Filters

The Filters column of the analysis pipeline conditions the signal before any analysis is applied. It holds classic frequency-selective filters (lowpass, highpass, bandpass, bandstop, notch, peaking EQ), frequency weightings, and a set of signal-conditioning items such as differentiation, integration, resampling and scaling. Items are applied one after another, top to bottom, to every selected channel.

The Filters Pool

Add an item with Insert → Filter…, or by right-clicking the column and choosing Insert…. The Insert dialog groups the items under Basic Filters and Filters, with a search box and a My Favorites category.
Items run in column order. The order matters: integrating and then highpass filtering does not give the same result as filtering first and integrating afterwards.
Every item has a check box. Unchecking it keeps the item in the pool but skips it during the run — the fastest way to compare "with" and "without".
Bypass is a deliberate do-nothing item. Use it as a placeholder so the pool structure stays visible while a stage is temporarily disabled.
Select the item to edit its parameters in the properties panel on the right. Parameter changes take effect on the next run.
The filtered signal feeds the Analyses column; it can also be written out from the Destination column, for example as a WAV file.

Basic Filters (IIR)

The frequency-selective filters are recursive (IIR) second-order sections. Their shape is set by a corner or centre frequency and by the quality factor Q; the peaking equaliser adds a gain.

Filter	Parameters	Default	Typical use
Lowpass Filter	Cutoff (Hz), Q, Order	1000 Hz, Q = 0.707, order 2	Remove high-frequency content and hiss; limit the band before decimation.
Highpass Filter	Cutoff (Hz), Q	20 Hz, Q = 0.707	Remove DC offset, wind rumble and low-frequency drift.
Bandpass Filter	Centre (Hz), Q	1000 Hz, Q = 1	Isolate one component, e.g. a gear-mesh order or a resonance, before listening or analysis.
Bandstop Filter	Centre (Hz), Q	1000 Hz, Q = 1	Remove a disturbing band so that what remains can be judged on its own.
Notch Filter	Centre (Hz), Q	50 Hz, Q = 10	Suppress a single narrow interference line, typically mains hum at 50/60 Hz and its harmonics.
Peaking EQ	Centre (Hz), Q, Gain (dB)	1000 Hz, Q = 1, 0 dB	Boost or cut a band by a defined amount, for auralisation and A/B listening.

Q, Bandwidth and Slope

Q is the ratio of the centre frequency to the −3 dB bandwidth. A higher Q gives a narrower and more selective filter; a lower Q gives a gentler, wider one.

Q = f_centre / (f_upper − f_lower)

For lowpass and highpass filters, Q = 0.707 (1/√2) gives the maximally flat passband. Raising Q above this value creates a peak at the corner frequency.
Set Order to make the transition steeper. Order 2 is a single second-order section (12 dB per octave); orders 4, 6 and 8 add sections and give 24, 36 and 48 dB per octave. The −3 dB point stays at the frequency you entered — the sections are not simply repeated, each one gets the Q that the Butterworth pole distribution requires.
Q is only free at order 2, where you can raise it above 0.707 to create a peak at the corner. From order 4 on, the Q of every section is fixed by the chosen characteristic; for bandpass and bandstop the Q field then sets the width of the band instead.
Notch filters need a high Q (10 and above) so that only the interference line is removed and useful signal on either side survives.

Characteristic

Characteristic decides what the filter optimises for. All three are available on lowpass, highpass, bandpass and bandstop, and all take effect from order 4 upwards (order 2 is a single section and behaves the same in every case).

Characteristic	Passband	Transition	Phase / waveform	Use it when
Butterworth (default)	Maximally flat, no ripple	Moderate	Moderate	The safe default. Level, spectrum and psychoacoustic work, where a flat passband matters more than the last few dB of stopband.
Chebyshev (type I)	Equiripple — ripple depth is yours to set	Steepest at a given order	Worst	You must separate two components that sit close together and cannot afford a higher order.
Bessel	Flat	Gentlest	Best — maximally flat group delay	Transients, impulses and anything you will listen to or judge by waveform shape: a Bessel filter delays all frequencies by nearly the same amount, so pulses keep their shape.

Chebyshev and the cutoff frequency: for Chebyshev the frequency you enter is the ripple band edge — the gain there equals the ripple you set (for example −1 dB), not −3 dB. This is the standard definition for this characteristic; do not read it as the filter having "moved" its cutoff when comparing against Butterworth. The passband is scaled so that its peaks reach 0 dB and never exceed it, so a Chebyshev filter cannot amplify.

These are recursive (IIR) filters, so they are not phase-linear: different frequencies are delayed by different amounts. This is irrelevant for level, spectrum and psychoacoustic analyses, but keep it in mind when the exact shape of a transient or a phase relationship between two channels matters.

Frequency Weighting

Frequency Weighting applies a standardised weighting curve to the signal. The node name shows the selected curve, for example Frequency Weighting (A).

Setting	Meaning	Typical use
None	Unweighted (linear) — the signal passes through unchanged.	Physical measurements; vibration.
A	Approximates the sensitivity of the ear at moderate levels; attenuates low and very high frequencies.	The default for almost all noise-emission and workplace-noise work; results are quoted in dB(A).
B	Intermediate curve between A and C.	Rarely required; kept for comparison with historical data.
C	Nearly flat over the audible range with a gentle roll-off at both ends.	Peak levels and situations where the low-frequency content must be retained; quoted in dB(C).
D	Strong emphasis of the 2–4 kHz region, where the ear is most sensitive to the noise of jet engines.	Aircraft noise (IEC 537); quoted in dB(D).

The weighting is realised as a cascade of second-order sections normalised to 0 dB at 1000 Hz: A, B and C from the analog prototypes of IEC 61672, D from the pole/zero definition of IEC 537.

Frequency weightings A, B, C and D are defined for sound pressure. Do not apply them to acceleration, force or voltage channels — the resulting figures have no defined meaning.

Signal Conditioning and Utility Items

Item	Parameters	Default	What it does
Differentiate	Differentiation order	1	Differentiates with respect to time, once per order. Displacement → velocity → acceleration.
Integrate	Integration order; Highpass mode (Off / Relative / Absolute); Relative frequency (×f_s); Absolute frequency (Hz)	1; Off; 0.001; 1 Hz	Integrates with respect to time, once per order. Acceleration → velocity → displacement. The optional highpass removes the DC drift that integration otherwise accumulates.
Envelope	Smoothing (ms)	0 ms (no smoothing)	Extracts the amplitude envelope of the signal. Smoothing suppresses the residual ripple of the envelope.
Delay	Delay (ms); Zero-pad start/end	0 ms; off	Shifts the channel in time. A positive value delays, a negative value advances. With zero-padding the length is preserved; without it the signal is trimmed.
Fade In/Out	Fade mode (Linear / Cosine); Fade in (s); Fade out (s)	Linear; 0.01 s; 0.01 s	Applies a fade at the beginning and end of the signal, avoiding the click that an abrupt start produces during playback and export.
Linear Mapping	Factor mode (Linear / dB); Factor; Offset	Linear; 1; 0	Scales and offsets the samples: y = factor × x + offset. In dB mode the factor is entered as a level change.
Unit Conversion	Factor mode (Linear / dB); Factor; Offset	Linear; 1; 0	The same scale-and-offset operation, meant for converting a channel into another unit with a known conversion factor.
Resample	Target sample rate (Hz)	48000 Hz	Converts the channel to another sample rate. All following pool items work at the new rate.
Vector Magnitude	—	—	Combines all channels sample by sample into one channel: √(x₁² + x₂² + …). Used for triaxial accelerometers.
Bypass	—	—	Passes the signal through unchanged.

Integration and drift: any offset in the input becomes a ramp after one integration and a parabola after two. Set Highpass mode to Absolute and choose a corner frequency well below the lowest frequency of interest whenever you integrate a measured acceleration signal.

Resample: when the target rate is lower than the source rate, an anti-aliasing lowpass is applied automatically before the rate is changed, so content above the new Nyquist frequency cannot fold back into the band. Upsampling needs no such filter and gets none.

IIR and FIR

All items in the Filters pool are IIR designs: they are efficient, need only a few coefficients, and reach a given selectivity with a very low order — at the price of a frequency-dependent phase. FIR (finite impulse response) designs can be made exactly phase-linear but need many more coefficients for the same selectivity. Deepsona uses linear-phase FIR stages internally where phase behaviour is critical — for example inside the fractional-octave and wavelet filter banks and in the decimation stages of the psychoacoustic analyses — so those results are not affected by the phase behaviour of the pool filters.

Practical Tips

Filter as little as possible. Every filter changes the signal; if an analysis already offers the band limit you need, use the analysis parameter instead of a pool filter.
Put a highpass first when the recording has a DC offset — it protects everything downstream, especially integration and envelope analysis.
To hear what a component sounds like on its own, chain a bandpass around it and send the result to a WAV destination.
Give each filter node a meaningful name (F2) such as "HP 20 Hz" — the name is carried into the result titles and reports.
Duplicate a pipeline and disable a filter in one copy to document its effect side by side in the same result page.