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/RTELMemoryPool.h"
#include "../../audiodriver/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:
            // Attributes
            int          MIDIKey;      ///< MIDI key number of the key that triggered the voice
            DiskThread*  pDiskThread;  ///< Pointer to the disk thread, to be able to order a disk stream and later to delete the stream again

            // Methods
            Voice();
           ~Voice();
            void Kill();
            void Render(uint Samples);
            void Reset();
            void SetOutput(AudioOutputDevice* pAudioOutputDevice);
            void SetEngine(Engine* pEngine);
            int  Trigger(Event* pNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument);
            inline bool IsActive() { return Active; }
        private:
            // Types
            enum playback_state_t {
                playback_state_ram,
                playback_state_disk,
                playback_state_end
            };

            // 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*                      pOutputLeft;        ///< Audio output channel buffer (left)
            float*                      pOutputRight;       ///< Audio output channel buffer (right)
            uint                        SampleRate;         ///< Sample rate of the engines output audio signal (in Hz)
            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
            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)
            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();
            static int   CalculateFilterUpdateMask();

            // Methods
            void        ProcessEvents(uint Samples);
            #if ENABLE_FILTER
            void        CalculateBiquadParameters(uint Samples);
            #endif // ENABLE_FILTER
            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, 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 0 //ENABLE_FILTER
                    UpdateFilter_Stereo(cutoff + FILTER_CUTOFF_MIN, resonance);
                #endif // ENABLE_FILTER

                #if USE_LINEAR_INTERPOLATION
                    #if ENABLE_FILTER
                        // left channel
                        pOutputLeft[i]    += this->FilterLeft.Apply(&bq_base, &bq_main, effective_volume * (pSrc[pos_int]   + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int])));
                        // right channel
                        pOutputRight[i++] += this->FilterRight.Apply(&bq_base, &bq_main, effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1])));
                    #else // no filter
                        // left channel
                        pOutputLeft[i]    += effective_volume * (pSrc[pos_int]   + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int]));
                        // right channel
                        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
                        pOutputLeft[i] += this->FilterLeft.Apply(&bq_base, &bq_main, effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
                    #else // no filter
                        pOutputRight[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
                        pOutputLeft[i++] += this->FilterRight.Apply(&bq_base, &bq_main, effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
                    #else // no filter
                        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,  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 0 //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(&bq_base, &bq_main, sample_point);
                #endif // ENABLE_FILTER

                pOutputLeft[i]    += sample_point;
                pOutputRight[i++] += sample_point;

                this->Pos += pitch;
            }
#if 0
            inline void UpdateFilter_Stereo(float cutoff, float& resonance) {
                if (!(++FilterUpdateCounter % FILTER_UPDATE_PERIOD) && (cutoff != FilterLeft.Cutoff() || resonance != FilterLeft.Resonance())) {
                    FilterLeft.SetParameters(cutoff, resonance, SampleRate);
                    FilterRight.SetParameters(cutoff, resonance, SampleRate);
                }
            }
            inline void UpdateFilter_Mono(float cutoff, float& resonance) {
                if (!(++FilterUpdateCounter % FILTER_UPDATE_PERIOD) && (cutoff != FilterLeft.Cutoff() || resonance != FilterLeft.Resonance())) {
                    FilterLeft.SetParameters(cutoff, resonance, SampleRate);
                }
            }
#endif
            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/RTELMemoryPool.h"
35	#include "../../audiodriver/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	// Attributes
84	int MIDIKey; ///< MIDI key number of the key that triggered the voice
85	DiskThread* pDiskThread; ///< Pointer to the disk thread, to be able to order a disk stream and later to delete the stream again
86
87	// Methods
88	Voice();
89	~Voice();
90	void Kill();
91	void Render(uint Samples);
92	void Reset();
93	void SetOutput(AudioOutputDevice* pAudioOutputDevice);
94	void SetEngine(Engine* pEngine);
95	int Trigger(Event* pNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument);
96	inline bool IsActive() { return Active; }
97	private:
98	// Types
99	enum playback_state_t {
100	playback_state_ram,
101	playback_state_disk,
102	playback_state_end
103	};
104
105	// Attributes
106	gig::Engine* pEngine; ///< Pointer to the sampler engine, to be able to access the event lists.
107	float Volume; ///< Volume level of the voice
108	float* pOutputLeft; ///< Audio output channel buffer (left)
109	float* pOutputRight; ///< Audio output channel buffer (right)
110	uint SampleRate; ///< Sample rate of the engines output audio signal (in Hz)
111	uint MaxSamplesPerCycle; ///< Size of each audio output buffer
112	double Pos; ///< Current playback position in sample
113	double PitchBase; ///< Basic pitch depth, stays the same for the whole life time of the voice
114	double PitchBend; ///< Current pitch value of the pitchbend wheel
115	::gig::Sample* pSample; ///< Pointer to the sample to be played back
116	::gig::Region* pRegion; ///< Pointer to the articulation information of the respective keyboard region of this voice
117	bool Active; ///< If this voice object is currently in usage
118	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
119	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
120	Stream::reference_t DiskStreamRef; ///< Reference / link to the disk stream
121	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.
122	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
123	int LoopCyclesLeft; ///< In case there is a RAMLoop and it's not an endless loop; reflects number of loop cycles left to be passed
124	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
125	EGADSR* pEG1; ///< Envelope Generator 1 (Amplification)
126	EGADSR* pEG2; ///< Envelope Generator 2 (Filter cutoff frequency)
127	EGDecay* pEG3; ///< Envelope Generator 3 (Pitch)
128	Filter FilterLeft;
129	Filter FilterRight;
130	midi_ctrl VCFCutoffCtrl;
131	midi_ctrl VCFResonanceCtrl;
132	int FilterUpdateCounter; ///< Used to update filter parameters all FILTER_UPDATE_PERIOD samples
133	static const float FILTER_CUTOFF_COEFF;
134	static const int FILTER_UPDATE_MASK;
135	VCAManipulator* pVCAManipulator;
136	VCFCManipulator* pVCFCManipulator;
137	VCOManipulator* pVCOManipulator;
138	LFO<gig::VCAManipulator>* pLFO1; ///< Low Frequency Oscillator 1 (Amplification)
139	LFO<gig::VCFCManipulator>* pLFO2; ///< Low Frequency Oscillator 2 (Filter cutoff frequency)
140	LFO<gig::VCOManipulator>* pLFO3; ///< Low Frequency Oscillator 3 (Pitch)
141	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).
142
143	// Static Methods
144	static float CalculateFilterCutoffCoeff();
145	static int CalculateFilterUpdateMask();
146
147	// Methods
148	void ProcessEvents(uint Samples);
149	#if ENABLE_FILTER
150	void CalculateBiquadParameters(uint Samples);
151	#endif // ENABLE_FILTER
152	void Interpolate(uint Samples, sample_t* pSrc, uint Skip);
153	void InterpolateAndLoop(uint Samples, sample_t* pSrc, uint Skip);
154	inline void InterpolateOneStep_Stereo(sample_t* pSrc, int& i, float& effective_volume, float& pitch, biquad_param_t& bq_base, biquad_param_t& bq_main) {
155	int pos_int = RTMath::DoubleToInt(this->Pos); // integer position
156	float pos_fract = this->Pos - pos_int; // fractional part of position
157	pos_int <<= 1;
158
159	#if 0 //ENABLE_FILTER
160	UpdateFilter_Stereo(cutoff + FILTER_CUTOFF_MIN, resonance);
161	#endif // ENABLE_FILTER
162
163	#if USE_LINEAR_INTERPOLATION
164	#if ENABLE_FILTER
165	// left channel
166	pOutputLeft[i] += this->FilterLeft.Apply(&bq_base, &bq_main, effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int])));
167	// right channel
168	pOutputRight[i++] += this->FilterRight.Apply(&bq_base, &bq_main, effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1])));
169	#else // no filter
170	// left channel
171	pOutputLeft[i] += effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int]));
172	// right channel
173	pOutputRight[i++] += effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1]));
174	#endif // ENABLE_FILTER
175	#else // polynomial interpolation
176	// calculate left channel
177	float xm1 = pSrc[pos_int];
178	float x0 = pSrc[pos_int+2];
179	float x1 = pSrc[pos_int+4];
180	float x2 = pSrc[pos_int+6];
181	float a = (3 * (x0 - x1) - xm1 + x2) / 2;
182	float b = 2 * x1 + xm1 - (5 * x0 + x2) / 2;
183	float c = (x1 - xm1) / 2;
184	#if ENABLE_FILTER
185	pOutputLeft[i] += this->FilterLeft.Apply(&bq_base, &bq_main, effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
186	#else // no filter
187	pOutputRight[i] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
188	#endif // ENABLE_FILTER
189
190	//calculate right channel
191	xm1 = pSrc[pos_int+1];
192	x0 = pSrc[pos_int+3];
193	x1 = pSrc[pos_int+5];
194	x2 = pSrc[pos_int+7];
195	a = (3 * (x0 - x1) - xm1 + x2) / 2;
196	b = 2 * x1 + xm1 - (5 * x0 + x2) / 2;
197	c = (x1 - xm1) / 2;
198	#if ENABLE_FILTER
199	pOutputLeft[i++] += this->FilterRight.Apply(&bq_base, &bq_main, effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
200	#else // no filter
201	pOutputRight[i++] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
202	#endif // ENABLE_FILTER
203	#endif // USE_LINEAR_INTERPOLATION
204
205	this->Pos += pitch;
206	}
207	inline void InterpolateOneStep_Mono(sample_t* pSrc, int& i, float& effective_volume, float& pitch, biquad_param_t& bq_base, biquad_param_t& bq_main) {
208	int pos_int = RTMath::DoubleToInt(this->Pos); // integer position
209	float pos_fract = this->Pos - pos_int; // fractional part of position
210
211	#if 0 //ENABLE_FILTER
212	UpdateFilter_Mono(cutoff + FILTER_CUTOFF_MIN, resonance);
213	#endif // ENABLE_FILTER
214
215	#if USE_LINEAR_INTERPOLATION
216	float sample_point = effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+1] - pSrc[pos_int]));
217	#else // polynomial interpolation
218	float xm1 = pSrc[pos_int];
219	float x0 = pSrc[pos_int+1];
220	float x1 = pSrc[pos_int+2];
221	float x2 = pSrc[pos_int+3];
222	float a = (3 * (x0 - x1) - xm1 + x2) / 2;
223	float b = 2 * x1 + xm1 - (5 * x0 + x2) / 2;
224	float c = (x1 - xm1) / 2;
225	float sample_point = effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
226	#endif // USE_LINEAR_INTERPOLATION
227
228	#if ENABLE_FILTER
229	sample_point = this->FilterLeft.Apply(&bq_base, &bq_main, sample_point);
230	#endif // ENABLE_FILTER
231
232	pOutputLeft[i] += sample_point;
233	pOutputRight[i++] += sample_point;
234
235	this->Pos += pitch;
236	}
237	#if 0
238	inline void UpdateFilter_Stereo(float cutoff, float& resonance) {
239	if (!(++FilterUpdateCounter % FILTER_UPDATE_PERIOD) && (cutoff != FilterLeft.Cutoff() \|\| resonance != FilterLeft.Resonance())) {
240	FilterLeft.SetParameters(cutoff, resonance, SampleRate);
241	FilterRight.SetParameters(cutoff, resonance, SampleRate);
242	}
243	}
244	inline void UpdateFilter_Mono(float cutoff, float& resonance) {
245	if (!(++FilterUpdateCounter % FILTER_UPDATE_PERIOD) && (cutoff != FilterLeft.Cutoff() \|\| resonance != FilterLeft.Resonance())) {
246	FilterLeft.SetParameters(cutoff, resonance, SampleRate);
247	}
248	}
249	#endif
250	inline float Constrain(float ValueToCheck, float Min, float Max) {
251	if (ValueToCheck > Max) ValueToCheck = Max;
252	else if (ValueToCheck < Min) ValueToCheck = Min;
253	return ValueToCheck;
254	}
255	};
256
257	}} // namespace LinuxSampler::gig
258
259	#endif // __LS_GIG_VOICE_H__