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Filter (signal processing)

Filter (signal processing) is a science topic covered in the lgStudy science library. This page brings together a partial reference excerpt, illustrations, worked examples, real-world applications and a short study plan, so you can understand Filter (signal processing) rather than just read about it. In short: In signal processing, a filter is a device or process that removes some unwanted components or features from a signal. Filtering is a class of signal processing, the defining feature of filters being the complete or partial suppression of some aspect of the signal.

Filter (signal processing) — main illustration
Filter (signal processing) — illustration

Key takeaways

  • Filter (signal processing) belongs to science; place it in that map before memorising details.
  • Learn the definition first, then one example that makes the definition concrete.
  • Connect Filter (signal processing) to a quantity you can measure, compute or draw — that is where exam questions come from.
  • Reproduce the core statement of Filter (signal processing) from memory before moving on to harder problems.

Reference excerpt

In signal processing, a filter is a device or process that removes some unwanted components or features from a signal. Filtering is a class of signal processing, the defining feature of filters being the complete or partial suppression of some aspect of the signal. Most often, this means removing some frequencies or frequency bands. However, filters do not exclusively act in the frequency domain; especially in the field of image processing many other targets for filtering exist. Correlations can be removed for certain frequency components and not for others without having to act in the frequency domain. Filters are widely used in electronics and telecommunication, in radio, television, audio recording, radar, control systems, music synthesis, image processing, computer graphics, and structural dynamics. There are many different bases of classifying filters and these overlap in many different ways; there is no simple hierarchical classification. Filters may be:

non-linear or linear time-variant or time-invariant, also known as shift invariance. If the filter operates in a spatial domain then the characterization is space invariance. causal or non-causal: A filter is non-causal if its present output depends on future input. Filters processing time-domain signals in real time must be causal, but not filters acting on spatial domain signals or deferred-time processing of time-domain signals. analog or digital discrete-time (sampled) or continuous-time passive or active type of continuous-time filter infinite impulse response (IIR) or finite impulse response (FIR) type of discrete-time or digital filter.

Linear continuous-time filters Linear continuous-time circuit is perhaps the most common meaning for filter in the analog signal processing world, and simply "filter" is often taken to be synonymous. These circuits are generally designed to remove certain frequencies and allow others to pass. Circuits that perform this function are generally linear in their response, or at least approximately so. Any nonlinearity would potentially result in the output signal containing frequency components not present in the input signal. The modern design methodology for linear continuous-time filters is called network synthesis. Some important filter families designed in this way are:

Chebyshev filter, has the best approximation to the ideal response of any filter for a specified order and ripple. Butterworth filter, has a maximally flat frequency response in the passband. Bessel filter, has a maximally flat phase delay in the passband. Elliptic filter, has the steepest cutoff of any filter for a specified order and ripple. The difference between these filter families is that they all use a different polynomial function to approximate to the ideal filter response. This results in each having a different transfer function. Another older, less-used methodology is the image parameter method. Filters designed by this methodology are archaically called "wave filters". Some important filters designed by this method are:

Constant k filter, the original and simplest form of wave filter. m-derived filter, a modification of the constant k with improved cutoff steepness and impedance matching.

Terminology Some terms used to describe and classify linear filters:

The frequency response can be classified into a number of different bandforms describing which frequency bands the filter passes (the passband) and which it rejects (the stopband): Low-pass filter – low frequencies are passed, high frequencies are attenuated. High-pass filter – high frequencies are passed, low frequencies are attenuated. Band-pass filter – only frequencies in a frequency band are passed. Band-stop filter or band-reject filter – only frequencies in a frequency band are attenuated. Notch filter – rejects just one specific frequency - an extreme band-stop filter. Comb filter – has multiple regularly spaced narrow passbands giving the bandform the appearance of a comb. All-pass filter – all frequencies are passed, but the phase of the output is modified. Cutoff frequency is the frequency beyond which the filter will not pass signals. It is usually measured at a specific attenuation such as 3 dB. Roll-off is the rate at which attenuation increases beyond the cut-off frequency. Transition band, the (usually narrow) band of frequencies between a passband and stopband. Ripple is the variation of the filter's insertion loss in the passband. The order of a filter is the degree of the approximating polynomial and in passive filters corresponds to the number of elements required to build it. Increasing order increases roll-off and brings the filter closer to the ideal response. One important application of filters is in telecommunication. Many telecommunication systems use frequency-division multiplexing, where the system designers divide a wide frequency band into many narrower frequency bands called "slots" or "channels", and each stream of information is allocated one of those channels. The people who design the filters at each transmitter and each receiver try to balance passing the desired signal through as accurately as possible, keeping interference to and from other cooperating transmitters and noise sources outside the system as low as possible, at reasonable cost. Multilevel and multiphase digital modulation systems require filters that have flat phase delay—are linear phase in the passband—to preserve pulse integrity in the time domain, giving less intersymbol interference than other kinds of filters. On the other hand, analog audio systems using analog transmission can tolerate much larger ripples in phase delay, and so designers of such systems often deliberately sacrifice linear phase to get filters that are better in other ways—better stop-band rejection, lower passband amplitude ripple, lower cost, etc.

Technologies Filters can be built in a number of different technologies. The same transfer function can be realised in several different ways, that is the mathematical properties of the filter are the same but the physical properties are quite different. Often the components in different technologies are directly analogous to each other and fulfill the same role in their respective filters. For instance, the resistors, inductors and capacitors of electronics correspond respectively to dampers, masses and springs in mechanics. Likewise, there are corresponding components in distributed-element filters.

… excerpt ends here. Continue reading the full article.

Illustrations

Filter (signal processing): A general finite impulse response filter with n stages, each with an independent delay, di and amplification gain, ai.
A general finite impulse response filter with n stages, each with an independent delay, di and amplification gain, ai.
Filter (signal processing): Crystal filter with a center frequency of 45 MHz and a bandwidth B3dB of 12 kHz.
Crystal filter with a center frequency of 45 MHz and a bandwidth B3dB of 12 kHz.
Filter (signal processing) illustration

Worked examples

Example 1 — a first encounter with Filter (signal processing)

Start with the simplest possible case. Write down what Filter (signal processing) claims or describes in one sentence, then invent the smallest concrete situation in which that sentence is true. In science, the smallest case is usually a single object, a single equation or a single measurement. Check that every symbol or term in your sentence has a meaning in that case.

Example 2 — changing one variable

Take the situation from Example 1 and change exactly one quantity: double it, halve it, or set it to zero. Predict what should happen to Filter (signal processing) before you calculate. Comparing your prediction with the result is the fastest way to find out whether you understand the idea or only the words.

Example 3 — an exam-style question

Typical questions about Filter (signal processing) ask you to (a) state it precisely, (b) apply it to given data, and (c) explain a limitation. Practise writing all three answers in under five minutes; the third part is what separates a full-mark answer from an average one.

Applications of Filter (signal processing)

In research
Filter (signal processing) appears in science research whenever the underlying quantities have to be modelled precisely. Papers usually cite it as a starting assumption and then explore where it breaks down.
In technology and industry
Engineering practice reuses Filter (signal processing) in design rules, simulations and safety margins. Knowing the idea lets you read a specification sheet and understand why the numbers look the way they do.
In the classroom
Filter (signal processing) is common in secondary-school and first-year university syllabi. It links to neighbouring topics Filter theory, Signal processing, Telecommunication theory, so understanding it makes those chapters shorter.
In everyday life
Look for Filter (signal processing) outside the textbook — in sport, cooking, traffic, electronics or the sky above you. An example you found yourself is remembered far longer than one you were given.

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How to study Filter (signal processing) in 20 minutes

  1. Read the reference excerpt below once, without taking notes.
  2. Close the page and write down what Filter (signal processing) means in your own words.
  3. Compare your version with the excerpt and mark what you missed.
  4. Work through the three examples above with pen and paper.
  5. Explain Filter (signal processing) out loud to somebody else — or to Teacher Smith in the lgStudy chat.

Frequently asked questions

What is Filter (signal processing) in simple terms?

In signal processing, a filter is a device or process that removes some unwanted components or features from a signal. Filtering is a class of signal processing, the defining feature of filters being the complete or partial suppression of some aspect of the signal.

Why does Filter (signal processing) matter?

Because it connects several science ideas at once: it gives you a definition you can apply, a quantity you can calculate, and a way to check whether a result is plausible.

How should I study Filter (signal processing)?

Read the excerpt, restate it from memory, then work through the examples and applications listed on this page. The five-step study plan above takes about twenty minutes.

What does this page cover?

It gives you a compact reference excerpt plus original lgStudy explanations, examples, applications and study material on Filter (signal processing).

Tags

  • Filter theory
  • Signal processing
  • Telecommunication theory

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