WASM의 오디오 처리 및 에셋 스트리밍

Audio DSP — mixing, filtering, resampling — is number-crunching that benefits from WASM speed. The browser exposes the Web Audio API, and WASM fills the heavy processing role.

An AudioWorklet runs on the audio render thread. You can call WASM-compiled DSP code from inside its process() callback for low-latency synthesis.

Audio frames are Float32Array blocks (typically 128 samples). You copy these into WASM linear memory, process, and read back.

A minimal DSP kernel multiplies each sample by a gain factor — easy to express in C and compile to WASM.

For visualizers, share an AnalyserNode or feed FFT magnitudes from WASM into your render loop so visuals track the beat in real time.

Big textures and models should not block startup. Use fetch with a streaming reader to load and decode assets incrementally while rendering placeholders.

Custom binary formats (compressed meshes, audio codecs) can be decoded by WASM far faster than hand-written JS, keeping the main thread free.

When streaming, watch WASM linear memory growth. Free decoded chunks promptly and reuse buffers to avoid memory.grow thrashing.

Audio demands tight timing. Keep per-block WASM work under the buffer duration (128 samples at 48 kHz is ~2.7 ms) or you will hear glitches.

Move heavy asset decoding to a Web Worker running its own WASM instance, then transfer the decoded bytes to the render thread to keep frame rate smooth.

A media app typically has: an AudioWorklet for sound, a Worker for asset decode, and the main thread for WebGL/WebGPU rendering — all powered by separate WASM instances.

Where should low-latency audio DSP run in a WASM app?

You combined WASM with the Web Audio API via AudioWorklet for DSP, and used streaming fetch + worker decoding for large assets. Mind latency budgets and memory growth to keep both sound and visuals smooth.