@audio/dynamics

Try it in the browser: Loudness meter and normalizer. Runs on this package, nothing is uploaded.

Dynamics processing — compressor, limiter, gate, expander, de-limiter, de-esser, ducker, softclip, compand, multiband. The family includes envelope-driven gain control, lookahead limiting, waveshaping and whole-buffer level correction. Part of audiojs.

	Kind	Gain function	Typical use
compressor	envelope	soft-knee above threshold	leveling vocals, mix glue
limiter	lookahead	brickwall at ceiling	master bus, peak control
gate	envelope	hard cut below threshold	silence between phrases
expander	envelope	gentle below-threshold reduction	soft gating, noise bed shaping
unlimit	dual envelope	transient-gated upward expansion	de-limiting, crest restoration
deesser	sidechain	sibilance band over voice body, cut within a range	harsh 's' in voice
ducker	ext. sidechain	compressor keyed by side signal	music-under-voice, podcast
softclip	waveshaper	static transfer curve	gentle peak limiting + coloration
compand	envelope	piecewise-linear transfer	SoX-style multi-segment
multiband	envelope × N bands	LR split + per-band up/down compression	mastering glue, OTT-style upward+downward

Usage

npm install @audio/dynamics

import { compressor, limiter } from '@audio/dynamics'

const sampleRate = 48000
const samples = Float32Array.from({ length: 4800 }, (_, i) =>
  0.8 * Math.sin(2 * Math.PI * 220 * i / sampleRate))
const opts = { sampleRate, threshold: -18, ratio: 4, attack: 5, release: 100 }
const compressed = compressor(samples, opts)
const limited = limiter(compressed, { sampleRate, ceiling: -1, lookahead: 5 })
console.log(limited.length) // 4800; samples and compressed are preserved

// Each writer owns its history. Keep it for successive chunks, then flush once.
const write = compressor(opts)
const blocks = [write(samples.subarray(0, 256)), write(samples.subarray(256)), write()]
console.log(blocks.reduce((n, block) => n + block.length, 0)) // 4800

For one processor: npm install @audio/dynamics-compressor, then import compressor from '@audio/dynamics-compressor'. The umbrella re-exports the same functions and types. Existing CompressorOpts, GateOpts and other legacy option names remain available alongside the leaf names (CompressorOptions, GateOptions, etc.).

Processing expectations

Functions	Buffer ownership / streaming	Sample-rate option
compressor, limiter, gate, expander, unlimit, deesser, ducker, compand, opto, fet, vca, varimu	Batch calls return a new buffer. Calling with options returns a writer; keep it across chunks and call with no arguments to flush.	sampleRate
softclip	Returns a new buffer. Its writer buffers until flush when oversample > 1.	fs
transientShaper	Mutates and returns the input. Reuse the same options object across chunks; a fresh object resets the envelopes.	fs
multiband, leveler	Mutate and return the input. Whole-buffer processing; no writer form.	fs
envelope	Returns a stateful sample → level function. Create another follower to reset.	sampleRate

PCM is mono Float32Array; process channels with separate state/writers. Sample rate defaults to 44100 Hz. Gains/thresholds are dB and attack/release/lookahead are milliseconds unless stated otherwise: softclip.drive and transient-shaper gains are linear, while leveler.frame is seconds. Do not substitute fs for sampleRate indiscriminately; only some processors accept that alias.

Batch compression and a fresh writer fed the same signal produce the same samples. Lookahead writers can return fewer samples until flush; concatenate every returned block, including the final flush. An empty chunk is a zero-length write, not a flush. Treat flush as the end of a signal and create a new writer to reset. Options are construction settings, not a live automation interface.

The leaf /audio exports are host processor factories with their own parameter metadata. Some controls restart their processor, and some factories require a whole render (streaming: false); check those limits before using an AudioWorklet adapter.

envelope

The common detector for the envelope-driven processors is a branching one-pole follower with separate attack/release time constants, peak or RMS detection.

import { envelope } from '@audio/dynamics'

let follow = envelope({ attack: 5, release: 100, detector: 'peak' })
let level = []
for (let x of samples) level.push(follow(x))

Param	Default
sampleRate	44100	—
attack	5	ms
release	50	ms
truePeak	false	hold the reconstructed waveform under the ceiling, not only its samples (ITU-R BS.1770 Annex 2): each interval read at 8 points through a 96-tap Kaiser sinc; 96 samples more delay
detector	'peak'	'peak' or 'rms'
rmsWindow	256	samples, for RMS detector

compressor

Feed-forward soft-knee downward compressor, as Giannoulis, Massberg & Reiss recommend: level in dB → quadratic soft-knee gain curve → gain reduction smoothed in the log domain by the smooth decoupled peak detector → linear gain applied to input. attack and release are the time constants of the gain reduction itself (1 − 1/e), at any depth of compression; the first sample over the threshold is already reduced.

Downward compression (above threshold, reduces gain) is one half of the canonical four-quadrant dynamics taxonomy — downward/upward compression, downward/upward expansion (Giannoulis, Massberg & Reiss 2012; Izhaki, Mixing Audio). Setting upThreshold engages the other compression half: upward compression lifts quiet passages toward the threshold instead of squashing loud ones — the "OTT up" half popularized by Xfer OTT. Both curves read the same envelope and sum in the dB domain, so a single compressor call can glue loud material down and lift quiet material up at once.

import { compressor } from '@audio/dynamics'

compressor(data, { threshold: -18, ratio: 4 })
compressor(data, { threshold: -24, ratio: 2, knee: 12, attack: 10, release: 200, makeup: 6 })

// upward + downward together (OTT-style): lift quiet passages, squash loud ones
compressor(data, { threshold: -18, ratio: 4, upThreshold: -40, upRatio: 2, upRange: 12 })

Param	Default
threshold	-20	dB
ratio	4	—
knee	6	dB (soft-knee width)
attack	5	ms
release	100	ms
makeup	0	dB
depth	1	scales the summed up+down gain before makeup (OTT "Depth" macro; 0 = identity)
upThreshold	null	dB; null disables upward compression
upRatio	2	— (1 is a mathematical no-op)
upKnee	6	dB
upRange	12	dB, max upward lift — without a ceiling, silence would take unbounded gain

Use when: vocals, bass, drum bus, mix glue; add upThreshold for OTT-style up+down "aggressive" glue.
Not for: peak control at the master bus — use limiter. Transparent loudness shaping — use compand with gentle slope.

limiter

Lookahead brickwall limiter. The gain each sample needs, its sliding minimum over the lookahead span and that minimum's moving average: the gain ramps down across lookahead ms into a peak, never stepping, and still covers every sample in transit; exponential release after it passes.

import { limiter } from '@audio/dynamics'

limiter(data, { ceiling: -0.3 })
limiter(data, { ceiling: -1, lookahead: 10, release: 100 })
limiter(data, { ceiling: -1, truePeak: true })   // dBTP: the waveform between samples too

Param	Default
ceiling	-0.3	dB (brickwall)
lookahead	5	ms (introduces delay)
release	50	ms
truePeak	false	hold the reconstructed waveform under the ceiling, not only its samples (ITU-R BS.1770 Annex 2): each interval read at 8 points through a 96-tap Kaiser sinc; 96 samples more delay

Use when: peak control at the master bus; truePeak for a delivery spec in dBTP (streaming's -1 dBTP, EBU R128's -1), where a DAC or a lossy decoder rebuilds peaks up to 3 dB over the samples. Full-band noise limited 20 dB stays within 0.005 dB of the ceiling, band-limited, with 16 samples of lookahead or more.
Not for: musical dynamics shaping — use compressor. Low-latency paths — use softclip.

gate

Noise gate with hysteresis, hold-then-close logic and look-ahead. Opens above threshold, closes only below closeThreshold (hysteresis prevents chatter around a single threshold); below it, signal is attenuated by range dB. A hold timer keeps the gate open after a drop-out; attack/release smooth the gain transitions. lookahead runs detection ahead of emission so the gate is already opening when a transient reaches the output — batch calls stay sample-aligned (no silence prefix, no dropped tail); block hosts get the delay declared as atom latency.

import { gate } from '@audio/dynamics'

gate(data, { threshold: -40 })
gate(data, { threshold: -35, range: -80, hold: 20, attack: 1, release: 150, lookahead: 5 })

Param	Default
threshold	-40	dB, open above
closeThreshold	threshold − 6	dB, close below (hysteresis)
range	-60	dB attenuation when closed
hold	10	ms
attack	0.1	ms (opening)
release	100	ms (closing)
lookahead	0	ms, detection leads emission

Use when: drum mics, voice dialogue with ambient noise, removing hiss between phrases.
Not for: subtle low-level reduction — use expander.

expander

Downward expander (mode: 'downward', default) — a softer gate. Below threshold, gain is reduced by (threshold − level) × (ratio − 1) dB, clamped at range.

mode: 'upward' switches to upward expansion — the de-compression complement, raising gain above threshold instead of cutting it below. Classical substrate for de-limiting: transient-aware upward expansion restores crest factor a brickwall limiter (or an over-eager mix bus compressor) flattened. Same four-quadrant taxonomy as compressor's upward mode (Giannoulis/Reiss; Izhaki, Mixing Audio).

import { expander } from '@audio/dynamics'

expander(data, { threshold: -30, ratio: 2 })

// de-limiting: expand transients back out above threshold
expander(data, { mode: 'upward', threshold: -20, ratio: 1.5, range: 20 })

Param	Default
mode	'downward'	'downward' | 'upward'
threshold	-30	dB
ratio	2	—
knee	6	dB
range	-40 (downward) / 20 (upward)	dB, max reduction (downward, negative) or max lift (upward, positive)
attack	5	ms
release	50	ms
truePeak	false	hold the reconstructed waveform under the ceiling, not only its samples (ITU-R BS.1770 Annex 2): each interval read at 8 points through a 96-tap Kaiser sinc; 96 samples more delay

Use when: gentle noise-floor suppression without the abruptness of a gate (downward); restoring dynamics to over-compressed or over-limited material (upward).
Not for: hard removal of sound between phrases — use gate.

unlimit

De-limiter. iZotope Ozone 12's "Unlimiter" (Sept 2025) created the de-limiting category with a trained ML model; this atom is the classical counterpart — transient-synchronous upward expansion, restoring the crest factor a brickwall limiter (or an over-eager bus compressor) flattened. Program-adaptive upward expansion gated to transients, not level — one more cell in the four-quadrant dynamics taxonomy (Giannoulis, Massberg & Reiss 2012, JAES 60(6); Izhaki, Mixing Audio), built on expander's upwardExpanderGain curve.

A fast envelope (fastAttack/fastRelease, near-instant) and a slow envelope (slowAttack/slowRelease, sluggish) both track the input; their gap in dB — transientness — rises sharply on attacks and sits near zero on sustained material. Gain lift follows transientness, not absolute level: an absolute-level upward expander would pump sustains; gating on the fast/slow gap instead is what makes this a de-limiter rather than a leveler.

import { unlimit } from '@audio/dynamics'

unlimit(data, { amount: 9, drive: 2 })                  // deliberate restoration
unlimit(data, { amount: 9, drive: 2, ceiling: -1 })      // guard restored peaks at -1 dBFS

Param	Default
amount	6	dB, max crest restoration (range 0–18)
drive	1	scales the deficit-driven restoration (1 = restore attacks to crestTarget); in adaptive: false mode, dB of lift per dB of transientness
adaptive	true	deficit mode (see below); false = raw proportional transient-following (a transient exaggerator, for manual sound design)
crestTarget	10	dB of transientness a healthy attack is expected to show; flattened attacks get lifted by what they're missing
ceiling	null	dBFS; post guard so restored peaks don't exceed it. null (default): peaks may exceed 0 dBFS by design (float domain)
fastAttack	0.5	ms
fastRelease	20	ms
slowAttack	20	ms
slowRelease	200	ms

The default mode lifts by transient deficit, not transient presence — the inverse-limiter insight: a brickwall limiter's fingerprint is attacks that are too small (3–8 dB of fast-over-slow transientness where healthy program shows 12–25 dB), so each onset gets back crestTarget − measured dB, and a naturally healthy attack gets structurally zero lift. Safety on dynamic material is a property of the curve, not a timid default: measured on the test fixture, defaults change never-limited program by ≤ 0.2 dB RMS while amount: 9 recovers ~4.5 dB of crest from a 9 dB-crushed brickwall (see tests). Three gates make that separation robust — peak-hold (deficit is judged against an attack's peak transientness, not its rise samples), a 10 ms attack window (a decaying tail keeps the fast envelope above the slow one for its whole length; a decay is not an onset), and a ~3 ms confirmation ramp (a healthy attack outruns crestTarget in ~1.5 ms, collapsing its own deficit before lift confirms; a limiter-flattened plateau is still standing).

Honest scope: this restores dynamics/crest — it cannot recover information a clipper already destroyed (pair with @audio/denoise-declip for that), and it does not un-mix limiter pumping artifacts baked into the waveform's history. Over-driving amount/drive invents transients that were never there. v1 is zero-latency with no lookahead — it reacts to a transient already underway, it cannot anticipate one; lookahead attack-anticipation is a future option.

Use when: restoring life to over-limited masters, streaming-loudness-flattened stems, squashed dialogue or game audio.
Not for: recovering clipped/distorted peaks — use a declipper. Undoing audible limiter pumping — remix from an earlier, unlimited stage if one exists.

deesser

Sibilance reduction. An 's' is told by its shape, not its level: the band where sibilance lies (over split) is measured against the voice body (its formants, under 3.5 kHz), in dB, as the dbx 902 compares its high band with the full band. The same 's' is caught on a quiet recording and a loud one, and a cymbal in a mix, which sits over the mix's body, is left alone. How far the band rises over threshold sets the cut, held within range, the Range of the dbx 902, FabFilter Pro-DS and Waves Sibilance. A pause is measured against the voice's running level, so hiss between words is not taken for an 's'. The sound is read lookahead ms ahead of what is cut (a declared latency, so a host aligns it): an 's' sets its level within a few ms, and a cut that waits for it leaves its onset whole. mode applies the cut: split (default) to the band over split alone, x − (1 − g)·HP(x) with HP linear-phase (±1.5 ms), so x − HP is exactly the rest: no band is lifted at the split, and with no 's' the input passes sample for sample; band to a linear-phase band at fc, Q wide; broadband to the whole sound.

import { deesser } from '@audio/dynamics'

deesser(data)                                    // at any recording level; aligned (a stream is `lookahead` late)
deesser(data, { mode: 'band', fc: 7500, Q: 1 })  // only that band moves
deesser(data, { threshold: 3, range: -6 })       // fewer esses, shallower

Param	Default
mode	'split'	'split' | 'band' | 'broadband'
split	3500	Hz, where the cut band starts in split and broadband modes, and the band watched
fc	6500	Hz, band mode: the band's centre (freq still accepted)
Q	1.4	band mode: its width, as a peaking EQ's (q still accepted)
threshold	0	dB of the sibilance band over the voice body where the cut starts; not a level
ratio	4
range	-8	dB, the deepest cut
knee	6	dB
attack	1	ms
release	15	ms
lookahead	5	ms, the output's delay in a stream (never under 1.5)

Against iZotope RX 12 De-ess, node bench/rx/deess.mjs in audio (2026-10): VoiceBank test speech (2 speakers) and ten Spoken Wikipedia narrations, 21.4 min; its sibilants (active frames with half their energy over 3.5 kHz, found on the clean speech), 1702 of them, made harsh by their 4–10 kHz band alone, 4–12 dB up; the clean speech the target. Every setting chosen on 28 other speakers and ten other narrations, by SNR to the clean speech; RX tuned: Classic, threshold −9, Fast, cutoff 4 kHz (defaults: −12, 2.5 kHz). Over the sibilants, the error to the clean speech taken away and their 4–10 kHz band left over the clean's (median); over the reels, SNR to the clean speech; and the clean speech through it: SNR to itself, the share of its other active frames (vowels, voiced consonants) moved by over 1 dB, its own sibilants' 4–10 kHz cut (median):

	harsh: error taken away	harsh: 4–10 kHz left	harsh: SNR to clean	clean: SNR to input	clean: others moved	clean: own sibilants cut
RX 12 defaults	9.5 dB	−2.0 dB	22.5 dB	18.7 dB	8.1 %	8.2 dB
RX 12 tuned	12.1 dB	0.4 dB	25.4 dB	20.4 dB	4.2 %	5.6 dB
0.3.0 (broadband)	8.9 dB	3.4 dB	21.9 dB	22.7 dB	1.1 %	1.3 dB
0.3.0, band	5.0 dB	4.1 dB	18.7 dB	24.4 dB	0.8 %	1.2 dB
0.4.0 (split)	12.5 dB	1.4 dB	26.2 dB	21.0 dB	1.9 %	3.9 dB
0.4.0, band	7.5 dB	3.1 dB	21.2 dB	23.2 dB	0.9 %	2.0 dB
0.4.0, broadband	11.5 dB	1.3 dB	23.9 dB	20.6 dB	2.3 %	4.0 dB

0.3.0 cut late (with no look-ahead an 's' sets the level its own cut answers, so its first ms pass), on the whole sound or on a bell at fc, and no deeper than 6 dB, under the 4–12 dB a harsh 's' carries. node scripts/deesser.js runs it over speech, singing and music and reads, per 10 ms frame, how far the 4–10 kHz band and the 0.3–4 kHz voice body moved: on 's' frames (active, half their energy above 3.5 kHz) the cut, elsewhere the share of frames moved by more than 1 dB. VoiceBank test set (824 utterances), 10 narrations (60 s each), VocalSet:

	's' frames, 4–10 kHz cut: median · 90th pct · most	voice frames moved > 1 dB	pauses moved > 1 dB
VoiceBank test, split	3.4 · 6.6 · 8.0 dB	2.4%	0.7%
narrations, split	4.5 · 8.0 · 8.0 dB	2.4%	0.9%
sung, split	5.4 · 8.0 · 8.0 dB	0.5%	2.4%
VoiceBank test, band	1.7 · 4.7 · 7.7 dB	1.2%	0.4%
narrations, band	2.4 · 5.6 · 8.0 dB	1.2%	0.4%
sung, band	3.4 · 6.3 · 7.4 dB	0.4%	1.7%

Music ("Vibe Ace", Brahms, the Nutcracker, a trumpet): no frame moved by 1 dB in either mode. A ride cymbal 18, 12 and 6 dB under "Vibe Ace", the mix at −16 LUFS, its 4–10 kHz band cut: split, a median 0 dB (90th percentile 0, 1.1 and 4.4 dB); band, 0 dB (0, 0.8, 3.0). { mode: 'band', fc: 7500, Q: 1 }: 1.7 · 5.7 · 8.0 dB on the VoiceBank test set, 1.2 %.

Use when: harsh 's' / 'sh' in a voice, bright vocal takes; mode: 'band' for a lighter hand on one band.
Not for: broadband brightness: use an EQ. Generic compression: use compressor. A path that cannot wait 5 ms: lookahead: 0 (1.5 ms, the split's own).

ducker

External-sidechain compressor. Main signal's gain tracks the level of a separate side signal.

import { ducker } from '@audio/dynamics'

// batch
let podcast = ducker(music, voice, { threshold: -30, range: -12 })

// streaming — callable takes (main, side); call with no args to flush
let duck = ducker({ threshold: -30, range: -15 })
let out1 = duck(musicBlock1, voiceBlock1)
let out2 = duck(musicBlock2, voiceBlock2)
let tail = duck()

Param	Default
threshold	-30	dB (on side level)
ratio	4	—
knee	6	dB
range	-24	dB, max reduction
attack	20	ms
release	300	ms

Use when: music-under-voice podcasts, dialogue ducking, sidechain-pumped mixes.
Not for: sidechain from the same signal — use compressor.

softclip

Static waveshaping — no time state, no pumping. Maps input through a fixed transfer curve; peaks saturate smoothly, introducing controlled harmonic content.

Hard/high-drive clipping generates harmonics above Nyquist that fold back as audible aliasing. oversample (1/2/4/8, default 1) runs the transfer at N× rate and decimates back down through a windowed-sinc anti-alias filter, same technique as @audio/saturate's oversampled shapers — oversample: 1 is the exact non-oversampled path (no resampling, zero cost).

import { softclip } from '@audio/dynamics'

softclip(data, { curve: 'tanh', drive: 1.5 })
softclip(data, { curve: 'cubic', drive: 2, ceiling: 0.9 })
softclip(data, { curve: 'hard', drive: 4, oversample: 4, fs: 44100 })   // clean high-drive clip

Param	Default
curve	'tanh'	'tanh', 'atan', 'cubic', 'sin', 'hard'
drive	1	input pre-gain
ceiling	1	output asymptote
oversample	1	1, 2, 4, 8 — anti-aliased oversampling
fs	44100	Hz, sample rate (only used when oversample > 1)

Use when: gentle peak control with musical saturation, avoiding pumping artifacts of a limiter, lo-fi character; oversample for hard/high-drive clipping that must stay alias-free.
Not for: transparent true-peak safety — use limiter. Clean gain reduction — use compressor.

compand

SoX-style multi-segment compander. Arbitrary piecewise-linear transfer in dB unifies compression, expansion, and gating under one curve — points below the identity line compress; above, they expand.

import { compand } from '@audio/dynamics'

// Default: compress above -20 dB
compand(data)

// Broadcast leveler: lift -40..-20 dB, compress above -10 dB
compand(data, {
  points: [[-90, -90], [-40, -30], [-20, -18], [-10, -10], [0, -4]],
  attack: 20, release: 500
})

Param	Default
points	[[-90,-90],[-60,-60],[-20,-20],[0,-8]]	[[inDb, outDb], ...]
attack	5	ms
release	200	ms

Use when: broadcast leveling, speech normalization, any time a single compressor's fixed ratio is too rigid.
Not for: simple threshold compression — use compressor.

multiband

Multiband compressor — Linkwitz-Riley crossover split, an independent compressor per band (upward half included), flat sum by construction (SoX mcompand class). The manifest (multiband/audio) is a 3-band "one-knob" mastering stage — one shared setting across low/mid/high, split at low/high. The kernel (multiband(data, opts)) takes N-1 crossover points and per-band settings directly, for full control; every bands entry is spread straight into compressor(), so upward compression and depth are already there per band.

import { multiband } from '@audio/dynamics'

// one-knob: shared setting across 3 bands split at 200/2000 Hz
multiband(data, { freqs: [200, 2000], bands: { threshold: -24, ratio: 3 } })

// per-band settings, N bands (mutates data in place)
multiband(data, {
  freqs: [400, 4000],
  bands: [
    { threshold: -24, ratio: 3 },               // low
    { threshold: -20, ratio: 4, makeup: 2 },     // mid
    null,                                        // high: pass through uncompressed
  ],
})

OTT-class upward+downward multiband — Xfer OTT's "upward + downward compression on 3 bands" recipe, reproduced with this atom's upThreshold/upRatio/depth:

let depth = 1   // OTT's "Depth" macro — 0 is a transparent pass, 1 is full effect, up to 2 overshoots it
multiband(data, {
  freqs: [88.3, 2500],   // OTT's own crossover points
  bands: [
    { threshold: -24, ratio: 4, upThreshold: -30, upRatio: 2, attack: 2, release: 35, depth },  // low
    { threshold: -24, ratio: 4, upThreshold: -30, upRatio: 2, attack: 5, release: 60, depth },  // mid
    { threshold: -24, ratio: 4, upThreshold: -30, upRatio: 2, attack: 2, release: 35, depth },  // high
  ],
})

Param	Default
freqs	[200, 2000]	Hz, N-1 crossover points for N bands
bands	—	per-band {threshold, ratio, knee, attack, release, makeup, upThreshold, upRatio, upKnee, upRange, depth}, or one object shared by all bands; null passes a band through uncompressed
order	4	Linkwitz-Riley crossover order (2, 4, 8)
fs	44100	Hz

Manifest params (3-band one-knob form): low, high, threshold, ratio, upThreshold, upRatio, depth, attack, release, makeup.

Use when: mastering-stage glue across the spectrum; OTT-style "upward + downward everywhere" aggressive multiband; taming one band without touching others.
Not for: single-band dynamics — use compressor directly.

Additional processors

Export	Behavior and detailed options
transientShaper	Attack/sustain gain shaping; state continues on the reused options object.
opto	Optical-style compression, RMS detection and program-dependent release.
fet	FET-style compression, fast peak detection.
vca	VCA-style compression, feed-forward peak detection and a firm knee.
varimu	Variable-mu-style compression, level-dependent ratio.
leveler	Dialogue gain riding, whole-buffer analysis and smoothing.

These compressor models describe gain-control behavior; they do not model a hardware unit's full circuit or coloration.

The umbrella also exposes pure dB gain-curve helpers: compressorGain(levelDb, threshold, ratio, kneeDb), upwardGain(levelDb, threshold, ratio, kneeDb, rangeDb?), upwardExpanderGain(levelDb, threshold, ratio, kneeDb, rangeDb), and unlimitGain(fastDb, slowDb, amount, drive). They return gain in dB and do not modify audio. See the corresponding leaf declarations for parameter details.

See also

• denoise — umbrella for everything noise; its gate/deesser are seconds-unit adapters over this package (2026-07 near-dupe merge)
• filter — biquads for deesser sidechain
• effect — modulation effects
• stretch — sibling package

References

• Giannoulis, D., Massberg, M. & Reiss, J.D. (2012). "Digital dynamic range compressor design — a tutorial and analysis." JAES, 60(6).
• Izhaki, R. Mixing Audio: Concepts, Practices and Tools. Focal Press / Routledge. Four-quadrant dynamics taxonomy — downward/upward compression, downward/upward expansion.
• Zölzer, U. (ed., 2011). DAFX — Digital Audio Effects (2nd ed.), chapter on dynamics processing.
• Reiss, J.D. & McPherson, A. (2014). Audio Effects — Theory, Implementation and Application, Ch. 6.
• Bristow-Johnson, R. (2005). "Audio EQ Cookbook." (RBJ biquad formulae, used in deesser sidechain.)
• dbx Professional Products (1996). 902 de-Esser owner's manual (18-2015-B): sibilance detected in dB of the high band against the full band, "regardless of variations in signal levels"; Range 0–20 dB, past its normal region an 's' is "swallowed".
• Sonnox. Oxford SuprEsser user guide, §3.3.2: auto threshold that follows the signal outside the band, with a 24 dB window.
• FabFilter. Pro-DS help: Threshold, Range ("scales the detected gain reduction so that it stays within a desired range"), Wide Band / Split Band. Waves. Sibilance user guide: Range 0 to −48 dB.
• SoX manual — compand (piecewise-linear compander semantics).

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