src/modules_pitch.cpp

   1 /* Calf DSP plugin pack
   2  * Assorted plugins
   3  *
   4  * Copyright (C) 2015 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen
   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
  22 #include <limits.h>
  23 #include <memory.h>
  24 #include <math.h>
  25 #include <calf/giface.h>
  26 #include <calf/modules_pitch.h>
  27 #include <calf/utils.h>
  28
  29 #if ENABLE_EXPERIMENTAL
  30
  31 // http://www.cs.otago.ac.nz/tartini/papers/A_Smarter_Way_to_Find_Pitch.pdf
  32
  33 using namespace calf_plugins;
  34
  35 pitch_audio_module::pitch_audio_module()
  36 {
  37 }
  38
  39 pitch_audio_module::~pitch_audio_module()
  40 {
  41 }
  42
  43 void pitch_audio_module::set_sample_rate(uint32_t sr)
  44 {
  45     srate = sr;
  46 }
  47
  48 void pitch_audio_module::params_changed()
  49 {
  50 }
  51
  52 void pitch_audio_module::activate()
  53 {
  54     write_ptr = 0;
  55     for (size_t i = 0; i < 2 * BufferSize; ++i)
  56         waveform[i] = spectrum[i] = autocorr[i] = 0;
  57     for (size_t i = 0; i < BufferSize; ++i)
  58         inputbuf[i] = 0;
  59 }
  60
  61 void pitch_audio_module::deactivate()
  62 {
  63 }
  64
  65 void pitch_audio_module::recompute()
  66 {
  67     // second half of the waveform always zero
  68     double sumsquares_acc = 0.;
  69     for (int i = 0; i < BufferSize; ++i)
  70     {
  71         float val = inputbuf[(i + write_ptr) & (BufferSize - 1)];
  72         float win = 0.54 - 0.46 * cos(i * M_PI / BufferSize);
  73         val *= win;
  74         waveform[i] = val;
  75         sumsquares[i] = sumsquares_acc;
  76         sumsquares_acc += val * val;
  77     }
  78     sumsquares[BufferSize] = sumsquares_acc;
  79         //waveform[i] = inputbuf[(i + write_ptr) & (BufferSize - 1)];
  80     transform.calculate(waveform, spectrum, false);
  81     pfft::complex temp[2 * BufferSize];
  82     for (int i = 0; i < 2 * BufferSize; ++i)
  83     {
  84         float val = std::abs(spectrum[i]);
  85         temp[i] = val * val;
  86     }
  87         //temp[i] = spectrum[i] * std::conj(spectrum[i]);
  88     transform.calculate(temp, autocorr, true);
  89     sumsquares_last = sumsquares_acc;
  90     float maxpt = 0;
  91     int maxpos = -1;
  92     int i;
  93     for (i = 2; i < BufferSize / 2; ++i)
  94     {
  95         float mag = 2.0 * autocorr[i].real() / (sumsquares[BufferSize] + sumsquares[BufferSize - i] - sumsquares[i]);
  96         magarr[i] = mag;
  97         if (mag > maxpt)
  98         {
  99             maxpt = mag;
 100             maxpos = i;
 101         }
 102     }
 103     for (i = 2; i < BufferSize / 2 && magarr[i + 1] < magarr[i]; ++i)
 104         ;
 105     float thr = *params[par_pd_threshold];
 106     for (; i < BufferSize / 2; ++i)
 107     {
 108         if (magarr[i] >= thr * maxpt)
 109         {
 110             while(i < BufferSize / 2 - 1 && magarr[i + 1] > magarr[i])
 111                 i++;
 112             maxpt = magarr[i];
 113             maxpos = i;
 114             break;
 115         }
 116     }
 117     if (maxpt > 0 && maxpos < BufferSize / 2 - 1)
 118     {
 119         float y1 = magarr[maxpos - 1];
 120         float y2 = magarr[maxpos];
 121         float y3 = magarr[maxpos + 1];
 122         float pos2 = maxpos + 0.5 * (y1 - y3) / (y1 - 2 * y2 + y3);
 123         dsp::note_desc desc = dsp::hz_to_note(srate / pos2, *params[par_tune]);
 124         *params[par_note]  = desc.note;
 125         *params[par_cents] = desc.cents;
 126         *params[par_freq]  = desc.freq;
 127         *params[par_clarity] = maxpt;
 128     }
 129     *params[par_clarity] = maxpt;
 130 }
 131
 132 bool pitch_audio_module::get_graph(int index, int subindex, int phase, float *data, int points, cairo_iface *context, int *mode) const
 133 {
 134     if (index == par_pd_threshold && subindex == 0)
 135     {
 136         context->set_source_rgba(1, 0, 0);
 137         for (int i = 0; i < points; i++)
 138         {
 139             float ac = autocorr[i * (BufferSize / 2 - 1) / (points - 1)].real();
 140             if (ac >= 0)
 141                 data[i] = sqrt(ac / sumsquares_last);
 142             else
 143                 data[i] = -sqrt(-ac / sumsquares_last);
 144         }
 145         return true;
 146     }
 147     if (index == par_pd_threshold && subindex == 1)
 148     {
 149         context->set_source_rgba(0, 0, 1);
 150         for (int i = 0; i < points; i++)
 151         {
 152             data[i] = 0.0625 * log(std::abs(spectrum[i * (BufferSize / 4 - 1) / (points - 1)]));
 153         }
 154         return true;
 155     }
 156     if (index == par_pd_threshold && subindex == 2)
 157     {
 158         context->set_source_rgba(0, 0, 0);
 159         for (int i = 0; i < points; i++)
 160         {
 161             int j = i * (BufferSize / 2 - 1) / (points - 1);
 162             float mag = magarr[j];
 163             data[i] = mag;
 164         }
 165         return true;
 166     }
 167     if (index == par_pd_threshold && subindex == 3)
 168     {
 169         context->set_source_rgba(0, 1, 1);
 170         for (int i = 0; i < points; i++)
 171         {
 172             int j = i * (BufferSize - 1) / (points - 1);
 173             float mag = std::abs(sumsquares[j]);
 174             data[i] = 0.25 * log(mag);
 175         }
 176         return true;
 177     }
 178     return false;
 179 }
 180
 181 uint32_t pitch_audio_module::process(uint32_t offset, uint32_t numsamples, uint32_t inputs_mask, uint32_t outputs_mask)
 182 {
 183     uint32_t endpos = offset + numsamples;
 184     bool has2nd = ins[1] != NULL;
 185
 186     int bperiod = BufferSize;
 187     int sd = *params[par_pd_subdivide];
 188     if (sd >= 1 && sd <= 8)
 189         bperiod /= sd;
 190
 191     // XXXKF note: not working, not optimized, just trying stuff out
 192     for (uint32_t i = offset; i < endpos; ++i)
 193     {
 194         float val = ins[0][i];
 195         inputbuf[write_ptr] = val;
 196         write_ptr = (write_ptr + 1) & (BufferSize - 1);
 197         if (!(write_ptr % bperiod))
 198             recompute();
 199         outs[0][i] = ins[0][i];
 200         if (has2nd)
 201             outs[1][i] = ins[1][i];
 202     }
 203     return outputs_mask;
 204 }
 205
 206 #endif