ofxVSTPlugin(audioMasterCallback audioMaster): AudioEffectX (audioMaster, 1, 1) {
plugin = ofxAudioPlugin_getPlugin(this);
#ifdef ofxAudioPlugin_IsSynth
setNumInputs(0);
isSynth();
#else
setNumInputs (2); // stereo in
#endif
setNumOutputs (2); // stereo out
setUniqueID (ofxAudioPlugin_PluginCode); // identify
canProcessReplacing (); // supports replacing output
fGain = 1.f; // default to 0 dB
vst_strncpy (programName, "Default", kVstMaxProgNameLen); // default program name
cEffect.numPrograms = 3;
cEffect.numParams = plugin->getNumParameters();
cEffect.numParams = 1;
parametersChanged();
}
//-------------------------------------------------------------------------------------------------------
e6::e6 (audioMasterCallback audioMaster)
: AudioEffectX (audioMaster, 1, nParams ) // 1 program, 1 parameter only
{
setNumInputs (0); // no in
setNumOutputs (0); // no out
setUniqueID ('e6e6'); // identify
// we don't do audio !
canProcessReplacing (false); // supports replacing output
canDoubleReplacing (false); // supports double precision processing
for ( int id=0; id<nParams; id++ )
{
sprintf( params[id].name,"param_%d", id );
strcpy( params[id].label, "db" );
params[id].value = 0.0f;
}
vst_strncpy (programName, "Default", kVstMaxProgNameLen); // default program name
logFile = fopen("e6.log", "wb" );
int res = sock.connect( "localhost", 9999 );
if ( res = SOCKET_ERROR )
{
fprintf( logFile, "could not connect to localhost on 9999\n" );
fflush( logFile );
}
else
{
fprintf( logFile, "connected to localhost on 9999\n" );
fflush( logFile );
}
}
SorolletVSTi::SorolletVSTi(audioMasterCallback audioMaster) : AudioEffectX(audioMaster, kNumPrograms, kNumParams)
{
if (audioMaster)
{
// TODO: load default / more varied programs!
// initialize programs
programs = new SorolletVSTiProgram[kNumPrograms];
for (VstInt32 i = 0; i < 16; i++)
channelPrograms[i] = i;
mSorolletVoice = new SorolletVoice();
if (programs)
{
setProgram(0);
}
setNumInputs(0); // no inputs
setNumOutputs(kNumOutputs); // 2 outputs
canProcessReplacing();
isSynth();
setUniqueID('so02');
currentEvent = EVENT_NULL;
initProcess();
suspend();
}
}
//-----------------------------------------------------------------------------
SurroundDelay::SurroundDelay (audioMasterCallback audioMaster)
: ADelay (audioMaster)
, plugInput (0)
, plugOutput (0)
{
// The first buffer is allocated in ADelay's constructor
for (int i = 1; i < MAX_CHANNELS; i++)
{
sBuffers[i] = new float[size];
}
setNumInputs (MAX_CHANNELS);
setNumOutputs (MAX_CHANNELS);
// We initialize the arrangements to default values.
// Nevertheless, the host should modify them via
// appropriate calls to setSpeakerArrangement.
allocateArrangement (&plugInput, MAX_CHANNELS);
plugInput->type = kSpeakerArr51;
allocateArrangement (&plugOutput, MAX_CHANNELS);
plugOutput->type = kSpeakerArr51;
setUniqueID ('SDlE'); // this should be unique, use the Steinberg web page for plugin Id registration
// create the editor
editor = new SDEditor (this);
resume ();
}
mdaDetune::mdaDetune(audioMasterCallback audioMaster): AudioEffectX(audioMaster, NPROGS, NPARAMS)
{
setNumInputs(2);
setNumOutputs(2);
setUniqueID('mdat'); ///identify mdaDetune-in here
DECLARE_VST_DEPRECATED(canMono) ();
canProcessReplacing();
programs[0].param[0] = 0.20f; //fine
programs[0].param[1] = 0.90f; //mix
programs[0].param[2] = 0.50f; //output
programs[0].param[3] = 0.50f; //chunksize
strcpy(programs[0].name, "Stereo Detune");
programs[1].param[0] = 0.20f;
programs[1].param[1] = 0.90f;
programs[1].param[2] = 0.50f;
programs[1].param[3] = 0.50f;
strcpy(programs[1].name,"Symphonic");
programs[2].param[0] = 0.8f;
programs[2].param[1] = 0.7f;
programs[2].param[2] = 0.50f;
programs[2].param[3] = 0.50f;
strcpy(programs[2].name,"Out Of Tune");
///initialise...
curProgram=0;
suspend();
semi = 3.0f * 0.20f * 0.20f * 0.20f;
dpos2 = (float)pow(1.0594631f, semi);
dpos1 = 1.0f / dpos2;
wet = 1.0f;
dry = wet - wet * 0.90f * 0.90f;
wet = (wet + wet - wet * 0.90f) * 0.90f;
}
mdaAmbience::mdaAmbience(audioMasterCallback audioMaster) : AudioEffectX(audioMaster, 1, 4) // programs, parameters
{
//inits here!
fParam0 = 0.7f; //size
fParam1 = 0.7f; //hf
fParam2 = 0.9f; //mix
fParam3 = 0.5f; //output
size = 0;
buf1 = new float[1024];
buf2 = new float[1024];
buf3 = new float[1024];
buf4 = new float[1024];
fil = 0.0f;
den = pos=0;
setNumInputs(2);
setNumOutputs(2);
setUniqueID("mdaAmb"); // identify here
DECLARE_LVZ_DEPRECATED(canMono) ();
canProcessReplacing();
strcpy(programName, "Small Space Ambience");
suspend(); // flush buffer
setParameter(0, 0.7f); //go and set initial values!
}
//-----------------------------------------------------------------------------
// Plug-In constructor
// 1) create programs
// 2) initialize Plug-in
CSoftExciter::CSoftExciter (audioMasterCallback audioMaster)
: AudioEffectX (audioMaster, kNumPrograms, kNumParams)
{
// create our inner plugin
//
// This also initializes all the controls; the plugin also
// stores all its presets so no need to write those objects any more!
m_pRAFXPlugIn = new CHarmonicSecond;
m_pRAFXPlugIn->initialize();
// if you are sr dependent
m_fSampleRate = getSampleRate();
// stereo IN
setNumInputs(2);
// stereo OUT
setNumOutputs(2);
// our GUID
setUniqueID ('ALNS');
// IF you have a custom GUI, un-comment this line otherwise will default to
// standard I/F
// cEffect.flags |= effFlagsHasEditor;
// PAUSE function
suspend (); // flush buffer
}
TimedFader::TimedFader(audioMasterCallback audioMaster) : AudioEffectX(audioMaster, 1, 3)
{
//inits here!
fOutput = 1.0f;
faderSteps = 15;
currentFaderFrameCount = 1;
setNumInputs(2);
setNumOutputs(2);
setUniqueID('tFd1'); // identify here
DECLARE_VST_DEPRECATED(canMono) ();
canProcessReplacing();
strcpy(programName, "Timed Fader 1");
setParameter(PARAM_PlayDuration, 30.0f); // 10 seconds //0.166667); //~30 seconds
setParameter(PARAM_FadeDuration, 3.0f); // 3 seconds
fPlaySecondsRemaining = fPlayDuration;
fFadeSecondsRemaining = fFadeDuration;
state = STATE_DONE;
muteGain = (float)pow(10.0f, 0.05f * -60.0f);
muteGain *= 0.01f;
fSecondsPerFrame = 1 / sampleRate;
// Create the editor
faderDisplay = new FaderDisplay(this);
editor = faderDisplay;
suspend(); // flush buffer
}
mdaRezFilter::mdaRezFilter(audioMasterCallback audioMaster) : AudioEffectX(audioMaster, 1, 10) // programs, parameters
{
//inits here!
fParam0 = 0.33f; //f
fParam1 = 0.70f; //q
fParam2 = 0.50f; //a
fParam3 = 0.85f; //fenv
fParam4 = 0.00f; //att
fParam5 = 0.50f; //rel
fParam6 = 0.70f; //lfo
fParam7 = 0.40f; //rate
fParam8 = 0.00f; //trigger
fParam9 = 0.75f; //max freq
fff = fq = fg = fmax = env = fenv = att = rel = 0.0f;
flfo = phi = dphi = bufl = buf0 = buf1 = buf2 = tthr = env2 = 0.0f;
lfomode = ttrig = tatt = 0;
setNumInputs(2);
setNumOutputs(2);
setUniqueID("mdaRezFilter");
DECLARE_LVZ_DEPRECATED(canMono) ();
canProcessReplacing();
strcpy(programName, "Resonant Filter");
suspend(); // flush buffer
setParameter(2, 0.5f); //go and set initial values!
}
NitroSynth::NitroSynth(audioMasterCallback pCallback)
: AudioEffectX(pCallback, 0, 128),
fSamplePeriod(1.0f/44100.0f),
bRecording(false),
iRecorderState(IDLE)
{
if ( pCallback )
{
setNumInputs(0);
setNumOutputs(2);
canProcessReplacing();
isSynth();
setUniqueID('SRTN');
}
for ( int i = 0 ; i < 8 ; i++ )
if ( i < 6 )
pInstruments[i] = new Instrument(new WaveDuty(), & pParameters[i]);
else
pInstruments[i] = new Instrument(new Noise(), & pParameters[i]);
#ifndef NO_VST_GUI
setEditor(new NitroSynthInterface(this));
#endif // NO_VST_GUI
}
mdaTalkBox::mdaTalkBox(audioMasterCallback audioMaster): AudioEffectX(audioMaster, NPROGS, NPARAMS)
{
setNumInputs(2);
setNumOutputs(2);
setUniqueID('mda&'); ///identify plug-in here
//canMono();
canProcessReplacing();
///initialise...
buf0 = new float[BUF_MAX];
buf1 = new float[BUF_MAX];
window = new float[BUF_MAX];
car0 = new float[BUF_MAX];
car1 = new float[BUF_MAX];
N = 1; //trigger window recalc
K = 0;
programs = new mdaTalkBoxProgram[NPROGS];
if(programs)
{
///differences from default program...
//programs[1].param[0] = 0.1f;
//strcpy(programs[1].name,"Another Program");
setProgram(0);
}
suspend();
}
mdaCombo::mdaCombo(audioMasterCallback audioMaster) : AudioEffectX(audioMaster, 1, 7) // programs, parameters
{
//inits here!
fParam1 = 1.00f; //select
fParam2 = 0.50f; //drive
fParam3 = 0.50f; //bias
fParam4 = 0.50f; //output
fParam5 = 0.00f; //stereo
fParam6 = 0.00f; //hpf freq
fParam7 = 0.50f; //hpf reso
size = 1024;
bufpos = 0;
buffer = new float[size];
buffe2 = new float[size];
setNumInputs(2);
setNumOutputs(2);
setUniqueID("mdaCombo"); // identify here
DECLARE_LVZ_DEPRECATED(canMono) ();
canProcessReplacing();
strcpy(programName, "Amp & Speaker Simulator");
suspend(); // flush buffer
setParameter(0, 0.f); //go and set initial values!
}
mdaLimiter::mdaLimiter(audioMasterCallback audioMaster) : AudioEffectX(audioMaster, 1, 5) // 1 program, 4 parameters
{
fParam1 = (float)0.60; //thresh
fParam2 = (float)0.60; //trim
fParam3 = (float)0.15; //attack
fParam4 = (float)0.50; //release
fParam5 = (float)0.40; //knee
setNumInputs(2); // stereo in
setNumOutputs(2); // stereo out
setUniqueID('mdaL'); // identify
DECLARE_VST_DEPRECATED(canMono) ();
canProcessReplacing(); // supports both accumulating and replacing output
strcpy(programName, "Limiter"); // default program name
if(fParam5>0.5) //soft knee
{
thresh = (float)pow(10.0, 1.0 - (2.0 * fParam1));
}
else //hard knee
{
thresh = (float)pow(10.0, (2.0 * fParam1) - 2.0);
}
trim = (float)(pow(10.0, (2.0 * fParam2) - 1.0));
att = (float)pow(10.0, -0.01 - 2.0 * fParam3);//wavelab overruns with zero???
rel = (float)pow(10.0, -2.0 - (3.0 * fParam4));
gain = 1.0;
}
mdaTracker::mdaTracker(audioMasterCallback audioMaster) : AudioEffectX(audioMaster, 1, 8) // programs, parameters
{
//inits here!
fParam1 = (float)0.00; //Mode
fParam2 = (float)1.00; //Dynamics
fParam3 = (float)1.00; //Mix
fParam4 = (float)0.97; //Tracking
fParam5 = (float)0.50; //Trnspose
fParam6 = (float)0.80; //Maximum Hz
fParam7 = (float)0.50; //Trigger dB
fParam8 = (float)0.50; //Output
setNumInputs(2);
setNumOutputs(2);
setUniqueID("mdaTracker"); // identify here
DECLARE_LVZ_DEPRECATED(canMono) ();
canProcessReplacing();
strcpy(programName, "Pitch Tracker");
dphi = 100.f/getSampleRate(); //initial pitch
min = (int32_t)(getSampleRate()/30.0); //lower limit
res1 = (float)cos(0.01); //p
res2 = (float)sin(0.01); //q
fi = fo = thr = phi = ddphi = trans = buf1 = buf2 = dn = bold = wet = dry = 0.0f;
dyn = env = rel = saw = dsaw = res1 = res2 = buf3 = buf4 = 0.0f;
max = min = num = sig = mode = 0;
setParameter(0, 0.0f);
}
//-------------------------------------------------------------------------------------------------------
gsBEncoder::gsBEncoder (audioMasterCallback audioMaster)
: AudioEffectX (audioMaster, 1, num_params()) // 1 program, N parameters
, gen(0)
, bufferSize(512)
, sampleRate(44100)
, blockSize(512)
{
inputBuffers.resize(num_inputs());
outputBuffers.resize(num_outputs());
for(int i=0; i < inputBuffers.size(); i++) {
inputBuffers[i] = new t_sample[bufferSize];
}
for(int i=0; i < outputBuffers.size(); i++) {
outputBuffers[i] = new t_sample[bufferSize];
}
gen = (CommonState *)create(sampleRate, blockSize);
setNumInputs (num_inputs());
setNumOutputs (num_outputs());
setUniqueID ('GSE_I'); // identify
canProcessReplacing (); // supports replacing output
canDoubleReplacing (); // supports double precision processing
vst_strncpy (programName, VST_NAME, kVstMaxProgNameLen); // default program name
}
CLlconVST::CLlconVST ( audioMasterCallback AudioMaster ) :
AudioEffectX ( AudioMaster, 1, 0 ), // 1 program with no parameters (=0)
Client ( LLCON_DEFAULT_PORT_NUMBER )
{
// stereo input/output
setNumInputs ( 2 );
setNumOutputs ( 2 );
setUniqueID ( 'Llco' );
// capabilities of llcon VST plugin
canProcessReplacing(); // supports replacing output
// set default program name
GetName ( strProgName );
// we want a single shot timer to shut down the connection if no
// processing is done anymore (VST host has stopped the stream)
TimerOnOff.setSingleShot ( true );
TimerOnOff.setInterval ( VST_STOP_TIMER_INTERVAL );
// connect timer event
connect ( &TimerOnOff, SIGNAL ( timeout() ),
this, SLOT ( OnTimerOnOff() ) );
// TODO settings
Client.SetServerAddr ( DEFAULT_SERVER_ADDRESS );
}
LowPass::LowPass(audioMasterCallback audioMaster)
: AudioEffectX(audioMaster, NUM_PRESETS, kNumParams)
{
programs = new LowPassProgram[numPrograms];
sr = getSampleRate();
if(sr<0.f) sr = 44100.f;
fcutoff = log(cutoffDefault/cutoffMin)/(log(cutoffMax/cutoffMin));
// instantiate callback
// Initialize the internal states
lambda = lp1 = lp2 = 0.;
if(programs) setProgram(0);
setUniqueID(ID);
setNumInputs(NUM_INPUTS);
setNumOutputs(NUM_OUTPUTS);
canProcessReplacing(CAN_PROCESS_REPLACING);
canMono(CAN_MONO);
}
mdaLooplex::mdaLooplex(audioMasterCallback audioMaster) : AudioEffectX(audioMaster, NPROGS, NPARAMS)
{
Fs = 44100.0f; //won't we know the sample rate by the time we need it?
//initialise...
bypass = bypassed = busy = status = 0; //0=not recorded 1=first pass 2=looping
bufmax = 882000;
buffer = new short[bufmax + 10];
memset(buffer, 0, (bufmax + 10) * sizeof(short));
bufpos = 0;
buflen = 0;
recreq = 0;
oldParam0 = oldParam1 = oldParam2 = 0.0f;
modwhl = 1.0f;
programs = new mdaLooplexProgram[NPROGS];
if(programs) setProgram(0);
if(audioMaster)
{
setNumInputs(NOUTS);
setNumOutputs(NOUTS);
canProcessReplacing();
//needIdle(); idle is broken in VST2.4
setUniqueID('MDA~'); ///
}
update();
suspend();
idleList.next = new IdleList(this, idleList.next); //add to idle list, start timer if not running...
}
Silence::Silence(audioMasterCallback audio_master)
: AudioEffectX(audio_master, 1, 1) {
setNumInputs(2);
setNumOutputs(2);
setUniqueID('Silence');
canProcessReplacing();
canDoubleReplacing();
vst_strncpy(program_name, "Default", kVstMaxProgNameLen);
}
mdaBandisto::mdaBandisto(audioMasterCallback audioMaster) : AudioEffectX(audioMaster, 1, 10) // programs, parameters
{
//inits here!
fParam1 = (float)1.00; //Listen: L/M/H/out
fParam2 = (float)0.40; //xover1
fParam3 = (float)0.50; //xover2
fParam4 = (float)0.50; //L drive (1)
fParam5 = (float)0.50; //M drive
fParam6 = (float)0.50; //H drive
fParam7 = (float)0.50; //L trim (2)
fParam8 = (float)0.50; //M trim
fParam9 = (float)0.50; //H trim
fParam10 = (float)0.0; //unipolar/bipolar
setNumInputs(2);
setNumOutputs(2);
setUniqueID("mdaBand"); // identify here
DECLARE_LVZ_DEPRECATED(canMono) ();
canProcessReplacing();
strcpy(programName, "Multi-Band Distortion");
//calcs here!
driv1 = (float)pow(10.0,(6.0 * fParam4 * fParam4) - 1.0);
driv2 = (float)pow(10.0,(6.0 * fParam5 *fParam5) - 1.0);
driv3 = (float)pow(10.0,(6.0 * fParam6 *fParam6) - 1.0);
valve = int(fParam10 > 0.0);
if(valve)
{
trim1 = (float)(0.5);
trim2 = (float)(0.5);
trim3 = (float)(0.5);
}
else
{
trim1 = 0.3f*(float)pow(10.0,(4.0 * pow(fParam4,3.f)));//(0.5 + 500.0 * pow(fParam4,6.0));
trim2 = 0.3f*(float)pow(10.0,(4.0 * pow(fParam5,3.f)));
trim3 = 0.3f*(float)pow(10.0,(4.0 * pow(fParam6,3.f)));
}
trim1 = (float)(trim1 * pow(10.0, 2.0 * fParam7 - 1.0));
trim2 = (float)(trim2 * pow(10.0, 2.0 * fParam8 - 1.0));
trim3 = (float)(trim3 * pow(10.0, 2.0 * fParam9 - 1.0));
switch(int(fParam1*3.9))
{
case 0: trim2=0.0; trim3=0.0; slev=0.0; break;
case 1: trim1=0.0; trim3=0.0; slev=0.0; break;
case 2: trim1=0.0; trim2=0.0; slev=0.0; break;
default: slev=0.5; break;
}
fi1 = (float)pow(10.0,fParam2 - 1.70); fo1=(float)(1.0 - fi1);
fi2 = (float)pow(10.0,fParam3 - 1.05); fo2=(float)(1.0 - fi2);
fb1 = fb2 = fb3 = 0.0f;
}
//-------------------------------------------------------------------------------------------------------
JackVST::JackVST (audioMasterCallback audioMaster)
: AudioEffectX (audioMaster, 1, 1) // 1 program, 1 parameter only
{
fGain = 1.; // default to 0 dB
setNumInputs (2); // stereo in
setNumOutputs (2); // stereo out
setUniqueID ('JACK-ASinsert'); // identify
canMono (); // makes sense to feed both inputs with the same signal
canProcessReplacing (); // supports both accumulating and replacing output
strcpy (fProgramName, "Default"); // default program name
fStatus = kIsOff;
}
Convolver2::Convolver2(audioMasterCallback audioMaster)
: AudioEffectX(audioMaster, 0, NUM_PARAMS) {
setNumInputs(NUM_INPUTS);
setNumOutputs(NUM_OUTPUTS);
setUniqueID(UNIQUE_ID);
editor = new Convolver2Editor(this);
core = new Convolver2Core(NUM_PARAMS, VERSION, DEF_PRODUCT);
core->setParameter(PRM_SAMPLE_RATE, getSampleRate(), true);
((Convolver2Editor*)editor)->setCore(core);
((Convolver2Core*)core)->setEditor((Convolver2Editor*)editor);
}
//------------------------------------------------------------------------------------
TutorialVST2Effect::TutorialVST2Effect (audioMasterCallback audioMaster)
: AudioEffectX (audioMaster, 1, kNumParameters)
{
setUniqueID (CCONST('G', 'U', 'I', '0'));
setNumInputs (2);
setNumOutputs (2);
parameters[kLeftVolumeParameter] = 1.f;
parameters[kRightVolumeParameter] = 1.f;
extern AEffGUIEditor* createEditor (AudioEffectX*);
setEditor (createEditor (this));
}
Vst2413r::Vst2413r(audioMasterCallback audioMaster)
: AudioEffectX(audioMaster, 0, 0),
driver_(44100)
{
if(audioMaster != NULL) {
setNumInputs(0);
setNumOutputs(1);
setUniqueID(kUniqueId);
canProcessReplacing();
isSynth();
}
suspend();
}
//-------------------------------------------------------------------------------------------------------
JackVST::JackVST (audioMasterCallback audioMaster)
: AudioEffectX (audioMaster, 1, 1),c_jar(NULL),c_error(JARInsert::kNoErr)// 1 program, 1 parameter only
{
fGain = 1.; // default to 0 dB
setNumInputs(2); // stereo in
setNumOutputs(2); // stereo out
setUniqueID('JACK'); // identify
canProcessReplacing(); // supports both accumulating and replacing output
strcpy(programName, "Default"); // default program name
c_jar = new JARInsert('vst ');
c_error = c_jar->GetError();
}
//-------------------------------------------------------------------------------------------------------
AGain::AGain (audioMasterCallback audioMaster)
: AudioEffectX (audioMaster, 1, 11) // 1 program, 1 parameter only
{
setNumInputs (2); // stereo in
setNumOutputs (2); // stereo out
setUniqueID ('Band'); // identify
canProcessReplacing (); // supports replacing output
canDoubleReplacing (); // supports double precision processing
fGain = 1.f; // default to 0 dB
vst_strncpy (programName, "Default", kVstMaxProgNameLen); // default program name
UdpLog::setup( 7777, "localhost" );
}
mdaLoudness::mdaLoudness(audioMasterCallback audioMaster): AudioEffectX(audioMaster, NPROGS, NPARAMS)
{
setNumInputs(2);
setNumOutputs(2);
setUniqueID('mdal');
DECLARE_VST_DEPRECATED(canMono) ();
canProcessReplacing();
programs = new mdaLoudnessProgram[NPROGS];
setProgram(0);
suspend();
}
plugin::plugin(audioMasterCallback audioMaster)
: AudioEffectX(audioMaster, 1, PARAMETERS)
{
setNumInputs(2);
setNumOutputs(2);
setUniqueID(ID);
canProcessReplacing();
canDoubleReplacing();
vst_strncpy(programName, "Init", kVstMaxProgNameLen);
#if (PARAMETERS > 0)
::construct_params(params);
#endif
::construct(&data);
}
//------------------------------------------------------------------------------
Reverb::Reverb (audioMasterCallback audioMaster)
: AudioEffectX (audioMaster, 1, kNumParams) // 1 program, 1 parameter only
{
setNumInputs (kNumInputs); // stereo in
setNumOutputs (kNumOutputs); // stereo out
setUniqueID ('Rvrb'); // identify
canProcessReplacing (); // supports replacing output
vst_strncpy (programName, "Default", kVstMaxProgNameLen); // default program name
// internal state var declaration and initialization
fs = getSampleRate();
pcoefs = coefs; // pointer for filter coef assignment
T60LowValue = 3.5; // time, in seconds, for low freqs to decay 60dB
T60LowKnob = SmartKnob::value2knob(T60LowValue, T60LowLimits, T60LowTaper);
T60HighValue = 0.5; // time, in seconds, for high freqs to decay 60dB
T60HighKnob = SmartKnob::value2knob(T60HighValue, T60HighLimits, T60HighTaper);
TransitionValue = 6700.0; // transition freq, in Hz, between low and high freqs
TransitionKnob = SmartKnob::value2knob(TransitionValue, TransitionLimits, TransitionTaper);
WetDryKnob = 0.2; // output (wet/dry) mix
for(int i=0; i<kNumDelays; i++){
dl[i].SetDelay(dlens[i]); // set reverb delay lengths
designShelf(pcoefs,dlens[i], TransitionValue, T60LowValue, T60HighValue); // design filters for feedback loop
fbfilt[i].SetCoefs(coefs); // assign filter coefs
}
ParametricFcValue = 5000.0;
ParametricFcKnob = SmartKnob::value2knob(ParametricFcValue, ParametricFcLimits, ParametricFcTaper);
ParametricGammaValue = dB2mag(0.0);
ParametricGammaKnob = SmartKnob::value2knob(ParametricGammaValue, ParametricGammaLimits, ParametricGammaTaper);
ParametricQValue = 1;
ParametricQKnob = SmartKnob::value2knob(ParametricQValue, ParametricQLimits, ParametricQTaper);
designParametric(parametric_coefs, ParametricFcValue, ParametricGammaValue, ParametricQValue);
parametric[0].setCoefs(parametric_coefs);
parametric[1].setCoefs(parametric_coefs);
}
mdaOverdrive::mdaOverdrive(audioMasterCallback audioMaster) : AudioEffectX(audioMaster, 1, 3) // 1 program, 3 parameters
{
fParam1 = 0.0f;
fParam2 = 0.0f;
fParam3 = 0.5f;
setNumInputs(2);
setNumOutputs(2);
setUniqueID('mdaO'); // identify
DECLARE_VST_DEPRECATED(canMono) ();
canProcessReplacing();
strcpy(programName, "Soft Overdrive");
filt1 = filt2 = 0.0f;
setParameter(0, 0.0f);
}