Single-pass noise reduction. 13 specialised methods + an auto-classifier.
| Domain | Targets | Quality | CPU | Best for | |
|---|---|---|---|---|---|
| denoise | meta | auto | — | varies | "just clean it" |
| gate | time | silence | ★ | very low | hard cut at threshold |
| dehum | time | mains hum | ★★★★ | very low | 50/60 Hz + harmonics |
| specsub | freq | broadband stationary | ★★ | medium | baseline |
| wiener | freq | broadband stationary | ★★★ | medium | general broadband |
| omlsa | freq | broadband non-stationary | ★★★★ | high | speech in changing noise |
| declick | time | impulses | ★★★★ | medium | vinyl pops, edit clicks |
| decrackle | time | dense impulses | ★★★ | medium | shellac crackle |
| declip | time | hard clipping | ★★★ | medium | restoration |
| dewind | time | LF rumble | ★★★ | very low | wind, handling noise |
| deplosive | time | LF bursts | ★★★ | low | mic plosives (p, b) |
| deesser | time | sibilance | ★★★★ | low | voice (s, sh) |
| debreath | time | inter-word noise | ★★★ | low | breath / hiss in pauses |
| dereverb | freq | late reverb | ★★ | medium | moderate room reverb |
For broader DSP needs use stretch, shift, pitch, beat.
npm install @audio/denoiseimport { denoise, dehum, wiener, declick } from '@audio/denoise'
let cleaned = denoise(samples) // auto-classify + dispatch
let unhummed = dehum(samples, { freq: 60 }) // explicit method
let { out, plan } = denoise(samples, { returnPlan: true }) // see what was chosen// Streaming — pass options first, then call repeatedly with chunks.
let write = wiener({ fs: 48000 })
write(block1)
write(block2)
write() // → flush remaining samplesMono
Float32Arrayin/out. State lives on theparamsobject; pass the same one across calls and biquad memory / spectral history persists. For stereo, process channels independently.
Content-aware auto-selector. Runs a single STFT classification sweep over the input and dispatches to the most suitable method.
denoise(data) // → cleaned Float32Array
denoise(data, { returnPlan: true }) // → { out, plan }
denoise(data, { force: 'wiener' }) // skip classifier| Param | Default | |
|---|---|---|
fs |
44100 |
Sample rate |
force |
— | One of 'dehum' | 'declick' | 'dewind' | 'deesser' | 'dereverb' | 'omlsa' | 'wiener' |
returnPlan |
false |
Return { out, plan } with classifier scores + chosen method |
Routing (in priority order):
- tonal hum (Goertzel — ≥2 of first 3 harmonics show 50× line/off-line ratio at 50 or 60 Hz)
- impulses (excess kurtosis of AR residual > 12)
- sibilance (high/mid band power ratio > 8)
- LF rumble (low/mid band power ratio > 3)
- non-stationary noise (CV of the rolling-minimum frame-energy floor > 0.3) → omlsa
- otherwise → wiener
Cascade of high-Q biquad notches at the fundamental + harmonics.
dehum(data, { freq: 60, harmonics: 4 })
dehum(data, { freq: 50, adaptive: true, drift: 0.5 }) // tracks slow mains drift| Param | Default | |
|---|---|---|
freq |
50 |
Fundamental (Hz) |
harmonics |
4 |
Number of notches placed |
Q |
30 |
Notch sharpness — higher = narrower |
adaptive |
false |
Goertzel sweep refines freq ± drift Hz |
Use when: mains buzz, ground-loop hum, fixed tonal interference.
Not for: broadband noise (use wiener/omlsa); shifting tones (use spectral methods).
Adaptive high-pass. Cutoff slides between cutoffMin and cutoffMax based on the LF/MF energy ratio.
dewind(data, { cutoffMin: 60, cutoffMax: 250 })| Param | Default | |
|---|---|---|
cutoffMin |
60 |
Hz — minimum cutoff (LF mostly clean) |
cutoffMax |
250 |
Hz — maximum cutoff (heavy rumble) |
order |
2 |
HP sections (each 12 dB/oct) |
Q |
0.707 |
Butterworth-ish |
blockSize |
1024 |
Coefficient update interval (samples) |
Use when: intermittent wind buffeting, handling thumps, low-frequency room modes — the adaptive cutoff opens on gusts and closes between them (measured: beats wiener on gusty wind at ~1/10 the CPU).
Not for: continuous rumble under speech — a time-domain cutoff can't separate overlapping spectra; use wiener/omlsa there (measured ~9 dB vs ~1 dB SNR gain). An LPC-null post-filter was evaluated and rejected: voiced speech is as AR-predictable as wind, so nulling wind poles whitens vowels too (LSD improves, SNR and speech level degrade).
Splits the signal into an LF band (< crossover) and its exact complement; ducks the LF band when its energy spikes above triggerRatio× the high band (a plosive signature). With no plosive present the output equals the input sample-for-sample — no crossover coloration.
deplosive(data, { triggerRatio: 4, attack: 0.005, release: 0.08 })| Param | Default | |
|---|---|---|
triggerRatio |
4 |
LF/high energy ratio that opens the duck |
attenuation |
-18 |
dB cut on the LF band when triggered |
crossover |
200 |
Hz — LF/high split point |
attack |
0.005 |
s |
release |
0.08 |
s |
Use when: mic plosives (p, b, t) producing low-frequency thuds.
Dynamic peaking EQ centred on the sibilance band. Detection runs on a HP side-chain; when the envelope exceeds threshold, a negative-gain peaking EQ at freq engages on the audio path. Re-computed every block samples for smooth gain riding. Backed by @audio/dynamics-deesser mode: 'band' since the 2026-07 near-dupe merge — same seconds-based options here.
deesser(data, { freq: 6500, threshold: -28, ratio: 4 })| Param | Default | |
|---|---|---|
freq |
6000 |
Sibilance centre (Hz) |
threshold |
-30 |
dBFS — engagement level |
ratio |
4 |
Compression ratio above threshold |
attack |
0.001 |
s — how fast the cut engages |
release |
0.05 |
s — how slowly it recovers |
Q |
1.4 |
Peaking EQ Q |
block |
64 |
Coefficient update interval (samples) |
Use when: voice post-production with hot s/sh; vocal bus de-essing.
Berouti spectral subtraction (1979). Estimates noise from the first noiseFrames (or tracks it via Minimum Statistics) and subtracts α(γ)·N̂(k) — an SNR-adaptive over-subtraction factor — from each magnitude frame, with a β·|Y(k)|² spectral floor. Pass an explicit alpha to force a fixed factor.
specsub(data, { beta: 0.02, noiseFrames: 6 }) // adaptive α(γ)
specsub(data, { alpha: 2, beta: 0.02 }) // fixed over-subtraction| Param | Default | |
|---|---|---|
alpha |
adaptive | Fixed over-subtraction factor; omit for Berouti α(γ) |
beta |
0.02 |
Spectral floor (fraction of the noisy spectrum) |
noiseFrames |
first 4 frames | Leading noise-only frames for the PSD bootstrap |
Use when: quick baseline; offline cleanup with a known noise-only preamble.
Not for: musical-noise-sensitive material — use wiener or omlsa.
MMSE Wiener / log-MMSE (Ephraim-Malah 1984/1985) with decision-directed a-priori SNR.
wiener(data, { rule: 'wiener' }) // pure Wiener gain
wiener(data) // defaults: 'mmse-lsa' rule| Param | Default | |
|---|---|---|
rule |
'mmse-lsa' |
'wiener' or 'mmse-lsa' (log-spectral, less musical noise) |
alpha |
0.98 |
Decision-directed smoothing (alias of alphaDD) |
frameSize |
2048 |
STFT frame |
hopSize |
frameSize/4 |
OLA hop |
noiseFrames |
first 4 frames | Leading noise-only frames for PSD bootstrap |
Use when: transparent broadband denoise; the "safe default" for stationary noise.
Optimally-Modified Log-Spectral Amplitude estimator (Cohen 2002) driven by IMCRA noise tracking. Combines an LSA gain with a minimum-gain floor weighted by speech presence probability:
G = G_LSA^p · G_min^(1-p).
omlsa(data)
omlsa(data, { gMinDb: -25 }) // less aggressive floor| Param | Default | |
|---|---|---|
gMinDb |
-20 |
dB floor for non-speech bins (alias of gMin) |
alpha |
0.92 |
Decision-directed smoothing (alias of alphaDD) |
frameSize |
2048 |
|
hopSize |
frameSize/4 |
Use when: speech in non-stationary noise (street, café, car); generally the highest-quality choice for noisy speech.
Detects impulses as AR-residual outliers (> threshold·σ); replaces each click region with an AR-LS interpolation (Janssen 1986 / Godsill-Rayner 1998).
declick(data, { threshold: 4, order: 60 })| Param | Default | |
|---|---|---|
threshold |
4 |
σ-multiple for click detection |
order |
60 |
AR model order |
guard |
2 |
Extra samples on each side of the detected click |
maxBurst |
64 |
Longest run repaired (longer → left as a real transient) |
Use when: vinyl pops, edit clicks, occasional impulse noise.
Not for: dense crackle (use decrackle); long dropouts (use arInterpolate directly).
Continuous AR-residual outlier detection with MAD-based threshold. Suited to high-rate impulse noise.
decrackle(data, { threshold: 3 })Use when: shellac / 78 RPM crackle; persistent low-amplitude clicks.
Detects runs of samples at ±clipLevel, fits AR on the un-clipped neighbourhood, extrapolates a sign-constrained interpolation.
declip(data, { clipLevel: 0.95 }) // explicit threshold
declip(data) // auto-detects clip level| Param | Default | |
|---|---|---|
clipLevel |
auto | Detected from histogram of |x| > 0.5 |
order |
100 |
AR model order |
maxRun |
order/2 |
Longest run that gets restored |
Use when: hard digital clipping with short clip runs.
Not for: sustained clipping covering many cycles (use sparsity-based methods).
Late-reverb suppression (Lebart, Boucher & Denbigh 2001 estimate, Habets-class gain). Models the late tail as a decaying sum of past frames' power, then applies a decision-directed Wiener gain on the signal-to-reverb ratio — the cross-frame smoothing suppresses the musical noise hard subtraction produces.
dereverb(data, { t60: 0.6, predelay: 0.04 })| Param | Default | |
|---|---|---|
t60 |
0.5 |
Assumed reverberation time (s) |
predelay |
0.04 |
Direct-sound passthrough (s) |
alpha |
1.5 |
Reverb-PSD over-estimation factor |
alphaDD |
0.98 |
Decision-directed SIR smoothing |
gMin |
0.05 |
Gain floor for reverb-dominated bins |
Use when: moderate room reverb (RT60 ≤ 1 s) on a single channel.
Not for: heavy reverb or convolutive distortion — use multi-channel WPE (out of scope).
Look-ahead noise gate with hysteresis. Backed by @audio/dynamics-gate since the 2026-07 near-dupe merge — same seconds-based options here; closeThreshold (default threshold − 6 dB) sets the hysteresis close level.
gate(data, { threshold: -45, attack: 0.005, release: 0.1, hold: 0.05, lookahead: 0.005 })Use when: silence enforcement; aggressive cut between phrases.
Not for: continuous denoise — use wiener/omlsa.
VAD-driven inverse gate. Uses energy + spectral flatness with a percentile-based noise floor; attenuates frames classified as non-speech with smooth attack/release.
debreath(data, { range: -10 }) // -10 dB on non-speech (default -12)Use when: breath, mouth noise, hiss in pauses on a voiceover.
import { snr, segSnr, lsd, nrr, speechAttenuation } from '@audio/denoise'
snr(reference, processed) // global SNR (dB)
segSnr(reference, processed) // segmental SNR (dB)
lsd(reference, processed) // log-spectral distance
nrr(noisyInput, processed) // noise reduction ratio
speechAttenuation(reference, processed) // dB lost on speech segments| Metric | Higher is better | What it captures |
|---|---|---|
snr |
✓ | Energy ratio reference / error |
segSnr |
✓ | Time-localised SNR — better correlates with perception |
lsd |
✗ | Mean log-magnitude error per bin |
nrr |
✓ | Floor reduction in non-speech regions |
speechAttenuation |
✗ | Loss of speech energy (over-aggressive denoising) |
import { stftBatch, stftStream, stftAnalyse } from '@audio/denoise'
import { vad, spp, ddSnr } from '@audio/denoise'
import { noiseProfile, minStats, imcra } from '@audio/denoise'stft*— analysis-modification-synthesis with Hann + ∑win² OLA reconstruction. Visit(mag, phase, state, ctx) => { mag, phase }.vad— frame-level activity (energy + spectral flatness, percentile floor).spp— per-bin Speech Presence Probability under Gaussian model.ddSnr— decision-directed a-priori SNR (Ephraim-Malah).noiseProfile— average PSD over leading frames.minStats— Martin (2001) minimum-statistics noise PSD tracker.imcra— Cohen (2003) Improved Minima-Controlled Recursive Averaging — drivesomlsa.
npm run measure produces a Markdown table of SNR / segSNR / LSD / NRR per method on canonical scenarios. Headline numbers on the included audio-lena fixture (8 s mono speech, 44.1 kHz):
| scenario | SNR-in | best method | SNR-out | NRR | ms |
|---|---|---|---|---|---|
| 60 Hz hum + harmonics | -5.2 dB | dehum |
15.0 dB | 6.3 dB | 5 |
| white noise (~13 dB SNR) | 13.2 dB | wiener |
19.8 dB | 0.3 dB | 82 |
| clicks (vinyl-style) | 24.1 dB | declick |
44.1 dB | — | 462 |
| 7 kHz sibilance | 2.0 dB | deesser |
9.5 dB | 1.9 dB | 5 |
Higher = better.
demo.html is a self-contained browser demo: pick a noise scenario, pick a method (or auto), inspect input/output waveforms, hear the difference, and read the live classifier scores.
- Boll, Suppression of Acoustic Noise in Speech Using Spectral Subtraction, IEEE TASSP 1979.
- Berouti, Schwartz, Makhoul, Enhancement of Speech Corrupted by Acoustic Noise, ICASSP 1979.
- Ephraim & Malah, Speech Enhancement Using a Minimum Mean-Square Error Short-Time Spectral Amplitude Estimator, IEEE TASSP 1984.
- Ephraim & Malah, Speech Enhancement Using a Minimum Mean-Square Error Log-Spectral Amplitude Estimator, IEEE TASSP 1985.
- Martin, Noise Power Spectral Density Estimation Based on Optimal Smoothing and Minimum Statistics, IEEE TSAP 2001.
- Cohen, Optimal Speech Enhancement Under Signal Presence Uncertainty Using Log-Spectral Amplitude Estimator, IEEE SPL 2002.
- Cohen, Noise Spectrum Estimation in Adverse Environments: Improved Minima Controlled Recursive Averaging, IEEE TSAP 2003.
- Janssen, Veldhuis & Vries, Adaptive Interpolation of Discrete-Time Signals That Can Be Modeled as Autoregressive Processes, IEEE TASSP 1986.
- Godsill & Rayner, Digital Audio Restoration, Springer 1998.
- Lebart, Boucher & Denbigh, A New Method Based on Spectral Subtraction for Speech Dereverberation, Acta Acustica 2001.
- RBJ Audio EQ Cookbook (biquad coefficients).
MIT