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

#define MAX_PITCH                       4  //FIXME: at the moment in octaves, should be changed into semitones
#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

// 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 Pitch, 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               Pitch;              ///< Current pitch depth (number of sample points to move on with each render step)
        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               EG1;
        GigFilter            FilterLeft;
        GigFilter            FilterRight;
        midi_ctrl            VCFCutoffCtrl;
        midi_ctrl            VCFResonanceCtrl;
        LFO*                 pLFO1;
        LFO*                 pLFO2;
        LFO*                 pLFO3;
        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).

        // 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   = double_to_int(this->Pos);  // integer position
            float pos_fract = this->Pos - pos_int;       // fractional part of position
            pos_int <<= 1;

            #if ENABLE_FILTER
                UpdateFilter_Stereo(cutoff + 20.0f, resonance); // 20Hz min.
            #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   = double_to_int(this->Pos);  // integer position
            float pos_fract = this->Pos - pos_int;       // fractional part of position

            #if ENABLE_FILTER
                UpdateFilter_Mono(cutoff + 20.0f, resonance); // 20Hz min.
            #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) {
            static int updatecounter = 0; // we update the filter all FILTER_UPDATE_PERIOD samples
            if (!(++updatecounter % 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) {
            static int updatecounter = 0; // we update the filter all FILTER_UPDATE_PERIOD samples
            if (!(++updatecounter % FILTER_UPDATE_PERIOD) && (cutoff != FilterLeft.Cutoff() || resonance != FilterLeft.Resonance())) {
                FilterLeft.SetParameters(cutoff, resonance, ModulationSystem::SampleRate());
            }
        }
        inline void ForceUpdateFilter_Stereo(float cutoff, float& resonance) {
            if (cutoff != FilterLeft.Cutoff() || resonance != FilterLeft.Resonance()) {
                FilterLeft.SetParameters(cutoff, resonance, ModulationSystem::SampleRate());
                FilterRight.SetParameters(cutoff, resonance, ModulationSystem::SampleRate());
            }
        }
        inline void ForceUpdateFilter_Mono(float cutoff, float& resonance) {
            if (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;
        }
        inline int double_to_int(double f) {
            #if ARCH_X86
            int i;
            __asm__ ("fistl %0" : "=m"(i) : "st"(f - 0.5) );
            return i;
            #else
            return (int) f;
            #endif // ARCH_X86
        }
};

#endif // __VOICE_H__
1	/***************************************************************************
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	#include "diskthread.h"
28	#include "ringbuffer.h"
29	#include "stream.h"
30	#include "gig.h"
31	#include "eg_vca.h"
32	#include "rtelmemorypool.h"
33	#include "audiothread.h"
34	#include "filter.h"
35	#include "lfo.h"
36
37	#define MAX_PITCH 4 //FIXME: at the moment in octaves, should be changed into semitones
38	#define USE_LINEAR_INTERPOLATION 1 ///< set to 0 if you prefer cubic interpolation (slower, better quality)
39	#define ENABLE_FILTER 0 ///< if set to 0 then filter (VCF) code is ignored on compile time
40	#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)
41	#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
42
43	// Uncomment following line to override external cutoff controller
44	//#define OVERRIDE_FILTER_CUTOFF_CTRL 1 ///< set to an arbitrary MIDI control change controller (e.g. 1 for 'modulation wheel')
45
46	// Uncomment following line to override external resonance controller
47	//#define OVERRIDE_FILTER_RES_CTRL 91 ///< set to an arbitrary MIDI control change controller (e.g. 91 for 'effect 1 depth')
48
49	// Uncomment following line to override filter type
50	//#define OVERRIDE_FILTER_TYPE gig::vcf_type_lowpass ///< either gig::vcf_type_lowpass, gig::vcf_type_bandpass or gig::vcf_type_highpass
51
52
53	/// Reflects a MIDI controller
54	struct midi_ctrl {
55	uint8_t controller; ///< MIDI control change controller number
56	uint8_t value; ///< Current MIDI controller value
57	float fvalue; ///< Transformed / effective value (e.g. volume level or filter cutoff frequency)
58	};
59
60	class Voice {
61	public:
62	// Attributes
63	int MIDIKey; ///< MIDI key number of the key that triggered the voice
64	uint ReleaseVelocity; ///< Reflects the release velocity value if a note-off command arrived for the voice.
65
66	// Static Attributes
67	static DiskThread* pDiskThread; ///< Pointer to the disk thread, to be able to order a disk stream and later to delete the stream again
68	static AudioThread* pEngine; ///< Pointer to the engine, to be able to access the event lists.
69
70	// Methods
71	Voice();
72	~Voice();
73	void Kill();
74	void Render(uint Samples);
75	void Reset();
76	int Trigger(ModulationSystem::Event* pNoteOnEvent, int Pitch, gig::Instrument* pInstrument);
77	inline bool IsActive() { return Active; }
78	inline void SetOutputLeft(float* pOutput, uint MaxSamplesPerCycle) { this->pOutputLeft = pOutput; this->MaxSamplesPerCycle = MaxSamplesPerCycle; }
79	inline void SetOutputRight(float* pOutput, uint MaxSamplesPerCycle) { this->pOutputRight = pOutput; this->MaxSamplesPerCycle = MaxSamplesPerCycle; }
80	private:
81	// Types
82	enum playback_state_t {
83	playback_state_ram,
84	playback_state_disk,
85	playback_state_end
86	};
87
88	// Attributes
89	float Volume; ///< Volume level of the voice
90	float* pOutputLeft; ///< Audio output buffer (left channel)
91	float* pOutputRight; ///< Audio output buffer (right channel)
92	uint MaxSamplesPerCycle; ///< Size of each audio output buffer
93	double Pos; ///< Current playback position in sample
94	double Pitch; ///< Current pitch depth (number of sample points to move on with each render step)
95	gig::Sample* pSample; ///< Pointer to the sample to be played back
96	gig::Region* pRegion; ///< Pointer to the articulation information of the respective keyboard region of this voice
97	bool Active; ///< If this voice object is currently in usage
98	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
99	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
100	Stream::reference_t DiskStreamRef; ///< Reference / link to the disk stream
101	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.
102	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
103	int LoopCyclesLeft; ///< In case there is a RAMLoop and it's not an endless loop; reflects number of loop cycles left to be passed
104	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
105	EG_VCA EG1;
106	GigFilter FilterLeft;
107	GigFilter FilterRight;
108	midi_ctrl VCFCutoffCtrl;
109	midi_ctrl VCFResonanceCtrl;
110	LFO* pLFO1;
111	LFO* pLFO2;
112	LFO* pLFO3;
113	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).
114
115	// Methods
116	void ProcessEvents(uint Samples);
117	void Interpolate(uint Samples, sample_t* pSrc, uint Skip);
118	void InterpolateAndLoop(uint Samples, sample_t* pSrc, uint Skip);
119	inline void InterpolateOneStep_Stereo(sample_t* pSrc, int& i, float& effective_volume, float& pitch, float& cutoff, float& resonance) {
120	int pos_int = double_to_int(this->Pos); // integer position
121	float pos_fract = this->Pos - pos_int; // fractional part of position
122	pos_int <<= 1;
123
124	#if ENABLE_FILTER
125	UpdateFilter_Stereo(cutoff + 20.0f, resonance); // 20Hz min.
126	#endif // ENABLE_FILTER
127
128	#if USE_LINEAR_INTERPOLATION
129	#if ENABLE_FILTER
130	// left channel
131	this->pOutputLeft[i] += this->FilterLeft.Apply(effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int])));
132	// right channel
133	this->pOutputRight[i++] += this->FilterRight.Apply(effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1])));
134	#else // no filter
135	// left channel
136	this->pOutputLeft[i] += effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int]));
137	// right channel
138	this->pOutputRight[i++] += effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1]));
139	#endif // ENABLE_FILTER
140	#else // polynomial interpolation
141	// calculate left channel
142	float xm1 = pSrc[pos_int];
143	float x0 = pSrc[pos_int+2];
144	float x1 = pSrc[pos_int+4];
145	float x2 = pSrc[pos_int+6];
146	float a = (3 * (x0 - x1) - xm1 + x2) / 2;
147	float b = 2 * x1 + xm1 - (5 * x0 + x2) / 2;
148	float c = (x1 - xm1) / 2;
149	#if ENABLE_FILTER
150	this->pOutputLeft[i] += this->FilterLeft.Apply(effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
151	#else // no filter
152	this->pOutputLeft[i] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
153	#endif // ENABLE_FILTER
154
155	//calculate right channel
156	xm1 = pSrc[pos_int+1];
157	x0 = pSrc[pos_int+3];
158	x1 = pSrc[pos_int+5];
159	x2 = pSrc[pos_int+7];
160	a = (3 * (x0 - x1) - xm1 + x2) / 2;
161	b = 2 * x1 + xm1 - (5 * x0 + x2) / 2;
162	c = (x1 - xm1) / 2;
163	#if ENABLE_FILTER
164	this->pOutputRight[i++] += this->FilterRight.Apply(effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
165	#else // no filter
166	this->pOutputRight[i++] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
167	#endif // ENABLE_FILTER
168	#endif // USE_LINEAR_INTERPOLATION
169
170	this->Pos += pitch;
171	}
172	inline void InterpolateOneStep_Mono(sample_t* pSrc, int& i, float& effective_volume, float& pitch, float& cutoff, float& resonance) {
173	int pos_int = double_to_int(this->Pos); // integer position
174	float pos_fract = this->Pos - pos_int; // fractional part of position
175
176	#if ENABLE_FILTER
177	UpdateFilter_Mono(cutoff + 20.0f, resonance); // 20Hz min.
178	#endif // ENABLE_FILTER
179
180	#if USE_LINEAR_INTERPOLATION
181	float sample_point = effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+1] - pSrc[pos_int]));
182	#else // polynomial interpolation
183	float xm1 = pSrc[pos_int];
184	float x0 = pSrc[pos_int+1];
185	float x1 = pSrc[pos_int+2];
186	float x2 = pSrc[pos_int+3];
187	float a = (3 * (x0 - x1) - xm1 + x2) / 2;
188	float b = 2 * x1 + xm1 - (5 * x0 + x2) / 2;
189	float c = (x1 - xm1) / 2;
190	float sample_point = effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
191	#endif // USE_LINEAR_INTERPOLATION
192
193	#if ENABLE_FILTER
194	sample_point = this->FilterLeft.Apply(sample_point);
195	#endif // ENABLE_FILTER
196
197	this->pOutputLeft[i] += sample_point;
198	this->pOutputRight[i++] += sample_point;
199
200	this->Pos += pitch;
201	}
202	inline void UpdateFilter_Stereo(float cutoff, float& resonance) {
203	static int updatecounter = 0; // we update the filter all FILTER_UPDATE_PERIOD samples
204	if (!(++updatecounter % FILTER_UPDATE_PERIOD) && (cutoff != FilterLeft.Cutoff() \|\| resonance != FilterLeft.Resonance())) {
205	FilterLeft.SetParameters(cutoff, resonance, ModulationSystem::SampleRate());
206	FilterRight.SetParameters(cutoff, resonance, ModulationSystem::SampleRate());
207	}
208	}
209	inline void UpdateFilter_Mono(float cutoff, float& resonance) {
210	static int updatecounter = 0; // we update the filter all FILTER_UPDATE_PERIOD samples
211	if (!(++updatecounter % FILTER_UPDATE_PERIOD) && (cutoff != FilterLeft.Cutoff() \|\| resonance != FilterLeft.Resonance())) {
212	FilterLeft.SetParameters(cutoff, resonance, ModulationSystem::SampleRate());
213	}
214	}
215	inline void ForceUpdateFilter_Stereo(float cutoff, float& resonance) {
216	if (cutoff != FilterLeft.Cutoff() \|\| resonance != FilterLeft.Resonance()) {
217	FilterLeft.SetParameters(cutoff, resonance, ModulationSystem::SampleRate());
218	FilterRight.SetParameters(cutoff, resonance, ModulationSystem::SampleRate());
219	}
220	}
221	inline void ForceUpdateFilter_Mono(float cutoff, float& resonance) {
222	if (cutoff != FilterLeft.Cutoff() \|\| resonance != FilterLeft.Resonance()) {
223	FilterLeft.SetParameters(cutoff, resonance, ModulationSystem::SampleRate());
224	}
225	}
226	inline float Constrain(float ValueToCheck, float Min, float Max) {
227	if (ValueToCheck > Max) ValueToCheck = Max;
228	else if (ValueToCheck < Min) ValueToCheck = Min;
229	return ValueToCheck;
230	}
231	inline int double_to_int(double f) {
232	#if ARCH_X86
233	int i;
234	__asm__ ("fistl %0" : "=m"(i) : "st"(f - 0.5) );
235	return i;
236	#else
237	return (int) f;
238	#endif // ARCH_X86
239	}
240	};
241
242	#endif // __VOICE_H__