v0_1_0/src/voice.h

/***************************************************************************
 *                                                                         *
 *   LinuxSampler - modular, streaming capable sampler                     *
 *                                                                         *
 *   Copyright (C) 2003 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 __VOICE_H__
#define __VOICE_H__

#include "global.h"
#include "rtmath.h"
#include "diskthread.h"
#include "ringbuffer.h"
#include "stream.h"
#include "gig.h"
#include "eg_vca.h"
#include "eg_d.h"
#include "rtelmemorypool.h"
#include "audiothread.h"
#include "filter.h"
#include "lfo.h"

#define USE_LINEAR_INTERPOLATION        1  ///< set to 0 if you prefer cubic interpolation (slower, better quality)
#define ENABLE_FILTER                   0  ///< 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)
#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


/// 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)
};

class Voice {
    public:
        // Attributes
        int                 MIDIKey;          ///< MIDI key number of the key that triggered the voice
        uint                ReleaseVelocity;  ///< Reflects the release velocity value if a note-off command arrived for the voice.

        // Static Attributes
        static DiskThread*  pDiskThread;  ///< Pointer to the disk thread, to be able to order a disk stream and later to delete the stream again
        static AudioThread* pEngine;      ///< Pointer to the engine, to be able to access the event lists.

        // Methods
        Voice();
       ~Voice();
        void Kill();
        void Render(uint Samples);
        void Reset();
        int  Trigger(ModulationSystem::Event* pNoteOnEvent, int PitchBend, gig::Instrument* pInstrument);
        inline bool IsActive()                                              { return Active; }
        inline void SetOutputLeft(float* pOutput, uint MaxSamplesPerCycle)  { this->pOutputLeft  = pOutput; this->MaxSamplesPerCycle = MaxSamplesPerCycle; }
        inline void SetOutputRight(float* pOutput, uint MaxSamplesPerCycle) { this->pOutputRight = pOutput; this->MaxSamplesPerCycle = MaxSamplesPerCycle; }
    private:
        // Types
        enum playback_state_t {
            playback_state_ram,
            playback_state_disk,
            playback_state_end
        };

        // Attributes
        float                Volume;             ///< Volume level of the voice
        float*               pOutputLeft;        ///< Audio output buffer (left channel)
        float*               pOutputRight;       ///< Audio output buffer (right channel)
        uint                 MaxSamplesPerCycle; ///< Size of each audio output buffer
        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
        bool                 Active;             ///< If this voice object is currently in usage
        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
        EG_VCA*              pEG1;               ///< Envelope Generator 1 (Amplification)
        EG_VCA*              pEG2;               ///< Envelope Generator 2 (Filter cutoff frequency)
        EG_D*                pEG3;               ///< Envelope Generator 3 (Pitch)
        GigFilter            FilterLeft;
        GigFilter            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;
        LFO<VCAManipulator>* pLFO1;              ///< Low Frequency Oscillator 1 (Amplification)
        LFO<VCFCManipulator>* pLFO2;             ///< Low Frequency Oscillator 2 (Filter cutoff frequency)
        LFO<VCOManipulator>* pLFO3;              ///< Low Frequency Oscillator 3 (Pitch)
        ModulationSystem::Event* pTriggerEvent;  ///< 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).

        // Static Methods
        static float CalculateFilterCutoffCoeff();

        // Methods
        void        ProcessEvents(uint Samples);
        void        Interpolate(uint Samples, sample_t* pSrc, uint Skip);
        void        InterpolateAndLoop(uint Samples, sample_t* pSrc, uint Skip);
        inline void InterpolateOneStep_Stereo(sample_t* pSrc, int& i, float& effective_volume, float& pitch, float& cutoff, float& resonance) {
            int   pos_int   = RTMath::DoubleToInt(this->Pos);  // integer position
            float pos_fract = this->Pos - pos_int;             // fractional part of position
            pos_int <<= 1;

            #if ENABLE_FILTER
                UpdateFilter_Stereo(cutoff + FILTER_CUTOFF_MIN, resonance);
            #endif // ENABLE_FILTER

            #if USE_LINEAR_INTERPOLATION
                #if ENABLE_FILTER
                    // left channel
                    this->pOutputLeft[i]    += this->FilterLeft.Apply(effective_volume * (pSrc[pos_int]   + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int])));
                    // right channel
                    this->pOutputRight[i++] += this->FilterRight.Apply(effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1])));
                #else // no filter
                    // left channel
                    this->pOutputLeft[i]    += effective_volume * (pSrc[pos_int]   + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int]));
                    // right channel
                    this->pOutputRight[i++] += effective_volume * (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 * (x0 - x1) - xm1 + x2) / 2;
                float b   = 2 * x1 + xm1 - (5 * x0 + x2) / 2;
                float c   = (x1 - xm1) / 2;
                #if ENABLE_FILTER
                    this->pOutputLeft[i] += this->FilterLeft.Apply(effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
                #else // no filter
                    this->pOutputLeft[i] += effective_volume * ((((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 * (x0 - x1) - xm1 + x2) / 2;
                b   = 2 * x1 + xm1 - (5 * x0 + x2) / 2;
                c   = (x1 - xm1) / 2;
                #if ENABLE_FILTER
                    this->pOutputRight[i++] += this->FilterRight.Apply(effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
                #else // no filter
                    this->pOutputRight[i++] += effective_volume * ((((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& effective_volume, float& pitch, float& cutoff, float& resonance) {
            int   pos_int   = RTMath::DoubleToInt(this->Pos);  // integer position
            float pos_fract = this->Pos - pos_int;             // fractional part of position

            #if ENABLE_FILTER
                UpdateFilter_Mono(cutoff + FILTER_CUTOFF_MIN, resonance);
            #endif // ENABLE_FILTER

            #if USE_LINEAR_INTERPOLATION
                float sample_point  = effective_volume * (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 * (x0 - x1) - xm1 + x2) / 2;
                float b   = 2 * x1 + xm1 - (5 * x0 + x2) / 2;
                float c   = (x1 - xm1) / 2;
                float sample_point = effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
            #endif // USE_LINEAR_INTERPOLATION

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

            this->pOutputLeft[i]    += sample_point;
            this->pOutputRight[i++] += sample_point;

            this->Pos += pitch;
        }
        inline void UpdateFilter_Stereo(float cutoff, float& resonance) {
            if (!(++FilterUpdateCounter % FILTER_UPDATE_PERIOD) && (cutoff != FilterLeft.Cutoff() || resonance != FilterLeft.Resonance())) {
                FilterLeft.SetParameters(cutoff, resonance, ModulationSystem::SampleRate());
                FilterRight.SetParameters(cutoff, resonance, ModulationSystem::SampleRate());
            }
        }
        inline void UpdateFilter_Mono(float cutoff, float& resonance) {
            if (!(++FilterUpdateCounter % FILTER_UPDATE_PERIOD) && (cutoff != FilterLeft.Cutoff() || resonance != FilterLeft.Resonance())) {
                FilterLeft.SetParameters(cutoff, resonance, ModulationSystem::SampleRate());
            }
        }
        inline float Constrain(float ValueToCheck, float Min, float Max) {
            if      (ValueToCheck > Max) ValueToCheck = Max;
            else if (ValueToCheck < Min) ValueToCheck = Min;
            return ValueToCheck;
        }
};

#endif // __VOICE_H__
1	schoenebeck	9	/***************************************************************************
2			* *
3			* LinuxSampler - modular, streaming capable sampler *
4			* *
5			* Copyright (C) 2003 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 __VOICE_H__
24			#define __VOICE_H__
25
26			#include "global.h"
27	schoenebeck	40	#include "rtmath.h"
28	schoenebeck	9	#include "diskthread.h"
29			#include "ringbuffer.h"
30			#include "stream.h"
31			#include "gig.h"
32	schoenebeck	30	#include "eg_vca.h"
33	schoenebeck	40	#include "eg_d.h"
34	schoenebeck	32	#include "rtelmemorypool.h"
35			#include "audiothread.h"
36	schoenebeck	38	#include "filter.h"
37	schoenebeck	39	#include "lfo.h"
38	schoenebeck	9
39	schoenebeck	18	#define USE_LINEAR_INTERPOLATION 1 ///< set to 0 if you prefer cubic interpolation (slower, better quality)
40	schoenebeck	38	#define ENABLE_FILTER 0 ///< if set to 0 then filter (VCF) code is ignored on compile time
41			#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)
42			#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
43	schoenebeck	40	#define FILTER_CUTOFF_MAX 10000.0f ///< maximum cutoff frequency (10kHz)
44			#define FILTER_CUTOFF_MIN 100.0f ///< minimum cutoff frequency (100Hz)
45	schoenebeck	9
46	schoenebeck	38	// Uncomment following line to override external cutoff controller
47			//#define OVERRIDE_FILTER_CUTOFF_CTRL 1 ///< set to an arbitrary MIDI control change controller (e.g. 1 for 'modulation wheel')
48
49			// Uncomment following line to override external resonance controller
50			//#define OVERRIDE_FILTER_RES_CTRL 91 ///< set to an arbitrary MIDI control change controller (e.g. 91 for 'effect 1 depth')
51
52			// Uncomment following line to override filter type
53			//#define OVERRIDE_FILTER_TYPE gig::vcf_type_lowpass ///< either gig::vcf_type_lowpass, gig::vcf_type_bandpass or gig::vcf_type_highpass
54
55
56			/// Reflects a MIDI controller
57			struct midi_ctrl {
58			uint8_t controller; ///< MIDI control change controller number
59	schoenebeck	39	uint8_t value; ///< Current MIDI controller value
60			float fvalue; ///< Transformed / effective value (e.g. volume level or filter cutoff frequency)
61	schoenebeck	38	};
62
63	schoenebeck	9	class Voice {
64			public:
65	schoenebeck	12	// Attributes
66	schoenebeck	32	int MIDIKey; ///< MIDI key number of the key that triggered the voice
67			uint ReleaseVelocity; ///< Reflects the release velocity value if a note-off command arrived for the voice.
68	schoenebeck	12
69	schoenebeck	32	// Static Attributes
70			static DiskThread* pDiskThread; ///< Pointer to the disk thread, to be able to order a disk stream and later to delete the stream again
71			static AudioThread* pEngine; ///< Pointer to the engine, to be able to access the event lists.
72
73	schoenebeck	9	// Methods
74	schoenebeck	32	Voice();
75	schoenebeck	9	~Voice();
76			void Kill();
77	schoenebeck	31	void Render(uint Samples);
78	schoenebeck	35	void Reset();
79	schoenebeck	40	int Trigger(ModulationSystem::Event* pNoteOnEvent, int PitchBend, gig::Instrument* pInstrument);
80	schoenebeck	31	inline bool IsActive() { return Active; }
81			inline void SetOutputLeft(float* pOutput, uint MaxSamplesPerCycle) { this->pOutputLeft = pOutput; this->MaxSamplesPerCycle = MaxSamplesPerCycle; }
82			inline void SetOutputRight(float* pOutput, uint MaxSamplesPerCycle) { this->pOutputRight = pOutput; this->MaxSamplesPerCycle = MaxSamplesPerCycle; }
83	schoenebeck	9	private:
84			// Types
85			enum playback_state_t {
86			playback_state_ram,
87			playback_state_disk,
88			playback_state_end
89			};
90
91			// Attributes
92	schoenebeck	31	float Volume; ///< Volume level of the voice
93			float* pOutputLeft; ///< Audio output buffer (left channel)
94			float* pOutputRight; ///< Audio output buffer (right channel)
95			uint MaxSamplesPerCycle; ///< Size of each audio output buffer
96			double Pos; ///< Current playback position in sample
97	schoenebeck	40	double PitchBase; ///< Basic pitch depth, stays the same for the whole life time of the voice
98			double PitchBend; ///< Current pitch value of the pitchbend wheel
99	schoenebeck	31	gig::Sample* pSample; ///< Pointer to the sample to be played back
100			gig::Region* pRegion; ///< Pointer to the articulation information of the respective keyboard region of this voice
101			bool Active; ///< If this voice object is currently in usage
102			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
103			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
104			Stream::reference_t DiskStreamRef; ///< Reference / link to the disk stream
105			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.
106			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
107			int LoopCyclesLeft; ///< In case there is a RAMLoop and it's not an endless loop; reflects number of loop cycles left to be passed
108	schoenebeck	32	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
109	schoenebeck	40	EG_VCA* pEG1; ///< Envelope Generator 1 (Amplification)
110			EG_VCA* pEG2; ///< Envelope Generator 2 (Filter cutoff frequency)
111			EG_D* pEG3; ///< Envelope Generator 3 (Pitch)
112	schoenebeck	38	GigFilter FilterLeft;
113			GigFilter FilterRight;
114			midi_ctrl VCFCutoffCtrl;
115			midi_ctrl VCFResonanceCtrl;
116	schoenebeck	40	int FilterUpdateCounter; ///< Used to update filter parameters all FILTER_UPDATE_PERIOD samples
117			static const float FILTER_CUTOFF_COEFF;
118			LFO<VCAManipulator>* pLFO1; ///< Low Frequency Oscillator 1 (Amplification)
119			LFO<VCFCManipulator>* pLFO2; ///< Low Frequency Oscillator 2 (Filter cutoff frequency)
120			LFO<VCOManipulator>* pLFO3; ///< Low Frequency Oscillator 3 (Pitch)
121			ModulationSystem::Event* pTriggerEvent; ///< 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).
122	schoenebeck	9
123	schoenebeck	40	// Static Methods
124			static float CalculateFilterCutoffCoeff();
125
126	schoenebeck	9	// Methods
127	schoenebeck	32	void ProcessEvents(uint Samples);
128			void Interpolate(uint Samples, sample_t* pSrc, uint Skip);
129			void InterpolateAndLoop(uint Samples, sample_t* pSrc, uint Skip);
130	schoenebeck	38	inline void InterpolateOneStep_Stereo(sample_t* pSrc, int& i, float& effective_volume, float& pitch, float& cutoff, float& resonance) {
131	schoenebeck	40	int pos_int = RTMath::DoubleToInt(this->Pos); // integer position
132			float pos_fract = this->Pos - pos_int; // fractional part of position
133	schoenebeck	26	pos_int <<= 1;
134	schoenebeck	39
135	schoenebeck	38	#if ENABLE_FILTER
136	schoenebeck	40	UpdateFilter_Stereo(cutoff + FILTER_CUTOFF_MIN, resonance);
137	schoenebeck	38	#endif // ENABLE_FILTER
138	schoenebeck	26
139			#if USE_LINEAR_INTERPOLATION
140	schoenebeck	38	#if ENABLE_FILTER
141			// left channel
142			this->pOutputLeft[i] += this->FilterLeft.Apply(effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int])));
143			// right channel
144			this->pOutputRight[i++] += this->FilterRight.Apply(effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1])));
145	schoenebeck	39	#else // no filter
146	schoenebeck	38	// left channel
147			this->pOutputLeft[i] += effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int]));
148			// right channel
149			this->pOutputRight[i++] += effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1]));
150			#endif // ENABLE_FILTER
151	schoenebeck	26	#else // polynomial interpolation
152			// calculate left channel
153			float xm1 = pSrc[pos_int];
154			float x0 = pSrc[pos_int+2];
155			float x1 = pSrc[pos_int+4];
156			float x2 = pSrc[pos_int+6];
157			float a = (3 * (x0 - x1) - xm1 + x2) / 2;
158			float b = 2 * x1 + xm1 - (5 * x0 + x2) / 2;
159			float c = (x1 - xm1) / 2;
160	schoenebeck	38	#if ENABLE_FILTER
161			this->pOutputLeft[i] += this->FilterLeft.Apply(effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
162			#else // no filter
163			this->pOutputLeft[i] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
164	schoenebeck	39	#endif // ENABLE_FILTER
165	schoenebeck	26
166			//calculate right channel
167			xm1 = pSrc[pos_int+1];
168			x0 = pSrc[pos_int+3];
169			x1 = pSrc[pos_int+5];
170			x2 = pSrc[pos_int+7];
171			a = (3 * (x0 - x1) - xm1 + x2) / 2;
172			b = 2 * x1 + xm1 - (5 * x0 + x2) / 2;
173			c = (x1 - xm1) / 2;
174	schoenebeck	38	#if ENABLE_FILTER
175			this->pOutputRight[i++] += this->FilterRight.Apply(effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
176			#else // no filter
177			this->pOutputRight[i++] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
178			#endif // ENABLE_FILTER
179	schoenebeck	26	#endif // USE_LINEAR_INTERPOLATION
180
181	schoenebeck	32	this->Pos += pitch;
182	schoenebeck	26	}
183	schoenebeck	38	inline void InterpolateOneStep_Mono(sample_t* pSrc, int& i, float& effective_volume, float& pitch, float& cutoff, float& resonance) {
184	schoenebeck	40	int pos_int = RTMath::DoubleToInt(this->Pos); // integer position
185			float pos_fract = this->Pos - pos_int; // fractional part of position
186	schoenebeck	39
187	schoenebeck	38	#if ENABLE_FILTER
188	schoenebeck	40	UpdateFilter_Mono(cutoff + FILTER_CUTOFF_MIN, resonance);
189	schoenebeck	38	#endif // ENABLE_FILTER
190	schoenebeck	26
191			#if USE_LINEAR_INTERPOLATION
192			float sample_point = effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+1] - pSrc[pos_int]));
193			#else // polynomial interpolation
194			float xm1 = pSrc[pos_int];
195			float x0 = pSrc[pos_int+1];
196			float x1 = pSrc[pos_int+2];
197			float x2 = pSrc[pos_int+3];
198			float a = (3 * (x0 - x1) - xm1 + x2) / 2;
199			float b = 2 * x1 + xm1 - (5 * x0 + x2) / 2;
200			float c = (x1 - xm1) / 2;
201			float sample_point = effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
202	schoenebeck	39	#endif // USE_LINEAR_INTERPOLATION
203	schoenebeck	26
204	schoenebeck	39	#if ENABLE_FILTER
205	schoenebeck	38	sample_point = this->FilterLeft.Apply(sample_point);
206			#endif // ENABLE_FILTER
207
208	schoenebeck	31	this->pOutputLeft[i] += sample_point;
209			this->pOutputRight[i++] += sample_point;
210	schoenebeck	26
211	schoenebeck	32	this->Pos += pitch;
212	schoenebeck	26	}
213	schoenebeck	39	inline void UpdateFilter_Stereo(float cutoff, float& resonance) {
214	schoenebeck	40	if (!(++FilterUpdateCounter % FILTER_UPDATE_PERIOD) && (cutoff != FilterLeft.Cutoff() \|\| resonance != FilterLeft.Resonance())) {
215	schoenebeck	38	FilterLeft.SetParameters(cutoff, resonance, ModulationSystem::SampleRate());
216			FilterRight.SetParameters(cutoff, resonance, ModulationSystem::SampleRate());
217			}
218			}
219	schoenebeck	39	inline void UpdateFilter_Mono(float cutoff, float& resonance) {
220	schoenebeck	40	if (!(++FilterUpdateCounter % FILTER_UPDATE_PERIOD) && (cutoff != FilterLeft.Cutoff() \|\| resonance != FilterLeft.Resonance())) {
221	schoenebeck	38	FilterLeft.SetParameters(cutoff, resonance, ModulationSystem::SampleRate());
222			}
223			}
224			inline float Constrain(float ValueToCheck, float Min, float Max) {
225			if (ValueToCheck > Max) ValueToCheck = Max;
226			else if (ValueToCheck < Min) ValueToCheck = Min;
227	schoenebeck	39	return ValueToCheck;
228	schoenebeck	38	}
229	schoenebeck	9	};
230
231			#endif // __VOICE_H__