Transverb: various approaches to adapt to "distance" changes without …

…crackly glitches

transverb/transverb-base.h

@@ -48,6 +48,7 @@ enum : dfx::ParameterID
  kTomsound,
  kFreeze,
  kAttenuateFeedbackByMixLevel,
+ kDistChangeMode,
 
  kNumParameters
 };
@@ -66,6 +67,8 @@ enum { kQualityMode_DirtFi, kQualityMode_HiFi, kQualityMode_UltraHiFi, kQualityM
 enum : unsigned int { kSpeedMode_Fine, kSpeedMode_Semitone, kSpeedMode_Octave, kSpeedMode_NumModes };
 static constexpr dfx::PropertyID kTransverbProperty_SpeedModeBase = dfx::kPluginProperty_EndOfList;
 
+enum { kDistChangeMode_Reverse, kDistChangeMode_AdHocVarispeed, kDistChangeMode_DistanceVarispeed, kDistChangeMode_BufferVarispeed, kDistChangeMode_LoopingBufferVarispeed, kDistChangeMode_Count };
+
 
 dfx::PropertyID speedModeIndexToPropertyID(size_t inIndex) noexcept;
 size_t speedModePropertyIDToIndex(dfx::PropertyID inPropertyID) noexcept;

transverb/transverb.h

@@ -26,6 +26,7 @@ To contact the author, use the contact form at http:https://destroyfx.org
 #include <cmath>
 #include <cstdint>
 #include <numeric>
+#include <optional>
 #include <vector>
 
 #include "dfxplugin.h"
@@ -57,6 +58,8 @@ class TransverbDSP final : public DfxPluginCore {
  dfx::SmoothedValue<float> mix, feed;
 
  double read = 0.;
+ std::optional<double> targetdist;
+ double distspeedfactor = 1.;
  std::vector<float> buf;
 
  dfx::IIRFilter filter;
@@ -83,17 +86,23 @@ class TransverbDSP final : public DfxPluginCore {
  static constexpr int mod_bipolar(int value, int modulo);
  static inline double fmod_bipolar(double value, double modulo);
 
+ // distance in samples of a read position from the current write position
+ double getdist(double read) const;
+ void processdist(double distnormalized, Head& head);
+
  // these store the parameter values
  int bsize = 0;
  dfx::SmoothedValue<float> drymix;
- long quality = 0;
- bool tomsound = false;
+ int distchangemode {};
 
  int writer = 0;
  std::array<Head, dfx::TV::kNumDelays> heads;
 
  int const MAXBUF; // the size of the audio buffer (dependent on sampling rate)
 
+ bool firstrendersincereset = false;
+ std::vector<float>& buftemp;
+
  std::vector<float> const firCoefficientsWindow;
 };
 
@@ -116,6 +125,10 @@ class Transverb final : public DfxPlugin {
  dfx::StatusCode dfx_SetProperty(dfx::PropertyID inPropertyID, dfx::Scope inScope, unsigned int inItemIndex,
  void const* inData, size_t inDataSize) override;
 
+ auto& getscratchbuffer() noexcept {
+ return buftemp;
+ }
+
 protected:
  size_t settings_sizeOfExtendedData() const noexcept override;
  void settings_saveExtendedData(void* outData, bool isPreset) override;
@@ -132,6 +145,7 @@ class Transverb final : public DfxPlugin {
  }
 
  std::array<uint32_t, dfx::TV::kNumDelays> speedModeStates {};
+ std::vector<float> buftemp; // shared between all DSP cores (memory optimization)
 };
 
 

transverb/transverbformalities.cpp

@@ -57,6 +57,7 @@ Transverb::Transverb(TARGET_API_BASE_INSTANCE_TYPE inInstance)
  initparameter_b(kTomsound, {"TOMSOUND", "TomSnd", "Tom7"}, false);
  initparameter_b(kFreeze, dfx::MakeParameterNames(dfx::kParameterNames_Freeze), false);
  initparameter_b(kAttenuateFeedbackByMixLevel, {"attenuate feedback by mix level", "AtnFdbk", "AtnFdb", "-fdb"}, false);
+ initparameter_list(kDistChangeMode, {"distance change mode", "DistMod", "DstMod", "DMod"}, kDistChangeMode_Reverse, kDistChangeMode_Reverse, kDistChangeMode_Count);
 
  setparameterenforcevaluelimits(kBsize, true);
  for (auto const parameterID : kDistParameters) {
@@ -67,10 +68,12 @@ Transverb::Transverb(TARGET_API_BASE_INSTANCE_TYPE inInstance)
  setparametervaluestring(kQuality, kQualityMode_HiFi, "hi-fi");
  setparametervaluestring(kQuality, kQualityMode_UltraHiFi, "ultra hi-fi");
 
- // distance parameters only have meaningful effect at zero speed which probably will never occur randomly,
- // and otherwise all they do is glitch a lot, so omit them
- addparameterattributes(kDist1, DfxParam::kAttribute_OmitFromRandomizeAll);
- addparameterattributes(kDist2, DfxParam::kAttribute_OmitFromRandomizeAll);
+ setparametervaluestring(kDistChangeMode, kDistChangeMode_Reverse, "reverse");
+ setparametervaluestring(kDistChangeMode, kDistChangeMode_AdHocVarispeed, "ad hoc varispeed");
+ setparametervaluestring(kDistChangeMode, kDistChangeMode_DistanceVarispeed, "distance varispeed");
+ setparametervaluestring(kDistChangeMode, kDistChangeMode_BufferVarispeed, "buffer varispeed");
+ setparametervaluestring(kDistChangeMode, kDistChangeMode_LoopingBufferVarispeed, "looping buffer varispeed");
+
  addparameterattributes(kFreeze, DfxParam::kAttribute_OmitFromRandomizeAll);
  addparameterattributes(kAttenuateFeedbackByMixLevel, DfxParam::kAttribute_OmitFromRandomizeAll);
 
@@ -109,8 +112,11 @@ void Transverb::dfx_PostConstructor() {
 TransverbDSP::TransverbDSP(DfxPlugin* inDfxPlugin)
  : DfxPluginCore(inDfxPlugin),
  MAXBUF(static_cast<int>(getparametermax_f(kBsize) * 0.001 * getsamplerate())),
+ buftemp(dynamic_cast<Transverb*>(inDfxPlugin)->getscratchbuffer()),
  firCoefficientsWindow(dfx::FIRFilter::generateKaiserWindow(kNumFIRTaps, 60.0f)) {
 
+ buftemp.assign(MAXBUF, 0.f);
+
  registerSmoothedAudioValue(&drymix);
 
  for (auto& head : heads) {
@@ -126,12 +132,15 @@ TransverbDSP::TransverbDSP(DfxPlugin* inDfxPlugin)
 void TransverbDSP::reset() {
 
  std::for_each(heads.begin(), heads.end(), [](Head& head){ head.reset(); });
+
+ firstrendersincereset = true;
 }
 
 void TransverbDSP::Head::reset() {
 
  smoothcount = 0;
  lastdelayval = 0.f;
+ targetdist.reset();
  filter.reset();
  speedHasChanged = true;
 
@@ -164,8 +173,6 @@ void TransverbDSP::processparameters() {
  heads[head].feed = *value;
  }
  }
- quality = getparameter_i(kQuality);
- tomsound = getparameter_b(kTomsound);
 
  if (auto const value = getparameterifchanged_f(kBsize))
  {
@@ -193,12 +200,12 @@ void TransverbDSP::processparameters() {
  std::for_each(heads.begin(), heads.end(), [bsize_f](Head& head){ head.read = fmod_bipolar(head.read, bsize_f); });
  }
 
+ distchangemode = getparameter_i(kDistChangeMode);
  for (size_t head = 0; head < kNumDelays; head++)
  {
  if (auto const dist = getparameterifchanged_f(kDistParameters[head]))
  {
- auto const bsize_f = static_cast<double>(bsize);
- heads[head].read = fmod_bipolar(static_cast<double>(writer) - (*dist * bsize_f), bsize_f);
+ processdist(*dist, heads[head]);
  }
  }
 
@@ -221,6 +228,64 @@ void TransverbDSP::processparameters() {
  heads[0].mix.setValueNow(getparametermin_f(kMix1));
  }
  }
+
+ firstrendersincereset = false;
+}
+
+double TransverbDSP::getdist(double read) const {
+
+ return fmod_bipolar(static_cast<double>(writer) - read, static_cast<double>(bsize));
+}
+
+void TransverbDSP::processdist(double distnormalized, Head& head) {
+
+ auto const bsize_f = static_cast<double>(bsize);
+ auto const distsamples = distnormalized * bsize_f;
+ auto const targetread = fmod_bipolar(static_cast<double>(writer) - distsamples, bsize_f);
+
+ if (firstrendersincereset)
+ {
+ head.read = targetread;
+ }
+ else
+ {
+ if (distchangemode == kDistChangeMode_Reverse)
+ {
+ head.targetdist = distsamples;
+ }
+ else if (auto const resamplerate = getdist(head.read) / std::max(distsamples, 1.); distchangemode == kDistChangeMode_AdHocVarispeed)
+ {
+ head.targetdist = distsamples;
+ head.distspeedfactor = resamplerate;
+ head.speedHasChanged = true;
+ }
+ else
+ {
+ constexpr double modulationthresholdsamples = 1.;
+ if (std::fabs(targetread - head.read) >= modulationthresholdsamples)
+ {
+ auto const bsizesteps = bsize_f / resamplerate;
+ bool const subslice = (distchangemode == kDistChangeMode_DistanceVarispeed);
+ bool const looping = (distchangemode == kDistChangeMode_LoopingBufferVarispeed);
+ auto const copylength_f = subslice ? distsamples : (looping ? bsize_f : std::min(bsize_f, bsizesteps));
+ auto const copylength = static_cast<size_t>(std::lround(copylength_f));
+ assert(copylength <= buftemp.size());
+ assert(static_cast<int>(copylength) <= bsize);
+ auto const sourcestart = subslice ? head.read : fmod_bipolar(static_cast<double>(writer) - (copylength_f * resamplerate), bsize_f);
+ for (size_t i = 0; i < copylength; i++)
+ {
+ auto const sourcepos = std::fmod(sourcestart + (resamplerate * static_cast<double>(i)), bsize_f);
+ buftemp[i] = interpolateHermite(head.buf.data(), sourcepos, bsize, writer);
+ }
+ auto const destinationstart = subslice ? std::lround(targetread) : mod_bipolar(writer - static_cast<int>(copylength), bsize);
+ auto const copylength1 = std::min(static_cast<size_t>(bsize - destinationstart), copylength);
+ auto const copylength2 = copylength - copylength1;
+ std::copy_n(buftemp.cbegin(), copylength1, std::next(head.buf.begin(), destinationstart));
+ std::copy_n(std::next(buftemp.cbegin(), copylength1), copylength2, head.buf.begin());
+ }
+ head.read = targetread;
+ }
+ }
 }
 
 

transverb/transverbprocess.cpp

@@ -36,11 +36,13 @@ using namespace dfx::TV;
 
 void TransverbDSP::process(float const* inAudio, float* outAudio, size_t numSampleFrames) {
 
- std::array<float, kNumDelays> delayvals {}; // delay buffer output values
  auto const bsize_float = static_cast<double>(bsize); // cut down on casting
+ auto const quality = getparameter_i(kQuality);
+ auto const tomsound = getparameter_b(kTomsound);
  auto const freeze = getparameter_b(kFreeze);
  int const writerIncrement = freeze ? 0 : 1;
  auto const attenuateFeedbackByMixLevel = getparameter_b(kAttenuateFeedbackByMixLevel);
+ std::array<float, kNumDelays> delayvals {}; // delay buffer output values
 
 
  ///////////// S O P H I A S O U N D //////////////
@@ -69,6 +71,9 @@ void TransverbDSP::process(float const* inAudio, float* outAudio, size_t numSamp
  for (size_t h = 0; h < kNumDelays; h++) // delay heads loop
  {
  auto const read_int = static_cast<int>(heads[h].read);
+ auto const reverseread = (distchangemode == kDistChangeMode_Reverse) && heads[h].targetdist.has_value();
+ auto const distcatchup = (distchangemode == kDistChangeMode_AdHocVarispeed) && heads[h].targetdist.has_value();
+ auto const speed = heads[h].speed.getValue() * (distcatchup ? heads[h].distspeedfactor : 1.);
 
  // filter setup
  if (quality == kQualityMode_UltraHiFi)
@@ -77,27 +82,27 @@ void TransverbDSP::process(float const* inAudio, float* outAudio, size_t numSamp
  {
  lowpasspos[h] = read_int;
  // check to see if we need to lowpass the first delay head and init coefficients if so
- if (heads[h].speed.getValue() > kUnitySpeed)
+ if (speed > kUnitySpeed)
  {
  filtermodes[h] = FilterMode::LowpassIIR;
- speed_ints[h] = static_cast<int>(heads[h].speed.getValue());
+ speed_ints[h] = static_cast<int>(speed);
  // it becomes too costly to try to IIR at higher speeds, so switch to FIR filtering
- if (heads[h].speed.getValue() >= kFIRSpeedThreshold)
+ if (speed >= kFIRSpeedThreshold)
  {
  filtermodes[h] = FilterMode::LowpassFIR;
  // compensate for gain lost from filtering
- mugs[h] = static_cast<float>(std::pow(heads[h].speed.getValue() / kFIRSpeedThreshold, 0.78));
+ mugs[h] = static_cast<float>(std::pow(speed / kFIRSpeedThreshold, 0.78));
  // update the coefficients only if necessary
  if (std::exchange(heads[h].speedHasChanged, false))
  {
- dfx::FIRFilter::calculateIdealLowpassCoefficients((samplerate / heads[h].speed.getValue()) * dfx::FIRFilter::kShelfStartLowpass,
+ dfx::FIRFilter::calculateIdealLowpassCoefficients((samplerate / speed) * dfx::FIRFilter::kShelfStartLowpass,
  samplerate, heads[h].firCoefficients, firCoefficientsWindow);
  heads[h].filter.reset();
  }
  }
  else if (std::exchange(heads[h].speedHasChanged, false))
  {
- heads[h].filter.setLowpassCoefficients((samplerate / heads[h].speed.getValue()) * dfx::IIRFilter::kShelfStartLowpass);
+ heads[h].filter.setLowpassCoefficients((samplerate / speed) * dfx::IIRFilter::kShelfStartLowpass);
  }
  }
  // we need to highpass the delay head to remove mega sub bass
@@ -106,7 +111,7 @@ void TransverbDSP::process(float const* inAudio, float* outAudio, size_t numSamp
  filtermodes[h] = FilterMode::Highpass;
  if (std::exchange(heads[h].speedHasChanged, false))
  {
- heads[h].filter.setHighpassCoefficients(kHighpassFilterCutoff / heads[h].speed.getValue());
+ heads[h].filter.setHighpassCoefficients(kHighpassFilterCutoff / speed);
  }
  }
  }
@@ -173,28 +178,40 @@ void TransverbDSP::process(float const* inAudio, float* outAudio, size_t numSamp
  // start smoothing if the writer has passed a reader or vice versa,
  // though not if reader and writer move at the same speed
  // (check the positions before wrapping around the heads)
- auto const nextRead = static_cast<int>(heads[h].read + heads[h].speed.getValue());
+ auto const nextRead = static_cast<int>(heads[h].read + speed);
  auto const nextWrite = writer + 1;
  bool const readCrossingAhead = (read_int < writer) && (nextRead >= nextWrite);
  bool const readCrossingBehind = (read_int >= writer) && (nextRead <= nextWrite);
- bool const speedIsUnity = heads[h].speed.getValue() == kUnitySpeed;
+ bool const speedIsUnity = (speed == kUnitySpeed);
  if ((readCrossingAhead || readCrossingBehind) && !speedIsUnity) {
  // check because, at slow speeds, it's possible to go into this twice or more in a row
  if (heads[h].smoothcount <= 0) {
  // store the most recent output as the head's smoothing sample
  heads[h].lastdelayval = delayvals[h];
  // truncate the smoothing duration if we're using too small of a buffer size
- auto const bufferReadSteps = static_cast<int>(bsize_float / heads[h].speed.getValue());
+ auto const bufferReadSteps = static_cast<int>(bsize_float / speed);
  auto const smoothdur = std::min(bufferReadSteps, kAudioSmoothingDur_samples);
  heads[h].smoothstep = 1.f / static_cast<float>(smoothdur); // the scalar step value
  heads[h].smoothcount = smoothdur; // set the counter to the total duration
  }
  }
 
  // update read heads, wrapping around if they have gone past the end of the buffer
- heads[h].read += heads[h].speed.getValue();
- if (heads[h].read >= bsize_float) {
- heads[h].read = fmod_bipolar(heads[h].read, bsize_float);
+ if (reverseread) {
+ heads[h].read -= speed;
+ while (heads[h].read < 0.) {
+ heads[h].read += bsize_float;
+ }
+ } else {
+ heads[h].read += speed;
+ if (heads[h].read >= bsize_float) {
+ heads[h].read = fmod_bipolar(heads[h].read, bsize_float);
+ }
+ }
+ if (distcatchup || reverseread) {
+ if (std::fabs(getdist(heads[h].read) - *heads[h].targetdist) < speed) {
+ heads[h].targetdist.reset();
+ }
  }
 
  // if we're doing IIR lowpass filtering,
@@ -204,40 +221,47 @@ void TransverbDSP::process(float const* inAudio, float* outAudio, size_t numSamp
  if (filtermodes[h] == FilterMode::LowpassIIR)
  {
  int lowpasscount = 0;
+ int const direction = reverseread ? -1 : 1;
  while (lowpasscount < speed_ints[h])
  {
  switch (speed_ints[h] - lowpasscount)
  {
  case 1:
  heads[h].filter.processToCacheH1(heads[h].buf[lowpasspos[h]]);
- lowpasspos[h] = (lowpasspos[h] + 1) % bsize;
+ lowpasspos[h] = mod_bipolar(lowpasspos[h] + (1 * direction), bsize);
  lowpasscount++;
  break;
  case 2:
  heads[h].filter.processToCacheH2(heads[h].buf.data(), dfx::math::ToUnsigned(lowpasspos[h]), dfx::math::ToUnsigned(bsize));
- lowpasspos[h] = (lowpasspos[h] + 2) % bsize;
+ lowpasspos[h] = mod_bipolar(lowpasspos[h] + (2 * direction), bsize);
  lowpasscount += 2;
  break;
  case 3:
  heads[h].filter.processToCacheH3(heads[h].buf.data(), dfx::math::ToUnsigned(lowpasspos[h]), dfx::math::ToUnsigned(bsize));
- lowpasspos[h] = (lowpasspos[h] + 3) % bsize;
+ lowpasspos[h] = mod_bipolar(lowpasspos[h] + (3 * direction), bsize);
  lowpasscount += 3;
  break;
  default:
  heads[h].filter.processToCacheH4(heads[h].buf.data(), dfx::math::ToUnsigned(lowpasspos[h]), dfx::math::ToUnsigned(bsize));
- lowpasspos[h] = (lowpasspos[h] + 4) % bsize;
+ lowpasspos[h] = mod_bipolar(lowpasspos[h] + (4 * direction), bsize);
  lowpasscount += 4;
  break;
  }
  }
  auto const nextread_int = static_cast<int>(heads[h].read);
  // check whether we need to consume one more sample
- bool const extrasample = ((lowpasspos[h] < nextread_int) && ((lowpasspos[h] + 1) == nextread_int)) ||
- ((lowpasspos[h] == (bsize - 1)) && (nextread_int == 0));
+ bool const extrasample = [=] {
+ if (reverseread) {
+ return ((lowpasspos[h] > nextread_int) && ((lowpasspos[h] - 1) == nextread_int)) ||
+ ((lowpasspos[h] == 0) && (nextread_int == (bsize - 1)));
+ }
+ return ((lowpasspos[h] < nextread_int) && ((lowpasspos[h] + 1) == nextread_int)) ||
+ ((lowpasspos[h] == (bsize - 1)) && (nextread_int == 0));
+ }();
  if (extrasample)
  {
  heads[h].filter.processToCacheH1(heads[h].buf[lowpasspos[h]]);
- lowpasspos[h] = (lowpasspos[h] + 1) % bsize;
+ lowpasspos[h] = mod_bipolar(lowpasspos[h] + (1 * direction), bsize);
  }
  }
  // it's simpler for highpassing;