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FILTER
Filter design is the process of designing a filter (in the sense in which the term is used in signal processing, statistics, and applied mathematics), often a linear shift-invariant filter, that satisfies a set of requirements, some of which are contradictory. The purpose is to find a realization of the filter that meets each of the requirements to a sufficient degree to make it useful. The filter design process can be described as an optimization problem where each requirement contributes with a term to an error function which should be minimized. Certain parts of the design process can be automated, but normally an experienced electrical engineer is needed to get a good result.

Typical design requirements

Typical requirements which are considered in the design process are:

The filter should have a specific frequency response The filter should have a specific phase shift or group delay The filter should have a specific impulse response The filter should be causal The filter should be stable The filter should be localized The computational complexity of the filter should be low The filter should be implemented in particular hardware or software

The frequency function

Typical examples of frequency function are:

A low-pass filter is used to cut unwanted high-frequency signals. A high-pass filter passes high frequencies fairly well; it is helpful as a filter to cut any unwanted low frequency components. A band-pass filter passes a limited range of frequencies. A band-stop filter passes frequencies above and below a certain range. A very narrow band-stop filter is known as a notch filter.

It must also be decided how the filter is going to be implemented:

Analog filter Analog sampled filter Digital filter Mechanical filter

Presentation by V.Prabakaran:CSE-IV yr

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Analog filter: Analog filters are a basic building block of signal processing much used in electronics. Amongst their many applications are the separation of an audio signal before application to bass, mid-range and tweeter loudspeakers; the combining and later separation of multiple telephone conversations onto a single channel; the selection of a chosen radio station in a radio receiver and rejection of others.

Passive linear electronic analogue filters are those filters which can be described with linear differential equations (linear); they are composed of capacitors, inductors and, sometimes, resistors (passive) and are designed to operate on continuously varying (analogue) signals. There are many linear filters which are not analogue in implementation (digital filter), and there are many electronic filters which may not have a passive topology both of which may have the same transfer function of the filters described in this article. Analogue filters are most often used in wave filtering applications, that is, where it is required to pass particular frequency components and to reject others from analogue (continuous-time) signals. Analogue filters have played an important part in the development of electronics. Especially in the field of telecommunications, filters have been of crucial importance in a number of technological breakthroughs and have been the source of enormous profits for telecommunications companies.

Ohm's model of the transmission line was simply resistance.

Digital filters There are two categories of digital filter: the recursive filter and the nonrecursive filter. These are often referred to as infinite impulse response (IIR) filters and finite impulse response (FIR) filters

Finite impulse response, or FIR, filters express each output sample as a weighted sum of the last N inputs, where N is the order of the filter. Since they do not use feedback, they are inherently stable. If the coefficients are symmetrical (the usual case), then such a filter is linear phase, so it delays signals of all frequencies equally. This is important in many applications. It is also straightforward to avoid overflow in an FIR filter. The main disadvantage is that they may require significantly more processing and memory resources than cleverly designed IIR variants. FIR filters are generally easier to design than IIR filters Presentation by V.Prabakaran:CSE-IV yr

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Infinite impulse response, or IIR, filters are the digital counterpart to analog filters. Such a filter contains internal state, and the output and the next internal state are determined by a linear combination of the previous inputs and outputs (in other words, they use feedback, which FIR filters normally do not). IIR filters normally require less computing resources than an FIR filter of similar performance. However, due to the feedback, high order IIR filters may have problems with instability, arithmetic overflow, and limit cycles, and require careful design to avoid such pitfalls. Digital filters may be more expensive than an equivalent analog filter due to their increased complexity, but they make practical many designs that are impractical or impossible as analog filters. Since digital filters use a sampling process and discrete-time processing, they experience latency (the difference in time between the input and the response), which is almost irrelevant in analog filters.

A general finite impulse response filter with n stages, each with an independent delay, di, and amplification gain

Analog sampled filter

An analog sampled filter an electronic filter that is a hybrid between an analog and a digital filter. The input signal is analog, and usually stored in capacitors. The time domain is discrete, however. Distinct analog samples are shifted through an array of holding capacitors as in a bucket brigade. Analog adders and amplifiers do the arithmetic in the signal domain, just as in an analog computer. Note that these filters are subject to aliasing phenomena just like a digital filter, and antialiasing filters will usually be required.

Mechanical filter
A mechanical filter is a signal processing filter usually used in place of an electronic filter at radio frequencies. Its purpose is the same as that of a normal electronic filter: to pass a range of signal frequencies, but to block others.
Presentation by V.Prabakaran:CSE-IV yr

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The filter acts on mechanical vibrations which are the analogue of the electrical signal. At the input and output of the filter, transducers convert the electrical signal into, and then back from, these mechanical vibrations. Steel and nickeliron alloys are common materials for mechanical filter components; nickel is sometimes used for the input and output couplings. Resonators in the filter made from these materials need to be machined to precisely adjust their resonance frequency before final assembly.

Mechanical filter transducers. a magnetostrictive transducer. b Langevin type piezoelectric transducer. c torsional piezoelectric transducer.

Adaptive filter
An adaptive filter is a filter that self-adjusts its transfer function according to an optimization algorithm driven by an error signal. Because of the complexity of the optimization algorithms, most adaptive filters are digital filters. By way of contrast, a non-adaptive filter has a static transfer function. Adaptive filters are required for some applications because some parameters of the desired processing operation (for instance, the locations of reflective surfaces in a reverberant space) are not known in advance. The adaptive filter uses feedback in the form of an error signal to refine its transfer function to match the changing parameters. Generally speaking, the adaptive process involves the use of a cost function, which is a criterion for optimum performance of the filter, to feed an algorithm, which determines how to modify filter transfer function to minimize the cost on the next iteration. As the power of digital signal processors has increased, adaptive filters have become much more common and are now routinely used in devices such as mobile phones and other communication devices, camcorders and digital cameras, and medical monitoring equipment.

Presentation by V.Prabakaran:CSE-IV yr

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Example application
Suppose a hospital is recording a heart beat (an ECG), which is being corrupted by a 50 Hz noise (the frequency coming from the power supply in many countries). However, due to slight variations in the power supply to the hospital, the noise signal may contain harmonics of the noise and the exact frequency of the noise may vary. One way to remove the noise is to filter the signal with a notch filter at 50 Hz. Such a static filter would need to remove all the frequencies in the vicinity of 50 Hz, which could excessively degrade the quality of the ECG since the heart beat would also likely have frequency components in the rejected range. To circumvent this potential loss of information, an adaptive filter could be used. The adaptive filter would take input both from the patient and from the power supply directly and would thus be able to track the actual frequency of the noise as it fluctuates. Such an adaptive technique generally allows for a filter with a smaller rejection range, which means, in our case, that the quality of the output signal is more accurate for medical diagnoses.

Block diagram
The block diagram, shown in the following figure, serves as a foundation for particular adaptive filter realisations, such as Least Mean Squares (LMS) and Recursive Least Squares (RLS). The idea behind the block diagram is that a variable filter extracts an estimate of the desired signal.

Applications of adaptive filters

Noise cancellation Signal prediction Adaptive feedback cancellation Echo cancellation

Presentation by V.Prabakaran:CSE-IV yr