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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, 2004 by Benno Senoner and Christian Schoenebeck * |
* Copyright (C) 2003, 2004 by Benno Senoner and Christian Schoenebeck * |
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* Copyright (C) 2005, 2006 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/global.h" |
#include "../../common/global.h" |
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#if DEBUG_HEADERS |
#include <gig.h> |
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# warning Voice.h included |
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#endif // DEBUG_HEADERS |
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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/Pool.h" |
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#include "../../audiodriver/AudioOutputDevice.h" |
#include "../../drivers/audio/AudioOutputDevice.h" |
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#include "../../lib/fileloader/libgig/gig.h" |
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#include "../common/BiquadFilter.h" |
#include "../common/BiquadFilter.h" |
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#include "Engine.h" |
#include "Engine.h" |
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#include "EngineChannel.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 "EGADSR.h" |
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#include "EGDecay.h" |
#include "EGDecay.h" |
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#include "Filter.h" |
#include "Filter.h" |
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#include "../common/LFO.h" |
#include "../common/LFOBase.h" |
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#include "SynthesisParam.h" |
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#define USE_LINEAR_INTERPOLATION 0 ///< set to 0 if you prefer cubic interpolation (slower, better quality) |
#include "SmoothVolume.h" |
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#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, this value will be aligned to a power of two) |
// include the appropriate (unsigned) triangle LFO implementation |
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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 |
#if CONFIG_UNSIGNED_TRIANG_ALGO == INT_MATH_SOLUTION |
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#define FILTER_CUTOFF_MAX 10000.0f ///< maximum cutoff frequency (10kHz) |
# include "../common/LFOTriangleIntMath.h" |
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#define FILTER_CUTOFF_MIN 100.0f ///< minimum cutoff frequency (100Hz) |
#elif CONFIG_UNSIGNED_TRIANG_ALGO == INT_ABS_MATH_SOLUTION |
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# include "../common/LFOTriangleIntAbsMath.h" |
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// Uncomment following line to override external cutoff controller |
#elif CONFIG_UNSIGNED_TRIANG_ALGO == DI_HARMONIC_SOLUTION |
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//#define OVERRIDE_FILTER_CUTOFF_CTRL 1 ///< set to an arbitrary MIDI control change controller (e.g. 1 for 'modulation wheel') |
# include "../common/LFOTriangleDiHarmonic.h" |
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#else |
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// Uncomment following line to override external resonance controller |
# error "Unknown or no (unsigned) triangle LFO implementation selected!" |
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//#define OVERRIDE_FILTER_RES_CTRL 91 ///< set to an arbitrary MIDI control change controller (e.g. 91 for 'effect 1 depth') |
#endif |
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// Uncomment following line to override filter type |
// include the appropriate (signed) triangle LFO implementation |
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//#define OVERRIDE_FILTER_TYPE ::gig::vcf_type_lowpass ///< either ::gig::vcf_type_lowpass, ::gig::vcf_type_bandpass or ::gig::vcf_type_highpass |
#if CONFIG_SIGNED_TRIANG_ALGO == INT_MATH_SOLUTION |
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# include "../common/LFOTriangleIntMath.h" |
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#elif CONFIG_SIGNED_TRIANG_ALGO == INT_ABS_MATH_SOLUTION |
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# include "../common/LFOTriangleIntAbsMath.h" |
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#elif CONFIG_SIGNED_TRIANG_ALGO == DI_HARMONIC_SOLUTION |
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# include "../common/LFOTriangleDiHarmonic.h" |
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#else |
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# error "Unknown or no (signed) triangle LFO implementation selected!" |
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#endif |
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namespace LinuxSampler { namespace gig { |
namespace LinuxSampler { namespace gig { |
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class Engine; |
class Engine; |
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class EGADSR; |
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class VCAManipulator; |
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class VCFCManipulator; |
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class VCOManipulator; |
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/// Reflects a MIDI controller |
/// Reflects a MIDI controller |
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struct midi_ctrl { |
struct midi_ctrl { |
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float fvalue; ///< Transformed / effective value (e.g. volume level or filter cutoff frequency) |
float fvalue; ///< Transformed / effective value (e.g. volume level or filter cutoff frequency) |
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}; |
}; |
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#if CONFIG_UNSIGNED_TRIANG_ALGO == INT_MATH_SOLUTION |
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typedef LFOTriangleIntMath<range_unsigned> LFOUnsigned; |
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#elif CONFIG_UNSIGNED_TRIANG_ALGO == INT_ABS_MATH_SOLUTION |
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typedef LFOTriangleIntAbsMath<range_unsigned> LFOUnsigned; |
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#elif CONFIG_UNSIGNED_TRIANG_ALGO == DI_HARMONIC_SOLUTION |
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typedef LFOTriangleDiHarmonic<range_unsigned> LFOUnsigned; |
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#endif |
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#if CONFIG_SIGNED_TRIANG_ALGO == INT_MATH_SOLUTION |
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typedef LFOTriangleIntMath<range_signed> LFOSigned; |
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#elif CONFIG_SIGNED_TRIANG_ALGO == INT_ABS_MATH_SOLUTION |
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typedef LFOTriangleIntAbsMath<range_signed> LFOSigned; |
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#elif CONFIG_SIGNED_TRIANG_ALGO == DI_HARMONIC_SOLUTION |
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typedef LFOTriangleDiHarmonic<range_signed> LFOSigned; |
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#endif |
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/** Gig Voice |
/** Gig Voice |
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* |
* |
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* Renders a voice for the Gigasampler format. |
* Renders a voice for the Gigasampler format. |
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*/ |
*/ |
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class Voice { |
class Voice { |
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public: |
public: |
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// Types |
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enum type_t { |
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type_normal, |
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type_release_trigger_required, ///< If the key of this voice will be released, it causes a release triggered voice to be spawned |
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type_release_trigger ///< Release triggered voice which cannot be killed by releasing its key |
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}; |
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// Attributes |
// Attributes |
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type_t Type; ///< Voice Type |
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int MIDIKey; ///< MIDI key number of the key that triggered the voice |
int MIDIKey; ///< MIDI key number of the key that triggered the voice |
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uint KeyGroup; |
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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 |
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// Methods |
// Methods |
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Voice(); |
Voice(); |
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~Voice(); |
virtual ~Voice(); |
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void Kill(); |
void Kill(Pool<Event>::Iterator& itKillEvent); |
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void Render(uint Samples); |
void Render(uint Samples); |
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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(EngineChannel* pEngineChannel, Pool<Event>::Iterator& itNoteOnEvent, int PitchBend, ::gig::DimensionRegion* pDimRgn, type_t VoiceType, int iKeyGroup); |
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inline bool IsActive() { return Active; } |
inline bool IsActive() { return PlaybackState; } |
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private: |
inline bool IsStealable() { return !itKillEvent && PlaybackState >= playback_state_ram; } |
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void UpdatePortamentoPos(Pool<Event>::Iterator& itNoteOffEvent); |
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//private: |
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// Types |
// Types |
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enum playback_state_t { |
enum playback_state_t { |
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playback_state_ram, |
playback_state_end = 0, |
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playback_state_disk, |
playback_state_init = 1, |
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playback_state_end |
playback_state_ram = 2, |
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playback_state_disk = 3 |
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}; |
}; |
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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. |
EngineChannel* pEngineChannel; |
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float Volume; ///< Volume level of the voice |
Engine* pEngine; ///< Pointer to the sampler engine, to be able to access the event lists. |
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float* pOutputLeft; ///< Audio output channel buffer (left) |
float VolumeLeft; ///< Left channel volume. This factor is calculated when the voice is triggered and doesn't change after that. |
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float* pOutputRight; ///< Audio output channel buffer (right) |
float VolumeRight; ///< Right channel volume. This factor is calculated when the voice is triggered and doesn't change after that. |
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uint SampleRate; ///< Sample rate of the engines output audio signal (in Hz) |
SmoothVolume CrossfadeSmoother; ///< Crossfade volume, updated by crossfade CC events |
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uint MaxSamplesPerCycle; ///< Size of each audio output buffer |
SmoothVolume VolumeSmoother; ///< Volume, updated by CC 7 (volume) events |
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SmoothVolume PanLeftSmoother; ///< Left channel volume, updated by CC 10 (pan) events |
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SmoothVolume PanRightSmoother; ///< Right channel volume, updated by CC 10 (pan) events |
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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 |
float 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 |
float PitchBend; ///< Current pitch value of the pitchbend wheel |
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float CutoffBase; ///< Cutoff frequency before control change, EG and LFO are applied |
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::gig::Sample* pSample; ///< Pointer to the sample to be played back |
::gig::Sample* pSample; ///< Pointer to the sample to be played back |
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::gig::Region* pRegion; ///< Pointer to the articulation information of the respective keyboard region of this voice |
::gig::DimensionRegion* pDimRgn; ///< Pointer to the articulation information of current dimension region of this voice |
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bool Active; ///< If this voice object is currently in usage |
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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 |
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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 |
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Stream::reference_t DiskStreamRef; ///< Reference / link to the disk stream |
Stream::reference_t DiskStreamRef; ///< Reference / link to the disk stream |
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int RealSampleWordsLeftToRead; ///< Number of samples left to read, not including the silence added for the interpolator |
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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. |
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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 |
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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 |
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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 |
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EGADSR* pEG1; ///< Envelope Generator 1 (Amplification) |
EGADSR EG1; ///< Envelope Generator 1 (Amplification) |
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EGADSR* pEG2; ///< Envelope Generator 2 (Filter cutoff frequency) |
EGADSR EG2; ///< Envelope Generator 2 (Filter cutoff frequency) |
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EGDecay* pEG3; ///< Envelope Generator 3 (Pitch) |
EGDecay EG3; ///< Envelope Generator 3 (Pitch) |
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Filter FilterLeft; |
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Filter FilterRight; |
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midi_ctrl VCFCutoffCtrl; |
midi_ctrl VCFCutoffCtrl; |
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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 |
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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; |
LFOUnsigned* pLFO1; ///< Low Frequency Oscillator 1 (Amplification) |
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VCAManipulator* pVCAManipulator; |
LFOUnsigned* pLFO2; ///< Low Frequency Oscillator 2 (Filter cutoff frequency) |
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VCFCManipulator* pVCFCManipulator; |
LFOSigned* pLFO3; ///< Low Frequency Oscillator 3 (Pitch) |
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VCOManipulator* pVCOManipulator; |
bool bLFO1Enabled; ///< Should we use the Amplitude LFO for this voice? |
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LFO<gig::VCAManipulator>* pLFO1; ///< Low Frequency Oscillator 1 (Amplification) |
bool bLFO2Enabled; ///< Should we use the Filter Cutoff LFO for this voice? |
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LFO<gig::VCFCManipulator>* pLFO2; ///< Low Frequency Oscillator 2 (Filter cutoff frequency) |
bool bLFO3Enabled; ///< Should we use the Pitch LFO for this voice? |
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LFO<gig::VCOManipulator>* pLFO3; ///< Low Frequency Oscillator 3 (Pitch) |
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). |
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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). |
//public: // FIXME: just made public for debugging (sanity check in Engine::RenderAudio()), should be changed to private before the final release |
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Pool<Event>::Iterator itKillEvent; ///< Event which caused this voice to be killed |
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//private: |
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int SynthesisMode; |
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float fFinalCutoff; |
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float fFinalResonance; |
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SynthesisParam finalSynthesisParameters; |
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Loop loop; |
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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 KillImmediately(); |
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#if ENABLE_FILTER |
void ProcessEvents(uint Samples); |
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void CalculateBiquadParameters(uint Samples); |
void Synthesize(uint Samples, sample_t* pSrc, uint Skip); |
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#endif // ENABLE_FILTER |
void processTransitionEvents(RTList<Event>::Iterator& itEvent, uint End); |
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void Interpolate(uint Samples, sample_t* pSrc, uint Skip); |
void processCCEvents(RTList<Event>::Iterator& itEvent, uint End); |
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void InterpolateAndLoop(uint Samples, sample_t* pSrc, uint Skip); |
void processPitchEvent(RTList<Event>::Iterator& itEvent); |
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inline void InterpolateOneStep_Stereo(sample_t* pSrc, int& i, float& effective_volume, float& pitch, biquad_param_t& bq_base, biquad_param_t& bq_main) { |
void processCrossFadeEvent(RTList<Event>::Iterator& itEvent); |
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int pos_int = RTMath::DoubleToInt(this->Pos); // integer position |
void processCutoffEvent(RTList<Event>::Iterator& itEvent); |
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float pos_fract = this->Pos - pos_int; // fractional part of position |
void processResonanceEvent(RTList<Event>::Iterator& itEvent); |
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pos_int <<= 1; |
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inline uint8_t CrossfadeAttenuation(uint8_t& CrossfadeControllerValue) { |
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#if 0 //ENABLE_FILTER |
uint8_t c = std::max(CrossfadeControllerValue, pDimRgn->AttenuationControllerThreshold); |
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UpdateFilter_Stereo(cutoff + FILTER_CUTOFF_MIN, resonance); |
c = (!pDimRgn->Crossfade.out_end) ? c /* 0,0,0,0 means no crossfade defined */ |
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#endif // ENABLE_FILTER |
: (c < pDimRgn->Crossfade.in_end) ? |
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((c <= pDimRgn->Crossfade.in_start) ? 0 |
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#if USE_LINEAR_INTERPOLATION |
: 127 * (c - pDimRgn->Crossfade.in_start) / (pDimRgn->Crossfade.in_end - pDimRgn->Crossfade.in_start)) |
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#if ENABLE_FILTER |
: (c <= pDimRgn->Crossfade.out_start) ? 127 |
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// left channel |
: (c < pDimRgn->Crossfade.out_end) ? 127 * (pDimRgn->Crossfade.out_end - c) / (pDimRgn->Crossfade.out_end - pDimRgn->Crossfade.out_start) |
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pOutputLeft[i] += this->FilterLeft.Apply(&bq_base, &bq_main, effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int]))); |
: 0; |
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// right channel |
return pDimRgn->InvertAttenuationController ? 127 - c : c; |
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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 |
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// left channel |
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pOutputLeft[i] += effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int])); |
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// right channel |
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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 |
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#else // polynomial interpolation |
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// calculate left channel |
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float xm1 = pSrc[pos_int]; |
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float x0 = pSrc[pos_int+2]; |
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float x1 = pSrc[pos_int+4]; |
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float x2 = pSrc[pos_int+6]; |
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float a = (3 * (x0 - x1) - xm1 + x2) / 2; |
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float b = 2 * x1 + xm1 - (5 * x0 + x2) / 2; |
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float c = (x1 - xm1) / 2; |
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#if ENABLE_FILTER |
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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 |
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pOutputRight[i] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0); |
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#endif // ENABLE_FILTER |
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//calculate right channel |
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xm1 = pSrc[pos_int+1]; |
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x0 = pSrc[pos_int+3]; |
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x1 = pSrc[pos_int+5]; |
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x2 = pSrc[pos_int+7]; |
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a = (3 * (x0 - x1) - xm1 + x2) / 2; |
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b = 2 * x1 + xm1 - (5 * x0 + x2) / 2; |
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c = (x1 - xm1) / 2; |
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#if ENABLE_FILTER |
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pOutputLeft[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 |
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pOutputRight[i++] += effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0); |
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#endif // ENABLE_FILTER |
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#endif // USE_LINEAR_INTERPOLATION |
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this->Pos += pitch; |
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206 |
} |
} |
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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 |
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float pos_fract = this->Pos - pos_int; // fractional part of position |
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#if 0 //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 |
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float sample_point = effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+1] - pSrc[pos_int])); |
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#else // polynomial interpolation |
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float xm1 = pSrc[pos_int]; |
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float x0 = pSrc[pos_int+1]; |
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float x1 = pSrc[pos_int+2]; |
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float x2 = pSrc[pos_int+3]; |
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float a = (3 * (x0 - x1) - xm1 + x2) / 2; |
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float b = 2 * x1 + xm1 - (5 * x0 + x2) / 2; |
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float c = (x1 - xm1) / 2; |
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float sample_point = effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0); |
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#endif // USE_LINEAR_INTERPOLATION |
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#if ENABLE_FILTER |
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sample_point = this->FilterLeft.Apply(&bq_base, &bq_main, sample_point); |
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#endif // ENABLE_FILTER |
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pOutputLeft[i] += sample_point; |
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pOutputRight[i++] += sample_point; |
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207 |
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this->Pos += pitch; |
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} |
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#if 0 |
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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); |
|
|
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 |
|
208 |
inline float Constrain(float ValueToCheck, float Min, float Max) { |
inline float Constrain(float ValueToCheck, float Min, float Max) { |
209 |
if (ValueToCheck > Max) ValueToCheck = Max; |
if (ValueToCheck > Max) ValueToCheck = Max; |
210 |
else if (ValueToCheck < Min) ValueToCheck = Min; |
else if (ValueToCheck < Min) ValueToCheck = Min; |