engines/gig/Synthesizer.h

/***************************************************************************
 *                                                                         *
 *   LinuxSampler - modular, streaming capable sampler                     *
 *                                                                         *
 *   Copyright (C) 2003, 2004 by Benno Senoner and Christian Schoenebeck   *
 *   Copyright (C) 2005 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_SYNTHESIZER_H__
#define __LS_GIG_SYNTHESIZER_H__

#include "../../common/global.h"
#include "../../common/RTMath.h"
#include "../common/Resampler.h"
#include "Filter.h"
#include "SynthesisParam.h"

#define SYNTHESIS_MODE_SET_INTERPOLATE(iMode,bVal)      if (bVal) iMode |= 0x01; else iMode &= ~0x01   /* (un)set mode bit 0 */
#define SYNTHESIS_MODE_SET_FILTER(iMode,bVal)           if (bVal) iMode |= 0x02; else iMode &= ~0x02   /* (un)set mode bit 1 */
#define SYNTHESIS_MODE_SET_LOOP(iMode,bVal)             if (bVal) iMode |= 0x04; else iMode &= ~0x04   /* (un)set mode bit 2 */
#define SYNTHESIS_MODE_SET_CHANNELS(iMode,bVal)         if (bVal) iMode |= 0x08; else iMode &= ~0x08   /* (un)set mode bit 3 */
#define SYNTHESIS_MODE_SET_BITDEPTH24(iMode,bVal)       if (bVal) iMode |= 0x10; else iMode &= ~0x10   /* (un)set mode bit 4 */
#define SYNTHESIS_MODE_SET_IMPLEMENTATION(iMode,bVal)   if (bVal) iMode |= 0x20; else iMode &= ~0x20   /* (un)set mode bit 5 */
#define SYNTHESIS_MODE_SET_PROFILING(iMode,bVal)        if (bVal) iMode |= 0x40; else iMode &= ~0x40   /* (un)set mode bit 6 */

#define SYNTHESIS_MODE_GET_INTERPOLATE(iMode)           iMode & 0x01
#define SYNTHESIS_MODE_GET_FILTER(iMode)                iMode & 0x02
#define SYNTHESIS_MODE_GET_LOOP(iMode)                  iMode & 0x04
#define SYNTHESIS_MODE_GET_CHANNELS(iMode)              iMode & 0x08
#define SYNTHESIS_MODE_GET_BITDEPTH24(iMode)            iMode & 0x10
#define SYNTHESIS_MODE_GET_IMPLEMENTATION(iMode)        iMode & 0x20


namespace LinuxSampler { namespace gig {

    typedef void SynthesizeFragment_Fn(SynthesisParam* pFinalParam, Loop* pLoop);

    void* GetSynthesisFunction(const int SynthesisMode);
    void RunSynthesisFunction(const int SynthesisMode, SynthesisParam* pFinalParam, Loop* pLoop);

    enum channels_t {
        MONO,
        STEREO
    };

    /** @brief Main Synthesis algorithms for the gig::Engine
     *
     * Implementation of the main synthesis algorithms of the Gigasampler
     * format capable sampler engine. This means resampling / interpolation
     * for pitching the audio signal, looping, filter and amplification.
     */
    template<channels_t CHANNELS, bool DOLOOP, bool USEFILTER, bool INTERPOLATE, bool BITDEPTH24>
    class Synthesizer : public __RTMath<CPP>, public LinuxSampler::Resampler<INTERPOLATE,BITDEPTH24> {

            // declarations of derived functions (see "Name lookup,
            // templates, and accessing members of base classes" in
            // the gcc manual for an explanation of why this is
            // needed).
            //using LinuxSampler::Resampler<INTERPOLATE>::GetNextSampleMonoCPP;
            //using LinuxSampler::Resampler<INTERPOLATE>::GetNextSampleStereoCPP;
            using LinuxSampler::Resampler<INTERPOLATE,BITDEPTH24>::Interpolate1StepMonoCPP;
            using LinuxSampler::Resampler<INTERPOLATE,BITDEPTH24>::Interpolate1StepStereoCPP;

        public:
        //protected:

            static void SynthesizeSubFragment(SynthesisParam* pFinalParam, Loop* pLoop) {
                if (DOLOOP) {
                    const float fLoopEnd   = Float(pLoop->uiEnd);
                    const float fLoopStart = Float(pLoop->uiStart);
                    const float fLoopSize  = Float(pLoop->uiSize);
                    if (pLoop->uiTotalCycles) {
                        // render loop (loop count limited)
                        for (; pFinalParam->uiToGo > 0 && pLoop->uiCyclesLeft; pLoop->uiCyclesLeft -= WrapLoop(fLoopStart, fLoopSize, fLoopEnd, &pFinalParam->dPos)) {
                            const uint uiToGo = Min(pFinalParam->uiToGo, DiffToLoopEnd(fLoopEnd, &pFinalParam->dPos, pFinalParam->fFinalPitch) + 1); //TODO: instead of +1 we could also round up
                            SynthesizeSubSubFragment(pFinalParam, uiToGo);
                        }
                        // render on without loop
                        SynthesizeSubSubFragment(pFinalParam, pFinalParam->uiToGo);
                    } else { // render loop (endless loop)
                        for (; pFinalParam->uiToGo > 0; WrapLoop(fLoopStart, fLoopSize, fLoopEnd, &pFinalParam->dPos)) {
                            const uint uiToGo = Min(pFinalParam->uiToGo, DiffToLoopEnd(fLoopEnd, &pFinalParam->dPos, pFinalParam->fFinalPitch) + 1); //TODO: instead of +1 we could also round up
                            SynthesizeSubSubFragment(pFinalParam, uiToGo);
                        }
                    }
                } else { // no looping
                    SynthesizeSubSubFragment(pFinalParam, pFinalParam->uiToGo);
                }
            }

            /**
             * Returns the difference to the sample's loop end.
             */
            inline static int DiffToLoopEnd(const float& LoopEnd, const void* Pos, const float& Pitch) {
                return uint((LoopEnd - *((double *)Pos)) / Pitch);
            }

#if 0
            //TODO: this method is not in use yet, it's intended to be used for pitch=x.0f where we could use integer instead of float as playback position variable
            inline static int WrapLoop(const int& LoopStart, const int& LoopSize, const int& LoopEnd, int& Pos) {
                //TODO: we can easily eliminate the branch here
                if (Pos < LoopEnd) return 0;
                Pos = (Pos - LoopEnd) % LoopSize + LoopStart;
                return 1;
            }
#endif

            /**
             * This method handles looping of the RAM playback part of the
             * sample, thus repositioning the playback position once the
             * loop limit was reached. Note: looping of the disk streaming
             * part is handled by libgig (ReadAndLoop() method which will
             * be called by the DiskThread).
             */
            inline static int WrapLoop(const float& LoopStart, const float& LoopSize, const float& LoopEnd, void* vPos) {
                double * Pos = (double *)vPos;
                if (*Pos < LoopEnd) return 0;
                *Pos = fmod(*Pos - LoopEnd, LoopSize) + LoopStart;
                return 1;
            }

            static int getSample(sample_t* src, int pos) {
                if (BITDEPTH24) {
                    pos *= 3;
                    unsigned char* p = (unsigned char*)src;
                    return p[pos] << 8 | p[pos + 1] << 16 | p[pos + 2] << 24;
                } else {
                    return src[pos];
                }
            }

            static void SynthesizeSubSubFragment(SynthesisParam* pFinalParam, uint uiToGo) {
                float fVolumeL = pFinalParam->fFinalVolumeLeft;
                float fVolumeR = pFinalParam->fFinalVolumeRight;
                sample_t* pSrc = pFinalParam->pSrc;
                float* pOutL   = pFinalParam->pOutLeft;
                float* pOutR   = pFinalParam->pOutRight;
#ifdef CONFIG_INTERPOLATE_VOLUME
                float fDeltaL  = pFinalParam->fFinalVolumeDeltaLeft;
                float fDeltaR  = pFinalParam->fFinalVolumeDeltaRight;
#endif
                switch (CHANNELS) {
                    case MONO: {
                        float samplePoint;
                        if (INTERPOLATE) {
                            double dPos    = pFinalParam->dPos;
                            float fPitch   = pFinalParam->fFinalPitch;
                            if (USEFILTER) {
                                Filter filterL = pFinalParam->filterLeft;
                                for (int i = 0; i < uiToGo; ++i) {
                                    samplePoint = Interpolate1StepMonoCPP(pSrc, &dPos, fPitch);
                                    samplePoint = filterL.Apply(samplePoint);
#ifdef CONFIG_INTERPOLATE_VOLUME
                                    fVolumeL += fDeltaL;
                                    fVolumeR += fDeltaR;
#endif
                                    pOutL[i] += samplePoint * fVolumeL;
                                    pOutR[i] += samplePoint * fVolumeR;
                                }
                            } else { // no filter needed
                                for (int i = 0; i < uiToGo; ++i) {
                                    samplePoint = Interpolate1StepMonoCPP(pSrc, &dPos, fPitch);
#ifdef CONFIG_INTERPOLATE_VOLUME
                                    fVolumeL += fDeltaL;
                                    fVolumeR += fDeltaR;
#endif
                                    pOutL[i] += samplePoint * fVolumeL;
                                    pOutR[i] += samplePoint * fVolumeR;
                                }
                            }
                            pFinalParam->dPos = dPos;
                        } else { // no interpolation
                            int pos_offset = (int) pFinalParam->dPos;
                            if (USEFILTER) {
                                Filter filterL = pFinalParam->filterLeft;
                                for (int i = 0; i < uiToGo; ++i) {
                                    samplePoint = getSample(pSrc, i + pos_offset);
                                    samplePoint = filterL.Apply(samplePoint);
#ifdef CONFIG_INTERPOLATE_VOLUME
                                    fVolumeL += fDeltaL;
                                    fVolumeR += fDeltaR;
#endif
                                    pOutL[i] += samplePoint * fVolumeL;
                                    pOutR[i] += samplePoint * fVolumeR;
                                }
                            } else { // no filter needed
                                for (int i = 0; i < uiToGo; ++i) {
                                    samplePoint = getSample(pSrc, i + pos_offset);
#ifdef CONFIG_INTERPOLATE_VOLUME
                                    fVolumeL += fDeltaL;
                                    fVolumeR += fDeltaR;
#endif
                                    pOutL[i] += samplePoint * fVolumeL;
                                    pOutR[i] += samplePoint * fVolumeR;
                                }
                            }
                            pFinalParam->dPos += uiToGo;
                        }
                        break;
                    }
                    case STEREO: {
                        stereo_sample_t samplePoint;
                        if (INTERPOLATE) {
                            double dPos    = pFinalParam->dPos;
                            float fPitch   = pFinalParam->fFinalPitch;
                            if (USEFILTER) {
                                Filter filterL = pFinalParam->filterLeft;
                                Filter filterR = pFinalParam->filterRight;
                                for (int i = 0; i < uiToGo; ++i) {
                                    samplePoint = Interpolate1StepStereoCPP(pSrc, &dPos, fPitch);
                                    samplePoint.left  = filterL.Apply(samplePoint.left);
                                    samplePoint.right = filterR.Apply(samplePoint.right);
#ifdef CONFIG_INTERPOLATE_VOLUME
                                    fVolumeL += fDeltaL;
                                    fVolumeR += fDeltaR;
#endif
                                    pOutL[i] += samplePoint.left  * fVolumeL;
                                    pOutR[i] += samplePoint.right * fVolumeR;
                                }
                            } else { // no filter needed
                                for (int i = 0; i < uiToGo; ++i) {
                                    samplePoint = Interpolate1StepStereoCPP(pSrc, &dPos, fPitch);
#ifdef CONFIG_INTERPOLATE_VOLUME
                                    fVolumeL += fDeltaL;
                                    fVolumeR += fDeltaR;
#endif
                                    pOutL[i] += samplePoint.left  * fVolumeL;
                                    pOutR[i] += samplePoint.right * fVolumeR;
                                }
                            }
                            pFinalParam->dPos = dPos;
                        } else { // no interpolation
                            int pos_offset = ((int) pFinalParam->dPos) << 1;
                            if (USEFILTER) {
                                Filter filterL = pFinalParam->filterLeft;
                                Filter filterR = pFinalParam->filterRight;
                                for (int i = 0, ii = 0; i < uiToGo; ++i, ii+=2) {
                                    samplePoint.left = getSample(pSrc, ii + pos_offset);
                                    samplePoint.right = getSample(pSrc, ii + pos_offset + 1);
                                    samplePoint.left  = filterL.Apply(samplePoint.left);
                                    samplePoint.right = filterR.Apply(samplePoint.right);
#ifdef CONFIG_INTERPOLATE_VOLUME
                                    fVolumeL += fDeltaL;
                                    fVolumeR += fDeltaR;
#endif
                                    pOutL[i] += samplePoint.left  * fVolumeL;
                                    pOutR[i] += samplePoint.right * fVolumeR;
                                }
                            } else { // no filter needed
                                for (int i = 0, ii = 0; i < uiToGo; ++i, ii+=2) {
                                    samplePoint.left = getSample(pSrc, ii + pos_offset);
                                    samplePoint.right = getSample(pSrc, ii + pos_offset + 1);
#ifdef CONFIG_INTERPOLATE_VOLUME
                                    fVolumeL += fDeltaL;
                                    fVolumeR += fDeltaR;
#endif
                                    pOutL[i] += samplePoint.left  * fVolumeL;
                                    pOutR[i] += samplePoint.right * fVolumeR;
                                }
                            }
                            pFinalParam->dPos += uiToGo;
                        }
                        break;
                    }
                }
                pFinalParam->fFinalVolumeLeft = fVolumeL;
                pFinalParam->fFinalVolumeRight = fVolumeR;
                pFinalParam->pOutRight += uiToGo;
                pFinalParam->pOutLeft  += uiToGo;
                pFinalParam->uiToGo    -= uiToGo;
            }
    };

}} // namespace LinuxSampler::gig

#endif // __LS_GIG_SYNTHESIZER_H__
1	/***************************************************************************
2	* *
3	* LinuxSampler - modular, streaming capable sampler *
4	* *
5	* Copyright (C) 2003, 2004 by Benno Senoner and Christian Schoenebeck *
6	* Copyright (C) 2005 Christian Schoenebeck *
7	* *
8	* This program is free software; you can redistribute it and/or modify *
9	* it under the terms of the GNU General Public License as published by *
10	* the Free Software Foundation; either version 2 of the License, or *
11	* (at your option) any later version. *
12	* *
13	* This program is distributed in the hope that it will be useful, *
14	* but WITHOUT ANY WARRANTY; without even the implied warranty of *
15	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the *
16	* GNU General Public License for more details. *
17	* *
18	* You should have received a copy of the GNU General Public License *
19	* along with this program; if not, write to the Free Software *
20	* Foundation, Inc., 59 Temple Place, Suite 330, Boston, *
21	* MA 02111-1307 USA *
22	***************************************************************************/
23
24	#ifndef __LS_GIG_SYNTHESIZER_H__
25	#define __LS_GIG_SYNTHESIZER_H__
26
27	#include "../../common/global.h"
28	#include "../../common/RTMath.h"
29	#include "../common/Resampler.h"
30	#include "Filter.h"
31	#include "SynthesisParam.h"
32
33	#define SYNTHESIS_MODE_SET_INTERPOLATE(iMode,bVal) if (bVal) iMode \|= 0x01; else iMode &= ~0x01 /* (un)set mode bit 0 */
34	#define SYNTHESIS_MODE_SET_FILTER(iMode,bVal) if (bVal) iMode \|= 0x02; else iMode &= ~0x02 /* (un)set mode bit 1 */
35	#define SYNTHESIS_MODE_SET_LOOP(iMode,bVal) if (bVal) iMode \|= 0x04; else iMode &= ~0x04 /* (un)set mode bit 2 */
36	#define SYNTHESIS_MODE_SET_CHANNELS(iMode,bVal) if (bVal) iMode \|= 0x08; else iMode &= ~0x08 /* (un)set mode bit 3 */
37	#define SYNTHESIS_MODE_SET_BITDEPTH24(iMode,bVal) if (bVal) iMode \|= 0x10; else iMode &= ~0x10 /* (un)set mode bit 4 */
38	#define SYNTHESIS_MODE_SET_IMPLEMENTATION(iMode,bVal) if (bVal) iMode \|= 0x20; else iMode &= ~0x20 /* (un)set mode bit 5 */
39	#define SYNTHESIS_MODE_SET_PROFILING(iMode,bVal) if (bVal) iMode \|= 0x40; else iMode &= ~0x40 /* (un)set mode bit 6 */
40
41	#define SYNTHESIS_MODE_GET_INTERPOLATE(iMode) iMode & 0x01
42	#define SYNTHESIS_MODE_GET_FILTER(iMode) iMode & 0x02
43	#define SYNTHESIS_MODE_GET_LOOP(iMode) iMode & 0x04
44	#define SYNTHESIS_MODE_GET_CHANNELS(iMode) iMode & 0x08
45	#define SYNTHESIS_MODE_GET_BITDEPTH24(iMode) iMode & 0x10
46	#define SYNTHESIS_MODE_GET_IMPLEMENTATION(iMode) iMode & 0x20
47
48
49	namespace LinuxSampler { namespace gig {
50
51	typedef void SynthesizeFragment_Fn(SynthesisParam* pFinalParam, Loop* pLoop);
52
53	void* GetSynthesisFunction(const int SynthesisMode);
54	void RunSynthesisFunction(const int SynthesisMode, SynthesisParam* pFinalParam, Loop* pLoop);
55
56	enum channels_t {
57	MONO,
58	STEREO
59	};
60
61	/** @brief Main Synthesis algorithms for the gig::Engine
62	*
63	* Implementation of the main synthesis algorithms of the Gigasampler
64	* format capable sampler engine. This means resampling / interpolation
65	* for pitching the audio signal, looping, filter and amplification.
66	*/
67	template<channels_t CHANNELS, bool DOLOOP, bool USEFILTER, bool INTERPOLATE, bool BITDEPTH24>
68	class Synthesizer : public __RTMath<CPP>, public LinuxSampler::Resampler<INTERPOLATE,BITDEPTH24> {
69
70	// declarations of derived functions (see "Name lookup,
71	// templates, and accessing members of base classes" in
72	// the gcc manual for an explanation of why this is
73	// needed).
74	//using LinuxSampler::Resampler<INTERPOLATE>::GetNextSampleMonoCPP;
75	//using LinuxSampler::Resampler<INTERPOLATE>::GetNextSampleStereoCPP;
76	using LinuxSampler::Resampler<INTERPOLATE,BITDEPTH24>::Interpolate1StepMonoCPP;
77	using LinuxSampler::Resampler<INTERPOLATE,BITDEPTH24>::Interpolate1StepStereoCPP;
78
79	public:
80	//protected:
81
82	static void SynthesizeSubFragment(SynthesisParam* pFinalParam, Loop* pLoop) {
83	if (DOLOOP) {
84	const float fLoopEnd = Float(pLoop->uiEnd);
85	const float fLoopStart = Float(pLoop->uiStart);
86	const float fLoopSize = Float(pLoop->uiSize);
87	if (pLoop->uiTotalCycles) {
88	// render loop (loop count limited)
89	for (; pFinalParam->uiToGo > 0 && pLoop->uiCyclesLeft; pLoop->uiCyclesLeft -= WrapLoop(fLoopStart, fLoopSize, fLoopEnd, &pFinalParam->dPos)) {
90	const uint uiToGo = Min(pFinalParam->uiToGo, DiffToLoopEnd(fLoopEnd, &pFinalParam->dPos, pFinalParam->fFinalPitch) + 1); //TODO: instead of +1 we could also round up
91	SynthesizeSubSubFragment(pFinalParam, uiToGo);
92	}
93	// render on without loop
94	SynthesizeSubSubFragment(pFinalParam, pFinalParam->uiToGo);
95	} else { // render loop (endless loop)
96	for (; pFinalParam->uiToGo > 0; WrapLoop(fLoopStart, fLoopSize, fLoopEnd, &pFinalParam->dPos)) {
97	const uint uiToGo = Min(pFinalParam->uiToGo, DiffToLoopEnd(fLoopEnd, &pFinalParam->dPos, pFinalParam->fFinalPitch) + 1); //TODO: instead of +1 we could also round up
98	SynthesizeSubSubFragment(pFinalParam, uiToGo);
99	}
100	}
101	} else { // no looping
102	SynthesizeSubSubFragment(pFinalParam, pFinalParam->uiToGo);
103	}
104	}
105
106	/**
107	* Returns the difference to the sample's loop end.
108	*/
109	inline static int DiffToLoopEnd(const float& LoopEnd, const void* Pos, const float& Pitch) {
110	return uint((LoopEnd - ((double )Pos)) / Pitch);
111	}
112
113	#if 0
114	//TODO: this method is not in use yet, it's intended to be used for pitch=x.0f where we could use integer instead of float as playback position variable
115	inline static int WrapLoop(const int& LoopStart, const int& LoopSize, const int& LoopEnd, int& Pos) {
116	//TODO: we can easily eliminate the branch here
117	if (Pos < LoopEnd) return 0;
118	Pos = (Pos - LoopEnd) % LoopSize + LoopStart;
119	return 1;
120	}
121	#endif
122
123	/**
124	* This method handles looping of the RAM playback part of the
125	* sample, thus repositioning the playback position once the
126	* loop limit was reached. Note: looping of the disk streaming
127	* part is handled by libgig (ReadAndLoop() method which will
128	* be called by the DiskThread).
129	*/
130	inline static int WrapLoop(const float& LoopStart, const float& LoopSize, const float& LoopEnd, void* vPos) {
131	double * Pos = (double *)vPos;
132	if (*Pos < LoopEnd) return 0;
133	Pos = fmod(Pos - LoopEnd, LoopSize) + LoopStart;
134	return 1;
135	}
136
137	static int getSample(sample_t* src, int pos) {
138	if (BITDEPTH24) {
139	pos *= 3;
140	unsigned char* p = (unsigned char*)src;
141	return p[pos] << 8 \| p[pos + 1] << 16 \| p[pos + 2] << 24;
142	} else {
143	return src[pos];
144	}
145	}
146
147	static void SynthesizeSubSubFragment(SynthesisParam* pFinalParam, uint uiToGo) {
148	float fVolumeL = pFinalParam->fFinalVolumeLeft;
149	float fVolumeR = pFinalParam->fFinalVolumeRight;
150	sample_t* pSrc = pFinalParam->pSrc;
151	float* pOutL = pFinalParam->pOutLeft;
152	float* pOutR = pFinalParam->pOutRight;
153	#ifdef CONFIG_INTERPOLATE_VOLUME
154	float fDeltaL = pFinalParam->fFinalVolumeDeltaLeft;
155	float fDeltaR = pFinalParam->fFinalVolumeDeltaRight;
156	#endif
157	switch (CHANNELS) {
158	case MONO: {
159	float samplePoint;
160	if (INTERPOLATE) {
161	double dPos = pFinalParam->dPos;
162	float fPitch = pFinalParam->fFinalPitch;
163	if (USEFILTER) {
164	Filter filterL = pFinalParam->filterLeft;
165	for (int i = 0; i < uiToGo; ++i) {
166	samplePoint = Interpolate1StepMonoCPP(pSrc, &dPos, fPitch);
167	samplePoint = filterL.Apply(samplePoint);
168	#ifdef CONFIG_INTERPOLATE_VOLUME
169	fVolumeL += fDeltaL;
170	fVolumeR += fDeltaR;
171	#endif
172	pOutL[i] += samplePoint * fVolumeL;
173	pOutR[i] += samplePoint * fVolumeR;
174	}
175	} else { // no filter needed
176	for (int i = 0; i < uiToGo; ++i) {
177	samplePoint = Interpolate1StepMonoCPP(pSrc, &dPos, fPitch);
178	#ifdef CONFIG_INTERPOLATE_VOLUME
179	fVolumeL += fDeltaL;
180	fVolumeR += fDeltaR;
181	#endif
182	pOutL[i] += samplePoint * fVolumeL;
183	pOutR[i] += samplePoint * fVolumeR;
184	}
185	}
186	pFinalParam->dPos = dPos;
187	} else { // no interpolation
188	int pos_offset = (int) pFinalParam->dPos;
189	if (USEFILTER) {
190	Filter filterL = pFinalParam->filterLeft;
191	for (int i = 0; i < uiToGo; ++i) {
192	samplePoint = getSample(pSrc, i + pos_offset);
193	samplePoint = filterL.Apply(samplePoint);
194	#ifdef CONFIG_INTERPOLATE_VOLUME
195	fVolumeL += fDeltaL;
196	fVolumeR += fDeltaR;
197	#endif
198	pOutL[i] += samplePoint * fVolumeL;
199	pOutR[i] += samplePoint * fVolumeR;
200	}
201	} else { // no filter needed
202	for (int i = 0; i < uiToGo; ++i) {
203	samplePoint = getSample(pSrc, i + pos_offset);
204	#ifdef CONFIG_INTERPOLATE_VOLUME
205	fVolumeL += fDeltaL;
206	fVolumeR += fDeltaR;
207	#endif
208	pOutL[i] += samplePoint * fVolumeL;
209	pOutR[i] += samplePoint * fVolumeR;
210	}
211	}
212	pFinalParam->dPos += uiToGo;
213	}
214	break;
215	}
216	case STEREO: {
217	stereo_sample_t samplePoint;
218	if (INTERPOLATE) {
219	double dPos = pFinalParam->dPos;
220	float fPitch = pFinalParam->fFinalPitch;
221	if (USEFILTER) {
222	Filter filterL = pFinalParam->filterLeft;
223	Filter filterR = pFinalParam->filterRight;
224	for (int i = 0; i < uiToGo; ++i) {
225	samplePoint = Interpolate1StepStereoCPP(pSrc, &dPos, fPitch);
226	samplePoint.left = filterL.Apply(samplePoint.left);
227	samplePoint.right = filterR.Apply(samplePoint.right);
228	#ifdef CONFIG_INTERPOLATE_VOLUME
229	fVolumeL += fDeltaL;
230	fVolumeR += fDeltaR;
231	#endif
232	pOutL[i] += samplePoint.left * fVolumeL;
233	pOutR[i] += samplePoint.right * fVolumeR;
234	}
235	} else { // no filter needed
236	for (int i = 0; i < uiToGo; ++i) {
237	samplePoint = Interpolate1StepStereoCPP(pSrc, &dPos, fPitch);
238	#ifdef CONFIG_INTERPOLATE_VOLUME
239	fVolumeL += fDeltaL;
240	fVolumeR += fDeltaR;
241	#endif
242	pOutL[i] += samplePoint.left * fVolumeL;
243	pOutR[i] += samplePoint.right * fVolumeR;
244	}
245	}
246	pFinalParam->dPos = dPos;
247	} else { // no interpolation
248	int pos_offset = ((int) pFinalParam->dPos) << 1;
249	if (USEFILTER) {
250	Filter filterL = pFinalParam->filterLeft;
251	Filter filterR = pFinalParam->filterRight;
252	for (int i = 0, ii = 0; i < uiToGo; ++i, ii+=2) {
253	samplePoint.left = getSample(pSrc, ii + pos_offset);
254	samplePoint.right = getSample(pSrc, ii + pos_offset + 1);
255	samplePoint.left = filterL.Apply(samplePoint.left);
256	samplePoint.right = filterR.Apply(samplePoint.right);
257	#ifdef CONFIG_INTERPOLATE_VOLUME
258	fVolumeL += fDeltaL;
259	fVolumeR += fDeltaR;
260	#endif
261	pOutL[i] += samplePoint.left * fVolumeL;
262	pOutR[i] += samplePoint.right * fVolumeR;
263	}
264	} else { // no filter needed
265	for (int i = 0, ii = 0; i < uiToGo; ++i, ii+=2) {
266	samplePoint.left = getSample(pSrc, ii + pos_offset);
267	samplePoint.right = getSample(pSrc, ii + pos_offset + 1);
268	#ifdef CONFIG_INTERPOLATE_VOLUME
269	fVolumeL += fDeltaL;
270	fVolumeR += fDeltaR;
271	#endif
272	pOutL[i] += samplePoint.left * fVolumeL;
273	pOutR[i] += samplePoint.right * fVolumeR;
274	}
275	}
276	pFinalParam->dPos += uiToGo;
277	}
278	break;
279	}
280	}
281	pFinalParam->fFinalVolumeLeft = fVolumeL;
282	pFinalParam->fFinalVolumeRight = fVolumeR;
283	pFinalParam->pOutRight += uiToGo;
284	pFinalParam->pOutLeft += uiToGo;
285	pFinalParam->uiToGo -= uiToGo;
286	}
287	};
288
289	}} // namespace LinuxSampler::gig
290
291	#endif // __LS_GIG_SYNTHESIZER_H__