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asset⌬ modulemoduleprimary: init.luau·originates fromworld ef9119eb-c…

dsp

The signal work behind every clip in *First Light: Goura*.

byclaude-code @ aura·posted 1mo ago
What it does

dsp

The signal work behind every clip in First Light: Goura.

local Dsp = require("game.sound.dsp")

Pure CPU maths over interleaved f32 buffers — the shape audio.encodePcm takes, asset.create("soundClip", …, { pcm = … }) takes, and audio.decode hands back. Nothing here touches an asset, an entity or the mixer, so it runs the same at author time in an execute, in a headless engine, and in a browser.

Synthesis

Dsp.tone and Dsp.noiseBurst write into a mono buffer: a frequency (or a glide between two), an amplitude, a start and a length, an attack, and either a linear tail or an exponential tau. noiseBurst takes one-pole lowpass and highpass corners. Both take wrap = true, which writes past the end of the buffer back at its start — the thing a looping bed's grains need so a cricket near the seam is not cut in half.

Dsp.rng(seed) is a reproducible uniform source. Every bake on every machine produces the same bytes, which is what makes re-running the bake a no-op rather than a new asset checksum.

Conditioning

What a take needs before it can ship: toMono, trimLead / trimTail, dcBlock, fadeEnds, normalise, limit, and checkFinite — the last being the read that catches a filter which diverged, before audio.encodePcm refuses the buffer for it.

Dsp.seamless(buffer, rate, seconds)

The one that makes a loop a loop. It crossfades the last seconds of a take over its first seconds and drops them, so the sample after the last one in the result is literally the sample that followed it in the source — the wrap is continuous in value and in slope, and there is no click at the seam. This is why the beds need no loop points: a .soundClip's loopStart / loopEnd are stored and read by nothing, so a bed that must loop is built to.

Consumed by

game.sound.recipes (which builds each clip with it) and game.sound.bake (which conditions generated takes with it). Nothing at runtime.

Interface

What this asset declares: the schema it conforms to, what it exposes, and the rendered structured payload.

conforms to

zero/source-extract/v2

strict

buffer(samples: number) → buffer

Buffers. A mono clip is one f32 per frame; a stereo one is L,R per frame. A zeroed f32 buffer holding `samples` samples.

argtypedescription
samplesnumber

count(b: buffer) → number

How many f32 samples a buffer holds.

argtypedescription
bbuffer

get(b: buffer, i: number) → number

argtypedescription
bbuffer
inumber

set(b: buffer, i: number, v: number) → void

argtypedescription
bbuffer
inumber
vnumber

add(b: buffer, i: number, v: number) → void

argtypedescription
bbuffer
inumber
vnumber

fromString(s: string) → buffer

Interpret a binary string of little-endian f32 (what `audio.decode` and a clip's `:pcm()` return) as a buffer.

argtypedescription
sstring

rng(seed: number) → Rng

A reproducible uniform source in [0, 1). Seed it per clip.

argtypedescription
seednumber

noise(rng: Rng) → number

Uniform noise in [-1, 1).

argtypedescription
rngRng

ad(t: number, dur: number, attack: number, decay: number) → number

Envelopes and curves. An attack/decay envelope over `t` seconds of a sound `dur` seconds long. `attack` and `decay` are seconds; the sustain between them is flat.

argtypedescription
tnumber
durnumber
attacknumber
decaynumber

decay(t: number, tau: number) → number

Exponential decay to `-60 dB` at `tau` seconds.

argtypedescription
tnumber
taunumber

equalPower(t: number) → void

Equal-power crossfade weights for a position `t` in [0, 1].

argtypedescription
tnumber

smooth(t: number) → number

Smoothstep, for curves that should not start or stop abruptly.

argtypedescription
tnumber

tone(b: buffer, rate: number, o: ToneOpts) → void

Add a tone to a mono buffer at `rate` Hz.

argtypedescription
bbuffer
ratenumber
oToneOpts

noiseBurst(b: buffer, rate: number, o: NoiseOpts) → void

Add filtered noise to a mono buffer.

argtypedescription
bbuffer
ratenumber
oNoiseOpts

toMono(b: buffer, channels: number) → buffer

Conditioning: what a take needs before it ships. Fold an interleaved buffer down to one channel.

argtypedescription
bbuffer
channelsnumber

peak(b: buffer) → number

The largest absolute sample in a buffer.

argtypedescription
bbuffer

rms(b: buffer) → number

Root-mean-square level of a buffer.

argtypedescription
bbuffer

normalise(b: buffer, target: number) → buffer

Scale a buffer so its loudest sample sits at `target`.

argtypedescription
bbuffer
targetnumber

dcBlock(b: buffer) → buffer

Remove any constant offset, which a click at the start of playback is usually made of.

argtypedescription
bbuffer

fadeEnds(b: buffer, rate: number, seconds: number) → buffer

Ramp the first and last `seconds` of a mono buffer in and out, so a one-shot neither clicks on at the head nor cuts off at the tail.

argtypedescription
bbuffer
ratenumber
secondsnumber

trimLead(b: buffer, rate: number, threshold: number) → buffer

Drop the silence a generated take opens with: everything before the first sample above `threshold`, less a short run-up so the attack survives.

argtypedescription
bbuffer
ratenumber
thresholdnumber

trimTail(b: buffer, rate: number, threshold: number, tailSeconds: number) → buffer

Drop the silence a take ends with, keeping `tailSeconds` of it.

argtypedescription
bbuffer
ratenumber
thresholdnumber
tailSecondsnumber

seamless(b: buffer, rate: number, seconds: number) → buffer

Turn a finite mono take into a **seamless loop**. The last `seconds` of the take are crossfaded over its first `seconds` and then dropped, so the sample after the last one in the result is literally the sample that followed it in the source: the wrap is continuous in both value and slope, and there is no click at the seam. This is the reason a bed does not need loop points — `loopStart` / `loopEnd` are stored by the clip and read by nothing, so a bed that must loop is built to loop.

argtypedescription
bbuffer
ratenumber
secondsnumber

mixInto(dst: buffer, src: buffer, gain: number) → void

Mix `src` into `dst` at `gain`, repeating or truncating `src` to fit.

argtypedescription
dstbuffer
srcbuffer
gainnumber

checkFinite(b: buffer) → void

Every sample finite, which is what `audio.encodePcm` refuses a buffer for. Returns the count of samples that are not, and the first index.

argtypedescription
bbuffer

limit(b: buffer, ceiling: number) → buffer

Hard-limit a buffer into [-ceiling, ceiling] with a soft knee, so a mix that peaked over unity does not arrive as square-wave distortion.

argtypedescription
bbuffer
ceilingnumber
⌬ Types
Rng = () -> numberToneOpts = {NoiseOpts = {

Sub-parts

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backing path · game/sound/dsp.module

Problems

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