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FM_BISON.cpp
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FM_BISON.cpp
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/*
FM. BISON hybrid FM synthesis
(C) njdewit technologies (visualizers.nl) & bipolaraudio.nl
MIT license applies, please see https://en.wikipedia.org/wiki/MIT_License or LICENSE in the project root!
*/
// Shut it, MSVC
#ifndef _CRT_SECURE_NO_WARNINGS
#define _CRT_SECURE_NO_WARNINGS
#endif
#include "FM_BISON.h"
#include "synth-global.h"
#include "patch/synth-patch-global.h"
#include "synth-DX7-LFO-table.h"
namespace SFM
{
static bool s_performStaticInit = true;
/* ----------------------------------------------------------------------------------------------------
Constructor/Destructor
------------------------------------------------------------------------------------------------------ */
Bison::Bison()
{
if (true == s_performStaticInit)
{
// Calculate LUTs & initialize random generator
InitializeRandomGenerator();
CalculateMIDIToFrequencyLUT();
InitializeFastCosine();
Supersaw::CalculateDetuneTable();
s_performStaticInit = false;
}
// Reset entire patch
m_patch.ResetToEngineDefaults();
// Initialize polyphony
const bool monophonic = Patch::VoiceMode::kMono == m_patch.voiceMode;
m_curPolyphony = (false == monophonic) ? m_patch.maxPolyVoices : 1;
Log("Instance of FM. BISON engine initalized");
Log("Suzie, call DR. BISON, tell him it's for me...");
/*
IMPORTANT: call SetSamplingProperties() before Render()
*/
}
Bison::~Bison()
{
DeleteRateDependentObjects();
Log("Instance of FM. BISON engine released");
}
/* ----------------------------------------------------------------------------------------------------
OnSetSamplingProperties() is called by JUCE, but what it does is set up the basic properties
necessary to start rendering so it can be used in any situation
------------------------------------------------------------------------------------------------------ */
// Called by JUCE's prepareToPlay()
void Bison::OnSetSamplingProperties(unsigned sampleRate, unsigned samplesPerBlock)
{
Log("BISON::OnSetSamplingProperties(" + std::to_string(sampleRate) + ", " + std::to_string(samplesPerBlock) + ")");
m_sampleRate = sampleRate;
m_samplesPerBlock = samplesPerBlock;
m_Nyquist = sampleRate>>1;
/*
Reset sample rate dependent global objects
*/
DeleteRateDependentObjects();
// Stop & reset all voices, clear slots & wipe requests
for (unsigned iVoice = 0; iVoice < kMaxPolyVoices; ++iVoice)
m_voices[iVoice].Reset(m_sampleRate);
for (unsigned iSlot = 0; iSlot < 128; ++iSlot)
m_keyToVoice[iSlot] = -1;
m_polyVoiceReq.clear();
m_polyVoiceReleaseReq.clear();
m_resetVoices = false;
// Reset BPM
m_BPM = 0.0;
m_resetPhaseBPM = true;
// Voice mode
m_curVoiceMode = m_patch.voiceMode;
// Reset monophonic state
m_monoSequence.clear();
m_monoVoiceReq.key = VoiceRequest::kInvalid;
m_monoVoiceReleaseReq.key = MonoVoiceReleaseRequest::kInvalid;
// Reset filter type
m_curFilterType = SvfLinearTrapOptimised2::NO_FLT_TYPE;
// Allocate intermediate buffers (a pair for each thread)
m_pBufL[0] = reinterpret_cast<float *>(mallocAligned(m_samplesPerBlock*sizeof(float), 16));
m_pBufL[1] = reinterpret_cast<float *>(mallocAligned(m_samplesPerBlock*sizeof(float), 16));
m_pBufR[0] = reinterpret_cast<float *>(mallocAligned(m_samplesPerBlock*sizeof(float), 16));
m_pBufR[1] = reinterpret_cast<float *>(mallocAligned(m_samplesPerBlock*sizeof(float), 16));
// Create effects
m_postPass = new PostPass(m_sampleRate, m_samplesPerBlock, m_Nyquist);
// Start global LFO phase
m_globalLFO = new Phase(m_sampleRate);
const float freqLFO = MIDI_To_DX7_LFO_Hz(m_patch.LFORate);
m_globalLFO->Initialize(freqLFO, m_sampleRate);
// Reset global interpolated parameters
m_curLFOBlend = { m_patch.LFOBlend, m_sampleRate, kDefParameterLatency, 0.f, 1.f };
m_curLFOModDepth = { m_patch.LFOModDepth, m_sampleRate, kDefParameterLatency, 0.f, 1.f };
m_curCutoff = { SVF_CutoffToHz(m_patch.cutoff, m_Nyquist), m_sampleRate, kDefParameterLatency * 10.f /* Longer */, kSVFMinFilterCutoffHz, kSVFMaxFilterCutoffHz};
m_curQ = { SVF_ResoToQ(m_patch.resonance), m_sampleRate, kDefParameterLatency, kSVFMinFilterQ, kSVFMaxFilterQ};
m_curPitchBend = { 0.f, m_sampleRate, kDefParameterLatency, -1.f, 1.f };
m_curAmpBend = { 1.f /* 0dB */, m_sampleRate, kDefParameterLatency, 0.f, 1.f };
m_curModulation = { 0.f, m_sampleRate, kDefParameterLatency * 1.5f /* Longer */, 0.f, 1.f };
m_curAftertouch = { 0.f, m_sampleRate, kDefParameterLatency * 3.f /* Longer */, 0.f, 1.f };
// Reset operator peaks (visualization)
for (float &peak : m_opPeaks)
peak = 0.f;
}
// Cleans up after OnSetSamplingProperties()
void Bison::DeleteRateDependentObjects()
{
// Allocate intermediate sample buffers
if (nullptr != m_pBufL[0]) // Good enough to assume
{
freeAligned(m_pBufL[0]);
freeAligned(m_pBufL[1]);
freeAligned(m_pBufR[0]);
freeAligned(m_pBufR[1]);
}
m_pBufL[0] = m_pBufL[1] = m_pBufR[0] = m_pBufR[1] = nullptr;
// Release post-pass
delete m_postPass;
m_postPass = nullptr;
// Delete global LFO
delete m_globalLFO;
m_globalLFO = nullptr;
}
/* ----------------------------------------------------------------------------------------------------
Voice management
Polyphonic mode works as you'd expect
Monophonic mode:
- Each note uses it's own velocity for all calculations, instead of carrying over the
initial one (like I can hear my Yamaha TG77 do)
- When a note is released, the last note in the sequence will play; to restart lift all keys
- Glide speed and velocity attenuation can be controlled through parameters
------------------------------------------------------------------------------------------------------ */
// Release voice, but retain key slot
void Bison::ReleaseVoice(int index)
{
SFM_ASSERT(index >= 0 && index < kMaxPolyVoices);
Voice &voice = m_voices[index];
SFM_ASSERT(true == voice.IsPlaying());
// Lift possible sustain & release
voice.m_sustained = false;
voice.OnRelease();
const int key = voice.m_key;
Log("Voice released: " + std::to_string(index) + " for key: " + std::to_string(key));
}
// Free voice & key slot immediately
void Bison::FreeVoice(int index)
{
SFM_ASSERT(index >= 0 && index < kMaxPolyVoices);
Voice &voice = m_voices[index];
SFM_ASSERT(false == voice.IsIdle());
// Set to idle, lift possible sustain
voice.m_state = Voice::kIdle;
voice.m_sustained = false;
// Decrease global count
SFM_ASSERT(m_voiceCount > 0);
--m_voiceCount;
// Free key
const int key = voice.m_key;
if (-1 != key)
{
FreeKey(voice.m_key);
voice.m_key = -1;
Log("Voice freed: " + std::to_string(index) + " for key: " + std::to_string(key));
}
else
Log("Voice freed: " + std::to_string(index));
}
// Steal voice (quick fade)
void Bison::StealVoice(int index)
{
SFM_ASSERT(index >= 0 && index < kMaxPolyVoices);
Voice &voice = m_voices[index];
SFM_ASSERT(false == voice.IsIdle());
// Can not steal what has already been stolen!
SFM_ASSERT(false == voice.IsStolen());
// Not active? Does not need to be stolen
if (true == voice.IsIdle())
return;
// Flag as stolen
voice.m_state = Voice::kStolen;
// Initiate fade out
const float curGlobalAmp = voice.m_globalAmp.Get();
voice.m_globalAmp.SetRate(m_sampleRate, kGlobalAmpCutTime);
voice.m_globalAmp.Set(curGlobalAmp);
voice.m_globalAmp.SetTarget(0.f);
// Free key (if not already done)
const int key = voice.m_key;
if (-1 != key)
{
FreeKey(key);
voice.m_key = -1;
Log("Voice stolen: " + std::to_string(index) + " for key: " + std::to_string(key));
}
else
Log("Voice stolen (not bound to key): " + std::to_string(index));
}
void Bison::NoteOn(unsigned key, float frequency, float velocity, unsigned timeStamp)
{
const bool monophonic = Patch::VoiceMode::kMono == m_patch.voiceMode;
SFM_ASSERT(key <= 127);
SFM_ASSERT(velocity >= 0.f && velocity <= 1.f);
// In case of duplicates honour the first NOTE_ON
for (const auto &request : m_polyVoiceReq)
if (request.key == key)
{
Log("Duplicate NoteOn() for key: " + std::to_string(key));
return;
}
VoiceRequest request;
request.key = key;
request.frequency = frequency;
request.velocity = velocity;
request.timeStamp = timeStamp;
const int index = GetVoice(key);
if (false == monophonic)
{
/* Polyphonic */
// Key bound to voice (retrigger)?
if (index >= 0)
{
Voice &voice = m_voices[index];
if (false == voice.IsIdle())
{
/* Polyphonic */
// Steal if still playing (performance fix)
if (false == voice.IsStolen())
StealVoice(index);
Log("NoteOn() retrigger: " + std::to_string(key) + ", voice: " + std::to_string(index));
}
}
// Issue request
if (m_polyVoiceReq.size() < m_curPolyphony)
{
m_polyVoiceReq.push_back(request);
}
else
{
// Replace last request (FIXME: honour time stamp?)
m_polyVoiceReq.pop_back();
m_polyVoiceReq.push_back(request);
}
}
else
{
/* Monophonic */
SFM_ASSERT(1 == m_curPolyphony);
Log("NoteOn() monophonic, key: " + std::to_string(key));
if (false == m_monoVoiceReq.MonoIsValid() || request.timeStamp <= m_monoVoiceReq.timeStamp)
{
m_monoVoiceReq = request;
Log("Monophonic: is audible request");
}
// Always add requests to sequence
m_monoSequence.emplace_front(request);
}
}
void Bison::NoteOff(unsigned key, unsigned timeStamp)
{
const bool monophonic = Patch::VoiceMode::kMono == m_patch.voiceMode;
SFM_ASSERT(key <= 127);
// In case of duplicates honour the first NOTE_OFF
for (auto request : m_polyVoiceReleaseReq)
if (request == key)
{
Log("Duplicate NoteOff() for key: " + std::to_string(key));
return;
}
if (false == monophonic)
{
/* Polyphonic */
const int index = GetVoice(key);
if (index >= 0)
{
// Issue request
m_polyVoiceReleaseReq.push_back(key);
}
// It might be that a deferred request matches this NOTE_OFF, in which case we get rid of the request
for (auto iReq = m_polyVoiceReq.begin(); iReq != m_polyVoiceReq.end(); ++iReq)
{
if (iReq->key == key)
{
// Erase and break since NoteOn() ensures there are no duplicates in the deque
m_polyVoiceReq.erase(iReq);
Log("Deferred NoteOn() removed due to matching NOTE_OFF for key: " + std::to_string(key));
break;
}
}
}
else
{
/* Monophonic */
Log("NoteOff() monophonic, key: " + std::to_string(key));
const int index = GetVoice(key);
if (index >= 0)
{
SFM_ASSERT(0 == index);
// Key actually playing?
if (m_voices[index].IsPlaying())
{
m_monoVoiceReleaseReq.key = key;
m_monoVoiceReleaseReq.timeStamp = timeStamp;
Log("Monophonic: is release request of playing note");
}
}
// Remove occurence
for (auto iReq = m_monoSequence.begin(); iReq != m_monoSequence.end(); ++iReq)
{
if (iReq->key == key)
{
m_monoSequence.erase(iReq);
Log("Monophonic: key removed from sequence");
break;
}
}
}
}
/* ----------------------------------------------------------------------------------------------------
Voice initialization helper functions
------------------------------------------------------------------------------------------------------ */
// Calc. operator frequency
static float CalcOpFreq(float fundamentalFreq, float detuneOffs, const PatchOperators::Operator &patchOp)
{
float frequency;
if (true == patchOp.fixed)
{
frequency = (float) patchOp.coarse;
SFM_ASSERT(frequency >= 0.f && frequency <= kMaxFixedHz);
// Fixed requency does not necessarily have to be below Nyquist
// Ratio, fine and detune controls are disabled in VST UI
}
else
{
frequency = fundamentalFreq;
const int coarse = patchOp.coarse; // Ratio
const int fine = patchOp.fine; // Semitones
const float detune = patchOp.detune + detuneOffs; // Cents
SFM_ASSERT(coarse >= kCoarseMin && coarse <= kCoarseMax);
SFM_ASSERT(abs(fine) <= kFineRange);
SFM_ASSERT(abs(detune) <= kDetuneRange);
frequency *= powf(2.f, (detune*0.01f)/12.f);
if (coarse < 0)
frequency /= abs(coarse-1);
else if (coarse > 1)
frequency *= coarse;
frequency *= powf(2.f, fine/12.f);
}
return frequency;
}
// Calc. multiplier for operator amplitude or modulation index
static float CalcOpLevel(unsigned key, float velocity, const PatchOperators::Operator &patchOp)
{
float multiplier = 1.f;
// Factor in velocity (linear)
multiplier = lerpf<float>(multiplier, multiplier*velocity, patchOp.velSens);
// Apply L/R breakpoint cut & level scaling (subtractive/additive & linear/exponential, like the DX7)
const unsigned breakpoint = patchOp.levelScaleBP;
if (true == patchOp.cutLeftOfLSBP && true == patchOp.cutRightOfLSBP)
{
// Range cut
const unsigned sides = 127-breakpoint;
unsigned left = sides/2;
const unsigned remainder = left % 12;
left += 12-remainder; // Align right to high C note
const unsigned right = left+breakpoint;
if (key < left || key > right)
multiplier = 0.f;
}
else if (true == patchOp.cutLeftOfLSBP && key < breakpoint)
// Cut left
multiplier = 0.f;
else if (true == patchOp.cutRightOfLSBP && key > breakpoint)
// Cut right
multiplier = 0.f;
// We didn't cut, so apply level scaling as usual
if (0.f != multiplier)
{
// Apply level scaling
const unsigned numSemis = patchOp.levelScaleRange;
if (0 != numSemis)
{
const bool keyIsLeftOfBP = key < breakpoint;
const bool keyIsRightOfBP = key > breakpoint;
const float levelStep = 1.f/numSemis;
int distance = 0;
float amount = 0.f;
bool isExponential = false;
if (true == keyIsLeftOfBP)
{
distance = breakpoint-key;
amount = patchOp.levelScaleL;
isExponential = patchOp.levelScaleExpL;
}
else if (true == keyIsRightOfBP)
{
distance = key-breakpoint;
amount = patchOp.levelScaleR;
isExponential = patchOp.levelScaleExpR;
}
// Calculate normalized distance from BP
distance = std::min<int>(numSemis, abs(distance));
const float linear = smoothstepf(distance*levelStep); // This (smoothstep) takes the edges off, results in a smoother glide
const float factor = false == isExponential ? linear : powf(linear, 1.f-linear) /* -EXP/+EXP */;
if (amount < 0.f)
// Fade out by gradually interpolating towards lower level
multiplier = lerpf<float>(multiplier, multiplier*(1.f-fabsf(amount)), factor);
else if (amount > 0.f)
// Fade in by adding to set level
multiplier = lerpf<float>(multiplier, std::min<float>(1.f, multiplier+fabsf(amount)), factor);
// Subtractive as well as additive scaling leave the multiplier (output) on the other side
// of the breakpoint intact; this makes it intuitive to use this feature (I think)
}
}
SFM_ASSERT(multiplier >= 0.f && multiplier <= 1.f);
// Return level multiplier!
return multiplier;
}
// Simply scales [-1..1] to [0.0..1.0]
SFM_INLINE static float CalcPanning(const PatchOperators::Operator &patchOp)
{
SFM_ASSERT(fabsf(patchOp.panning) <= 1.f);
return 0.5f*patchOp.panning + 0.5f;
}
SFM_INLINE static float CalcPhaseJitter(float jitter)
{
SFM_ASSERT(jitter >= 0.f && jitter <= 1.f);
return jitter*mt_randf()*0.25f; // [0..90] deg.
}
SFM_INLINE static float CalcPhaseShift(const Voice::Operator &voiceOp, const PatchOperators::Operator &patchOp)
{
float shift = 0.f;
if (false == patchOp.keySync)
{
shift = voiceOp.oscillator.GetPhase() * mt_randf(); // FIXME: keep phase running or at least have global ones for each operator
}
return shift;
}
// Returns [-1..1]
SFM_INLINE static float CalcOpCutoffKeyTracking(unsigned key, float cutoffKeyTrack)
{
SFM_ASSERT(key >= 0 && key <= 127);
const float normalizedKey = key/127.f;
/* const */ float tracking = cutoffKeyTrack;
SFM_ASSERT_BINORM(tracking);
return tracking*normalizedKey;
}
// Set up (static) operator filter
static void SetOperatorFilters(unsigned key, unsigned sampleRate, Biquad &filter, SvfLinearTrapOptimised2 &modFilter, const PatchOperators::Operator &patchOp)
{
SFM_ASSERT(sampleRate > 0);
const unsigned Nyquist = sampleRate/2;
// filter.resetState(); //
filter.reset(); // <- This *is* necesasry after 20/05/2022 (see impl.)
modFilter.resetState(); //
const float normQ = patchOp.resonance;
// Calculate keytracking (only for LPF & HPF)
float tracking = -1.f, cutoffNormFrom = -1.f, cutoffNormTo = -1.f;
if (PatchOperators::Operator::kLowpassFilter == patchOp.filterType || PatchOperators::Operator::kHighpassFilter == patchOp.filterType)
{
const float cutoffKeyTrack = patchOp.cutoffKeyTrack;
tracking = CalcOpCutoffKeyTracking(key, cutoffKeyTrack);
cutoffNormFrom = patchOp.cutoff;
cutoffNormTo = (tracking >= 0.f) ? 1.f : 0.f;
tracking = fabsf(tracking);
}
float cutoffNorm = -1.f; // Causes assertion if not set for kLowpassFilter/kHighpassFilter
const float biQ = 0.01f + 19.99f*normQ; // See Biquad.h
switch (patchOp.filterType)
{
default:
SFM_ASSERT(false);
case PatchOperators::Operator::kNoFilter:
filter.setBiquad(bq_type_none, 0.f, 0.f, 0.f);
break;
case PatchOperators::Operator::kLowpassFilter:
cutoffNorm = lerpf<float>(cutoffNormFrom, cutoffNormTo, tracking); // Keytrack
filter.setBiquad(bq_type_lowpass, BQ_CutoffToHz(cutoffNorm, Nyquist)/sampleRate, biQ, 0.f);
break;
case PatchOperators::Operator::kHighpassFilter:
cutoffNorm = lerpf<float>(cutoffNormFrom, 1.f-cutoffNormTo, tracking); // Keytrack
filter.setBiquad(bq_type_highpass, BQ_CutoffToHz(cutoffNorm, Nyquist)/sampleRate, biQ, 0.f);
break;
// FIXME: why not track the next ones too?
case PatchOperators::Operator::kBandpassFilter:
filter.setBiquad(bq_type_bandpass, BQ_CutoffToHz(patchOp.cutoff, Nyquist)/sampleRate, biQ, 0.f);
break;
case PatchOperators::Operator::kPeakFilter:
SFM_ASSERT(patchOp.peakdB >= kMinOpFilterPeakdB && patchOp.peakdB <= kMaxOpFilterPeakdB);
filter.setBiquad(bq_type_peak, BQ_CutoffToHz(patchOp.cutoff, Nyquist)/sampleRate, biQ, patchOp.peakdB);
break;
}
switch (patchOp.waveform)
{
// These waveforms shall remain unaltered
case Oscillator::kSine:
case Oscillator::kCosine:
case Oscillator::kPolyTriangle:
case Oscillator::kSupersaw: // Checked and it's not necessary (though at first I assumed it would be)
modFilter.updateNone();
break;
// Filter the remaining waveforms a little to "take the top off"
default:
modFilter.updateLowpassCoeff(SVF_CutoffToHz(kModulatorLP, Nyquist), kSVFLowestFilterQ, sampleRate);
break;
}
}
// Calc. tracking (linear or 'acoustically curved')
SFM_INLINE static float CalcKeyTracking(unsigned key, const PatchOperators::Operator &patchOp)
{
SFM_ASSERT(key >= 0 && key <= 127);
const float normalizedKey = key/127.f;
SFM_ASSERT(patchOp.envKeyTrack >= 0.f && patchOp.envKeyTrack <= 1.f);
return (false == patchOp.acousticEnvKeyTrack)
? 1.f - 0.9f*patchOp.envKeyTrack*normalizedKey
: AcousticTrackingCurve(normalizedKey, patchOp.envKeyTrack); // See impl. for details
}
// Calc. voice soft clipping (distortion) amount
SFM_INLINE static float CalcSoftClip(float opVelocity, const PatchOperators::Operator &patchOp)
{
const float softClip = lerpf<float>(patchOp.drive, patchOp.drive*opVelocity, patchOp.velSens);
return softClip;
}
// Calc. LFO frequencies
SFM_INLINE static void CalcLFOFreq(float &frequency /* Set to base freq. */, float &modFrequency, int speedAdj)
{
SFM_ASSERT(frequency > 0.f);
SFM_ASSERT(speedAdj >= kMinLFOModSpeed && speedAdj <= kMaxLFOModSpeed);
const float freqSpeedAdj = powf(2.f, float(speedAdj));
modFrequency = frequency*freqSpeedAdj;
}
// Initializes LFOs (used on top of Voice::Sample())
void Bison::InitializeLFOs(Voice &voice, float jitter)
{
// Calc. shift
float phaseShift = (true == m_patch.LFOKeySync)
? 0.f // Synchronized
: m_globalLFO->Get(); // Free running
// Add jitter
phaseShift += CalcPhaseJitter(jitter);
// Frequencies
float frequency = m_globalLFO->GetFrequency(), modFrequency;
CalcLFOFreq(frequency, modFrequency, m_patch.LFOModSpeed);
// Set up LFOs
voice.m_LFO1.Initialize(m_patch.LFOWaveform1, frequency, m_sampleRate, phaseShift);
voice.m_LFO2.Initialize(m_patch.LFOWaveform2, frequency, m_sampleRate, phaseShift);
voice.m_modLFO.Initialize(m_patch.LFOWaveform3, modFrequency, m_sampleRate, phaseShift);
}
/* ----------------------------------------------------------------------------------------------------
Voice initialization; there's a separate function for a monophonic voice
------------------------------------------------------------------------------------------------------ */
// Initialize new voice
void Bison::InitializeVoice(const VoiceRequest &request, unsigned iVoice)
{
Voice &voice = m_voices[iVoice];
// No voice reset, this function should initialize all necessary components
// and be able to use previous values such as oscillator phase to enable/disable
// Voice not sustained
voice.m_sustained = false;
// Offset in samples (to be within a single Render() pass)
SFM_ASSERT(request.timeStamp <= m_samplesPerBlock);
voice.m_sampleOffs = request.timeStamp;
const unsigned key = request.key; // Key
const float jitter = m_patch.jitter; // Jitter
const float velocity = request.velocity; // Velocity
// Store key & velocity immediately (used by CalcOpLevel())
voice.m_key = key;
voice.m_velocity = velocity;
// Get fundamental freq. (using JUCE-supplied freq. for now)
float fundamentalFreq = (-1.f == request.frequency)
? float(g_MIDIToFreqLUT[key])
: request.frequency;
// Note frequency jitter
const float noteJitter = jitter*mt_randfc()*kMaxNoteJitter;
fundamentalFreq *= powf(2.f, (noteJitter*0.01f)/12.f);
voice.m_fundamentalFreq = fundamentalFreq;
// Initialize LFO
InitializeLFOs(voice, jitter);
// Get dry FM patch
PatchOperators &patchOps = m_patch.operators;
// Default glide (in case frequency is manipulated whilst playing)
voice.m_freqGlide = kDefPolyFreqGlide;
// Acoustic scaling: more velocity can mean longer envelope decay phase
// This is specifically designed for piano, guitar et cetera: strum or strike
// harder and the decay phase will be longer
const float envAcousticScaling = 1.f + (velocity*velocity)*m_patch.acousticScaling;
// Set up voice operators
for (unsigned iOp = 0; iOp < kNumOperators; ++iOp)
{
const PatchOperators::Operator &patchOp = patchOps.operators[iOp];
Voice::Operator &voiceOp = voice.m_operators[iOp];
voiceOp.enabled = patchOp.enabled;
voiceOp.isCarrier = patchOp.isCarrier;
voiceOp.envGain.Reset();
if (true == voiceOp.enabled)
{
// Operator velocity
const float opVelocity = (false == patchOp.velocityInvert) ? velocity : 1.f-velocity;
// (Re)set constant/static filters
SetOperatorFilters(key, m_sampleRate, voiceOp.filter, voiceOp.modFilter, patchOp);
// Store detune jitter
voiceOp.detuneOffs = jitter*mt_randfc()*patchOp.detune*kMaxDetuneJitter;
const float frequency = CalcOpFreq(fundamentalFreq, voiceOp.detuneOffs, patchOp);
// Get amplitude & index
const float level = CalcOpLevel(key, opVelocity, patchOp);
const float amplitude = patchOp.output*level, index = patchOp.index*level;
voiceOp.oscillator.Initialize(
patchOp.waveform, frequency, m_sampleRate, CalcPhaseShift(voiceOp, patchOp), patchOp.supersawDetune, patchOp.supersawMix);
// Set supersaw parameters for interpolation
voiceOp.supersawDetune.SetRate(m_sampleRate, kDefParameterLatency);
voiceOp.supersawDetune.Set(patchOp.supersawDetune);
voiceOp.supersawMix.SetRate(m_sampleRate, kDefParameterLatency);
voiceOp.supersawMix.Set(patchOp.supersawMix);
// Set (static) amplitude
voiceOp.amplitude.SetRate(m_sampleRate, kDefParameterLatency);
voiceOp.amplitude.Set(amplitude);
// Set (static) index
voiceOp.index.SetRate(m_sampleRate, kDefParameterLatency);
voiceOp.index.Set(index);
// No interpolation
voiceOp.curFreq.SetRate(m_sampleRate, kDefPolyFreqGlide);
voiceOp.curFreq.Set(frequency);
voiceOp.setFrequency = frequency;
// Envelope key tracking
const float envKeyTracking = CalcKeyTracking(key, patchOp);
// Start envelope
voiceOp.envelope.Start(patchOp.envParams, m_sampleRate, patchOp.isCarrier, envKeyTracking, envAcousticScaling);
// Modulation sources
voiceOp.modulators[0] = patchOp.modulators[0];
voiceOp.modulators[1] = patchOp.modulators[1];
voiceOp.modulators[2] = patchOp.modulators[2];
// Feedback
voiceOp.iFeedback = patchOp.feedback;
voiceOp.feedbackAmt = { patchOp.feedbackAmt, m_sampleRate, kDefParameterLatency, 0.f, 1.f };
voiceOp.feedback = 0.f;
// LFO influence
voiceOp.ampMod = patchOp.ampMod;
voiceOp.pitchMod = patchOp.pitchMod;
voiceOp.panMod = patchOp.panMod;
// Panning
voiceOp.panning = { CalcPanning(patchOp), m_sampleRate, kDefParameterLatency, 0.f, 1.f };
// Soft clip (distortion)
voiceOp.softClip = { CalcSoftClip(opVelocity, patchOp), m_sampleRate, kDefParameterLatency, 0.f, 1.f };
}
}
// Reset main filter
voice.m_filterSVF.resetState();
// Start filter envelope
voice.m_filterEnvelope.Start(m_patch.filterEnvParams, m_sampleRate, false, 1.f, envAcousticScaling);
// Start pitch envelope
voice.m_pitchBendRange = m_patch.pitchBendRange;
voice.m_pitchEnvelope.Start(m_patch.pitchEnvParams, m_sampleRate);
// Voice is now playing
voice.m_state = Voice::kPlaying;
++m_voiceCount;
// Store (new) index in key slot
SFM_ASSERT(-1 == GetVoice(key));
SetKey(key, iVoice);
voice.PostInitialize();
}
// Specialized function for monophonic voices
// Not ideal due to some code duplication, but easier to modify and follow
void Bison::InitializeMonoVoice(const VoiceRequest &request)
{
Voice &voice = m_voices[0];
// No voice reset, this function should initialize all necessary components
// and be able to use previous values such as oscillator phase to enable/disable
// Reset (do *not* glide) if voice has been let go
bool reset = voice.IsReleasing() || voice.IsDone();
// Voice not sustained
voice.m_sustained = false;
const unsigned key = request.key; // Key
const float jitter = m_patch.jitter; // Jitter
const float velocity = request.velocity;
// Store key & velocity immediately (used by CalcOpLevel())
voice.m_key = key;
voice.m_velocity = velocity;
// Get fundamental freq. (using JUCE-supplied freq. for now)
float fundamentalFreq = (-1.f == request.frequency)
? float(g_MIDIToFreqLUT[key])
: request.frequency;
// Note frequency jitter
const float noteJitter = jitter * (-1.f + mt_randf()*2.f);
fundamentalFreq *= powf(2.f, (noteJitter*kMaxNoteJitter*0.01f)/12.f);
voice.m_fundamentalFreq = fundamentalFreq;
if (true == reset)
{
// Initialize LFO
InitializeLFOs(voice, jitter);
}
// See InitializeVoice()
const float envAcousticScaling = 1.f + (velocity*velocity)*m_patch.acousticScaling;
// Get patch operators
PatchOperators &patchOps = m_patch.operators;
// Calc. attenuated glide (using new velocity, feels more natural)
float monoGlide = m_patch.monoGlide;
float glideAtt = 1.f - m_patch.monoAtt*request.velocity;
voice.m_freqGlide = monoGlide*glideAtt;
// Set up voice operators
for (unsigned iOp = 0; iOp < kNumOperators; ++iOp)
{
const PatchOperators::Operator &patchOp = patchOps.operators[iOp];
Voice::Operator &voiceOp = voice.m_operators[iOp];
voiceOp.enabled = patchOp.enabled;
voiceOp.isCarrier = patchOp.isCarrier;
voiceOp.envGain.Reset();
if (true == voiceOp.enabled)
{
// Operator velocity
const float opVelocity = (false == patchOp.velocityInvert) ? velocity : 1.f-velocity;
if (true == reset)
{
// (Re)set constant/static filters
SetOperatorFilters(key, m_sampleRate, voiceOp.filter, voiceOp.modFilter, patchOp);
}
// Store detune jitter
voiceOp.detuneOffs = jitter*mt_randfc()*patchOp.detune*kMaxDetuneJitter;
const float frequency = CalcOpFreq(fundamentalFreq, voiceOp.detuneOffs, patchOp);
// Get amplitude & index
const float level = CalcOpLevel(key, opVelocity, patchOp);
const float amplitude = patchOp.output*level, index = patchOp.index*level;
if (true == reset)
{
voiceOp.oscillator.Initialize(
patchOp.waveform, frequency, m_sampleRate, CalcPhaseShift(voiceOp, patchOp), patchOp.supersawDetune, patchOp.supersawMix);
// Set supersaw parameters for interpolation
voiceOp.supersawDetune.SetRate(m_sampleRate, kDefParameterLatency);
voiceOp.supersawDetune.Set(patchOp.supersawDetune);
voiceOp.supersawMix.SetRate(m_sampleRate, kDefParameterLatency);
voiceOp.supersawMix.Set(patchOp.supersawMix);
// Set amplitude
voiceOp.amplitude.SetRate(m_sampleRate, kDefParameterLatency);
voiceOp.amplitude.Set(amplitude);
// Set index
voiceOp.index.SetRate(m_sampleRate, kDefParameterLatency);
voiceOp.index.Set(index);
// Set freq.
voiceOp.curFreq.SetRate(m_sampleRate, voice.m_freqGlide);
voiceOp.curFreq.Set(frequency);
const float envKeyTracking = CalcKeyTracking(key, patchOp);
voiceOp.envelope.Start(patchOp.envParams, m_sampleRate, patchOp.isCarrier, envKeyTracking, envAcousticScaling);
}
else
{
// Glide
voiceOp.supersawDetune.SetTarget(patchOp.supersawDetune);
voiceOp.supersawMix.SetTarget(patchOp.supersawMix);
voiceOp.amplitude.SetTarget(amplitude);
voiceOp.index.SetTarget(index);
// See synth-interpolated-parameter.h to see why I do this in this particular order
const float curFreq = voiceOp.curFreq.Get();
voiceOp.curFreq.SetRate(m_sampleRate, voice.m_freqGlide);
voiceOp.curFreq.Set(curFreq);
voiceOp.curFreq.SetTarget(frequency);
}
voiceOp.setFrequency = frequency;
// Modulation sources
voiceOp.modulators[0] = patchOp.modulators[0];
voiceOp.modulators[1] = patchOp.modulators[1];
voiceOp.modulators[2] = patchOp.modulators[2];
// Feedback
voiceOp.iFeedback = patchOp.feedback;
voiceOp.feedbackAmt = { patchOp.feedbackAmt, m_sampleRate, kDefParameterLatency, 0.f, 1.f };
voiceOp.feedback = 0.f;
// LFO influence
voiceOp.ampMod = patchOp.ampMod;