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DAQ fundamentals / 6 min read

Sample Rate, Fmax and the Anti-Alias Filter: Getting the Front End Right

An alias is indistinguishable from a real fault once it is in the file. A visual walk through sampling, folding, Fmax, lines and the anti-alias filter — and how to pick a DAQ channel that cannot lie to you.

01

An alias is not noise — it is a perfect impostor

Most acquisition mistakes are recoverable: re-capture, re-route a cable, rescale the units. Aliasing is the exception. Any frequency content above half the sample rate does not disappear — it folds back into the measured band and lands on a legitimate-looking frequency. A 9 kHz component captured at 8 kS/s is stored as a clean, stable 1 kHz tone.

Once saved, the alias is sample-for-sample identical to a genuine tone at the folded frequency. No post-processing can separate them, because the information needed to tell them apart was destroyed at the instant of sampling. In condition-monitoring terms, an alias can fabricate a bearing tone — or sit on top of a real one and corrupt its amplitude.

Folding: where out-of-band energy lands (sample rate 8 kS/s) measured band frequency (kHz) amplitude 0 2 4 6 8 10 Nyquist = fs/2 = 4 kHz real component, 9 kHz alias: 1 kHz the 9 kHz tone is stored as a 1 kHz line — inside the band, looking real
Folding on the frequency axis: with an 8 kS/s rate, Nyquist is 4 kHz. A real 9 kHz component reflects back into the measured band and is recorded as a 1 kHz line.
02

The wagon-wheel effect, on your DAQ

A filmed wagon wheel appears to spin slowly backwards because the frame rate undersamples the spokes. Every frame is a true photograph, yet the motion you perceive never happened. Sampling a voltage does exactly the same thing.

Watch the dots below: every sample lies exactly on the true 9 kHz signal — the ADC never records a wrong value. But connect the samples, which is all an FFT can do, and they trace a perfect 1 kHz sine. This is why the theorem demands sampling above twice the highest frequency present at the input, not just the highest frequency you care about.

A 9 kHz signal, sampled at only 8 kS/s grey: true signal · dotted verticals: sample instants · dots: what the ADC stores time → amplitude red: the 1 kHz alias — every sample is a true reading, yet the stored waveform never existed
Every dot is a genuine reading from the grey 9 kHz signal, yet the stored record is the red 1 kHz sine — a signal that was never present.
03

Fmax, lines and the resolution you actually get

Three chained settings decide what a spectrum can show. Fmax is the highest displayed frequency; analysers conventionally sample at 2.56 times Fmax to leave room for a real anti-alias filter. The line count divides 0-to-Fmax into bins, so resolution is delta-f = Fmax divided by lines — and the capture time per average is T = lines divided by Fmax, which is exactly 1 over delta-f.

That identity is the whole trade-off: finer resolution is bought with longer capture, nothing else. Sideband families — gear mesh sidebands at shaft speed, rotor-bar spacing — need delta-f several times finer than the spacing, and therefore seconds of steady data. A wide survey capture and a fine zoom capture are different measurements; plan both.

frequency → amplitude two real components, 3 Hz apart 3,200 lines at Fmax 1 kHz → Δf ≈ 0.31 Hz — capture T = 3.2 s both components resolved: two distinct peaks capture time T = 3.2 s 400 lines at Fmax 1 kHz → Δf = 2.5 Hz — capture T = 0.4 s the same two components smear into one broad peak capture time T = 0.4 s Δf = Fmax ÷ lines  ·  capture time T = lines ÷ Fmax = 1 / Δf
Same signal, same Fmax, different line counts. 3,200 lines resolves two components 3 Hz apart but needs 3.2 s; 400 lines finishes in 0.4 s and reports one broad peak.
04

The anti-alias filter: the mistake you cannot undo later

The only defence against aliasing is an analogue low-pass filter ahead of the ADC. It must exist in hardware: a filter applied in software afterwards can only delete real in-band content, while the aliases stay put, disguised as legitimate frequencies.

No real filter is a brick wall, so an honest front end never claims usable bandwidth all the way to Nyquist — there must be a guard band where the roll-off works. You can read this in the specifications: the PhonoVibe HD samples at 128 kHz per channel and quotes 0.5 Hz to 60 kHz of bandwidth, below the 64 kHz Nyquist limit; the 2-channel PhonoVibe D and the Eco X bench quote 20 kHz against a 48 kHz rate, with Nyquist at 24 kHz. A datasheet claiming bandwidth equal to half the sample rate is a warning sign, not a feature.

The anti-alias filter: analogue, before the ADC (PhonoVibe HD, 128 kS/s) frequency (kHz) response 0 60 64 80 guard band: roll-off happens here → Nyquist 64 kHz passband: specified 0.5 Hz – 60 kHz out-of-band tone: crushed before it can fold
A trustworthy channel specifies its bandwidth below Nyquist, leaving a guard band for the filter roll-off. Out-of-band energy is attenuated before the converter ever sees it.
05

Choosing a channel that cannot lie: a short checklist

Bandwidth honesty first: specified bandwidth comfortably below half the sample rate. Then simultaneous sampling — multiplexed channels corrupt cross-channel phase and quietly invalidate FRFs and coherence. Every PhonoVibe model, from the 2-channel D to the 16-channel HD, samples all inputs simultaneously with a 24-bit ADC per channel, and 24-bit range means small bearing tones stay visible next to large 1x components without gain juggling.

The rest of the chain matters equally: clean 24 V, 4 mA IEPE excitation (built into PhonoVibe; the TSP 02 and TSP 04 conditioners provide the same at unity gain, 0.5 Hz to 50 kHz, for instruments without it), a sensor matched to the job — such as the AC153-1D low-frequency accelerometer for machinery below 10 Hz — low-noise coaxial cable (CA-101 type) against triboelectric noise, and TEDS recognition to kill manual sensitivity-entry errors. The bundled TVIB TSAP 201 software then keeps sample rate, span, lines and averaging under your control rather than behind an auto mode.

06

Case study: commissioning a gearbox test cell

A representative workflow, not a specific customer: a reliability team instruments a motor-gearbox rig running near 2,980 RPM — shaft rate just under 50 Hz, a 23-tooth pinion putting gear mesh around 1.14 kHz with harmonics above that. A 4-channel PhonoVibe Q (128 kHz sampling, 0.5 Hz to 60 kHz bandwidth) is wired with IEPE accelerometers on low-noise coaxial cable at the bearing housings.

They capture in two passes, as sketched below — a wide survey to confirm where the energy really is, then a fine-resolution zoom on the mesh region so sidebands at the 50 Hz shaft spacing separate cleanly. One commissioning habit worth copying: if a suspicious line appears, re-capture at a different sample rate. A real component stays put; an alias moves, because its apparent frequency depends on the fold.

Pass 1 — survey: high Fmax, find the energy 0 – 50 kHz mesh ≈ 1.14 kHz + harmonics Pass 2 — zoom: low span, high lines mesh region, Δf ≪ 50 Hz sidebands at 50 Hz shaft spacing — resolved One instrument, two settings: the survey confirms where the energy is; the zoom buys resolution with capture time.
The two-pass plan: the survey pass locates the mesh energy across the full band; the zoom pass trades capture time for the resolution sideband analysis needs.
07

Where to learn this properly

The theory here — sampling, folding, FFT resolution, windowing — is covered in TIERA's free primers at 101.tieraonline.in: Signal Processing 101 and Accelerometer & DAQ 101. They are free introductory courses for onboarding and refreshing fundamentals, not accredited ISO certifications.

For formal, assessed competence in vibration analysis, TIERA runs the TCAT programme — details on the services page at /services — with proctored examinations at exams.tieraonline.in. A sensible path: primers first, hands-on time with a DAQ and a known signal source, then the formal programme when certification matters.

The kit for this job

TIERA instruments that do this work.

PhonoVibe Series — Sound & Vibration DAQ

PhonoVibe Series — Sound & Vibration DAQ

Quotes usable bandwidth well below Nyquist — 0.5 Hz – 60 kHz at 128 kHz on the Q/O/HD — a genuine guard band for the anti-alias roll-off, not a 'half the sample rate' promise.

ADC resolution
24-bit
Sampling (Q/O/HD)
128 kHz
Bandwidth (Q/O/HD)
0.5 Hz – 60 kHz
Bandwidth (D, 2-ch)
2 Hz – 20 kHz at 48 kHz
Sensor power
24 V, 4 mA (IEPE/ICP/CCLD)
TSP Series IEPE Signal Conditioners

TSP Series IEPE Signal Conditioners

Clean constant-current excitation at unity gain for front ends without built-in sensor power, so the noise floor is set by the sensor — not the supply.

Excitation
24 VDC, 4 mA constant
Frequency response
0.5 Hz – 50,000 Hz
Gain
1 (unity)
Connectors
BNC in / BNC out
T-Calibro Vibration Calibration System

T-Calibro Vibration Calibration System

A front end you never second-guess also needs a known sensitivity: back-to-back verification of your accelerometers on your own bench, with certificates from the bundled software.

Method
Back-to-back comparison
Reference frequency
159.2 Hz (ISO 16063 convention)
Compatibility
All IEPE/ICP accelerometers
Records
Automated certificate generation
From TIERA

Measure with a front end you never have to second-guess

The checklist in this post is how TIERA specifies its own hardware. Every PhonoVibe DAQ quotes its usable bandwidth below Nyquist — the 4-, 8- and 16-channel Q, O and HD sample at 128 kHz and specify 0.5 Hz to 60 kHz; the 2-channel D specifies 2 Hz to 20 kHz at 48 kHz — leaving a genuine guard band for the anti-alias roll-off instead of promising 'half the sample rate'. All models are 24-bit with simultaneous sampling on every input, so the cross-channel phase your FRFs and coherence depend on survives capture.

A channel you can trust also needs clean excitation and a known sensitivity. PhonoVibe builds in 24 V, 4 mA IEPE/ICP/CCLD power with TEDS recognition; for instruments without sensor power, the TSP 02 and TSP 04 conditioners supply the same excitation at unity gain from 0.5 Hz to 50 kHz. Every DAQ ships with a factory calibration certificate, and the T-Calibro back-to-back calibrator lets you verify accelerometer sensitivities on your own bench against a traceable reference, with certificates generated by its software.

  • PhonoVibe Q / O / HD — 128 kHz sampling with 0.5 Hz to 60 kHz specified bandwidth: a real guard band below the 64 kHz Nyquist limit
  • 24-bit ADC and simultaneous sampling on every channel, from the 2-channel D to the 16-channel HD — no multiplexing, no phase corruption
  • Built-in IEPE/ICP/CCLD excitation (24 V, 4 mA) and TEDS; TSP 02 / TSP 04 conditioners extend the same to third-party instruments
  • Factory calibration certificate with every unit, plus the T-Calibro back-to-back calibration system for in-house sensor verification
Learn this properly

Where this sits on the TIERA learning ladder.

The theory behind this article is covered free, in full, by the TIERA 101 primers: Signal Processing 101, Accelerometer & DAQ 101. They are self-paced, interactive, and end in an exam and a certificate.

The free 101 primers teach the sampling and FFT theory; the formal TCAT programme (see /services) adds structured coursework and proctored examinations at exams.tieraonline.in for teams that need assessed, certificated competence.

TIERA 101 is a free introductory primer, not an accredited ISO certification, and its hours do not count towards the formal training ISO 18436 requires.