
	Where results make sense

Topic: Sampling (signal processing)

Related Topics

Digital signal processing

Signal processing

Signal (information theory)

Signal

Frequency spectrum

Discrete Fourier transform

Terminal equipment

Electronics

Data terminal equipment

Quantization

Channel coding

Computer

Speech recognition

Data communication equipment

Control theory

In the News (Sat 28 Nov 09)

Northern Trust

Marvin Harrison

Peter Chernin

Gary Locke

Match Play

Penelope Cruz

Mickey Rourke

AR Rahman

Danny Boyle

Hugh Jackman

	InformIT: Understanding Digital Signal Processing: Periodic Sampling > ALIASING: SIGNAL AMBIGUITY IN THE FREQUENCY ...
		Periodic sampling, the process of representing a continuous signal with a sequence of discrete data values, pervades the field of digital signal processing.
		There is a frequency-domain ambiguity associated with discrete-time signal samples that does not exist in the continuous signal world, and we can appreciate the effects of this uncertainty by understanding the sampled nature of discrete data.
		Figure 2-3 Shark's tooth pattern: (a) aliasing at multiples of the sampling frequency; (b) aliasing of the 7-kHz sinewave to 1 kHz, 13 kHz, and 19 kHz.
	www.informit.com /articles/article.aspx?p=345472 (0 words)

	The SpectrumWare Approach to Wireless Signal Processing
		However, given the rates at which the sampling and processing technologies are improving, we are convinced that the bandwidth and variety of computationally accessible signals, as well as the complexity of their processing, will expand.
		Temporal decoupling allows the samples to be processed in large blocks, and a whole new range of algorithms that leverage statistical behavior may be brought to bear on the problem space.
		Furthermore, this sample processing may be performed in bursts as dictated by the behavior of the application program and the time-slicing of the processor among applications.
	www.tns.lcs.mit.edu /publications/winet95/winet95.html (7237 words)

	Digital signal processing
		DSP and analog signal processing are subfields of signal processing.
		A sequence of samples from a measuring device produces a time or spatial domain representation, whereas a discrete Fourier transform produces the frequency domain information, that is the frequency spectrum.
		In the discretization stage, the space of signals is partioned into equivalence classes and discretization is carried out by replacing the signal with representative signal of the corresponding equivalence class.
	www.brainyencyclopedia.com /encyclopedia/d/di/digital_signal_processing.html (1450 words)

	Signal Processing
		Two key properties that we rely on in signal processing are "linearity" and "spatial invariance".
		The student discovers the basics of aliasing and is able to adjust a signal's frequency and the frequency at which it is sampled to build an intuitive understanding of all these concepts.
		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)

	Understanding Digital Signal Processing: Periodic Sampling > ALIASING: SIGNAL AMBIGUITY IN THE FREQUENCY DOMAIN
		Periodic sampling, the process of representing a continuous signal with a sequence of discrete data values, pervades the field of digital signal processing.
		There is a frequency-domain ambiguity associated with discrete-time signal samples that does not exist in the continuous signal world, and we can appreciate the effects of this uncertainty by understanding the sampled nature of discrete data.
		Figure 2-3 Shark's tooth pattern: (a) aliasing at multiples of the sampling frequency; (b) aliasing of the 7-kHz sinewave to 1 kHz, 13 kHz, and 19 kHz.
	www.phptr.com /articles/article.asp?p=345472&rl=1 (1579 words)

	A CMU Graphics Lab Project: Plenoptic Sampling
		From a spectral analysis of light field signals and using the sampling theorem, we mathematically derive the analytical functions to determine the minimum sampling rate for light field rendering.
		The minimum sampling rate for light field rendering is obtained by compacting the replicas of the spectral support of the sampled light field within the smallest interval.
		Note that sampling points in the figure have been chosen to be slightly above the minimum sampling curve due to quantization and also the number of images in the following figure means the number of sampling image along one direction.
	graphics.cs.cmu.edu /projects/plenoptic-sampling/ps_projectpage.htm (858 words)

	Communication & Signal Processing engineering - Sampling system SNR and SFDR
		Process Gain: My thought was it it is proportional to the oversampling the ADC is capable of.
		Processing gain in radar usually means using a complex outgoing signal (for example: a chirp, perhaps a Barker Code), and then decoding it during receive (for processing gain).
		Processing gain or spreading gain also builds SNR but after the processing so don't confuse that with the ADC effective bits for your application--they are important taking both into account only for a specific system--remember that all signals in the ADC bandwidth need to be accomodated without saturating the ADC.
	www.eng-tips.com /viewthread.cfm?qid=166202&page=1 (2067 words)

	Algebraic Theory of Signal Processing
		The algebraic signal processing theory is a new approach to and an extension of linear signal processing (henceforth called SP), that is, SP built around the concepts of filters, spectrum, Fourier transform, and others.
		The algebraic signal processing theory is a theory of linear signal processing.
		The theory is built on top of the concept of a signal model, which is defined as the triple (A, M, Phi), where A is a chosen algebra of filters, M an associated A-module of signals, and Phi a generalization of the z-transform.
	www.ece.cmu.edu /~smart/research.html (1683 words)

	Digital Alias-free Signal Processing
		DASP represents advanced DSP techniques for digital processing of signals in a considerably extended frequency range not necessarily limited by the used sampling rate as usual.
		Adapts the sampling and quantization operations to the specifics of the particular signal processing case
		Is using special signal processing algorithms matched to the specifics of the digitizing mode chosen for the particular signal processing case including algorithms suitable for processing non-equispaced signal sample value sequences
	www.edi.lv /dasp-web (0 words)

	Signal Processing in OSW
		However, it is common practice to assume the relationship between sample index n, time t, and sampling rate S and refer to the digital waveform as y(n) [Chaudhary 2001].
		OSW automatically determines the sample rate for a particular signal based on the input and/or output devices to which it is connected.
		Signals are usually transmitted in blocks (otherwise known as vectors) of 64 samples.
	osw.sourceforge.net /manual/signal_processing.htm (2864 words)

	RLE :: Digital Signal Processing Group
		The Digital Signal Processing Group in the MIT Research Laboratory of Electronics focuses on developing general methods for signal processing that can be applied to a wide range of applications.
		In the general area of signal processing algorithms, we have been investigating several areas of mathematics such as geometric algebras and frame theory with the goal of developing new classes of algorithms with broad application.
		An important trend in signal processing technology is the increasing deployment of distributed signal processing systems; i.e., systems in which signal measurement, storage and processing capacity are separated by a communication network.
	www.rle.mit.edu /dspg/research.html (883 words)

	Understanding Digital Signal Processing: Periodic Sampling > SAMPLING BANDPASS SIGNALS (Site not responding. Last check: )
		Conforming to the Nyquist criterion (sampling at twice the highest frequency content of the signal) implies that the sampling frequency must be a minimum of 45 MHz.
		The processes of sampling and frequency translation are intimately bound together in the world of digital signal processing, and every sampling operation inherently results in spectral replications.
		For example, in the m = 1 range of permissible sampling frequencies, it is much easier to perform subsequent digital filtering or other processing on the signal samples whose spectrum is that of Figure 2-9(b), as opposed to the spectrum in Figure 2-9(a).
	www.phptr.com /articles/article.asp?p=345472&seqNum=3 (2331 words)

	An Introduction to Sampling Theory (Site not responding. Last check: )
		When the continuous analog signal is sampled at a frequency F, the resulting discrete signal has more frequency components than did the analog signal.
		The Sampling Theorem states that a signal can be exactly reproduced if it is sampled at a frequency F, where F is greater than twice the maximum frequency in the signal.
		Aliasing is the presence of unwanted components in the reconstructed signal.
	www2.egr.uh.edu /~glover/applets/Sampling/Sampling.html (926 words)

	Glossary of Vibration and Shock terms
		In a sampled data system, the analog input must be sampled at a rate at least twice the bandwidth of the signal to avoid loss of data (Nyquist Theorem).
		A current loop created when a signal source and a signal measurement device are grounded at two separate points on a ground bus through which noise currents flow.
		In the output signal of a device, distortion caused by the presence of frequencies not present in the input signal.
	www.vibrationandshock.com /glossary.htm (7123 words)

	IEEE Signal Processing Magazine
		Sensors, signal processing hardware, and algorithms are under increasing pressure to accommodate ever larger and higher-dimensional data sets; ever faster capture, sampling, and processing rates; ever lower power consumption; communication over ever more difficult channels; and radically new sensing modalities.
		The traditional approach to digital signal processing is to sample first (at at least twice the signal bandwidth) and then ask questions (process) later.
		Compressive sampling fuses the theories of approximation, compression, inverse problems, and random matrices and is being developed by a multidisciplinary community of researchers from signal processing, applied mathematics, and computer science.
	apollo.ee.columbia.edu /spm/?i=cfp/Mar08 (483 words)

	Signal Processing in Processing: Sampling and Quantization
		Any signal, in order to be processed by numerical computing devices, have to be reduced to a sequence of discrete samples, and each sample must be represented using a finite number of bits.
		Sampling is, for one-dimensional signals, the operation that transforms a continuous-time signal (such as, for instance, the air pressure fluctuation at the entrance of the ear canal) into a discrete-time signal, that is a sequence of numbers.
		In that case, the sampling rate is given by the frame rate, or number of pictures per second, and has to be related with the spinning velocity of the wheels.
	cnx.org /content/m13045/latest (2160 words)

	DSP Courses
		Introduction to signal processing for discrete-time and continuous-time signals.
		An introduction to advanced signal processing methods that are used in a variety of application areas.
		Techniques for signal and state estimation in the presence of measurement and process noise with the emphasis on Wiener and Kalman filtering.
	csip.ece.gatech.edu /Classes/index.shtml (443 words)

	Home Theater Glossary
		The routing of delayed, processor-modified or processor-created ambience signals, in addition to those already on a recording (even one that is monophonic), to surround speakers to simulate a concert-hall effect.
		In audio, the result of an analog signal's being overdriven to the extent that its peak levels cannot be accommodated, and therefore are "clipped" off from the audible signal.
		The circuit that routes the proper electrical signals (highs, midrange, bass) to the various drivers in a loudspeaker system (if it is a passive design) and to the various amplifiers in a bitmapped system (if it is an active design).
	www.yanman.com /HomeTheater/HTGlossary.htm (5452 words)

	Mathtools.net : Java/Signal processing
		This is a powerful Java applet that is used to produce different digital signals, such as sine, square, trigon, saw, white noise, beat wave or sweep wave.
		A Java applet for signal sampling at various sampling frequencies, and signal reconstruction from samples using various low-pass filter cutoff frequencies.
		You can sketch one period of a period N = 5 discrete-time signal with the mouse and observe the magnitude and phase spectra, and the behavior of the corresponding complex frequency components.
	www.mathtools.net /Java/Signal_processing/index.html (568 words)

	Analog Signal Processing
		The input signal (which comes from a CCD camera) is converted to a digital representation and stored in a video memory bank.
		The processed images are then placed in an output memory bank and now the result is converted back to analog (NTSC) for display.
		A double correlated sampling technique is used to achieve high power supply rejection, low dc offset and low 1/f noise.
	amsc.tamu.edu /signal.htm (799 words)

	Upsampling vs. Oversampling for Digital Audio — Audioholics Home Theater Reviews and News
		In one case we are sampling at some frequency, f s and in the other case we are at twice the same sampling rate.
		Oversampling is performed at the ADC and then the signal is sent to the digital processing chain that does the filtering and any DSP operations.
		Without increasing the sample rate, we would need to design a very sharp filter that would have to cutoff at just past 20kHz and be 80-100dB down at 22kHz.
	www.audioholics.com /techtips/specsformats/upsamplingvsoversampling1.php (1391 words)

	upfirdn (Signal Processing Toolbox)
		FIR filtering the upsampled signal data with the impulse response sequence given in the vector or matrix
		However, when these arguments are arrays, each column is treated as a separate signal or filter.
		This example implements a seven-channel filter bank by convolving seven different filters with one input signal, then downsamples by five.
	www.cs.berkeley.edu /titan/sww/software/matlab/toolbox/signal/upfirdn.html (521 words)

	Part I: Fourier Transforms and Sampling
		An example of a 8 Hz tone sampled at 20 Hz [45kb] shows that the sample values fluctuate with a period of 4 Hz.
		There is exactly the same kind of relationship for sampled signals; the integral in equation (1b) is replaced by a summation, and the continuous times and frequencies are replaced by discrete values.
		The convolution theorem also proves that a signal that is finite in time has an infinite spectrum: the response can always be expressed as a convolution with a sinc function, which extends the spectrum to infinity.
	www.silcom.com /~aludwig/Signal_processing/Signal_processing.htm (2926 words)

	DAQ/DIO/Adapter-PCI + PCMCIA + PXI + CompactPCI + MOXA + MAGMA
		The input signals are mechanically connected to the SAK-Board by BNC connectors, Lemo connectors or screw terminals.
		By this lot of data processing can be moved from pc system to the DAQ board, this process relieves the host computer and gives the chance for realtime reactions with guaranteed latency times.
		Depending on the measurement signal and the sampling rate of the data acquisition board SAK-Amplifiers with a smaller signal bandwidth are available.
	www.hacker-technology.com /15080.html?sessionidkey=sessionidval (1400 words)

	Signal Processing
		AFTER you have sampled the data and converted it from a continuous analog trace to a series of numbers, it is usually not possible to reverse the process entirely.
		If the process is stationary, the value of the cross-correlation should depend only on time delay, not on the precise values of the two times.
		Thus, for example, the signals from two hot-film sensors placed very close together in the flow may be expected to be virtually identical, except for a factor accounting for the difference in the calibration constants of the two sensors.
	www.adl.gatech.edu /classes/flodiag/dsp/b01.html (3127 words)

	Signal Processing \| Software Radio - Signal Processing - Digital Signal Processing
		The important question is how to sample and reconstruct an analog signal while preserving the full information of the original....
		Developments in digital signal processing and data conversion are enabling the commercialization of software-based wideband receiver, thereby reducing the cost, size, complexity and power consumption of a base station.
		Sampling is the first step in the process of converting a continuous analog signal to a sequence of digital numbers.
	www.rfdesign.com /software_radio/signal_processing (774 words)

	Frequency Response & Sampling Rate: Digital Signal Processing
		For an analog instrument designed to measure changing input signals, a key parameter is frequency response, or bandpass, stated in hertz (Hz).
		For a digital instrument used in the same application, the key parameter is the sampling rate, which is the number of samples that can be acquired in a specified period of time, typically one second.
		Sampling rate is sometimes specified in terms of frequency response (Hz), but while the two are related, they are not the same, and should not be used interchangeably.
	www.vishay.com /brands/measurements_group/guide/ta/dsp/dspp.htm (125 words)

Try your search on: Qwika (all wikis)