daniel/reasampler
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Fix filter DSP click-train flush and finish namespace nesting + test coverage

Browse Source
This commit is contained in:
Daniel Harvey
2026-07-30 07:00:41 -04:00
parent 78e112d1f9
commit 7af3c0c630
9 changed files with 97 additions and 35 deletions
Download Patch File Download Diff File Expand all files Collapse all files
src/core/instrument/engine/filter/CLAUDE.md
+9 -4
View File
@@ -3,7 +3,11 @@
## Scope
The pure 2-pole resonant low/high-pass a sounding voice runs. No REAPER, no VST3, no
allocation, no I/O. Four files, one responsibility each:
allocation, no I/O. Everything here lives in `reasampler::instrument::engine::filter`,
nested per the directory-mirrors-namespace convention — this keeps `FilterMode` and
friends out of `reasampler::instrument::engine` proper, where `zone_params.h` lives, since
this module has no call site yet to force the collision into the open at compile time.
Four files, one responsibility each:
- `filter_params` — the control domain: `FilterMode`, normalized [0,1] knob position →
cutoff Hz and Q, and the exact inverses.
@@ -43,9 +47,10 @@ is 200 Hz and 2/3 is 2 kHz) and takes no sample rate. The persisted value is the
normalized knob position, so a rate-derived endpoint would make one preset sound different
at 44.1k and 96k. The Nyquist clamp (`kFilterNyquistFraction`, 0.48) is a property of the
bilinear transform — `tan(pi*fc/sr)` diverges at Nyquist — so it lives in `biquadCoeffs`
where the rate is already a parameter. 20 kHz is under 0.48·sr at every supported rate, so
the clamp never eats live knob travel; the source's hardcoded 23 kHz endpoint did exactly
that at 44.1k.
where the rate is already a parameter. 20 kHz is under 0.48·sr at 44.1k and above, so the
clamp never eats live knob travel there; the source's hardcoded 23 kHz endpoint did exactly
that at 44.1k. Below 44.1k (e.g. 32k, 22.05k) the clamp still handles the math correctly —
it just legitimately eats the top of the knob travel at those rates.
`biquadCoeffs` with a non-positive sample rate returns pass-through coefficients. It does
**not** fall back to 44100 — that would breach the standing no-hardcoded-sample-rates
src/core/instrument/engine/filter/filter_coeffs.cpp
+2 -2
View File
@@ -2,7 +2,7 @@
#include <cmath>
namespace reasampler::instrument::engine {
namespace reasampler::instrument::engine::filter {
namespace {
// M_PI is not standard C++ and is absent on MSVC without _USE_MATH_DEFINES.
@@ -40,4 +40,4 @@ BiquadCoeffs biquadCoeffs(FilterMode mode, float cutoffHz, float q, double sampl
return c;
}
} // namespace reasampler::instrument::engine
} // namespace reasampler::instrument::engine::filter
src/core/instrument/engine/filter/filter_coeffs.h
+2 -2
View File
@@ -7,7 +7,7 @@
#include "core/instrument/engine/filter/filter_params.h"
namespace reasampler::instrument::engine {
namespace reasampler::instrument::engine::filter {
// Already normalized by a0. The denominator is 1 + a1*z^-1 + a2*z^-2, so the difference
// equation SUBTRACTS the a terms: y = b0*x + b1*x1 + b2*x2 - a1*y1 - a2*y2.
@@ -28,4 +28,4 @@ inline constexpr double kFilterNyquistFraction = 0.48;
// no-hardcoded-sample-rates ruling means we refuse to invent a rate rather than assume 44.1k.
BiquadCoeffs biquadCoeffs(FilterMode mode, float cutoffHz, float q, double sampleRate);
} // namespace reasampler::instrument::engine
} // namespace reasampler::instrument::engine::filter
src/core/instrument/engine/filter/filter_params.cpp
+2 -2
View File
@@ -2,7 +2,7 @@
#include <cmath>
namespace reasampler::instrument::engine {
namespace reasampler::instrument::engine::filter {
namespace {
double clamp01(double v) { return v < 0.0 ? 0.0 : (v > 1.0 ? 1.0 : v); }
@@ -56,4 +56,4 @@ float filterNormFromQ(float q) {
return static_cast<float>(clamp01((-k.b + std::sqrt(d)) / (2.0 * k.c)));
}
} // namespace reasampler::instrument::engine
} // namespace reasampler::instrument::engine::filter
src/core/instrument/engine/filter/filter_params.h
+4 -3
View File
@@ -5,7 +5,7 @@
#pragma once
namespace reasampler::instrument::engine {
namespace reasampler::instrument::engine::filter {
enum class FilterMode { LowPass, HighPass };
@@ -13,7 +13,8 @@ enum class FilterMode { LowPass, HighPass };
// norm 2/3 is 2 kHz. NOT derived from the sample rate — a rate-dependent endpoint would make
// one saved preset sound different at 44.1k and 96k, and at 44.1k the top of the travel would
// be dead against the Nyquist clamp (the ported firmware's 23 kHz endpoint had exactly that
// defect). 20 kHz sits under 0.48*sr at every rate we support, so the whole knob stays live.
// defect). 20 kHz sits under 0.48*sr at 44.1 kHz and above; below that (e.g. 32 kHz, 22.05 kHz)
// the clamp still handles it correctly, it just eats the top of the knob travel at those rates.
inline constexpr float kFilterCutoffMinHz = 20.0f;
inline constexpr float kFilterCutoffMaxHz = 20000.0f;
@@ -36,4 +37,4 @@ float filterQFromNorm(float norm);
// Exact inverse of filterQFromNorm; out-of-range Q clamps to 0 or 1.
float filterNormFromQ(float q);
} // namespace reasampler::instrument::engine
} // namespace reasampler::instrument::engine::filter
src/core/instrument/engine/filter/filter_saturate.h
+2 -2
View File
@@ -4,7 +4,7 @@
#pragma once
namespace reasampler::instrument::engine {
namespace reasampler::instrument::engine::filter {
// Rational tanh approximation inside +/-threshold, continued past it with a gentle 0.1 slope
// anchored at the threshold value so the curve stays continuous rather than hard-clipping.
@@ -24,4 +24,4 @@ inline float tanhSaturate(float x, float threshold, float a, float b) {
// path produces, not for audio-level signals — do not reuse it as a general waveshaper.
inline float feedbackSaturate(float x) { return tanhSaturate(x, 2.0f, 27.0f, 9.0f); }
} // namespace reasampler::instrument::engine
} // namespace reasampler::instrument::engine::filter
src/core/instrument/engine/filter/voice_filter.cpp
+2 -2
View File
@@ -1,6 +1,6 @@
#include "core/instrument/engine/filter/voice_filter.h"
namespace reasampler::instrument::engine {
namespace reasampler::instrument::engine::filter {
void VoiceFilter::prepare(const FilterSettings& settings, double sampleRate) {
mode_ = settings.mode;
@@ -26,4 +26,4 @@ bool VoiceFilter::isSilent() const {
return true;
}
} // namespace reasampler::instrument::engine
} // namespace reasampler::instrument::engine::filter
src/core/instrument/engine/filter/voice_filter.h
+27 -9
View File
@@ -11,7 +11,7 @@
#include "core/instrument/engine/filter/filter_params.h"
#include "core/instrument/engine/filter/filter_saturate.h"
namespace reasampler::instrument::engine {
namespace reasampler::instrument::engine::filter {
// Normalized control positions, as the editor moves them and the persisted state carries them.
struct FilterSettings {
@@ -23,6 +23,13 @@ struct FilterSettings {
// Share of the last output fed back into the high-pass input at full resonance. Driven by the
// raw control position rather than by Q: Q reaches 10, and scaling the feedback by it would
// push the loop gain past unity at the top of the range.
//
// The HP/LP resonance asymmetry this produces is a known ear call reserved for Daniel, not a
// bug: measured peak/passband at res=1.0, fc=1kHz/sr=48k is LP 9.98 (flat at every input level)
// vs HP 7.44 (input 0.001-0.1), 7.59 (0.3), 8.52 (1.0) — HP resonance is level-dependent because
// feedbackSaturate's threshold (+/-2.0) is an absolute level, not a fraction of the signal.
// Retuning this constant alone cannot make the two modes track, since it does not touch that
// level-dependence.
inline constexpr float kHighPassFeedbackShare = 0.24f;
// Below this the recursion has decayed past -600 dB. Flushing keeps the history out of the
@@ -56,7 +63,8 @@ public:
// The high-pass numerator collapses toward zero as cutoff falls, taking the resonance
// with it; feeding a saturated share of the last output back into the input restores
// the character the coefficients alone stop producing down there.
// the character the coefficients alone stop producing down there. The 0.9f pre-scale is
// carried from the source firmware, uncalibrated here — no derivation is known for it.
const float in = (mode_ == FilterMode::HighPass)
? x - fbAmount_ * feedbackSaturate(s.fb * 0.9f)
: x;
@@ -69,15 +77,25 @@ public:
s.y2 = s.y1;
s.y1 = y;
// Snap the WHOLE state once the recursion as a whole has decayed past -600 dB.
// Zeroing individual samples instead does not work: a resonator swings through zero
// twice a cycle, so a per-sample flush injects a step in phase with the resonance,
// which the resonance then amplifies — the filter limit-cycles at the floor forever
// rather than going quiet. Testing y1 AND y2 tests the envelope, not one sample.
// Snap the RECURSIVE half of the state once it has decayed past -600 dB. Only y1/y2
// are flushed (and only they are tested) — x1/x2 is an FIR tail that shifts out within
// two samples on its own, and a high-pass has an exact DC null (b1 == -2*b0 bit-exactly),
// so under a constant/DC-biased input y decays to zero while x1/x2 sit at the input
// level; clearing x1/x2 too would discard that history and the next sample would
// recompute a full-amplitude step from b0*in alone, re-ringing forever (a click train).
// Zeroing individual samples instead of the pair does not work either: a resonator
// swings through zero twice a cycle, so a per-sample flush injects a step in phase with
// the resonance, which the resonance then amplifies — the filter limit-cycles at the
// floor forever rather than going quiet. Testing y1 AND y2 tests the envelope, not one
// sample.
if (s.y1 > -kFilterDenormalFloor && s.y1 < kFilterDenormalFloor &&
s.y2 > -kFilterDenormalFloor && s.y2 < kFilterDenormalFloor) {
s = State{};
s.y1 = 0.0f;
s.y2 = 0.0f;
}
// Stored unconditionally even in LP mode, where nothing reads it: the mode branch above
// already exists, but gating this one store on it buys nothing a dead-store-eliminating
// compiler doesn't already do for free, at the cost of a second branch on the mode.
s.fb = s.y1;
return y;
}
@@ -110,4 +128,4 @@ private:
static_assert(!std::is_polymorphic_v<VoiceFilter>, "no vtable on the per-sample path");
static_assert(std::is_trivially_copyable_v<VoiceFilter>, "state is plain values, never owned");
} // namespace reasampler::instrument::engine
} // namespace reasampler::instrument::engine::filter