Guided by:Dr. Sanjay Upadhyay Asst.

Professor IIT Roorkee

Made by:Faraz Ansari Mtech. Iyr Prod. & Ind. Systems Engg.

Digital Signals
Are

discrete time signals. Are not continuous function of time. Exists at only discrete times.

Filters

Filters shape the frequency spectrum of a signal. Filters generally do not add frequency components to a signal that are not there to begin with.

They boost or attenuate selected frequency regions.

The range of frequencies passed by a filter is known as pass band and the range not passed as stop band. The boundary between stopping and passing is known as cut-off frequency.

Analog v/s Digital filters

Analog filters Electronic components are cheap. Large dynamic range in amplitude and frequency. Real-time. Low stability of resistors, capacitors and inductors due to temperature Difficult to get the components accuracy as calculated by the formula. Digital filters

Better performance than analog filters. Are programmable. The characteristics are predictable. Filter design software packages can accurately evaluate the performance of a filter Alternative digital designs are available by tools to adapt the filter to the application.

Function of digital filter

Digital filters are used for two tasks
Separation

of different frequency components in signals if contaminated by

Noise Interference Other signals

Restoration

some ways

of signals that have been distorted in

Improvement & correction of an audio signal recording which is distorted by poor equipment. Deblurring of an image from improperly focused lenses

Classification of filter
Filters are classified according to the frequency ranges they transmit or reject. They are classified as:i. ii. iii. iv.

Low pass filter. High pass filter. Band pass filter. Band stop filter.

 Low

pass filter- Has a pass band that allows all frequencies from zero upto some frequency to be transmitted. These are generally used as most of the useful information being transmitted is low frequency. Since noise tend to occur at high frequency.

 High

pass filter Has a pass band which allows all frequencies from some value up to infinity to be transmitted.

 Band

pass filter- It allows all the frequencies within a specified band to be transmitted.

 Stop

band filter- It stops all frequencies with a particular band from being transmitted.

Implementation of Digital filter

Impulse
An

impulse is a very short pulsea waveform that has significant amplitude only for a very short time. filters, we use a one-sample pulse, or unit impulse.
response of the filter to the unit impulse is the filters Impulse Response (IR).

For

The

Implementation of Digital filter

Implementation of Digital Filter is achieved in two ways
By

convolution (FIR- finite impulse response filter)

In these each sample in the output is calculated by weighting the samples in the input and adding them together.

By

recursion(IIR- Infinite Impulse response filter)

These are extension of convolution by using previously calculated values from the output, besides the point from the input.

Mathematics Behind

Notations
x-

is the input signal y- represents the output signal n- represents the sample index.

e.g.:- x[0] represents first sample of input, y[0] represents first sample of output Similarly, x[n] represents current input sample x[n-1] represents previous input sample

FIR Maths

Simple average filter Output = half of current input + half of previous input y[n] = (0.5 x[n]) + (0.5 x[n-1]) Simple difference filter Output = half of current input - half of previous input y[n] = (0.5 x[n]) - (0.5 x[n-1])

But filters generally use more than one sample, with independently dependent coefficients. y[n] = (a0 x[n]) (a1 x[n-1]) (ai x[n-i])

The order of the filter is equal to the number of samples looked back. Generally, the higher the orderthe more samples are looked back to take an average or differencethe more attenuation of frequencies

IIR Maths
Since

based on recursive approach, the introduction of feedback loop creates possibility of an infinite impulse. Hence even a simple averaging filter transforms to an Exponential time averaging filter, equivalent to an infinitely long FIR. y[n] = (0.5 x[n]) + (0.5 y[n-1]) y[n] = (1/2 x[n]) + (1/4 x[n-1]) + (1/8 x[n-2]).

Complicated filters
More

complicated filters can generally be built by combinations of FIR and IIR filters.

FIR contributes the notches in the frequency spectrum While IIR contributes for the peaks.

Conclusion
Digital Their

filters are an important part of DSP.

extraordinary performance is one keys that have made DSP popular. filters are reprogrammable as per requirements.

Digital

More

filters.

precise than analog component based

References
Bolton,

W., Mechatronics, edition 3, Pearson Publications Ltd.

Smith,

S.W., The scientists and engineers guide to digital signal processing, Edition 2, California technical Publishing Ltd. Material.

Internet