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faustvst.cpp
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/************************************************************************
************************************************************************
FAUST Architecture File
Copyright (C) 2014-2016 Albert Graef <[email protected]>
---------------------------------------------------------------------
This program is free software; you can redistribute it and/or modify
it under the terms of the GNU Lesser General Public License as
published by the Free Software Foundation; either version 3 of the
License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with the GNU C Library; if not, write to the Free
Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA
02111-1307 USA.
************************************************************************
************************************************************************/
/* VST architecture for Faust synths. */
/* NOTE: This requires one of the Boost headers (boost/circular_buffer.hpp),
so to compile Faust programs created with this architecture you need to
have at least the Boost headers installed somewhere on your include path
(the Boost libraries aren't needed). */
#include <cstdlib>
#include <cstdint>
#include <cmath>
#include <list>
#include <map>
#include <set>
// generic Faust dsp and UI classes
#include <faust/dsp/dsp.h>
#include <faust/gui/UI.h>
using namespace std;
typedef pair<const char*,const char*> strpair;
struct Meta : std::map<const char*, const char*>
{
void declare(const char *key, const char *value)
{
(*this)[key] = value;
}
const char* get(const char *key, const char *def)
{
if (this->find(key) != this->end())
return (*this)[key];
else
return def;
}
};
/******************************************************************************
*******************************************************************************
VECTOR INTRINSICS
*******************************************************************************
*******************************************************************************/
<<includeIntrinsic>>
/***************************************************************************
VST UI interface
***************************************************************************/
#include <string.h>
enum ui_elem_type_t {
UI_BUTTON, UI_CHECK_BUTTON,
UI_V_SLIDER, UI_H_SLIDER, UI_NUM_ENTRY,
UI_V_BARGRAPH, UI_H_BARGRAPH,
UI_END_GROUP, UI_V_GROUP, UI_H_GROUP, UI_T_GROUP
};
struct ui_elem_t {
ui_elem_type_t type;
const char *label;
int port;
float *zone;
void *ref;
float init, min, max, step;
};
class VSTUI : public UI
{
public:
bool is_instr;
int nelems, nports;
ui_elem_t *elems;
map< int, list<strpair> > metadata;
VSTUI(int maxvoices = 0);
virtual ~VSTUI();
protected:
void add_elem(ui_elem_type_t type, const char *label = NULL);
void add_elem(ui_elem_type_t type, const char *label, float *zone);
void add_elem(ui_elem_type_t type, const char *label, float *zone,
float init, float min, float max, float step);
void add_elem(ui_elem_type_t type, const char *label, float *zone,
float min, float max);
bool have_freq, have_gain, have_gate;
bool is_voice_ctrl(const char *label);
public:
virtual void addButton(const char* label, float* zone);
virtual void addCheckButton(const char* label, float* zone);
virtual void addVerticalSlider(const char* label, float* zone, float init, float min, float max, float step);
virtual void addHorizontalSlider(const char* label, float* zone, float init, float min, float max, float step);
virtual void addNumEntry(const char* label, float* zone, float init, float min, float max, float step);
virtual void addHorizontalBargraph(const char* label, float* zone, float min, float max);
virtual void addVerticalBargraph(const char* label, float* zone, float min, float max);
virtual void addSoundfile(const char* label, const char* filename, Soundfile** sf_zone) {}
virtual void openTabBox(const char* label);
virtual void openHorizontalBox(const char* label);
virtual void openVerticalBox(const char* label);
virtual void closeBox();
virtual void run();
virtual void declare(float* zone, const char* key, const char* value);
};
VSTUI::VSTUI(int maxvoices)
{
is_instr = maxvoices>0;
have_freq = have_gain = have_gate = false;
nelems = nports = 0;
elems = NULL;
}
VSTUI::~VSTUI()
{
if (elems) free(elems);
}
void VSTUI::declare(float* zone, const char* key, const char* value)
{
map< int, list<strpair> >::iterator it = metadata.find(nelems);
if (it != metadata.end())
it->second.push_back(strpair(key, value));
else
metadata[nelems] = list<strpair>(1, strpair(key, value));
}
inline void VSTUI::add_elem(ui_elem_type_t type, const char *label)
{
ui_elem_t *elems1 = (ui_elem_t*)realloc(elems, (nelems+1)*sizeof(ui_elem_t));
if (elems1)
elems = elems1;
else
return;
elems[nelems].type = type;
elems[nelems].label = label;
elems[nelems].port = -1;
elems[nelems].zone = NULL;
elems[nelems].ref = NULL;
elems[nelems].init = 0.0;
elems[nelems].min = 0.0;
elems[nelems].max = 0.0;
elems[nelems].step = 0.0;
nelems++;
}
#define portno(label) (is_voice_ctrl(label)?-1:nports++)
inline void VSTUI::add_elem(ui_elem_type_t type, const char *label, float *zone)
{
ui_elem_t *elems1 = (ui_elem_t*)realloc(elems, (nelems+1)*sizeof(ui_elem_t));
if (elems1)
elems = elems1;
else
return;
elems[nelems].type = type;
elems[nelems].label = label;
elems[nelems].port = portno(label);
elems[nelems].zone = zone;
elems[nelems].ref = NULL;
elems[nelems].init = 0.0;
elems[nelems].min = 0.0;
elems[nelems].max = 1.0;
elems[nelems].step = 1.0;
nelems++;
}
inline void VSTUI::add_elem(ui_elem_type_t type, const char *label, float *zone, float init, float min, float max, float step)
{
ui_elem_t *elems1 = (ui_elem_t*)realloc(elems, (nelems+1)*sizeof(ui_elem_t));
if (elems1)
elems = elems1;
else
return;
elems[nelems].type = type;
elems[nelems].label = label;
elems[nelems].port = portno(label);
elems[nelems].zone = zone;
elems[nelems].ref = NULL;
elems[nelems].init = init;
elems[nelems].min = min;
elems[nelems].max = max;
elems[nelems].step = step;
nelems++;
}
inline void VSTUI::add_elem(ui_elem_type_t type, const char *label, float *zone, float min, float max)
{
ui_elem_t *elems1 = (ui_elem_t*)realloc(elems, (nelems+1)*sizeof(ui_elem_t));
if (elems1)
elems = elems1;
else
return;
elems[nelems].type = type;
elems[nelems].label = label;
elems[nelems].port = portno(label);
elems[nelems].zone = zone;
elems[nelems].ref = NULL;
elems[nelems].init = 0.0;
elems[nelems].min = min;
elems[nelems].max = max;
elems[nelems].step = 0.0;
nelems++;
}
inline bool VSTUI::is_voice_ctrl(const char *label)
{
if (!is_instr)
return false;
else if (!have_freq && !strcmp(label, "freq"))
return (have_freq = true);
else if (!have_gain && !strcmp(label, "gain"))
return (have_gain = true);
else if (!have_gate && !strcmp(label, "gate"))
return (have_gate = true);
else
return false;
}
void VSTUI::addButton(const char* label, float* zone)
{ add_elem(UI_BUTTON, label, zone); }
void VSTUI::addCheckButton(const char* label, float* zone)
{ add_elem(UI_CHECK_BUTTON, label, zone); }
void VSTUI::addVerticalSlider(const char* label, float* zone, float init, float min, float max, float step)
{ add_elem(UI_V_SLIDER, label, zone, init, min, max, step); }
void VSTUI::addHorizontalSlider(const char* label, float* zone, float init, float min, float max, float step)
{ add_elem(UI_H_SLIDER, label, zone, init, min, max, step); }
void VSTUI::addNumEntry(const char* label, float* zone, float init, float min, float max, float step)
{ add_elem(UI_NUM_ENTRY, label, zone, init, min, max, step); }
void VSTUI::addHorizontalBargraph(const char* label, float* zone, float min, float max)
{ add_elem(UI_H_BARGRAPH, label, zone, min, max); }
void VSTUI::addVerticalBargraph(const char* label, float* zone, float min, float max)
{ add_elem(UI_V_BARGRAPH, label, zone, min, max); }
void VSTUI::openTabBox(const char* label)
{ add_elem(UI_T_GROUP, label); }
void VSTUI::openHorizontalBox(const char* label)
{ add_elem(UI_H_GROUP, label); }
void VSTUI::openVerticalBox(const char* label)
{ add_elem(UI_V_GROUP, label); }
void VSTUI::closeBox()
{ add_elem(UI_END_GROUP); }
void VSTUI::run() {}
//----------------------------------------------------------------------------
// FAUST generated signal processor
//----------------------------------------------------------------------------
<<includeclass>>
//----------------------------------------------------------------------------
// VST interface
//----------------------------------------------------------------------------
#line 286 "faustvst.cpp"
#include <assert.h>
#include <stdio.h>
#include <stdlib.h>
#include <boost/circular_buffer.hpp>
// Some boilerplate code pilfered from the mda Linux vst source code.
#include "pluginterfaces/vst2.x/aeffectx.h"
extern "C" {
#define VST_EXPORT __attribute__ ((visibility ("default")))
extern VST_EXPORT AEffect * VSTPluginMain(audioMasterCallback audioMaster);
// This is for legacy (<2.4) VST hosts which look for the 'main' entry point.
AEffect *main_plugin (audioMasterCallback audioMaster) asm ("main");
#define main main_plugin
VST_EXPORT AEffect * main(audioMasterCallback audioMaster)
{
return VSTPluginMain(audioMaster);
}
}
/* Setting NVOICES at compile time overrides meta data in the Faust source. If
set, this must be an integer value >= 0. A nonzero value indicates an
instrument (VSTi) plugin with the given maximum number of voices. Use 1 for
a monophonic synthesizer, and 0 for a simple effect plugin. If NVOICES
isn't defined at compile time then the number of voices of an instrument
plugin can also be set with the global "nvoices" meta data key in the Faust
source. This setting also adds a special "polyphony" control to the plugin
which can be used to dynamically adjust the actual number of voices in the
range 1..NVOICES. */
//#define NVOICES 16
/* This enables special polyphony/tuning controls on the GUI (VSTi only). */
#ifndef VOICE_CTRLS
#define VOICE_CTRLS 1
#endif
/* This enables a special "tuning" control in a VSTi plugin which lets you
select the MTS tuning to be used for the synth. In order to use this, you
just drop some sysex (.syx) files with MTS octave-based tunings in 1- or
2-byte format into the ~/.fautvst/tuning directory (these can be generated
with the author's sclsyx program, https://bitbucket.org/agraef/sclsyx).
The control will only be shown if any .syx files were found at startup. 0
selects the default tuning (standard 12-tone equal temperament), i>0 the
tuning in the ith sysex file (in alphabetic order). */
#ifndef FAUST_MTS
#define FAUST_MTS 1
#endif
/* This allows various manifest data to be generated from the corresponding
metadata (author, name, description, license) in the Faust source. */
#ifndef FAUST_META
#define FAUST_META 1
#endif
/* This enables automatic MIDI controller mapping based on the midi:ctrl
attributes in the Faust source. We have this enabled by default, but you
may have to disable it if the custom controller mapping gets in the way of
the automation facilities that the host provides. (But then again if the
host wants to do its own controller mapping then it probably won't, or at
least shouldn't, send us the MIDI controllers in the first place.) */
#ifndef FAUST_MIDICC
#define FAUST_MIDICC 1
#endif
/* This enables or disables the plugin's custom Qt GUI (see the Qt-specific
part at the end of this module). This is disabled by default, but enabled
with gui=1 in the Makefile or the -gui option of the faust2vst script. */
#ifndef FAUST_UI
#define FAUST_UI 0
#endif
// You can define these for various debugging output items.
//#define DEBUG_META 1 // recognized MIDI controller metadata
//#define DEBUG_VOICES 1 // triggering of synth voices
//#define DEBUG_VOICE_ALLOC 1 // voice allocation
//#define DEBUG_MIDI 1 // incoming MIDI messages
//#define DEBUG_NOTES 1 // note messages
//#define DEBUG_MIDICC 1 // controller messages
//#define DEBUG_RPN 1 // RPN messages (pitch bend range, master tuning)
//#define DEBUG_MTS 1 // MTS messages (octave/scale tuning)
// Note and voice data structures.
struct NoteInfo {
uint8_t ch;
int8_t note;
};
struct VoiceData {
// Octave tunings (offsets in semitones) per MIDI channel.
float tuning[16][12];
// Allocated voices per MIDI channel and note.
int8_t notes[16][128];
// Free and used voices.
int n_free, n_used;
boost::circular_buffer<int> free_voices;
boost::circular_buffer<int> used_voices;
NoteInfo *note_info;
// Voices queued for note-offs (zero-length notes).
set<int> queued;
// Last gate value during run() for each voice. We need to keep track of
// these so that we can force the Faust synth to retrigger a note when
// needed.
float *lastgate;
// Current pitch bend and pitch bend range on each MIDI channel, in semitones.
float bend[16], range[16];
// Current coarse, fine and total master tuning on each MIDI channel (tuning
// offset relative to A4 = 440 Hz, in semitones).
float coarse[16], fine[16], tune[16];
VoiceData(int n) : free_voices(n), used_voices(n) { }
};
#if FAUST_MTS
// Helper classes to read and store MTS tunings.
#include <sys/types.h>
#include <sys/stat.h>
#include <dirent.h>
#include <string>
#include <vector>
struct MTSTuning {
char *name; // name of the tuning
int len; // length of sysex data in bytes
unsigned char *data; // sysex data
MTSTuning() : name(0), len(0), data(0) {}
MTSTuning& operator=(const MTSTuning &t)
{
if (this == &t) return *this;
if (name) free(name); if (data) free(data);
name = 0; data = 0; len = t.len;
if (t.name) {
name = strdup(t.name); assert(name);
}
if (t.data) {
data = (unsigned char*)malloc(len); assert(data);
memcpy(data, t.data, len);
}
return *this;
}
MTSTuning(const MTSTuning& t) : name(0), len(0), data(0)
{ *this = t; }
MTSTuning(const char *filename);
~MTSTuning()
{ if (name) free(name); if (data) free(data); }
};
MTSTuning::MTSTuning(const char *filename)
{
FILE *fp = fopen(filename, "rb");
name = 0; len = 0; data = 0;
if (!fp) return;
struct stat st;
if (fstat(fileno(fp), &st)) return;
len = st.st_size;
data = (unsigned char*)calloc(len, 1);
if (!data) {
len = 0; fclose(fp);
return;
}
assert(len > 0);
if (fread(data, 1, len, fp) < len) {
free(data); len = 0; data = 0; fclose(fp);
return;
}
fclose(fp);
// Do some basic sanity checks.
if (data[0] != 0xf0 || data[len-1] != 0xf7 || // not a sysex message
(data[1] != 0x7e && data[1] != 0x7f) || data[3] != 8 || // not MTS
!((len == 21 && data[4] == 8) ||
(len == 33 && data[4] == 9))) { // no 1- or 2-byte tuning
free(data); len = 0; data = 0;
return;
}
// Name of the tuning is the basename of the file, without the trailing .syx
// suffix.
string nm = filename;
size_t p = nm.rfind(".syx");
if (p != string::npos) nm.erase(p);
p = nm.rfind('/');
if (p != string::npos) nm.erase(0, p+1);
name = strdup(nm.c_str());
assert(name);
}
struct MTSTunings {
vector<MTSTuning> tuning;
MTSTunings() {}
MTSTunings(const char *path);
};
static bool compareByName(const MTSTuning &a, const MTSTuning &b)
{
return strcmp(a.name, b.name) < 0;
}
MTSTunings::MTSTunings(const char *path)
{
DIR *dp = opendir(path);
if (!dp) return;
struct dirent *d;
while ((d = readdir(dp))) {
string nm = d->d_name;
if (nm.length() > 4 && nm.substr(nm.length()-4) == ".syx") {
string pathname = path;
pathname += "/";
pathname += nm;
MTSTuning t(pathname.c_str());
if (t.data) tuning.push_back(t);
}
}
closedir(dp);
// sort found tunings by name
sort(tuning.begin(), tuning.end(), compareByName);
}
#endif
#if FAUST_MIDICC
static float ctrlval(const ui_elem_t &el, uint8_t v)
{
// Translate the given MIDI controller value to the range and stepsize
// indicated by the Faust control.
switch (el.type) {
case UI_BUTTON: case UI_CHECK_BUTTON:
return (float)(v>=64);
default:
/* Continuous controllers. The problem here is that the range 0..127 is
not symmetric. We'd like to map 64 to the center of the range
(max-min)/2 and at the same time retain the full control range
min..max. So let's just pretend that there are 128 controller values
and map value 127 to the max value anyway. */
if (v==127)
return el.max;
else
// XXXFIXME: We might want to add proper quantization according to
// el.step here.
return el.min+(el.max-el.min)*v/128;
}
}
#endif
/***************************************************************************/
/* Polyphonic Faust plugin data structure. XXXTODO: At present this is just a
big struct which exposes all requisite data. Some more work is needed to
make the interface a bit more abstract and properly encapsulate the
internal data structures, so that implementation details can be changed
more easily. */
struct VSTPlugin {
const int maxvoices; // maximum number of voices (zero if not an instrument)
const int ndsps; // number of dsp instances (1 if maxvoices==0)
int nvoices; // current number of voices (<= maxvoices)
bool active; // activation status
bool modified; // keep track of modified controls
int rate; // sampling rate
mydsp **dsp; // the dsps
VSTUI **ui; // their Faust interface descriptions
int n_in, n_out; // number of input and output control ports
int poly, tuning; // polyphony and tuning ports
int *ctrls; // Faust ui elements (indices into ui->elems)
float *ports; // port data (plugin-side control values)
float *portvals; // cached port data from the last run
float *midivals[16]; // per-channel midi data
int *inctrls, *outctrls; // indices for active and passive controls
int freq, gain, gate; // indices of voice controls
const char **units; // unit names (control meta data)
unsigned n_samples; // current block size
float **outbuf; // audio buffers for mixing down the voices
float **inbuf; // dummy input buffer used for retriggering notes
std::map<uint8_t,int> ctrlmap; // MIDI controller map (control meta data)
// Current RPN MSB and LSB numbers, as set with controllers 101 and 100.
uint8_t rpn_msb[16], rpn_lsb[16];
// Current data entry MSB and LSB numbers, as set with controllers 6 and 38.
uint8_t data_msb[16], data_lsb[16];
// Synth voice data (instruments only).
VoiceData *vd;
// Static methods. These all use static data so they can be invoked before
// instantiating a plugin.
// Global meta data (dsp name, author, etc.).
static Meta *meta;
static void init_meta()
{
if (!meta && (meta = new Meta)) {
// We allocate the temporary dsp object on the heap here, to prevent
// large dsp objects from running out of stack in environments where
// stack space is precious (e.g., Reaper). Note that if any of these
// allocations fail then no meta data will be available, but at least we
// won't make the host crash and burn.
mydsp* tmp_dsp = new mydsp();
if (tmp_dsp) {
tmp_dsp->metadata(meta);
delete tmp_dsp;
}
}
}
static const char *meta_get(const char *key, const char *deflt)
{
init_meta();
return meta?meta->get(key, deflt):deflt;
}
static const char *pluginName()
{
return meta_get("name", "mydsp");
}
static const char *pluginAuthor()
{
return meta_get("author", "");
}
static const char *pluginDescription()
{
return meta_get("description", "");
}
static const char *pluginVersion()
{
return meta_get("version", "0.0");
}
// Load a collection of sysex files with MTS tunings in ~/.faust/tuning.
static int n_tunings;
#if FAUST_MTS
static MTSTunings *mts;
static MTSTunings *load_sysex_data()
{
if (!mts) {
string mts_path;
// Look for FAUST_HOME. If that isn't set, try $HOME/.faust. If HOME
// isn't set either, just assume a .faust subdir of the cwd.
const char *home = getenv("FAUST_HOME");
if (home)
mts_path = home;
else {
home = getenv("HOME");
if (home) {
mts_path = home;
mts_path += "/.faust";
} else
mts_path = ".faust";
}
// MTS tunings are looked for in this subdir.
mts_path += "/tuning";
mts = new MTSTunings(mts_path.c_str());
#ifdef __APPLE__
if (!mts || mts->tuning.size() == 0) {
// Also check ~/Library/Faust/Tuning on the Mac.
home = getenv("HOME");
if (home) {
if (mts) delete mts;
mts_path = home;
mts_path += "/Library/Faust/Tuning";
mts = new MTSTunings(mts_path.c_str());
}
}
#endif
n_tunings = mts->tuning.size();
}
return mts;
}
#endif
// The number of voices of an instrument plugin. We get this information
// from the global meta data (nvoices key) of the dsp module if present, and
// you can also override this setting at compile time by defining the
// NVOICES macro. If neither is defined or the value is zero then the plugin
// becomes a simple audio effect instead.
static int numVoices()
{
#ifdef NVOICES
return NVOICES;
#else
const char *numVoices = meta_get("nvoices", "0");
int nvoices = atoi(numVoices);
if (nvoices < 0) nvoices = 0;
return nvoices;
#endif
}
// The number of controls of the dsp. Some plugin interfaces need that
// information beforehand, so we create a dummy instance of the UI data to
// retrieve it. For instrument plugins, we also reserve extra ports for the
// polyphony and tuning controls, if applicable.
static int numControls()
{
const int num_voices = numVoices();
// Allocate temporary dsp object on the heap (see comments under init_meta
// for explanation).
mydsp *dsp = new mydsp();
if (!dsp) return 0;
VSTUI ui(num_voices);
dsp->buildUserInterface(&ui);
delete dsp;
// reserve one extra port for the polyphony control (instruments only)
int num_extra = (num_voices>0);
#if FAUST_MTS
// likewise for the tuning control
if (num_voices>0 && load_sysex_data())
num_extra += (mts->tuning.size()>0);
#endif
return ui.nports+num_extra;
}
// Instance methods.
VSTPlugin(const int num_voices, const int sr)
: maxvoices(num_voices), ndsps(num_voices<=0?1:num_voices),
vd(num_voices>0?new VoiceData(num_voices):0)
{
// Initialize static data.
init_meta();
#if FAUST_MTS
// Synth: load tuning sysex data if present.
if (num_voices>0) load_sysex_data();
#endif
// Allocate data structures and set some reasonable defaults.
dsp = (mydsp**)calloc(ndsps, sizeof(mydsp*));
ui = (VSTUI**)calloc(ndsps, sizeof(VSTUI*));
assert(dsp && ui);
if (vd) {
vd->note_info = (NoteInfo*)calloc(ndsps, sizeof(NoteInfo));
vd->lastgate = (float*)calloc(ndsps, sizeof(float));
assert(vd->note_info && vd->lastgate);
}
active = modified = false;
rate = sr;
nvoices = maxvoices;
n_in = n_out = 0;
poly = maxvoices/2;
tuning = 0;
freq = gain = gate = -1;
for (int i = 0; i < 16; i++) {
rpn_msb[i] = rpn_lsb[i] = 0x7f;
data_msb[i] = data_lsb[i] = 0;
}
if (vd) {
vd->n_free = maxvoices;
for (int i = 0; i < maxvoices; i++) {
vd->free_voices.push_back(i);
vd->lastgate[i] = 0.0f;
}
for (int i = 0; i < 16; i++) {
vd->bend[i] = 0.0f;
vd->range[i] = 2.0f;
vd->coarse[i] = vd->fine[i] = vd->tune[i] = 0.0f;
for (int j = 0; j < 12; j++)
vd->tuning[i][j] = 0.0f;
}
vd->n_used = 0;
memset(vd->notes, 0xff, sizeof(vd->notes));
}
n_samples = 0;
ctrls = inctrls = outctrls = NULL;
inbuf = outbuf = NULL;
ports = portvals = NULL;
units = NULL;
memset(midivals, 0, sizeof(midivals));
// Initialize the Faust DSPs.
for (int i = 0; i < ndsps; i++) {
dsp[i] = new mydsp();
ui[i] = new VSTUI(num_voices);
dsp[i]->init(rate);
dsp[i]->buildUserInterface(ui[i]);
}
// The ports are numbered as follows: 0..k-1 are the control ports, then
// come the n audio input ports, then the m audio output ports, and
// finally the midi input port and the polyphony and tuning controls. This
// mimics the port layout of faust-lv2, but should work fine with other
// kinds of plugin architectures as well.
int k = ui[0]->nports, p = 0, q = 0;
int n = dsp[0]->getNumInputs(), m = dsp[0]->getNumOutputs();
// Allocate tables for the built-in control elements and their ports.
ctrls = (int*)calloc(k, sizeof(int));
inctrls = (int*)calloc(k, sizeof(int));
outctrls = (int*)calloc(k, sizeof(int));
ports = (float*)calloc(k, sizeof(float));
portvals = (float*)calloc(k, sizeof(float));
units = (const char**)calloc(k, sizeof(const char*));
assert(k == 0 || (ctrls && inctrls && outctrls &&
ports && portvals && units));
for (int ch = 0; ch < 16; ch++) {
midivals[ch] = (float*)calloc(k, sizeof(float));
assert(k == 0 || midivals[ch]);
}
// Scan the Faust UI for active and passive controls which become the
// input and output control ports of the plugin, respectively.
for (int i = 0, j = 0; i < ui[0]->nelems; i++) {
const char *unit = NULL;
switch (ui[0]->elems[i].type) {
case UI_T_GROUP: case UI_H_GROUP: case UI_V_GROUP: case UI_END_GROUP:
// control groups (ignored right now)
break;
case UI_H_BARGRAPH: case UI_V_BARGRAPH:
// passive controls (output ports)
ctrls[j++] = i;
outctrls[q++] = i;
{
std::map< int, list<strpair> >::iterator it =
ui[0]->metadata.find(i);
if (it != ui[0]->metadata.end()) {
for (std::list<strpair>::iterator jt = it->second.begin();
jt != it->second.end(); jt++) {
const char *key = jt->first, *val = jt->second;
#if DEBUG_META
fprintf(stderr, "ctrl '%s' meta: '%s' -> '%s'\n",
ui[0]->elems[i].label, key, val);
#endif
if (strcmp(key, "unit") == 0)
unit = val;
}
}
int p = ui[0]->elems[i].port;
units[p] = unit;
}
break;
default:
// active controls (input ports)
if (maxvoices == 0)
goto noinstr;
else if (freq == -1 &&
!strcmp(ui[0]->elems[i].label, "freq"))
freq = i;
else if (gain == -1 &&
!strcmp(ui[0]->elems[i].label, "gain"))
gain = i;
else if (gate == -1 &&
!strcmp(ui[0]->elems[i].label, "gate"))
gate = i;
else {
noinstr:
std::map< int, list<strpair> >::iterator it =
ui[0]->metadata.find(i);
if (it != ui[0]->metadata.end()) {
// Scan for controller mappings and other control meta data.
for (std::list<strpair>::iterator jt = it->second.begin();
jt != it->second.end(); jt++) {
const char *key = jt->first, *val = jt->second;
#if DEBUG_META
fprintf(stderr, "ctrl '%s' meta: '%s' -> '%s'\n",
ui[0]->elems[i].label, key, val);
#endif
if (strcmp(key, "unit") == 0) {
unit = val;
#if FAUST_MIDICC
} else if (strcmp(key, "midi") == 0) {
unsigned num;
if (sscanf(val, "ctrl %u", &num) < 1) continue;
#if 0 // enable this to get feedback about controller assignments
const char *dsp_name = pluginName();
fprintf(stderr, "%s: cc %d -> %s\n", dsp_name, num,
ui[0]->elems[i].label);
#endif
ctrlmap.insert(std::pair<uint8_t,int>(num, p));
#endif
}
}
}
ctrls[j++] = i;
inctrls[p++] = i;
int p = ui[0]->elems[i].port;
float val = ui[0]->elems[i].init;
assert(p>=0);
portvals[p] = ports[p] = val;
units[p] = unit;
for (int ch = 0; ch < 16; ch++)
midivals[ch][p] = val;
}
break;
}
}
// Realloc the inctrls and outctrls vectors to their appropriate sizes.
inctrls = (int*)realloc(inctrls, p*sizeof(int));
assert(p == 0 || inctrls);
outctrls = (int*)realloc(outctrls, q*sizeof(int));
assert(q == 0 || outctrls);
n_in = p; n_out = q;
if (maxvoices > 0) {
// Initialize the mixdown buffer.
outbuf = (float**)calloc(m, sizeof(float*));
assert(m == 0 || outbuf);
// We start out with a blocksize of 512 samples here. Hopefully this is
// enough for most realtime hosts so that we can avoid reallocations
// later when we know what the actual blocksize is.
n_samples = 512;
for (int i = 0; i < m; i++) {
outbuf[i] = (float*)malloc(n_samples*sizeof(float));
assert(outbuf[i]);
}
// Initialize a 1-sample dummy input buffer used for retriggering notes.
inbuf = (float**)calloc(n, sizeof(float*));
assert(n == 0 || inbuf);
for (int i = 0; i < m; i++) {
inbuf[i] = (float*)malloc(sizeof(float));
assert(inbuf[i]);
*inbuf[i] = 0.0f;
}
}
}
~VSTPlugin()
{
const int n = dsp[0]->getNumInputs();
const int m = dsp[0]->getNumOutputs();
for (int i = 0; i < ndsps; i++) {
delete dsp[i];
delete ui[i];
}
free(ctrls);
free(inctrls);
free(outctrls);
free(ports);
free(portvals);
free(units);
for (int ch = 0; ch < 16; ch++)
free(midivals[ch]);
if (inbuf) {
for (int i = 0; i < n; i++)
free(inbuf[i]);
free(inbuf);
}
if (outbuf) {
for (int i = 0; i < m; i++)
free(outbuf[i]);
free(outbuf);
}
free(dsp);
free(ui);
if (vd) {
free(vd->note_info);
free(vd->lastgate);
delete vd;
}
}
// Voice allocation.
#if DEBUG_VOICE_ALLOC
void print_voices(const char *msg)
{
fprintf(stderr, "%s: notes =", msg);
for (uint8_t ch = 0; ch < 16; ch++)
for (int note = 0; note < 128; note++)
if (vd->notes[ch][note] >= 0)
fprintf(stderr, " [%d] %d(#%d)", ch, note, vd->notes[ch][note]);
fprintf(stderr, "\nqueued (%d):", vd->queued.size());
for (int i = 0; i < nvoices; i++)
if (vd->queued.find(i) != vd->queued.end()) fprintf(stderr, " #%d", i);
fprintf(stderr, "\nused (%d):", vd->n_used);
for (boost::circular_buffer<int>::iterator it = vd->used_voices.begin();
it != vd->used_voices.end(); it++)
fprintf(stderr, " #%d->%d", *it, vd->note_info[*it].note);
fprintf(stderr, "\nfree (%d):", vd->n_free);
for (boost::circular_buffer<int>::iterator it = vd->free_voices.begin();
it != vd->free_voices.end(); it++)
fprintf(stderr, " #%d", *it);
fprintf(stderr, "\n");
}
#endif
int alloc_voice(uint8_t ch, int8_t note, int8_t vel)
{
int i = vd->notes[ch][note];
if (i >= 0) {
// note already playing on same channel, retrigger it
voice_off(i);
voice_on(i, note, vel, ch);
// move this voice to the end of the used list
for (boost::circular_buffer<int>::iterator it =
vd->used_voices.begin();
it != vd->used_voices.end(); it++) {
if (*it == i) {
vd->used_voices.erase(it);
vd->used_voices.push_back(i);
break;
}
}
#if DEBUG_VOICE_ALLOC
print_voices("retrigger");
#endif
return i;
} else if (vd->n_free > 0) {
// take voice from free list
int i = vd->free_voices.front();
vd->free_voices.pop_front();
vd->n_free--;
vd->used_voices.push_back(i);
vd->note_info[i].ch = ch;
vd->note_info[i].note = note;
vd->n_used++;
voice_on(i, note, vel, ch);
vd->notes[ch][note] = i;
#if DEBUG_VOICE_ALLOC
print_voices("alloc");
#endif
return i;
} else {
// steal a voice
assert(vd->n_used > 0);
// FIXME: Maybe we should look for the oldest note on the *current*
// channel here, but this is faster.
int i = vd->used_voices.front();
int oldch = vd->note_info[i].ch;
int oldnote = vd->note_info[i].note;
voice_off(i);
vd->notes[oldch][oldnote] = -1;