/[svn]/linuxsampler/trunk/src/engines/gig/Voice.h
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Contents of /linuxsampler/trunk/src/engines/gig/Voice.h

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Revision 233 - (show annotations) (download) (as text)
Tue Sep 7 09:32:21 2004 UTC (19 years, 6 months ago) by schoenebeck
File MIME type: text/x-c++hdr
File size: 14580 byte(s)
* added support for layers
* fixed initial pitch calculation which did not honor the sample's own
  sample rate

1 /***************************************************************************
2 * *
3 * LinuxSampler - modular, streaming capable sampler *
4 * *
5 * Copyright (C) 2003, 2004 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 __LS_GIG_VOICE_H__
24 #define __LS_GIG_VOICE_H__
25
26 #include "../../common/global.h"
27
28 #if DEBUG_HEADERS
29 # warning Voice.h included
30 #endif // DEBUG_HEADERS
31
32 #include "../../common/RTMath.h"
33 #include "../../common/RingBuffer.h"
34 #include "../../common/RTELMemoryPool.h"
35 #include "../../drivers/audio/AudioOutputDevice.h"
36 #include "../../lib/fileloader/libgig/gig.h"
37 #include "../common/BiquadFilter.h"
38 #include "Engine.h"
39 #include "Stream.h"
40 #include "DiskThread.h"
41
42 #include "EGDecay.h"
43 #include "Filter.h"
44 #include "../common/LFO.h"
45
46 #define USE_LINEAR_INTERPOLATION 0 ///< set to 0 if you prefer cubic interpolation (slower, better quality)
47 #define ENABLE_FILTER 1 ///< if set to 0 then filter (VCF) code is ignored on compile time
48 #define FILTER_UPDATE_PERIOD 64 ///< amount of sample points after which filter parameters (cutoff, resonance) are going to be updated (higher value means less CPU load, but also worse parameter resolution, this value will be aligned to a power of two)
49 #define FORCE_FILTER_USAGE 0 ///< if set to 1 then filter is always used, if set to 0 filter is used only in case the instrument file defined one
50 #define FILTER_CUTOFF_MAX 10000.0f ///< maximum cutoff frequency (10kHz)
51 #define FILTER_CUTOFF_MIN 100.0f ///< minimum cutoff frequency (100Hz)
52
53 // Uncomment following line to override external cutoff controller
54 //#define OVERRIDE_FILTER_CUTOFF_CTRL 1 ///< set to an arbitrary MIDI control change controller (e.g. 1 for 'modulation wheel')
55
56 // Uncomment following line to override external resonance controller
57 //#define OVERRIDE_FILTER_RES_CTRL 91 ///< set to an arbitrary MIDI control change controller (e.g. 91 for 'effect 1 depth')
58
59 // Uncomment following line to override filter type
60 //#define OVERRIDE_FILTER_TYPE ::gig::vcf_type_lowpass ///< either ::gig::vcf_type_lowpass, ::gig::vcf_type_bandpass or ::gig::vcf_type_highpass
61
62 namespace LinuxSampler { namespace gig {
63
64 class Engine;
65 class EGADSR;
66 class VCAManipulator;
67 class VCFCManipulator;
68 class VCOManipulator;
69
70 /// Reflects a MIDI controller
71 struct midi_ctrl {
72 uint8_t controller; ///< MIDI control change controller number
73 uint8_t value; ///< Current MIDI controller value
74 float fvalue; ///< Transformed / effective value (e.g. volume level or filter cutoff frequency)
75 };
76
77 /** Gig Voice
78 *
79 * Renders a voice for the Gigasampler format.
80 */
81 class Voice {
82 public:
83 // Attributes
84 int MIDIKey; ///< MIDI key number of the key that triggered the voice
85 DiskThread* pDiskThread; ///< Pointer to the disk thread, to be able to order a disk stream and later to delete the stream again
86
87 // Methods
88 Voice();
89 ~Voice();
90 void Kill();
91 void Render(uint Samples);
92 void Reset();
93 void SetOutput(AudioOutputDevice* pAudioOutputDevice);
94 void SetEngine(Engine* pEngine);
95 int Trigger(Event* pNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument, int iLayer = 0);
96 inline bool IsActive() { return Active; }
97 private:
98 // Types
99 enum playback_state_t {
100 playback_state_ram,
101 playback_state_disk,
102 playback_state_end
103 };
104
105 // Attributes
106 gig::Engine* pEngine; ///< Pointer to the sampler engine, to be able to access the event lists.
107 float Volume; ///< Volume level of the voice
108 double Pos; ///< Current playback position in sample
109 double PitchBase; ///< Basic pitch depth, stays the same for the whole life time of the voice
110 double PitchBend; ///< Current pitch value of the pitchbend wheel
111 ::gig::Sample* pSample; ///< Pointer to the sample to be played back
112 ::gig::Region* pRegion; ///< Pointer to the articulation information of the respective keyboard region of this voice
113 bool Active; ///< If this voice object is currently in usage
114 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
115 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
116 Stream::reference_t DiskStreamRef; ///< Reference / link to the disk stream
117 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.
118 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
119 int LoopCyclesLeft; ///< In case there is a RAMLoop and it's not an endless loop; reflects number of loop cycles left to be passed
120 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
121 EGADSR* pEG1; ///< Envelope Generator 1 (Amplification)
122 EGADSR* pEG2; ///< Envelope Generator 2 (Filter cutoff frequency)
123 EGDecay* pEG3; ///< Envelope Generator 3 (Pitch)
124 Filter FilterLeft;
125 Filter FilterRight;
126 midi_ctrl VCFCutoffCtrl;
127 midi_ctrl VCFResonanceCtrl;
128 int FilterUpdateCounter; ///< Used to update filter parameters all FILTER_UPDATE_PERIOD samples
129 static const float FILTER_CUTOFF_COEFF;
130 static const int FILTER_UPDATE_MASK;
131 VCAManipulator* pVCAManipulator;
132 VCFCManipulator* pVCFCManipulator;
133 VCOManipulator* pVCOManipulator;
134 LFO<gig::VCAManipulator>* pLFO1; ///< Low Frequency Oscillator 1 (Amplification)
135 LFO<gig::VCFCManipulator>* pLFO2; ///< Low Frequency Oscillator 2 (Filter cutoff frequency)
136 LFO<gig::VCOManipulator>* pLFO3; ///< Low Frequency Oscillator 3 (Pitch)
137 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).
138
139 // Static Methods
140 static float CalculateFilterCutoffCoeff();
141 static int CalculateFilterUpdateMask();
142
143 // Methods
144 void ProcessEvents(uint Samples);
145 #if ENABLE_FILTER
146 void CalculateBiquadParameters(uint Samples);
147 #endif // ENABLE_FILTER
148 void Interpolate(uint Samples, sample_t* pSrc, uint Skip);
149 void InterpolateAndLoop(uint Samples, sample_t* pSrc, uint Skip);
150 inline void InterpolateOneStep_Stereo(sample_t* pSrc, int& i, float& effective_volume, float& pitch, biquad_param_t& bq_base, biquad_param_t& bq_main) {
151 int pos_int = RTMath::DoubleToInt(this->Pos); // integer position
152 float pos_fract = this->Pos - pos_int; // fractional part of position
153 pos_int <<= 1;
154
155 #if USE_LINEAR_INTERPOLATION
156 #if ENABLE_FILTER
157 // left channel
158 pEngine->pOutputLeft[i] += this->FilterLeft.Apply(&bq_base, &bq_main, effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int])));
159 // right channel
160 pEngine->pOutputRight[i++] += this->FilterRight.Apply(&bq_base, &bq_main, effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1])));
161 #else // no filter
162 // left channel
163 pEngine->pOutputLeft[i] += effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int]));
164 // right channel
165 pEngine->pOutputRight[i++] += effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1]));
166 #endif // ENABLE_FILTER
167 #else // polynomial interpolation
168 // calculate left channel
169 float xm1 = pSrc[pos_int];
170 float x0 = pSrc[pos_int+2];
171 float x1 = pSrc[pos_int+4];
172 float x2 = pSrc[pos_int+6];
173 float a = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f;
174 float b = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f;
175 float c = (x1 - xm1) * 0.5f;
176 #if ENABLE_FILTER
177 pEngine->pOutputLeft[i] += this->FilterLeft.Apply(&bq_base, &bq_main, effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
178 #else // no filter
179 pEngine->pOutputLeft[i] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
180 #endif // ENABLE_FILTER
181
182 //calculate right channel
183 xm1 = pSrc[pos_int+1];
184 x0 = pSrc[pos_int+3];
185 x1 = pSrc[pos_int+5];
186 x2 = pSrc[pos_int+7];
187 a = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f;
188 b = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f;
189 c = (x1 - xm1) * 0.5f;
190 #if ENABLE_FILTER
191 pEngine->pOutputRight[i++] += this->FilterRight.Apply(&bq_base, &bq_main, effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
192 #else // no filter
193 pEngine->pOutputRight[i++] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
194 #endif // ENABLE_FILTER
195 #endif // USE_LINEAR_INTERPOLATION
196
197 this->Pos += pitch;
198 }
199
200 inline void InterpolateOneStep_Mono(sample_t* pSrc, int& i, float& effective_volume, float& pitch, biquad_param_t& bq_base, biquad_param_t& bq_main) {
201 int pos_int = RTMath::DoubleToInt(this->Pos); // integer position
202 float pos_fract = this->Pos - pos_int; // fractional part of position
203
204 #if USE_LINEAR_INTERPOLATION
205 float sample_point = effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+1] - pSrc[pos_int]));
206 #else // polynomial interpolation
207 float xm1 = pSrc[pos_int];
208 float x0 = pSrc[pos_int+1];
209 float x1 = pSrc[pos_int+2];
210 float x2 = pSrc[pos_int+3];
211 float a = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f;
212 float b = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f;
213 float c = (x1 - xm1) * 0.5f;
214 float sample_point = effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
215 #endif // USE_LINEAR_INTERPOLATION
216
217 #if ENABLE_FILTER
218 sample_point = this->FilterLeft.Apply(&bq_base, &bq_main, sample_point);
219 #endif // ENABLE_FILTER
220
221 pEngine->pOutputLeft[i] += sample_point;
222 pEngine->pOutputRight[i++] += sample_point;
223
224 this->Pos += pitch;
225 }
226
227 inline float Constrain(float ValueToCheck, float Min, float Max) {
228 if (ValueToCheck > Max) ValueToCheck = Max;
229 else if (ValueToCheck < Min) ValueToCheck = Min;
230 return ValueToCheck;
231 }
232 };
233
234 }} // namespace LinuxSampler::gig
235
236 #endif // __LS_GIG_VOICE_H__

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