ScriptProcessorNode ran audio processing callbacks directly on the browser's main UI thread. Because the main thread also handles DOM rendering, user input events, CSS layout recalculations, and JavaScript garbage collection, any main thread hesitation longer than a few milliseconds causes audio buffer underruns, producing audible stuttering, clicks, and pops. AudioWorklet executes inside a dedicated AudioWorkletGlobalScope running on a high-priority, real-time operating system audio rendering thread completely decoupled from the DOM. This ensures reliable 128-sample real-time processing without dropouts.

In Web Audio, audio is rendered in discrete blocks of exactly 128 samples per quantum. At a standard sample rate of 48,000 Hz, 128 samples represents only 2.667 milliseconds of audio (128 / 48000 s). To avoid audio underrun glitches, the AudioWorkletProcessor.process() method must complete all sample computations, filter equations, and buffer copies in well under 2.67 ms (ideally in under 0.3 ms). To meet this real-time deadline, developers must never allocate memory (no new object creations), avoid garbage collection triggers, and utilize WebAssembly SIMD for vector mathematical operations.

When transferring continuous audio data (such as mic recording or streaming PCM) between the main/worker thread and the AudioWorklet thread, traditional Mutex locks are forbidden: blocking an audio thread on a lock leads to priority inversion and severe buffer starvation. An SPSC ring buffer uses a pre-allocated SharedArrayBuffer with atomic head and tail pointers manipulated via Atomics.load() and Atomics.store(). Because only one thread writes and one thread reads, pointers can be updated without locks or thread synchronization stalls, achieving lock-free zero-copy stream transfer.

Custom AudioParams defined in an AudioWorklet can operate at two distinct sampling rates: k-rate parameters evaluate once per 128-sample render quantum (providing a single float value per block, ideal for coarse control like master volume or slow LFOs), while a-rate parameters calculate a distinct value for every single individual audio sample within the 128-sample array. a-rate automation is essential for high-speed frequency modulation (FM synthesis), rapid filter sweeps, and click-free envelopes, preventing stepping artifacts.