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Digital Signal Processing

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1 Signal sampling

conducted on analog signals by digital means (as opposed

to Analog signal processing where the signal processing

is is carried out by an analog process). It consist in the

mathematical manipulation of an information signal to

modify or improve it in some way. It is characterized by

the representation of discrete time, discrete frequency, or

other discrete domain signals by a sequence of numbers

or symbols and the processing of these signals.

operations. Nonlinear signal processing is closely related

to nonlinear system identication[3] and can be implemented in the time, frequency, and spatio-temporal domains.

choose the domain in which to process a signal by making an informed assumption (or by trying dierent possibilities) as to which domain best represents the essential characteristics of the signal. A sequence of samples

from a measuring device produces a temporal or spatial

domain representation, whereas a discrete Fourier transform produces the frequency domain information, that is,

The increasing use of computers has resulted in the increased use of, and need for, digital signal processing. To

digitally analyze and manipulate an analog signal, it must

be digitized with an analog-to-digital converter. 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 quanThe goal of DSP is usually to measure, lter and/or com- tization is carried out by replacing the signal with reppress continuous real-world analog signals. Usually, the resentative signal of the corresponding equivalence class.

rst step is conversion of the signal from an analog to a In the quantization stage, the representative signal values

digital form, by sampling and then digitizing it using an are approximated by values from a nite set.

analog-to-digital converter (ADC), which turns the ana- The NyquistShannon sampling theorem states that a siglog signal into a stream of discrete digital values. Often, nal can be exactly reconstructed from its samples if the

however, the required output signal is also analog, which sampling frequency is greater than twice the highest frerequires a digital-to-analog converter (DAC). Even if this quency of the signal, but this requires an innite number

process is more complex than analog processing and has of samples. In practice, the sampling frequency is often

a discrete value range, the application of computational signicantly higher than twice that required by the sigpower to signal processing allows for many advantages nals limited bandwidth.

over analog processing in many applications, such as error

detection and correction in transmission as well as data Some (continuous-time) periodic signals become nonperiodic after sampling, and some non-periodic signals

compression.[1]

become periodic after sampling. In general, for a periDigital signal processing and analog signal processing are odic signal with period T to be periodic (with period N)

subelds of signal processing. DSP applications include after sampling with sampling interval Ts, the following

audio and speech signal processing, sonar and radar sig- must be satised:

nal processing, sensor array processing, spectral estimation, statistical signal processing, digital image processing, signal processing for communications, control of systems, biomedical signal processing, seismic data process- Ts N = T k

ing, among others. DSP algorithms have long been run

where k is an integer.[4]

on standard computers, as well as on specialized processors called digital signal processors, and on purposebuilt hardware such as application-specic integrated circuit (ASICs). Currently, there are additional technologies 2 DSP domains

used for digital signal processing including more powerful general purpose microprocessors, eld-programmable In DSP, engineers usually study digital signals in

gate arrays (FPGAs), digital signal controllers (mostly for one of the following domains: time domain (oneindustrial applications such as motor control), and stream dimensional signals), spatial domain (multidimensional

processors, among others.[2]

signals), frequency domain, and wavelet domains. They

2 DSP DOMAINS

cross-correlation of the signal with itself over varying intervals of time or space.

Main article: Frequency domain

2.1

frequency domain usually through the Fourier transform.

The Fourier transform converts the signal information to

Main article: Time domain

a magnitude and phase component of each frequency.

Often the Fourier transform is converted to the power

The most common processing approach in the time or

spectrum, which is the magnitude of each frequency comspace domain is enhancement of the input signal through

ponent squared.

a method called ltering. Digital ltering generally consists of some linear transformation of a number of sur- The most common purpose for analysis of signals in the

rounding samples around the current sample of the input frequency domain is analysis of signal properties. The

or output signal. There are various ways to characterize engineer can study the spectrum to determine which frequencies are present in the input signal and which are

lters; for example:

missing.

A linear lter is a linear transformation of input

samples; other lters are non-linear. Linear lters

satisfy the superposition condition, i.e. if an input

is a weighted linear combination of dierent signals,

the output is an equally weighted linear combination

of the corresponding output signals.

is often needed. This can be obtained from the Fourier

transform. With some applications, how the phase varies

with frequency can be a signicant consideration.

achieved by converting to the frequency domain, applying the lter and then converting back to the time domain.

A causal lter uses only previous samples of the This is a fast, O(n log n) operation, and can give esseninput or output signals; while a non-causal lter tially any lter shape including excellent approximations

uses future input samples. A non-causal lter can to brickwall lters.

usually be changed into a causal lter by adding a There are some commonly used frequency domain transdelay to it.

formations. For example, the cepstrum converts a signal

to the frequency domain through Fourier transform, takes

A time-invariant lter has constant properties over the logarithm, then applies another Fourier transform.

time; other lters such as adaptive lters change in This emphasizes the harmonic structure of the original

spectrum.

time.

Frequency domain analysis is also called spectrum- or

A stable lter produces an output that converges spectral analysis.

to a constant value with time, or remains bounded

within a nite interval. An unstable lter can

produce an output that grows without bounds, with

bounded or even zero input.

A nite impulse response (FIR) lter uses only the 2.3 Z-plane analysis

input signals, while an innite impulse response

lter (IIR) uses both the input signal and previous

Main article: Z-transform

samples of the output signal. FIR lters are always

stable, while IIR lters may be unstable.

Whereas analog lters are usually analyzed in terms of

transfer functions in the s plane using Laplace transforms,

A lter can be represented by a block diagram, which can digital lters are analyzed in the z plane in terms of Zthen be used to derive a sample processing algorithm to transforms. A digital lter may be described in the z

implement the lter with hardware instructions. A lter plane by its characteristic collection of zeroes and poles.

may also be described as a dierence equation, a collec- The z plane provides a means for mapping digital fretion of zeroes and poles or, if it is an FIR lter, an impulse quency (samples/second) to real and imaginary z comresponse or step response.

ponents, where z = rej for continuous periodic signals

The output of a linear digital lter to any given input and = 2F ( F is the digital frequency). This is useful

may be calculated by convolving the input signal with the for providing a visualization of the frequency response of

impulse response.

a digital system or signal.

2.4

Wavelet

4 Implementation

Depending on the requirements of the application, digital

In numerical analysis and functional analysis, a discrete signal processing tasks can be implemented on general

purpose computers.

Often when the processing requirement is not real-time,

processing is economically done with an existing generalpurpose computer and the signal data (either input or output) exists in data les. This is essentially no dierent

from any other data processing, except DSP mathematical techniques (such as the FFT) are used, and the sampled data is usually assumed to be uniformly sampled

in time or space. For example: processing digital photographs with software such as Photoshop.

in JPEG2000. The original image is high-pass ltered, yielding

the three large images, each describing local changes in brightness (details) in the original image. It is then low-pass ltered

and downscaled, yielding an approximation image; this image is

high-pass ltered to produce the three smaller detail images, and

low-pass ltered to produce the nal approximation image in the

upper-left.

DSP is often implemented using specialized microprocessors such as the DSP56000, the TMS320, or the

SHARC. These often process data using xed-point

arithmetic, though some more powerful versions use

oating point. For faster applications FPGAs[5] might be

used. Beginning in 2007, multicore implementations of

DSPs have started to emerge from companies including

Freescale and Stream Processors, Inc. For faster applications with vast usage, ASICs might be designed specifically. For slow applications, a traditional slower processor such as a microcontroller may be adequate. Also

a growing number of DSP applications are now being

implemented on embedded systems using powerful PCs

with multi-core processors.

5 Techniques

wavelet transform (DWT) is any wavelet transform for

which the wavelets are discretely sampled. As with other

wavelet transforms, a key advantage it has over Fourier

transforms is temporal resolution: it captures both frequency and location information (location in time).

Bilinear transform

Discrete Fourier transform

Discrete-time Fourier transform

Filter design

Applications

Minimum phase

The main applications of DSP are audio signal processing, audio compression, digital image processing,

video compression, speech processing, speech recognition, digital communications, radar, sonar, Financial

signal processing, seismology and biomedicine. Specic examples are speech compression and transmission in digital mobile phones, room correction of sound

in hi- and sound reinforcement applications, weather

forecasting, economic forecasting, seismic data processing, analysis and control of industrial processes,

medical imaging such as CAT scans and MRI, MP3 compression, computer graphics, image manipulation, hi-

loudspeaker crossovers and equalization, and audio effects for use with electric guitar ampliers.

Transfer function

Z-transform

Goertzel algorithm

s-plane

6 Related elds

Analog signal processing

Automatic control

8 FURTHER READING

Computer Engineering

Computer Science

Data compression

Dataow programming

Electrical engineering

Fourier Analysis

Information theory

Machine Learning

Real-time computing

Stream processing

Telecommunication

Time series

Wavelet

Sen M. Kuo, Woon-Seng Gan: Digital Signal Processors: Architectures, Implementations, and Applications, Prentice Hall, ISBN 0-13-035214-4

Bernard Mulgrew, Peter Grant, John Thompson:

Digital Signal Processing - Concepts and Applications, Palgrave Macmillan, ISBN 0-333-96356-3

Steven W. Smith (2002). Digital Signal Processing: A Practical Guide for Engineers and Scientists.

Newnes. ISBN 0-7506-7444-X.

Paul A. Lynn, Wolfgang Fuerst: Introductory Digital

Signal Processing with Computer Applications, John

Wiley & Sons, ISBN 0-471-97984-8

James D. Broesch: Digital Signal Processing Demystied, Newnes, ISBN 1-878707-16-7

John G. Proakis, Dimitris Manolakis: Digital Signal Processing: Principles, Algorithms and Applications, 4th ed, Pearson, April 2006, ISBN 9780131873742

Hari Krishna Garg: Digital Signal Processing Algorithms, CRC Press, ISBN 0-8493-7178-3

References

Digital Signal Processing: Instant access. ButterworthHeinemann. p. 3.

P. Gaydecki: Foundations Of Digital Signal Processing: Theory, Algorithms And Hardware Design,

Institution of Electrical Engineers, ISBN 0-85296431-5

Processing and Applications (2nd ed. ed.). Elsevier. ISBN

0-7506-6344-8.

for Digital Signal Processing, Prentice Hall, ISBN 013-179144-3

[3] Billings S.A. Nonlinear System Identication: NARMAX Methods in the Time, Frequency, and SpatioTemporal Domains. Wiley, 2013

Signal Processing Handbook, CRC Press, ISBN 08493-8572-5

[4] Oppenheim, Alan V.; Schafer, Ronald W.; Buck, John R.,

Discrete-Time Signal Processing (3rd ed.), p. 15

[5] JpFix. FPGA-Based Image Processing Accelerator.

Retrieved 2008-05-10.

Handbook: Theory and Implementation for Radar,

Sonar, and Medical Imaging Real-Time Systems,

CRC Press, ISBN 0-8493-3691-0

Joyce Van De Vegte: Fundamentals of Digital Signal

Processing, Prentice Hall, ISBN 0-13-016077-6

Further reading

Alan V. Oppenheim, Ronald W. Schafer, John R.

Buck : Discrete-Time Signal Processing, Prentice

Hall, ISBN 0-13-754920-2

Boaz Porat: A Course in Digital Signal Processing,

Wiley, ISBN 0-471-14961-6

Ashfaq Khan: Digital Signal Processing Fundamentals, Charles River Media, ISBN 1-58450-281-9

Jonathan M. Blackledge, Martin Turner: Digital Signal Processing: Mathematical and Computational

Methods, Software Development and Applications,

Horwood Publishing, ISBN 1-898563-48-9

Processing, Prentice Hall, ISBN 0-13-108989-7

Essentials of the Communications Revolution, American Radio Relay League, ISBN 0-87259-819-5

Computer Science Perspective, Wiley, ISBN 0-47129546-9

Hands-On Approach, McGraw-Hill, ISBN 0-07829744-3

5

James H. McClellan, Ronald W. Schafer, Mark A.

Yoder: Signal Processing First, Prentice Hall, ISBN

0-13-090999-8

John G. Proakis: A Self-Study Guide for Digital Signal Processing, Prentice Hall, ISBN 0-13-143239-7

N. Ahmed and K.R. Rao (1975). Orthogonal

Transforms for Digital Signal Processing. SpringerVerlag (Berlin Heidelberg New York), ISBN 3540-06556-3.

9.1

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