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 "../../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

            // Methods
            Voice();
           ~Voice();
            void Kill(Event* pKillEvent);
            void KillImmediately();
            void Render(uint Samples);
            void Reset();
            void SetOutput(AudioOutputDevice* pAudioOutputDevice);
            void SetEngine(Engine* pEngine);
            int  Trigger(Event* pNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument, int iLayer = 0, bool ReleaseTriggerVoice = false);
            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                       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
            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).
            Event*                      pKillEvent;         ///< Event which caused this voice to be killed

            // 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 USE_LINEAR_INTERPOLATION
                    #if ENABLE_FILTER
                        // left channel
                        pEngine->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
                        pEngine->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
                        pEngine->pOutputLeft[i]    += effective_volume * (pSrc[pos_int]   + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int]));
                        // right channel
                        pEngine->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   = (3.0f * (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, effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
                    #else // no filter
                        pEngine->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   = (3.0f * (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, effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
                    #else // no filter
                        pEngine->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 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   = (3.0f * (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 = 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

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

                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	schoenebeck	53	/***************************************************************************
2			* *
3			* LinuxSampler - modular, streaming capable sampler *
4			* *
5	schoenebeck	56	* Copyright (C) 2003, 2004 by Benno Senoner and Christian Schoenebeck *
6	schoenebeck	53	* *
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	schoenebeck	203	#include "../../drivers/audio/AudioOutputDevice.h"
36	schoenebeck	53	#include "../../lib/fileloader/libgig/gig.h"
37	schoenebeck	80	#include "../common/BiquadFilter.h"
38	schoenebeck	53	#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	schoenebeck	80	#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	schoenebeck	53	#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	schoenebeck	242	// 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	schoenebeck	53	// Attributes
91	schoenebeck	242	type_t Type; ///< Voice Type
92	schoenebeck	53	int MIDIKey; ///< MIDI key number of the key that triggered the voice
93	schoenebeck	239	uint KeyGroup;
94	schoenebeck	53	DiskThread* pDiskThread; ///< Pointer to the disk thread, to be able to order a disk stream and later to delete the stream again
95
96			// Methods
97			Voice();
98			~Voice();
99	schoenebeck	239	void Kill(Event* pKillEvent);
100			void KillImmediately();
101	schoenebeck	53	void Render(uint Samples);
102			void Reset();
103			void SetOutput(AudioOutputDevice* pAudioOutputDevice);
104			void SetEngine(Engine* pEngine);
105	schoenebeck	242	int Trigger(Event* pNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument, int iLayer = 0, bool ReleaseTriggerVoice = false);
106	schoenebeck	53	inline bool IsActive() { return Active; }
107			private:
108			// Types
109			enum playback_state_t {
110			playback_state_ram,
111			playback_state_disk,
112			playback_state_end
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	schoenebeck	236	float CrossfadeVolume; ///< Current attenuation level caused by a crossfade (only if a crossfade is defined of course)
119	schoenebeck	53	double Pos; ///< Current playback position in sample
120			double PitchBase; ///< Basic pitch depth, stays the same for the whole life time of the voice
121			double PitchBend; ///< Current pitch value of the pitchbend wheel
122			::gig::Sample* pSample; ///< Pointer to the sample to be played back
123			::gig::Region* pRegion; ///< Pointer to the articulation information of the respective keyboard region of this voice
124	schoenebeck	236	::gig::DimensionRegion* pDimRgn; ///< Pointer to the articulation information of current dimension region of this voice
125	schoenebeck	53	bool Active; ///< If this voice object is currently in usage
126			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
127			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
128			Stream::reference_t DiskStreamRef; ///< Reference / link to the disk stream
129			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.
130			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
131			int LoopCyclesLeft; ///< In case there is a RAMLoop and it's not an endless loop; reflects number of loop cycles left to be passed
132			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
133			EGADSR* pEG1; ///< Envelope Generator 1 (Amplification)
134			EGADSR* pEG2; ///< Envelope Generator 2 (Filter cutoff frequency)
135			EGDecay* pEG3; ///< Envelope Generator 3 (Pitch)
136			Filter FilterLeft;
137			Filter FilterRight;
138			midi_ctrl VCFCutoffCtrl;
139			midi_ctrl VCFResonanceCtrl;
140			int FilterUpdateCounter; ///< Used to update filter parameters all FILTER_UPDATE_PERIOD samples
141			static const float FILTER_CUTOFF_COEFF;
142	schoenebeck	80	static const int FILTER_UPDATE_MASK;
143	schoenebeck	53	VCAManipulator* pVCAManipulator;
144			VCFCManipulator* pVCFCManipulator;
145			VCOManipulator* pVCOManipulator;
146			LFO<gig::VCAManipulator>* pLFO1; ///< Low Frequency Oscillator 1 (Amplification)
147			LFO<gig::VCFCManipulator>* pLFO2; ///< Low Frequency Oscillator 2 (Filter cutoff frequency)
148			LFO<gig::VCOManipulator>* pLFO3; ///< Low Frequency Oscillator 3 (Pitch)
149			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).
150	schoenebeck	239	Event* pKillEvent; ///< Event which caused this voice to be killed
151	schoenebeck	53
152			// Static Methods
153			static float CalculateFilterCutoffCoeff();
154	schoenebeck	80	static int CalculateFilterUpdateMask();
155	schoenebeck	53
156			// Methods
157			void ProcessEvents(uint Samples);
158	schoenebeck	80	#if ENABLE_FILTER
159			void CalculateBiquadParameters(uint Samples);
160			#endif // ENABLE_FILTER
161	schoenebeck	53	void Interpolate(uint Samples, sample_t* pSrc, uint Skip);
162			void InterpolateAndLoop(uint Samples, sample_t* pSrc, uint Skip);
163	schoenebeck	80	inline void InterpolateOneStep_Stereo(sample_t* pSrc, int& i, float& effective_volume, float& pitch, biquad_param_t& bq_base, biquad_param_t& bq_main) {
164	schoenebeck	53	int pos_int = RTMath::DoubleToInt(this->Pos); // integer position
165			float pos_fract = this->Pos - pos_int; // fractional part of position
166			pos_int <<= 1;
167
168			#if USE_LINEAR_INTERPOLATION
169			#if ENABLE_FILTER
170			// left channel
171	schoenebeck	225	pEngine->pOutputLeft[i] += this->FilterLeft.Apply(&bq_base, &bq_main, effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int])));
172	schoenebeck	53	// right channel
173	schoenebeck	225	pEngine->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])));
174	schoenebeck	53	#else // no filter
175			// left channel
176	schoenebeck	225	pEngine->pOutputLeft[i] += effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int]));
177	schoenebeck	53	// right channel
178	schoenebeck	225	pEngine->pOutputRight[i++] += effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1]));
179	schoenebeck	53	#endif // ENABLE_FILTER
180			#else // polynomial interpolation
181			// calculate left channel
182			float xm1 = pSrc[pos_int];
183			float x0 = pSrc[pos_int+2];
184			float x1 = pSrc[pos_int+4];
185			float x2 = pSrc[pos_int+6];
186	letz	99	float a = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f;
187			float b = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f;
188			float c = (x1 - xm1) * 0.5f;
189	schoenebeck	53	#if ENABLE_FILTER
190	schoenebeck	225	pEngine->pOutputLeft[i] += this->FilterLeft.Apply(&bq_base, &bq_main, effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
191	schoenebeck	53	#else // no filter
192	schoenebeck	225	pEngine->pOutputLeft[i] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
193	schoenebeck	53	#endif // ENABLE_FILTER
194
195			//calculate right channel
196			xm1 = pSrc[pos_int+1];
197			x0 = pSrc[pos_int+3];
198			x1 = pSrc[pos_int+5];
199			x2 = pSrc[pos_int+7];
200	letz	99	a = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f;
201			b = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f;
202			c = (x1 - xm1) * 0.5f;
203	schoenebeck	53	#if ENABLE_FILTER
204	schoenebeck	225	pEngine->pOutputRight[i++] += this->FilterRight.Apply(&bq_base, &bq_main, effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
205	schoenebeck	53	#else // no filter
206	schoenebeck	225	pEngine->pOutputRight[i++] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
207	schoenebeck	53	#endif // ENABLE_FILTER
208			#endif // USE_LINEAR_INTERPOLATION
209
210			this->Pos += pitch;
211			}
212	schoenebeck	97
213	schoenebeck	80	inline void InterpolateOneStep_Mono(sample_t* pSrc, int& i, float& effective_volume, float& pitch, biquad_param_t& bq_base, biquad_param_t& bq_main) {
214	schoenebeck	53	int pos_int = RTMath::DoubleToInt(this->Pos); // integer position
215			float pos_fract = this->Pos - pos_int; // fractional part of position
216
217			#if USE_LINEAR_INTERPOLATION
218			float sample_point = effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+1] - pSrc[pos_int]));
219			#else // polynomial interpolation
220			float xm1 = pSrc[pos_int];
221			float x0 = pSrc[pos_int+1];
222			float x1 = pSrc[pos_int+2];
223			float x2 = pSrc[pos_int+3];
224	letz	99	float a = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f;
225			float b = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f;
226			float c = (x1 - xm1) * 0.5f;
227	schoenebeck	53	float sample_point = effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
228			#endif // USE_LINEAR_INTERPOLATION
229
230			#if ENABLE_FILTER
231	schoenebeck	80	sample_point = this->FilterLeft.Apply(&bq_base, &bq_main, sample_point);
232	schoenebeck	53	#endif // ENABLE_FILTER
233
234	schoenebeck	225	pEngine->pOutputLeft[i] += sample_point;
235			pEngine->pOutputRight[i++] += sample_point;
236	schoenebeck	53
237			this->Pos += pitch;
238			}
239	schoenebeck	97
240	schoenebeck	236	inline float CrossfadeAttenuation(uint8_t& CrossfadeControllerValue) {
241			return (CrossfadeControllerValue <= pDimRgn->Crossfade.in_start) ? 0.0f
242			: (CrossfadeControllerValue < pDimRgn->Crossfade.in_end) ? float(CrossfadeControllerValue - pDimRgn->Crossfade.in_start) / float(pDimRgn->Crossfade.in_end - pDimRgn->Crossfade.in_start)
243			: (CrossfadeControllerValue <= pDimRgn->Crossfade.out_start) ? 1.0f
244			: (CrossfadeControllerValue < pDimRgn->Crossfade.out_end) ? float(CrossfadeControllerValue - pDimRgn->Crossfade.out_start) / float(pDimRgn->Crossfade.out_end - pDimRgn->Crossfade.out_start)
245			: 0.0f;
246			}
247
248	schoenebeck	53	inline float Constrain(float ValueToCheck, float Min, float Max) {
249			if (ValueToCheck > Max) ValueToCheck = Max;
250			else if (ValueToCheck < Min) ValueToCheck = Min;
251			return ValueToCheck;
252			}
253			};
254
255			}} // namespace LinuxSampler::gig
256
257			#endif // __LS_GIG_VOICE_H__