Digital signal processing

Digital signal processing (DSP) is the use of digital processing, such as by computers or more specialized
digital signal processors, to perform a wide variety of signal processing operations. The digital signals
processed in this manner are a sequence of numbers that represent samples of a continuous variable in a
domain such as time, space, or frequency. In digital electronics, a digital signal is represented as a pulse
train,[1][2] which is typically generated by the switching of a transistor.[3]

Digital signal processing and analog signal processing are subfields of signal processing. DSP applications
include audio and speech processing, sonar, radar and other sensor array processing, spectral density
estimation, statistical signal processing, digital image processing, data compression, video coding, audio
coding, image compression, signal processing for telecommunications, control systems, biomedical
engineering, and seismology, among others.

DSP can involve linear or nonlinear operations. Nonlinear signal processing is closely related to nonlinear
system identification[4] and can be implemented in the time, frequency, and spatio-temporal domains.

The application of digital computation to signal processing allows for many advantages over analog
processing in many applications, such as error detection and correction in transmission as well as data
compression.[5] Digital signal processing is also fundamental to digital technology, such as digital
telecommunication and wireless communications.[6] DSP is applicable to both streaming data and static
(stored) data.

Signal sampling
To digitally analyze and manipulate an analog signal, it must be digitized with an analog-to-digital
converter (ADC).[7] Sampling is usually carried out in two stages, discretization and quantization.
Discretization means that the signal is divided into equal intervals of time, and each interval is represented
by a single measurement of amplitude. Quantization means each amplitude measurement is approximated
by a value from a finite set. Rounding real numbers to integers is an example.

The Nyquist–Shannon sampling theorem states that a signal can be exactly reconstructed from its samples if
the sampling frequency is greater than twice the highest frequency component in the signal. In practice, the
sampling frequency is often significantly higher than this.[8] It is common to use an anti-aliasing filter to
limit the signal bandwidth to comply with the sampling theorem, however careful selection of this filter is
required because the reconstructed signal will be the filtered signal plus residual aliasing from imperfect
stop band rejection instead of the original (unfiltered) signal.

Theoretical DSP analyses and derivations are typically performed on discrete-time signal models with no
amplitude inaccuracies (quantization error), "created" by the abstract process of sampling. Numerical
methods require a quantized signal, such as those produced by an ADC. The processed result might be a
frequency spectrum or a set of statistics. But often it is another quantized signal that is converted back to
analog form by a digital-to-analog converter (DAC).

Domains
DSP engineers usually study digital signals in one of the following domains: time domain (one-dimensional
signals), spatial domain (multidimensional signals), frequency domain, and wavelet domains. They choose
the domain in which to process a signal by making an informed assumption (or by trying different
possibilities) as to which domain best represents the essential characteristics of the signal and the processing
to be applied to it. A sequence of samples from a measuring device produces a temporal or spatial domain
representation, whereas a discrete Fourier transform produces the frequency domain representation.

Time and space domains

Time domain refers to the analysis of signals with respect to time. Similarly, space domain refers to the
analysis of signals with respect to position, e.g., pixel location for the case of image processing.

The most common processing approach in the time or space domain is enhancement of the input signal
through a method called filtering. Digital filtering generally consists of some linear transformation of a
number of surrounding samples around the current sample of the input or output signal. The surrounding
samples may be identified with respect to time or space. The output of a linear digital filter to any given
input may be calculated by convolving the input signal with an impulse response.

Frequency domain

Signals are converted from time or space domain to the frequency domain usually through use of the
Fourier transform. The Fourier transform converts the time or space information to a magnitude and phase
component of each frequency. With some applications, how the phase varies with frequency can be a
significant consideration. Where phase is unimportant, often the Fourier transform is converted to the power
spectrum, which is the magnitude of each frequency component squared.

The most common purpose for analysis of signals in the frequency domain is analysis of signal properties.
The engineer can study the spectrum to determine which frequencies are present in the input signal and
which are missing. Frequency domain analysis is also called spectrum- or spectral analysis.

Filtering, particularly in non-realtime work can also be achieved in the frequency domain, applying the
filter and then converting back to the time domain. This can be an efficient implementation and can give
essentially any filter response including excellent approximations to brickwall filters.

There are some commonly used frequency domain transformations. For example, the cepstrum converts a
signal to the frequency domain through Fourier transform, takes the logarithm, then applies another Fourier
transform. This emphasizes the harmonic structure of the original spectrum.

Z-plane analysis

Digital filters come in both infinite impulse response (IIR) and finite impulse response (FIR) types. Whereas
FIR filters are always stable, IIR filters have feedback loops that may become unstable and oscillate. The Z-
transform provides a tool for analyzing stability issues of digital IIR filters. It is analogous to the Laplace
transform, which is used to design and analyze analog IIR filters.

Autoregression analysis
A signal is represented as linear combination of its previous samples. Coefficients of the combination are
called autoregression coefficients. This method has higher frequency resolution and can process shorter
signals compared to the Fourier transform.[9] Prony's method can be used to estimate phases, amplitudes,
initial phases and decays of the components of signal.[10][9] Components are assumed to be complex
decaying exponents.[10][9]

Time-frequency analysis

A time-frequency representation of signal can capture both temporal evolution and frequency structure of
analyzed signal. Temporal and frequency resolution are limited by the principle of uncertainty and the
tradeoff is adjusted by the width of analysis window. Linear techniques such as Short-time Fourier
transform, wavelet transform, filter bank,[11] non-linear (e.g., Wigner–Ville transform[10]) and
autoregressive methods (e.g. segmented Prony method)[10][12][13] are used for representation of signal on
the time-frequency plane. Non-linear and segmented Prony methods can provide higher resolution, but may
produce undesirable artifacts. Time-frequency analysis is usually used for analysis of non-stationary signals.
For example, methods of fundamental frequency estimation, such as RAPT and PEFAC[14] are based on
windowed spectral analysis.

Wavelet

In numerical analysis and functional analysis, a discrete

wavelet transform 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. The accuracy of
the joint time-frequency resolution is limited by the
uncertainty principle of time-frequency.

Empirical mode decomposition

Empirical mode decomposition is based on

decomposition signal into intrinsic mode functions
(IMFs). IMFs are quasiharmonical oscillations that are
extracted from the signal.[15]
An example of the 2D discrete wavelet transform
that is used in JPEG2000. The original image is
Implementation high-pass filtered, yielding the three large images,
each describing local changes in brightness
DSP algorithms may be run on general-purpose (details) in the original image. It is then low-pass
computers and digital signal processors. DSP filtered and downscaled, yielding an approximation
algorithms are also implemented on purpose-built image; this image is high-pass filtered to produce
hardware such as application-specific integrated circuit the three smaller detail images, and low-pass
(ASICs). Additional technologies for digital signal filtered to produce the final approximation image in
processing include more powerful general purpose the upper-left.
microprocessors, graphics processing units, field-
programmable gate arrays (FPGAs), digital signal
controllers (mostly for industrial applications such as motor control), and stream processors.[16]
For systems that do not have a real-time computing requirement and the signal data (either input or output)
exists in data files, processing may be done economically with a general-purpose computer. This is
essentially no different from any other data processing, except DSP mathematical techniques (such as the
DCT and FFT) are used, and the sampled data is usually assumed to be uniformly sampled in time or
space. An example of such an application is processing digital photographs with software such as
Photoshop.

When the application requirement is real-time, DSP is often implemented using specialized or dedicated
processors or microprocessors, sometimes using multiple processors or multiple processing cores. These
may process data using fixed-point arithmetic or floating point. For more demanding applications FPGAs
may be used.[17] For the most demanding applications or high-volume products, ASICs might be designed
specifically for the application.

Parallel implementations of DSP algorithms, utilising multi-core CPU and many-core GPU architectures,
are developed to improve the performances in terms of latency of these algorithms.[18]

Native processing is done by the computer's CPU rather than by DSP or outboard processing, which is
done by additional third-party DSP chips located on extension cards or external hardware boxes or racks.
Many digital audio workstations such as Logic Pro, Cubase, Digital Performer and Pro Tools LE use native
processing. Others, such as Pro Tools HD, Universal Audio's UAD-1 and TC Electronic's Powercore use
DSP processing.

Applications
General application areas for DSP include

Audio signal processing Data transmission

Audio data compression e.g. MP3 Radar
Video data compression Sonar
Computer graphics Financial signal processing
Digital image processing Economic forecasting
Photo manipulation Seismology
Speech processing Biomedicine
Speech recognition Weather forecasting
Specific examples include speech coding and transmission in digital mobile phones, room correction of
sound in hi-fi and sound reinforcement applications, analysis and control of industrial processes, medical
imaging such as CAT scans and MRI, audio crossovers and equalization, digital synthesizers, and audio
effects units.[19]

Techniques
Bilinear transform LTI system theory
Discrete Fourier transform Minimum phase
Discrete-time Fourier transform s-plane
Filter design Transfer function
Goertzel algorithm Z-transform
Least-squares spectral analysis

Related fields
Analog signal processing Fourier analysis
Automatic control Information theory
Computer engineering Machine learning
Computer science Real-time computing
Data compression Stream processing
Dataflow programming Telecommunication
Discrete cosine transform Time series
Electrical engineering Wavelet

Further reading
Ahmed, Nasir; Rao, Kamisetty Ramamohan (7 August 1975). Orthogonal Transforms for
Digital Signal Processing. New York: Springer-Verlag. doi:10.1109/ICASSP.1976.1170121
(https://doi.org/10.1109%2FICASSP.1976.1170121). ISBN 978-3540065562.
LCCN 73018912 (https://lccn.loc.gov/73018912). OCLC 438821458 (https://www.worldcat.or
g/oclc/438821458). OL 22806004M (https://openlibrary.org/books/OL22806004M).
S2CID 10776771 (https://api.semanticscholar.org/CorpusID:10776771).
Jonathan M. Blackledge, Martin Turner: Digital Signal Processing: Mathematical and
Computational Methods, Software Development and Applications, Horwood Publishing,
ISBN 1-898563-48-9
James D. Broesch: Digital Signal Processing Demystified, Newnes, ISBN 1-878707-16-7
Dyer, Stephen A.; Harms, Brian K. (13 August 1993). "Digital Signal Processing" (https://boo
ks.google.com/books?id=vL-bB7GALAwC&pg=PA104). In Yovits, Marshall C. (ed.).
Advances in Computers. Vol. 37. Academic Press. pp. 59–118. doi:10.1016/S0065-
2458(08)60403-9 (https://doi.org/10.1016%2FS0065-2458%2808%2960403-9). ISBN 978-
0120121373. ISSN 0065-2458 (https://www.worldcat.org/issn/0065-2458). LCCN 59015761
(https://lccn.loc.gov/59015761). OCLC 858439915 (https://www.worldcat.org/oclc/85843991
5). OL 10070096M (https://openlibrary.org/books/OL10070096M).
Paul M. Embree, Damon Danieli: C++ Algorithms for Digital Signal Processing, Prentice
Hall, ISBN 0-13-179144-3
Hari Krishna Garg: Digital Signal Processing Algorithms, CRC Press, ISBN 0-8493-7178-3
 P. Gaydecki: Foundations Of Digital Signal Processing: Theory, Algorithms And Hardware
Design, Institution of Electrical Engineers, ISBN 0-85296-431-5
Ashfaq Khan: Digital Signal Processing Fundamentals, Charles River Media, ISBN 1-
58450-281-9
Sen M. Kuo, Woon-Seng Gan: Digital Signal Processors: Architectures, Implementations,
and Applications, Prentice Hall, ISBN 0-13-035214-4
Paul A. Lynn, Wolfgang Fuerst: Introductory Digital Signal Processing with Computer
Applications, John Wiley & Sons, ISBN 0-471-97984-8
Richard G. Lyons: Understanding Digital Signal Processing, Prentice Hall, ISBN 0-13-
108989-7
Vijay Madisetti, Douglas B. Williams: The Digital Signal Processing Handbook, CRC Press,
ISBN 0-8493-8572-5
James H. McClellan, Ronald W. Schafer, Mark A. Yoder: Signal Processing First, Prentice
Hall, ISBN 0-13-090999-8
Bernard Mulgrew, Peter Grant, John Thompson: Digital Signal Processing – Concepts and
Applications, Palgrave Macmillan, ISBN 0-333-96356-3
Boaz Porat: A Course in Digital Signal Processing, Wiley, ISBN 0-471-14961-6
John G. Proakis, Dimitris Manolakis: Digital Signal Processing: Principles, Algorithms and
Applications, 4th ed, Pearson, April 2006, ISBN 978-0131873742
John G. Proakis: A Self-Study Guide for Digital Signal Processing, Prentice Hall, ISBN 0-13-
143239-7
Charles A. Schuler: Digital Signal Processing: A Hands-On Approach, McGraw-Hill, ISBN 0-
07-829744-3
Doug Smith: Digital Signal Processing Technology: Essentials of the Communications
Revolution, American Radio Relay League, ISBN 0-87259-819-5
Smith, Steven W. (2002). Digital Signal Processing: A Practical Guide for Engineers and
Scientists (http://www.dspguide.com). Newnes. ISBN 0-7506-7444-X.
Stein, Jonathan Yaakov (2000-10-09). Digital Signal Processing, a Computer Science
Perspective. Wiley. ISBN 0-471-29546-9.
Stergiopoulos, Stergios (2000). Advanced Signal Processing Handbook: Theory and
Implementation for Radar, Sonar, and Medical Imaging Real-Time Systems. CRC Press.
ISBN 0-8493-3691-0.
Van De Vegte, Joyce (2001). Fundamentals of Digital Signal Processing. Prentice Hall.
ISBN 0-13-016077-6.
Oppenheim, Alan V.; Schafer, Ronald W. (2001). Discrete-Time Signal Processing. Pearson.
ISBN 1-292-02572-7.
Hayes, Monson H. Statistical digital signal processing and modeling. John Wiley & Sons,
2009. (with MATLAB scripts (https://www.mathworks.com/matlabcentral/fileexchange/2183-s
tatistical-digital-signal-processing-and-modeling?s_tid=prof_contriblnk))

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