12.0);
 255     ampctl = 1.0 + (queue_vel - 1.0) * *params[par_vel2amp];
 256     fltctl = 1.0 + (queue_vel - 1.0) * *params[par_vel2filter];
 257     set_frequency();
 258     osc1.waveform = waves[wave1].get_level(osc1.phasedelta);
 259     osc2.waveform = waves[wave2].get_level(osc2.phasedelta);
 260     if (!osc1.waveform) osc1.waveform = silence;
 261     if (!osc2.waveform) osc2.waveform = silence;
 262
 263     if (!running)
 264     {
 265         if (legato >= 2)
 266             porta_time = -1.f;
 267         osc1.reset();
 268         osc2.reset();
 269         filter.reset();
 270         filter2.reset();
 271         switch((int)*params[par_oscmode])
 272         {
 273         case 1:
 274             osc2.phase = 0x80000000;
 275             break;
 276         case 2:
 277             osc2.phase = 0x40000000;
 278             break;
 279         case 3:
 280             osc1.phase = osc2.phase = 0x40000000;
 281             break;
 282         case 4:
 283             osc1.phase = 0x40000000;
 284             osc2.phase = 0xC0000000;
 285             break;
 286         case 5:
 287             // rand() is crap, but I don't have any better RNG in Calf yet
 288             osc1.phase = rand() << 16;
 289             osc2.phase = rand() << 16;
 290             break;
 291         default:
 292             break;
 293         }
 294         envelope.note_on();
 295         running = true;
 296     }
 297     if (legato >= 2 && !gate)
 298         porta_time = -1.f;
 299     gate = true;
 300     stopping = false;
 301     if (!(legato & 1) || envelope.released()) {
 302         envelope.note_on();
 303     }
 304     envelope.advance();
 305     queue_note_on = -1;
 306 }
 307
 308 void monosynth_audio_module::set_sample_rate(uint32_t sr) {
 309     srate = sr;
 310     crate = sr / step_size;
 311     odcr = (float)(1.0 / crate);
 312     phaseshifter.set_ap(1000.f, sr);
 313     fgain = 0.f;
 314     fgain_delta = 0.f;
 315 }
 316
 317 void monosynth_audio_module::calculate_step()
 318 {
 319     if (queue_note_on != -1)
 320         delayed_note_on();
 321     else if (stopping)
 322     {
 323         running = false;
 324         dsp::zero(buffer, step_size);
 325         if (is_stereo_filter())
 326             dsp::zero(buffer2, step_size);
 327         return;
 328     }
 329     float porta_total_time = *params[par_portamento] * 0.001f;
 330
 331     if (porta_total_time >= 0.00101f && porta_time >= 0) {
 332         // XXXKF this is criminal, optimize!
 333         float point = porta_time / porta_total_time;
 334         if (point >= 1.0f) {
 335             freq = target_freq;
 336             porta_time = -1;
 337         } else {
 338             freq = start_freq + (target_freq - start_freq) * point;
 339             // freq = start_freq * pow(target_freq / start_freq, point);
 340             porta_time += odcr;
 341         }
 342     }
 343     set_frequency();
 344     envelope.advance();
 345     float env = envelope.value;
 346     cutoff = *params[par_cutoff] * pow(2.0f, env * fltctl * *params[par_envmod] * (1.f / 1200.f));
 347     if (*params[par_keyfollow] > 0.01f)
 348         cutoff *= pow(freq / 264.f, *params[par_keyfollow]);
 349     cutoff = dsp::clip(cutoff , 10.f, 18000.f);
 350     float resonance = *params[par_resonance];
 351     float e2r = *params[par_envtores];
 352     float e2a = *params[par_envtoamp];
 353     resonance = resonance * (1 - e2r) + (0.7 + (resonance - 0.7) * env * env) * e2r;
 354     float cutoff2 = dsp::clip(cutoff * separation, 10.f, 18000.f);
 355     float newfgain = 0.f;
 356     if (filter_type != last_filter_type)
 357     {
 358         filter.y2 = filter.y1 = filter.x2 = filter.x1 = filter.y1;
 359         filter2.y2 = filter2.y1 = filter2.x2 = filter2.x1 = filter2.y1;
 360         last_filter_type = filter_type;
 361     }
 362     switch(filter_type)
 363     {
 364     case flt_lp12:
 365         filter.set_lp_rbj(cutoff, resonance, srate);
 366         filter2.set_null();
 367         newfgain = min(0.7f, 0.7f / resonance) * ampctl;
 368         break;
 369     case flt_hp12:
 370         filter.set_hp_rbj(cutoff, resonance, srate);
 371         filter2.set_null();
 372         newfgain = min(0.7f, 0.7f / resonance) * ampctl;
 373         break;
 374     case flt_lp24:
 375         filter.set_lp_rbj(cutoff, resonance, srate);
 376         filter2.set_lp_rbj(cutoff2, resonance, srate);
 377         newfgain = min(0.5f, 0.5f / resonance) * ampctl;
 378         break;
 379     case flt_lpbr:
 380         filter.set_lp_rbj(cutoff, resonance, srate);
 381         filter2.set_br_rbj(cutoff2, 1.0 / resonance, srate);
 382         newfgain = min(0.5f, 0.5f / resonance) * ampctl;
 383         break;
 384     case flt_hpbr:
 385         filter.set_hp_rbj(cutoff, resonance, srate);
 386         filter2.set_br_rbj(cutoff2, 1.0 / resonance, srate);
 387         newfgain = min(0.5f, 0.5f / resonance) * ampctl;
 388         break;
 389     case flt_2lp12:
 390         filter.set_lp_rbj(cutoff, resonance, srate);
 391         filter2.set_lp_rbj(cutoff2, resonance, srate);
 392         newfgain = min(0.7f, 0.7f / resonance) * ampctl;
 393         break;
 394     case flt_bp6:
 395         filter.set_bp_rbj(cutoff, resonance, srate);
 396         filter2.set_null();
 397         newfgain = ampctl;
 398         break;
 399     case flt_2bp6:
 400         filter.set_bp_rbj(cutoff, resonance, srate);
 401         filter2.set_bp_rbj(cutoff2, resonance, srate);
 402         newfgain = ampctl;
 403         break;
 404     }
 405     float aenv = env;
 406     /* isn't as good as expected
 407     if (e2a > 1.0) { // extra-steep release on amplitude envelope only
 408         if (envelope.state == adsr::RELEASE && env < envelope.sustain) {
 409             aenv -= (envelope.sustain - env) * (e2a - 1.0);
 410             if (aenv < 0.f) aenv = 0.f;
 411             printf("aenv = %f\n", aenv);
 412         }
 413         e2a = 1.0;
 414     }
 415     */
 416     newfgain *= 1.0 - (1.0 - aenv) * e2a;
 417     fgain_delta = (newfgain - fgain) * (1.0 / step_size);
 418     switch(filter_type)
 419     {
 420     case flt_lp24:
 421     case flt_lpbr:
 422     case flt_hpbr: // Oomek's wish
 423         calculate_buffer_ser();
 424         break;
 425     case flt_lp12:
 426     case flt_hp12:
 427     case flt_bp6:
 428         calculate_buffer_single();
 429         break;
 430     case flt_2lp12:
 431     case flt_2bp6:
 432         calculate_buffer_stereo();
 433         break;
 434     }
 435     if (envelope.state == adsr::STOP)
 436     {
 437         enum { ramp = step_size * 4 };
 438         for (int i = 0; i < step_size; i++)
 439             buffer[i] *= (ramp - i - stop_count) * (1.0f / ramp);
 440         if (is_stereo_filter())
 441             for (int i = 0; i < step_size; i++)
 442                 buffer2[i] *= (ramp - i - stop_count) * (1.0f / ramp);
 443         stop_count += step_size;
 444         if (stop_count >= ramp)
 445             stopping = true;
 446     }
 447 }
 448
 449 void monosynth_audio_module::control_change(int controller, int value)
 450 {
 451     switch(controller)
 452     {
 453         case 120: // all sounds off
 454         case 123: // all notes off
 455             gate = false;
 456             envelope.note_off();
 457             stack.clear();
 458             break;
 459     }
 460 }
 461