voice.h

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/*
 * Copyright © 2014-2023 Synthstrom Audible Limited
 *
 * This file is part of The Synthstrom Audible Deluge Firmware.
 *
 * The Synthstrom Audible Deluge Firmware is free software: you can redistribute it and/or modify it under the
 * terms of the GNU 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 General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License along with this program.
 * If not, see <https://www.gnu.org/licenses/>.
 */
 
#ifndef VOICE_H
#define VOICE_H
 
#include <patcher.h>
#include <voicesampleplaybackguide.h>
#include "definitions.h"
#include "envelope.h"
#include "lfo.h"
#include "voiceunisonpart.h"
#include "FilterSet.h"
 
class StereoSample;
class FilterSetConfig;
class ModelStackWithVoice;
 
class Voice final {
public:
    Voice();
 
    Patcher patcher;
 
    VoiceUnisonPart unisonParts[maxNumUnison]; // Stores all oscillator positions and stuff, for each Source within each Unison too
    VoiceSamplePlaybackGuide guides[NUM_SOURCES]; // Stores overall info on each Source (basically just sample memory bounds), for the play-through associated with this Voice right now.
 
    Sound* assignedToSound;
 
    int32_t paramFinalValues[FIRST_GLOBAL_PARAM]; // This is just for the *local* params, specific to this Voice only
    int32_t sourceValues[NUM_PATCH_SOURCES]; // At the start of this list are local copies of the "global" ones. It's cheaper to copy them here than to pick and choose where the Patcher looks for them
 
    int32_t localExpressionSourceValuesBeforeSmoothing[NUM_EXPRESSION_DIMENSIONS];
 
    Envelope envelopes[numEnvelopes];
    LFO lfo;
 
    FilterSet filterSets[2];
    int inputCharacteristics[2]; // Contains what used to be called noteCodeBeforeArpeggiation, and fromMIDIChannel
    int noteCodeAfterArpeggiation;
 
    uint32_t portaEnvelopePos;
    int32_t portaEnvelopeMaxAmplitude;
 
    uint32_t lastSaturationTanHWorkingValue[2];
 
    int32_t overallOscAmplitudeLastTime;
    int32_t sourceAmplitudesLastTime[NUM_SOURCES];
    int32_t modulatorAmplitudeLastTime[numModulators];
    uint32_t sourceWaveIndexesLastTime[NUM_SOURCES];
 
    int32_t filterGainLastTime;
 
    bool doneFirstRender;
    bool previouslyIgnoredNoteOff;
    uint8_t whichExpressionSourcesCurrentlySmoothing;
    uint8_t whichExpressionSourcesFinalValueChanged;
 
    uint32_t orderSounded;
 
    int32_t overrideAmplitudeEnvelopeReleaseRate;
 
    Voice* nextUnassigned;
 
    void setAsUnassigned(ModelStackWithVoice* modelStack, bool deletingSong = false);
    bool render(ModelStackWithVoice* modelStack, int32_t* soundBuffer, int numSamples, bool soundRenderingInStereo, bool applyingPanAtVoiceLevel, uint32_t sourcesChanged, FilterSetConfig* filterSetConfig, int32_t externalPitchAdjust);
 
    void calculatePhaseIncrements(ModelStackWithVoice* modelStack);
    bool sampleZoneChanged(ModelStackWithVoice* modelStack, int s, int markerType);
    bool noteOn(ModelStackWithVoice* modelStack, int newNoteCodeBeforeArpeggiation, int newNoteCodeAfterArpeggiation, uint8_t velocity, uint32_t newSampleSyncLength, int32_t ticksLate, uint32_t samplesLate, bool resetEnvelopes, int fromMIDIChannel, const int16_t* mpeValues);
    void noteOff(ModelStackWithVoice* modelStack, bool allowReleaseStage = true);
    bool doFastRelease(uint32_t releaseIncrement = 4096);
    void randomizeOscPhases(Sound* sound);
    void changeNoteCode(ModelStackWithVoice* modelStack, int newNoteCodeBeforeArpeggiation, int newNoteCodeAfterArpeggiation, int newInputMIDIChannel, const int16_t* newMPEValues);
    bool hasReleaseStage();
    void unassignStuff();
    uint32_t getPriorityRating();
    void expressionEventImmediate(Sound* sound, int32_t voiceLevelValue, int s);
    void expressionEventSmooth(int32_t newValue, int s);
 
private:
    //inline int32_t doFM(uint32_t *carrierPhase, uint32_t* lastShiftedPhase, uint32_t carrierPhaseIncrement, uint32_t phaseShift);
 
    void renderOsc(int s, int type, int32_t amplitude, int32_t* thisSample, int32_t* bufferEnd, int numSamples, uint32_t phaseIncrementNow, uint32_t phaseWidth, uint32_t* thisPhase, bool applyAmplitude, int32_t amplitudeIncrement, bool doOscSync, uint32_t resetterPhase, uint32_t resetterPhaseIncrement, uint32_t retriggerPhase, int32_t waveIndexIncrement);
    void renderBasicSource(Sound* sound, ParamManagerForTimeline* paramManager, int s, int32_t* oscBuffer, int numSamples, int32_t sourceAmplitude, bool* unisonPartBecameInactive, int32_t overallPitchAdjust, bool doOscSync, uint32_t* oscSyncPos, uint32_t* oscSyncPhaseIncrements, int32_t amplitudeIncrement, uint32_t* getPhaseIncrements, bool getOutAfterPhaseIncrements, int32_t waveIndexIncrement);
    bool adjustPitch(uint32_t* phaseIncrement, int32_t adjustment);
 
    void renderSineWaveWithFeedback(int32_t* thisSample, int numSamples, uint32_t* phase, int32_t amplitude, uint32_t phaseIncrement, int32_t feedbackAmount, int32_t* lastFeedbackValue, bool add, int32_t amplitudeIncrement);
    void renderFMWithFeedback(int32_t* thisSample, int numSamples, int32_t* fmBuffer, uint32_t* phase, int32_t amplitude, uint32_t phaseIncrement, int32_t feedbackAmount, int32_t* lastFeedbackValue, int32_t amplitudeIncrement);
    void renderFMWithFeedbackAdd(int32_t* thisSample, int numSamples, int32_t* fmBuffer, uint32_t* phase, int32_t amplitude, uint32_t phaseIncrement, int32_t feedbackAmount, int32_t* lastFeedbackValue, int32_t amplitudeIncrement);
    bool areAllUnisonPartsInactive(ModelStackWithVoice* modelStackWithVoice);
    void setupPorta(Sound* sound);
    int32_t combineExpressionValues(Sound* sound, int whichExpressionDimension);
 
};
 
#endif // VOICE_H