engines/gig/Voice.h

/***************************************************************************
 *                                                                         *
 *   LinuxSampler - modular, streaming capable sampler                     *
 *                                                                         *
 *   Copyright (C) 2003, 2004 by Benno Senoner and Christian Schoenebeck   *
 *                                                                         *
 *   This program 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 2 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, write to the Free Software           *
 *   Foundation, Inc., 59 Temple Place, Suite 330, Boston,                 *
 *   MA  02111-1307  USA                                                   *
 ***************************************************************************/

#ifndef __LS_GIG_VOICE_H__
#define __LS_GIG_VOICE_H__

#include "../../common/global.h"

#if DEBUG_HEADERS
# warning Voice.h included
#endif // DEBUG_HEADERS

#include "../../common/RTMath.h"
#include "../../common/RingBuffer.h"
#include "../../common/Pool.h"
#include "../../drivers/audio/AudioOutputDevice.h"
#include "../../lib/fileloader/libgig/gig.h"
#include "../common/BiquadFilter.h"
#include "Engine.h"
#include "Stream.h"
#include "DiskThread.h"

#include "EGDecay.h"
#include "Filter.h"
#include "../common/LFO.h"

#define USE_LINEAR_INTERPOLATION        0  ///< set to 0 if you prefer cubic interpolation (slower, better quality)
#define ENABLE_FILTER                   1  ///< if set to 0 then filter (VCF) code is ignored on compile time
#define FILTER_UPDATE_PERIOD            64 ///< amount of sample points after which filter parameters (cutoff, resonance) are going to be updated (higher value means less CPU load, but also worse parameter resolution, this value will be aligned to a power of two)
#define FORCE_FILTER_USAGE              0  ///< if set to 1 then filter is always used, if set to 0 filter is used only in case the instrument file defined one
#define FILTER_CUTOFF_MAX               10000.0f ///< maximum cutoff frequency (10kHz)
#define FILTER_CUTOFF_MIN               100.0f   ///< minimum cutoff frequency (100Hz)

// Uncomment following line to override external cutoff controller
//#define OVERRIDE_FILTER_CUTOFF_CTRL   1  ///< set to an arbitrary MIDI control change controller (e.g. 1 for 'modulation wheel')

// Uncomment following line to override external resonance controller
//#define OVERRIDE_FILTER_RES_CTRL      91  ///< set to an arbitrary MIDI control change controller (e.g. 91 for 'effect 1 depth')

// Uncomment following line to override filter type
//#define OVERRIDE_FILTER_TYPE          ::gig::vcf_type_lowpass  ///< either ::gig::vcf_type_lowpass, ::gig::vcf_type_bandpass or ::gig::vcf_type_highpass

namespace LinuxSampler { namespace gig {

    class Engine;
    class EGADSR;
    class VCAManipulator;
    class VCFCManipulator;
    class VCOManipulator;

    /// Reflects a MIDI controller
    struct midi_ctrl {
        uint8_t controller; ///< MIDI control change controller number
        uint8_t value;      ///< Current MIDI controller value
        float   fvalue;     ///< Transformed / effective value (e.g. volume level or filter cutoff frequency)
    };

    /** Gig Voice
     *
     * Renders a voice for the Gigasampler format.
     */
    class Voice {
        public:
            // Types
            enum type_t {
                type_normal,
                type_release_trigger_required,  ///< If the key of this voice will be released, it causes a release triggered voice to be spawned
                type_release_trigger            ///< Release triggered voice which cannot be killed by releasing its key
            };

            // Attributes
            type_t       Type;         ///< Voice Type
            int          MIDIKey;      ///< MIDI key number of the key that triggered the voice
            uint         KeyGroup;
            DiskThread*  pDiskThread;  ///< Pointer to the disk thread, to be able to order a disk stream and later to delete the stream again
            RTList<Voice>::Iterator itChildVoice; ///< Points to the next layer voice (if any). This field is currently only used by the voice stealing algorithm.

            // Methods
            Voice();
           ~Voice();
            void Kill(Pool<Event>::Iterator& itKillEvent);
            void Render(uint Samples);
            void Reset();
            void SetOutput(AudioOutputDevice* pAudioOutputDevice);
            void SetEngine(Engine* pEngine);
            int  Trigger(Pool<Event>::Iterator& itNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument, int iLayer, bool ReleaseTriggerVoice, bool VoiceStealing);
            inline bool IsActive() { return PlaybackState; }
        private:
            // Types
            enum playback_state_t {
                playback_state_end  = 0,
                playback_state_ram  = 1,
                playback_state_disk = 2
            };

            // Attributes
            gig::Engine*                pEngine;            ///< Pointer to the sampler engine, to be able to access the event lists.
            float                       Volume;             ///< Volume level of the voice
            float                       PanLeft;
            float                       PanRight;
            float                       CrossfadeVolume;    ///< Current attenuation level caused by a crossfade (only if a crossfade is defined of course)
            double                      Pos;                ///< Current playback position in sample
            double                      PitchBase;          ///< Basic pitch depth, stays the same for the whole life time of the voice
            double                      PitchBend;          ///< Current pitch value of the pitchbend wheel
            ::gig::Sample*              pSample;            ///< Pointer to the sample to be played back
            ::gig::Region*              pRegion;            ///< Pointer to the articulation information of the respective keyboard region of this voice
            ::gig::DimensionRegion*     pDimRgn;            ///< Pointer to the articulation information of current dimension region of this voice
            playback_state_t            PlaybackState;      ///< When a sample will be triggered, it will be first played from RAM cache and after a couple of sample points it will switch to disk streaming and at the end of a disk stream we have to add null samples, so the interpolator can do it's work correctly
            bool                        DiskVoice;          ///< If the sample is very short it completely fits into the RAM cache and doesn't need to be streamed from disk, in that case this flag is set to false
            Stream::reference_t         DiskStreamRef;      ///< Reference / link to the disk stream
            unsigned long               MaxRAMPos;          ///< The upper allowed limit (not actually the end) in the RAM sample cache, after that point it's not safe to chase the interpolator another time over over the current cache position, instead we switch to disk then.
            bool                        RAMLoop;            ///< If this voice has a loop defined which completely fits into the cached RAM part of the sample, in this case we handle the looping within the voice class, else if the loop is located in the disk stream part, we let the disk stream handle the looping
            int                         LoopCyclesLeft;     ///< In case there is a RAMLoop and it's not an endless loop; reflects number of loop cycles left to be passed
            uint                        Delay;              ///< Number of sample points the rendering process of this voice should be delayed (jitter correction), will be set to 0 after the first audio fragment cycle
            EGADSR*                     pEG1;               ///< Envelope Generator 1 (Amplification)
            EGADSR*                     pEG2;               ///< Envelope Generator 2 (Filter cutoff frequency)
            EGDecay*                    pEG3;               ///< Envelope Generator 3 (Pitch)
            Filter                      FilterLeft;
            Filter                      FilterRight;
            midi_ctrl                   VCFCutoffCtrl;
            midi_ctrl                   VCFResonanceCtrl;
            int                         FilterUpdateCounter; ///< Used to update filter parameters all FILTER_UPDATE_PERIOD samples
            static const float          FILTER_CUTOFF_COEFF;
            static const int            FILTER_UPDATE_MASK;
            VCAManipulator*             pVCAManipulator;
            VCFCManipulator*            pVCFCManipulator;
            VCOManipulator*             pVCOManipulator;
            LFO<gig::VCAManipulator>*   pLFO1;              ///< Low Frequency Oscillator 1 (Amplification)
            LFO<gig::VCFCManipulator>*  pLFO2;             ///< Low Frequency Oscillator 2 (Filter cutoff frequency)
            LFO<gig::VCOManipulator>*   pLFO3;              ///< Low Frequency Oscillator 3 (Pitch)
            Pool<Event>::Iterator       itTriggerEvent;      ///< First event on the key's list the voice should process (only needed for the first audio fragment in which voice was triggered, after that it will be set to NULL).
        public: // FIXME: just made public for debugging (sanity check in Engine::RenderAudio()), should be changed to private before the final release
            Pool<Event>::Iterator       itKillEvent;         ///< Event which caused this voice to be killed
        private:


            // Static Methods
            static float CalculateFilterCutoffCoeff();
            static int   CalculateFilterUpdateMask();

            // Methods
            void        KillImmediately();
            void        ProcessEvents(uint Samples);
            #if ENABLE_FILTER
            void        CalculateBiquadParameters(uint Samples);
            #endif // ENABLE_FILTER
            void        InterpolateNoLoop(uint Samples, sample_t* pSrc, uint Skip);
            void        InterpolateAndLoop(uint Samples, sample_t* pSrc, uint Skip);

            inline void InterpolateMono(sample_t* pSrc, int& i) {
                InterpolateOneStep_Mono(pSrc, i,
                                        pEngine->pSynthesisParameters[Event::destination_vca][i] * PanLeft,
                                        pEngine->pSynthesisParameters[Event::destination_vca][i] * PanRight,
                                        pEngine->pSynthesisParameters[Event::destination_vco][i],
                                        pEngine->pBasicFilterParameters[i],
                                        pEngine->pMainFilterParameters[i]);
            }

            inline void InterpolateStereo(sample_t* pSrc, int& i) {
                InterpolateOneStep_Stereo(pSrc, i,
                                          pEngine->pSynthesisParameters[Event::destination_vca][i] * PanLeft,
                                          pEngine->pSynthesisParameters[Event::destination_vca][i] * PanRight,
                                          pEngine->pSynthesisParameters[Event::destination_vco][i],
                                          pEngine->pBasicFilterParameters[i],
                                          pEngine->pMainFilterParameters[i]);
            }

            inline void InterpolateOneStep_Stereo(sample_t* pSrc, int& i, float volume_left, float volume_right, float& pitch, biquad_param_t& bq_base, biquad_param_t& bq_main) {
                int   pos_int   = RTMath::DoubleToInt(this->Pos);  // integer position
                float pos_fract = this->Pos - pos_int;             // fractional part of position
                pos_int <<= 1;

                #if USE_LINEAR_INTERPOLATION
                    #if ENABLE_FILTER
                        // left channel
                        pEngine->pOutputLeft[i]    += this->FilterLeft.Apply(&bq_base, &bq_main, volume_left * (pSrc[pos_int]   + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int])));
                        // right channel
                        pEngine->pOutputRight[i++] += this->FilterRight.Apply(&bq_base, &bq_main, volume_right * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1])));
                    #else // no filter
                        // left channel
                        pEngine->pOutputLeft[i]    += volume_left * (pSrc[pos_int]   + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int]));
                        // right channel
                        pEngine->pOutputRight[i++] += volume_right * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1]));
                    #endif // ENABLE_FILTER
                #else // polynomial interpolation
                    // calculate left channel
                    float xm1 = pSrc[pos_int];
                    float x0  = pSrc[pos_int+2];
                    float x1  = pSrc[pos_int+4];
                    float x2  = pSrc[pos_int+6];
                    float a   = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f;
                    float b   = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f;
                    float c   = (x1 - xm1) * 0.5f;
                    #if ENABLE_FILTER
                        pEngine->pOutputLeft[i] += this->FilterLeft.Apply(&bq_base, &bq_main, volume_left * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
                    #else // no filter
                        pEngine->pOutputLeft[i] += volume_left * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
                    #endif // ENABLE_FILTER

                    //calculate right channel
                    xm1 = pSrc[pos_int+1];
                    x0  = pSrc[pos_int+3];
                    x1  = pSrc[pos_int+5];
                    x2  = pSrc[pos_int+7];
                    a   = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f;
                    b   = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f;
                    c   = (x1 - xm1) * 0.5f;
                    #if ENABLE_FILTER
                        pEngine->pOutputRight[i++] += this->FilterRight.Apply(&bq_base, &bq_main, volume_right * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
                    #else // no filter
                        pEngine->pOutputRight[i++] += volume_right * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
                    #endif // ENABLE_FILTER
                #endif // USE_LINEAR_INTERPOLATION

                this->Pos += pitch;
            }

            inline void InterpolateOneStep_Mono(sample_t* pSrc, int& i, float volume_left, float volume_right, float& pitch,  biquad_param_t& bq_base, biquad_param_t& bq_main) {
                int   pos_int   = RTMath::DoubleToInt(this->Pos);  // integer position
                float pos_fract = this->Pos - pos_int;             // fractional part of position

                #if USE_LINEAR_INTERPOLATION
                    float sample_point  = pSrc[pos_int] + pos_fract * (pSrc[pos_int+1] - pSrc[pos_int]);
                #else // polynomial interpolation
                    float xm1 = pSrc[pos_int];
                    float x0  = pSrc[pos_int+1];
                    float x1  = pSrc[pos_int+2];
                    float x2  = pSrc[pos_int+3];
                    float a   = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f;
                    float b   = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f;
                    float c   = (x1 - xm1) * 0.5f;
                    float sample_point =  (((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0;
                #endif // USE_LINEAR_INTERPOLATION

                #if ENABLE_FILTER
                    sample_point = this->FilterLeft.Apply(&bq_base, &bq_main, sample_point);
                #endif // ENABLE_FILTER

                pEngine->pOutputLeft[i]    += sample_point * volume_left;
                pEngine->pOutputRight[i++] += sample_point * volume_right;

                this->Pos += pitch;
            }

            inline float CrossfadeAttenuation(uint8_t& CrossfadeControllerValue) {
                return (CrossfadeControllerValue <= pDimRgn->Crossfade.in_start)  ? 0.0f
                     : (CrossfadeControllerValue < pDimRgn->Crossfade.in_end)     ? float(CrossfadeControllerValue - pDimRgn->Crossfade.in_start) / float(pDimRgn->Crossfade.in_end - pDimRgn->Crossfade.in_start)
                     : (CrossfadeControllerValue <= pDimRgn->Crossfade.out_start) ? 1.0f
                     : (CrossfadeControllerValue < pDimRgn->Crossfade.out_end)    ? float(CrossfadeControllerValue - pDimRgn->Crossfade.out_start) / float(pDimRgn->Crossfade.out_end - pDimRgn->Crossfade.out_start)
                     : 0.0f;
            }

            inline float Constrain(float ValueToCheck, float Min, float Max) {
                if      (ValueToCheck > Max) ValueToCheck = Max;
                else if (ValueToCheck < Min) ValueToCheck = Min;
                return ValueToCheck;
            }
    };

}} // namespace LinuxSampler::gig

#endif // __LS_GIG_VOICE_H__
1	/***************************************************************************
2	* *
3	* LinuxSampler - modular, streaming capable sampler *
4	* *
5	* Copyright (C) 2003, 2004 by Benno Senoner and Christian Schoenebeck *
6	* *
7	* This program is free software; you can redistribute it and/or modify *
8	* it under the terms of the GNU General Public License as published by *
9	* the Free Software Foundation; either version 2 of the License, or *
10	* (at your option) any later version. *
11	* *
12	* This program is distributed in the hope that it will be useful, *
13	* but WITHOUT ANY WARRANTY; without even the implied warranty of *
14	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the *
15	* GNU General Public License for more details. *
16	* *
17	* You should have received a copy of the GNU General Public License *
18	* along with this program; if not, write to the Free Software *
19	* Foundation, Inc., 59 Temple Place, Suite 330, Boston, *
20	* MA 02111-1307 USA *
21	***************************************************************************/
22
23	#ifndef __LS_GIG_VOICE_H__
24	#define __LS_GIG_VOICE_H__
25
26	#include "../../common/global.h"
27
28	#if DEBUG_HEADERS
29	# warning Voice.h included
30	#endif // DEBUG_HEADERS
31
32	#include "../../common/RTMath.h"
33	#include "../../common/RingBuffer.h"
34	#include "../../common/Pool.h"
35	#include "../../drivers/audio/AudioOutputDevice.h"
36	#include "../../lib/fileloader/libgig/gig.h"
37	#include "../common/BiquadFilter.h"
38	#include "Engine.h"
39	#include "Stream.h"
40	#include "DiskThread.h"
41
42	#include "EGDecay.h"
43	#include "Filter.h"
44	#include "../common/LFO.h"
45
46	#define USE_LINEAR_INTERPOLATION 0 ///< set to 0 if you prefer cubic interpolation (slower, better quality)
47	#define ENABLE_FILTER 1 ///< if set to 0 then filter (VCF) code is ignored on compile time
48	#define FILTER_UPDATE_PERIOD 64 ///< amount of sample points after which filter parameters (cutoff, resonance) are going to be updated (higher value means less CPU load, but also worse parameter resolution, this value will be aligned to a power of two)
49	#define FORCE_FILTER_USAGE 0 ///< if set to 1 then filter is always used, if set to 0 filter is used only in case the instrument file defined one
50	#define FILTER_CUTOFF_MAX 10000.0f ///< maximum cutoff frequency (10kHz)
51	#define FILTER_CUTOFF_MIN 100.0f ///< minimum cutoff frequency (100Hz)
52
53	// Uncomment following line to override external cutoff controller
54	//#define OVERRIDE_FILTER_CUTOFF_CTRL 1 ///< set to an arbitrary MIDI control change controller (e.g. 1 for 'modulation wheel')
55
56	// Uncomment following line to override external resonance controller
57	//#define OVERRIDE_FILTER_RES_CTRL 91 ///< set to an arbitrary MIDI control change controller (e.g. 91 for 'effect 1 depth')
58
59	// Uncomment following line to override filter type
60	//#define OVERRIDE_FILTER_TYPE ::gig::vcf_type_lowpass ///< either ::gig::vcf_type_lowpass, ::gig::vcf_type_bandpass or ::gig::vcf_type_highpass
61
62	namespace LinuxSampler { namespace gig {
63
64	class Engine;
65	class EGADSR;
66	class VCAManipulator;
67	class VCFCManipulator;
68	class VCOManipulator;
69
70	/// Reflects a MIDI controller
71	struct midi_ctrl {
72	uint8_t controller; ///< MIDI control change controller number
73	uint8_t value; ///< Current MIDI controller value
74	float fvalue; ///< Transformed / effective value (e.g. volume level or filter cutoff frequency)
75	};
76
77	/** Gig Voice
78	*
79	* Renders a voice for the Gigasampler format.
80	*/
81	class Voice {
82	public:
83	// Types
84	enum type_t {
85	type_normal,
86	type_release_trigger_required, ///< If the key of this voice will be released, it causes a release triggered voice to be spawned
87	type_release_trigger ///< Release triggered voice which cannot be killed by releasing its key
88	};
89
90	// Attributes
91	type_t Type; ///< Voice Type
92	int MIDIKey; ///< MIDI key number of the key that triggered the voice
93	uint KeyGroup;
94	DiskThread* pDiskThread; ///< Pointer to the disk thread, to be able to order a disk stream and later to delete the stream again
95	RTList<Voice>::Iterator itChildVoice; ///< Points to the next layer voice (if any). This field is currently only used by the voice stealing algorithm.
96
97	// Methods
98	Voice();
99	~Voice();
100	void Kill(Pool<Event>::Iterator& itKillEvent);
101	void Render(uint Samples);
102	void Reset();
103	void SetOutput(AudioOutputDevice* pAudioOutputDevice);
104	void SetEngine(Engine* pEngine);
105	int Trigger(Pool<Event>::Iterator& itNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument, int iLayer, bool ReleaseTriggerVoice, bool VoiceStealing);
106	inline bool IsActive() { return PlaybackState; }
107	private:
108	// Types
109	enum playback_state_t {
110	playback_state_end = 0,
111	playback_state_ram = 1,
112	playback_state_disk = 2
113	};
114
115	// Attributes
116	gig::Engine* pEngine; ///< Pointer to the sampler engine, to be able to access the event lists.
117	float Volume; ///< Volume level of the voice
118	float PanLeft;
119	float PanRight;
120	float CrossfadeVolume; ///< Current attenuation level caused by a crossfade (only if a crossfade is defined of course)
121	double Pos; ///< Current playback position in sample
122	double PitchBase; ///< Basic pitch depth, stays the same for the whole life time of the voice
123	double PitchBend; ///< Current pitch value of the pitchbend wheel
124	::gig::Sample* pSample; ///< Pointer to the sample to be played back
125	::gig::Region* pRegion; ///< Pointer to the articulation information of the respective keyboard region of this voice
126	::gig::DimensionRegion* pDimRgn; ///< Pointer to the articulation information of current dimension region of this voice
127	playback_state_t PlaybackState; ///< When a sample will be triggered, it will be first played from RAM cache and after a couple of sample points it will switch to disk streaming and at the end of a disk stream we have to add null samples, so the interpolator can do it's work correctly
128	bool DiskVoice; ///< If the sample is very short it completely fits into the RAM cache and doesn't need to be streamed from disk, in that case this flag is set to false
129	Stream::reference_t DiskStreamRef; ///< Reference / link to the disk stream
130	unsigned long MaxRAMPos; ///< The upper allowed limit (not actually the end) in the RAM sample cache, after that point it's not safe to chase the interpolator another time over over the current cache position, instead we switch to disk then.
131	bool RAMLoop; ///< If this voice has a loop defined which completely fits into the cached RAM part of the sample, in this case we handle the looping within the voice class, else if the loop is located in the disk stream part, we let the disk stream handle the looping
132	int LoopCyclesLeft; ///< In case there is a RAMLoop and it's not an endless loop; reflects number of loop cycles left to be passed
133	uint Delay; ///< Number of sample points the rendering process of this voice should be delayed (jitter correction), will be set to 0 after the first audio fragment cycle
134	EGADSR* pEG1; ///< Envelope Generator 1 (Amplification)
135	EGADSR* pEG2; ///< Envelope Generator 2 (Filter cutoff frequency)
136	EGDecay* pEG3; ///< Envelope Generator 3 (Pitch)
137	Filter FilterLeft;
138	Filter FilterRight;
139	midi_ctrl VCFCutoffCtrl;
140	midi_ctrl VCFResonanceCtrl;
141	int FilterUpdateCounter; ///< Used to update filter parameters all FILTER_UPDATE_PERIOD samples
142	static const float FILTER_CUTOFF_COEFF;
143	static const int FILTER_UPDATE_MASK;
144	VCAManipulator* pVCAManipulator;
145	VCFCManipulator* pVCFCManipulator;
146	VCOManipulator* pVCOManipulator;
147	LFO<gig::VCAManipulator>* pLFO1; ///< Low Frequency Oscillator 1 (Amplification)
148	LFO<gig::VCFCManipulator>* pLFO2; ///< Low Frequency Oscillator 2 (Filter cutoff frequency)
149	LFO<gig::VCOManipulator>* pLFO3; ///< Low Frequency Oscillator 3 (Pitch)
150	Pool<Event>::Iterator itTriggerEvent; ///< First event on the key's list the voice should process (only needed for the first audio fragment in which voice was triggered, after that it will be set to NULL).
151	public: // FIXME: just made public for debugging (sanity check in Engine::RenderAudio()), should be changed to private before the final release
152	Pool<Event>::Iterator itKillEvent; ///< Event which caused this voice to be killed
153	private:
154
155
156	// Static Methods
157	static float CalculateFilterCutoffCoeff();
158	static int CalculateFilterUpdateMask();
159
160	// Methods
161	void KillImmediately();
162	void ProcessEvents(uint Samples);
163	#if ENABLE_FILTER
164	void CalculateBiquadParameters(uint Samples);
165	#endif // ENABLE_FILTER
166	void InterpolateNoLoop(uint Samples, sample_t* pSrc, uint Skip);
167	void InterpolateAndLoop(uint Samples, sample_t* pSrc, uint Skip);
168
169	inline void InterpolateMono(sample_t* pSrc, int& i) {
170	InterpolateOneStep_Mono(pSrc, i,
171	pEngine->pSynthesisParameters[Event::destination_vca][i] * PanLeft,
172	pEngine->pSynthesisParameters[Event::destination_vca][i] * PanRight,
173	pEngine->pSynthesisParameters[Event::destination_vco][i],
174	pEngine->pBasicFilterParameters[i],
175	pEngine->pMainFilterParameters[i]);
176	}
177
178	inline void InterpolateStereo(sample_t* pSrc, int& i) {
179	InterpolateOneStep_Stereo(pSrc, i,
180	pEngine->pSynthesisParameters[Event::destination_vca][i] * PanLeft,
181	pEngine->pSynthesisParameters[Event::destination_vca][i] * PanRight,
182	pEngine->pSynthesisParameters[Event::destination_vco][i],
183	pEngine->pBasicFilterParameters[i],
184	pEngine->pMainFilterParameters[i]);
185	}
186
187	inline void InterpolateOneStep_Stereo(sample_t* pSrc, int& i, float volume_left, float volume_right, float& pitch, biquad_param_t& bq_base, biquad_param_t& bq_main) {
188	int pos_int = RTMath::DoubleToInt(this->Pos); // integer position
189	float pos_fract = this->Pos - pos_int; // fractional part of position
190	pos_int <<= 1;
191
192	#if USE_LINEAR_INTERPOLATION
193	#if ENABLE_FILTER
194	// left channel
195	pEngine->pOutputLeft[i] += this->FilterLeft.Apply(&bq_base, &bq_main, volume_left * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int])));
196	// right channel
197	pEngine->pOutputRight[i++] += this->FilterRight.Apply(&bq_base, &bq_main, volume_right * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1])));
198	#else // no filter
199	// left channel
200	pEngine->pOutputLeft[i] += volume_left * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int]));
201	// right channel
202	pEngine->pOutputRight[i++] += volume_right * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1]));
203	#endif // ENABLE_FILTER
204	#else // polynomial interpolation
205	// calculate left channel
206	float xm1 = pSrc[pos_int];
207	float x0 = pSrc[pos_int+2];
208	float x1 = pSrc[pos_int+4];
209	float x2 = pSrc[pos_int+6];
210	float a = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f;
211	float b = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f;
212	float c = (x1 - xm1) * 0.5f;
213	#if ENABLE_FILTER
214	pEngine->pOutputLeft[i] += this->FilterLeft.Apply(&bq_base, &bq_main, volume_left * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
215	#else // no filter
216	pEngine->pOutputLeft[i] += volume_left * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
217	#endif // ENABLE_FILTER
218
219	//calculate right channel
220	xm1 = pSrc[pos_int+1];
221	x0 = pSrc[pos_int+3];
222	x1 = pSrc[pos_int+5];
223	x2 = pSrc[pos_int+7];
224	a = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f;
225	b = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f;
226	c = (x1 - xm1) * 0.5f;
227	#if ENABLE_FILTER
228	pEngine->pOutputRight[i++] += this->FilterRight.Apply(&bq_base, &bq_main, volume_right * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
229	#else // no filter
230	pEngine->pOutputRight[i++] += volume_right * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
231	#endif // ENABLE_FILTER
232	#endif // USE_LINEAR_INTERPOLATION
233
234	this->Pos += pitch;
235	}
236
237	inline void InterpolateOneStep_Mono(sample_t* pSrc, int& i, float volume_left, float volume_right, float& pitch, biquad_param_t& bq_base, biquad_param_t& bq_main) {
238	int pos_int = RTMath::DoubleToInt(this->Pos); // integer position
239	float pos_fract = this->Pos - pos_int; // fractional part of position
240
241	#if USE_LINEAR_INTERPOLATION
242	float sample_point = pSrc[pos_int] + pos_fract * (pSrc[pos_int+1] - pSrc[pos_int]);
243	#else // polynomial interpolation
244	float xm1 = pSrc[pos_int];
245	float x0 = pSrc[pos_int+1];
246	float x1 = pSrc[pos_int+2];
247	float x2 = pSrc[pos_int+3];
248	float a = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f;
249	float b = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f;
250	float c = (x1 - xm1) * 0.5f;
251	float sample_point = (((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0;
252	#endif // USE_LINEAR_INTERPOLATION
253
254	#if ENABLE_FILTER
255	sample_point = this->FilterLeft.Apply(&bq_base, &bq_main, sample_point);
256	#endif // ENABLE_FILTER
257
258	pEngine->pOutputLeft[i] += sample_point * volume_left;
259	pEngine->pOutputRight[i++] += sample_point * volume_right;
260
261	this->Pos += pitch;
262	}
263
264	inline float CrossfadeAttenuation(uint8_t& CrossfadeControllerValue) {
265	return (CrossfadeControllerValue <= pDimRgn->Crossfade.in_start) ? 0.0f
266	: (CrossfadeControllerValue < pDimRgn->Crossfade.in_end) ? float(CrossfadeControllerValue - pDimRgn->Crossfade.in_start) / float(pDimRgn->Crossfade.in_end - pDimRgn->Crossfade.in_start)
267	: (CrossfadeControllerValue <= pDimRgn->Crossfade.out_start) ? 1.0f
268	: (CrossfadeControllerValue < pDimRgn->Crossfade.out_end) ? float(CrossfadeControllerValue - pDimRgn->Crossfade.out_start) / float(pDimRgn->Crossfade.out_end - pDimRgn->Crossfade.out_start)
269	: 0.0f;
270	}
271
272	inline float Constrain(float ValueToCheck, float Min, float Max) {
273	if (ValueToCheck > Max) ValueToCheck = Max;
274	else if (ValueToCheck < Min) ValueToCheck = Min;
275	return ValueToCheck;
276	}
277	};
278
279	}} // namespace LinuxSampler::gig
280
281	#endif // __LS_GIG_VOICE_H__