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reasampler/src/core/instrument/engine/voice.h
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daniel 3cb22e984d loop: fix the crossfade seam's residual discontinuity, plus six review minors
Normalizes crossfadeWeight over crossfade-1 so the last rendered frame lands at exactly the incoming tap instead of a residual step; corrects the CLAUDE.md invariant and seam test to match. Shares lerpSource/crossfadedSource/maxCrossfade, fixes stale docs/constants, and clears crossfade on the loop-OFF gesture.
2026-07-31 17:59:58 -04:00

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#pragma once
// voice.h — one sounding voice: a repitched, enveloped read over the loaded capture.
//
// The PER-SAMPLE render half (advanceFrame and everything it calls) is defined INLINE here
// on purpose: VoiceEngine::render's inner loop lives in another TU, and with no LTO
// configured an out-of-line render would put a call — and the envelope ticks behind it —
// across a TU boundary on the hottest path in the program. The per-NOTE half (start /
// retune / release / hardStop / presize) is cold enough to live in voice.cpp.
#include <cmath>
#include <cstdint>
#include <vector>
#include "core/audio/peaks.h"
#include "core/instrument/engine/envelopes.h"
#include "core/instrument/engine/filter/filter_params.h"
#include "core/instrument/engine/filter/voice_filter.h"
#include "core/instrument/engine/live_params.h"
#include "core/instrument/engine/loop/loop_span.h"
#include "core/instrument/engine/pitch_shift.h"
#include "core/instrument/engine/play_params.h"
#include "core/instrument/engine/velocity_curve.h"
namespace reasampler {
using audio::AudioSample;
using instrument::engine::PitchShifter;
using instrument::engine::VelocityCurve;
using instrument::engine::VelocityPoint;
using instrument::engine::loop::ResolvedLoop;
using instrument::engine::loop::crossfadeWeight;
using instrument::engine::loop::crossfadedSource;
using instrument::engine::loop::lerpSource;
// 2^((note - rootNote) / 12). note == rootNote -> 1.0. Pure equal temperament; no
// reference-frequency needed.
inline double pitchRatio(int note, int rootNote) {
return std::pow(2.0, static_cast<double>(note - rootNote) / 12.0);
}
// 2^(((note - rootNote) * keyTrack) / 12) — keyTrack scales the semitone offset before the
// ET conversion. keyTrack == 1.0 is bit-identical to pitchRatio(note, rootNote)
// ((note-root)*1.0 is exact in IEEE-754 for an integer-valued double, feeding the same
// std::pow call); 0.0 means every key plays the root pitch; 2.0 doubles the tracking rate.
// At the root note the offset is 0 regardless of keyTrack.
inline double keyTrackedRatio(int note, int rootNote, double keyTrack) {
const double semis = static_cast<double>(note - rootNote) * keyTrack;
return std::pow(2.0, semis / 12.0);
}
// One octave expressed in the cutoff control's normalized domain, read out of the filter
// module's OWN inverse rather than re-derived from its endpoints — the log law belongs to
// filter_params, and a second copy here could drift from it. Evaluated at note-on only.
inline double filterNormPerOctave() {
namespace flt = instrument::engine::filter;
return static_cast<double>(flt::filterNormFromCutoffHz(2.0f * flt::kFilterCutoffMinHz) -
flt::filterNormFromCutoffHz(flt::kFilterCutoffMinHz));
}
// Takeover declick: a restart of a sounding voice (mono retrigger takeover/fallback or a
// poly at-cap steal) hard-cuts the old tone in one frame — a step discontinuity that clicks.
// When the caller opts in (start()'s declickTakeover), start() records the last rendered
// output as a pre-cut reference, and the first frame after the restart seeds a compensation
// equal to (reference - that frame's raw new output), summed in ungated and decaying by
// kDeclickDecay/frame — so the boundary frame reproduces the old level exactly regardless of
// the new envelope's first value, and the residue fades to the -80 dB floor in a few ms.
// An earlier revision gated the compensation by (1 - newAmp): any restart whose new
// amplitude was instantly ~1 (a zero-attack Trigger or Gate) got zero
// compensation and kept the full click — the difference-seed has no such hole. Off by
// default so the bare core stays byte-identical to the pre-fix engine; the processor
// shell opts in.
inline constexpr double kDeclickDecay = 0.95; // per-frame decay of the compensation
inline constexpr double kDeclickFloor = 1e-4; // below this the ramp is done (~ -80 dB)
// A single voice: one active note playing the loaded capture, repitched and enveloped.
// Reads the sample by fractional frame position with linear interpolation, advancing by the
// pitch ratio; loops the sustain region for held notes past the loop end.
class Voice {
public:
// Plays `sample` (a stable reference the caller must keep alive — the engine's loaded
// instrument owns it), repitched from its root by `sample.keyTrack`. Play-mode /
// AHDSR / pitch-engine params are read from sample.play (frames, resolved from stored
// seconds at load). Preserve shifters must already be pre-sized
// (presizePreserveShifters, off-thread) — start() only reset()s + warm()s them (RT-safe,
// no allocation) since it runs on the audio thread inside process(). Byte-identical to
// the bare engine when sample.play is default. `velocityCurve` maps note-on velocity to
// amp gain, evaluated once here (off the per-frame path). `declickTakeover`: when true
// and this voice is currently active (a takeover/steal restart, not a fresh start), arms
// the difference-seeded declick compensation on the first frame after the restart (see
// kDeclickDecay above). A fresh start never declicks.
void start(int note, int velocity, const SampleData& sample, bool declickTakeover = false);
// Mono legato takeover: re-pitch this active voice to `note` without touching the
// amplitude envelope, read position, or shifter state — pitch moves, no re-attack. Both
// engines pick the new baseRatio_ up on the next frame. No-op on an idle voice.
void retune(int note);
// Gate off. In Gate mode enters the AHDSR release; in Trigger mode a no-op (Trigger
// ignores note-off and plays through to its play length).
void release();
// Hard stop (CC 120 semantics): immediately silences this voice regardless of play mode,
// no release ramp. Stops a ringing Trigger one-shot instantly (release() cannot).
// RT-safe: no allocation, no lock.
void hardStop() { active_ = false; }
// True while producing (or about to produce) sound, including any declick ring-out
// tail past the note's playable span.
bool active() const { return active_; }
// True while sounding a playable note — active and the amplitude envelope hasn't
// finished. A voice ringing out a declick tail past note end is active() but not
// soundingNote(); the Preserve-cap count and the mono-legato takeover predicate must
// ignore a ramp-only past-end voice or a new note-on could be dropped/silently muted.
bool soundingNote() const { return active_ && !amplitudeDone_; }
int note() const { return note_; }
// Monotonic age counter for the engine's oldest-first stealing policy. Set by the engine.
std::uint64_t startOrder() const { return startOrder_; }
void setStartOrder(std::uint64_t order) { startOrder_ = order; }
bool releasing() const { return releasing_; }
// The pitch engine this voice is running (for the engine's Preserve-voice tally). Only
// meaningful while active().
PitchEngine pitchEngine() const { return pitchEngine_; }
// Applies the live-parameter block to a voice that is already sounding (or, with `snap`,
// to one just started). Called at BLOCK boundaries by VoiceEngine — never per frame — so
// the per-sample shape is unchanged; every continuous control glides toward its new value
// from here rather than jumping to it. `snap` takes the values outright — glides AND
// envelopes: a fresh note has nothing to glide from, and its copy may predate the edit.
//
// What is NOT here is the point: velocity and its curve result, the note number and the
// pitch ratio, and the decoded PCM stay latched at note-on.
void applyLive(const instrument::engine::LiveValues& live, bool snap);
// Pre-sizes this voice's Preserve pitch shifters (both channels) to `windowFrames`, off
// the audio thread (allocates; also sizes the prime scratch buffer), so start() — which
// runs inside process() — never allocates. <= 1 leaves the shifters pass-through.
// Idempotent: a re-presize to the same window is a cheap no-op.
void presizePreserveShifters(std::int64_t windowFrames);
// Renders one frame's contribution, advancing the read head and envelope by one output
// frame. Returns 0.0 (and goes idle) once the envelope finishes or the sample runs out
// with no loop. Already velocity- and envelope-scaled — the engine sums voices directly.
// Mono path (channel 0 only).
AudioSample renderFrame() {
AudioSample discard = 0.0f;
return advanceFrame(/*stereo=*/false, discard);
}
// Writes this frame's per-channel contribution into `l`/`r` and advances the read head +
// envelope by exactly one frame (the envelope ticks once per frame, shared across both
// channels). A mono sample writes the same value to both (dual-mono/centered). Goes idle
// on the same conditions as the mono path, writing 0 to both.
void renderFrameStereo(AudioSample& l, AudioSample& r) {
r = 0.0f;
l = advanceFrame(/*stereo=*/true, r);
}
private:
// This frame's amplitude in [0,1] from the active envelope. Gate: AHDSR ticks once per
// output frame (envelope time is wall-clock, independent of read rate). Trigger: the AHD
// is evaluated at the source offset (readPos - startFrame) so its stages anchor to source
// frames regardless of pitch engine. Sets amplitudeDone_ on finish so advanceFrame frees
// the voice.
double tickAmplitude() {
double amp;
if (playMode_ == PlayMode::Gate) {
amp = env_.tick();
if (env_.finished()) amplitudeDone_ = true;
} else {
amp = ampAhd_.amplitudeAt(sourceOffset());
if (ampAhd_.finished()) amplitudeDone_ = true;
}
return amp;
}
// Frames into the Trigger play span at the current read head — the domain both
// sustain-less envelopes are evaluated over.
double sourceOffset() const { return readPos_ - static_cast<double>(startFrame_); }
// Advances the filter envelope and re-solves the corner from the modulated cutoff. The
// solve is UNQUANTIZED: the corner tracks the envelope continuously, so a sweep glides
// rather than staircasing. State preservation across the solve is voice_filter's own
// contract (voice_filter.h / filter/CLAUDE.md). Do not reintroduce a step quantizer on the
// control value to save the solve — setCutoffNorm exists to make the solve cheap instead.
//
// Two exact skips, neither of which rounds the control: filterModAmount_ is fixed for the
// note's lifetime, so a zero depth can only ever re-derive the cutoff already solved; and a
// held envelope (sustain, or finished) reproduces the previous position bit-for-bit. Both
// compare the value itself, so they can never suppress a move the ear would hear.
// filterSolved_ == false (forced by start()/retune() via updateFilterCutoffBase) falls
// through both so a moved base always re-solves.
void tickFilterCutoff() {
if (filterModAmount_ == 0.0 && filterSolved_) return;
// The filter envelope takes the amp's shape under the active mode — AHDSR in Gate,
// the source-offset AHD in Trigger. playMode_ is fixed for the note's lifetime, so the
// branch is perfectly predicted.
const double envOut = (playMode_ == PlayMode::Gate)
? filterEnv_.tick()
: filterAhd_.amplitudeAt(sourceOffset());
double cut = static_cast<double>(filterBaseCutoff_) + filterModAmount_ * envOut;
if (cut < 0.0) cut = 0.0;
if (cut > 1.0) cut = 1.0;
const float cutNorm = static_cast<float>(cut);
if (filterSolved_ && cutNorm == filterSolvedCutoff_) return;
filterSolvedCutoff_ = cutNorm;
filterSolved_ = true;
filter_.setCutoffNorm(cutNorm, filterRate_);
}
// The cutoff position before the envelope: the stored knob position plus this note's
// velocity offset and key-tracking. Evaluated at note-on, at a legato retune (both move
// the note), and when a live move changes the knob position or the key-track depth —
// never per frame.
double filterCutoffBaseTarget(int note) const {
double base = filterCutoffNorm_ + filterVelOffset_;
if (filterKeyTrack_ != 0.0 && sample_ != nullptr) {
base += filterKeyTrack_ *
(static_cast<double>(note - sample_->rootNote) / 12.0) *
filterNormPerOctave();
}
if (base < 0.0) base = 0.0;
if (base > 1.0) base = 1.0;
return base;
}
// Takes the base outright (no glide) — a note-on or a retune is a new note position, not a
// knob move, so there is nothing to glide from.
void updateFilterCutoffBase(int note) {
const double base = filterCutoffBaseTarget(note);
rBaseCutoff_.set(base);
filterBaseCutoff_ = static_cast<float>(base);
filterSolved_ = false; // forces the next frame to solve
}
// The full solve, from the tone-control ramps' current values, at the current base cutoff —
// the same shape start() performs, and it leaves the same solved-cutoff bookkeeping behind
// so an unmoved live block reproduces start()'s state exactly. State is preserved across
// prepare() by contract (voice_filter.h), which is what makes a live tone move glide
// rather than click.
void prepareFilterFromRamps() {
filterSettings_.resonanceNorm = static_cast<float>(rResonance_.value);
filterSettings_.morphNorm = static_cast<float>(rMorph_.value);
filterSettings_.driveNorm = static_cast<float>(rDrive_.value);
filterSettings_.cutoffNorm = filterBaseCutoff_;
filter_.prepare(filterSettings_, filterRate_);
filterSolvedCutoff_ = filterBaseCutoff_;
filterSolved_ = true;
}
// Advances the five live filter-control glides by one frame. Q, morph and drive are
// prepare()-cadence constants, so a move on any of them costs the full solve while the
// glide runs (~20 ms) and nothing once it lands; the base cutoff and the mod depth feed
// tickFilterCutoff's own cheap cutoff-only solve instead.
void tickFilterRamps() {
bool tone = false;
if (rResonance_.tick()) tone = true;
if (rMorph_.tick()) tone = true;
if (rDrive_.tick()) tone = true;
if (rBaseCutoff_.tick()) {
filterBaseCutoff_ = static_cast<float>(rBaseCutoff_.value);
filterSolved_ = false;
}
if (rModAmount_.tick()) filterModAmount_ = rModAmount_.value;
if (tone) prepareFilterFromRamps();
filterRamping_ = rResonance_.moving() || rMorph_.moving() || rDrive_.moving() ||
rBaseCutoff_.moving() || rModAmount_.moving();
}
// Seeds the takeover compensation on the first frame after a restart: the ramp is the
// actual discontinuity — (pre-cut reference - the new voice's raw output this frame) —
// applied ungated so the boundary frame reproduces the old level exactly.
void seedDeclick() {
// The weight starts at 1.0 so this frame's output is `out*(1-1) + ref*1 == ref` —
// exact boundary identity whatever the new envelope's first value. Each subsequent
// frame adds `w*(ref outCurrent)` then decays w, so output is provably bounded by
// max(|ref|, |outCurrent|) — mid-ramp overshoot is impossible even if outCurrent
// rises while the weight is still significant. (An earlier revision stored the frozen
// difference (ref x₀), which could exceed full scale if outₙ rose while that
// residue was still large.)
declickPending_ = false;
declickWeight_ = 1.0; // one weight for both channels
// ref is already clamped to ±1.0 at start(). Activate only when it's above the floor —
// if ref ≈ 0 there is nothing to blend.
declickActive_ = (declickRefL_ > kDeclickFloor || declickRefL_ < -kDeclickFloor ||
declickRefR_ > kDeclickFloor || declickRefR_ < -kDeclickFloor);
}
// Rings the voice's last rendered output out instead of hard-cutting it when the read head
// reaches the end of its span, on the PRESERVE path only. Varispeed's final sample is real
// source content at its natural end and its stop is left byte-identical; Preserve's is
// recycled synthetic tail (freezeTail stops the writer a full window before the read head
// arrives), whose level bears no relation to the source's own ending — cutting it at
// whatever amplitude the splice machinery happens to be at is the end-of-sample click.
// Reuses the takeover blend so the boundary frame reproduces the last level exactly.
void seedTerminalDeclick() {
if (pitchEngine_ != PitchEngine::Preserve) return;
declickRefL_ = (lastOutL_ > 1.0) ? 1.0 : (lastOutL_ < -1.0) ? -1.0 : lastOutL_;
declickRefR_ = (lastOutR_ > 1.0) ? 1.0 : (lastOutR_ < -1.0) ? -1.0 : lastOutR_;
declickWeight_ = 1.0;
declickActive_ = (declickRefL_ > kDeclickFloor || declickRefL_ < -kDeclickFloor ||
declickRefR_ > kDeclickFloor || declickRefR_ < -kDeclickFloor);
}
// Shared read/advance for both render paths: computes the interpolated per-channel
// value(s) at the current read head, ticks the amplitude + pitch envelopes once, applies
// the pitch engine, advances the head, and latches idle on exhaustion. `stereo` selects
// whether the second channel is read (into `outR`). Returns the channel-0 value.
//
// INLINE BY CONSTRAINT — see the file header.
AudioSample advanceFrame(bool stereo, AudioSample& outR) {
if (!active_ || sample_ == nullptr) {
if (stereo) outR = 0.0f;
return 0.0f;
}
const std::vector<AudioSample>& pcm = sample_->frames;
const std::int64_t frameCount = static_cast<std::int64_t>(pcm.size());
// Read the second channel only for a genuinely stereo sample; a mono sample plays
// dual-mono (channel 0 duplicated), so `pcmR` aliases channel 0 in that case.
const bool haveR = stereo && sample_->channelCount() == 2;
const std::vector<AudioSample>& pcmR = haveR ? sample_->framesR : pcm;
// Loop-aware sustain (Gate only — Trigger is a one-shot with no sustain loop). The
// span was folded once at note-on (loop_span.h); an invalid or absent loop leaves
// loop_.active false and this whole path off. Under Preserve the loop is over the
// source read (loop the source, shift the output).
const ResolvedLoop& loop = loop_;
if (loop.active) {
const double loopLen = static_cast<double>(loop.length);
while (readPos_ >= static_cast<double>(loop.end)) {
readPos_ -= loopLen; // wrap by exactly one loop length, preserving phase.
}
}
// Trigger frees once the read head reaches playEnd; the envelope also finishes at the
// same count, either latches idle.
const bool triggerRanOff =
playMode_ == PlayMode::Trigger && readPos_ >= static_cast<double>(playEnd_);
// Ran off the sample end with no usable loop -> voice is done, except an in-flight
// takeover declick rings out here instead of hard-cutting — dropping it would
// re-introduce a step on exactly the path the ramp exists for (a restart whose new
// play span ends within the ramp). With no declick (the common case) this is
// byte-identical to the plain idle-out.
if (triggerRanOff || readPos_ >= static_cast<double>(frameCount)) {
// The NOTE is over the moment the read head leaves its span, whether or not a ramp
// still rings: no later frame can carry envelope output. Latching here is what keeps
// a ringing-out voice out of soundingNote() — the Preserve cap would otherwise
// refuse a new onset, and mono legato would retune a voice already past its end
// (silencing the new note) for the whole ~4 ms ramp.
amplitudeDone_ = true;
if (declickPending_) seedDeclick();
if (!declickActive_) seedTerminalDeclick();
if (declickActive_) {
// Bounded blend at silence: outCurrent == 0, so the blend is
// w*(ref 0) == w*ref. The weight decays by kDeclickDecay each frame,
// floor-checked on the weight itself.
const double l = declickWeight_ * declickRefL_;
const double r = declickWeight_ * declickRefR_; // same weight both channels
declickWeight_ *= kDeclickDecay;
if (declickWeight_ < kDeclickFloor && declickWeight_ > -kDeclickFloor) {
declickActive_ = false;
active_ = false;
}
lastOutL_ = l;
lastOutR_ = stereo ? r : l;
if (stereo) outR = static_cast<AudioSample>(r);
return static_cast<AudioSample>(l);
}
active_ = false;
if (stereo) outR = 0.0f;
return 0.0f;
}
// Envelopes tick once per output frame. Pitch envelope biases pitch under either engine.
const double amp = tickAmplitude();
// Peer of the read-head exhaustion path above: a Trigger AHD whose stages end BEFORE
// the play span (a zero decay, which the shape deliberately keeps expressible) cuts the
// same synthetic Preserve tail at whatever level it was at. Seeded from lastOut, which
// still holds the PREVIOUS frame — this one is already silent. Gate is left out of THIS
// site only: its amplitude reaches zero through a release, so nothing here is cut
// mid-level. The exhaustion path above deliberately does NOT exclude Gate — a held Gate
// note whose source runs out with no loop is cut at its sustain level, and under
// Preserve that cut lands on the same recycled synthetic tail.
if (amplitudeDone_ && amp == 0.0 && !declickActive_ &&
playMode_ == PlayMode::Trigger) {
seedTerminalDeclick();
}
const double gain = amp * velocityGain_;
const double pitchEnvSemis = pitchEnv_.tick();
// 2^(semis/12); when the envelope is off (semis exactly 0) this is 1.0 and skips the
// pow entirely — no per-frame transcendental on the common path.
const double envFactor =
(pitchEnvSemis == 0.0) ? 1.0 : std::pow(2.0, pitchEnvSemis / 12.0);
// Both pitch branches leave the UNENVELOPED post-pitch signal here; the filter acts on
// it and the amp gain is applied afterwards, so the pipeline is pitch -> filter -> amp
// and the amp envelope shapes the filtered result (drive included).
double outL, outRlocal = 0.0;
if (pitchEngine_ == PitchEngine::Preserve && shiftL_.configured()) {
// Feed the shifters the source stream at unity rate (duration held) and transpose
// the output by 2^((note-root + pitchEnvSemis)/12) — pitch envelope adds to the
// shift amount, not the read rate. The feed runs one window ahead of readPos_ (the
// rings were primed with that window at start()), under the same sustain-loop wrap
// rule, reading integer source frames (nothing to interpolate). Past the last real
// frame the shifter's writer is frozen — it recycles the real tail it already holds.
if (loop.active) {
while (feedPos_ >= loop.end) feedPos_ -= loop.length;
}
// feedPos_ runs one window ahead of readPos_; the last real source frame is
// playEnd_-1 for Trigger or frameCount-1 for Gate. Once feedPos_ reaches that bound
// the source is exhausted — feeding the held last sample instead would give the
// splice correlation a DC plateau it can't align on (periodic troughs at the splice
// cadence, growing toward the note end). Freezing the shifter's writer means no
// padding ever enters the ring, so the splice machinery keeps recycling the frozen
// all-real tail — a continuous tone through the voice's own end. The sustain-loop
// path never gets here: the wrap above keeps feedPos_ < loop.end forever.
const std::int64_t feedBound =
(playMode_ == PlayMode::Trigger && playEnd_ > 0 && playEnd_ < frameCount)
? playEnd_ : frameCount;
const bool exhausted = feedPos_ >= feedBound;
if (exhausted) shiftL_.freezeTail(); // idempotent; input ignored while frozen
const bool feedOk = (!exhausted && feedPos_ >= 0 && feedPos_ < frameCount);
// Crossfaded on the way IN to the shifter, not on the way out: loop the source,
// shift the output.
const double feedXw = crossfadeWeight(loop, static_cast<double>(feedPos_));
const AudioSample feedL =
feedOk ? crossfadedSource(pcm, loop, feedPos_, feedXw) : 0.0f;
const double shift = baseRatio_ * envFactor;
shiftL_.setShiftRatio(shift);
const double shiftedL = static_cast<double>(shiftL_.process(feedL));
outL = shiftedL;
if (stereo) {
if (haveR && shiftR_.configured()) {
// Genuine stereo (linked lag): channel 1's shifter FOLLOWS channel 0's
// splice decisions via processLinked — one correlation search, one lag, one
// splice schedule for both channels (standard stereo SOLA). An independent
// per-channel search re-drew an inter-channel offset of up to +/-maxLag at
// every splice: stereo image wander at the splice cadence + mono-sum
// combing. Each shifter is still processed EXACTLY ONCE per output frame
// (never twice — that would advance its heads twice and corrupt the state).
// Gated on haveR so a MONO sample never touches shiftR_ — start() only
// primes it for genuinely stereo samples, and a stale un-primed ring must
// not leak a previous note.
if (exhausted) shiftR_.freezeTail();
const AudioSample feedR =
feedOk ? crossfadedSource(pcmR, loop, feedPos_, feedXw) : 0.0f;
shiftR_.setShiftRatio(shift);
outRlocal =
static_cast<double>(shiftR_.processLinked(feedR, shiftL_.lastSplice()));
} else {
// Mono sample in stereo mode (dual-mono): shiftL_ already produced the
// shifted value from the mono feed; mirror it to R. Do NOT call
// shiftL_.process again this frame.
outRlocal = shiftedL;
}
}
++feedPos_;
// Preserve advances the read head at the SOURCE rate (duration preserved).
ratio_ = 1.0;
} else {
// VARISPEED: pitch and duration coupled. The read rate carries the repitch; the
// pitch envelope multiplies the ratio for the read-rate bias (unchanged idiom when
// the envelope is off -> pitchEnvSemis == 0 -> factor 1.0 -> byte-identical).
//
// Linear interpolation between the two bracketing SOURCE frames at the read head.
// For the loop case, the second point wraps to loopStart so the seam is continuous.
const std::int64_t i0 = static_cast<std::int64_t>(readPos_);
const double frac = readPos_ - static_cast<double>(i0);
std::int64_t i1 = i0 + 1;
if (loop.active && i1 >= loop.end) {
i1 = loop.start; // seamless wrap for the interpolation partner.
}
const bool i0ok = (i0 >= 0 && i0 < frameCount);
const bool i1ok = (i1 >= 0 && i1 < frameCount);
const double srcL = (i0ok ? static_cast<double>(pcm[i0]) : 0.0) +
((i1ok ? static_cast<double>(pcm[i1]) : 0.0) -
(i0ok ? static_cast<double>(pcm[i0]) : 0.0)) * frac;
outL = srcL;
if (stereo) {
const double srcR = (i0ok ? static_cast<double>(pcmR[i0]) : 0.0) +
((i1ok ? static_cast<double>(pcmR[i1]) : 0.0) -
(i0ok ? static_cast<double>(pcmR[i0]) : 0.0)) * frac;
outRlocal = srcR;
}
// Loop crossfade: blend toward the same read head one loop length earlier, which
// is the material the wrap is about to hand over to. Zero outside the fade region
// (and always, with no fade dialled), so the un-crossfaded read stays exactly the
// shape it was.
const double xw = crossfadeWeight(loop, readPos_);
if (xw > 0.0) {
const double tap = readPos_ - static_cast<double>(loop.length);
outL += xw * (lerpSource(pcm, frameCount, tap) - outL);
if (stereo) outRlocal += xw * (lerpSource(pcmR, frameCount, tap) - outRlocal);
}
ratio_ = baseRatio_ * envFactor;
}
// Skipped whole when disengaged (the default), so an un-filtered render stays
// bit-identical to the pre-filter engine.
if (filterOn_) {
if (filterRamping_) tickFilterRamps(); // false at rest: one predicted branch
tickFilterCutoff();
outL = static_cast<double>(filter_.process(0, static_cast<float>(outL)));
// Dual-mono feeds channel 1 the value channel 0 already carried, so mirroring the
// filtered result is exactly what a second identical filter would produce — one
// less kernel pass per frame for the same samples.
if (stereo) {
outRlocal = haveR
? static_cast<double>(filter_.process(1, static_cast<float>(outRlocal)))
: outL;
}
}
outL *= gain;
if (stereo) outRlocal *= gain;
// Takeover declick (bounded-blend revision): on the FIRST frame after a takeover/steal
// restart, seed the blend weight at 1.0 so this frame's output is
// outₙ*(1w) + ref*w = out*(11) + ref*1 = ref (exact boundary identity).
// Each subsequent frame the blend add is `w*(ref outCurrent)` and then w decays by
// kDeclickDecay. The output is therefore bounded by max(|ref|, |outCurrent|) in every
// frame — mid-ramp overshoot from a rising outCurrent is structurally impossible.
// [An earlier revision added the frozen difference (ref x₀) ungated; if outₙ rose
// while the residue was still large the sum could exceed ±1 by up to ~+3.8 dB on an
// extreme retrig.] Inactive (the common case) costs one branch; the blend itself costs
// one extra subtract.
if (declickPending_) seedDeclick();
if (declickActive_) {
const double addL = declickWeight_ * (declickRefL_ - outL);
const double addR = declickWeight_ * (declickRefR_ - (stereo ? outRlocal : outL));
outL += addL;
if (stereo) outRlocal += addR;
declickWeight_ *= kDeclickDecay; // one shared weight — both channels decay together
if (declickWeight_ < kDeclickFloor && declickWeight_ > -kDeclickFloor) {
declickActive_ = false;
}
}
if (stereo) outR = static_cast<AudioSample>(outRlocal);
// Track the value this voice actually contributed THIS frame (post-gain, incl. any
// running declick) — a future takeover restart seeds its declick from exactly this. In
// a mono render the R track mirrors L (dual-mono semantics, matching the stereo mirror
// of a mono sample), so a later stereo takeover still has a sane R seed.
lastOutL_ = outL;
lastOutR_ = stereo ? outRlocal : outL;
readPos_ += ratio_;
// A finished amplitude envelope frees the voice — unless a takeover declick still
// rings: the envelope contributes 0 from here on, so the remaining frames are the bare
// ramp fading out (bounded: the ramp floors within ~4 ms). Baseline unchanged.
if (amplitudeDone_ && !declickActive_) {
active_ = false;
}
return static_cast<AudioSample>(outL);
}
bool active_ = false;
bool releasing_ = false;
int note_ = 0;
double velocityGain_ = 1.0;
double baseRatio_ = 1.0; // 2^((note-root)/12): the un-modulated repitch ratio
double ratio_ = 1.0; // fractional source frames advanced per output frame (this frame)
double readPos_ = 0.0; // fractional frame index into the sample
const SampleData* sample_ = nullptr;
// Gate uses env_ (AHDSR); Trigger uses ampAhd_ — only one active per voice (selected by
// playMode_ at start). playEnd_ is Trigger's source-frame stop (frees when
// readPos_ >= playEnd_).
PlayMode playMode_ = PlayMode::Gate;
AdsrEnvelope env_;
AhdEnvelope ampAhd_;
std::int64_t startFrame_ = 0; // clamped initial read frame; the span-offset origin
std::int64_t playEnd_ = 0; // Trigger: source-frame end; Gate: unused
bool amplitudeDone_ = false; // set when the active amplitude envelope finished
// The sustain loop folded ONCE at note-on: the sample, the play mode and the stored span
// are all fixed for the note's lifetime, so re-deriving validity per frame bought nothing.
// Shared by the output anchor, the Preserve feed, and the start()-time ring prime.
ResolvedLoop loop_;
// The voice's OWN filter and filter envelope — per-voice, never shared, so two notes at
// different envelope phases are filtered independently. filterCutoffNorm_ keeps the
// unmodulated knob position the base is rebuilt from. Q, morph and drive are note-constants
// solved once by start()'s prepare(), which is why every later re-solve is cutoff-only.
// filterRate_ <= 0 makes prepare() bypass rather than invent a rate.
instrument::engine::filter::VoiceFilter filter_;
AdsrEnvelope filterEnv_; // Gate
AhdEnvelope filterAhd_; // Trigger
bool filterOn_ = false;
double filterRate_ = 0.0;
double filterCutoffNorm_ = 1.0;
double filterModAmount_ = 0.0;
double filterVelOffset_ = 0.0; // velAmount * velocityCurve.eval(velocity), fixed per note
double filterKeyTrack_ = 0.0;
instrument::engine::filter::FilterSettings filterSettings_{}; // the note's tone controls
float filterBaseCutoff_ = 1.0f; // cutoff before the envelope, clamped
float filterSolvedCutoff_ = 1.0f; // the position the live coefficients were solved from
bool filterSolved_ = false; // false forces the next frame to solve
// Live-parameter glides (live_params.h). Every one is parked at its target unless a move
// is in flight, so filterRamping_ is false and the per-sample path keeps the pre-live
// engine's exact shape. All five live in the filter's control domains — the envelopes
// need no ramp here, because holding normalized stage position is continuous by
// construction and their two genuine level steps are absorbed inside AdsrEnvelope /
// PitchEnvelope themselves.
bool filterRamping_ = false;
instrument::engine::ValueRamp rBaseCutoff_;
instrument::engine::ValueRamp rModAmount_;
instrument::engine::ValueRamp rResonance_;
instrument::engine::ValueRamp rMorph_;
instrument::engine::ValueRamp rDrive_;
// pitchEngine_ selects Varispeed (ratio bias) vs Preserve (source-rate read + shifter).
// shiftL_/shiftR_ transpose the Preserve output per channel. pitchEnv_ rides either engine.
//
// The shifter rings are primed at start() with the first window of the actual upcoming
// source (silence past the end) — output frame 0 is source frame `start`, no ring-fill
// silence, and splices always land in real history. feedPos_ is the integer source frame
// fed to the shifters next; it runs exactly one window ahead of readPos_ under the same
// sustain-loop wrap rule. Once feedPos_ passes the last real frame (Gate: sample end;
// Trigger: playEnd_), the shifters' writers freeze — no padding enters the rings and the
// splice machinery recycles the frozen real tail through the note end (see advanceFrame).
// primeBuf_ is the presized scratch the prime stream is assembled into.
PitchEngine pitchEngine_ = PitchEngine::Varispeed;
PitchEnvelope pitchEnv_;
PitchShifter shiftL_;
PitchShifter shiftR_;
std::int64_t feedPos_ = 0;
std::vector<AudioSample> primeBuf_;
// lastOut{L,R}_ track the voice's most recent rendered output. A takeover/steal start()
// records them as declickRef{L,R}_ and sets declickPending_; the first frame after the
// restart calls seedDeclick to arm the bounded blend:
// outₙ = outₙ*(1w) + ref*w, w = declickWeight_ (one weight, shared by both channels so
// L/R can never diverge), starting at 1.0 and decaying by kDeclickDecay each frame.
// lastOut is not zeroed by start() — a second same-block takeover (no frame rendered
// between) must record the same pre-cut reference, not a phantom 0. The whole declick
// state is cleared on a fresh (non-takeover) start.
bool declickPending_ = false;
bool declickActive_ = false;
double declickRefL_ = 0.0; // clamped pre-cut reference (bounded blend target)
double declickRefR_ = 0.0;
double declickWeight_ = 0.0; // blend weight w; 1.0 on seed, decays by kDeclickDecay/frame
double lastOutL_ = 0.0;
double lastOutR_ = 0.0;
std::uint64_t startOrder_ = 0;
};
} // namespace reasampler