--- linuxsampler/trunk/src/engines/gig/Voice.h 2004/04/27 09:21:58 56 +++ linuxsampler/trunk/src/engines/gig/Voice.h 2004/10/08 20:51:39 271 @@ -32,8 +32,9 @@ #include "../../common/RTMath.h" #include "../../common/RingBuffer.h" #include "../../common/RTELMemoryPool.h" -#include "../../audiodriver/AudioOutputDevice.h" +#include "../../drivers/audio/AudioOutputDevice.h" #include "../../lib/fileloader/libgig/gig.h" +#include "../common/BiquadFilter.h" #include "Engine.h" #include "Stream.h" #include "DiskThread.h" @@ -42,9 +43,9 @@ #include "Filter.h" #include "../common/LFO.h" -#define USE_LINEAR_INTERPOLATION 1 ///< set to 0 if you prefer cubic interpolation (slower, better quality) -#define ENABLE_FILTER 0 ///< if set to 0 then filter (VCF) code is ignored on compile time -#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 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 +#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 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 FILTER_CUTOFF_MAX 10000.0f ///< maximum cutoff frequency (10kHz) #define FILTER_CUTOFF_MIN 100.0f ///< minimum cutoff frequency (100Hz) @@ -79,19 +80,29 @@ */ class Voice { public: + // Types + enum type_t { + type_normal, + type_release_trigger_required, ///< If the key of this voice will be released, it causes a release triggered voice to be spawned + type_release_trigger ///< Release triggered voice which cannot be killed by releasing its key + }; + // Attributes + type_t Type; ///< Voice Type int MIDIKey; ///< MIDI key number of the key that triggered the voice + uint KeyGroup; DiskThread* pDiskThread; ///< Pointer to the disk thread, to be able to order a disk stream and later to delete the stream again // Methods Voice(); ~Voice(); - void Kill(); + void Kill(Pool::Iterator& itKillEvent); + void KillImmediately(); void Render(uint Samples); void Reset(); void SetOutput(AudioOutputDevice* pAudioOutputDevice); void SetEngine(Engine* pEngine); - int Trigger(Event* pNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument); + int Trigger(Pool::Iterator& itNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument, int iLayer = 0, bool ReleaseTriggerVoice = false); inline bool IsActive() { return Active; } private: // Types @@ -104,15 +115,15 @@ // Attributes gig::Engine* pEngine; ///< Pointer to the sampler engine, to be able to access the event lists. float Volume; ///< Volume level of the voice - float* pOutputLeft; ///< Audio output channel buffer (left) - float* pOutputRight; ///< Audio output channel buffer (right) - uint SampleRate; ///< Sample rate of the engines output audio signal (in Hz) - uint MaxSamplesPerCycle; ///< Size of each audio output buffer + float PanLeft; + float PanRight; + float CrossfadeVolume; ///< Current attenuation level caused by a crossfade (only if a crossfade is defined of course) double Pos; ///< Current playback position in sample double PitchBase; ///< Basic pitch depth, stays the same for the whole life time of the voice double PitchBend; ///< Current pitch value of the pitchbend wheel ::gig::Sample* pSample; ///< Pointer to the sample to be played back ::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 bool Active; ///< If this voice object is currently in usage 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 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 @@ -130,41 +141,62 @@ midi_ctrl VCFResonanceCtrl; int FilterUpdateCounter; ///< Used to update filter parameters all FILTER_UPDATE_PERIOD samples static const float FILTER_CUTOFF_COEFF; + static const int FILTER_UPDATE_MASK; VCAManipulator* pVCAManipulator; VCFCManipulator* pVCFCManipulator; VCOManipulator* pVCOManipulator; LFO* pLFO1; ///< Low Frequency Oscillator 1 (Amplification) LFO* pLFO2; ///< Low Frequency Oscillator 2 (Filter cutoff frequency) LFO* pLFO3; ///< Low Frequency Oscillator 3 (Pitch) - 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::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). + Pool::Iterator itKillEvent; ///< Event which caused this voice to be killed // Static Methods static float CalculateFilterCutoffCoeff(); + static int CalculateFilterUpdateMask(); // Methods void ProcessEvents(uint Samples); - void Interpolate(uint Samples, sample_t* pSrc, uint Skip); + #if ENABLE_FILTER + void CalculateBiquadParameters(uint Samples); + #endif // ENABLE_FILTER + void InterpolateNoLoop(uint Samples, sample_t* pSrc, uint Skip); void InterpolateAndLoop(uint Samples, sample_t* pSrc, uint Skip); - inline void InterpolateOneStep_Stereo(sample_t* pSrc, int& i, float& effective_volume, float& pitch, float& cutoff, float& resonance) { + + inline void InterpolateMono(sample_t* pSrc, int& i) { + InterpolateOneStep_Mono(pSrc, i, + pEngine->pSynthesisParameters[Event::destination_vca][i] * PanLeft, + pEngine->pSynthesisParameters[Event::destination_vca][i] * PanRight, + pEngine->pSynthesisParameters[Event::destination_vco][i], + pEngine->pBasicFilterParameters[i], + pEngine->pMainFilterParameters[i]); + } + + inline void InterpolateStereo(sample_t* pSrc, int& i) { + InterpolateOneStep_Stereo(pSrc, i, + pEngine->pSynthesisParameters[Event::destination_vca][i] * PanLeft, + pEngine->pSynthesisParameters[Event::destination_vca][i] * PanRight, + pEngine->pSynthesisParameters[Event::destination_vco][i], + pEngine->pBasicFilterParameters[i], + pEngine->pMainFilterParameters[i]); + } + + inline void InterpolateOneStep_Stereo(sample_t* pSrc, int& i, float volume_left, float volume_right, 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 pos_int <<= 1; - #if ENABLE_FILTER - UpdateFilter_Stereo(cutoff + FILTER_CUTOFF_MIN, resonance); - #endif // ENABLE_FILTER - #if USE_LINEAR_INTERPOLATION #if ENABLE_FILTER // left channel - 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, volume_left * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int]))); // right channel - 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, volume_right * (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])); + pEngine->pOutputLeft[i] += volume_left * (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])); + pEngine->pOutputRight[i++] += volume_right * (pSrc[pos_int+1] + pos_fract * (pSrc[pos_int+3] - pSrc[pos_int+1])); #endif // ENABLE_FILTER #else // polynomial interpolation // calculate left channel @@ -172,13 +204,13 @@ 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; + float a = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f; + float b = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f; + float c = (x1 - xm1) * 0.5f; #if ENABLE_FILTER - 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, volume_left * ((((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); + pEngine->pOutputLeft[i] += volume_left * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0); #endif // ENABLE_FILTER //calculate right channel @@ -186,59 +218,54 @@ 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; + a = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f; + b = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f; + c = (x1 - xm1) * 0.5f; #if ENABLE_FILTER - 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, volume_right * ((((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); + pEngine->pOutputRight[i++] += volume_right * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0); #endif // ENABLE_FILTER #endif // USE_LINEAR_INTERPOLATION this->Pos += pitch; } - inline void InterpolateOneStep_Mono(sample_t* pSrc, int& i, float& effective_volume, float& pitch, float& cutoff, float& resonance) { + + inline void InterpolateOneStep_Mono(sample_t* pSrc, int& i, float volume_left, float volume_right, 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 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])); + float sample_point = 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); + float a = (3.0f * (x0 - x1) - xm1 + x2) * 0.5f; + float b = 2.0f * x1 + xm1 - (5.0f * x0 + x2) * 0.5f; + float c = (x1 - xm1) * 0.5f; + float sample_point = (((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0; #endif // USE_LINEAR_INTERPOLATION #if ENABLE_FILTER - sample_point = this->FilterLeft.Apply(sample_point); + sample_point = this->FilterLeft.Apply(&bq_base, &bq_main, sample_point); #endif // ENABLE_FILTER - pOutputLeft[i] += sample_point; - pOutputRight[i++] += sample_point; + pEngine->pOutputLeft[i] += sample_point * volume_left; + pEngine->pOutputRight[i++] += sample_point * volume_right; this->Pos += pitch; } - 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); - } + + inline float CrossfadeAttenuation(uint8_t& CrossfadeControllerValue) { + return (CrossfadeControllerValue <= pDimRgn->Crossfade.in_start) ? 0.0f + : (CrossfadeControllerValue < pDimRgn->Crossfade.in_end) ? float(CrossfadeControllerValue - pDimRgn->Crossfade.in_start) / float(pDimRgn->Crossfade.in_end - pDimRgn->Crossfade.in_start) + : (CrossfadeControllerValue <= pDimRgn->Crossfade.out_start) ? 1.0f + : (CrossfadeControllerValue < pDimRgn->Crossfade.out_end) ? float(CrossfadeControllerValue - pDimRgn->Crossfade.out_start) / float(pDimRgn->Crossfade.out_end - pDimRgn->Crossfade.out_start) + : 0.0f; } + inline float Constrain(float ValueToCheck, float Min, float Max) { if (ValueToCheck > Max) ValueToCheck = Max; else if (ValueToCheck < Min) ValueToCheck = Min;