--- linuxsampler/trunk/src/engines/gig/Voice.h 2004/09/15 13:59:08 242 +++ linuxsampler/trunk/src/engines/gig/Voice.h 2005/01/23 21:24:16 348 @@ -31,7 +31,7 @@ #include "../../common/RTMath.h" #include "../../common/RingBuffer.h" -#include "../../common/RTELMemoryPool.h" +#include "../../common/Pool.h" #include "../../drivers/audio/AudioOutputDevice.h" #include "../../lib/fileloader/libgig/gig.h" #include "../common/BiquadFilter.h" @@ -43,8 +43,6 @@ #include "Filter.h" #include "../common/LFO.h" -#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) @@ -92,43 +90,45 @@ 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 + RTList::Iterator itChildVoice; ///< Points to the next layer voice (if any). This field is currently only used by the voice stealing algorithm. // Methods Voice(); ~Voice(); - void Kill(Event* pKillEvent); - void KillImmediately(); + void Kill(Pool::Iterator& itKillEvent); void Render(uint Samples); void Reset(); void SetOutput(AudioOutputDevice* pAudioOutputDevice); void SetEngine(Engine* pEngine); - int Trigger(Event* pNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument, int iLayer = 0, bool ReleaseTriggerVoice = false); - inline bool IsActive() { return Active; } - private: + int Trigger(Pool::Iterator& itNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument, int iLayer, bool ReleaseTriggerVoice, bool VoiceStealing); + inline bool IsActive() { return PlaybackState; } + //private: // Types enum playback_state_t { - playback_state_ram, - playback_state_disk, - playback_state_end + playback_state_end = 0, + playback_state_ram = 1, + playback_state_disk = 2 }; // 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 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 + float PitchBase; ///< Basic pitch depth, stays the same for the whole life time of the voice + float 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 Stream::reference_t DiskStreamRef; ///< Reference / link to the disk stream + int RealSampleWordsLeftToRead; ///< Number of samples left to read, not including the silence added for the interpolator 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. 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 - 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 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 EGADSR* pEG1; ///< Envelope Generator 1 (Amplification) EGADSR* pEG2; ///< Envelope Generator 2 (Filter cutoff frequency) @@ -146,103 +146,31 @@ 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). - Event* pKillEvent; ///< Event which caused this voice to be killed + 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). + //public: // FIXME: just made public for debugging (sanity check in Engine::RenderAudio()), should be changed to private before the final release + Pool::Iterator itKillEvent; ///< Event which caused this voice to be killed + //private: + int SynthesisMode; // Static Methods static float CalculateFilterCutoffCoeff(); static int CalculateFilterUpdateMask(); // Methods - void ProcessEvents(uint Samples); - #if ENABLE_FILTER - void CalculateBiquadParameters(uint Samples); - #endif // ENABLE_FILTER - void Interpolate(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, 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 USE_LINEAR_INTERPOLATION - #if ENABLE_FILTER - // left channel - pEngine->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 - 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]))); - #else // no filter - // left channel - pEngine->pOutputLeft[i] += effective_volume * (pSrc[pos_int] + pos_fract * (pSrc[pos_int+2] - pSrc[pos_int])); - // right channel - pEngine->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 = (3.0f * (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 - pEngine->pOutputLeft[i] += this->FilterLeft.Apply(&bq_base, &bq_main, effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0)); - #else // no filter - pEngine->pOutputLeft[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 = (3.0f * (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 - pEngine->pOutputRight[i++] += this->FilterRight.Apply(&bq_base, &bq_main, effective_volume * ((((a * pos_fract) + b) * pos_fract + c) * pos_fract + x0)); - #else // no filter - pEngine->pOutputRight[i++] += effective_volume * ((((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, 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 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 = (3.0f * (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 = 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 - - pEngine->pOutputLeft[i] += sample_point; - pEngine->pOutputRight[i++] += sample_point; - - this->Pos += pitch; - } + void KillImmediately(); + void ProcessEvents(uint Samples); + void CalculateBiquadParameters(uint Samples); + void Synthesize(uint Samples, sample_t* pSrc, uint Skip); 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; + float att = (!pDimRgn->Crossfade.out_end) ? CrossfadeControllerValue / 127.0f /* 0,0,0,0 means no crossfade defined */ + : (CrossfadeControllerValue < pDimRgn->Crossfade.in_end) ? + ((CrossfadeControllerValue <= pDimRgn->Crossfade.in_start) ? 0.0f + : 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(pDimRgn->Crossfade.out_end - CrossfadeControllerValue) / float(pDimRgn->Crossfade.out_end - pDimRgn->Crossfade.out_start) + : 0.0f; + return pDimRgn->InvertAttenuationController ? 1 - att : att; } inline float Constrain(float ValueToCheck, float Min, float Max) {