Multiple start and end ranges in a single AudioLoopPlayer
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@d-healey Thanks, aggh it's a shame, maybe down the line then
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You could probably fake it if you're willing to use multiple, linked loop players.
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@rglides
Possible with c++ nodes. I can give you one. -
@rglides Haha yeah I tried, linking more than one AudioWaveform to a single LoopPlayer means an update in one tile just updates both tiles, so I don't think it's possible like that
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@modularsamples Oh really? yes please! I'll dm you
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Just a heads up to people, I've solved this with a custom c++ node recently.
I'll make a post about it once I've cleaned the example up better.
The sampler is a c++ scriptnode node, that loads a sample via external data and you can use hise scripting to make different notes or voices play back different parts of the sample.
The sampler also has x fade looping and parameters for pitch bends etc, all controllable in a polyphonic and per voice manner.
It partially integrates with the audio waveform component.
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@griffinboy Oh nice. I've done a similar thing, would be interested to compare code. Mine has: sample-start/end, loop start/end, loop on/off, alternate on/off, and start-link, which loops the start of the voice to the loop start instead. I've got a bit of a problem with my uptime not properly resetting when you activate alternate or move a marker during a voice, but I haven't had time to diagnose it yet.
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I'll give you a peek, but this isn't meant for other people to try and use yet (it's not properly cleaned up or optimized).
This is the polyphonic sampler node. We then use the Hise Midi processing scripts to target notes / voices and send event data to control the parameters in this node. Meaning that we end up with polyphonic control over the parameters on a voice / note basis. The c++ sampler allows for safe modulation of any parameter, which means we can simply update it from the Hise scripts without worrying.#pragma once #include <JuceHeader.h> #include <array> #include <vector> #include <cmath> #include <algorithm> #include <random> #include <limits> #include <new> #include <atomic> namespace project { #ifndef M_PI #define M_PI 3.14159265358979323846 #endif #if defined(_MSC_VER) #define FORCE_INLINE __forceinline #else #define FORCE_INLINE inline __attribute__((always_inline)) #endif using namespace juce; using namespace hise; using namespace scriptnode; static constexpr int FIXED_SHIFT = 16; static constexpr int64_t FIXED_ONE = (int64_t)1 << FIXED_SHIFT; static constexpr int64_t FIXED_MASK = FIXED_ONE - 1; struct SampleSettings { double pitchOffsetCents = 0.0; float volumeMult = 1.0f; float panning = 0.0f; float playbackStartInSamples = 0.0f; float loopStartInSamples = 0.0f; float loopEndInSamples = 0.0f; bool loopMode = false; float xfadeLengthInSamples = 0.0f; }; struct SamplePlayback { const float* sourceL = nullptr; const float* sourceR = nullptr; bool active = false; float amplitude = 1.0f; SampleSettings settings; int64_t phaseAcc = 0; int64_t phaseInc = 0; float playbackStart = 0.0f; float loopStart = 0.0f; float loopEnd = 0.0f; SamplePlayback() noexcept = default; SamplePlayback(const std::array<const float*, 2>& src, float amp, const SampleSettings& s, int bufferLength, double baseDelta) { sourceL = src[0]; sourceR = src[1]; settings = s; amplitude = amp * s.volumeMult; active = true; float pbStart = s.playbackStartInSamples; if (pbStart < 0.f) pbStart = 0.f; if (pbStart > float(bufferLength - 1)) pbStart = float(bufferLength - 1); playbackStart = pbStart; float lStart = s.loopStartInSamples; float lEnd = s.loopEndInSamples; if (!s.loopMode) { lStart = pbStart; lEnd = s.loopEndInSamples; } else { if (lStart < 0.f) lStart = 0.f; if (lStart > float(bufferLength - 1)) lStart = float(bufferLength - 1); if (lEnd < 0.f) lEnd = 0.f; if (lEnd > float(bufferLength - 1)) lEnd = float(bufferLength - 1); } loopStart = lStart; loopEnd = lEnd; phaseAcc = (int64_t)std::llround(pbStart * FIXED_ONE); double centsFact = std::pow(2.0, (s.pitchOffsetCents / 1200.0)); double effectiveSpeed = baseDelta * centsFact; phaseInc = (int64_t)std::llround(effectiveSpeed * FIXED_ONE); } // Marked as inline to help reduce overhead in performance-critical paths. FORCE_INLINE void updateSettings(const SampleSettings& s, int bufferLength) { settings = s; float pbStart = s.playbackStartInSamples; if (pbStart < 0.f) pbStart = 0.f; if (pbStart > float(bufferLength - 1)) pbStart = float(bufferLength - 1); playbackStart = pbStart; float lStart = s.loopStartInSamples; float lEnd = s.loopEndInSamples; if (!s.loopMode) { lStart = pbStart; lEnd = s.loopEndInSamples; } else { if (lStart < 0.f) lStart = 0.f; if (lStart > float(bufferLength - 1)) lStart = float(bufferLength - 1); if (lEnd < 0.f) lEnd = 0.f; if (lEnd > float(bufferLength - 1)) lEnd = float(bufferLength - 1); } loopStart = lStart; loopEnd = lEnd; } // This synthesis function is critical in performance: it is FORCE_INLINE to encourage inlining // and precomputes frequently used values outside the inner loops. FORCE_INLINE int vectorSynthesize(float* outL, float* outR, int blockSize, const AudioBuffer<float>* preXfadeBuffer, float preXfadeLength) { if (!active) return 0; int processed = 0; const float invFixedOne = 1.f / FIXED_ONE; const float leftGain = 0.5f * (1.f - settings.panning); const float rightGain = 0.5f * (1.f + settings.panning); const float ampLeft = amplitude * leftGain; const float ampRight = amplitude * rightGain; const bool loopEnabled = settings.loopMode; const float X = settings.xfadeLengthInSamples; const float endSample = loopEnd; const float crossfadeStartF = loopEnabled ? (endSample - X) : 0.f; const float piOverTwo = float(M_PI * 0.5f); const int64_t fixedEnd = int64_t(endSample * FIXED_ONE); int64_t fixedLoopStart = 0; int64_t fixedCrossfadeStart = 0; int64_t fixedX = 0; if (loopEnabled) { fixedLoopStart = int64_t(loopStart * FIXED_ONE); fixedX = (int64_t)std::llround(X * FIXED_ONE); fixedCrossfadeStart = fixedEnd - fixedX; } while (processed < blockSize) { if (!loopEnabled) { if (phaseAcc >= fixedEnd) { active = false; break; } int64_t samplesToBoundary = (fixedEnd - phaseAcc + phaseInc - 1) / phaseInc; int n = (samplesToBoundary > (blockSize - processed)) ? (blockSize - processed) : int(samplesToBoundary); for (int i = 0; i < n; i++) { int idx = int(phaseAcc >> FIXED_SHIFT); float frac = float(phaseAcc & FIXED_MASK) * invFixedOne; float sampL = sourceL[idx] + frac * (sourceL[idx + 1] - sourceL[idx]); float sampR = sourceR[idx] + frac * (sourceR[idx + 1] - sourceR[idx]); outL[processed + i] += sampL * ampLeft; outR[processed + i] += sampR * ampRight; phaseAcc += phaseInc; } processed += n; } else { if (phaseAcc < fixedLoopStart) { int64_t samplesToBoundary = (fixedLoopStart - phaseAcc + phaseInc - 1) / phaseInc; int n = (samplesToBoundary > (blockSize - processed)) ? (blockSize - processed) : int(samplesToBoundary); for (int i = 0; i < n; i++) { int idx = int(phaseAcc >> FIXED_SHIFT); float frac = float(phaseAcc & FIXED_MASK) * invFixedOne; float sampL = sourceL[idx] + frac * (sourceL[idx + 1] - sourceL[idx]); float sampR = sourceR[idx] + frac * (sourceR[idx + 1] - sourceR[idx]); outL[processed + i] += sampL * ampLeft; outR[processed + i] += sampR * ampRight; phaseAcc += phaseInc; } processed += n; } else if (phaseAcc >= fixedEnd) { int64_t excess = phaseAcc - fixedEnd; phaseAcc = fixedLoopStart + fixedX + excess; continue; } else if (phaseAcc < fixedCrossfadeStart) { int64_t samplesToBoundary = (fixedCrossfadeStart - phaseAcc + phaseInc - 1) / phaseInc; int n = (samplesToBoundary > (blockSize - processed)) ? (blockSize - processed) : int(samplesToBoundary); for (int i = 0; i < n; i++) { int idx = int(phaseAcc >> FIXED_SHIFT); float frac = float(phaseAcc & FIXED_MASK) * invFixedOne; float sampL = sourceL[idx] + frac * (sourceL[idx + 1] - sourceL[idx]); float sampR = sourceR[idx] + frac * (sourceR[idx + 1] - sourceR[idx]); outL[processed + i] += sampL * ampLeft; outR[processed + i] += sampR * ampRight; phaseAcc += phaseInc; } processed += n; } else { int64_t samplesToBoundary = (fixedEnd - phaseAcc + phaseInc - 1) / phaseInc; int n = (samplesToBoundary > (blockSize - processed)) ? (blockSize - processed) : int(samplesToBoundary); if (preXfadeBuffer) { const float* xfadeL = preXfadeBuffer->getReadPointer(0); const float* xfadeR = preXfadeBuffer->getReadPointer(1); int xfadeBufferLen = preXfadeBuffer->getNumSamples(); for (int i = 0; i < n; i++) { int idxPhase = int(phaseAcc >> FIXED_SHIFT); float frac = float(phaseAcc & FIXED_MASK) * invFixedOne; float currentP = float(phaseAcc) * invFixedOne; float posInXfade = currentP - crossfadeStartF; int idx = int(posInXfade); float subFrac = posInXfade - idx; if (idx < 0) idx = 0; else if (idx >= xfadeBufferLen - 1) idx = xfadeBufferLen - 2; float sampL = xfadeL[idx] + subFrac * (xfadeL[idx + 1] - xfadeL[idx]); float sampR = xfadeR[idx] + subFrac * (xfadeR[idx + 1] - xfadeR[idx]); outL[processed + i] += sampL * ampLeft; outR[processed + i] += sampR * ampRight; phaseAcc += phaseInc; } } else { for (int i = 0; i < n; i++) { int idxPhase = int(phaseAcc >> FIXED_SHIFT); float frac = float(phaseAcc & FIXED_MASK) * invFixedOne; float currentP = float(phaseAcc) * invFixedOne; float alpha = (currentP - crossfadeStartF) / X; float crossAngle = alpha * piOverTwo; float tailGain = std::cos(crossAngle); float headGain = std::sin(crossAngle); float sampTailL = sourceL[idxPhase] + frac * (sourceL[idxPhase + 1] - sourceL[idxPhase]); float sampTailR = sourceR[idxPhase] + frac * (sourceR[idxPhase + 1] - sourceR[idxPhase]); float headPos = float(fixedLoopStart) * invFixedOne + (currentP - crossfadeStartF); int idxHead = int(headPos); float fracHead = headPos - idxHead; float sampHeadL = sourceL[idxHead] + fracHead * (sourceL[idxHead + 1] - sourceL[idxHead]); float sampHeadR = sourceR[idxHead] + fracHead * (sourceR[idxHead + 1] - sourceR[idxHead]); float mixL = tailGain * sampTailL + headGain * sampHeadL; float mixR = tailGain * sampTailR + headGain * sampHeadR; outL[processed + i] += mixL * ampLeft; outR[processed + i] += mixR * ampRight; phaseAcc += phaseInc; } } processed += n; } } } return processed; } }; struct Voice { int midiNote = 60; bool isActive = false; float velocity = 1.0f; SamplePlayback playback; void reset(int note, float vel, const std::array<const float*, 2>& sample, int bufferLength, double baseDelta, const SampleSettings& settings) { midiNote = note; velocity = vel; isActive = true; new (&playback) SamplePlayback(sample, velocity, settings, bufferLength, baseDelta); } }; template <int NV> struct Griffin_Sampler : public data::base { SNEX_NODE(Griffin_Sampler); struct MetadataClass { SN_NODE_ID("Griffin_Sampler"); }; static constexpr bool isModNode() { return false; } static constexpr bool isPolyphonic() { return NV > 1; } static constexpr bool hasTail() { return true; } static constexpr bool isSuspendedOnSilence() { return false; } static constexpr int getFixChannelAmount() { return 2; } static constexpr int NumTables = 0; static constexpr int NumSliderPacks = 0; static constexpr int NumAudioFiles = 1; static constexpr int NumFilters = 0; static constexpr int NumDisplayBuffers = 0; PolyData<Voice, NV> voices; ExternalData sampleData; AudioBuffer<float> sampleBuffer; std::array<const float*, 2> sample{ nullptr, nullptr }; std::array<float, 128> pitchRatios{}; double sampleRate = 44100.0; double sampleRateRatio = 1.0; float sampleStartPercent = 0.0f; float loopStartPercent = 0.0f; float loopEndPercent = 1.0f; float playbackStartOffsetInSamples = 0.0f; float loopStartOffsetInSamples = 0.0f; float loopEndOffsetInSamples = 0.0f; double globalPitchOffsetFactor = 1.0; float xfadeFraction = 0.0f; float xfadeLengthInSamples = 0.0f; bool loopMode = false; std::mt19937 randomGen; std::atomic<AudioBuffer<float>*> precomputedXfadeBuffer{ nullptr }; void setExternalData(const ExternalData& ed, int) { sampleData = ed; AudioSampleBuffer tempBuf = ed.toAudioSampleBuffer(); int numSamples = tempBuf.getNumSamples(); int numChannels = tempBuf.getNumChannels(); if (numSamples <= 0) { int fallbackLen = 8; int chs = (numChannels > 0 ? numChannels : 2); AudioSampleBuffer fallback(chs, fallbackLen); fallback.clear(); sampleBuffer.makeCopyOf(fallback, true); } else { sampleBuffer.makeCopyOf(tempBuf, true); } sample[0] = sampleBuffer.getReadPointer(0); if (numChannels > 1) sample[1] = sampleBuffer.getReadPointer(1); else sample[1] = sample[0]; updateDerivedParameters(); } void updateDerivedParameters() { int currentSampleLength = sampleBuffer.getNumSamples(); if (currentSampleLength < 1) currentSampleLength = 1; playbackStartOffsetInSamples = sampleStartPercent * float(currentSampleLength - 1); loopStartOffsetInSamples = loopStartPercent * float(currentSampleLength - 1); loopEndOffsetInSamples = loopEndPercent * float(currentSampleLength - 1); float regionLen = loopEndOffsetInSamples - loopStartOffsetInSamples; if (regionLen < 0.f) regionLen = 0.f; float maxXfade = regionLen * 0.5f; float desiredXfade = xfadeFraction * regionLen; if (desiredXfade > maxXfade) desiredXfade = maxXfade; xfadeLengthInSamples = desiredXfade; if (loopMode && xfadeLengthInSamples > 0.f && sampleBuffer.getNumSamples() > 0) { int xfadeSamples = std::max(1, (int)std::round(xfadeLengthInSamples)); auto* newXfadeBuffer = new AudioBuffer<float>(2, xfadeSamples); for (int ch = 0; ch < 2; ++ch) { float* dest = newXfadeBuffer->getWritePointer(ch); const float* src = sampleBuffer.getReadPointer(std::min(ch, sampleBuffer.getNumChannels() - 1)); for (int i = 0; i < xfadeSamples; i++) { float pos = float(i); float alpha = (xfadeSamples > 1 ? pos / float(xfadeSamples - 1) : 0.f); float tailGain = std::cos(alpha * (float(M_PI) * 0.5f)); float headGain = std::sin(alpha * (float(M_PI) * 0.5f)); float tailPos = loopEndOffsetInSamples - xfadeLengthInSamples + pos; float headPos = loopStartOffsetInSamples + pos; int tailIdx = int(tailPos); int headIdx = int(headPos); float tailFrac = tailPos - tailIdx; float headFrac = headPos - headIdx; int numSamples = sampleBuffer.getNumSamples(); int tailIdx1 = std::min(tailIdx + 1, numSamples - 1); int headIdx1 = std::min(headIdx + 1, numSamples - 1); float tailSample = src[tailIdx] + tailFrac * (src[tailIdx1] - src[tailIdx]); float headSample = src[headIdx] + headFrac * (src[headIdx1] - src[headIdx]); dest[i] = tailGain * tailSample + headGain * headSample; } } AudioBuffer<float>* oldBuffer = precomputedXfadeBuffer.exchange(newXfadeBuffer); if (oldBuffer) delete oldBuffer; } else { AudioBuffer<float>* oldBuffer = precomputedXfadeBuffer.exchange(nullptr); if (oldBuffer) delete oldBuffer; } } void reset() { for (auto& voice : voices) voice.isActive = false; } void prepare(PrepareSpecs specs) { sampleRate = specs.sampleRate; initPitchRatios(); updateDerivedParameters(); voices.prepare(specs); std::random_device rd; randomGen.seed(rd()); } void handleHiseEvent(HiseEvent& e) { if (e.isNoteOn()) { auto& voice = voices.get(); double baseDelta = pitchRatios[e.getNoteNumber()] * sampleRateRatio * globalPitchOffsetFactor; SampleSettings settings; settings.pitchOffsetCents = 0.0; settings.volumeMult = 1.0f; settings.panning = 0.0f; settings.playbackStartInSamples = playbackStartOffsetInSamples; settings.loopStartInSamples = loopStartOffsetInSamples; settings.loopEndInSamples = loopEndOffsetInSamples; settings.loopMode = loopMode; settings.xfadeLengthInSamples = xfadeLengthInSamples; voice.reset(e.getNoteNumber(), e.getFloatVelocity(), sample, sampleBuffer.getNumSamples(), baseDelta, settings); } } template <typename ProcessDataType> void process(ProcessDataType& data) { auto& fixData = data.template as<ProcessData<getFixChannelAmount()>>(); auto audioBlock = fixData.toAudioBlock(); auto* leftChannel = audioBlock.getChannelPointer(0); auto* rightChannel = audioBlock.getChannelPointer(1); int totalSamples = data.getNumSamples(); if (sampleBuffer.getNumSamples() == 0) { audioBlock.clear(); return; } // Pre-clear output buffers std::fill(leftChannel, leftChannel + totalSamples, 0.f); std::fill(rightChannel, rightChannel + totalSamples, 0.f); AudioBuffer<float>* currentXfadeBuffer = precomputedXfadeBuffer.load(); for (auto& voice : voices) { if (!voice.isActive) continue; int n = voice.playback.vectorSynthesize(leftChannel, rightChannel, totalSamples, currentXfadeBuffer, xfadeLengthInSamples); if (!voice.playback.active) voice.isActive = false; } // Lock external sample data for thread-safe UI update. { DataReadLock lock(sampleData); bool uiUpdated = false; for (auto& voice : voices) { if (voice.isActive) { // Convert fixed-point phase accumulator to a float sample position. float currentPos = float(voice.playback.phaseAcc) / float(FIXED_ONE); sampleData.setDisplayedValue(currentPos); uiUpdated = true; break; } } if (!uiUpdated) sampleData.setDisplayedValue(0.0); } } template <typename FrameDataType> void processFrame(FrameDataType&) {} template <int P> void setParameter(double v) { if constexpr (P == 0) { globalPitchOffsetFactor = std::pow(2.0, v / 12.0); for (auto& voice : voices) { if (voice.isActive) { double newBaseDelta = pitchRatios[voice.midiNote] * sampleRateRatio * globalPitchOffsetFactor; double centsFact = std::pow(2.0, (voice.playback.settings.pitchOffsetCents / 1200.0)); double effectiveSpeed = newBaseDelta * centsFact; voice.playback.phaseInc = (int64_t)std::llround(effectiveSpeed * FIXED_ONE); } } } else if constexpr (P == 1) { sampleStartPercent = (float)v; updateDerivedParameters(); } else if constexpr (P == 2) { loopStartPercent = (float)v; updateDerivedParameters(); } else if constexpr (P == 3) { loopEndPercent = (float)v; updateDerivedParameters(); } else if constexpr (P == 4) { xfadeFraction = (float)v; if (xfadeFraction < 0.f) xfadeFraction = 0.f; if (xfadeFraction > 1.f) xfadeFraction = 1.f; updateDerivedParameters(); } else if constexpr (P == 5) { loopMode = (v >= 0.5); updateDerivedParameters(); } if constexpr (P >= 1 && P <= 5) { for (auto& voice : voices) { if (voice.isActive) { SampleSettings newSettings; newSettings.pitchOffsetCents = voice.playback.settings.pitchOffsetCents; newSettings.volumeMult = voice.playback.settings.volumeMult; newSettings.panning = voice.playback.settings.panning; newSettings.playbackStartInSamples = playbackStartOffsetInSamples; newSettings.loopStartInSamples = loopStartOffsetInSamples; newSettings.loopEndInSamples = loopEndOffsetInSamples; newSettings.loopMode = loopMode; newSettings.xfadeLengthInSamples = xfadeLengthInSamples; voice.playback.updateSettings(newSettings, sampleBuffer.getNumSamples()); } } } } void initPitchRatios() { for (int i = 0; i < 128; ++i) pitchRatios[i] = std::pow(2.0f, float(i - 60) / 12.0f); } void createParameters(ParameterDataList& data) { { parameter::data pitchParam("Pitch (semitones)", { -48.0, 24.0, 0.01 }); registerCallback<0>(pitchParam); pitchParam.setDefaultValue(0.0); data.add(std::move(pitchParam)); } { parameter::data startParam("Playhead Start", { 0.0, 1.0, 0.00001 }); registerCallback<1>(startParam); startParam.setDefaultValue(0.0); data.add(std::move(startParam)); } { parameter::data loopStartParam("Loop Start", { 0.0, 1.0, 0.00001 }); registerCallback<2>(loopStartParam); loopStartParam.setDefaultValue(0.0); data.add(std::move(loopStartParam)); } { parameter::data loopEndParam("Sample End", { 0.0, 1.0, 0.00001 }); registerCallback<3>(loopEndParam); loopEndParam.setDefaultValue(1.0); data.add(std::move(loopEndParam)); } { parameter::data xfadeParam("Xfade Length", { 0.0, 1.0, 0.0001 }); registerCallback<4>(xfadeParam); xfadeParam.setDefaultValue(0.0); data.add(std::move(xfadeParam)); } { parameter::data loopModeParam("Loop Mode", { 0.0, 1.0, 1.0 }); registerCallback<5>(loopModeParam); loopModeParam.setDefaultValue(0.0); data.add(std::move(loopModeParam)); } } }; } // namespace project -
@griffinboy any chance you've got a version to share? I'm trying to get something like this going myself.
Not experienced in c++ at all tho. -
Ah I don't unfortunately, not anything I can share freely at least.
You can paste the code I sent above, into chat gpt and ask it questions about the workings of it
