/[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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Tue Apr 27 09:21:58 2004 UTC (19 years, 11 months ago) by schoenebeck
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updated copyright header for 2004

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 "../../audiodriver/AudioOutputDevice.h"
36 #include "../../lib/fileloader/libgig/gig.h"
37 #include "Engine.h"
38 #include "Stream.h"
39 #include "DiskThread.h"
40
41 #include "EGDecay.h"
42 #include "Filter.h"
43 #include "../common/LFO.h"
44
45 #define USE_LINEAR_INTERPOLATION 1 ///< set to 0 if you prefer cubic interpolation (slower, better quality)
46 #define ENABLE_FILTER 0 ///< if set to 0 then filter (VCF) code is ignored on compile time
47 #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)
48 #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
49 #define FILTER_CUTOFF_MAX 10000.0f ///< maximum cutoff frequency (10kHz)
50 #define FILTER_CUTOFF_MIN 100.0f ///< minimum cutoff frequency (100Hz)
51
52 // Uncomment following line to override external cutoff controller
53 //#define OVERRIDE_FILTER_CUTOFF_CTRL 1 ///< set to an arbitrary MIDI control change controller (e.g. 1 for 'modulation wheel')
54
55 // Uncomment following line to override external resonance controller
56 //#define OVERRIDE_FILTER_RES_CTRL 91 ///< set to an arbitrary MIDI control change controller (e.g. 91 for 'effect 1 depth')
57
58 // Uncomment following line to override filter type
59 //#define OVERRIDE_FILTER_TYPE ::gig::vcf_type_lowpass ///< either ::gig::vcf_type_lowpass, ::gig::vcf_type_bandpass or ::gig::vcf_type_highpass
60
61 namespace LinuxSampler { namespace gig {
62
63 class Engine;
64 class EGADSR;
65 class VCAManipulator;
66 class VCFCManipulator;
67 class VCOManipulator;
68
69 /// Reflects a MIDI controller
70 struct midi_ctrl {
71 uint8_t controller; ///< MIDI control change controller number
72 uint8_t value; ///< Current MIDI controller value
73 float fvalue; ///< Transformed / effective value (e.g. volume level or filter cutoff frequency)
74 };
75
76 /** Gig Voice
77 *
78 * Renders a voice for the Gigasampler format.
79 */
80 class Voice {
81 public:
82 // Attributes
83 int MIDIKey; ///< MIDI key number of the key that triggered the voice
84 DiskThread* pDiskThread; ///< Pointer to the disk thread, to be able to order a disk stream and later to delete the stream again
85
86 // Methods
87 Voice();
88 ~Voice();
89 void Kill();
90 void Render(uint Samples);
91 void Reset();
92 void SetOutput(AudioOutputDevice* pAudioOutputDevice);
93 void SetEngine(Engine* pEngine);
94 int Trigger(Event* pNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument);
95 inline bool IsActive() { return Active; }
96 private:
97 // Types
98 enum playback_state_t {
99 playback_state_ram,
100 playback_state_disk,
101 playback_state_end
102 };
103
104 // Attributes
105 gig::Engine* pEngine; ///< Pointer to the sampler engine, to be able to access the event lists.
106 float Volume; ///< Volume level of the voice
107 float* pOutputLeft; ///< Audio output channel buffer (left)
108 float* pOutputRight; ///< Audio output channel buffer (right)
109 uint SampleRate; ///< Sample rate of the engines output audio signal (in Hz)
110 uint MaxSamplesPerCycle; ///< Size of each audio output buffer
111 double Pos; ///< Current playback position in sample
112 double PitchBase; ///< Basic pitch depth, stays the same for the whole life time of the voice
113 double PitchBend; ///< Current pitch value of the pitchbend wheel
114 ::gig::Sample* pSample; ///< Pointer to the sample to be played back
115 ::gig::Region* pRegion; ///< Pointer to the articulation information of the respective keyboard region of this voice
116 bool Active; ///< If this voice object is currently in usage
117 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
118 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
119 Stream::reference_t DiskStreamRef; ///< Reference / link to the disk stream
120 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.
121 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
122 int LoopCyclesLeft; ///< In case there is a RAMLoop and it's not an endless loop; reflects number of loop cycles left to be passed
123 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
124 EGADSR* pEG1; ///< Envelope Generator 1 (Amplification)
125 EGADSR* pEG2; ///< Envelope Generator 2 (Filter cutoff frequency)
126 EGDecay* pEG3; ///< Envelope Generator 3 (Pitch)
127 Filter FilterLeft;
128 Filter FilterRight;
129 midi_ctrl VCFCutoffCtrl;
130 midi_ctrl VCFResonanceCtrl;
131 int FilterUpdateCounter; ///< Used to update filter parameters all FILTER_UPDATE_PERIOD samples
132 static const float FILTER_CUTOFF_COEFF;
133 VCAManipulator* pVCAManipulator;
134 VCFCManipulator* pVCFCManipulator;
135 VCOManipulator* pVCOManipulator;
136 LFO<gig::VCAManipulator>* pLFO1; ///< Low Frequency Oscillator 1 (Amplification)
137 LFO<gig::VCFCManipulator>* pLFO2; ///< Low Frequency Oscillator 2 (Filter cutoff frequency)
138 LFO<gig::VCOManipulator>* pLFO3; ///< Low Frequency Oscillator 3 (Pitch)
139 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).
140
141 // Static Methods
142 static float CalculateFilterCutoffCoeff();
143
144 // Methods
145 void ProcessEvents(uint Samples);
146 void Interpolate(uint Samples, sample_t* pSrc, uint Skip);
147 void InterpolateAndLoop(uint Samples, sample_t* pSrc, uint Skip);
148 inline void InterpolateOneStep_Stereo(sample_t* pSrc, int& i, float& effective_volume, float& pitch, float& cutoff, float& resonance) {
149 int pos_int = RTMath::DoubleToInt(this->Pos); // integer position
150 float pos_fract = this->Pos - pos_int; // fractional part of position
151 pos_int <<= 1;
152
153 #if ENABLE_FILTER
154 UpdateFilter_Stereo(cutoff + FILTER_CUTOFF_MIN, resonance);
155 #endif // ENABLE_FILTER
156
157 #if USE_LINEAR_INTERPOLATION
158 #if ENABLE_FILTER
159 // left channel
160 pOutputLeft[i] += this->FilterLeft.Apply(effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int])));
161 // right channel
162 pOutputRight[i++] += this->FilterRight.Apply(effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1])));
163 #else // no filter
164 // left channel
165 pOutputLeft[i] += effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int]));
166 // right channel
167 pOutputRight[i++] += effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1]));
168 #endif // ENABLE_FILTER
169 #else // polynomial interpolation
170 // calculate left channel
171 float xm1 = pSrc[pos_int];
172 float x0 = pSrc[pos_int+2];
173 float x1 = pSrc[pos_int+4];
174 float x2 = pSrc[pos_int+6];
175 float a = (3 * (x0 - x1) - xm1 + x2) / 2;
176 float b = 2 * x1 + xm1 - (5 * x0 + x2) / 2;
177 float c = (x1 - xm1) / 2;
178 #if ENABLE_FILTER
179 pOutputLeft[i] += this->FilterLeft.Apply(effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
180 #else // no filter
181 pOutputRight[i] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
182 #endif // ENABLE_FILTER
183
184 //calculate right channel
185 xm1 = pSrc[pos_int+1];
186 x0 = pSrc[pos_int+3];
187 x1 = pSrc[pos_int+5];
188 x2 = pSrc[pos_int+7];
189 a = (3 * (x0 - x1) - xm1 + x2) / 2;
190 b = 2 * x1 + xm1 - (5 * x0 + x2) / 2;
191 c = (x1 - xm1) / 2;
192 #if ENABLE_FILTER
193 pOutputLeft[i++] += this->FilterRight.Apply(effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
194 #else // no filter
195 pOutputRight[i++] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
196 #endif // ENABLE_FILTER
197 #endif // USE_LINEAR_INTERPOLATION
198
199 this->Pos += pitch;
200 }
201 inline void InterpolateOneStep_Mono(sample_t* pSrc, int& i, float& effective_volume, float& pitch, float& cutoff, float& resonance) {
202 int pos_int = RTMath::DoubleToInt(this->Pos); // integer position
203 float pos_fract = this->Pos - pos_int; // fractional part of position
204
205 #if ENABLE_FILTER
206 UpdateFilter_Mono(cutoff + FILTER_CUTOFF_MIN, resonance);
207 #endif // ENABLE_FILTER
208
209 #if USE_LINEAR_INTERPOLATION
210 float sample_point = effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+1] - pSrc[pos_int]));
211 #else // polynomial interpolation
212 float xm1 = pSrc[pos_int];
213 float x0 = pSrc[pos_int+1];
214 float x1 = pSrc[pos_int+2];
215 float x2 = pSrc[pos_int+3];
216 float a = (3 * (x0 - x1) - xm1 + x2) / 2;
217 float b = 2 * x1 + xm1 - (5 * x0 + x2) / 2;
218 float c = (x1 - xm1) / 2;
219 float sample_point = effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
220 #endif // USE_LINEAR_INTERPOLATION
221
222 #if ENABLE_FILTER
223 sample_point = this->FilterLeft.Apply(sample_point);
224 #endif // ENABLE_FILTER
225
226 pOutputLeft[i] += sample_point;
227 pOutputRight[i++] += sample_point;
228
229 this->Pos += pitch;
230 }
231 inline void UpdateFilter_Stereo(float cutoff, float& resonance) {
232 if (!(++FilterUpdateCounter % FILTER_UPDATE_PERIOD) && (cutoff != FilterLeft.Cutoff() || resonance != FilterLeft.Resonance())) {
233 FilterLeft.SetParameters(cutoff, resonance, SampleRate);
234 FilterRight.SetParameters(cutoff, resonance, SampleRate);
235 }
236 }
237 inline void UpdateFilter_Mono(float cutoff, float& resonance) {
238 if (!(++FilterUpdateCounter % FILTER_UPDATE_PERIOD) && (cutoff != FilterLeft.Cutoff() || resonance != FilterLeft.Resonance())) {
239 FilterLeft.SetParameters(cutoff, resonance, SampleRate);
240 }
241 }
242 inline float Constrain(float ValueToCheck, float Min, float Max) {
243 if (ValueToCheck > Max) ValueToCheck = Max;
244 else if (ValueToCheck < Min) ValueToCheck = Min;
245 return ValueToCheck;
246 }
247 };
248
249 }} // namespace LinuxSampler::gig
250
251 #endif // __LS_GIG_VOICE_H__

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