--- linuxsampler/trunk/src/engines/gig/Voice.cpp 2004/10/14 21:31:26 285 +++ linuxsampler/trunk/src/engines/gig/Voice.cpp 2005/02/23 19:14:14 407 @@ -22,6 +22,8 @@ #include "EGADSR.h" #include "Manipulator.h" +#include "../../common/Features.h" +#include "Synthesizer.h" #include "Voice.h" @@ -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() { @@ -108,13 +121,19 @@ * @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(Pool::Iterator& itNoteOnEvent, 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; MIDIKey = itNoteOnEvent->Param.Note.Key; @@ -126,8 +145,7 @@ 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: @@ -146,7 +164,7 @@ // 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++) - itChildVoice = pEngine->LaunchVoice(itNoteOnEvent, iNewLayer, ReleaseTriggerVoice); + itChildVoice = pEngine->LaunchVoice(itNoteOnEvent, iNewLayer, ReleaseTriggerVoice, VoiceStealing); break; case ::gig::dimension_velocity: DimValues[i] = itNoteOnEvent->Param.Note.Velocity; @@ -159,7 +177,7 @@ DimValues[i] = (uint) ReleaseTriggerVoice; break; case ::gig::dimension_keyboard: - DimValues[i] = (uint) itNoteOnEvent->Param.Note.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) { @@ -258,8 +282,6 @@ PanLeft = 1.0f - float(RTMath::Max(pDimRgn->Pan, 0)) / 63.0f; PanRight = 1.0f - float(RTMath::Min(pDimRgn->Pan, 0)) / -64.0f; - pSample = pDimRgn->pSample; // sample won't change until the voice is finished - Pos = pDimRgn->SampleStartOffset; // offset where we should start playback of sample (0 - 2000 sample points) // Check if the sample needs disk streaming or is too short for that @@ -296,15 +318,15 @@ // 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(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) + Volume *= pDimRgn->SampleAttenuation; // setup EG 1 (VCA EG) { @@ -344,7 +366,6 @@ } - #if ENABLE_FILTER // setup EG 2 (VCF Cutoff EG) { // get current value of EG2 controller @@ -381,7 +402,6 @@ pDimRgn->EG2Release + eg2release, Delay); } - #endif // ENABLE_FILTER // setup EG 3 (VCO EG) @@ -428,7 +448,7 @@ Delay); } - #if ENABLE_FILTER + // setup LFO 2 (VCF Cutoff LFO) { uint16_t lfo2_internal_depth; @@ -465,7 +485,7 @@ pEngine->SampleRate, Delay); } - #endif // ENABLE_FILTER + // setup LFO 3 (VCO LFO) { @@ -504,13 +524,14 @@ 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 @@ -598,16 +619,12 @@ 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 +642,46 @@ */ 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, itTriggerEvent, this->Pos, this->PitchBase * this->PitchBend, itKillEvent); - #if ENABLE_FILTER pEG2->Process(Samples, pEngine->pMIDIKeyInfo[MIDIKey].pEvents, itTriggerEvent, this->Pos, this->PitchBase * this->PitchBend); - #endif // ENABLE_FILTER - pEG3->Process(Samples); + 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) { @@ -683,20 +704,38 @@ 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; @@ -705,13 +744,10 @@ 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; @@ -730,6 +766,8 @@ pLFO1->Reset(); pLFO2->Reset(); pLFO3->Reset(); + FilterLeft.Reset(); + FilterRight.Reset(); DiskStreamRef.pStream = NULL; DiskStreamRef.hStream = 0; DiskStreamRef.State = Stream::state_unused; @@ -755,22 +793,18 @@ } while (itCCEvent) { if (itCCEvent->Param.CC.Controller) { // if valid MIDI controller - #if ENABLE_FILTER if (itCCEvent->Param.CC.Controller == VCFCutoffCtrl.controller) { *pEngine->pSynthesisEvents[Event::destination_vcfc]->allocAppend() = *itCCEvent; } if (itCCEvent->Param.CC.Controller == VCFResonanceCtrl.controller) { *pEngine->pSynthesisEvents[Event::destination_vcfr]->allocAppend() = *itCCEvent; } - #endif // ENABLE_FILTER if (itCCEvent->Param.CC.Controller == pLFO1->ExtController) { pLFO1->SendEvent(itCCEvent); } - #if ENABLE_FILTER if (itCCEvent->Param.CC.Controller == pLFO2->ExtController) { pLFO2->SendEvent(itCCEvent); } - #endif // ENABLE_FILTER if (itCCEvent->Param.CC.Controller == pLFO3->ExtController) { pLFO3->SendEvent(itCCEvent); } @@ -815,7 +849,11 @@ itVCOEvent = itNextVCOEvent; } - if (!pVCOEventList->isEmpty()) this->PitchBend = pitch; + if (!pVCOEventList->isEmpty()) { + this->PitchBend = pitch; + SYNTHESIS_MODE_SET_INTERPOLATE(SynthesisMode, true); + SYNTHESIS_MODE_SET_CONSTPITCH(SynthesisMode, false); + } } // process volume / attenuation events (TODO: we only handle and _expect_ crossfade events here ATM !) @@ -847,7 +885,6 @@ if (!pVCAEventList->isEmpty()) this->CrossfadeVolume = crossfadevolume; } - #if ENABLE_FILTER // process filter cutoff events { RTList* pCutoffEventList = pEngine->pSynthesisEvents[Event::destination_vcfc]; @@ -904,131 +941,65 @@ } 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); } /** @@ -1056,6 +1027,10 @@ * @param itKillEvent - event which caused the voice to be killed */ 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; }