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Topic: Filter signal processing

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Frequency spectrum

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	Digital signal processing - Wikipedia, the free encyclopedia
		DSP and analog signal processing are subfields of signal processing.
		A filter may also be described as a difference equation, a collection of zeroes and poles or, if it is an FIR filter, an impulse response or step response.
		Filters can also be represented by block diagrams which can then be used to derive a sample processing algorithm to implement the filter using hardware instructions.
	en.wikipedia.org /wiki/Digital_signal_processing (1333 words)

	Electronic filter - Wikipedia, the free encyclopedia
		A filter in which the signal passes through an inductor, or in which a capacitor provides a path to earth, therefore presents less attenuation to low-frequency signals than high-frequency signals and is a low-pass filter.
		Active filters are implemented using a combination of passive and active (amplifying) components.
		Electrical signals are converted to a mechanical wave in a piezoelectric crystal; this wave is delayed as it propagates across the crystal, before being converted back to an electrical signal by further electrodes.
	en.wikipedia.org /wiki/Filter_(signal_processing) (1031 words)

	Filter Design (Signal Processing Toolbox) (Site not responding. Last check: 2007-10-08)
		The filters are optimal in the sense that they minimize the maximum error between the desired frequency response and the actual frequency response; they are sometimes called minimax filters.
		Another possibility is a filter that has as a transition region the line connecting the passband with the stopband; this can help control "runaway" magnitude response in wide transition regions.
		Differentiation of a signal in the time domain is equivalent to multiplication of the signal's Fourier transform by an imaginary ramp function.
	caligari.dartmouth.edu /doc/matlab6.1/toolbox/signal/filterd9.html (985 words)

	Electronic filter -- Facts, Info, and Encyclopedia article (Site not responding. Last check: 2007-10-08)
		The oldest forms of electronic filters are passive analog linear filters, constructed using only (An electrical device that resists the flow of electrical current) resistors, (An electrical device characterized by its capacity to store an electric charge) capacitors and (An electrical device that introduces inductance into a circuit) inductors.
		The simplest electronic implementations of linear filters are based on combinations of (An electrical device that resists the flow of electrical current) resistors, (An electrical device that introduces inductance into a circuit) inductors and (An electrical device characterized by its capacity to store an electric charge) capacitors.
		A filter is said to have a high Q if it selects or rejects a narrow range of frequencies compared with the absolute frequency at which it operates.
	www.absoluteastronomy.com /encyclopedia/e/el/electronic_filter.htm (1381 words)

	Bandwidth - Wikipedia, the free encyclopedia (Site not responding. Last check: 2007-10-08)
		For instance, this signal would require a lowpass filter with cutoff frequency of at least W to stay intact.
		The bandwidth of an electronic filter is the part of the filter's frequency response that lies within 3 dB compared to the center frequency of its peak.
		In signal processing and control theory, the bandwidth is the frequency at which the closed-loop system gain drops to −3 dB.
	www.eastcleveland.us /project/wikipedia/index.php/Bandwidth (516 words)

	Signal Processing
		Users are able to sketch and resize their own filters or use previously generated filters and see how each filter affects the scaling of a subregion of an image.
		Two key properties that we rely on in signal processing are "linearity" and "spatial invariance".
		Two box functions are displayed on a signal graph and their area of overlap (which, for the box function, corresponds to the value of the convolution at a particular point) is shaded.
	www.cs.brown.edu /exploratories/freeSoftware/catalogs/signal_processing.html (722 words)

	Digital Signal Processing Filter Terminology
		Filter attenuation is the ratio, at a given frequency, of the signal amplitude at the output of the filter over the signal amplitude at the input of the filter, defined as
		For an FIR filter, the filter coefficients are, by definition, the impulse response of the filter.
		With a constant group delay of 0.04 seconds, the 1 Hz input sinewave is delayed at the filter output by 0.25 radians, the 3 Hz sinewave is delayed by 0.75 radians, the 5 Hz sinewave by 1.25 radians, and the 7 Hz sinewave by 1.75 radians.
	www.dspguru.com /info/terms/filtterm/index2.htm (2987 words)

	Filter article - Filter Filter (chemistry) Filter (mathematics) subset partially ordered - What-Means.com (Site not responding. Last check: 2007-10-08)
		Filter (signal processing) — a device or algorithm for signal processing (e.g.
		Filter (water) — a device to physically separate impurities from water
		Filter (part) — a device that provides a physical barrier to unwanted impuries (e.g.
	www.what-means.com /encyclopedia/Filter (187 words)

	Signal Processing Laboratory
		We believe that signal processing is not merely limited to the acquisition, filtering, prediction, enhancement, processing, analysis, compression, or display of data.
		We view Signal Processing as a key technology, the usefulness of which is best perceived when applied and put to use in concrete engineering or multidisciplinary problems.
		Signal processing is in itself multidisciplinary: it is closely related to information theory, control and system theory, probability and statistics, computational mathematics, numerical analysis, matrix analysis, and applied functional and harmonic analysis.
	www.ieeta.pt /spl (241 words)

	Wiener filter (Site not responding. Last check: 2007-10-08)
		The Wiener filter is a filter (signal processing) proposed by Norbert Wiener during the 1940s and published 1.
		By creating a filter that filters only on the frequency domain it is possible for the filter to pass noise.
		The input to the Wiener filter is assumed to be additive noise.
	read-and-go.hopto.org /Estimation/Wiener-filter.html (276 words)

	DSA-2000 Digital Spectrum Analyzer - CANBERRA, An AREVA Group Company
		Unlike conventional systems, which digitize the detector preamplifier signals at the end of the signal processing chain, the DSA-2000 digitizes the detector preamplifier signals at the front end of the signal processing chain.
		Signal filtering functions previously implemented in analog circuits are limited.
		Digital signal processing allows filter algorithms and pulse shapes that are not realizable using conventional analog processing techniques.
	www.canberra.com /products/636.asp (1221 words)

	Intro. to Signal Processing:Fourier filter (Site not responding. Last check: 2007-10-08)
		The Fourier filter is a type of filtering or smoothing function that is based on the frequency components of a signal.
		The assumption is made here that the frequency components of the signal fall predominantly at low frequencies and those of the noise fall predominantly at high frequencies.
		The signal at the top left seems to be only random noise, but its power spectrum (top right) shows that high-frequency components dominate the signal.
	www.wam.umd.edu /~toh/spectrum/FourierFilter.html (256 words)

	OTL - File # 3301
		The filter provides a region of silence, where there is no audio signal, without significantly affecting the audio signal in another region in any environment (e.g., rooms, cars, etc.).
		The method obtains a filter by: (i) measuring the room acoustical responses at multiple seating positions, and (ii) applying the measured responses to a Digital Signal Processing algorithm for computing the desired filter.
		The filter, once determined, does not need to be recomputed unless significant changes occur in the environment.
	www.usc.edu /academe/otl/3301.htm (174 words)

	LPBF-01G : Low Pass Bessel Filter Module
		The amplified signal is applied to the output filter.
		The output filter can also be bypassed by using a toggle switch.
		Filter and gain are monitored at BNC connectors (1V / step).
	www.alascience.com /products/module_lpbf01g.html (278 words)

	DIGITAL SIGNAL PROCESSING AND DIGITAL FILTER DESIGN
		When these filters are implemented on Digital Signal Processor with a special purpose-hardware, each filter coefficient has to be represented by a finite number of bits 'b' smaller than that used on a computer.
		The simplest and the most widely used approach to the problem is the rounding of the optimal infinite precision coefficients to its 'b' bits representation.
		However, the filters obtained are degraded and in most case there exists another set of finite word length coefficients which gives the best Chebyshev approximation to the desired frequency response 'HD(ejw)'.
	www.prip.tuwien.ac.at /~nabil/SignalProcessing.html (808 words)

	Elec 531: Digital Signal Processing (Site not responding. Last check: 2007-10-08)
		An undergraduate course in signals and systems is prerequisite, and the background of an undergraduate digital signal processing course is strongly recommended, although not required.
		This course focuses on the fundamentals of digital signal processing, and includes in-depth treatment of digital filter design and various FFT algorithms in the first 2/3 of the semester.
		The last 1/3 of the semester is dedicated to more advanced topics, where the students are introduced to multirate signal processing, filter bank theory, and adaptive filtering theory.
	www.utdallas.edu /~aria/courses/elec531 (159 words)

	Audio Signal Processing Basics
		There is a signal processing glossary on a page of its own.
		Conclusion: in audio applications, when the phase response isn't critical, it is often profitable to use IIR filters because of their efficiency.
		Therefore, when a whole regular sound signal is transformed, the changes in frequency content cannot be observed.
	www.cs.tut.fi /sgn/arg/intro/basics.html (1064 words)

	filter (Signal Processing Toolbox)
		Filter data with a recursive (IIR) or nonrecursive (FIR) filter.
		The filter realization is the transposed direct form II structure [1], which can handle both FIR and IIR filters.
		The input-output description of this filtering operation in the z-transform domain is a rational transfer function:
	cens.ioc.ee /local/man/matlab/toolbox/signal/filter.html (190 words)

	Educational Matlab GUIs
		The user is allowed to control the parameters of both the input sinusoid and the analog LTI filter.
		The user is allowed to control the parameters of both the input sinusoid and the digital filter.
		Signals can be dragged around with the mouse with results displayed in real-time.
	users.ece.gatech.edu /mcclella/matlabGUIs (509 words)

	Automatic Speech Recognition: Java Applets (Site not responding. Last check: 2007-10-08)
		Dynamic Time-Warping: learn how dynamic programming is applied to the problem of time-warping and comparing a speech signal to a recognition model.
		Provides for direct manipulation of the pole and zero locations, analysis of the system frequency and impulse responses, and a three dimensional view of the pole/zero interaction.
		Filter Design: a hands-on approach to exploring digital filter design from a parameter-driven perspective.
	www.cavs.msstate.edu /hse/ies/projects/speech/software/demonstrations/applets (222 words)

	Digital Signal Processing
		A particular strength is in digital filtering: DSPworks takes specified digital filters and allows them to be applied to generated or recorded signals.
		This is especially important in testing IIR filters, where the filter response in practice depends on the signals which are encountered and cannot be completely determined theoretically at the time of designing the filter.
		It is particularly well suited to testing of digital filters in simulation and on the target hardware.
	www.mds.com /Products/product.asp?prod=S030 (168 words)

	Geometry in Action: Signal Processing (Site not responding. Last check: 2007-10-08)
		Digital image compression and transmission is a problem that (with the growth of the world wide web) is rapidly growing in prominence, and may be a fertile source of links between geometry and signal processing.
		Designing two-dimensional filter banks based on geometric decompositions of the frequency domain, W.
		The basic idea is to solve various "advancing front" type problems such as finding shortest paths around obstacles, by evolving a surface in one higher dimension that describes the dynamics of the front.
	www.ics.uci.edu /~eppstein/gina/sigproc.html (234 words)

	Signal Processing Group
		This research aims at the design of optimal wavelets and filter banks for applications like image and audio compression, the design of low-delay filterbanks for real-time subband processing, and the design of modulated filterbanks with integer coefficients for efficient real-time implementations.
		Karp, A. Mertins, and G. Schuller, "Efficient biorthogonal cosine-modulated filter banks," Signal Processing, vol.
		Mertins, ``Subspace approach for the design of cosine-modulated filter banks with linear-phase prototype filter,'' IEEE Trans.
	www.uni-oldenburg.de /sigproc/research/wavelets-and-filterbanks.html (140 words)

	ARRLWeb: DSP - Digital Signal Processing
		Digital Signal Processing uses software techniques to accomplish many things that have traditionally been accomplished in hardware, plus a number of things that cannot easily be done using physical components.
		Once a signal is converted to digital form, DSP software can filter it, shape it and process it to remove noise.
		With Signals, Samples and Stuff: A DSP Tutorial -- Part 1 you will find a four part series on digital signal processing.
	www.arrl.org /tis/info/dsp.html (1176 words)

	Signal Processing Research: Greg Kochanski (Site not responding. Last check: 2007-10-08)
		However, applying the source-filter model is conceptually unsatisfactory, because one cannot uniquely reconstruct both the source signal and filter from a single audio signal.
		I have developed an algorithm for reconstructing an estimate of the source signal from sound propagated through the throat, using an array of external microphones.
		[ `A Quasi-Glottogram signal,' Kochanski, G. and Shih, Chilin] This signal can, in principle, provide a direct view of the source, without the complexities of the time-varying filter that is provided by the vocal tract.
	kochanski.org /gpk/research/qgg.html (202 words)

	Signal Processing Primer (Site not responding. Last check: 2007-10-08)
		Wet/Dry Mix: Most signal processing units or software have a 'wet/dry mix' to determine the proportion of mix between the original (dry) and 'effected' (wet) signals.
		Many composers chose to control their wet/dry mix from the mixing console and not the signal processing unit, leaving it set to 100% wet.
		May alter dynamic nature of input signal with high ratios--used to create more "punchy" sounds when applied to input with sharp transients, like a kick drum.
	www.indiana.edu /~emusic/effects.htm (425 words)

	Filter Design and Analysis Tool (Signal Processing Toolbox) (Site not responding. Last check: 2007-10-08)
		Note The Filter Design and Analysis Tool (FDATool) requires resolution greater than 640 x 480.
		The Filter Design and Analysis Tool (FDATool) is a powerful user interface for designing and analyzing filters.
		FDATool enables you to quickly design digital FIR or IIR filters by setting filter performance specifications, by importing filters from your MATLAB workspace, or by directly specifying filter coefficients.
	www.weizmann.ac.il /matlab/toolbox/signal/fdtool.html (187 words)

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