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"

#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)

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;
        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) {
            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 USE_LINEAR_INTERPOLATION
                // 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]));
            #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;
                this->pOutputLeft[i] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);

                //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;
                this->pOutputRight[i++] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
            #endif // USE_LINEAR_INTERPOLATION

            this->Pos += pitch;
        }
        inline void InterpolateOneStep_Mono(sample_t* pSrc, int& i, float& effective_volume, float& pitch) {
            int   pos_int   = double_to_int(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 * (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

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

            this->Pos += pitch;
        }
        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
35	#define MAX_PITCH 4 //FIXME: at the moment in octaves, should be changed into semitones
36	#define USE_LINEAR_INTERPOLATION 1 ///< set to 0 if you prefer cubic interpolation (slower, better quality)
37
38	class Voice {
39	public:
40	// Attributes
41	int MIDIKey; ///< MIDI key number of the key that triggered the voice
42	uint ReleaseVelocity; ///< Reflects the release velocity value if a note-off command arrived for the voice.
43
44	// Static Attributes
45	static DiskThread* pDiskThread; ///< Pointer to the disk thread, to be able to order a disk stream and later to delete the stream again
46	static AudioThread* pEngine; ///< Pointer to the engine, to be able to access the event lists.
47
48	// Methods
49	Voice();
50	~Voice();
51	void Kill();
52	void Render(uint Samples);
53	void Reset();
54	int Trigger(ModulationSystem::Event* pNoteOnEvent, int Pitch, gig::Instrument* pInstrument);
55	inline bool IsActive() { return Active; }
56	inline void SetOutputLeft(float* pOutput, uint MaxSamplesPerCycle) { this->pOutputLeft = pOutput; this->MaxSamplesPerCycle = MaxSamplesPerCycle; }
57	inline void SetOutputRight(float* pOutput, uint MaxSamplesPerCycle) { this->pOutputRight = pOutput; this->MaxSamplesPerCycle = MaxSamplesPerCycle; }
58	private:
59	// Types
60	enum playback_state_t {
61	playback_state_ram,
62	playback_state_disk,
63	playback_state_end
64	};
65
66	// Attributes
67	float Volume; ///< Volume level of the voice
68	float* pOutputLeft; ///< Audio output buffer (left channel)
69	float* pOutputRight; ///< Audio output buffer (right channel)
70	uint MaxSamplesPerCycle; ///< Size of each audio output buffer
71	double Pos; ///< Current playback position in sample
72	double Pitch; ///< Current pitch depth (number of sample points to move on with each render step)
73	gig::Sample* pSample; ///< Pointer to the sample to be played back
74	gig::Region* pRegion; ///< Pointer to the articulation information of the respective keyboard region of this voice
75	bool Active; ///< If this voice object is currently in usage
76	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
77	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
78	Stream::reference_t DiskStreamRef; ///< Reference / link to the disk stream
79	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.
80	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
81	int LoopCyclesLeft; ///< In case there is a RAMLoop and it's not an endless loop; reflects number of loop cycles left to be passed
82	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
83	EG_VCA EG1;
84	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).
85
86	// Methods
87	void ProcessEvents(uint Samples);
88	void Interpolate(uint Samples, sample_t* pSrc, uint Skip);
89	void InterpolateAndLoop(uint Samples, sample_t* pSrc, uint Skip);
90	inline void InterpolateOneStep_Stereo(sample_t* pSrc, int& i, float& effective_volume, float& pitch) {
91	int pos_int = double_to_int(this->Pos); // integer position
92	float pos_fract = this->Pos - pos_int; // fractional part of position
93	pos_int <<= 1;
94
95	#if USE_LINEAR_INTERPOLATION
96	// left channel
97	this->pOutputLeft[i] += effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int]));
98	// right channel
99	this->pOutputRight[i++] += effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1]));
100	#else // polynomial interpolation
101	// calculate left channel
102	float xm1 = pSrc[pos_int];
103	float x0 = pSrc[pos_int+2];
104	float x1 = pSrc[pos_int+4];
105	float x2 = pSrc[pos_int+6];
106	float a = (3 * (x0 - x1) - xm1 + x2) / 2;
107	float b = 2 * x1 + xm1 - (5 * x0 + x2) / 2;
108	float c = (x1 - xm1) / 2;
109	this->pOutputLeft[i] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
110
111	//calculate right channel
112	xm1 = pSrc[pos_int+1];
113	x0 = pSrc[pos_int+3];
114	x1 = pSrc[pos_int+5];
115	x2 = pSrc[pos_int+7];
116	a = (3 * (x0 - x1) - xm1 + x2) / 2;
117	b = 2 * x1 + xm1 - (5 * x0 + x2) / 2;
118	c = (x1 - xm1) / 2;
119	this->pOutputRight[i++] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
120	#endif // USE_LINEAR_INTERPOLATION
121
122	this->Pos += pitch;
123	}
124	inline void InterpolateOneStep_Mono(sample_t* pSrc, int& i, float& effective_volume, float& pitch) {
125	int pos_int = double_to_int(this->Pos); // integer position
126	float pos_fract = this->Pos - pos_int; // fractional part of position
127
128	#if USE_LINEAR_INTERPOLATION
129	float sample_point = effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+1] - pSrc[pos_int]));
130	#else // polynomial interpolation
131	float xm1 = pSrc[pos_int];
132	float x0 = pSrc[pos_int+1];
133	float x1 = pSrc[pos_int+2];
134	float x2 = pSrc[pos_int+3];
135	float a = (3 * (x0 - x1) - xm1 + x2) / 2;
136	float b = 2 * x1 + xm1 - (5 * x0 + x2) / 2;
137	float c = (x1 - xm1) / 2;
138	float sample_point = effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
139	#endif // USE_LINEAR_INTERPOLATION
140
141	this->pOutputLeft[i] += sample_point;
142	this->pOutputRight[i++] += sample_point;
143
144	this->Pos += pitch;
145	}
146	inline int double_to_int(double f) {
147	#if ARCH_X86
148	int i;
149	__asm__ ("fistl %0" : "=m"(i) : "st"(f - 0.5) );
150	return i;
151	#else
152	return (int) f;
153	#endif // ARCH_X86
154	}
155	};
156
157	#endif // __VOICE_H__