trunk/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"

#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 controller value
};

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