/[svn]/linuxsampler/trunk/src/engines/gig/Voice.h
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revision 80 by schoenebeck, Sun May 23 19:16:33 2004 UTC revision 348 by schoenebeck, Sun Jan 23 21:24:16 2005 UTC
# Line 31  Line 31 
31    
32  #include "../../common/RTMath.h"  #include "../../common/RTMath.h"
33  #include "../../common/RingBuffer.h"  #include "../../common/RingBuffer.h"
34  #include "../../common/RTELMemoryPool.h"  #include "../../common/Pool.h"
35  #include "../../audiodriver/AudioOutputDevice.h"  #include "../../drivers/audio/AudioOutputDevice.h"
36  #include "../../lib/fileloader/libgig/gig.h"  #include "../../lib/fileloader/libgig/gig.h"
37  #include "../common/BiquadFilter.h"  #include "../common/BiquadFilter.h"
38  #include "Engine.h"  #include "Engine.h"
# Line 43  Line 43 
43  #include "Filter.h"  #include "Filter.h"
44  #include "../common/LFO.h"  #include "../common/LFO.h"
45    
 #define USE_LINEAR_INTERPOLATION        0  ///< set to 0 if you prefer cubic interpolation (slower, better quality)  
 #define ENABLE_FILTER                   1  ///< if set to 0 then filter (VCF) code is ignored on compile time  
46  #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)  #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)
47  #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  #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
48  #define FILTER_CUTOFF_MAX               10000.0f ///< maximum cutoff frequency (10kHz)  #define FILTER_CUTOFF_MAX               10000.0f ///< maximum cutoff frequency (10kHz)
# Line 80  namespace LinuxSampler { namespace gig { Line 78  namespace LinuxSampler { namespace gig {
78       */       */
79      class Voice {      class Voice {
80          public:          public:
81                // Types
82                enum type_t {
83                    type_normal,
84                    type_release_trigger_required,  ///< If the key of this voice will be released, it causes a release triggered voice to be spawned
85                    type_release_trigger            ///< Release triggered voice which cannot be killed by releasing its key
86                };
87    
88              // Attributes              // Attributes
89                type_t       Type;         ///< Voice Type
90              int          MIDIKey;      ///< MIDI key number of the key that triggered the voice              int          MIDIKey;      ///< MIDI key number of the key that triggered the voice
91                uint         KeyGroup;
92              DiskThread*  pDiskThread;  ///< Pointer to the disk thread, to be able to order a disk stream and later to delete the stream again              DiskThread*  pDiskThread;  ///< Pointer to the disk thread, to be able to order a disk stream and later to delete the stream again
93                RTList<Voice>::Iterator itChildVoice; ///< Points to the next layer voice (if any). This field is currently only used by the voice stealing algorithm.
94    
95              // Methods              // Methods
96              Voice();              Voice();
97             ~Voice();             ~Voice();
98              void Kill();              void Kill(Pool<Event>::Iterator& itKillEvent);
99              void Render(uint Samples);              void Render(uint Samples);
100              void Reset();              void Reset();
101              void SetOutput(AudioOutputDevice* pAudioOutputDevice);              void SetOutput(AudioOutputDevice* pAudioOutputDevice);
102              void SetEngine(Engine* pEngine);              void SetEngine(Engine* pEngine);
103              int  Trigger(Event* pNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument);              int  Trigger(Pool<Event>::Iterator& itNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument, int iLayer, bool ReleaseTriggerVoice, bool VoiceStealing);
104              inline bool IsActive() { return Active; }              inline bool IsActive() { return PlaybackState; }
105          private:          //private:
106              // Types              // Types
107              enum playback_state_t {              enum playback_state_t {
108                  playback_state_ram,                  playback_state_end  = 0,
109                  playback_state_disk,                  playback_state_ram  = 1,
110                  playback_state_end                  playback_state_disk = 2
111              };              };
112    
113              // Attributes              // Attributes
114              gig::Engine*                pEngine;            ///< Pointer to the sampler engine, to be able to access the event lists.              gig::Engine*                pEngine;            ///< Pointer to the sampler engine, to be able to access the event lists.
115              float                       Volume;             ///< Volume level of the voice              float                       Volume;             ///< Volume level of the voice
116              float*                      pOutputLeft;        ///< Audio output channel buffer (left)              float                       PanLeft;
117              float*                      pOutputRight;       ///< Audio output channel buffer (right)              float                       PanRight;
118              uint                        SampleRate;         ///< Sample rate of the engines output audio signal (in Hz)              float                       CrossfadeVolume;    ///< Current attenuation level caused by a crossfade (only if a crossfade is defined of course)
             uint                        MaxSamplesPerCycle; ///< Size of each audio output buffer  
119              double                      Pos;                ///< Current playback position in sample              double                      Pos;                ///< Current playback position in sample
120              double                      PitchBase;          ///< Basic pitch depth, stays the same for the whole life time of the voice              float                       PitchBase;          ///< Basic pitch depth, stays the same for the whole life time of the voice
121              double                      PitchBend;          ///< Current pitch value of the pitchbend wheel              float                       PitchBend;          ///< Current pitch value of the pitchbend wheel
122              ::gig::Sample*              pSample;            ///< Pointer to the sample to be played back              ::gig::Sample*              pSample;            ///< Pointer to the sample to be played back
123              ::gig::Region*              pRegion;            ///< Pointer to the articulation information of the respective keyboard region of this voice              ::gig::Region*              pRegion;            ///< Pointer to the articulation information of the respective keyboard region of this voice
124              bool                        Active;             ///< If this voice object is currently in usage              ::gig::DimensionRegion*     pDimRgn;            ///< Pointer to the articulation information of current dimension region of this voice
125              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              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
126              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              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
127              Stream::reference_t         DiskStreamRef;      ///< Reference / link to the disk stream              Stream::reference_t         DiskStreamRef;      ///< Reference / link to the disk stream
128                int                         RealSampleWordsLeftToRead; ///< Number of samples left to read, not including the silence added for the interpolator
129              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.              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.
130              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              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
131              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                        LoopCyclesLeft;     ///< In case there is a RAMLoop and it's not an endless loop; reflects number of loop cycles left to be passed
132              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              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
133              EGADSR*                     pEG1;               ///< Envelope Generator 1 (Amplification)              EGADSR*                     pEG1;               ///< Envelope Generator 1 (Amplification)
134              EGADSR*                     pEG2;               ///< Envelope Generator 2 (Filter cutoff frequency)              EGADSR*                     pEG2;               ///< Envelope Generator 2 (Filter cutoff frequency)
# Line 138  namespace LinuxSampler { namespace gig { Line 146  namespace LinuxSampler { namespace gig {
146              LFO<gig::VCAManipulator>*   pLFO1;              ///< Low Frequency Oscillator 1 (Amplification)              LFO<gig::VCAManipulator>*   pLFO1;              ///< Low Frequency Oscillator 1 (Amplification)
147              LFO<gig::VCFCManipulator>*  pLFO2;             ///< Low Frequency Oscillator 2 (Filter cutoff frequency)              LFO<gig::VCFCManipulator>*  pLFO2;             ///< Low Frequency Oscillator 2 (Filter cutoff frequency)
148              LFO<gig::VCOManipulator>*   pLFO3;              ///< Low Frequency Oscillator 3 (Pitch)              LFO<gig::VCOManipulator>*   pLFO3;              ///< Low Frequency Oscillator 3 (Pitch)
149              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).              Pool<Event>::Iterator       itTriggerEvent;      ///< 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).
150            //public: // FIXME: just made public for debugging (sanity check in Engine::RenderAudio()), should be changed to private before the final release
151                Pool<Event>::Iterator       itKillEvent;         ///< Event which caused this voice to be killed
152            //private:
153                int                         SynthesisMode;
154    
155              // Static Methods              // Static Methods
156              static float CalculateFilterCutoffCoeff();              static float CalculateFilterCutoffCoeff();
157              static int   CalculateFilterUpdateMask();              static int   CalculateFilterUpdateMask();
158    
159              // Methods              // Methods
160              void        ProcessEvents(uint Samples);              void KillImmediately();
161              #if ENABLE_FILTER              void ProcessEvents(uint Samples);
162              void        CalculateBiquadParameters(uint Samples);              void CalculateBiquadParameters(uint Samples);
163              #endif // ENABLE_FILTER              void Synthesize(uint Samples, sample_t* pSrc, uint Skip);
164              void        Interpolate(uint Samples, sample_t* pSrc, uint Skip);  
165              void        InterpolateAndLoop(uint Samples, sample_t* pSrc, uint Skip);              inline float CrossfadeAttenuation(uint8_t& CrossfadeControllerValue) {
166              inline void InterpolateOneStep_Stereo(sample_t* pSrc, int& i, float& effective_volume, float& pitch, biquad_param_t& bq_base, biquad_param_t& bq_main) {                  float att = (!pDimRgn->Crossfade.out_end) ? CrossfadeControllerValue / 127.0f /* 0,0,0,0 means no crossfade defined */
167                  int   pos_int   = RTMath::DoubleToInt(this->Pos);  // integer position                            : (CrossfadeControllerValue < pDimRgn->Crossfade.in_end) ?
168                  float pos_fract = this->Pos - pos_int;             // fractional part of position                                  ((CrossfadeControllerValue <= pDimRgn->Crossfade.in_start) ? 0.0f
169                  pos_int <<= 1;                                  : float(CrossfadeControllerValue - pDimRgn->Crossfade.in_start) / float(pDimRgn->Crossfade.in_end - pDimRgn->Crossfade.in_start))
170                              : (CrossfadeControllerValue <= pDimRgn->Crossfade.out_start) ? 1.0f
171                  #if 0 //ENABLE_FILTER                            : (CrossfadeControllerValue < pDimRgn->Crossfade.out_end) ? float(pDimRgn->Crossfade.out_end - CrossfadeControllerValue) / float(pDimRgn->Crossfade.out_end - pDimRgn->Crossfade.out_start)
172                      UpdateFilter_Stereo(cutoff + FILTER_CUTOFF_MIN, resonance);                            : 0.0f;
173                  #endif // ENABLE_FILTER                  return pDimRgn->InvertAttenuationController ? 1 - att : att;
   
                 #if USE_LINEAR_INTERPOLATION  
                     #if ENABLE_FILTER  
                         // left channel  
                         pOutputLeft[i]    += this->FilterLeft.Apply(&bq_base, &bq_main, effective_volume * (pSrc[pos_int]   + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int])));  
                         // right channel  
                         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])));  
                     #else // no filter  
                         // left channel  
                         pOutputLeft[i]    += effective_volume * (pSrc[pos_int]   + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int]));  
                         // right channel  
                         pOutputRight[i++] += effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1]));  
                     #endif // ENABLE_FILTER  
                 #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;  
                     #if ENABLE_FILTER  
                         pOutputLeft[i] += this->FilterLeft.Apply(&bq_base, &bq_main, effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));  
                     #else // no filter  
                         pOutputRight[i] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);  
                     #endif // ENABLE_FILTER  
   
                     //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;  
                     #if ENABLE_FILTER  
                         pOutputLeft[i++] += this->FilterRight.Apply(&bq_base, &bq_main, effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0));  
                     #else // no filter  
                         pOutputRight[i++] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0);  
                     #endif // ENABLE_FILTER  
                 #endif // USE_LINEAR_INTERPOLATION  
   
                 this->Pos += pitch;  
174              }              }
             inline void InterpolateOneStep_Mono(sample_t* pSrc, int& i, float& effective_volume, float& pitch,  biquad_param_t& bq_base, biquad_param_t& bq_main) {  
                 int   pos_int   = RTMath::DoubleToInt(this->Pos);  // integer position  
                 float pos_fract = this->Pos - pos_int;             // fractional part of position  
   
                 #if 0 //ENABLE_FILTER  
                     UpdateFilter_Mono(cutoff + FILTER_CUTOFF_MIN, resonance);  
                 #endif // ENABLE_FILTER  
   
                 #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  
   
                 #if ENABLE_FILTER  
                     sample_point = this->FilterLeft.Apply(&bq_base, &bq_main, sample_point);  
                 #endif // ENABLE_FILTER  
175    
                 pOutputLeft[i]    += sample_point;  
                 pOutputRight[i++] += sample_point;  
   
                 this->Pos += pitch;  
             }  
 #if 0  
             inline void UpdateFilter_Stereo(float cutoff, float& resonance) {  
                 if (!(++FilterUpdateCounter % FILTER_UPDATE_PERIOD) && (cutoff != FilterLeft.Cutoff() || resonance != FilterLeft.Resonance())) {  
                     FilterLeft.SetParameters(cutoff, resonance, SampleRate);  
                     FilterRight.SetParameters(cutoff, resonance, SampleRate);  
                 }  
             }  
             inline void UpdateFilter_Mono(float cutoff, float& resonance) {  
                 if (!(++FilterUpdateCounter % FILTER_UPDATE_PERIOD) && (cutoff != FilterLeft.Cutoff() || resonance != FilterLeft.Resonance())) {  
                     FilterLeft.SetParameters(cutoff, resonance, SampleRate);  
                 }  
             }  
 #endif  
176              inline float Constrain(float ValueToCheck, float Min, float Max) {              inline float Constrain(float ValueToCheck, float Min, float Max) {
177                  if      (ValueToCheck > Max) ValueToCheck = Max;                  if      (ValueToCheck > Max) ValueToCheck = Max;
178                  else if (ValueToCheck < Min) ValueToCheck = Min;                  else if (ValueToCheck < Min) ValueToCheck = Min;
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