The audio lab
Change it. See it. Hear it.
Three hands-on experiments in signal processing and acoustics.
Press a listen button to play a four-second sample. No microphone or uploads needed.
Signal processing
Find the signal in the noise.
A repeating musical tone, mixed with broadband hiss. How much can a simple filter help?
Shape the sound
Lower values mean more noise.
Lower cutoffs remove more hiss—and can dull the wanted signal.
What passes through?
Filter responseOutput SNR compares filtered signal power with filtered noise power. Signal retained shows output signal power relative to the clean input; listen for distortion as well as noise reduction.
Sensor conditioning, speech front ends and embedded audio. Filtering is a useful baseline; separating overlapping speech and noise calls for more advanced methods.
What this experiment models
A deterministic four-second harmonic melody at 24 kHz, with white noise scaled to the chosen input SNR, processed by a second-order Butterworth low-pass filter. The clean and noise components are filtered separately to calculate the displayed SNR. Playback uses the same gain across comparisons; there is no per-sample loudness normalization. This is classical DSP, not a trained speech-enhancement model.
Spatial acoustics
Point the listening beam.
Two tones arrive from different directions. Steer a virtual microphone array toward the sound you want.
Control the array
Microphone spacing: 4 cm. More microphones increase the aperture and sharpen directional selectivity.
Directional sensitivity
0° = broadside0 dB means fully retained; negative values mean attenuation. Directional response depends on frequency. Both sources can remain audible.
Microphone arrays, directional sensing and audio capture on devices. Geometry, frequency, reverberation and compute budgets shape real-world performance.
What this experiment models
An ideal uniform linear delay-and-sum array in the far field, with sound speed 343 m/s and equal source amplitudes. The chart uses the normalized complex array factor. Playback applies that factor’s gain and phase to each synthetic tone. It models two steady tones, not speech, moving sources, reverberation, microphone mismatch or an actual hardware recording.
Active noise control / ANC
Can sound cancel sound?
Match an opposing tone to a 180 Hz hum. Adjust its phase and amplitude to find the quiet point.
Tune the opposing sound
180° opposes the original tone; 0° reinforces it.
100% matches the amplitude of the original hum.
Two waves. One result.
At one observation pointNegative dB means attenuation; positive dB means reinforcement. Exact cancellation produces silence in this ideal simulation.
Real ANC depends on microphone and speaker placement, acoustic transfer paths, timing and stability. Cancellation at one point does not imply a quiet room; moving the listener or changing the sound changes the problem.
What this experiment models
Linear superposition of two steady 180 Hz tones at one observation point. Residual amplitude relative to the original is the magnitude of 1 + a·exp(jφ), where a is the amplitude ratio and φ is the phase offset. Playback uses the actual summed waveform with the same gain as the original. The phase and amplitude are set manually: this illustrates the cancellation principle, not a feedback controller, adaptive ANC algorithm or measured headphone performance. Graphs use normalized amplitude. All sounds are synthesized in the browser.