audio

Audio playback, editing and analysis

• Any Format — fast wasm codecs, no ffmpeg.
• Non-destructive — virtual edits, infinite undo, instant clone.
• Stream-first — playback/encode during decode, realtime editing.
• Paged — no 2Gb memory limit, open 10Gb+ files.
• Analysis — loudness, spectrum, beats, pitch, chords, key.
• Modular – pluggable ops, tree-shakable.
• CLI — playback, batch processing, scripting, unix pipes, tab completion.
• Cross-platform — browsers, node, deno, bun.

Start   Recipes   API   CLI   FAQ   Plugins   Architecture   Comparison

Start

Node

npm i audio

import audio from 'audio'
audio('voice.mp3').trim().normalize('podcast').fade(0.3, 0.5).save('clean.mp3')

Browser

<script type="module">
  import audio from 'https://esm.sh/audio'
  audio('./song.mp3').trim().normalize().fade(0.5, 2).clip({ at: 60, duration: 30 }).play()
</script>

CLI

npm i -g audio # or: npx audio …
audio voice.wav trim normalize podcast fade 0.3s -0.5s save clean.mp3

Skill

npx skills add audiojs/audio

MCP

claude mcp add audio -- npx -y audio --mcp
Prompt: make ~/Desktop/interview.m4a podcast-ready and tell me the loudness before and after

Recipes

Clean up

// master a raw take
let a = audio('raw-take.wav')
a.trim(-30).normalize('podcast').fade(0.3, 0.5)
await a.save('clean.wav')

// full restoration chain via ecosystem plugins (see API › Plugins)
a.gate(-45).dehum().deesser().compressor({ threshold: -18 }).limiter({ ceiling: -1 })

// cut 2:00–2:15, smooth the splice
a.remove({ at: 120, duration: 15 }).fade(0.1, { at: 120 })

// find clipped blocks
let clips = await a.stat('clipping')

// does it pass? each rule of the spec, measured
let { pass, rules } = await a.check('podcast')         // acx, podcast, streaming, broadcast, netflix

Master & deliver

// master a song to a reference track: its tone (mid and side), width and loudness, under -1 dBTP
audio('mix.wav').master(await audio('reference.wav')).save('master.wav')

// audiobook chapter for ACX: RMS -23..-18 dB, peaks under -3 dB, floor under -60 dB, room tone at each end, 44.1 kHz
let ch = audio('chapter-01.wav')
  .highpass(80).omlsa({ gMin: -12 }).compressor({ threshold: -24, ratio: 2.5 })
  .normalize(-20, 'rms', { ceiling: -3.5 })
  .trim().pad(1.5, 2).roomtone()                      // room tone, not digital silence
  .resample(44100)
console.log(await ch.check('acx'))                    // passed 10 of 10 real narrations (.work/pro.md)
await ch.save('chapter-01.mp3', { bitrate: 192 })

// tighten pauses in a talking-head video, then hand the cuts to the video editor
let talk = audio('talk.mp4').shrink(0.3)
await talk.save('talk.m4a')                            // the sound, cut
fs.writeFileSync('talk.edl', await talk.cuts('edl'))   // the same cuts for the picture (Premiere, Resolve)

Compose

// podcast montage
let ep = audio([intro, interview.trim().normalize('podcast'), outro], { crossfade: 0.5 })
await ep.save('episode.mp3')

// voiceover over music
music.gain(-12).mix(voice, { at: 2 })

// ringtone: the chorus + fades
audio('song.mp3').crop({ at: 45, duration: 30 }).fade(0.5, 2).normalize().save('ringtone.mp3')

// split an audiobook into chapters
let [ch1, ch2, ch3] = audio('audiobook.mp3').split(1800, 3600)

// glitch: stutter + reverse
let v = a.clip({ at: 1, duration: 0.25 })
audio([v, v, v, v]).reverse({ at: 0.25, duration: 0.25 })

Analyze

// waveform bars — and progressively, as it decodes
let [mins, peaks] = await a.stat(['min', 'max'], { bins: canvas.width })
a.on('data', ({ delta }) => appendBars(delta.max[0], delta.min[0]))

// features for ML
let mfcc = await a.stat('cepstrum', { bins: 13 })
let [loud, rms] = await a.stat(['loudness', 'rms'])

// notes, chords, key
let notes = await a.stat('notes')    // [{time, duration, freq, midi, note, clarity}]
let chords = await a.stat('chords')  // [{time, duration, label, root, quality, confidence}]
let key = await a.stat('key')        // {tonic, mode, label, confidence}

Record & generate

// mic take
let a = audio()
a.record()
// …later
a.stop()
a.trim().normalize()

// tone — any t => sample function
let tone = audio.from(t => Math.sin(440 * Math.PI * 2 * t), { duration: 2 })

// sonify data
let s = audio.from(t => Math.sin((200 + data[t / 0.2 | 0]) * Math.PI * 2 * t) * 0.5, { duration: data.length * 0.2 })

Automate

a.gain(t => -12 * (0.5 + 0.5 * Math.cos(t * Math.PI * 4)))  // 2Hz tremolo in dB
a.lowpass(t => 400 + 4000 * t)                              // filter sweep
a.pan({ t: [0, 2, 4], v: [-1, 1, -1] })                     // curve L→R→L over 4s, serializable
music.ducker({ key: voice })                                // sidechain (plugin)

Stream & persist

// stream to network — encode/playback during decode
for await (let chunk of audio('2hour-mix.flac').highpass(40)) socket.send(chunk[0].buffer)

// serialize edits, restore later
let json = JSON.stringify(a)     // { source, edits, ... }
let b = audio(JSON.parse(json))  // re-decode + replay edits

API

Create

Method	Description
audio(source, opts?)	decode from file, URL, bytes, or a byte stream. Returns instantly — decodes in background; streams decode as they arrive.
audio.from(source, opts?)	wrap existing PCM, AudioBuffer, silence, or function. Sync, no I/O.

let a = audio('voice.mp3')                // file path
let b = audio('https://cdn.ex/track.mp3') // URL
let c = audio(inputEl.files[0])           // Blob, File, Response, ArrayBuffer
let d = audio()                           // empty, ready for .push() or .record()
let e = audio([intro, body, outro])       // concat (virtual, no copy)
let f = audio([a, b, c], { crossfade: 2 })  // concat with 2s crossfade
let g = audio(process.stdin)              // byte stream: pipe, socket, fetch body – decodes as it arrives
// opts: { sampleRate, channels, crossfade, curve, storage: 'memory' | 'persistent' | 'auto' }

await a    // await for decode — if you need .duration, full stats etc

let a = audio.from([left, right])                 // Float32Array[] channels
let b = audio.from(3, { channels: 2 })           // 3s silence
let c = audio.from(t => Math.sin(440*TAU*t), { duration: 2 })  // generator
let d = audio.from(audioBuffer)                   // Web Audio AudioBuffer
let e = audio.from(int16arr, { format: 'int16' }) // typed array + format

Properties

Property	Description
.duration	total seconds, after edits.
.channels	channel count.
.sampleRate	sample rate.
.length	samples per channel.
.currentTime	what the speakers play now, in seconds: their latency compensated, smooth; at pause it holds where playback resumes.
.playing	true during playback.
.paused	true when paused.
.volume	0..1 linear. Settable.
.muted	mute, independent of volume. Settable.
.loop	settable, mid-playback too: the span repeats, each seam a 10 ms equal-power crossfade.
.playbackRate	0.0625..16, settable mid-playback; glides click-free (~50 ms, tape-style varispeed). .speed() bakes it.
.ended	true when playback reached the end, not after stop().
.seeking	true during a seek.
.played	promise, resolves when playback sounds.
.recording	true during mic recording.
.ready	promise, resolves when fully decoded.
.source	original source.
.bitDepth	stored sample depth of the source: 16, 24, 32 (float); null for lossy or generated audio. Lossless save keeps it.
.pages	Float32Array page store.
.stats	per-block stats (peak, rms, etc.).
.edits	edit list.
.version	increments on each edit.

Structure

Method	Description
.trim(threshold?)	strip leading/trailing silence (dB, default auto). On a live stream a given threshold streams, holding a silent tail until sound resumes; the automatic one reads the whole input, so it waits for the end.
.shrink(gap?, threshold?)	shorten silent pauses to gap seconds (default 0.3); 0 removes them. Streams with a given threshold, like trim. ≡ FFmpeg silenceremove, Audacity truncate-silence
.crop({at, duration})	keep range, discard rest.
.remove(at, duration, crossfade?)	delete range, close gap. crossfade ('10ms') makes the splice an equal-power crossfade centered on the cut; the length stays the same.
.insert(source, at?, crossfade?)	insert audio (default: at end), or a number of seconds of silence; crossfade fades both seams.
.copy({at?, duration?})	copy range (default: all) to this instance's clipboard.
.cut({at?, duration?, crossfade?})	copy, then remove.
.paste({at?, crossfade?})	insert the clipboard (default: at end).
.clip({at, duration})	zero-copy excerpt as a new instance.
.split(...offsets)	zero-copy excerpts between timestamps.
.pad(before, after?)	silence at edges (seconds).
.repeat(n)	repeat n times.
.reverse({at?, duration?})	reverse audio or range.
.speed(rate)	changes pitch and duration together.
.stretch(factor)	changes duration, keeps pitch (phase-locked vocoder). A t => f or {t, v} factor slides the tempo; duration becomes ∫factor dt.
.warp(markers)	move moments in time: [[from, to], …] in seconds. Between markers the audio stretches to fit, pitch kept; start and end stay. ≡ Logic Flex Time, Ableton warp markers
.pitch(semitones)	changes pitch, keeps duration.
.remix(channels)	channel count, or a map: [1, 0] swaps L/R.

Every op takes a trailing {at, duration, channel} range, except channel-changing remix and crossover. Times are seconds or strings ('1:30', '2m'); negative counts from the end.

a.trim(-30)                               // strip silence below -30dB
a.remove({ at: '2m', duration: 15 })      // delete 2:00–2:15, close gap
a.remove(12.3, 0.4, '10ms')               // cut a breath, crossfaded: no click
a.insert(intro, { at: 0 })                // prepend; .insert(3) appends 3s silence
a.copy(60, 30).paste(120)                 // duplicate the chorus at 2:00
a.cut(2, 1).paste(5)                      // move 2s–3s to 5s of the shortened timeline
let [pt1, pt2] = a.split('30m')           // zero-copy parts
let hook = a.clip({ at: 60, duration: 30 })  // zero-copy excerpt
a.stretch(1.1)                            // 10% longer, same pitch
a.warp([[1, 1], [2, 2.4], [3, 3]])        // the hit at 2s lands at 2.4s; 1s–3s keeps its length
a.pitch(-2)                               // 2 semitones down, same tempo
a.remix([0, 0])                           // L→both; .remix(1) for mono

Process

Method	Description
.gain(dB, opts?)	{ unit: 'linear' } takes a multiplier.
.fade(in, out?, curve?)	curves 'linear' 'exp' 'log' 'cos'. {start, end} levels 0..1 fade between any levels (a duck); {mid} skews the half-amplitude point. ≡ Audacity adjustable-fade
.normalize(target?, mode?)	remove DC, normalize. Loudness targets hold a true-peak ceiling, -1 dBTP by default: a lookahead limiter, then the loudness it took made back up. Presets per Apple Podcasts, Spotify, EBU R 128 (ITU-R BS.1770-4): 'podcast' -16 LUFS 'streaming' -14 LUFS 'broadcast' -23 LUFS -18, 'lufs' any loudness; -3 peak dB; no arg: peak 0 dBFS; 'rms' mode an audio instance: its integrated loudness { ceiling: -2 } dBTP, false off { dc: false } keep DC { adaptive: true } on a live stream, start at once: the gain follows what it has heard, the ceiling (the target itself in peak mode) guards what it hasn't. Without it, one gain for the whole selection: a live stream waits for its end. ≡ FFmpeg loudnorm
.roomtone(threshold?)	fill digital silence (≥ 10 ms under -90 dBFS: edited-out pauses, pad()) with the recording's own room tone. .trim().pad(1.5, 2).roomtone() gives an audiobook chapter its room tone at each end (ACX rejects digital silence). ≡ iZotope RX Ambience Match
.mix(source, at?, gain?)	overlay at at seconds, source level gain dB. ≡ FFmpeg amix weights
.crossfade(source, duration?, curve?)	append with overlap, default 0.5s. 'cos' (default) suits similar material; 'equal' (equal-power) keeps loudness across unrelated tracks. With no source, .crossfade({ at, duration }) crossfades across the range, as an editor crossfades a selection: the audio before it fades into the audio after it, and the range goes. ≡ FFmpeg acrossfade
.pan(value, opts?)	−1 left, 0 center, 1 right.
.write(data, {at?})	overwrite from at with raw PCM or another sound, as a tape records over what is there; what runs past the end extends it.
.transform(fn)	inline (input, output, ctx) => void. Not serialized.

a.gain(-3)                                // reduce 3dB
a.gain(6, { at: 10, duration: 5 })        // boost range
a.gain(t => -12 * Math.cos(t * TAU))      // automate over time
a.fade(0.5, -2, 'exp')                    // 0.5s in, 2s exp fade-out
a.normalize('podcast')                    // -16 LUFS, -1 dBTP
a.normalize(-27, 'lufs', { ceiling: -2 }) // Netflix
a.mix(voice, { at: 2 })                   // overlay at 2s
a.mix(bed, 0, -18)                        // music bed, 18 dB under
a.crossfade(next, 2)                      // 2s crossfade into next
a.crossfade(song2, 4, 'equal')            // equal-power, for unrelated tracks
a.pan(-0.3, { at: 10, duration: 5 })      // pan left for range

Filter

Method	Description
.highpass(freq, order?), .lowpass(freq, order?)	Butterworth pass filter; order 2 (12 dB/oct, default), 4 (24), 6, 8.
.bandpass(freq, Q?), .notch(freq, Q?)	band-pass / notch.
.allpass(freq, Q?)	phase shift, unity magnitude.
.lowshelf(freq, dB), .highshelf(freq, dB)	shelf EQ.
.eq(freq, gain, Q?)	parametric EQ.
.filter(type, ...params)	by type name, or a custom filter function.

All biquads.

a.highpass(80).lowshelf(200, -3)          // rumble + mud
a.eq(3000, 2, 1.5).highshelf(8000, 3)     // presence + air
a.notch(50)                               // remove hum
a.allpass(1000)                           // phase shift at 1kHz
a.filter(customFn, { cutoff: 2000 })      // custom filter function

Effect

Method	Description
.vocals(mode?, {model?})	mid/side: 'isolate' (default) keeps center, 'remove' keeps sides. model separates with a trained model instead, 'umxhq' (Open-Unmix, MIT weights) or 'htdemucs' (Hybrid Transformer Demucs, higher SDR, weights for research only), through the optional @audio/neural-separate; 'remove' then subtracts the model's vocals. Weights are exported locally (how) or served from weights. ≡ SoX oops; Demucs, Open-Unmix
.dither(bits?, {shape?})	TPDF, default 16-bit. shape: true adds 2nd-order noise shaping: quantization noise moves above ~Nyquist/2, audibly quieter.
.crossfeed(freq?, level?)	headphone crossfeed, default 700 Hz, 0.3. ≡ SoX earwax, bs2b
.resample(rate, {type?})	upsampling defaults to linear, downsampling to anti-aliased windowed sinc, its taps widening with the ratio. type: 'sinc' or 'linear' forces one.
.crossover(...freqs)	N split frequencies → N+1 bands × channels, band-major. Linkwitz-Riley 4th order; bands sum back flat. ≡ FFmpeg acrossover
.match(ref, amount?)	match EQ: up to 8 parametric bands fit to the reference/source spectrum ratio. Tone only; loudness stays with normalize. Streams {lookahead} s behind (10), refitting as it hears more. { midside: true } matches a stereo pair's mid and side apart, and the side level to the reference's width. ≡ iZotope Ozone Match EQ
.master(ref, opts?)	master to a reference track: match in mid and side, then normalize to the reference's integrated loudness under -1 dBTP ({ ceiling }). ≡ Matchering
.spectral(band?, gain?, {at, duration})	gain on a time × frequency region, band = [lo, hi] Hz; default removes it. ≡ Audacity spectral edit, FFmpeg afftfilt
.repair(band?, {at, duration, method?, window?})	rebuild a damaged range (dropout, beep, click burst) from its surroundings. method 'auto' (default) transplants the passage that joins seamlessly, searched in the window s (10) before the range; failing that, AR interpolation up to 70 ms, a sinusoidal bridge beyond. 'ar', 'sinusoidal', 'similarity', 'spectral' force one. ≡ iZotope RX Spectral Repair
.denoise(reduction?, threshold?, {noise})	remove a noise that holds still (hiss, hum and buzz, a fan, room tone, tape), learned where it plays alone: noise is that { at, duration } of the op's input, or several, or a print saved from stat('print'). It goes reduction dB down (12) everywhere, or in the op's own { at, duration }; what stays is the same noise, quieter, without musical tones. threshold (dB) raises the print: more of the quiet counts as noise. OM-LSA on the held noise (@audio/denoise-omlsa), each channel its own print; a live source renders once the range has arrived. VoiceBank+DEMAND PESQ, the noise learned from the half second before each speaker starts: noisy 1.97, omlsa() 2.40, denoise() 2.48. For noise that moves: omlsa(), deepfilter(). ≡ iZotope RX Spectral De-noise (Learn), Adobe Audition Noise Reduction (noise print), Audacity Noise Reduction
.deepfilter(limit?, floor?, {weights?, device?}), .rnnoise(limit?)	neural speech denoising through the optional @audio/neural-denoise: it also removes noise that moves (keys, traffic, a busy room). deepfilter runs DeepFilterNet3, its 8 MB model downloaded once, over the whole input before rendering; rnnoise streams RNNoise, weights in the package, 30 ms behind. deepfilter takes the noise limit dB down (12), or further, to floor dB under the voice's loudness (−45; false: the limit only): a narration keeps its room tone, noisy speech loses its noise. VoiceBank+DEMAND PESQ: noisy 1.97, wiener() 2.19, rnnoise() 2.46, deepfilter({ floor: false }) 2.67, deepfilter() 3.10, deepfilter(0) 3.16. limit 0 lifts the limit; rnnoise's 20 keeps it from removing the voice. Speech only: both drop music and singing. ≡ DeepFilterNet, RNNoise

a.vocals()                                // isolate center-panned vocals
a.vocals('remove')                        // remove vocals (karaoke)
a.vocals({ model: 'umxhq' })              // vocals by a separation model
a.dither(16)                              // TPDF dither to 16-bit
a.dither(16, { shape: true })             // noise-shaped
a.crossfeed()                             // headphone crossfeed
a.resample(48000)                         // resample to 48kHz (linear)
a.resample(96000, { type: 'sinc' })       // high-quality windowed-sinc
a.match(reference, 0.7)                   // 70% of the way to its tone
a.spectral([1000, 4000], -30, { at: 2.1, duration: 0.3 })  // a cough
a.repair({ at: 1.2, duration: 0.05 })     // a dropout
a.repair({ at: 42, duration: 1 })         // a lost second of music: the passage that fits
a.denoise({ noise: { at: 1.2, duration: 0.5 } })  // hiss learned from a pause, 12 dB down everywhere
a.deepfilter()                            // speech out of noise: the noise 45 dB under the voice, room tone kept
a.rnnoise()                               // the same, streaming

I/O

Method	Description
await .read(opts?)	rendered PCM. { format, channel } to convert. A source still arriving is waited for: a range until it has arrived, all of it until the end (an endless stream: read ranges, or stream()).
await .save(path, opts?)	encode + write, format from extension. Lossless keeps the source depth; { bitDepth, bitrate, quality, codec } set the encoder; m4a and mp3 write markers as chapters. Output streams as it encodes, headers patched with their totals at the end (a pipe keeps them "unknown"); m4a from a live source is fragmented. A video source saved to .mp4/.mov keeps its picture: only the audio track changes.
await .encode(format?, opts?)	encode to Uint8Array.
await .cuts(format?, opts?)	the edits as a cut list of the source file: 'edl' (CMX 3600: Premiere, Resolve, Avid), 'fcpxml' (Final Cut Pro, Resolve), 'otio' (OpenTimelineIO); none gives { fps, clips }. Cuts land on the video's frames (its MP4/MOV track), else { fps } (30). The CLI's save cuts.edl writes one.
.clone()	independent edits, shared pages.
.push(data, format?)	feed PCM into a pushable instance; .stop() finalizes.

let pcm = await a.read()                              // Float32Array[]
let raw = await a.read({ format: 'int16', channel: 0 })
for await (let block of a) send(block)                 // async-iterable over blocks
await a.save('out.mp3')                                // format from extension
await a.save('book.mp3', { bitrate: 192 })             // ACX: 192 kbps CBR
await a.save('master.wav', { bitDepth: 24 })           // 24-bit
await a.save('talk.mp4')                               // video in, video out
let bytes = await a.encode('flac')                     // Uint8Array
let b = a.clone()                                      // independent copy, shared pages

let src = audio()                                      // pushable source
src.push(buf, 'int16')                                 // feed PCM
src.stop()                                             // finalize

Playback / Recording

Method	Description
.play(opts?)	{ at, duration, loop, volume, rate, paused }. at defaults to currentTime (the start once ended); playing already, it jumps there without a gap.
.play({ from: b })	take over b's playback where it is (its span, loop, volume, rate, pause), crossfaded, no gap; b stops.
.pause(), .resume(), .seek(t), .stop()	each ramps over 5 ms, none clicks; seek crossfades, and in a loop stays in its span. stop() also ends recording.
.record(opts?)	mic. { deviceId, sampleRate, channels }.
audio.context	the page's one AudioContext, which playback uses: made on first use, resumed by the first gesture; set your own before playing.

Playback renders up to 2 s ahead into an AudioWorklet on audio.context (Node: @audio/speaker), so a busy main thread doesn't stop it, and sounds within milliseconds of play() (the device's own latency aside). An edit to the playing instance is heard ~50 ms later where it happens: the audio rendered ahead gives way, crossfaded. A source still arriving (decoding, pushed) plays what has come and goes on as more comes. Any channel count plays as it is; the device downmixes.

a.play({ at: 30, duration: 10 })          // play 30s–40s
await a.played                            // wait for sound
a.volume = 0.5; a.loop = true             // live adjustments
a.muted = true                            // mute without changing volume
a.playbackRate = 1.5                      // tape-style speed ramp
a.pause(); a.seek(60); a.resume()         // jump to 1:00
a.highpass(80)                            // an edit while playing: heard where it happens
b.play({ from: a })                       // b takes over at the same place, crossfaded
b.stop()                                  // end playback or recording

await audio.context.audioWorklet.addModule('./scrub.js')  // your own nodes, on the same context
let scrub = new AudioWorkletNode(audio.context, 'scrub')

let mic = audio()
mic.record({ sampleRate: 16000, channels: 1 })
mic.stop()

Metering

Method	Description
.meter(what, cb?)	live per-block stats of what plays, delivered as it is heard: rms, peak, ms, min, max, dc, clipping, spectrum, or your own. Without cb, read .value. Returns { value, stop() }. The same on a worker facade: measured in the worker, delivered on the page.

Option	Description
type	stat name, array of names, or omit for all block stats.
channel	n for one channel, [n, m] per-channel, or omit for scalar avg (mirrors a.stat()).
smoothing	one-pole EMA time constant τ, in seconds.
hold	peak-hold decay τ, in seconds.
bins, fMin, fMax	spectrum resolution and range (when type: 'spectrum').

a.meter('rms', v => draw(v))                                       // scalar avg across channels
a.meter(['rms', 'peak'], v => draw(v))                             // { rms, peak }
a.meter({ type: 'rms', channel: [0, 1] }, v => draw(v))            // [L, R]
a.meter({ type: 'spectrum', bins: 64, smoothing: 0.15 }, drawFFT)  // Float32Array of mel bins
a.meter({}, ({ delta, offset }) => draw(delta))                    // no type → all block stats

let m = a.meter({ type: 'rms' })                                   // pull form
requestAnimationFrame(function tick() { draw(m.value); requestAnimationFrame(tick) })
m.stop()                                                           // release

Analysis

Method	Description
await .stat(name, opts?)	one value; with { bins } a Float32Array; an array of names gives an array. { channel: n } one channel, [n, m] per channel; {at, duration} sub-range.
await .detect(opts?)	{ bpm, confidence, beats, onsets } in one pass; { channel } as in stat.
await .check(spec)	pass or fail against a delivery spec: { pass, rules: [{ name, value, unit, min, max, pass }] }. 'acx' (RMS, peak, noise floor, room tone, 44.1 kHz), 'podcast' (Apple: -16 LUFS ±1, ≤ -1 dBTP), 'streaming' (Spotify: plays at -14 LUFS, ≤ -1 dBTP), 'broadcast' (EBU R 128: -23 ±0.2 LUFS, ≤ -1 dBTP), 'netflix' (dialog -27 ±2 LUFS, ≤ -2 dBTP). Each limit cites its source in fn/check.js.

Stat	Description
'db'	peak amplitude in dBFS.
'rms'	RMS amplitude, linear (the CLI prints dBFS).
'noisefloor'	RMS of the quietest 0.4 s, dB: the room between words (ACX Check's measure, sample-exact).
'print'	the noise print of a range, as denoise({ noise }) takes it: dB in 1025 bands 23.4375 Hz apart, 0 to 24 kHz (white noise of RMS 0.01 prints −40).
'peak'	max(|min|, |max|), linear.
'loudness'	integrated LUFS (ITU-R BS.1770-4; surround channels weighted, LFE excluded).
'momentary', 'shortterm'	maximum 400 ms / 3 s loudness, LUFS (EBU Tech 3341).
'dialog'	loudness of the speech only, LUFS: speech found automatically (AES TD1008 dialog loudness).
'dc'	DC offset.
'clipping'	clipped samples, at 16-bit full scale (±32767/32768) or beyond (scalar: timestamps, binned: counts).
'silence'	silent ranges as {at, duration}.
'crest'	peak/RMS in dB. Sine ≈ 3dB, square ≈ 0dB.
'centroid'	spectral centroid in Hz (brightness).
'flatness'	spectral flatness: 0 tonal, 1 noise.
'correlation'	L/R phase correlation, −1 to +1. Mono returns 1.
'max', 'min'	peak envelope per bin, for waveforms.
'spectrum'	mel spectrum in dB (A-weighted); of several channels, their mean power.
'cepstrum'	MFCCs.
'bpm'	tempo.
'beats', 'onsets'	timestamps as Float64Array (seconds).
'notes'	[{time, duration, freq, midi, note, clarity}] (pYIN + Tony note HMM); with robust: true, the same through noise and rooms (a neural pYIN stage 1); with poly: true, polyphonic [{time, duration, freq, midi, note, velocity, bends}] (Basic Pitch).
'chords'	[{time, duration, label, root, quality, bass, confidence}]: Chordino on NNLS chroma (Mauch & Dixon 2010), matched to the reference plugin; labels like 'Am', 'G7', 'C/E', 'N'.
'key'	{tonic, mode, label, confidence} (Krumhansl-Schmuckler).

Opts: bpm, beats, onsets take { minBpm, maxBpm, delta, frameSize, hopSize }; notes takes { minFreq, maxFreq, frameSize, hopSize, minDuration }; chords, key take { frameSize, hopSize, tuning } (frames of 16384 samples at 44.1 kHz, 0.34 to 0.51 s at other rates, every eighth of a frame; concert A read from the audio unless tuning in Hz is given); chords also boostN (no-chord bias, 0.1); key also method: 'nnls' | 'pcp'. chords needs @audio/mir-nnls-chroma and @audio/mir-chordino, key needs @audio/mir-nnls-chroma: GPL-2.0-or-later translations of the reference plugins, installed by choice (npm i @audio/mir-nnls-chroma @audio/mir-chordino); key with method: 'pcp' needs only the MIT @audio/mir-chroma and @audio/mir-key, installed with audio unless optional dependencies are skipped. notes with robust: true needs @audio/neural-pitch (weights inside): a network's pitch candidates in place of YIN's keep the notes where YIN loses them (Vocadito onsets F 0.76 against 0.53 at 0 dB SNR) and trail it slightly on clean audio, so YIN stays the default. notes with poly: true takes { minFreq, maxFreq, minDuration, onsetThreshold, frameThreshold } and needs @audio/neural-transcribe, whose model downloads on first use; bends are cents from the note's pitch per 11.6 ms frame, in 33.3-cent steps (in-tune notes read 0).

let loud = await a.stat('loudness')                       // LUFS
let [db, clips] = await a.stat(['db', 'clipping'])        // multiple at once
let spec = await a.stat('spectrum', { bins: 128 })        // frequency bins
let [min, max] = await a.stat(['min', 'max'], { bins: 800 }) // peak envelope for canvas rendering
await a.stat('rms', { channel: 0 })                       // left only → number
await a.stat('rms', { channel: [0, 1] })                  // per-channel → [n, n]
let gaps = await a.stat('silence', { threshold: -40 })    // [{at, duration}, ...]
let bpm = await a.stat('bpm')                             // 120.5
let beats = await a.stat('beats')                         // Float64Array [0, 0.5, 1, ...]
let { bpm, confidence, beats, onsets } = await a.detect() // full pipeline, one pass
let notes = await a.stat('notes')                         // [{time, duration, freq, midi, note: 'A4', clarity}]
let chords = await a.stat('chords')                       // [{time, duration, label: 'Am', confidence}]
let k = await a.stat('key')                               // {label: 'C', mode: 'major', confidence}

Meta

Property	Description
.meta	tags: {title, artist, album, year, bpm, key, comment, pictures, raw, ...}. Writable. meta.raw holds format-specific blocks untouched (WAV bext/iXML, ID3v2 frames, FLAC blocks).
.meta.pictures	cover art [{mime, type, description, data, url}]. .url is a lazy Blob URL (browser) or data URL (Node).
.markers	[{time, label}] in output seconds; edits shift or drop them.
.mark(time, label?)	a marker at time, seconds of the audio as edited so far; later edits carry it. Chainable.
.regions	[{at, duration, label}]; edits shift or drop them.

Parsed on decode, written on save; round-trips WAV, MP3, FLAC.

let a = await audio('song.mp3')
a.meta.title                     // 'Track Name'
a.meta.artist = 'Me'             // mutate
img.src = a.meta.pictures[0].url // lazy Blob URL

a.crop({ at: 10, duration: 30 })
a.markers                         // re-projected — outside markers dropped, inside shifted

await a.save('edited.mp3')        // tags + pictures preserved
await a.save('stripped.wav', { meta: false })   // opt out

Utility

Method	Description
.on(event, fn), .off(event?, fn?)	subscribe / unsubscribe.
.undo(n?)	returns the undone edit, for redo via .run().
.run(...edits)	apply ['type', opts] edits: op params (value, freq, …) plus range keys.
.dispose()	release resources. Supports using.

Event	Description
'data'	pages decoded/pushed. Payload: { delta, offset, sampleRate, channels }.
'change'	any edit or undo.
'metadata'	stream header decoded. Payload: { sampleRate, channels, estDuration }: seconds it lasts, from the header where it says (WAV, AIFF, FLAC, an MP3's Xing or VBRI frame, a constant bitrate), else from its size; null when neither is known.
'timeupdate'	playback position, as heard (~50 times a second). Payload: currentTime.
'play'	playback started or resumed.
'pause'	playback paused.
'volumechange'	volume or muted changed.
'ended'	playback ended: at its end, by stop(), or taken over by play({ from }); not in a loop.
'progress'	during save/encode. Payload: { offset, total } in seconds.

a.on('data', ({ delta }) => draw(delta))  // decode progress
a.on('timeupdate', t => ui.update(t))     // playback position

a.run(
  ['gain', { value: -3, at: 10, duration: 5 }],
  ['crop', { at: 1, duration: 2 }],
  ['fade', { in: 1, curve: 'exp' }],
  ['insert', { source: ref, at: 2 }],
)
a.undo()                                  // undo last edit
b.run(...a.edits)                         // replay onto another file
JSON.stringify(a); audio(json)            // serialize / restore

Plugins

Method	Description
audio.use(...plugins)	register an @audio contract factory, a stat { stat, compute }, a codec { codec, test?, decode?, encode? }, a function receiving audio, or a registry name. Registry plugins need no use: a.compressor() and a.stat('truepeak') load them on first use.
audio.op(name, descriptor)	register an op: a process function or { params, process, plan, resolve }.
audio.op(name?)	one descriptor, or all ops.
audio.stat(name, descriptor)	register a stat: (chs, ctx) => [...] or { block, reduce, query }.

Plugins also run without the engine: audio/batch over a whole signal, audio/stream over live chunks. Plugin tutorial.

import { compressor } from '@audio/dynamics-compressor/audio'
audio.use(compressor)                       // bring-your-own factory

a.freeverb({ room: 0.8 })                   // registry plugin: loads on first render; tail composes
music.ducker({ key: voice })                // sidechain via the key option
await a.stat('truepeak')                    // stat plugins land on a.stat()

audio.op('crush', { params: ['bits'], process: (input, output, ctx) => {
  let steps = 2 ** (ctx.bits ?? 8)
  for (let c = 0; c < input.length; c++)
    for (let i = 0; i < input[c].length; i++)
      output[c][i] = Math.round(input[c][i] * steps) / steps
}})
a.crush(4)                                  // custom op, chainable like built-ins

Worker

Call	Description
audioWorker(source, opts?)	same API, engine in a Worker; the main thread keeps a few-KB facade.
audio(source, { worker: true })	same, once audio/worker is imported.
{ worker: new Worker(url) }	your own worker entry: codecs, plugins, your own code and messages, then audio/worker, which talks on a port of its own.
expose(a) → id, audioWorker.adopt(id, { worker })	hand an instance your worker made to the page as a facade.
audioWorker.context	the page's AudioContext, the same as audio.context.

Across the boundary clip(), split(), clone() return promises; op errors emit 'error'; functions don't cross, use {t, v} curves. play() renders in the worker straight into the page's AudioWorklet: the main thread can stall for seconds without a dropout. Architecture.

import audioWorker from 'audio/worker'
let a = audioWorker('track.mp3')            // decode/edits/stats/encode in a Worker
a.gain(-3).fade(0.5)
let [mins, maxs] = await a.stat(['min','max'], { bins: 640 })  // transferred, zero-copy
a.play()                                    // rendered in the worker, played by an AudioWorklet (Node: @audio/speaker)

// your own worker: its messages stay its own, and what it makes plays on the page
import audio from 'audio'                   // worker.js
import { expose } from 'audio/worker'
self.onmessage = ({ data }) => self.postMessage({ out: expose(audio(data.file).gain(-3)) })

let worker = new Worker('./worker.js', { type: 'module' })   // page
worker.onmessage = ({ data }) => audioWorker.adopt(data.out, { worker }).play()

CLI

npm i -g audio, or without installing: npx audio …

audio [source] [transforms...] [sink] [options]

A pipeline: a source produces audio, transforms reshape it, a sink consumes it. The default sink is stat — printing an overview.

# sources
FILE         path, URL, or glob  ('*.wav' for batch)
-            stdin (or omit when piping)
record       capture from microphone

# transforms (chained left-to-right)
gain         fade        trim        normalize   crop
clip         remove      reverse     repeat      pad
speed        stretch     pitch       insert      mix
crossfade    remix       pan         split       resample
highpass     lowpass     eq          lowshelf    highshelf
notch        bandpass    allpass     vocals      dither
crossfeed    shrink      crossover   match       spectral
repair       copy        cut         paste

# sinks (terminate the chain — at most one)
stat [NAMES...]    print analysis (default)
play [loop]        open player UI
save PATH          encode and write (or `-` for stdout); `192k` bitrate, `24bit` depth

# options
-f --force         overwrite existing output
--format FMT       override output format
--macro FILE       apply edits from JSON
--cue FILE         split at cue-sheet tracks (with split)
--verbose          show progress
--help, -h         help (or per-op: `audio gain --help`)
--mcp              serve the CLI to AI agents as an MCP tool (stdio)

# named options, after an op or sink: name:value
normalize -27 lufs ceiling:-2     ducker key:voice.wav     save out.m4a codec:alac

# compatibility shortcuts
-p ⇔ play     -l ⇔ play loop     -o PATH ⇔ save PATH

Playback

␣ pause · ←/→ seek ±10s · ⇧←/⇧→ seek ±60s · ↑/↓ volume · l loop · s save as · q quit

# play full song
audio song.mp3 play

# play fragment
audio song.mp3 10s..15s play

# play and loop a hook
audio song.mp3 30s..45s play loop

# play with effects applied live (streamable ops)
audio song.mp3 normalize broadcast highpass 80hz play

Edit

# clean up
audio raw-take.wav trim -30db normalize podcast fade 0.3s -0.5s save clean.wav

# scope a range (applies to whole chain)
audio in.wav 1s..10s gain -3db save out.wav

# range on a single op
audio in.wav gain -3db 1s..10s save out.wav

# filter chain
audio in.mp3 highpass 80hz lowshelf 200hz -3db save out.wav

# concat
audio intro.mp3 + content.wav + outro.mp3 trim normalize fade 0.5s -2s save ep.mp3

# crossfade into next
audio track1.mp3 crossfade track2.mp3 2s save mixed.wav

# voiceover
audio bg.mp3 gain -12db mix narration.wav 2s save mixed.wav

# music bed ducked under the voice (sidechain)
audio bed.mp3 ducker key:voice.wav mix voice.wav save episode.wav

# loudness to any target, true peak held; delivery settings
audio book.wav normalize -20 lufs save book.mp3 192k

# fix a video's sound, keep the picture
audio talk.mp4 highpass 80hz 4 normalize podcast save talk.clean.mp4

# master to a reference track (tone, width, loudness)
audio mix.wav master reference.wav save master.wav

# shorten pauses; the same cuts as an EDL for the video editor (.fcpxml, .otio too)
audio talk.mp4 shrink 0.3 save talk.edl

# split
audio audiobook.mp3 split 30m 60m save 'chapter-{i}.mp3'
audio album.wav split --cue album.cue save '{i} - {title}.mp3'   # cue-sheet tracks, tagged

# record
audio record 30s save voice.wav

Analysis

# overview (default sink)
audio speech.wav

# range overview — `audio FILE 0..10s` ⇔ `audio FILE stat 0..10s`
audio speech.wav 0..10s

# specific stats
audio speech.wav stat loudness rms

# tempo / beat grid / onsets
audio track.mp3 stat bpm
audio track.mp3 stat beats onsets

# loudness to spec: integrated, max momentary / short-term, speech only, true peak
audio mix.wav stat loudness momentary shortterm dialog truepeak

# pass or fail against a delivery spec (exit 1 on a fail); --json for scripts
audio episode.wav normalize podcast check podcast
audio chapter.wav check acx --json

# pitch / chords / key
audio song.mp3 stat notes
audio song.mp3 stat chords
audio song.mp3 stat key

# spectrum / cepstrum with bin count
audio speech.wav stat spectrum 128
audio speech.wav stat cepstrum 13

# stat after transforms (transforms apply, then stat)
audio speech.wav gain -3db stat db

Batch

audio '*.wav' trim normalize podcast save '{name}.clean.{ext}'
audio '*.wav' gain -3db save '{name}.out.{ext}'
audio 'chapters/*.mp3' check acx                      # a whole audiobook: one line per chapter

Stdin/stdout

cat in.wav | audio gain -3db save -      > out.wav
curl -s https://ex.com/speech.mp3 | audio normalize save clean.wav

Tab completion

eval "$(audio --completions zsh)"       # add to ~/.zshrc
eval "$(audio --completions bash)"      # add to ~/.bashrc
audio --completions fish | source       # fish

FAQ

What formats are supported?

Decode: WAV, MP3, FLAC, OGG Vorbis, Opus, AAC, AIFF, CAF, WebM, AMR, WMA, QOA via decode. Encode: WAV, MP3, FLAC, Opus, OGG, AIFF via encode. Codecs are WASM-based, lazy-loaded on first use.

Does it need ffmpeg or native addons?

No, pure JS + WASM. For CLI, you can install globally: npm i -g audio.

How big is the bundle?

~20K gzipped core. Codecs load on demand via import(), so unused formats aren't fetched.

How does it handle large files?

Audio is stored in fixed-size pages. In the browser, cold pages can evict to OPFS when memory exceeds budget — auto-sized from navigator.storage.estimate() (quota/4, 64MB..2GB), overridable via {budget}. Stats stay resident (~7 MB for 2h stereo).

Are edits destructive?

No. a.gain(-3).trim() pushes entries to an edit list — source pages aren't touched. Edits replay on read() / save() / for await.

Can I use it in the browser?

Yes, same API. See Browser for bundle options and import maps.

Does it need the full file before I can work with it?

No. Playback, edits, and structural ops (crop, repeat, pad, insert, etc.) all stream incrementally during decode — output begins before the file finishes loading. The edit plan recompiles as data arrives, tracking a safe output boundary per op. Only ops that depend on total length (open-end reverse, negative at) wait for full decode.

TypeScript?

Yes, ships with audio.d.ts.

Does it have feature parity with FFmpeg / SoX / librosa?

Yes — the audiojs ecosystem covers the practical baseline of FFmpeg filters, SoX effects, librosa analysis, Pedalboard and MIREX, all as @audio/* plugins audio wires through one API (the few uncovered items are esoteric or deliberately skipped). Every effect, filter, generator and analyzer lives in the registry — call one by name and it loads on first use. Coverage matrix: docs/comparison.md.

How is this different from SoX / FFmpeg / Audacity / librosa / Web Audio / Tone.js?

In one line: audio is the only one that runs the same API in Node and the browser, with non-destructive lazy edits that stream during decode. The native tools (SoX, FFmpeg) are faster on raw throughput but have no JS API, browser, or undo; the browser libs (Web Audio, Tone.js, Howler) are real-time graphs, not file editors. Full feature and performance matrices vs pydub, librosa, aubio, essentia, Pedalboard, SoX, FFmpeg, Audacity and MATLAB are in docs/comparison.md.

Built with

• decode – codec decoding (13+ formats)
• encode – codec encoding
• filter – filters (weighting, EQ, auditory)
• speaker – audio output
• mic – audio input
• pitch – pitch, chord, key analysis
• audio-type – format detection
• pcm-convert – PCM format conversion

MIT · ॐ