/*************************************************************************** * * * LinuxSampler - modular, streaming capable sampler * * * * Copyright (C) 2003 by Benno Senoner and Christian Schoenebeck * * * * This program is free software; you can redistribute it and/or modify * * it under the terms of the GNU General Public License as published by * * the Free Software Foundation; either version 2 of the License, or * * (at your option) any later version. * * * * This program is distributed in the hope that it will be useful, * * but WITHOUT ANY WARRANTY; without even the implied warranty of * * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * * GNU General Public License for more details. * * * * You should have received a copy of the GNU General Public License * * along with this program; if not, write to the Free Software * * Foundation, Inc., 59 Temple Place, Suite 330, Boston, * * MA 02111-1307 USA * ***************************************************************************/ #include "voice.h" // FIXME: no support for layers (nor crossfades) yet DiskThread* Voice::pDiskThread = NULL; AudioThread* Voice::pEngine = NULL; Voice::Voice() { Active = false; } Voice::~Voice() { } /** * Initializes and triggers the voice, a disk stream will be launched if * needed. * * @param pNoteOnEvent - event that caused triggering of this voice * @param Pitch - MIDI detune factor (-8192 ... +8191) * @param pInstrument - points to the loaded instrument which provides sample wave(s) and articulation data * @returns 0 on success, a value < 0 if something failed */ int Voice::Trigger(ModulationSystem::Event* pNoteOnEvent, int Pitch, gig::Instrument* pInstrument) { Active = true; MIDIKey = pNoteOnEvent->Key; pRegion = pInstrument->GetRegion(MIDIKey); PlaybackState = playback_state_ram; // we always start playback from RAM cache and switch then to disk if needed Pos = 0; ReleaseVelocity = 127; // default release velocity value Delay = pNoteOnEvent->FragmentPos(); pTriggerEvent = pNoteOnEvent; if (!pRegion) { std::cerr << "Audio Thread: No Region defined for MIDI key " << MIDIKey << std::endl << std::flush; Kill(); return -1; } //TODO: current MIDI controller values are not taken into account yet gig::DimensionRegion* pDimRgn = NULL; for (int i = pRegion->Dimensions - 1; i >= 0; i--) { // Check if instrument has a velocity split if (pRegion->pDimensionDefinitions[i].dimension == gig::dimension_velocity) { uint DimValues[5] = {0,0,0,0,0}; DimValues[i] = pNoteOnEvent->Velocity; pDimRgn = pRegion->GetDimensionRegionByValue(DimValues[4],DimValues[3],DimValues[2],DimValues[1],DimValues[0]); break; } } if (!pDimRgn) { // if there was no velocity split pDimRgn = pRegion->GetDimensionRegionByValue(0,0,0,0,0); } pSample = pDimRgn->pSample; // sample won't change until the voice is finished // Check if the sample needs disk streaming or is too short for that long cachedsamples = pSample->GetCache().Size / pSample->FrameSize; DiskVoice = cachedsamples < pSample->SamplesTotal; if (DiskVoice) { // voice to be streamed from disk MaxRAMPos = cachedsamples - (MaxSamplesPerCycle << MAX_PITCH) / pSample->Channels; //TODO: this calculation is too pessimistic and may better be moved to Render() method, so it calculates MaxRAMPos dependent to the current demand of sample points to be rendered (e.g. in case of JACK) // check if there's a loop defined which completely fits into the cached (RAM) part of the sample if (pSample->Loops && pSample->LoopEnd <= MaxRAMPos) { RAMLoop = true; LoopCyclesLeft = pSample->LoopPlayCount; } else RAMLoop = false; if (pDiskThread->OrderNewStream(&DiskStreamRef, pSample, MaxRAMPos, !RAMLoop) < 0) { dmsg(1,("Disk stream order failed!\n")); Kill(); return -1; } dmsg(4,("Disk voice launched (cached samples: %d, total Samples: %d, MaxRAMPos: %d, RAMLooping: %s)\n", cachedsamples, pSample->SamplesTotal, MaxRAMPos, (RAMLoop) ? "yes" : "no")); } else { // RAM only voice MaxRAMPos = cachedsamples; if (pSample->Loops) { RAMLoop = true; LoopCyclesLeft = pSample->LoopPlayCount; } else RAMLoop = false; dmsg(4,("RAM only voice launched (Looping: %s)\n", (RAMLoop) ? "yes" : "no")); } // Pitch according to keyboard position (if 'PitchTrack' is set) and given detune factor this->Pitch = ((double) Pitch / 8192.0) / 12.0 + ((pDimRgn->PitchTrack) ? pow(2, ((double) (MIDIKey - (int) pDimRgn->UnityNote) + (double) pDimRgn->FineTune / 100.0) / 12.0) : pow(2, ((double) pDimRgn->FineTune / 100.0) / 12.0)); Volume = pDimRgn->GetVelocityAttenuation(pNoteOnEvent->Velocity); // get current value of EG1 controller double eg1controllervalue; switch (pDimRgn->EG1Controller.type) { case gig::eg1_ctrl_t::type_none: // no controller defined eg1controllervalue = 0; break; case gig::eg1_ctrl_t::type_channelaftertouch: eg1controllervalue = 0; // TODO: aftertouch not yet supported break; case gig::eg1_ctrl_t::type_velocity: eg1controllervalue = pNoteOnEvent->Velocity; break; case gig::eg1_ctrl_t::type_controlchange: // MIDI control change controller eg1controllervalue = pEngine->ControllerTable[pDimRgn->EG1Controller.controller_number]; break; } if (pDimRgn->EG1ControllerInvert) eg1controllervalue = 127 - eg1controllervalue; // calculate influence of EG1 controller on EG1's parameters (TODO: needs to be fine tuned) double eg1attack = (pDimRgn->EG1ControllerAttackInfluence) ? 0.0001 * (double) (1 << pDimRgn->EG1ControllerAttackInfluence) * eg1controllervalue : 0.0; double eg1decay = (pDimRgn->EG1ControllerDecayInfluence) ? 0.0001 * (double) (1 << pDimRgn->EG1ControllerDecayInfluence) * eg1controllervalue : 0.0; double eg1release = (pDimRgn->EG1ControllerReleaseInfluence) ? 0.0001 * (double) (1 << pDimRgn->EG1ControllerReleaseInfluence) * eg1controllervalue : 0.0; EG1.Trigger(pDimRgn->EG1PreAttack, pDimRgn->EG1Attack + eg1attack, pDimRgn->EG1Hold, pSample->LoopStart, pDimRgn->EG1Decay1 + eg1decay, pDimRgn->EG1Decay2 + eg1decay, pDimRgn->EG1InfiniteSustain, pDimRgn->EG1Sustain, pDimRgn->EG1Release + eg1release, Delay); #if ENABLE_FILTER #if FORCE_FILTER_USAGE FilterLeft.Enabled = FilterRight.Enabled = true; #else // use filter only if instrument file told so FilterLeft.Enabled = FilterRight.Enabled = pDimRgn->VCFEnabled; #endif // FORCE_FILTER_USAGE if (pDimRgn->VCFEnabled) { #ifdef OVERRIDE_FILTER_CUTOFF_CTRL VCFCutoffCtrl.controller = OVERRIDE_FILTER_CUTOFF_CTRL; #else // use the one defined in the instrument file switch (pDimRgn->VCFCutoffController) { case gig::vcf_cutoff_ctrl_modwheel: VCFCutoffCtrl.controller = 1; break; case gig::vcf_cutoff_ctrl_effect1: VCFCutoffCtrl.controller = 12; break; case gig::vcf_cutoff_ctrl_effect2: VCFCutoffCtrl.controller = 13; break; case gig::vcf_cutoff_ctrl_breath: VCFCutoffCtrl.controller = 2; break; case gig::vcf_cutoff_ctrl_foot: VCFCutoffCtrl.controller = 4; break; case gig::vcf_cutoff_ctrl_sustainpedal: VCFCutoffCtrl.controller = 64; break; case gig::vcf_cutoff_ctrl_softpedal: VCFCutoffCtrl.controller = 67; break; case gig::vcf_cutoff_ctrl_genpurpose7: VCFCutoffCtrl.controller = 82; break; case gig::vcf_cutoff_ctrl_genpurpose8: VCFCutoffCtrl.controller = 83; break; case gig::vcf_cutoff_ctrl_aftertouch: //TODO: not implemented yet case gig::vcf_cutoff_ctrl_none: default: VCFCutoffCtrl.controller = 0; break; } #endif // OVERRIDE_FILTER_CUTOFF_CTRL #ifdef OVERRIDE_FILTER_RES_CTRL VCFResonanceCtrl.controller = OVERRIDE_FILTER_RES_CTRL; #else // use the one defined in the instrument file switch (pDimRgn->VCFResonanceController) { case gig::vcf_res_ctrl_genpurpose3: VCFResonanceCtrl.controller = 18; break; case gig::vcf_res_ctrl_genpurpose4: VCFResonanceCtrl.controller = 19; break; case gig::vcf_res_ctrl_genpurpose5: VCFResonanceCtrl.controller = 80; break; case gig::vcf_res_ctrl_genpurpose6: VCFResonanceCtrl.controller = 81; break; case gig::vcf_res_ctrl_none: default: VCFResonanceCtrl.controller = 0; } #endif // OVERRIDE_FILTER_RES_CTRL #ifndef OVERRIDE_FILTER_TYPE FilterLeft.SetType(pDimRgn->VCFType); FilterRight.SetType(pDimRgn->VCFType); #else // override filter type FilterLeft.SetType(OVERRIDE_FILTER_TYPE); FilterRight.SetType(OVERRIDE_FILTER_TYPE); #endif // OVERRIDE_FILTER_TYPE VCFCutoffCtrl.value = pEngine->ControllerTable[VCFCutoffCtrl.controller]; VCFResonanceCtrl.value = pEngine->ControllerTable[VCFResonanceCtrl.controller]; // calculate cutoff frequency float cutoff = (!VCFCutoffCtrl.controller && pDimRgn->VCFVelocityScale) ? (float) pNoteOnEvent->Velocity * pDimRgn->VCFVelocityScale * 0.31f + 20.0f // 20Hz..5kHz : (float) (127 - VCFCutoffCtrl.value) * 39.4f + 20.0f; // 20Hz..5kHz (inverted) // calculate resonance float resonance = (float) VCFResonanceCtrl.value * 0.00787f; // 0.0..1.0 if (pDimRgn->VCFKeyboardTracking) { resonance += (float) (pNoteOnEvent->Key - pDimRgn->VCFKeyboardTrackingBreakpoint) * 0.00787f; } Constrain(resonance, 0.0, 1.0); // correct resonance if outside allowed value range (0.0..1.0) FilterLeft.SetParameters(cutoff, resonance, ModulationSystem::SampleRate()); FilterRight.SetParameters(cutoff, resonance, ModulationSystem::SampleRate()); } else { VCFCutoffCtrl.controller = 0; VCFResonanceCtrl.controller = 0; } #endif // ENABLE_FILTER // ************************************************ // TODO: ARTICULATION DATA HANDLING IS MISSING HERE // ************************************************ return 0; // success } /** * Renders the audio data for this voice for the current audio fragment. * The sample input data can either come from RAM (cached sample or sample * part) or directly from disk. The output signal will be rendered by * resampling / interpolation. If this voice is a disk streaming voice and * the voice completely played back the cached RAM part of the sample, it * will automatically switch to disk playback for the next RenderAudio() * call. * * @param Samples - number of samples to be rendered in this audio fragment cycle */ void Voice::Render(uint Samples) { // Reset the synthesis parameter matrix ModulationSystem::ResetDestinationParameter(ModulationSystem::destination_vca, this->Volume); ModulationSystem::ResetDestinationParameter(ModulationSystem::destination_vco, this->Pitch); #if ENABLE_FILTER ModulationSystem::ResetDestinationParameter(ModulationSystem::destination_vcfc, FilterLeft.Cutoff()); // intialize with last cutoff value from previous render cycle ModulationSystem::ResetDestinationParameter(ModulationSystem::destination_vcfr, FilterLeft.Resonance()); // intialize with last resonance value from previous render cycle // Reset synthesis event lists (except VCO, as VCO events apply channel wide currently) pEngine->pSynthesisEvents[ModulationSystem::destination_vcfc]->clear(); pEngine->pSynthesisEvents[ModulationSystem::destination_vcfr]->clear(); #endif // ENABLE_FILTER // Apply events to the synthesis parameter matrix ProcessEvents(Samples); // Let all modulators write their parameter changes to the synthesis parameter matrix for the current audio fragment EG1.Process(Samples, pEngine->pMIDIKeyInfo[MIDIKey].pEvents, pTriggerEvent, this->Pos, this->Pitch); switch (this->PlaybackState) { case playback_state_ram: { if (RAMLoop) InterpolateAndLoop(Samples, (sample_t*) pSample->GetCache().pStart, Delay); else Interpolate(Samples, (sample_t*) pSample->GetCache().pStart, Delay); if (DiskVoice) { // check if we reached the allowed limit of the sample RAM cache if (Pos > MaxRAMPos) { dmsg(5,("Voice: switching to disk playback (Pos=%f)\n", Pos)); this->PlaybackState = playback_state_disk; } } else if (Pos >= pSample->GetCache().Size / pSample->FrameSize) { this->PlaybackState = playback_state_end; } } break; case playback_state_disk: { if (!DiskStreamRef.pStream) { // check if the disk thread created our ordered disk stream in the meantime DiskStreamRef.pStream = pDiskThread->AskForCreatedStream(DiskStreamRef.OrderID); if (!DiskStreamRef.pStream) { std::cout << stderr << "Disk stream not available in time!" << std::endl << std::flush; Kill(); return; } DiskStreamRef.pStream->IncrementReadPos(pSample->Channels * (double_to_int(Pos) - MaxRAMPos)); Pos -= double_to_int(Pos); } // add silence sample at the end if we reached the end of the stream (for the interpolator) if (DiskStreamRef.State == Stream::state_end && DiskStreamRef.pStream->GetReadSpace() < (MaxSamplesPerCycle << MAX_PITCH) / pSample->Channels) { DiskStreamRef.pStream->WriteSilence((MaxSamplesPerCycle << MAX_PITCH) / pSample->Channels); this->PlaybackState = playback_state_end; } sample_t* ptr = DiskStreamRef.pStream->GetReadPtr(); // get the current read_ptr within the ringbuffer where we read the samples from Interpolate(Samples, ptr, Delay); DiskStreamRef.pStream->IncrementReadPos(double_to_int(Pos) * pSample->Channels); Pos -= double_to_int(Pos); } break; case playback_state_end: Kill(); // free voice break; } // Reset delay Delay = 0; pTriggerEvent = NULL; // If release stage finished, let the voice be killed if (EG1.GetStage() == EG_VCA::stage_end) this->PlaybackState = playback_state_end; } /** * Resets voice variables. Should only be called if rendering process is * suspended / not running. */ void Voice::Reset() { DiskStreamRef.pStream = NULL; DiskStreamRef.hStream = 0; DiskStreamRef.State = Stream::state_unused; DiskStreamRef.OrderID = 0; Active = false; } /** * Process the control change event lists of the engine for the current * audio fragment. Event values will be applied to the synthesis parameter * matrix. * * @param Samples - number of samples to be rendered in this audio fragment cycle */ void Voice::ProcessEvents(uint Samples) { #if ENABLE_FILTER // dispatch control change events ModulationSystem::Event* pCCEvent = pEngine->pCCEvents->first(); while (pCCEvent) { if (pCCEvent->Controller == VCFCutoffCtrl.controller) { pEngine->pSynthesisEvents[ModulationSystem::destination_vcfc]->alloc_assign(*pCCEvent); } if (pCCEvent->Controller == VCFResonanceCtrl.controller) { pEngine->pSynthesisEvents[ModulationSystem::destination_vcfr]->alloc_assign(*pCCEvent); } pCCEvent = pEngine->pCCEvents->next(); } #endif // ENABLE_FILTER // process pitch events RTEList* pVCOEventList = pEngine->pSynthesisEvents[ModulationSystem::destination_vco]; ModulationSystem::Event* pVCOEvent = pVCOEventList->first(); while (pVCOEvent) { ModulationSystem::Event* pNextVCOEvent = pVCOEventList->next(); // calculate the influence length of this event (in sample points) uint end = (pNextVCOEvent) ? pNextVCOEvent->FragmentPos() : Samples; this->Pitch += ((double) pVCOEvent->Pitch / 8192.0) / 12.0; // +- one semitone // apply pitch value to the pitch parameter sequence for (uint i = pVCOEvent->FragmentPos(); i < end; i++) { ModulationSystem::pDestinationParameter[ModulationSystem::destination_vco][i] = this->Pitch; } pVCOEvent = pNextVCOEvent; } #if ENABLE_FILTER // process filter cutoff events RTEList* pCutoffEventList = pEngine->pSynthesisEvents[ModulationSystem::destination_vcfc]; ModulationSystem::Event* pCutoffEvent = pCutoffEventList->first(); while (pCutoffEvent) { ModulationSystem::Event* pNextCutoffEvent = pCutoffEventList->next(); // calculate the influence length of this event (in sample points) uint end = (pNextCutoffEvent) ? pNextCutoffEvent->FragmentPos() : Samples; // convert absolute controller value to differential int ctrldelta = pCutoffEvent->Value - VCFCutoffCtrl.value; VCFCutoffCtrl.value = pCutoffEvent->Value; float cutoffdelta = (float) ctrldelta * -39.4f; // (20Hz)..5kHz (inverted) // apply cutoff frequency to the cutoff parameter sequence for (uint i = pCutoffEvent->FragmentPos(); i < end; i++) { ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfc][i] += cutoffdelta; } pCutoffEvent = pNextCutoffEvent; } // process filter resonance events RTEList* pResonanceEventList = pEngine->pSynthesisEvents[ModulationSystem::destination_vcfr]; ModulationSystem::Event* pResonanceEvent = pResonanceEventList->first(); while (pResonanceEvent) { ModulationSystem::Event* pNextResonanceEvent = pResonanceEventList->next(); // calculate the influence length of this event (in sample points) uint end = (pNextResonanceEvent) ? pNextResonanceEvent->FragmentPos() : Samples; // convert absolute controller value to differential int ctrldelta = pResonanceEvent->Value - VCFResonanceCtrl.value; VCFResonanceCtrl.value = pResonanceEvent->Value; float resonancedelta = (float) ctrldelta * 0.00787f; // 0.0..1.0 // apply cutoff frequency to the cutoff parameter sequence for (uint i = pResonanceEvent->FragmentPos(); i < end; i++) { ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfr][i] += resonancedelta; } pResonanceEvent = pNextResonanceEvent; } #endif // ENABLE_FILTER } /** * Interpolates the input audio data (no loop). * * @param Samples - number of sample points to be rendered in this audio * fragment cycle * @param pSrc - pointer to input sample data * @param Skip - number of sample points to skip in output buffer */ void Voice::Interpolate(uint Samples, sample_t* pSrc, uint Skip) { int i = Skip; // FIXME: assuming either mono or stereo if (this->pSample->Channels == 2) { // Stereo Sample while (i < Samples) { InterpolateOneStep_Stereo(pSrc, i, ModulationSystem::pDestinationParameter[ModulationSystem::destination_vca][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vco][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfc][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfr][i]); } #if ENABLE_FILTER // to save the last filter setting for the next render cycle (only needed when we update the filter not-sample-accurate) ForceUpdateFilter_Stereo(ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfc][Samples - 1], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfr][Samples - 1]); #endif // ENABLE_FILTER } else { // Mono Sample while (i < Samples) { InterpolateOneStep_Mono(pSrc, i, ModulationSystem::pDestinationParameter[ModulationSystem::destination_vca][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vco][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfc][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfr][i]); } #if ENABLE_FILTER // to save the last filter setting for the next render cycle (only needed when we update the filter not-sample-accurate) ForceUpdateFilter_Mono(ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfc][Samples - 1], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfr][Samples - 1]); #endif // ENABLE_FILTER } } /** * Interpolates the input audio data, this method honors looping. * * @param Samples - number of sample points to be rendered in this audio * fragment cycle * @param pSrc - pointer to input sample data * @param Skip - number of sample points to skip in output buffer */ void Voice::InterpolateAndLoop(uint Samples, sample_t* pSrc, uint Skip) { int i = Skip; // FIXME: assuming either mono or stereo if (pSample->Channels == 2) { // Stereo Sample if (pSample->LoopPlayCount) { // render loop (loop count limited) while (i < Samples && LoopCyclesLeft) { InterpolateOneStep_Stereo(pSrc, i, ModulationSystem::pDestinationParameter[ModulationSystem::destination_vca][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vco][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfc][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfr][i]); if (Pos > pSample->LoopEnd) { Pos = pSample->LoopStart + fmod(Pos - pSample->LoopEnd, pSample->LoopSize);; LoopCyclesLeft--; } } // render on without loop while (i < Samples) { InterpolateOneStep_Stereo(pSrc, i, ModulationSystem::pDestinationParameter[ModulationSystem::destination_vca][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vco][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfc][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfr][i]); } } else { // render loop (endless loop) while (i < Samples) { InterpolateOneStep_Stereo(pSrc, i, ModulationSystem::pDestinationParameter[ModulationSystem::destination_vca][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vco][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfc][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfr][i]); if (Pos > pSample->LoopEnd) { Pos = pSample->LoopStart + fmod(Pos - pSample->LoopEnd, pSample->LoopSize); } } } #if ENABLE_FILTER // to save the last filter setting for the next render cycle (only needed when we update the filter not-sample-accurate) ForceUpdateFilter_Stereo(ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfc][Samples - 1], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfr][Samples - 1]); #endif // ENABLE_FILTER } else { // Mono Sample if (pSample->LoopPlayCount) { // render loop (loop count limited) while (i < Samples && LoopCyclesLeft) { InterpolateOneStep_Mono(pSrc, i, ModulationSystem::pDestinationParameter[ModulationSystem::destination_vca][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vco][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfc][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfr][i]); if (Pos > pSample->LoopEnd) { Pos = pSample->LoopStart + fmod(Pos - pSample->LoopEnd, pSample->LoopSize);; LoopCyclesLeft--; } } // render on without loop while (i < Samples) { InterpolateOneStep_Mono(pSrc, i, ModulationSystem::pDestinationParameter[ModulationSystem::destination_vca][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vco][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfc][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfr][i]); } } else { // render loop (endless loop) while (i < Samples) { InterpolateOneStep_Mono(pSrc, i, ModulationSystem::pDestinationParameter[ModulationSystem::destination_vca][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vco][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfc][i], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfr][i]); if (Pos > pSample->LoopEnd) { Pos = pSample->LoopStart + fmod(Pos - pSample->LoopEnd, pSample->LoopSize);; } } } #if ENABLE_FILTER // to save the last filter setting for the next render cycle (only needed when we update the filter not-sample-accurate) ForceUpdateFilter_Mono(ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfc][Samples - 1], ModulationSystem::pDestinationParameter[ModulationSystem::destination_vcfr][Samples - 1]); #endif // ENABLE_FILTER } } /** * Immediately kill the voice. */ void Voice::Kill() { if (DiskVoice && DiskStreamRef.State != Stream::state_unused) { pDiskThread->OrderDeletionOfStream(&DiskStreamRef); } Reset(); }