Commit 057d630 switched this repo from GPL-3.0 to Apache-2.0, which the
build contradicts: src/lib.rs calls nih_export_vst3! unconditionally and
nih_plug is taken with default features, so every build links vst3-sys —
GPLv3-or-later by its own authors' choice. deploy.ps1 then installs that
binary. The declared licence and the artifact's actual terms disagreed,
and the original GPL-3.0 was correct.
Given the intent to distribute non-profit and open-source, GPL costs
nothing here, so this restores it rather than dropping VST3 to keep a
permissive licence.
Also adds the ISC notice owed to Robbert van der Helm: the plugin
scaffolding derives from the NIH-plug cookiecutter template and example
plugins, whose one condition is that the copyright and permission notice
travel with all copies. The README credited the template under Reference
Material, which is courtesy, not compliance. ISC is GPL-compatible, so the
notice is retained in NOTICE rather than superseded.
Corrects the README's claim that Steinberg's SDK licence must be accepted
before distributing. No SDK source is vendored or redistributed, so there
is nothing to accept at build time. The binding constraint is vst3-sys's
GPLv3; Steinberg's trademark in "VST" is a separate regime that no source
licence resolves.
Copyright in the Irene <irene@novoyuuparosk.org> commits is held by the
repository owner (an agent identity), so no third-party agreement was
needed for this relicense.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01WHt4DykxXaQnyaJfyS8KCn
5 ms was too short to be an RMS average at all. Squaring the detector signal
puts a component at twice the fundamental, and a one-pole that does not span a
cycle passes it straight through, so the detector ripples and the gain
intermodulates with the bass. Measured peak-to-peak detector ripple at 5 ms:
30 Hz 50 Hz 80 Hz 120 Hz 200 Hz 500 Hz 1 kHz
4.42 2.72 1.72 1.15 0.69 0.28 0.14 dB
That lands squarely on the low band, whose crossover reaches down to 30 Hz,
and attack is allowed to go to 0 ms so nothing downstream is guaranteed to
mask it. Above ~500 Hz the same window was so short that RMS mode barely
differed from peak, making the mode switch do less than it appeared to.
The fix is per-band rather than one longer constant, because the required
window is set by how low the band reaches and a multiband already knows that.
Cost is response time (95% of a step: ~12 ms at 5 ms, ~117 ms at 50 ms), so
each band gets only what it needs:
LOW 50 ms 30 Hz ripple 4.42 -> 0.46 dB; slow is wanted down here anyway
MID 20 ms body without dulling the band carrying most transients
HIGH 10 ms nothing up there ripples, so stay responsive
ALL 35 ms full-band, so it has to cover the low content too
RMS_WINDOW_MS moves from a compressor constant to a table in lib.rs indexed by
the band consts, and Compressor::prepare takes the window - the compressor has
no business knowing which band it is.
rms_window_suppresses_low_frequency_ripple pins the property: the LOW window
holds 30 Hz under 1 dB, a 5 ms window is far worse on the same signal (so the
bound cannot pass for the wrong reason), ripple falls monotonically with window
length, and the HIGH band's short window stays clean at 5 kHz.
This changes how the plugin sounds in RMS mode - needs an ear, not just tests.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01LU1NRL7fAaZTtFMTTPn2b1
Found by reading the audio-thread code, not by profiling - no profiler is
set up in this project yet. All four are algorithmic or arithmetic wins,
none change the DSP result; the existing tests still pass.
- oversampler: max_true_peak recomputed its circular-history tap index
(including a % TAPS_PER_PHASE) inside the phase x channel loop, though
the index depends only on pos/k - 8x redundant. Hoisted to a per-sample
precomputed index array. This runs unconditionally every sample (it
feeds the limiter's true-peak detector, not gated by metering), so it
was the highest-value fix.
- limiter: the sliding-window max rescanned the entire look-ahead ring
(~73 samples at 48kHz/1.5ms) every sample - O(window) per sample.
Replaced with an O(1)-amortized monotonic-deque sliding max, backed by
preallocated ring arrays and RT-safe. Covered by a new test against a
naive brute-force reference. Also precomputed the constant true-peak
margin gain in prepare() instead of a powf every sample.
- compressor: replaced per-sample log10/powf(10, x/20) with the
equivalent ln/exp forms, mirroring nih-plug's own gain_to_db_fast /
db_to_gain_fast. Called 4x/sample (3 bands + 'All').
- both: replaced % capacity ring-index math (a real integer division,
since capacity is not a power of two) with a conditional-subtract
wrap_add helper. Left the oversampler's % TAPS_PER_PHASE alone - that
divisor is a compile-time constant LLVM already turns into a multiply.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01LU1NRL7fAaZTtFMTTPn2b1
Rework Low Curve from a one-sided saturating bend into a bipolar mid-bulge: the
low region is anchored at BOTH the silence floor and the knee, Low Slope tilts
the straight line between them, and Low Curve (-1..1) bows that line in the
middle (4*t*(1-t), peak +/-12 dB) without moving either endpoint. Positive
bulges up (boost the quiet middle), negative down (suppress). Still serial.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Add a per-channel below-threshold shaper composed in series ahead of the comp:
gain = low_shape(level) + comp(level + low_shape(level)). The compressor's
threshold now sees the shaped level, so a Low Slope boost lifts quiet material
up into compression (and a cut pulls it out). Anchored at the -60 dB silence
floor. Low Curve bends the shaper toward a bounded saturation so the serial
composition doesn't blow up (0 = straight line).
gain_computer split into comp_gain_db + low_gain_db and composed; shared with
the editor gain-curve display. Slider order rearranged to read in signal order
(pre-gain -> low shaper -> compressor -> output). Defaults (slope 1, curve 0)
reproduce the plain compressor; 17 tests pass.
Known: the bipolar behaviour isn't final yet (milestone commit).
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Add editor/gain_curve.rs: a square panel beside the scrolling plot showing the
channel transfer (out vs in, -60..0 dB) for whichever channel the plot tab
selects. Plots the full wet path — out = (in + pre_gain) + gain_reduction + makeup
— using the shared Compressor::gain_computer (now pub) so it matches the DSP and
the GR meter. Unity-reference diagonal + threshold marker; mix not folded in.
Update README structure/status to reflect the editor/ widget module and the
completed Stage 6 visualisers.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Replace the per-channel bypass toggle with a smoothed dry/wet `mix` (0..100%,
default 100%). The blend is applied at the compressor output:
out = delayed_input * ((1 - mix) + mix * wet_gain)
Dry and wet share the same delayed input, so it's phase-aligned (parallel
compression, no comb filtering). mix=0 is bit-identical to the old bypass.
The detector now runs even at mix 0, so the GR meter shows the wet gain
reduction regardless of mix, while the level/plot out trace reads the mixed
output. "Bands at 0% = simple full-band comp via All" still holds.
Update README + parameter docs (bypass -> mix).
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Stage 6 begins. Add lock-free Meters (atomics shared audio->GUI) with a
peak-with-decay ballistic, published once per block and gated on the editor
being open. Editor draws a per-channel level + gain-reduction meter panel and
a ceiling lamp fed by the output limiter. Compressor/Limiter expose
gain_reduction_db() for this.
Also add a smoothed per-channel pre-gain applied before each compressor (and
before the All compressor), driving the signal into compression and on into
the limiter for a compressed semi-distortion. Pairs with makeup for full
per-channel input/output gain-staging. Compressor DSP untouched; 16 tests pass.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Investigated the planned global FTZ for the IIR filters and found NIH-plug already
handles it: process_wrapper wraps process()/reset() in a ScopedFtz guard that
enables CPU Flush-To-Zero (x86 MXCSR bit 15 / AArch64 FPCR bit 24, via inline asm,
restored on drop) on the vst3, clap, and standalone paths. SSE is baseline on
x86_64 so FTZ is always active for our build.
So adding our own guard would just duplicate the framework. Instead, removed the
now-redundant flush_denormal() from compressor.rs (the biquads and envelope/RMS
tails already relied on this FTZ) for a single consistent story, and rewrote the
README 'Denormal flushing' note to document that the framework handles it (FTZ,
not DAZ — sufficient for our feed-forward IIR).
No functional change; 10 unit tests still pass.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
- Compressor: switchable peak / RMS detection (EnumParam<DetectionMode> in lib.rs
-> use_rms bool in CompressorSettings; DSP stays framework-agnostic). RMS is a
one-pole running mean of the linked squared level with a hardcoded 5 ms window,
updated whenever active so peak<->RMS switching is seamless. New unit test
(RMS compresses a sine less than peak); 6 tests total.
- README: reconciled Implementation Order checklists with actual progress
(Stages 1-2 done; look-ahead/latency + basic UI pulled forward), annotated the
project structure (implemented vs planned), and corrected the Latency and
Denormal-flushing notes to match the code (set_latency_samples once / constant
latency; in-code denormal flush). Overview and goals left unchanged.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Implements the single full-band feed-forward compressor (the engine that will be
reused per band + for the 'All' channel). Design follows Giannoulis et al. 2012:
log-domain gain computer with a quadratic soft knee feeding a smooth decoupled
peak detector for attack/release ballistics. Stereo-linked peak detection.
Look-ahead uses a fixed audio delay with a constant reported latency (set once in
initialize); the knob only moves the detector tap within that delay. This avoids
renegotiating latency from process(), which crashed FL Studio when the look-ahead
was adjusted during playback.
- src/dsp/compressor.rs: Compressor + CompressorSettings, RT-safe (no alloc in
process; buffers sized in prepare; envelope denormals flushed in-code)
- src/lib.rs: nested CompressorParams (threshold/ratio/knee/attack/release/makeup/
bypass) + global look-ahead; egui ParamSlider grid; latency reported once
- 5 unit tests (static curve, knee continuity, steady-state convergence, constant
latency); Cargo.toml lib crate-type added so tests link
- README: 'All' channel architecture already documented; look-ahead spec updated
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
mikkeli
Mikkeli Matlock
irene