/[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 Jun 1 16:19:40 2004 UTC (19 years, 10 months ago) by letz
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* Do polynomial interpolation computation in float

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 "../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);
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 float* pOutputLeft; ///< Audio output channel buffer (left)
109 float* pOutputRight; ///< Audio output channel buffer (right)
110 uint SampleRate; ///< Sample rate of the engines output audio signal (in Hz)
111 uint MaxSamplesPerCycle; ///< Size of each audio output buffer
112 double Pos; ///< Current playback position in sample
113 double PitchBase; ///< Basic pitch depth, stays the same for the whole life time of the voice
114 double PitchBend; ///< Current pitch value of the pitchbend wheel
115 ::gig::Sample* pSample; ///< Pointer to the sample to be played back
116 ::gig::Region* pRegion; ///< Pointer to the articulation information of the respective keyboard region of this voice
117 bool Active; ///< If this voice object is currently in usage
118 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
119 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
120 Stream::reference_t DiskStreamRef; ///< Reference / link to the disk stream
121 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.
122 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
123 int LoopCyclesLeft; ///< In case there is a RAMLoop and it's not an endless loop; reflects number of loop cycles left to be passed
124 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
125 EGADSR* pEG1; ///< Envelope Generator 1 (Amplification)
126 EGADSR* pEG2; ///< Envelope Generator 2 (Filter cutoff frequency)
127 EGDecay* pEG3; ///< Envelope Generator 3 (Pitch)
128 Filter FilterLeft;
129 Filter FilterRight;
130 midi_ctrl VCFCutoffCtrl;
131 midi_ctrl VCFResonanceCtrl;
132 int FilterUpdateCounter; ///< Used to update filter parameters all FILTER_UPDATE_PERIOD samples
133 static const float FILTER_CUTOFF_COEFF;
134 static const int FILTER_UPDATE_MASK;
135 VCAManipulator* pVCAManipulator;
136 VCFCManipulator* pVCFCManipulator;
137 VCOManipulator* pVCOManipulator;
138 LFO<gig::VCAManipulator>* pLFO1; ///< Low Frequency Oscillator 1 (Amplification)
139 LFO<gig::VCFCManipulator>* pLFO2; ///< Low Frequency Oscillator 2 (Filter cutoff frequency)
140 LFO<gig::VCOManipulator>* pLFO3; ///< Low Frequency Oscillator 3 (Pitch)
141 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).
142
143 // Static Methods
144 static float CalculateFilterCutoffCoeff();
145 static int CalculateFilterUpdateMask();
146
147 // Methods
148 void ProcessEvents(uint Samples);
149 #if ENABLE_FILTER
150 void CalculateBiquadParameters(uint Samples);
151 #endif // ENABLE_FILTER
152 void Interpolate(uint Samples, sample_t* pSrc, uint Skip);
153 void InterpolateAndLoop(uint Samples, sample_t* pSrc, uint Skip);
154 inline void InterpolateOneStep_Stereo(sample_t* pSrc, int& i, float& effective_volume, float& pitch, biquad_param_t& bq_base, biquad_param_t& bq_main) {
155 int pos_int = RTMath::DoubleToInt(this->Pos); // integer position
156 float pos_fract = this->Pos - pos_int; // fractional part of position
157 pos_int <<= 1;
158
159 #if USE_LINEAR_INTERPOLATION
160 #if ENABLE_FILTER
161 // left channel
162 pOutputLeft[i] += this->FilterLeft.Apply(&bq_base, &bq_main, effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int])));
163 // right channel
164 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])));
165 #else // no filter
166 // left channel
167 pOutputLeft[i] += effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int]));
168 // right channel
169 pOutputRight[i++] += effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1]));
170 #endif // ENABLE_FILTER
171 #else // polynomial interpolation
172 // calculate left channel
173 float xm1 = pSrc[pos_int];
174 float x0 = pSrc[pos_int+2];
175 float x1 = pSrc[pos_int+4];
176 float x2 = pSrc[pos_int+6];
177 float a = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f;
178 float b = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f;
179 float c = (x1 - xm1) * 0.5f;
180 #if ENABLE_FILTER
181 pOutputLeft[i] += this->FilterLeft.Apply(&bq_base, &bq_main, effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
182 #else // no filter
183 pOutputLeft[i] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
184 #endif // ENABLE_FILTER
185
186 //calculate right channel
187 xm1 = pSrc[pos_int+1];
188 x0 = pSrc[pos_int+3];
189 x1 = pSrc[pos_int+5];
190 x2 = pSrc[pos_int+7];
191 a = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f;
192 b = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f;
193 c = (x1 - xm1) * 0.5f;
194 #if ENABLE_FILTER
195 pOutputRight[i++] += this->FilterRight.Apply(&bq_base, &bq_main, effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));
196 #else // no filter
197 pOutputRight[i++] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
198 #endif // ENABLE_FILTER
199 #endif // USE_LINEAR_INTERPOLATION
200
201 this->Pos += pitch;
202 }
203
204 inline void InterpolateOneStep_Mono(sample_t* pSrc, int& i, float& effective_volume, float& pitch, biquad_param_t& bq_base, biquad_param_t& bq_main) {
205 int pos_int = RTMath::DoubleToInt(this->Pos); // integer position
206 float pos_fract = this->Pos - pos_int; // fractional part of position
207
208 #if USE_LINEAR_INTERPOLATION
209 float sample_point = effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+1] - pSrc[pos_int]));
210 #else // polynomial interpolation
211 float xm1 = pSrc[pos_int];
212 float x0 = pSrc[pos_int+1];
213 float x1 = pSrc[pos_int+2];
214 float x2 = pSrc[pos_int+3];
215 float a = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f;
216 float b = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f;
217 float c = (x1 - xm1) * 0.5f;
218 float sample_point = effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);
219 #endif // USE_LINEAR_INTERPOLATION
220
221 #if ENABLE_FILTER
222 sample_point = this->FilterLeft.Apply(&bq_base, &bq_main, sample_point);
223 #endif // ENABLE_FILTER
224
225 pOutputLeft[i] += sample_point;
226 pOutputRight[i++] += sample_point;
227
228 this->Pos += pitch;
229 }
230
231 inline float Constrain(float ValueToCheck, float Min, float Max) {
232 if (ValueToCheck > Max) ValueToCheck = Max;
233 else if (ValueToCheck < Min) ValueToCheck = Min;
234 return ValueToCheck;
235 }
236 };
237
238 }} // namespace LinuxSampler::gig
239
240 #endif // __LS_GIG_VOICE_H__

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