Browser-based noise cancelling with AudioWorklet noise generation, NLMS adaptive predictor ANC, Wiener spectral suppression for speech and broadband noise, A-weighted analysis, 5-band EQ, and adaptive masking.
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Interpret your results
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Analyze ambient noise with your microphone and generate adaptive masking sounds through your headphones.
dB(A) applies A-weighting to approximate perceived loudness. Adaptive mode uses this for accurate tracking.
–6 dB/octave — deep rumble for traffic, HVAC, fans.
Auto-adjusts masking using A-weighted ambient analysis
The Noise Canceller uses your browser's Web Audio API with AudioWorklet processors running on a dedicated audio rendering thread. It captures ambient sound through your microphone, analyses it in real time, and generates both adaptive masking noise and active anti-noise through your headphones.
Three ANC modes run entirely inside AudioWorklet processors for minimum latency:
Captures mic audio, inverts its phase through a cascaded 18 dB/octave lowpass filter, and outputs the anti-noise. Simple and immediate, but limited by system latency — effective only for frequencies below roughly 1/(2 × latency). At typical browser latency of 14–40 ms, this works for sub-bass content.
The Normalized Least Mean Squares (NLMS) adaptive filter overcomes the latency barrier. It works by learning the periodic structure of ambient noise:
For periodic noise (fans, AC, hums, engine drone), the filter converges within seconds and can effectively cancel the tonal components. A convergence meter shows real-time MSE (mean squared error) — when it drops below 0.01, cancellation is active. The filter length controls how many frequency components can be modelled, while the learning rate controls the speed-stability tradeoff.
The most versatile mode, effective against non-periodic noise like speech, typing, keyboard clicks, and broadband hiss. It uses a 1024-point FFT with 75% overlap (256-sample hops) to analyze the frequency spectrum of incoming audio.
The AudioContext uses latencyHint: 0 (numeric zero) to request the absolute minimum latency the platform supports. The mic stream disables echo cancellation, noise suppression, and auto gain control to eliminate processing latency. Mono input (channelCount: 1) halves the processing load. The system measures and displays baseLatency + outputLatency so you can see your actual round-trip performance (Firefox typically achieves ~14 ms, Chrome ~19–40 ms).
Dual meters show raw dB and A-weighted dB(A) per ISO 61672-1. The adaptive mode uses dB(A) with an attack/release envelope for natural masking volume adjustments.
All processing runs in your browser. No audio is recorded, stored, or transmitted. The mic stream and all AudioWorklet processors are terminated when you stop the tool.