Playback Filters

Playback filters let you change what you hear without changing the measurement. They are used to isolate a suspicious component — a whine, a rattle, a hum — or to remove it and judge how much of the overall impression it accounts for. The filters run live on the replay signal; the stored measurement file is never modified.

Opening the Playback Filters Panel

  1. Open a Mark Editor document tab (see the Mark Editor topic).
  2. Switch the right-hand dock to the Playback Filters panel.

The panel is bound to the active Mark Editor document. If no such tab is in front, the panel shows the hint “Switch to a Mark Analyzer document page to edit the playback filters”. Each Mark Editor document carries its own filter chain.

Playback filtering is a licensed add-on. If the Mark Editor entries in the View menu and the toolbar are greyed out, the module is not enabled in your licence.

The Filter Chain

The chain is a series of independent slots. The replay signal passes through every enabled slot in order, so the resulting response is the product of all of them. A slot that is switched off is bypassed completely.

Source Slot 1 62.5 Hz Q 6 · −24 dB Slot 2 125 Hz Q 6 · −24 dB Slot 3 (off) 250 Hz bypassed Ear Frequency dB notch 1 notch 2
ControlFunction
+Adds a filter slot. The chain holds at most 6 slots.
−Removes the last slot. The chain keeps at least 3 slots.
Power symbolSwitches a slot on or off. Green = active, grey = bypassed; a bypassed slot is drawn dimmed.
Type buttonShows a short symbol for the current filter shape. Click it to choose another type from the menu.

A new chain starts with three slots, at 62.5 Hz, 125 Hz and 250 Hz, all switched off, so nothing is altered until you deliberately enable a slot.

Filter Types

TypeEffectTypical use
AllpassLeaves the magnitude unchanged.Neutral slot; a placeholder in the chain.
LowpassPasses everything below the cut-off.Removing high-frequency hiss or whine to judge the low-frequency character.
HighpassPasses everything above the cut-off.Removing wind, engine rumble or structural low-frequency content.
BandpassPasses a band around the centre frequency.Isolating one component so you can hear it alone.
BandstopSuppresses a band around the centre frequency.Removing one component to hear what is left.
Parametric BandpassThe dry signal summed with a band-pass branch, scaled by the gain (see below).Attenuating a tonal peak by an exact amount, or exaggerating it to confirm the diagnosis.
Parametric LowpassThe dry signal summed with a low-pass branch.Gentle correction of the low end.
Parametric HighpassThe dry signal summed with a high-pass branch.Gentle correction of the high end.

The parametric types are a dry/wet mix, not a symmetric bell

A parametric filter is not a bell curve that lifts or lowers a band as a whole. It sums the dry signal with a filtered branch, scaled by the gain:

H(f) = 1 + A · Hprototype(f), where A = 10dB/20 − 1

Several consequences are worth knowing, because they are not what a bell curve would do:

So a parametric band-pass at −24 dB does not sound like a band-stop: the first is a narrow cancellation notch, the second is the stop band of the filter itself. Use a parametric type to attenuate a band by a stated amount; use a band-stop to remove it.

Each type uses its own prototype characteristic rather than one shared design: Chebyshev for band-pass, Butterworth for band-stop and for low-/high-pass, Bessel for the three parametric types and for all-pass. This affects skirt steepness and group delay, so the same order behaves differently across types.

The blue curve on the spectrum plot is the combined response of the whole chain, and all eight types draw one. To display it, see the Filter Cursor section of Mark Analyzer — the lines can be dragged to change frequency, and dragging during playback takes effect immediately.

Filter Parameters

Every slot has the same four fields:

FieldMeaningRange / default
Frequency (Hz)Cut-off frequency for lowpass and highpass, centre frequency for bandpass, bandstop and the parametric types.Default 1000 Hz for a new slot
OrderSteepness of the slope. Higher values make the transition sharper and the filter more selective.1, 2, 3 or 4; default 2
BandwidthWidth of the affected band, expressed in the unit chosen next to the field.Default quality factor 6
Level (dB)Gain or attenuation applied by a parametric filter. Negative values cut, positive values boost. Whole decibels only.0 dB for a new slot

Values are committed when you leave the field, so type the number and press Tab or click elsewhere. Editing a field while playback is looping takes effect on the next pass.

A high order combined with a narrow bandwidth produces a very selective filter and can make the result ring audibly. If a filtered signal sounds hollow or metallic, reduce the order or widen the bandwidth before drawing conclusions from what you hear.

Bandwidth Units

The label next to the bandwidth field is clickable and selects how the width is expressed. The number in the field is converted automatically, so switching the unit does not change the filter.

UnitDescription
QualityQuality factor Q = centre frequency / bandwidth. Larger Q means a narrower filter.
HzAbsolute bandwidth in hertz.
BarkBandwidth on the Bark scale, i.e. in terms of critical bands, which matches how the ear resolves frequency.
1/n octaveBandwidth as a fraction of an octave; enter the value of n.
The alternative units are offered for the peaking (parametric bandpass) filter. The other filter types express their width as a quality factor, and the panel switches the unit back to Quality when you change such a slot to another type.

Setting Filters from the Spectrum Chart

Rather than typing numbers, you can set a filter directly on the spectrum you are looking at:

  1. Right-click an FFT chart in the Mark Editor document and choose Add filter.
  2. The Playback Filters panel opens and guide lines appear on the spectrum.
  3. Drag the centre line to change the frequency, and the handles to change gain and bandwidth.
  4. The slot fields follow the drag, and playback picks up the new setting.

This is the fastest way to place a notch exactly on a peak you can see: put the line on the peak, pull the gain down, and listen to whether the annoyance disappears.

A/B Listening Comparison

The point of a filter chain is the comparison, not the filter. Two ways to make it:

Filtered against unfiltered

  1. Enable Loop so a short section repeats.
  2. Start playback on the marked section.
  3. Toggle the slot's power symbol on and off between passes.

Because the loop re-renders the section on every pass, the change is applied on the next repetition and you hear the same passage with and without the filter.

Two recordings against each other

Use Pin A / Pin B and A ↔ B in the Player to switch between two files, as described in the Playback topic. Keep the filter chain, the gain and the volume unchanged while switching, otherwise you are comparing the settings and not the recordings.

Change one thing at a time. Toggling a single slot while everything else stays fixed tells you what that component contributes; changing frequency, gain and slot at once tells you nothing.

Filters and Exported Audio

Tips and Notes


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