--- linuxsampler/trunk/src/engines/gig/Voice.cpp 2004/09/18 14:12:36 245 +++ linuxsampler/trunk/src/engines/gig/Voice.cpp 2005/02/11 13:13:54 368 @@ -22,6 +22,8 @@ #include "EGADSR.h" #include "Manipulator.h" +#include "../../common/Features.h" +#include "Synthesizer.h" #include "Voice.h" @@ -45,7 +47,7 @@ Voice::Voice() { pEngine = NULL; pDiskThread = NULL; - Active = false; + PlaybackState = playback_state_end; pEG1 = NULL; pEG2 = NULL; pEG3 = NULL; @@ -56,6 +58,17 @@ pLFO2 = NULL; pLFO3 = NULL; KeyGroup = 0; + SynthesisMode = 0; // set all mode bits to 0 first + // select synthesis implementation (currently either pure C++ or MMX+SSE(1)) + #if ARCH_X86 + SYNTHESIS_MODE_SET_IMPLEMENTATION(SynthesisMode, Features::supportsMMX() && Features::supportsSSE()); + #else + SYNTHESIS_MODE_SET_IMPLEMENTATION(SynthesisMode, false); + #endif + SYNTHESIS_MODE_SET_PROFILING(SynthesisMode, true); + + FilterLeft.Reset(); + FilterRight.Reset(); } Voice::~Voice() { @@ -103,31 +116,36 @@ * Initializes and triggers the voice, a disk stream will be launched if * needed. * - * @param pNoteOnEvent - event that caused triggering of this voice + * @param itNoteOnEvent - event that caused triggering of this voice * @param PitchBend - MIDI detune factor (-8192 ... +8191) * @param pInstrument - points to the loaded instrument which provides sample wave(s) and articulation data * @param iLayer - layer number this voice refers to (only if this is a layered sound of course) * @param ReleaseTriggerVoice - if this new voice is a release trigger voice (optional, default = false) - * @returns 0 on success, a value < 0 if something failed + * @param VoiceStealing - wether the voice is allowed to steal voices for further subvoices + * @returns 0 on success, a value < 0 if the voice wasn't triggered + * (either due to an error or e.g. because no region is + * defined for the given key) */ - int Voice::Trigger(Event* pNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument, int iLayer, bool ReleaseTriggerVoice) { + int Voice::Trigger(Pool::Iterator& itNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument, int iLayer, bool ReleaseTriggerVoice, bool VoiceStealing) { if (!pInstrument) { dmsg(1,("voice::trigger: !pInstrument\n")); exit(EXIT_FAILURE); } + if (itNoteOnEvent->FragmentPos() > pEngine->MaxSamplesPerCycle) { // FIXME: should be removed before the final release (purpose: just a sanity check for debugging) + dmsg(1,("Voice::Trigger(): ERROR, TriggerDelay > Totalsamples\n")); + } Type = type_normal; - Active = true; - MIDIKey = pNoteOnEvent->Key; + MIDIKey = itNoteOnEvent->Param.Note.Key; pRegion = pInstrument->GetRegion(MIDIKey); PlaybackState = playback_state_ram; // we always start playback from RAM cache and switch then to disk if needed - Delay = pNoteOnEvent->FragmentPos(); - pTriggerEvent = pNoteOnEvent; - pKillEvent = NULL; + Delay = itNoteOnEvent->FragmentPos(); + itTriggerEvent = itNoteOnEvent; + itKillEvent = Pool::Iterator(); + itChildVoice = Pool::Iterator(); if (!pRegion) { - std::cerr << "gig::Voice: No Region defined for MIDI key " << MIDIKey << std::endl << std::flush; - KillImmediately(); + dmsg(4, ("gig::Voice: No Region defined for MIDI key %d\n", MIDIKey)); return -1; } @@ -135,7 +153,7 @@ // get current dimension values to select the right dimension region //FIXME: controller values for selecting the dimension region here are currently not sample accurate - uint DimValues[5] = {0,0,0,0,0}; + uint DimValues[8] = { 0 }; for (int i = pRegion->Dimensions - 1; i >= 0; i--) { switch (pRegion->pDimensionDefinitions[i].dimension) { case ::gig::dimension_samplechannel: @@ -146,10 +164,10 @@ // if this is the 1st layer then spawn further voices for all the other layers if (iLayer == 0) for (int iNewLayer = 1; iNewLayer < pRegion->pDimensionDefinitions[i].zones; iNewLayer++) - pEngine->LaunchVoice(pNoteOnEvent, iNewLayer, ReleaseTriggerVoice); + itChildVoice = pEngine->LaunchVoice(itNoteOnEvent, iNewLayer, ReleaseTriggerVoice, VoiceStealing); break; case ::gig::dimension_velocity: - DimValues[i] = pNoteOnEvent->Velocity; + DimValues[i] = itNoteOnEvent->Param.Note.Velocity; break; case ::gig::dimension_channelaftertouch: DimValues[i] = 0; //TODO: we currently ignore this dimension @@ -159,7 +177,7 @@ DimValues[i] = (uint) ReleaseTriggerVoice; break; case ::gig::dimension_keyboard: - DimValues[i] = (uint) pNoteOnEvent->Key; + DimValues[i] = (uint) pEngine->CurrentKeyDimension; break; case ::gig::dimension_modwheel: DimValues[i] = pEngine->ControllerTable[1]; @@ -237,7 +255,13 @@ std::cerr << "gig::Voice::Trigger() Error: Unknown dimension\n" << std::flush; } } - pDimRgn = pRegion->GetDimensionRegionByValue(DimValues[4],DimValues[3],DimValues[2],DimValues[1],DimValues[0]); + pDimRgn = pRegion->GetDimensionRegionByValue(DimValues); + + pSample = pDimRgn->pSample; // sample won't change until the voice is finished + if (!pSample || !pSample->SamplesTotal) return -1; // no need to continue if sample is silent + + // select channel mode (mono or stereo) + SYNTHESIS_MODE_SET_CHANNELS(SynthesisMode, pSample->Channels == 2); // get starting crossfade volume level switch (pDimRgn->AttenuationController.type) { @@ -245,7 +269,7 @@ CrossfadeVolume = 1.0f; //TODO: aftertouch not supported yet break; case ::gig::attenuation_ctrl_t::type_velocity: - CrossfadeVolume = CrossfadeAttenuation(pNoteOnEvent->Velocity); + CrossfadeVolume = CrossfadeAttenuation(itNoteOnEvent->Param.Note.Velocity); break; case ::gig::attenuation_ctrl_t::type_controlchange: //FIXME: currently not sample accurate CrossfadeVolume = CrossfadeAttenuation(pEngine->ControllerTable[pDimRgn->AttenuationController.controller_number]); @@ -255,10 +279,8 @@ CrossfadeVolume = 1.0f; } - PanLeft = float(RTMath::Max(pDimRgn->Pan, 0)) / -64.0f; - PanRight = float(RTMath::Min(pDimRgn->Pan, 0)) / 63.0f; - - pSample = pDimRgn->pSample; // sample won't change until the voice is finished + PanLeft = 1.0f - float(RTMath::Max(pDimRgn->Pan, 0)) / 63.0f; + PanRight = 1.0f - float(RTMath::Min(pDimRgn->Pan, 0)) / -64.0f; Pos = pDimRgn->SampleStartOffset; // offset where we should start playback of sample (0 - 2000 sample points) @@ -296,15 +318,13 @@ // calculate initial pitch value { - double pitchbasecents = pDimRgn->FineTune * 10 + (int) pEngine->ScaleTuning[MIDIKey % 12]; + double pitchbasecents = pDimRgn->FineTune + (int) pEngine->ScaleTuning[MIDIKey % 12]; if (pDimRgn->PitchTrack) pitchbasecents += (MIDIKey - (int) pDimRgn->UnityNote) * 100; this->PitchBase = RTMath::CentsToFreqRatio(pitchbasecents) * (double(pSample->SamplesPerSecond) / double(pEngine->pAudioOutputDevice->SampleRate())); this->PitchBend = RTMath::CentsToFreqRatio(((double) PitchBend / 8192.0) * 200.0); // pitchbend wheel +-2 semitones = 200 cents } - - Volume = pDimRgn->GetVelocityAttenuation(pNoteOnEvent->Velocity) / 32768.0f; // we downscale by 32768 to convert from int16 value range to DSP value range (which is -1.0..1.0) - + Volume = pDimRgn->GetVelocityAttenuation(itNoteOnEvent->Param.Note.Velocity) / 32768.0f; // we downscale by 32768 to convert from int16 value range to DSP value range (which is -1.0..1.0) // setup EG 1 (VCA EG) { @@ -318,7 +338,7 @@ eg1controllervalue = 0; // TODO: aftertouch not yet supported break; case ::gig::eg1_ctrl_t::type_velocity: - eg1controllervalue = pNoteOnEvent->Velocity; + eg1controllervalue = itNoteOnEvent->Param.Note.Velocity; break; case ::gig::eg1_ctrl_t::type_controlchange: // MIDI control change controller eg1controllervalue = pEngine->ControllerTable[pDimRgn->EG1Controller.controller_number]; @@ -344,7 +364,6 @@ } - #if ENABLE_FILTER // setup EG 2 (VCF Cutoff EG) { // get current value of EG2 controller @@ -357,7 +376,7 @@ eg2controllervalue = 0; // TODO: aftertouch not yet supported break; case ::gig::eg2_ctrl_t::type_velocity: - eg2controllervalue = pNoteOnEvent->Velocity; + eg2controllervalue = itNoteOnEvent->Param.Note.Velocity; break; case ::gig::eg2_ctrl_t::type_controlchange: // MIDI control change controller eg2controllervalue = pEngine->ControllerTable[pDimRgn->EG2Controller.controller_number]; @@ -381,7 +400,6 @@ pDimRgn->EG2Release + eg2release, Delay); } - #endif // ENABLE_FILTER // setup EG 3 (VCO EG) @@ -428,7 +446,7 @@ Delay); } - #if ENABLE_FILTER + // setup LFO 2 (VCF Cutoff LFO) { uint16_t lfo2_internal_depth; @@ -465,7 +483,7 @@ pEngine->SampleRate, Delay); } - #endif // ENABLE_FILTER + // setup LFO 3 (VCO LFO) { @@ -504,13 +522,14 @@ Delay); } - #if ENABLE_FILTER + #if FORCE_FILTER_USAGE - FilterLeft.Enabled = FilterRight.Enabled = true; + const bool bUseFilter = true; #else // use filter only if instrument file told so - FilterLeft.Enabled = FilterRight.Enabled = pDimRgn->VCFEnabled; + const bool bUseFilter = pDimRgn->VCFEnabled; #endif // FORCE_FILTER_USAGE - if (pDimRgn->VCFEnabled) { + SYNTHESIS_MODE_SET_FILTER(SynthesisMode, bUseFilter); + if (bUseFilter) { #ifdef OVERRIDE_FILTER_CUTOFF_CTRL VCFCutoffCtrl.controller = OVERRIDE_FILTER_CUTOFF_CTRL; #else // use the one defined in the instrument file @@ -585,29 +604,25 @@ // calculate cutoff frequency float cutoff = (!VCFCutoffCtrl.controller) - ? exp((float) (127 - pNoteOnEvent->Velocity) * (float) pDimRgn->VCFVelocityScale * 6.2E-5f * FILTER_CUTOFF_COEFF) * FILTER_CUTOFF_MAX + ? exp((float) (127 - itNoteOnEvent->Param.Note.Velocity) * (float) pDimRgn->VCFVelocityScale * 6.2E-5f * FILTER_CUTOFF_COEFF) * FILTER_CUTOFF_MAX : exp((float) VCFCutoffCtrl.value * 0.00787402f * FILTER_CUTOFF_COEFF) * FILTER_CUTOFF_MAX; // calculate resonance float resonance = (float) VCFResonanceCtrl.value * 0.00787f; // 0.0..1.0 if (pDimRgn->VCFKeyboardTracking) { - resonance += (float) (pNoteOnEvent->Key - pDimRgn->VCFKeyboardTrackingBreakpoint) * 0.00787f; + resonance += (float) (itNoteOnEvent->Param.Note.Key - pDimRgn->VCFKeyboardTrackingBreakpoint) * 0.00787f; } Constrain(resonance, 0.0, 1.0); // correct resonance if outside allowed value range (0.0..1.0) VCFCutoffCtrl.fvalue = cutoff - FILTER_CUTOFF_MIN; VCFResonanceCtrl.fvalue = resonance; - FilterLeft.SetParameters(cutoff, resonance, pEngine->SampleRate); - FilterRight.SetParameters(cutoff, resonance, pEngine->SampleRate); - FilterUpdateCounter = -1; } else { VCFCutoffCtrl.controller = 0; VCFResonanceCtrl.controller = 0; } - #endif // ENABLE_FILTER return 0; // success } @@ -625,42 +640,46 @@ */ void Voice::Render(uint Samples) { + // select default values for synthesis mode bits + SYNTHESIS_MODE_SET_INTERPOLATE(SynthesisMode, (PitchBase * PitchBend) != 1.0f); + SYNTHESIS_MODE_SET_CONSTPITCH(SynthesisMode, true); + SYNTHESIS_MODE_SET_LOOP(SynthesisMode, false); + // Reset the synthesis parameter matrix + pEngine->ResetSynthesisParameters(Event::destination_vca, this->Volume * this->CrossfadeVolume * pEngine->GlobalVolume); pEngine->ResetSynthesisParameters(Event::destination_vco, this->PitchBase); - #if ENABLE_FILTER pEngine->ResetSynthesisParameters(Event::destination_vcfc, VCFCutoffCtrl.fvalue); pEngine->ResetSynthesisParameters(Event::destination_vcfr, VCFResonanceCtrl.fvalue); - #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 - pEG1->Process(Samples, pEngine->pMIDIKeyInfo[MIDIKey].pEvents, pTriggerEvent, this->Pos, this->PitchBase * this->PitchBend, pKillEvent); - #if ENABLE_FILTER - pEG2->Process(Samples, pEngine->pMIDIKeyInfo[MIDIKey].pEvents, pTriggerEvent, this->Pos, this->PitchBase * this->PitchBend); - #endif // ENABLE_FILTER - pEG3->Process(Samples); + pEG1->Process(Samples, pEngine->pMIDIKeyInfo[MIDIKey].pEvents, itTriggerEvent, this->Pos, this->PitchBase * this->PitchBend, itKillEvent); + pEG2->Process(Samples, pEngine->pMIDIKeyInfo[MIDIKey].pEvents, itTriggerEvent, this->Pos, this->PitchBase * this->PitchBend); + if (pEG3->Process(Samples)) { // if pitch EG is active + SYNTHESIS_MODE_SET_INTERPOLATE(SynthesisMode, true); + SYNTHESIS_MODE_SET_CONSTPITCH(SynthesisMode, false); + } pLFO1->Process(Samples); - #if ENABLE_FILTER pLFO2->Process(Samples); - #endif // ENABLE_FILTER - pLFO3->Process(Samples); - - - #if ENABLE_FILTER - CalculateBiquadParameters(Samples); // calculate the final biquad filter parameters - #endif // ENABLE_FILTER + if (pLFO3->Process(Samples)) { // if pitch LFO modulation is active + SYNTHESIS_MODE_SET_INTERPOLATE(SynthesisMode, true); + SYNTHESIS_MODE_SET_CONSTPITCH(SynthesisMode, false); + } + if (SYNTHESIS_MODE_GET_FILTER(SynthesisMode)) + CalculateBiquadParameters(Samples); // calculate the final biquad filter parameters switch (this->PlaybackState) { case playback_state_ram: { - if (RAMLoop) InterpolateAndLoop(Samples, (sample_t*) pSample->GetCache().pStart, Delay); - else InterpolateNoLoop(Samples, (sample_t*) pSample->GetCache().pStart, Delay); + if (RAMLoop) SYNTHESIS_MODE_SET_LOOP(SynthesisMode, true); // enable looping + + // render current fragment + Synthesize(Samples, (sample_t*) pSample->GetCache().pStart, Delay); + if (DiskVoice) { // check if we reached the allowed limit of the sample RAM cache if (Pos > MaxRAMPos) { @@ -683,43 +702,58 @@ KillImmediately(); return; } - DiskStreamRef.pStream->IncrementReadPos(pSample->Channels * (RTMath::DoubleToInt(Pos) - MaxRAMPos)); - Pos -= RTMath::DoubleToInt(Pos); + DiskStreamRef.pStream->IncrementReadPos(pSample->Channels * (int(Pos) - MaxRAMPos)); + Pos -= int(Pos); + RealSampleWordsLeftToRead = -1; // -1 means no silence has been added yet } + const int sampleWordsLeftToRead = DiskStreamRef.pStream->GetReadSpace(); + // 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() < (pEngine->MaxSamplesPerCycle << MAX_PITCH) / pSample->Channels) { - DiskStreamRef.pStream->WriteSilence((pEngine->MaxSamplesPerCycle << MAX_PITCH) / pSample->Channels); - this->PlaybackState = playback_state_end; + if (DiskStreamRef.State == Stream::state_end) { + const int maxSampleWordsPerCycle = (pEngine->MaxSamplesPerCycle << MAX_PITCH) * pSample->Channels + 6; // +6 for the interpolator algorithm + if (sampleWordsLeftToRead <= maxSampleWordsPerCycle) { + // remember how many sample words there are before any silence has been added + if (RealSampleWordsLeftToRead < 0) RealSampleWordsLeftToRead = sampleWordsLeftToRead; + DiskStreamRef.pStream->WriteSilence(maxSampleWordsPerCycle - sampleWordsLeftToRead); + } } sample_t* ptr = DiskStreamRef.pStream->GetReadPtr(); // get the current read_ptr within the ringbuffer where we read the samples from - InterpolateNoLoop(Samples, ptr, Delay); - DiskStreamRef.pStream->IncrementReadPos(RTMath::DoubleToInt(Pos) * pSample->Channels); - Pos -= RTMath::DoubleToInt(Pos); + + // render current audio fragment + Synthesize(Samples, ptr, Delay); + + const int iPos = (int) Pos; + const int readSampleWords = iPos * pSample->Channels; // amount of sample words actually been read + DiskStreamRef.pStream->IncrementReadPos(readSampleWords); + Pos -= iPos; // just keep fractional part of Pos + + // change state of voice to 'end' if we really reached the end of the sample data + if (RealSampleWordsLeftToRead >= 0) { + RealSampleWordsLeftToRead -= readSampleWords; + if (RealSampleWordsLeftToRead <= 0) this->PlaybackState = playback_state_end; + } } break; case playback_state_end: - KillImmediately(); // free voice + std::cerr << "gig::Voice::Render(): entered with playback_state_end, this is a bug!\n" << std::flush; break; } - // Reset synthesis event lists (except VCO, as VCO events apply channel wide currently) pEngine->pSynthesisEvents[Event::destination_vca]->clear(); - #if ENABLE_FILTER pEngine->pSynthesisEvents[Event::destination_vcfc]->clear(); pEngine->pSynthesisEvents[Event::destination_vcfr]->clear(); - #endif // ENABLE_FILTER // Reset delay Delay = 0; - pTriggerEvent = NULL; + itTriggerEvent = Pool::Iterator(); - // If release stage finished, let the voice be killed - if (pEG1->GetStage() == EGADSR::stage_end) this->PlaybackState = playback_state_end; + // If sample stream or release stage finished, kill the voice + if (PlaybackState == playback_state_end || pEG1->GetStage() == EGADSR::stage_end) KillImmediately(); } /** @@ -730,11 +764,15 @@ pLFO1->Reset(); pLFO2->Reset(); pLFO3->Reset(); + FilterLeft.Reset(); + FilterRight.Reset(); DiskStreamRef.pStream = NULL; DiskStreamRef.hStream = 0; DiskStreamRef.State = Stream::state_unused; DiskStreamRef.OrderID = 0; - Active = false; + PlaybackState = playback_state_end; + itTriggerEvent = Pool::Iterator(); + itKillEvent = Pool::Iterator(); } /** @@ -747,282 +785,219 @@ void Voice::ProcessEvents(uint Samples) { // dispatch control change events - Event* pCCEvent = pEngine->pCCEvents->first(); + RTList::Iterator itCCEvent = pEngine->pCCEvents->first(); if (Delay) { // skip events that happened before this voice was triggered - while (pCCEvent && pCCEvent->FragmentPos() <= Delay) pCCEvent = pEngine->pCCEvents->next(); + while (itCCEvent && itCCEvent->FragmentPos() <= Delay) ++itCCEvent; } - while (pCCEvent) { - if (pCCEvent->Controller) { // if valid MIDI controller - #if ENABLE_FILTER - if (pCCEvent->Controller == VCFCutoffCtrl.controller) { - pEngine->pSynthesisEvents[Event::destination_vcfc]->alloc_assign(*pCCEvent); + while (itCCEvent) { + if (itCCEvent->Param.CC.Controller) { // if valid MIDI controller + if (itCCEvent->Param.CC.Controller == VCFCutoffCtrl.controller) { + *pEngine->pSynthesisEvents[Event::destination_vcfc]->allocAppend() = *itCCEvent; } - if (pCCEvent->Controller == VCFResonanceCtrl.controller) { - pEngine->pSynthesisEvents[Event::destination_vcfr]->alloc_assign(*pCCEvent); + if (itCCEvent->Param.CC.Controller == VCFResonanceCtrl.controller) { + *pEngine->pSynthesisEvents[Event::destination_vcfr]->allocAppend() = *itCCEvent; } - #endif // ENABLE_FILTER - if (pCCEvent->Controller == pLFO1->ExtController) { - pLFO1->SendEvent(pCCEvent); + if (itCCEvent->Param.CC.Controller == pLFO1->ExtController) { + pLFO1->SendEvent(itCCEvent); } - #if ENABLE_FILTER - if (pCCEvent->Controller == pLFO2->ExtController) { - pLFO2->SendEvent(pCCEvent); + if (itCCEvent->Param.CC.Controller == pLFO2->ExtController) { + pLFO2->SendEvent(itCCEvent); } - #endif // ENABLE_FILTER - if (pCCEvent->Controller == pLFO3->ExtController) { - pLFO3->SendEvent(pCCEvent); + if (itCCEvent->Param.CC.Controller == pLFO3->ExtController) { + pLFO3->SendEvent(itCCEvent); } if (pDimRgn->AttenuationController.type == ::gig::attenuation_ctrl_t::type_controlchange && - pCCEvent->Controller == pDimRgn->AttenuationController.controller_number) { // if crossfade event - pEngine->pSynthesisEvents[Event::destination_vca]->alloc_assign(*pCCEvent); + itCCEvent->Param.CC.Controller == pDimRgn->AttenuationController.controller_number) { // if crossfade event + *pEngine->pSynthesisEvents[Event::destination_vca]->allocAppend() = *itCCEvent; } } - pCCEvent = pEngine->pCCEvents->next(); + ++itCCEvent; } // process pitch events { - RTEList* pVCOEventList = pEngine->pSynthesisEvents[Event::destination_vco]; - Event* pVCOEvent = pVCOEventList->first(); + RTList* pVCOEventList = pEngine->pSynthesisEvents[Event::destination_vco]; + RTList::Iterator itVCOEvent = pVCOEventList->first(); if (Delay) { // skip events that happened before this voice was triggered - while (pVCOEvent && pVCOEvent->FragmentPos() <= Delay) pVCOEvent = pVCOEventList->next(); + while (itVCOEvent && itVCOEvent->FragmentPos() <= Delay) ++itVCOEvent; } // apply old pitchbend value until first pitch event occurs if (this->PitchBend != 1.0) { - uint end = (pVCOEvent) ? pVCOEvent->FragmentPos() : Samples; + uint end = (itVCOEvent) ? itVCOEvent->FragmentPos() : Samples; for (uint i = Delay; i < end; i++) { pEngine->pSynthesisParameters[Event::destination_vco][i] *= this->PitchBend; } } float pitch; - while (pVCOEvent) { - Event* pNextVCOEvent = pVCOEventList->next(); + while (itVCOEvent) { + RTList::Iterator itNextVCOEvent = itVCOEvent; + ++itNextVCOEvent; // calculate the influence length of this event (in sample points) - uint end = (pNextVCOEvent) ? pNextVCOEvent->FragmentPos() : Samples; + uint end = (itNextVCOEvent) ? itNextVCOEvent->FragmentPos() : Samples; - pitch = RTMath::CentsToFreqRatio(((double) pVCOEvent->Pitch / 8192.0) * 200.0); // +-two semitones = +-200 cents + pitch = RTMath::CentsToFreqRatio(((double) itVCOEvent->Param.Pitch.Pitch / 8192.0) * 200.0); // +-two semitones = +-200 cents // apply pitch value to the pitch parameter sequence - for (uint i = pVCOEvent->FragmentPos(); i < end; i++) { + for (uint i = itVCOEvent->FragmentPos(); i < end; i++) { pEngine->pSynthesisParameters[Event::destination_vco][i] *= pitch; } - pVCOEvent = pNextVCOEvent; + itVCOEvent = itNextVCOEvent; + } + if (!pVCOEventList->isEmpty()) { + this->PitchBend = pitch; + SYNTHESIS_MODE_SET_INTERPOLATE(SynthesisMode, true); + SYNTHESIS_MODE_SET_CONSTPITCH(SynthesisMode, false); } - if (pVCOEventList->last()) this->PitchBend = pitch; } // process volume / attenuation events (TODO: we only handle and _expect_ crossfade events here ATM !) { - RTEList* pVCAEventList = pEngine->pSynthesisEvents[Event::destination_vca]; - Event* pVCAEvent = pVCAEventList->first(); + RTList* pVCAEventList = pEngine->pSynthesisEvents[Event::destination_vca]; + RTList::Iterator itVCAEvent = pVCAEventList->first(); if (Delay) { // skip events that happened before this voice was triggered - while (pVCAEvent && pVCAEvent->FragmentPos() <= Delay) pVCAEvent = pVCAEventList->next(); + while (itVCAEvent && itVCAEvent->FragmentPos() <= Delay) ++itVCAEvent; } float crossfadevolume; - while (pVCAEvent) { - Event* pNextVCAEvent = pVCAEventList->next(); + while (itVCAEvent) { + RTList::Iterator itNextVCAEvent = itVCAEvent; + ++itNextVCAEvent; // calculate the influence length of this event (in sample points) - uint end = (pNextVCAEvent) ? pNextVCAEvent->FragmentPos() : Samples; + uint end = (itNextVCAEvent) ? itNextVCAEvent->FragmentPos() : Samples; - crossfadevolume = CrossfadeAttenuation(pVCAEvent->Value); + crossfadevolume = CrossfadeAttenuation(itVCAEvent->Param.CC.Value); float effective_volume = crossfadevolume * this->Volume * pEngine->GlobalVolume; // apply volume value to the volume parameter sequence - for (uint i = pVCAEvent->FragmentPos(); i < end; i++) { + for (uint i = itVCAEvent->FragmentPos(); i < end; i++) { pEngine->pSynthesisParameters[Event::destination_vca][i] = effective_volume; } - pVCAEvent = pNextVCAEvent; + itVCAEvent = itNextVCAEvent; } - if (pVCAEventList->last()) this->CrossfadeVolume = crossfadevolume; + if (!pVCAEventList->isEmpty()) this->CrossfadeVolume = crossfadevolume; } - #if ENABLE_FILTER // process filter cutoff events { - RTEList* pCutoffEventList = pEngine->pSynthesisEvents[Event::destination_vcfc]; - Event* pCutoffEvent = pCutoffEventList->first(); + RTList* pCutoffEventList = pEngine->pSynthesisEvents[Event::destination_vcfc]; + RTList::Iterator itCutoffEvent = pCutoffEventList->first(); if (Delay) { // skip events that happened before this voice was triggered - while (pCutoffEvent && pCutoffEvent->FragmentPos() <= Delay) pCutoffEvent = pCutoffEventList->next(); + while (itCutoffEvent && itCutoffEvent->FragmentPos() <= Delay) ++itCutoffEvent; } float cutoff; - while (pCutoffEvent) { - Event* pNextCutoffEvent = pCutoffEventList->next(); + while (itCutoffEvent) { + RTList::Iterator itNextCutoffEvent = itCutoffEvent; + ++itNextCutoffEvent; // calculate the influence length of this event (in sample points) - uint end = (pNextCutoffEvent) ? pNextCutoffEvent->FragmentPos() : Samples; + uint end = (itNextCutoffEvent) ? itNextCutoffEvent->FragmentPos() : Samples; - cutoff = exp((float) pCutoffEvent->Value * 0.00787402f * FILTER_CUTOFF_COEFF) * FILTER_CUTOFF_MAX - FILTER_CUTOFF_MIN; + cutoff = exp((float) itCutoffEvent->Param.CC.Value * 0.00787402f * FILTER_CUTOFF_COEFF) * FILTER_CUTOFF_MAX - FILTER_CUTOFF_MIN; // apply cutoff frequency to the cutoff parameter sequence - for (uint i = pCutoffEvent->FragmentPos(); i < end; i++) { + for (uint i = itCutoffEvent->FragmentPos(); i < end; i++) { pEngine->pSynthesisParameters[Event::destination_vcfc][i] = cutoff; } - pCutoffEvent = pNextCutoffEvent; + itCutoffEvent = itNextCutoffEvent; } - if (pCutoffEventList->last()) VCFCutoffCtrl.fvalue = cutoff; // needed for initialization of parameter matrix next time + if (!pCutoffEventList->isEmpty()) VCFCutoffCtrl.fvalue = cutoff; // needed for initialization of parameter matrix next time } // process filter resonance events { - RTEList* pResonanceEventList = pEngine->pSynthesisEvents[Event::destination_vcfr]; - Event* pResonanceEvent = pResonanceEventList->first(); + RTList* pResonanceEventList = pEngine->pSynthesisEvents[Event::destination_vcfr]; + RTList::Iterator itResonanceEvent = pResonanceEventList->first(); if (Delay) { // skip events that happened before this voice was triggered - while (pResonanceEvent && pResonanceEvent->FragmentPos() <= Delay) pResonanceEvent = pResonanceEventList->next(); + while (itResonanceEvent && itResonanceEvent->FragmentPos() <= Delay) ++itResonanceEvent; } - while (pResonanceEvent) { - Event* pNextResonanceEvent = pResonanceEventList->next(); + while (itResonanceEvent) { + RTList::Iterator itNextResonanceEvent = itResonanceEvent; + ++itNextResonanceEvent; // calculate the influence length of this event (in sample points) - uint end = (pNextResonanceEvent) ? pNextResonanceEvent->FragmentPos() : Samples; + uint end = (itNextResonanceEvent) ? itNextResonanceEvent->FragmentPos() : Samples; // convert absolute controller value to differential - int ctrldelta = pResonanceEvent->Value - VCFResonanceCtrl.value; - VCFResonanceCtrl.value = pResonanceEvent->Value; + int ctrldelta = itResonanceEvent->Param.CC.Value - VCFResonanceCtrl.value; + VCFResonanceCtrl.value = itResonanceEvent->Param.CC.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++) { + for (uint i = itResonanceEvent->FragmentPos(); i < end; i++) { pEngine->pSynthesisParameters[Event::destination_vcfr][i] += resonancedelta; } - pResonanceEvent = pNextResonanceEvent; + itResonanceEvent = itNextResonanceEvent; } - if (pResonanceEventList->last()) VCFResonanceCtrl.fvalue = pResonanceEventList->last()->Value * 0.00787f; // needed for initialization of parameter matrix next time + if (!pResonanceEventList->isEmpty()) VCFResonanceCtrl.fvalue = pResonanceEventList->last()->Param.CC.Value * 0.00787f; // needed for initialization of parameter matrix next time } - #endif // ENABLE_FILTER } - #if ENABLE_FILTER /** * Calculate all necessary, final biquad filter parameters. * * @param Samples - number of samples to be rendered in this audio fragment cycle */ void Voice::CalculateBiquadParameters(uint Samples) { - if (!FilterLeft.Enabled) return; - biquad_param_t bqbase; biquad_param_t bqmain; float prev_cutoff = pEngine->pSynthesisParameters[Event::destination_vcfc][0]; float prev_res = pEngine->pSynthesisParameters[Event::destination_vcfr][0]; - FilterLeft.SetParameters(&bqbase, &bqmain, prev_cutoff, prev_res, pEngine->SampleRate); + FilterLeft.SetParameters( &bqbase, &bqmain, prev_cutoff + FILTER_CUTOFF_MIN, prev_res, pEngine->SampleRate); + FilterRight.SetParameters(&bqbase, &bqmain, prev_cutoff + FILTER_CUTOFF_MIN, prev_res, pEngine->SampleRate); pEngine->pBasicFilterParameters[0] = bqbase; pEngine->pMainFilterParameters[0] = bqmain; float* bq; for (int i = 1; i < Samples; i++) { // recalculate biquad parameters if cutoff or resonance differ from previous sample point - if (!(i & FILTER_UPDATE_MASK)) if (pEngine->pSynthesisParameters[Event::destination_vcfr][i] != prev_res || - pEngine->pSynthesisParameters[Event::destination_vcfc][i] != prev_cutoff) { - prev_cutoff = pEngine->pSynthesisParameters[Event::destination_vcfc][i]; - prev_res = pEngine->pSynthesisParameters[Event::destination_vcfr][i]; - FilterLeft.SetParameters(&bqbase, &bqmain, prev_cutoff, prev_res, pEngine->SampleRate); + if (!(i & FILTER_UPDATE_MASK)) { + if (pEngine->pSynthesisParameters[Event::destination_vcfr][i] != prev_res || + pEngine->pSynthesisParameters[Event::destination_vcfc][i] != prev_cutoff) + { + prev_cutoff = pEngine->pSynthesisParameters[Event::destination_vcfc][i]; + prev_res = pEngine->pSynthesisParameters[Event::destination_vcfr][i]; + FilterLeft.SetParameters( &bqbase, &bqmain, prev_cutoff + FILTER_CUTOFF_MIN, prev_res, pEngine->SampleRate); + FilterRight.SetParameters(&bqbase, &bqmain, prev_cutoff + FILTER_CUTOFF_MIN, prev_res, pEngine->SampleRate); + } } //same as 'pEngine->pBasicFilterParameters[i] = bqbase;' bq = (float*) &pEngine->pBasicFilterParameters[i]; - bq[0] = bqbase.a1; - bq[1] = bqbase.a2; - bq[2] = bqbase.b0; - bq[3] = bqbase.b1; - bq[4] = bqbase.b2; + bq[0] = bqbase.b0; + bq[1] = bqbase.b1; + bq[2] = bqbase.b2; + bq[3] = bqbase.a1; + bq[4] = bqbase.a2; // same as 'pEngine->pMainFilterParameters[i] = bqmain;' bq = (float*) &pEngine->pMainFilterParameters[i]; - bq[0] = bqmain.a1; - bq[1] = bqmain.a2; - bq[2] = bqmain.b0; - bq[3] = bqmain.b1; - bq[4] = bqmain.b2; + bq[0] = bqmain.b0; + bq[1] = bqmain.b1; + bq[2] = bqmain.b2; + bq[3] = bqmain.a1; + bq[4] = bqmain.a2; } } - #endif // ENABLE_FILTER /** - * Interpolates the input audio data (without looping). + * Synthesizes the current audio fragment for this voice. * * @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::InterpolateNoLoop(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) InterpolateStereo(pSrc, i); - } - else { // Mono Sample - while (i < Samples) InterpolateMono(pSrc, i); - } - } - - /** - * 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) { - InterpolateStereo(pSrc, i); - if (Pos > pSample->LoopEnd) { - Pos = pSample->LoopStart + fmod(Pos - pSample->LoopEnd, pSample->LoopSize);; - LoopCyclesLeft--; - } - } - // render on without loop - while (i < Samples) InterpolateStereo(pSrc, i); - } - else { // render loop (endless loop) - while (i < Samples) { - InterpolateStereo(pSrc, i); - if (Pos > pSample->LoopEnd) { - Pos = pSample->LoopStart + fmod(Pos - pSample->LoopEnd, pSample->LoopSize); - } - } - } - } - else { // Mono Sample - if (pSample->LoopPlayCount) { - // render loop (loop count limited) - while (i < Samples && LoopCyclesLeft) { - InterpolateMono(pSrc, i); - if (Pos > pSample->LoopEnd) { - Pos = pSample->LoopStart + fmod(Pos - pSample->LoopEnd, pSample->LoopSize);; - LoopCyclesLeft--; - } - } - // render on without loop - while (i < Samples) InterpolateMono(pSrc, i); - } - else { // render loop (endless loop) - while (i < Samples) { - InterpolateMono(pSrc, i); - if (Pos > pSample->LoopEnd) { - Pos = pSample->LoopStart + fmod(Pos - pSample->LoopEnd, pSample->LoopSize);; - } - } - } - } + void Voice::Synthesize(uint Samples, sample_t* pSrc, uint Skip) { + RunSynthesisFunction(SynthesisMode, *this, Samples, pSrc, Skip); } /** @@ -1047,11 +1022,15 @@ * of a voice, a kill process cannot be cancalled and is therefore * usually used for voice stealing and key group conflicts. * - * @param pKillEvent - event which caused the voice to be killed + * @param itKillEvent - event which caused the voice to be killed */ - void Voice::Kill(Event* pKillEvent) { - if (pTriggerEvent && pKillEvent->FragmentPos() <= pTriggerEvent->FragmentPos()) return; - this->pKillEvent = pKillEvent; + void Voice::Kill(Pool::Iterator& itKillEvent) { + //FIXME: just two sanity checks for debugging, can be removed + if (!itKillEvent) dmsg(1,("gig::Voice::Kill(): ERROR, !itKillEvent !!!\n")); + if (itKillEvent && !itKillEvent.isValid()) dmsg(1,("gig::Voice::Kill(): ERROR, itKillEvent invalid !!!\n")); + + if (itTriggerEvent && itKillEvent->FragmentPos() <= itTriggerEvent->FragmentPos()) return; + this->itKillEvent = itKillEvent; } }} // namespace LinuxSampler::gig