22 |
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23 |
#include "EGADSR.h" |
#include "EGADSR.h" |
24 |
#include "Manipulator.h" |
#include "Manipulator.h" |
25 |
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#include "../../common/Features.h" |
26 |
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#include "Synthesizer.h" |
27 |
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28 |
#include "Voice.h" |
#include "Voice.h" |
29 |
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|
58 |
pLFO2 = NULL; |
pLFO2 = NULL; |
59 |
pLFO3 = NULL; |
pLFO3 = NULL; |
60 |
KeyGroup = 0; |
KeyGroup = 0; |
61 |
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SynthesisMode = 0; // set all mode bits to 0 first |
62 |
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// select synthesis implementation (currently either pure C++ or MMX+SSE(1)) |
63 |
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#if ARCH_X86 |
64 |
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SYNTHESIS_MODE_SET_IMPLEMENTATION(SynthesisMode, Features::supportsMMX() && Features::supportsSSE()); |
65 |
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#else |
66 |
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SYNTHESIS_MODE_SET_IMPLEMENTATION(SynthesisMode, false); |
67 |
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#endif |
68 |
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SYNTHESIS_MODE_SET_PROFILING(SynthesisMode, true); |
69 |
} |
} |
70 |
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|
71 |
Voice::~Voice() { |
Voice::~Voice() { |
119 |
* @param iLayer - layer number this voice refers to (only if this is a layered sound of course) |
* @param iLayer - layer number this voice refers to (only if this is a layered sound of course) |
120 |
* @param ReleaseTriggerVoice - if this new voice is a release trigger voice (optional, default = false) |
* @param ReleaseTriggerVoice - if this new voice is a release trigger voice (optional, default = false) |
121 |
* @param VoiceStealing - wether the voice is allowed to steal voices for further subvoices |
* @param VoiceStealing - wether the voice is allowed to steal voices for further subvoices |
122 |
* @returns 0 on success, a value < 0 if something failed |
* @returns 0 on success, a value < 0 if the voice wasn't triggered |
123 |
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* (either due to an error or e.g. because no region is |
124 |
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* defined for the given key) |
125 |
*/ |
*/ |
126 |
int Voice::Trigger(Pool<Event>::Iterator& itNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument, int iLayer, bool ReleaseTriggerVoice, bool VoiceStealing) { |
int Voice::Trigger(Pool<Event>::Iterator& itNoteOnEvent, int PitchBend, ::gig::Instrument* pInstrument, int iLayer, bool ReleaseTriggerVoice, bool VoiceStealing) { |
127 |
if (!pInstrument) { |
if (!pInstrument) { |
128 |
dmsg(1,("voice::trigger: !pInstrument\n")); |
dmsg(1,("voice::trigger: !pInstrument\n")); |
129 |
exit(EXIT_FAILURE); |
exit(EXIT_FAILURE); |
130 |
} |
} |
131 |
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if (itNoteOnEvent->FragmentPos() > pEngine->MaxSamplesPerCycle) { // FIXME: should be removed before the final release (purpose: just a sanity check for debugging) |
132 |
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dmsg(1,("Voice::Trigger(): ERROR, TriggerDelay > Totalsamples\n")); |
133 |
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} |
134 |
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135 |
Type = type_normal; |
Type = type_normal; |
136 |
MIDIKey = itNoteOnEvent->Param.Note.Key; |
MIDIKey = itNoteOnEvent->Param.Note.Key; |
142 |
itChildVoice = Pool<Voice>::Iterator(); |
itChildVoice = Pool<Voice>::Iterator(); |
143 |
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|
144 |
if (!pRegion) { |
if (!pRegion) { |
145 |
std::cerr << "gig::Voice: No Region defined for MIDI key " << MIDIKey << std::endl << std::flush; |
dmsg(4, ("gig::Voice: No Region defined for MIDI key %d\n", MIDIKey)); |
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KillImmediately(); |
|
146 |
return -1; |
return -1; |
147 |
} |
} |
148 |
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150 |
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|
151 |
// get current dimension values to select the right dimension region |
// get current dimension values to select the right dimension region |
152 |
//FIXME: controller values for selecting the dimension region here are currently not sample accurate |
//FIXME: controller values for selecting the dimension region here are currently not sample accurate |
153 |
uint DimValues[5] = {0,0,0,0,0}; |
uint DimValues[8] = { 0 }; |
154 |
for (int i = pRegion->Dimensions - 1; i >= 0; i--) { |
for (int i = pRegion->Dimensions - 1; i >= 0; i--) { |
155 |
switch (pRegion->pDimensionDefinitions[i].dimension) { |
switch (pRegion->pDimensionDefinitions[i].dimension) { |
156 |
case ::gig::dimension_samplechannel: |
case ::gig::dimension_samplechannel: |
174 |
DimValues[i] = (uint) ReleaseTriggerVoice; |
DimValues[i] = (uint) ReleaseTriggerVoice; |
175 |
break; |
break; |
176 |
case ::gig::dimension_keyboard: |
case ::gig::dimension_keyboard: |
177 |
DimValues[i] = (uint) itNoteOnEvent->Param.Note.Key; |
DimValues[i] = (uint) pEngine->CurrentKeyDimension; |
178 |
break; |
break; |
179 |
case ::gig::dimension_modwheel: |
case ::gig::dimension_modwheel: |
180 |
DimValues[i] = pEngine->ControllerTable[1]; |
DimValues[i] = pEngine->ControllerTable[1]; |
252 |
std::cerr << "gig::Voice::Trigger() Error: Unknown dimension\n" << std::flush; |
std::cerr << "gig::Voice::Trigger() Error: Unknown dimension\n" << std::flush; |
253 |
} |
} |
254 |
} |
} |
255 |
pDimRgn = pRegion->GetDimensionRegionByValue(DimValues[4],DimValues[3],DimValues[2],DimValues[1],DimValues[0]); |
pDimRgn = pRegion->GetDimensionRegionByValue(DimValues); |
256 |
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257 |
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pSample = pDimRgn->pSample; // sample won't change until the voice is finished |
258 |
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if (!pSample || !pSample->SamplesTotal) return -1; // no need to continue if sample is silent |
259 |
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260 |
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// select channel mode (mono or stereo) |
261 |
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SYNTHESIS_MODE_SET_CHANNELS(SynthesisMode, pSample->Channels == 2); |
262 |
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|
263 |
// get starting crossfade volume level |
// get starting crossfade volume level |
264 |
switch (pDimRgn->AttenuationController.type) { |
switch (pDimRgn->AttenuationController.type) { |
279 |
PanLeft = 1.0f - float(RTMath::Max(pDimRgn->Pan, 0)) / 63.0f; |
PanLeft = 1.0f - float(RTMath::Max(pDimRgn->Pan, 0)) / 63.0f; |
280 |
PanRight = 1.0f - float(RTMath::Min(pDimRgn->Pan, 0)) / -64.0f; |
PanRight = 1.0f - float(RTMath::Min(pDimRgn->Pan, 0)) / -64.0f; |
281 |
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pSample = pDimRgn->pSample; // sample won't change until the voice is finished |
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282 |
Pos = pDimRgn->SampleStartOffset; // offset where we should start playback of sample (0 - 2000 sample points) |
Pos = pDimRgn->SampleStartOffset; // offset where we should start playback of sample (0 - 2000 sample points) |
283 |
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|
284 |
// Check if the sample needs disk streaming or is too short for that |
// Check if the sample needs disk streaming or is too short for that |
315 |
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|
316 |
// calculate initial pitch value |
// calculate initial pitch value |
317 |
{ |
{ |
318 |
double pitchbasecents = pDimRgn->FineTune * 10 + (int) pEngine->ScaleTuning[MIDIKey % 12]; |
double pitchbasecents = pDimRgn->FineTune + (int) pEngine->ScaleTuning[MIDIKey % 12]; |
319 |
if (pDimRgn->PitchTrack) pitchbasecents += (MIDIKey - (int) pDimRgn->UnityNote) * 100; |
if (pDimRgn->PitchTrack) pitchbasecents += (MIDIKey - (int) pDimRgn->UnityNote) * 100; |
320 |
this->PitchBase = RTMath::CentsToFreqRatio(pitchbasecents) * (double(pSample->SamplesPerSecond) / double(pEngine->pAudioOutputDevice->SampleRate())); |
this->PitchBase = RTMath::CentsToFreqRatio(pitchbasecents) * (double(pSample->SamplesPerSecond) / double(pEngine->pAudioOutputDevice->SampleRate())); |
321 |
this->PitchBend = RTMath::CentsToFreqRatio(((double) PitchBend / 8192.0) * 200.0); // pitchbend wheel +-2 semitones = 200 cents |
this->PitchBend = RTMath::CentsToFreqRatio(((double) PitchBend / 8192.0) * 200.0); // pitchbend wheel +-2 semitones = 200 cents |
322 |
} |
} |
323 |
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324 |
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->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) |
325 |
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|
326 |
// setup EG 1 (VCA EG) |
// setup EG 1 (VCA EG) |
327 |
{ |
{ |
328 |
// get current value of EG1 controller |
// get current value of EG1 controller |
361 |
} |
} |
362 |
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|
363 |
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#if ENABLE_FILTER |
|
364 |
// setup EG 2 (VCF Cutoff EG) |
// setup EG 2 (VCF Cutoff EG) |
365 |
{ |
{ |
366 |
// get current value of EG2 controller |
// get current value of EG2 controller |
397 |
pDimRgn->EG2Release + eg2release, |
pDimRgn->EG2Release + eg2release, |
398 |
Delay); |
Delay); |
399 |
} |
} |
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#endif // ENABLE_FILTER |
|
400 |
|
|
401 |
|
|
402 |
// setup EG 3 (VCO EG) |
// setup EG 3 (VCO EG) |
443 |
Delay); |
Delay); |
444 |
} |
} |
445 |
|
|
446 |
#if ENABLE_FILTER |
|
447 |
// setup LFO 2 (VCF Cutoff LFO) |
// setup LFO 2 (VCF Cutoff LFO) |
448 |
{ |
{ |
449 |
uint16_t lfo2_internal_depth; |
uint16_t lfo2_internal_depth; |
480 |
pEngine->SampleRate, |
pEngine->SampleRate, |
481 |
Delay); |
Delay); |
482 |
} |
} |
483 |
#endif // ENABLE_FILTER |
|
484 |
|
|
485 |
// setup LFO 3 (VCO LFO) |
// setup LFO 3 (VCO LFO) |
486 |
{ |
{ |
519 |
Delay); |
Delay); |
520 |
} |
} |
521 |
|
|
522 |
#if ENABLE_FILTER |
|
523 |
#if FORCE_FILTER_USAGE |
#if FORCE_FILTER_USAGE |
524 |
FilterLeft.Enabled = FilterRight.Enabled = true; |
SYNTHESIS_MODE_SET_FILTER(SynthesisMode, true); |
525 |
#else // use filter only if instrument file told so |
#else // use filter only if instrument file told so |
526 |
FilterLeft.Enabled = FilterRight.Enabled = pDimRgn->VCFEnabled; |
SYNTHESIS_MODE_SET_FILTER(SynthesisMode, pDimRgn->VCFEnabled); |
527 |
#endif // FORCE_FILTER_USAGE |
#endif // FORCE_FILTER_USAGE |
528 |
if (pDimRgn->VCFEnabled) { |
if (pDimRgn->VCFEnabled) { |
529 |
#ifdef OVERRIDE_FILTER_CUTOFF_CTRL |
#ifdef OVERRIDE_FILTER_CUTOFF_CTRL |
613 |
VCFCutoffCtrl.fvalue = cutoff - FILTER_CUTOFF_MIN; |
VCFCutoffCtrl.fvalue = cutoff - FILTER_CUTOFF_MIN; |
614 |
VCFResonanceCtrl.fvalue = resonance; |
VCFResonanceCtrl.fvalue = resonance; |
615 |
|
|
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FilterLeft.SetParameters(cutoff, resonance, pEngine->SampleRate); |
|
|
FilterRight.SetParameters(cutoff, resonance, pEngine->SampleRate); |
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|
616 |
FilterUpdateCounter = -1; |
FilterUpdateCounter = -1; |
617 |
} |
} |
618 |
else { |
else { |
619 |
VCFCutoffCtrl.controller = 0; |
VCFCutoffCtrl.controller = 0; |
620 |
VCFResonanceCtrl.controller = 0; |
VCFResonanceCtrl.controller = 0; |
621 |
} |
} |
|
#endif // ENABLE_FILTER |
|
622 |
|
|
623 |
return 0; // success |
return 0; // success |
624 |
} |
} |
636 |
*/ |
*/ |
637 |
void Voice::Render(uint Samples) { |
void Voice::Render(uint Samples) { |
638 |
|
|
639 |
|
// select default values for synthesis mode bits |
640 |
|
SYNTHESIS_MODE_SET_INTERPOLATE(SynthesisMode, (PitchBase * PitchBend) != 1.0f); |
641 |
|
SYNTHESIS_MODE_SET_CONSTPITCH(SynthesisMode, true); |
642 |
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SYNTHESIS_MODE_SET_LOOP(SynthesisMode, false); |
643 |
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|
644 |
// Reset the synthesis parameter matrix |
// Reset the synthesis parameter matrix |
645 |
|
|
646 |
pEngine->ResetSynthesisParameters(Event::destination_vca, this->Volume * this->CrossfadeVolume * pEngine->GlobalVolume); |
pEngine->ResetSynthesisParameters(Event::destination_vca, this->Volume * this->CrossfadeVolume * pEngine->GlobalVolume); |
647 |
pEngine->ResetSynthesisParameters(Event::destination_vco, this->PitchBase); |
pEngine->ResetSynthesisParameters(Event::destination_vco, this->PitchBase); |
|
#if ENABLE_FILTER |
|
648 |
pEngine->ResetSynthesisParameters(Event::destination_vcfc, VCFCutoffCtrl.fvalue); |
pEngine->ResetSynthesisParameters(Event::destination_vcfc, VCFCutoffCtrl.fvalue); |
649 |
pEngine->ResetSynthesisParameters(Event::destination_vcfr, VCFResonanceCtrl.fvalue); |
pEngine->ResetSynthesisParameters(Event::destination_vcfr, VCFResonanceCtrl.fvalue); |
|
#endif // ENABLE_FILTER |
|
|
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|
650 |
|
|
651 |
// Apply events to the synthesis parameter matrix |
// Apply events to the synthesis parameter matrix |
652 |
ProcessEvents(Samples); |
ProcessEvents(Samples); |
653 |
|
|
|
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|
654 |
// Let all modulators write their parameter changes to the synthesis parameter matrix for the current audio fragment |
// Let all modulators write their parameter changes to the synthesis parameter matrix for the current audio fragment |
655 |
pEG1->Process(Samples, pEngine->pMIDIKeyInfo[MIDIKey].pEvents, itTriggerEvent, this->Pos, this->PitchBase * this->PitchBend, itKillEvent); |
pEG1->Process(Samples, pEngine->pMIDIKeyInfo[MIDIKey].pEvents, itTriggerEvent, this->Pos, this->PitchBase * this->PitchBend, itKillEvent); |
|
#if ENABLE_FILTER |
|
656 |
pEG2->Process(Samples, pEngine->pMIDIKeyInfo[MIDIKey].pEvents, itTriggerEvent, this->Pos, this->PitchBase * this->PitchBend); |
pEG2->Process(Samples, pEngine->pMIDIKeyInfo[MIDIKey].pEvents, itTriggerEvent, this->Pos, this->PitchBase * this->PitchBend); |
657 |
#endif // ENABLE_FILTER |
if (pEG3->Process(Samples)) { // if pitch EG is active |
658 |
pEG3->Process(Samples); |
SYNTHESIS_MODE_SET_INTERPOLATE(SynthesisMode, true); |
659 |
|
SYNTHESIS_MODE_SET_CONSTPITCH(SynthesisMode, false); |
660 |
|
} |
661 |
pLFO1->Process(Samples); |
pLFO1->Process(Samples); |
|
#if ENABLE_FILTER |
|
662 |
pLFO2->Process(Samples); |
pLFO2->Process(Samples); |
663 |
#endif // ENABLE_FILTER |
if (pLFO3->Process(Samples)) { // if pitch LFO modulation is active |
664 |
pLFO3->Process(Samples); |
SYNTHESIS_MODE_SET_INTERPOLATE(SynthesisMode, true); |
665 |
|
SYNTHESIS_MODE_SET_CONSTPITCH(SynthesisMode, false); |
666 |
|
} |
|
#if ENABLE_FILTER |
|
|
CalculateBiquadParameters(Samples); // calculate the final biquad filter parameters |
|
|
#endif // ENABLE_FILTER |
|
667 |
|
|
668 |
|
if (SYNTHESIS_MODE_GET_FILTER(SynthesisMode)) |
669 |
|
CalculateBiquadParameters(Samples); // calculate the final biquad filter parameters |
670 |
|
|
671 |
switch (this->PlaybackState) { |
switch (this->PlaybackState) { |
672 |
|
|
673 |
case playback_state_ram: { |
case playback_state_ram: { |
674 |
if (RAMLoop) InterpolateAndLoop(Samples, (sample_t*) pSample->GetCache().pStart, Delay); |
if (RAMLoop) SYNTHESIS_MODE_SET_LOOP(SynthesisMode, true); // enable looping |
675 |
else InterpolateNoLoop(Samples, (sample_t*) pSample->GetCache().pStart, Delay); |
|
676 |
|
// render current fragment |
677 |
|
Synthesize(Samples, (sample_t*) pSample->GetCache().pStart, Delay); |
678 |
|
|
679 |
if (DiskVoice) { |
if (DiskVoice) { |
680 |
// check if we reached the allowed limit of the sample RAM cache |
// check if we reached the allowed limit of the sample RAM cache |
681 |
if (Pos > MaxRAMPos) { |
if (Pos > MaxRAMPos) { |
698 |
KillImmediately(); |
KillImmediately(); |
699 |
return; |
return; |
700 |
} |
} |
701 |
DiskStreamRef.pStream->IncrementReadPos(pSample->Channels * (RTMath::DoubleToInt(Pos) - MaxRAMPos)); |
DiskStreamRef.pStream->IncrementReadPos(pSample->Channels * (int(Pos) - MaxRAMPos)); |
702 |
Pos -= RTMath::DoubleToInt(Pos); |
Pos -= int(Pos); |
703 |
|
RealSampleWordsLeftToRead = -1; // -1 means no silence has been added yet |
704 |
} |
} |
705 |
|
|
706 |
|
const int sampleWordsLeftToRead = DiskStreamRef.pStream->GetReadSpace(); |
707 |
|
|
708 |
// add silence sample at the end if we reached the end of the stream (for the interpolator) |
// add silence sample at the end if we reached the end of the stream (for the interpolator) |
709 |
if (DiskStreamRef.State == Stream::state_end && DiskStreamRef.pStream->GetReadSpace() < (pEngine->MaxSamplesPerCycle << MAX_PITCH) / pSample->Channels) { |
if (DiskStreamRef.State == Stream::state_end) { |
710 |
DiskStreamRef.pStream->WriteSilence((pEngine->MaxSamplesPerCycle << MAX_PITCH) / pSample->Channels); |
const int maxSampleWordsPerCycle = (pEngine->MaxSamplesPerCycle << MAX_PITCH) * pSample->Channels + 6; // +6 for the interpolator algorithm |
711 |
this->PlaybackState = playback_state_end; |
if (sampleWordsLeftToRead <= maxSampleWordsPerCycle) { |
712 |
|
// remember how many sample words there are before any silence has been added |
713 |
|
if (RealSampleWordsLeftToRead < 0) RealSampleWordsLeftToRead = sampleWordsLeftToRead; |
714 |
|
DiskStreamRef.pStream->WriteSilence(maxSampleWordsPerCycle - sampleWordsLeftToRead); |
715 |
|
} |
716 |
} |
} |
717 |
|
|
718 |
sample_t* ptr = DiskStreamRef.pStream->GetReadPtr(); // get the current read_ptr within the ringbuffer where we read the samples from |
sample_t* ptr = DiskStreamRef.pStream->GetReadPtr(); // get the current read_ptr within the ringbuffer where we read the samples from |
719 |
InterpolateNoLoop(Samples, ptr, Delay); |
|
720 |
DiskStreamRef.pStream->IncrementReadPos(RTMath::DoubleToInt(Pos) * pSample->Channels); |
// render current audio fragment |
721 |
Pos -= RTMath::DoubleToInt(Pos); |
Synthesize(Samples, ptr, Delay); |
722 |
|
|
723 |
|
const int iPos = (int) Pos; |
724 |
|
const int readSampleWords = iPos * pSample->Channels; // amount of sample words actually been read |
725 |
|
DiskStreamRef.pStream->IncrementReadPos(readSampleWords); |
726 |
|
Pos -= iPos; // just keep fractional part of Pos |
727 |
|
|
728 |
|
// change state of voice to 'end' if we really reached the end of the sample data |
729 |
|
if (RealSampleWordsLeftToRead >= 0) { |
730 |
|
RealSampleWordsLeftToRead -= readSampleWords; |
731 |
|
if (RealSampleWordsLeftToRead <= 0) this->PlaybackState = playback_state_end; |
732 |
|
} |
733 |
} |
} |
734 |
break; |
break; |
735 |
|
|
738 |
break; |
break; |
739 |
} |
} |
740 |
|
|
|
|
|
741 |
// Reset synthesis event lists (except VCO, as VCO events apply channel wide currently) |
// Reset synthesis event lists (except VCO, as VCO events apply channel wide currently) |
742 |
pEngine->pSynthesisEvents[Event::destination_vca]->clear(); |
pEngine->pSynthesisEvents[Event::destination_vca]->clear(); |
|
#if ENABLE_FILTER |
|
743 |
pEngine->pSynthesisEvents[Event::destination_vcfc]->clear(); |
pEngine->pSynthesisEvents[Event::destination_vcfc]->clear(); |
744 |
pEngine->pSynthesisEvents[Event::destination_vcfr]->clear(); |
pEngine->pSynthesisEvents[Event::destination_vcfr]->clear(); |
|
#endif // ENABLE_FILTER |
|
745 |
|
|
746 |
// Reset delay |
// Reset delay |
747 |
Delay = 0; |
Delay = 0; |
760 |
pLFO1->Reset(); |
pLFO1->Reset(); |
761 |
pLFO2->Reset(); |
pLFO2->Reset(); |
762 |
pLFO3->Reset(); |
pLFO3->Reset(); |
763 |
|
FilterLeft.Reset(); |
764 |
|
FilterRight.Reset(); |
765 |
DiskStreamRef.pStream = NULL; |
DiskStreamRef.pStream = NULL; |
766 |
DiskStreamRef.hStream = 0; |
DiskStreamRef.hStream = 0; |
767 |
DiskStreamRef.State = Stream::state_unused; |
DiskStreamRef.State = Stream::state_unused; |
787 |
} |
} |
788 |
while (itCCEvent) { |
while (itCCEvent) { |
789 |
if (itCCEvent->Param.CC.Controller) { // if valid MIDI controller |
if (itCCEvent->Param.CC.Controller) { // if valid MIDI controller |
|
#if ENABLE_FILTER |
|
790 |
if (itCCEvent->Param.CC.Controller == VCFCutoffCtrl.controller) { |
if (itCCEvent->Param.CC.Controller == VCFCutoffCtrl.controller) { |
791 |
*pEngine->pSynthesisEvents[Event::destination_vcfc]->allocAppend() = *itCCEvent; |
*pEngine->pSynthesisEvents[Event::destination_vcfc]->allocAppend() = *itCCEvent; |
792 |
} |
} |
793 |
if (itCCEvent->Param.CC.Controller == VCFResonanceCtrl.controller) { |
if (itCCEvent->Param.CC.Controller == VCFResonanceCtrl.controller) { |
794 |
*pEngine->pSynthesisEvents[Event::destination_vcfr]->allocAppend() = *itCCEvent; |
*pEngine->pSynthesisEvents[Event::destination_vcfr]->allocAppend() = *itCCEvent; |
795 |
} |
} |
|
#endif // ENABLE_FILTER |
|
796 |
if (itCCEvent->Param.CC.Controller == pLFO1->ExtController) { |
if (itCCEvent->Param.CC.Controller == pLFO1->ExtController) { |
797 |
pLFO1->SendEvent(itCCEvent); |
pLFO1->SendEvent(itCCEvent); |
798 |
} |
} |
|
#if ENABLE_FILTER |
|
799 |
if (itCCEvent->Param.CC.Controller == pLFO2->ExtController) { |
if (itCCEvent->Param.CC.Controller == pLFO2->ExtController) { |
800 |
pLFO2->SendEvent(itCCEvent); |
pLFO2->SendEvent(itCCEvent); |
801 |
} |
} |
|
#endif // ENABLE_FILTER |
|
802 |
if (itCCEvent->Param.CC.Controller == pLFO3->ExtController) { |
if (itCCEvent->Param.CC.Controller == pLFO3->ExtController) { |
803 |
pLFO3->SendEvent(itCCEvent); |
pLFO3->SendEvent(itCCEvent); |
804 |
} |
} |
843 |
|
|
844 |
itVCOEvent = itNextVCOEvent; |
itVCOEvent = itNextVCOEvent; |
845 |
} |
} |
846 |
if (!pVCOEventList->isEmpty()) this->PitchBend = pitch; |
if (!pVCOEventList->isEmpty()) { |
847 |
|
this->PitchBend = pitch; |
848 |
|
SYNTHESIS_MODE_SET_INTERPOLATE(SynthesisMode, true); |
849 |
|
SYNTHESIS_MODE_SET_CONSTPITCH(SynthesisMode, false); |
850 |
|
} |
851 |
} |
} |
852 |
|
|
853 |
// process volume / attenuation events (TODO: we only handle and _expect_ crossfade events here ATM !) |
// process volume / attenuation events (TODO: we only handle and _expect_ crossfade events here ATM !) |
879 |
if (!pVCAEventList->isEmpty()) this->CrossfadeVolume = crossfadevolume; |
if (!pVCAEventList->isEmpty()) this->CrossfadeVolume = crossfadevolume; |
880 |
} |
} |
881 |
|
|
|
#if ENABLE_FILTER |
|
882 |
// process filter cutoff events |
// process filter cutoff events |
883 |
{ |
{ |
884 |
RTList<Event>* pCutoffEventList = pEngine->pSynthesisEvents[Event::destination_vcfc]; |
RTList<Event>* pCutoffEventList = pEngine->pSynthesisEvents[Event::destination_vcfc]; |
935 |
} |
} |
936 |
if (!pResonanceEventList->isEmpty()) VCFResonanceCtrl.fvalue = pResonanceEventList->last()->Param.CC.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 |
937 |
} |
} |
|
#endif // ENABLE_FILTER |
|
938 |
} |
} |
939 |
|
|
|
#if ENABLE_FILTER |
|
940 |
/** |
/** |
941 |
* Calculate all necessary, final biquad filter parameters. |
* Calculate all necessary, final biquad filter parameters. |
942 |
* |
* |
943 |
* @param Samples - number of samples to be rendered in this audio fragment cycle |
* @param Samples - number of samples to be rendered in this audio fragment cycle |
944 |
*/ |
*/ |
945 |
void Voice::CalculateBiquadParameters(uint Samples) { |
void Voice::CalculateBiquadParameters(uint Samples) { |
|
if (!FilterLeft.Enabled) return; |
|
|
|
|
946 |
biquad_param_t bqbase; |
biquad_param_t bqbase; |
947 |
biquad_param_t bqmain; |
biquad_param_t bqmain; |
948 |
float prev_cutoff = pEngine->pSynthesisParameters[Event::destination_vcfc][0]; |
float prev_cutoff = pEngine->pSynthesisParameters[Event::destination_vcfc][0]; |
949 |
float prev_res = pEngine->pSynthesisParameters[Event::destination_vcfr][0]; |
float prev_res = pEngine->pSynthesisParameters[Event::destination_vcfr][0]; |
950 |
FilterLeft.SetParameters(&bqbase, &bqmain, prev_cutoff, prev_res, pEngine->SampleRate); |
FilterLeft.SetParameters(&bqbase, &bqmain, prev_cutoff, prev_res, pEngine->SampleRate); |
951 |
|
FilterRight.SetParameters(&bqbase, &bqmain, prev_cutoff, prev_res, pEngine->SampleRate); |
952 |
pEngine->pBasicFilterParameters[0] = bqbase; |
pEngine->pBasicFilterParameters[0] = bqbase; |
953 |
pEngine->pMainFilterParameters[0] = bqmain; |
pEngine->pMainFilterParameters[0] = bqmain; |
954 |
|
|
955 |
float* bq; |
float* bq; |
956 |
for (int i = 1; i < Samples; i++) { |
for (int i = 1; i < Samples; i++) { |
957 |
// recalculate biquad parameters if cutoff or resonance differ from previous sample point |
// recalculate biquad parameters if cutoff or resonance differ from previous sample point |
958 |
if (!(i & FILTER_UPDATE_MASK)) if (pEngine->pSynthesisParameters[Event::destination_vcfr][i] != prev_res || |
if (!(i & FILTER_UPDATE_MASK)) { |
959 |
pEngine->pSynthesisParameters[Event::destination_vcfc][i] != prev_cutoff) { |
if (pEngine->pSynthesisParameters[Event::destination_vcfr][i] != prev_res || |
960 |
prev_cutoff = pEngine->pSynthesisParameters[Event::destination_vcfc][i]; |
pEngine->pSynthesisParameters[Event::destination_vcfc][i] != prev_cutoff) |
961 |
prev_res = pEngine->pSynthesisParameters[Event::destination_vcfr][i]; |
{ |
962 |
FilterLeft.SetParameters(&bqbase, &bqmain, prev_cutoff, prev_res, pEngine->SampleRate); |
prev_cutoff = pEngine->pSynthesisParameters[Event::destination_vcfc][i]; |
963 |
|
prev_res = pEngine->pSynthesisParameters[Event::destination_vcfr][i]; |
964 |
|
FilterLeft.SetParameters(&bqbase, &bqmain, prev_cutoff, prev_res, pEngine->SampleRate); |
965 |
|
FilterRight.SetParameters(&bqbase, &bqmain, prev_cutoff, prev_res, pEngine->SampleRate); |
966 |
|
} |
967 |
} |
} |
968 |
|
|
969 |
//same as 'pEngine->pBasicFilterParameters[i] = bqbase;' |
//same as 'pEngine->pBasicFilterParameters[i] = bqbase;' |
970 |
bq = (float*) &pEngine->pBasicFilterParameters[i]; |
bq = (float*) &pEngine->pBasicFilterParameters[i]; |
971 |
bq[0] = bqbase.a1; |
bq[0] = bqbase.b0; |
972 |
bq[1] = bqbase.a2; |
bq[1] = bqbase.b1; |
973 |
bq[2] = bqbase.b0; |
bq[2] = bqbase.b2; |
974 |
bq[3] = bqbase.b1; |
bq[3] = bqbase.a1; |
975 |
bq[4] = bqbase.b2; |
bq[4] = bqbase.a2; |
976 |
|
|
977 |
// same as 'pEngine->pMainFilterParameters[i] = bqmain;' |
// same as 'pEngine->pMainFilterParameters[i] = bqmain;' |
978 |
bq = (float*) &pEngine->pMainFilterParameters[i]; |
bq = (float*) &pEngine->pMainFilterParameters[i]; |
979 |
bq[0] = bqmain.a1; |
bq[0] = bqmain.b0; |
980 |
bq[1] = bqmain.a2; |
bq[1] = bqmain.b1; |
981 |
bq[2] = bqmain.b0; |
bq[2] = bqmain.b2; |
982 |
bq[3] = bqmain.b1; |
bq[3] = bqmain.a1; |
983 |
bq[4] = bqmain.b2; |
bq[4] = bqmain.a2; |
984 |
} |
} |
985 |
} |
} |
|
#endif // ENABLE_FILTER |
|
986 |
|
|
987 |
/** |
/** |
988 |
* Interpolates the input audio data (without looping). |
* Synthesizes the current audio fragment for this voice. |
989 |
* |
* |
990 |
* @param Samples - number of sample points to be rendered in this audio |
* @param Samples - number of sample points to be rendered in this audio |
991 |
* fragment cycle |
* fragment cycle |
992 |
* @param pSrc - pointer to input sample data |
* @param pSrc - pointer to input sample data |
993 |
* @param Skip - number of sample points to skip in output buffer |
* @param Skip - number of sample points to skip in output buffer |
994 |
*/ |
*/ |
995 |
void Voice::InterpolateNoLoop(uint Samples, sample_t* pSrc, uint Skip) { |
void Voice::Synthesize(uint Samples, sample_t* pSrc, uint Skip) { |
996 |
int i = Skip; |
RunSynthesisFunction(SynthesisMode, *this, Samples, pSrc, 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);; |
|
|
} |
|
|
} |
|
|
} |
|
|
} |
|
997 |
} |
} |
998 |
|
|
999 |
/** |
/** |