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* * |
* * |
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* LinuxSampler - modular, streaming capable sampler * |
* LinuxSampler - modular, streaming capable sampler * |
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* * |
* * |
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* Copyright (C) 2003 by Benno Senoner and Christian Schoenebeck * |
* Copyright (C) 2003, 2004 by Benno Senoner and Christian Schoenebeck * |
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* * |
* * |
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* This program is free software; you can redistribute it and/or modify * |
* This program is free software; you can redistribute it and/or modify * |
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* it under the terms of the GNU General Public License as published by * |
* it under the terms of the GNU General Public License as published by * |
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#include "../../common/RTMath.h" |
#include "../../common/RTMath.h" |
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#include "../../common/RingBuffer.h" |
#include "../../common/RingBuffer.h" |
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#include "../../common/RTELMemoryPool.h" |
#include "../../common/RTELMemoryPool.h" |
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#include "../../audiodriver/AudioOutputDevice.h" |
#include "../../drivers/audio/AudioOutputDevice.h" |
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#include "../../lib/fileloader/libgig/gig.h" |
#include "../../lib/fileloader/libgig/gig.h" |
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#include "../common/BiquadFilter.h" |
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#include "Engine.h" |
#include "Engine.h" |
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#include "Stream.h" |
#include "Stream.h" |
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#include "DiskThread.h" |
#include "DiskThread.h" |
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#include "Filter.h" |
#include "Filter.h" |
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#include "../common/LFO.h" |
#include "../common/LFO.h" |
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#define USE_LINEAR_INTERPOLATION 1 ///< set to 0 if you prefer cubic interpolation (slower, better quality) |
#define USE_LINEAR_INTERPOLATION 0 ///< set to 0 if you prefer cubic interpolation (slower, better quality) |
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#define ENABLE_FILTER 0 ///< if set to 0 then filter (VCF) code is ignored on compile time |
#define ENABLE_FILTER 1 ///< if set to 0 then filter (VCF) code is ignored on compile time |
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#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) |
#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) |
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#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 |
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#define FILTER_CUTOFF_MAX 10000.0f ///< maximum cutoff frequency (10kHz) |
#define FILTER_CUTOFF_MAX 10000.0f ///< maximum cutoff frequency (10kHz) |
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#define FILTER_CUTOFF_MIN 100.0f ///< minimum cutoff frequency (100Hz) |
#define FILTER_CUTOFF_MIN 100.0f ///< minimum cutoff frequency (100Hz) |
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void Reset(); |
void Reset(); |
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void SetOutput(AudioOutputDevice* pAudioOutputDevice); |
void SetOutput(AudioOutputDevice* pAudioOutputDevice); |
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void SetEngine(Engine* pEngine); |
void SetEngine(Engine* pEngine); |
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int Trigger(Event* pNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument); |
int Trigger(Event* pNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument, int iLayer = 0); |
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inline bool IsActive() { return Active; } |
inline bool IsActive() { return Active; } |
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private: |
private: |
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// Types |
// Types |
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// Attributes |
// Attributes |
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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. |
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float Volume; ///< Volume level of the voice |
float Volume; ///< Volume level of the voice |
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float* pOutputLeft; ///< Audio output channel buffer (left) |
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float* pOutputRight; ///< Audio output channel buffer (right) |
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uint SampleRate; ///< Sample rate of the engines output audio signal (in Hz) |
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uint MaxSamplesPerCycle; ///< Size of each audio output buffer |
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double Pos; ///< Current playback position in sample |
double Pos; ///< Current playback position in sample |
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double PitchBase; ///< Basic pitch depth, stays the same for the whole life time of the voice |
double PitchBase; ///< Basic pitch depth, stays the same for the whole life time of the voice |
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double PitchBend; ///< Current pitch value of the pitchbend wheel |
double PitchBend; ///< Current pitch value of the pitchbend wheel |
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midi_ctrl VCFResonanceCtrl; |
midi_ctrl VCFResonanceCtrl; |
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int FilterUpdateCounter; ///< Used to update filter parameters all FILTER_UPDATE_PERIOD samples |
int FilterUpdateCounter; ///< Used to update filter parameters all FILTER_UPDATE_PERIOD samples |
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static const float FILTER_CUTOFF_COEFF; |
static const float FILTER_CUTOFF_COEFF; |
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static const int FILTER_UPDATE_MASK; |
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VCAManipulator* pVCAManipulator; |
VCAManipulator* pVCAManipulator; |
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VCFCManipulator* pVCFCManipulator; |
VCFCManipulator* pVCFCManipulator; |
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VCOManipulator* pVCOManipulator; |
VCOManipulator* pVCOManipulator; |
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// Static Methods |
// Static Methods |
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static float CalculateFilterCutoffCoeff(); |
static float CalculateFilterCutoffCoeff(); |
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static int CalculateFilterUpdateMask(); |
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// Methods |
// Methods |
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void ProcessEvents(uint Samples); |
void ProcessEvents(uint Samples); |
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#if ENABLE_FILTER |
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void CalculateBiquadParameters(uint Samples); |
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#endif // ENABLE_FILTER |
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void Interpolate(uint Samples, sample_t* pSrc, uint Skip); |
void Interpolate(uint Samples, sample_t* pSrc, uint Skip); |
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void InterpolateAndLoop(uint Samples, sample_t* pSrc, uint Skip); |
void InterpolateAndLoop(uint Samples, sample_t* pSrc, uint Skip); |
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inline void InterpolateOneStep_Stereo(sample_t* pSrc, int& i, float& effective_volume, float& pitch, float& cutoff, float& resonance) { |
inline void InterpolateOneStep_Stereo(sample_t* pSrc, int& i, float& effective_volume, float& pitch, biquad_param_t& bq_base, biquad_param_t& bq_main) { |
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int pos_int = RTMath::DoubleToInt(this->Pos); // integer position |
int pos_int = RTMath::DoubleToInt(this->Pos); // integer position |
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float pos_fract = this->Pos - pos_int; // fractional part of position |
float pos_fract = this->Pos - pos_int; // fractional part of position |
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pos_int <<= 1; |
pos_int <<= 1; |
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#if ENABLE_FILTER |
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UpdateFilter_Stereo(cutoff + FILTER_CUTOFF_MIN, resonance); |
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#endif // ENABLE_FILTER |
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#if USE_LINEAR_INTERPOLATION |
#if USE_LINEAR_INTERPOLATION |
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#if ENABLE_FILTER |
#if ENABLE_FILTER |
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// left channel |
// left channel |
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pOutputLeft[i] += this->FilterLeft.Apply(effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int]))); |
pEngine->pOutputLeft[i] += this->FilterLeft.Apply(&bq_base, &bq_main, effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int]))); |
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// right channel |
// right channel |
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pOutputRight[i++] += this->FilterRight.Apply(effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1]))); |
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]))); |
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#else // no filter |
#else // no filter |
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// left channel |
// left channel |
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pOutputLeft[i] += effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int])); |
pEngine->pOutputLeft[i] += effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int])); |
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// right channel |
// right channel |
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pOutputRight[i++] += effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1])); |
pEngine->pOutputRight[i++] += effective_volume * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1])); |
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#endif // ENABLE_FILTER |
#endif // ENABLE_FILTER |
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#else // polynomial interpolation |
#else // polynomial interpolation |
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// calculate left channel |
// calculate left channel |
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float x0 = pSrc[pos_int+2]; |
float x0 = pSrc[pos_int+2]; |
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float x1 = pSrc[pos_int+4]; |
float x1 = pSrc[pos_int+4]; |
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float x2 = pSrc[pos_int+6]; |
float x2 = pSrc[pos_int+6]; |
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float a = (3 * (x0 - x1) - xm1 + x2) / 2; |
float a = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f; |
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float b = 2 * x1 + xm1 - (5 * x0 + x2) / 2; |
float b = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f; |
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float c = (x1 - xm1) / 2; |
float c = (x1 - xm1) * 0.5f; |
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#if ENABLE_FILTER |
#if ENABLE_FILTER |
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pOutputLeft[i] += this->FilterLeft.Apply(effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0)); |
pEngine->pOutputLeft[i] += this->FilterLeft.Apply(&bq_base, &bq_main, effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0)); |
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#else // no filter |
#else // no filter |
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pOutputRight[i] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0); |
pEngine->pOutputLeft[i] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0); |
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#endif // ENABLE_FILTER |
#endif // ENABLE_FILTER |
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//calculate right channel |
//calculate right channel |
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x0 = pSrc[pos_int+3]; |
x0 = pSrc[pos_int+3]; |
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x1 = pSrc[pos_int+5]; |
x1 = pSrc[pos_int+5]; |
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x2 = pSrc[pos_int+7]; |
x2 = pSrc[pos_int+7]; |
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a = (3 * (x0 - x1) - xm1 + x2) / 2; |
a = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f; |
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b = 2 * x1 + xm1 - (5 * x0 + x2) / 2; |
b = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f; |
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c = (x1 - xm1) / 2; |
c = (x1 - xm1) * 0.5f; |
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#if ENABLE_FILTER |
#if ENABLE_FILTER |
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pOutputLeft[i++] += this->FilterRight.Apply(effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0)); |
pEngine->pOutputRight[i++] += this->FilterRight.Apply(&bq_base, &bq_main, effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0)); |
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#else // no filter |
#else // no filter |
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pOutputRight[i++] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0); |
pEngine->pOutputRight[i++] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0); |
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#endif // ENABLE_FILTER |
#endif // ENABLE_FILTER |
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#endif // USE_LINEAR_INTERPOLATION |
#endif // USE_LINEAR_INTERPOLATION |
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this->Pos += pitch; |
this->Pos += pitch; |
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} |
} |
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inline void InterpolateOneStep_Mono(sample_t* pSrc, int& i, float& effective_volume, float& pitch, float& cutoff, float& resonance) { |
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inline void InterpolateOneStep_Mono(sample_t* pSrc, int& i, float& effective_volume, float& pitch, biquad_param_t& bq_base, biquad_param_t& bq_main) { |
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int pos_int = RTMath::DoubleToInt(this->Pos); // integer position |
int pos_int = RTMath::DoubleToInt(this->Pos); // integer position |
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float pos_fract = this->Pos - pos_int; // fractional part of position |
float pos_fract = this->Pos - pos_int; // fractional part of position |
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#if ENABLE_FILTER |
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UpdateFilter_Mono(cutoff + FILTER_CUTOFF_MIN, resonance); |
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#endif // ENABLE_FILTER |
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#if USE_LINEAR_INTERPOLATION |
#if USE_LINEAR_INTERPOLATION |
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float sample_point = effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+1] - pSrc[pos_int])); |
float sample_point = effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+1] - pSrc[pos_int])); |
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#else // polynomial interpolation |
#else // polynomial interpolation |
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float x0 = pSrc[pos_int+1]; |
float x0 = pSrc[pos_int+1]; |
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float x1 = pSrc[pos_int+2]; |
float x1 = pSrc[pos_int+2]; |
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float x2 = pSrc[pos_int+3]; |
float x2 = pSrc[pos_int+3]; |
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float a = (3 * (x0 - x1) - xm1 + x2) / 2; |
float a = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f; |
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float b = 2 * x1 + xm1 - (5 * x0 + x2) / 2; |
float b = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f; |
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float c = (x1 - xm1) / 2; |
float c = (x1 - xm1) * 0.5f; |
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float sample_point = effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0); |
float sample_point = effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0); |
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#endif // USE_LINEAR_INTERPOLATION |
#endif // USE_LINEAR_INTERPOLATION |
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#if ENABLE_FILTER |
#if ENABLE_FILTER |
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sample_point = this->FilterLeft.Apply(sample_point); |
sample_point = this->FilterLeft.Apply(&bq_base, &bq_main, sample_point); |
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#endif // ENABLE_FILTER |
#endif // ENABLE_FILTER |
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pOutputLeft[i] += sample_point; |
pEngine->pOutputLeft[i] += sample_point; |
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pOutputRight[i++] += sample_point; |
pEngine->pOutputRight[i++] += sample_point; |
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this->Pos += pitch; |
this->Pos += pitch; |
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} |
} |
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inline void UpdateFilter_Stereo(float cutoff, float& resonance) { |
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if (!(++FilterUpdateCounter % FILTER_UPDATE_PERIOD) && (cutoff != FilterLeft.Cutoff() || resonance != FilterLeft.Resonance())) { |
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FilterLeft.SetParameters(cutoff, resonance, SampleRate); |
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FilterRight.SetParameters(cutoff, resonance, SampleRate); |
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} |
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} |
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inline void UpdateFilter_Mono(float cutoff, float& resonance) { |
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if (!(++FilterUpdateCounter % FILTER_UPDATE_PERIOD) && (cutoff != FilterLeft.Cutoff() || resonance != FilterLeft.Resonance())) { |
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FilterLeft.SetParameters(cutoff, resonance, SampleRate); |
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} |
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} |
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inline float Constrain(float ValueToCheck, float Min, float Max) { |
inline float Constrain(float ValueToCheck, float Min, float Max) { |
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if (ValueToCheck > Max) ValueToCheck = Max; |
if (ValueToCheck > Max) ValueToCheck = Max; |
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else if (ValueToCheck < Min) ValueToCheck = Min; |
else if (ValueToCheck < Min) ValueToCheck = Min; |