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BandPass Academy Amateur radio licence exam practice

Units published 16/31 Built 28 Sep 2026

E1 · Published 26 September 2026 ·

Digital Signal Processing: Sampling, Quantising and Filtering

A modern transceiver converts the radio signal to numbers, does everything to the numbers, and converts them back. This unit covers the rules that conversion must obey, and what happens when it does not.

  • Band none
  • Study time 18
  • Level Core
  • questions 10
Waveform and antenna diagram for Signal Processing and Digital Communications
Signal Processing and Digital Communications — waveform overlay and radiation pattern

Pre-flight briefing

Two rules govern the conversion

The sampling theorem says that a signal must be sampled at least twice as fast as the highest frequency it contains. Sample at 8,000 samples per second and the highest frequency that can be represented unambiguously is 4,000 Hz - the Nyquist frequency. Anything above that does not disappear; it folds back into the band and appears at a frequency that is not really there, an effect called aliasing. A 5 kHz component sampled at 8 kHz reappears as a 3 kHz tone, and once it has been aliased there is no way to tell it apart from a genuine 3 kHz signal.

That is why a low-pass filter precedes the analogue-to-digital converter. The anti-aliasing filter removes energy above the Nyquist frequency before conversion, because after conversion it is too late. On the output side a reconstruction filter does the same job in reverse, smoothing the staircase of samples back into a continuous waveform.

The second rule concerns amplitude. Quantisation assigns each sample to the nearest of a finite set of levels, and the error is at most half a step. Each additional bit doubles the number of levels and improves the signal-to-noise ratio by about 6 dB, so an ideal 16-bit converter has roughly 96 dB of dynamic range while an 8-bit converter has about 48 dB - which is why a receiver's performance is often described by its ADC resolution.

What the numbers then allow

Once the signal is a sequence of numbers, filtering becomes arithmetic. A digital filter is a set of multiplications and additions whose coefficients determine the response: it can be linear phase, which no analogue filter achieves exactly, it can have a bandwidth of a few hertz at audio rates, and it never drifts with temperature or age because its coefficients are stored values. Changing the coefficients changes the filter - which is how one piece of hardware becomes a dozen different bandwidths in a modern receiver.

The fast Fourier transform converts a block of samples from the time domain to the frequency domain, computing what a spectrum analyser displays. In a receiver it produces the waterfall and panadapter display; in a transmitter it is used for modulation and for shaping. Because it processes a block of samples at once, it is a batch calculation rather than a continuous process, and its resolution depends on how many samples are in the block.

Decimation reduces the sample rate by discarding samples in a controlled way after filtering, which eases the processing load once a narrow signal has been isolated. Interpolation does the reverse, raising the effective rate before conversion back to analogue. Together they allow a receiver to spend its computation where the signal actually is.

Why this replaced the analogue chain

A software defined radio moves the boundary between hardware and software as far toward the antenna as the technology allows: instead of mixing, filtering and detecting with discrete components, it digitises a wide slice of spectrum and does all of it arithmetically. The advantages are repeatability, flexibility and the ability to implement filters and detectors that have no practical analogue equivalent.

The costs are real too. A wideband front end accepts every strong signal in the band it digitises, so it needs a high dynamic range and good linearity; a converter that clips on a strong nearby signal generates spurious responses across the whole captured spectrum. Digital processing also consumes power and generates its own clock-related spurs, which is why the analogue front end has not disappeared - it has simply moved.

Terms you must be able to define
TermWhat the examiner wants to hear
Nyquist frequencyHalf the sampling rate: the highest frequency that can be represented without aliasing.
AliasingFolding of signals above the Nyquist frequency back into the band as false responses.
QuantisationAssigning each sample to one of a finite set of levels; about 6 dB of dynamic range per bit.
Anti-aliasing filterA low-pass filter placed before the ADC to remove energy above the Nyquist frequency.
FFTFast Fourier transform: converts a block of samples from the time to the frequency domain.

Question deck 10 questions

E2A01 Recall

What does the sampling theorem require?

E2A02 Core

What is aliasing?

E2A03 Core

What is the purpose of an anti-aliasing filter in a digital receiver?

E2A04 Core

How much dynamic range does each additional bit of an analogue-to-digital converter add?

E2A05 Core

What does a fast Fourier transform produce from a block of samples?

E2A06 Core

What is a significant advantage of a digital filter over an analogue one?

E2A07 Stretch

What is decimation in a digital signal processing chain?

E2A08 Stretch

What is a principal risk of a wideband software defined radio front end?

E2A09 Core

What is quantisation error and how is it reduced?

E2A10 Stretch

Why does a modern receiver often describe its performance in terms of ADC resolution?