src/calf/modules_tools.h

   1 /* Calf DSP plugin pack
   2  * Assorted 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 02111-1307, USA.
  20  */
  21 #ifndef CALF_MODULES_TOOLS_H
  22 #define CALF_MODULES_TOOLS_H
  23
  24 #include <assert.h>
  25 #include <fftw3.h>
  26 #include <limits.h>
  27 #include "biquad.h"
  28 #include "inertia.h"
  29 #include "audio_fx.h"
  30 #include "analyzer.h"
  31 #include "giface.h"
  32 #include "metadata.h"
  33 #include "loudness.h"
  34 #include <math.h>
  35 #include "plugin_tools.h"
  36 #include "bypass.h"
  37
  38 namespace calf_plugins {
  39
  40 struct ladspa_plugin_info;
  41
  42 /**********************************************************************
  43  * STEREO TOOLS by Markus Schmidt
  44 **********************************************************************/
  45
  46 class stereo_audio_module:
  47     public audio_module<stereo_metadata>
  48 {
  49     typedef stereo_audio_module AM;
  50     float LL, LR, RL, RR;
  51     uint32_t srate;
  52     bool active;
  53     dsp::bypass bypass;
  54     float meter_inL, meter_inR, meter_outL, meter_outR, meter_phase;
  55     vumeters meters;
  56
  57     float * buffer;
  58     unsigned int pos;
  59     unsigned int buffer_size;
  60     static inline float sign(float x) {
  61         if(x < 0) return -1.f;
  62         if(x > 0) return 1.f;
  63         return 0.f;
  64     }
  65     float _phase, _phase_sin_coef, _phase_cos_coef, _sc_level, _inv_atan_shape;
  66 public:
  67     stereo_audio_module();
  68     ~stereo_audio_module();
  69     void params_changed();
  70     void activate();
  71     void set_sample_rate(uint32_t sr);
  72     void deactivate();
  73     uint32_t process(uint32_t offset, uint32_t numsamples, uint32_t inputs_mask, uint32_t outputs_mask);
  74 };
  75
  76 /**********************************************************************
  77  * MONO INPUT by Markus Schmidt
  78 **********************************************************************/
  79
  80 class mono_audio_module:
  81     public audio_module<mono_metadata>
  82 {
  83     typedef mono_audio_module AM;
  84     uint32_t srate;
  85     bool active;
  86     dsp::bypass bypass;
  87     float meter_in, meter_outL, meter_outR;
  88     vumeters meters;
  89
  90     float * buffer;
  91     unsigned int pos;
  92     unsigned int buffer_size;
  93     static inline float sign(float x) {
  94         if(x < 0) return -1.f;
  95         if(x > 0) return 1.f;
  96         return 0.f;
  97     }
  98     float _phase, _phase_sin_coef, _phase_cos_coef, _sc_level, _inv_atan_shape;
  99 public:
 100     mono_audio_module();
 101     ~mono_audio_module();
 102     void params_changed();
 103     void activate();
 104     void set_sample_rate(uint32_t sr);
 105     void deactivate();
 106     uint32_t process(uint32_t offset, uint32_t numsamples, uint32_t inputs_mask, uint32_t outputs_mask);
 107 };
 108
 109 /**********************************************************************
 110  * ANALYZER by Markus Schmidt and Christian Holschuh
 111 **********************************************************************/
 112
 113 class analyzer_audio_module:
 114     public audio_module<analyzer_metadata>, public phase_graph_iface,
 115     public frequency_response_line_graph
 116 {
 117     typedef analyzer_audio_module AM;
 118     uint32_t srate;
 119     bool active;
 120     float envelope;
 121     float attack_coef;
 122     float release_coef;
 123     uint32_t clip_L, clip_R;
 124     float meter_L, meter_R;
 125     analyzer _analyzer;
 126 public:
 127     analyzer_audio_module();
 128     void params_changed();
 129     void activate();
 130     void set_sample_rate(uint32_t sr);
 131     void deactivate();
 132     uint32_t process(uint32_t offset, uint32_t numsamples, uint32_t inputs_mask, uint32_t outputs_mask);
 133     bool get_phase_graph(float ** _buffer, int * _length, int * _mode, bool * _use_fade, float * _fade, int * _accuracy, bool * _display) const;
 134     bool get_graph(int index, int subindex, int phase, float *data, int points, cairo_iface *context, int *mode) const;
 135     bool get_moving(int index, int subindex, int &direction, float *data, int x, int y, int &offset, uint32_t &color) const;
 136     bool get_gridline(int index, int subindex, int phase, float &pos, bool &vertical, std::string &legend, cairo_iface *context) const;
 137     bool get_layers(int index, int generation, unsigned int &layers) const;
 138     ~analyzer_audio_module();
 139 protected:
 140     static const int max_phase_buffer_size = 8192;
 141     int phase_buffer_size;
 142     float *phase_buffer;
 143     int ppos;
 144     int plength;
 145 };
 146
 147 };
 148 #endif