Paper3

Abstract

Digital Signal Processing

Authors

Raghava Joshi

Email-id:- raghu_joshi1987@yahoo.com prathimahesh

Phone no: 9885013498 ph no:9985625814.

This paper deals with various types of digital signal processing and their uses. With the advent of
digital computers, signal processing had made rapid advances .The phenomenal growth of digital
signal processing (DSP) has been attributed to the availability of digital signal processor in late
seventies or early eighties.

This has opened up possibility of replacing analog filters, correlators , spectrum analyzers by
versatile but inexpensive DSP based equipment .The theory of DSP has been a step ahead of
hardware developments. We have new signal processing algorithms

For example singular value decomposition (SVD) based algorithms waiting for faster but
inexpensive digital signal processor for their exploitation . Digital signal processor

Technology has finally blossomed .Powerful methods of signal processing are used to detect and
analyse signals, make transform of different kinds and observe changes in real time with the help
of these DSP device based circuitry.

The digital signal processing deals with three main features

They are discrete signals and systems, digital filtering , spectrum analysis . Discrete signal and
system is about signal analysis and modeling of systems the study of discrete signals and systems
is also interest in communication , control system, circuit theory. A discrete signal is often derived
by sampling continuous signal and digital signal is derived by quantizing the discrete signal thus
sampling rate and quantization step size are of immediate concern

The digital filter like analog filter are primarily used to enhance the signal in presence of noise.
The digital filters are used for interpolation, extrapolation, equalization detection.

Spectrum deals with study of energy/power distribution as a function of frequency (energy for
deterministic signal and power for stochastic signal).

The spectrum of signal is essential when we are looking for discrete sinusoids in a signal, as in
communication , radar and sonar and also in optimum filters.

INTRODUCTION

The development of digital signal processing dates from the 1960's with the use of mainframe
digital computers for number-crunching applications such as the Fast Fourier Transform (FFT),
which allows the frequency spectrum of a signal to be computed rapidly. These techniques were
not widely used at that time, because suitable computing equipment was generally available only
in universities and other scientific research institutions. DSP, or Digital Signal Processing, as the
term suggests, is the processing of signals by digital means. A signal here means an electrical
signal carried by a wire or telephone line, or perhaps by a radio wave. More generally, however, a
signal is a stream of information representing anything from stock prices to data from a remote-
sensing satellite. The term "digital" comes from "digit", meaning a number, so "digital" literally
means numerical; the French word for digital is numerique . A digital signal consists of a stream of
numbers, usually in binary form. The processing of a digital signal is done by performing
numerical calculations. The signal is often strongly affected by "mains pickup" due to electrical
interference from the mains supply. Processing the signal using a filter circuit can remove or at
least reduce the unwanted part of the signal. The filtering of signals to improve signal quality or to
extract important information is done by DSP techniques rather than by analog electronics.

Digital Signal Processor chips - specialized microprocessors with architectures designed

specifically for the types of operations required in digital signal processing. DSP chips are capable
of carrying out millions of floating point operations per second, and like their better-known
general-purpose cousins, faster and more powerful versions are continually being introduced.
DSPs can also be embedded within complex "system-on-chip" devices, often containing both
analog and digital circuitry. DSP technology is nowadays commonplace in such devices as mobile
phones, multimedia computers, video recorders, C D players, hard disc drive controllers and
modems, and will soon replace analog circuitry in TV sets and telephones AnJ important
application of DSP is in signal compression and decompression. Signal compression is used in
digital cellular phones to allow a greater number of calls to be handled simultaneously within each
local "cell".

DSP signal compression technology allows people not only to talk to one another but also to see
one another on their computer screens, using small video cameras mounted on the computer
monitors, with only a conventional telephone line linking them together. In audio C D systems, DSP
technology is used to perform complex error detection and correction on the raw data as it is read
from the C D. The architecture of a DSP chip is designed to carry out such operations incredibly
fast, processing hundreds of millions of samples every second, to provide real-time performance:
that is, the ability to process a signal "live" as it is sampled and then output the processed signal,
for example to a loudspeaker or video display. All of the practical examples of DSP applications
mentioned earlier, such as hard disc drives and mobile phones, demand real-time operation.

Classification of Signals

• C ontinuous and discrete time signals

• C ontinuous and discrete amplitude signals

• Deterministic and random signals

• Digital and analog signals

• Multichannel and multidimensional signals

Continuous time signal : This signal can be defined at any time instant .The exponential
function and sinusoidal function are the examples of continuous time signals.

Discrete time signal : This signal is defined only at sampling instants. These signals are
basically represented as array of sample values

Continuous amplitude signals : The amplitude variation is continuous in such signals .Note that
the continuous amplitude signals can be discrete or continuous in time.

Discrete amplitude signals: These signals take only discrete amplitude levels.Here

Note that the discrete amplitude signals can be continuous or discrete in time.

Digital signals: The signals which are discrete in time as well as amplitude are called digital
signals .All the signal representation in computers and digital signal processors use digital signals.
The digital signal can be binary (one bit), octal(3 bit)

Hex(4bit) ,16 bit ,32 bit or even 64 bit. The complete amplitude range of the analog signal is
represented by these bit lengths. If the analog symbol has the amplitude range of 16 volts peak,
then each level will be of one volt.

Analog signals : The signals which are continuous in time as well as amplitude are called analog
signals. For example , the exponential function and sinusoidal function.

Deterministic signal : A signal which is completely described by the mathematical model is

called deterministic signal. The value of the deterministic signal can be evaluated at any time
without uncertainty. For example , the sinusoidal signal.

x (t) =Acos(wt)

Random signal : The signals which cannot be described by the mathematical model are called
random signals. For example the noise signal or speech signals are random signals. The random
signals can be described with the help of their statistical properties.

Multichannel signals : When different signals are recorded from the same source they are
called multichannel signals. For example, EC G signal can be recorded in 3 leads or 12 leads for
the same person. This results in 3 channel or 12 channel EC G

Signal. The multichannel signals are useful in studying correlation properties of the source.

Multidimensional signal : When the amplitude of the signal depends upon two or more
independent variables, it is called multidimensional signal. For example , the intensity or
brightness at any point in the picture or image is the function of its x and y position . Hence it
becomes two dimensional signal. The intensity of any point on the TV screen is the function of its x
and y position as well as time. Hence it becomes three dimensional signal.

Discrete time System

Previously we discussed about time discrete signals and their classification . Now we discuss about
discrete time systems. The discrete time systems is a device or algorithm that performs some
prescribed operation on the discrete time signal. Thus the discrete time system has an input or
excitation and the output or response. As shown in figure y(n) is response to the excitation x(n).
The input output relation ship

For a discrete time system is represented as,

y(n)=T [x(n)]

or

x(n) y(n)

Discrete time System

Here, T represents transformation operation. This transformation operation depends upon the
characteristics of the discrete time system.

DIGITAL FILTERING

In signal processing, the function of a filter is to remove unwanted parts of the signal, such as
random noise, or to extract useful parts of the signal, such as the components lying within a
certain frequency range.

The following block diagram illustrates the basic idea.

raw (unfiltered) signal -> FILTER -> filtered signal

There are two main kinds of filter, analog and digital

An analog filter uses analog electronic circuits made up from components such as resistors,
capacitors and op amps to produce the required filtering effect. These filter circuits are widely
used in such applications as noise reduction, video signal enhancement, graphic equalisers in hi-fi
systems, and many other areas. At all stages, the signal being filtered is an electrical voltage or
current which is the direct analogue of the physical quantity involved. A digital filter uses a digital
processor to perform numerical calculations on sampled values of the signal. The processor may
be a general-purpose computer such as a PC , or a specialised DSP (Digital Signal Processor) chip.

The analog input signal must first be sampled and digitised using an ADC (analog to digital
converter). The resulting binary numbers, representing successive sampled values of the input
signal, are transferred to the processor, which carries out numerical calculations on them. These
calculations typically involve multiplying the input values by constants and adding the products
together. If necessary, the results of these calculations, which now represent sampled values of
the filtered signal, are output through a DAC (digital to analog converter) to convert the signal
back to analog form.

the signal is represented by a sequence of numbers, rather than a voltage or current.

The following diagram shows the basic setup of such a system.

Digital filters can achieve virtually any filtering effect that can be expressed as a mathematical
function or algorithm. Digital filters process digitized or sampled signals. They perform an
extended sequence of multiplications and additions carried out at a uniformly spaced sample
interval. These signals are passed through structures that shift the clocked data into adders, delay
blocks, and multipliers. These structures change the mathematical values in a predetermined way.
The resulting data represents the filtered or transformed signal. Distortion and noise can be
introduced into digital filters simply by the conversion of analog signals into digital data, the digital
filtering process itself, and conversion of processed data back into analog. When fixed-point
processing is used, additional noise and distortion may be added during the filtering process
because the filter consists of large numbers of multiplications and additions that produce errors,
creating truncation noise. Increasing the bit resolution beyond 16 b reduces this filter noise.

Any digital filtering means that accepts as its input a set of one or more digital signals from which
it generates as its output a second set of digital signals .While being strictly correct, but it does
demonstrate the possible extent of application of digital-filter concepts and terminology. Digital
filters can be used in any signal-manipulating application where analog or continuous filters can be
used. They can be used in exacting applications where analog filters fail because of time- or other
parameter-dependent coefficient drift in continuous systems. Because of the ease and precision of
setting the filter coefficients, adaptive and learning digital filters are comparatively simple and
particularly effective to implement. As digital technology becomes more ubiquitous, digital filters
are increasingly acknowledged as the most versatile and cost-effective solutions to filtering
problems.

The number of functions that can be performed by a digital filter far exceeds that which can be
performed by an analog, or continuous, filter. By controlling the accuracy of the calculations within
the filter (that is, the arithmetic word length), it is possible to produce filters whose performance
comes arbitrarily close to the performance expected of the perfect models. For example,
theoretical designs that require perfect cancellation can be implemented with great fidelity by
digital filters. Filters are signal conditioners. Each functions by accepting an input signal, blocking
prespecified frequency components, and passing the original signal minus those components to
the output. For example, a typical phone line acts as a filter that limits frequencies to a range
considerably smaller than the range of frequencies human beings can hear. If the digital filter
under consideration is not a linear, time-invariant filter, the transfer function cannot be used.
There are many filter types, but the most common are lowpass, highpass, bandpass, and
bandstop. A lowpass filter allows only low frequency signals (below some specified cutoff) through
to its output, so it can be used to eliminate high frequencies. A lowpass filter is handy, in that
regard, for limiting the uppermost range of frequencies in an audio signal; it's the type of filter
that a phone line resembles. A highpass filter does just the opposite, by rejecting only frequency
components below some threshold.

Finite Impulse Response (FIR)

A finite impulse response (FIR) filter is a filter structure that can be used to implement almost
any sort of frequency response digitally. An FIR filter is usually implemented by using a series of
delays, multipliers, and adders to create the filter's output. Figure shows the basic block diagram
for an FIR filter of length N . The delays result in operating on prior input samples. The h k values
are the coefficients used for multiplication, so that the output at time n is the summation of all the
delayed samples multiplied by the appropriate coefficients.

The logical structure of an FIR filter

Advantages of digital filters

Digital filters can easily realize performance characteristics far beyond what are implementable
with analog filters. It is not particularly difficult, for example, to create a 1000 Hz low-pass filter
which can achieve near-perfect transmission of a 999 Hz input while entirely blocking a 1001 Hz
signal. Analog filters cannot discriminate between such closely spaced signals.

Also, for complex multi-stage filtering operations, digital filters have the potential to attain much
better signal to noise ratio than analog filters. This is because whereas at each intermediate stage
the analog filter adds more noise to the signal, the digital filter performs noiseless mathematical
operations at each intermediate step in the transform. The primary source of noise in a digital
filter is to be found in the initial ADC –analog to digital conversion step, where in addition to any
circuit noise introduced, the signal is subject to an unavoidable quantization error which is due to
the finite resolution of the digital representation of the signal.

Note also that frequency components exceeding half the sampling rate of the filter (Nyquist
sampling) will be confounded (or aliased) by the filter. Thus a small anti-aliasing filter is always
placed ahead of the analog to digital conversion circuitry to prevent these high-frequency
components from aliasing.

Discrete signal or discrete-time signal is a time series, perhaps a signal that has been
sampled from a continuous time-signal. Unlike a continuous-time signal, a discrete-time signal is
not a function of a continuous-time argument, but is a sequence of quantities; that is, a function
over a domain of discrete integers. Each value in the sequence is called a sample.

When a discrete-time signal is a sequence corresponding to uniformly spaced times, it has an

associated sampling rate; the sampling rate is not apparent in the data sequence, so may be
associated as a separate data item.

The discrete time signal are obtained by

time sampling of continuous time signals.
Hence the discrete time signals are
defined only at sampling instants. Let us
consider the exponential signal
Digital signals
A digital signal is discrete-time signal that takes on only a discrete set of values. It typically
derives from a discrete signal that has been quantized..

C ommon practical digital signals are represented as 8-bit (256 levels), 16-bit (65,536 levels), 32-
bit (4.3 billion levels), and so on, though any number of quantization levels is possible, not just
powers of two.

A continuous signal or a continuous-time signal is a varying quantity (a signal) that is

expressed as a function of a real-valued domain, usually time. The function of time need not be
continuous.

The signal is defined over a domain, which may or may not be finite, and there is a functional
mapping from the domain to the value of the signal. The continuity of the time variable, in
connection with the law of density of real numbers, means that the signal value can be found at
any arbitrary point in time.

A typical example of an infinite duration signal is:

A finite duration counterpart of the above signal could be:

and f ( t ) = 0 otherwise.

The value of a finite (or infinite) duration signal may or may not be finite. For example,

and f ( t ) = 0 otherwise,

is a finite duration signal but it takes an infinite value for

In many disciplines, the convention is that a continuous signal must always have a finite value,
which makes more sense in the case of physical signals.

For some purposes, infinite singularities are acceptable as long as the signal is integrable over any
finite interval (for example, the t - 1 signal is not integrable, but t - 2 is).

Any analogue signal is continuous by nature. Discrete signals, used in digital signal processing,
can be obtained by sampling and quantization of continuous signals.

C ontinuous signal may also be defined over an independent variable other than time. Another
very common independent variable is space and is particularly useful in image processing, where
two space dimensions are used.

Spectrum analysis

A spectrum analyzer is a device used to examine the spectral composition of some electrical,
acoustic, or optical waveform

It measures the power spectrum. There are analog and digital spectrum analyzers:

• An analog spectrum analyzer uses either a variable band pass filter whose mid-frequency is
automatically tuned (shifted, swept) through the range of frequencies of which the spectrum is to
be measured or a super-heterodyne receiver where the local oscillator is swept through a range of
frequencies.

• A digital spectrum analyzer computes the Fast Fourier transform (FFT), a mathematical process
that transforms a waveform into the components of its frequency spectrum.

Spectrum Analyzers require high dynamic range in order to capture bandwidths over wide input
frequency ranges. High-speed and high performance ADC s offer the speed and signal-to-noise
required for accurate measurement of signals and distortion. Highly accurate clock and DDS
products provide sampling clocks and sweep tuning of the Spectrum Analyzer, and amplifiers help
to drive the ADC s and increase signal levels in the down-conversion chains. Analog Devices has
all the key components for your next generation design.

The spectrum analyzer shows baseband

spectra of analog waveforms . The
baseband analyzer is integrated with the
instruments it serves, so there are no
additional settings to adjust.
Conclusion

Digital signal processing applications are so diverse that they make it necessary to have a number
of implementation alternatives. These are summarized in table 1. C learly, no one solution is best
in all cases. The challenge for the system implementers is to choose

With the increasing use of computers the usage and need of digital signal processing has
increased. In order to use an analog signal on a computer it must be digitized with an analog to
digital converter (ADC ). Sampling is usually carried out in two stages, discretization and
quantization. In the discretization stage, the space of signals is partitioned into equivalence classes
and discretization is carried out by replacing the signal with representative signal of the
corresponding equivalence class. In the quantization stage the representative signal values are
approximated by values from a finite set. the solution that best meets their system and market
requirements.

Real time signal processing is taking the digital revolution to the next step, making equipment that
is more personal, more powerful, and more interconnected than most people ever imagined
possible. Over the years, different technologies have powered the most innovative creations from
the mainframe and minicomputer eras to the PC and today's Internet era. C onsumers are driving
real time functionality, demanding equipment that is extremely fast, portable, and flexible. To
meet those needs, designers are facing more pressures than ever, but they also have more
options than ever to address them. areful evaluation of each option clearly shows several viable
alternatives for embedded applications. For implementing today's real time signal processing
applications, however, DSP is very often the best choice. No digital technology has more strengths
than DSP nor better meets the stringent criteria of today's developer.

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