Showing papers on "Sampling (signal processing) published in 2000"

Plenoptic sampling

Abstract: This paper studies the problem of plenoptic sampling in image-based rendering (IBR). 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 spectral support of a light field signal is bounded by the minimum and maximum depths only, no matter how complicated the spectral support might be because of depth variations in the scene. 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. Given the minimum and maximum depths, a reconstruction filter with an optimal and constant depth can be designed to achieve anti-aliased light field rendering.Plenoptic sampling goes beyond the minimum number of images needed for anti-aliased light field rendering. More significantly, it utilizes the scene depth information to determine the minimum sampling curve in the joint image and geometry space. The minimum sampling curve quantitatively describes the relationship among three key elements in IBR systems: scene complexity (geometrical and textural information), the number of image samples, and the output resolution. Therefore, plenoptic sampling bridges the gap between image-based rendering and traditional geometry-based rendering. Experimental results demonstrate the effectiveness of our approach.

Filterbank reconstruction of bandlimited signals from nonuniform and generalized samples

Abstract: This paper introduces a filterbank interpretation of various sampling strategies, which leads to efficient interpolation and reconstruction methods An identity, which is referred to as the interpolation identity, is developed and is used to obtain particularly efficient discrete-time systems for interpolation of generalized samples as well as a class of nonuniform samples, to uniform Nyquist samples, either for further processing in that form or for conversion to continuous time The interpolation identity also leads to new sampling strategies including an extension of Papoulis' (1977) generalized sampling expansion

A 90-dB SNR 2.5-MHz output-rate ADC using cascaded multibit delta-sigma modulation at 8/spl times/ oversampling ratio

Abstract: A 16-b 2.5-MHz output-rate analog-to-digital converter (ADC) for wireline communications and high-speed instrumentation has been developed. A 2-1-1 cascaded delta-sigma modulator (DSM) employing 4-b quantizers in every stage makes all quantization noise sources negligible at 8/spl times/ oversampling ratio, Data weighted averaging with bi-directional rotation eliminates tones generated by multibit digital-to-analog converter (DAC) nonlinearity to increase the spurious-free dynamic-range (SFDR). Switched-capacitor design techniques using low-threshold transistors reduce front-end sampling distortion. The 24.8 mm/sup 2/ chip in 0.5-/spl mu/m CMOS also integrates the decimation filter and voltage reference. The ADC achieves 90-dB signal-to-noise ratio (SNR) in the 1.25-MHz bandwidth and 102-dB SFDR with 270-mW power dissipation.

Principles of fractional delay filters

Abstract: In numerous applications, such as communications, audio and music technology, speech coding and synthesis, antenna and transducer arrays, and time delay estimation, not only the sampling frequency but the actual sampling instants are of crucial importance. Digital fractional delay (FD) filters provide a useful building block that can be used for fine-tuning the sampling instants, i.e., implement the required bandlimited interpolation. In this paper an overview of design techniques and applications is given.

Digital PWM controller and current estimator for a low-power switching converter

Abstract: This paper describes design and implementation of a digital controller for an experimental low-power converter in a battery-powered system with power management. Multiple operating modes are used to maintain high efficiency over wide ranges of input voltages and loads. A current estimation technique to perform load-dependent mode switching is proposed and tested. A digital PID regulator design example is described, with emphasis on practical limitations imposed by the fixed-point arithmetic and the delay due to sampling and processing.

Multiple frequency fluorescence lifetime imaging microscopy.

Abstract: The experimental configuration and the computational algorithms for performing multiple frequency fluorescence lifetime imaging microscopy (mfFLIM) are described. The mfFLIM experimental set-up enables the simultaneous homodyne detection of fluorescence emission modulated at a set of harmonic frequencies. This was achieved in practice by using monochromatic laser light as an excitation source modulated at a harmonic set of frequencies. A minimum of four frequencies were obtained by the use of two standing wave acousto-optic modulators placed in series. Homodyne detection at each of these frequencies was performed simultaneously by mixing with matching harmonics present in the gain characteristics of a microchannel plate (MCP) image intensifier. These harmonics arise as a natural consequence of applying a high frequency sinusoidal voltage to the photocathode of the device, which switches the flow of photoelectrons 'on' and 'off' as the sinus voltage swings from negative to positive. By changing the bias of the sinus it was possible to control the duration of the 'on' state of the intensifier relative to its 'off' state, enabling the amplitude of the higher harmonic content in the gain to be controlled. Relative modulation depths of 400% are theoretically possible from this form of square-pulse modulation. A phase-dependent integrated image is formed by the sum of the mixed frequencies on the phosphor of the MCP. Sampling this signal over a full period of the fundamental harmonic enables each harmonic to be resolved, provided that the Nyquist sampling criterion is satisfied for the highest harmonic component in the signal. At each frequency both the phase and modulation parameters can be estimated from a Fourier analysis of the data. These parameters enable the fractional populations and fluorescence lifetimes of individual components of a complex fluorescence decay to be resolved on a pixel-by-pixel basis using a non-linear fit to the dispersion relationships. The fitting algorithms were tested on a simulated data set and were successful in disentangling two populations having 1 ns and 4 ns fluorescence lifetimes. Spatial invariance of the lifetimes was exploited to improve the accuracy significantly. Multiple frequency fluorescence lifetime imaging microscopy was then successfully applied to resolve the fluorescence lifetimes and fluorescence intensity contributions in a rhodamine dye mixture in solution, and green fluorescent protein variants co-expressed in live cells.

Multichannel cochlear implant with neural response telemetry

Abstract: A circuit and method for cochlear implant telemetry where digital data is encoded into an RF signal. The RF signal is applied via a rectifier diode to a first switch matrix S1 and a second switch matrix S2, with S1 being coupled to a first sampling capacitor C1 and S2 being coupled to a second sampling capacitor C2. A local oscillator signal with period T is applied that controls S1 and S2, cyclically coupling C1 and C2 to the RF signal, a first input to a comparator, and ground. The comparator compares the first input to a DC reference voltage. The output of the comparator is then sampled via a flip flop clocked by the local oscillator, with the flip flop outputting a data bit stream representative of the envelope of the RF signal having encoded information.

Sampling of the diffraction field.

Abstract: When optical signals, like diffraction patterns, are processed by digital means the choice of sampling density and geometry is important during analog-to-digital conversion. Continuous band-limited signals can be sampled and recovered from their samples in accord with the Nyquist sampling criteria. The specific form of the convolution kernel that describes the Fresnel diffraction allows another, alternative, full-reconstruction procedure of an object from the samples of its diffraction pattern when the object is space limited. This alternative procedure is applicable and yields full reconstruction even when the diffraction pattern is undersampled and the Nyquist criteria are severely violated. Application of the new procedure to practical diffraction-related phenomena, like in-line holography, improves the processing efficiency without creating any associated artifacts on the reconstructed-object pattern.

Method and apparatus for RF common-mode noise rejection in a DSL receiver

Abstract: A receiver (50) for high-speed data communications, which receives a differential signal through a pair of signal lines (20, 22). The receiver includes a common-mode choke (52), which has first and second signal windings (24, 26), which are respectively coupled in series to the pair of signal lines so as to attenuate common-mode interference in the differential signal. The choke also has a sampling winding (54), which is inductively coupled to the signal windings so as to generate a sampled signal responsive to current flowing in the signal windings. Signal processing circuitry (62) is coupled to receive the sampled signal from the sampling winding and to receive the differential signal from the signal windings and to process the differential signal responsive to the sampled signal.

Method and apparatus for anti-aliasing in a graphics system

Abstract: PROBLEM TO BE SOLVED: To accomplish an efficient full scene anti-aliasing. SOLUTION: This graphics system can separately select three sampling positions in each super sampling pixel about a 2×2 pixel group (quadruplet). A 12-bit multi-sample application range mask decides which is enabled among twelve samples in the pixel quadruplet. The super sampling pixel is filtered during copyout operation from a local memory to an external buffer memory by using a pixel blend filter for connecting seven samples (three samples from present pixels and two samples from pixels just over and under the present pixels, respectively) from the three vertically displaced pixels. A weighted average is calculated on the basis of the enabled sample in order to decide the last color of the pixels.

An efficient algorithm for sample rate conversion from CD to DAT

Abstract: This letter unveils an efficient algorithm for sampling rate conversion (SRC) technique from 44.1 kHz compact disc (CD) to 48 kHz digital audio tape (DAT). This method involves upsampling the input signal by two, and then passing the interpolated signal through a fractional delay filter that employs a simple decimation. This method can also be used for SRC from DAT to CD without changing the filter coefficients. The proposed algorithm is simulated in Matlab and can be implemented in a realtime digital signal processor (DSP). Compared with other existing methods, the proposed method has the advantage that it requires fewer million instructions per second (MIPS) and memory.

Signal-to-noise ratio and spatial resolution in x-ray electronic imagers: is the MTF a relevant parameter?

Abstract: In most imaging detectors, the modulation transfer function (MTF) is regarded as a good parameter to describe spatial resolution. This is undoubtedly valid for visual observation. However, the detectability of a detail is essentially a matter of signal-to-noise ratio, which is not accounted for by the MTF. In x-ray imaging, signal-to-noise ratio in the image is generally limited by incident photons statistics, often larger than readout noises. Therefore, the MTF of the detector applies to both signal and noise, and does not impair the image content. Contrast can easily be restored by image processing without altering the signal-to-noise ratio. However, a number of effects may alter very differently noise and signal: (i) If the MTF significantly extends beyond half the sampling frequency, the aliasing introduced by spatial sampling can severely enhance the noise and cancel the benefit of the good signal transfer. This is illustrated by synthetic images which simulate the response of imagers with different MTFs to the same test pattern in the presence of quantum noise. (ii) Parallax and blurring by the x-ray spot size or motion are shown to degrade the transfer properties of signal, but do not affect the quantum noise; they must be treated separately. Contrary to the x-ray converter MTF, parallax directly impacts the detective quantum efficiency (DQE). Finally, it is shown that only the detective quantum efficiency can reliably describe the spatial resolution of an x-ray imaging detector in the presence of noise, parallax and blurring.

Series connected multi-stage linear FET amplifier circuit

Abstract: A sampling circuit is arranged so that source voltages VDD and VEE, which are to be applied to two inverters at the latter stages in a signal path on the p-channel transistor side, are shifted to the positive side with respect to source voltages VCC and VSS that are applied to the other inverters. With such a power supply construction, video signals on the low-potential side in a video signal line are picked up by the n-channel transistor and video signals on the high-potential side are picked up by the p-channel transistor, and the resulting signals are supplied to a data signal line. This arrangement makes it possible to reduce the gate input voltage upon conduction of the sampling switch. Moreover, by shifting the levels of the source voltages as described above, it becomes possible to ensure writing and holding operations even in the case of having signals with a small amplitude. Therefore, even in the case when devices having low withstand voltage are used, no damage is caused on the circuit characteristics.

Modulated filter banks with arbitrary system delay: efficient implementations and the time-varying case

Abstract: We present a new method for the design and implementation of modulated filter banks with perfect reconstruction. It is based on the decomposition of the analysis and synthesis polyphase matrices into a product of two different types of simple matrices, replacing the polyphase filtering part in a modulated filter bank. Special consideration is given to cosine-modulated as well as time-varying filter banks. The new structure provides several advantages. First of all, it allows an easy control of the input-output system delay, which can be chosen in single steps of input sampling rate, independent of the filter length. This property can be used in audio coding applications to reduce pre-echoes. Second, it results in a structure that is nearly twice as efficient as performing the polyphase filtering directly. Perfect reconstruction is a structurally inherent feature of the new formulation, even for nonlinear operations or time-varying coefficients. Hence, the structure is especially suited for the design of time-varying filter banks where both the number of bands as well as the prototype filters can be changed while maintaining perfect reconstruction and critical sampling. Further, a proof of effective completeness is given, and the design of equal magnitude-response analysis and synthesis filter banks is described. Filter design can be performed by nonconstrained optimization of the matrix coefficients according to a given cost function. Design and audio-coding application examples are given to show the performance of the new filter bank.

Bioelectrical impedance measuring device and body composition measuring apparatus

Abstract: There are provided a bioelectrical impedance measuring device which calculates bioelectrical impedance parameter values by use of an AD converter incorporated in a low-cost general-purpose microcontroller and a body composition measuring apparatus using the device. The bioelectrical impedance measuring device measures voltages generated in a living body according to alternating currents of predetermined frequencies applied to the living body and comprises digital data acquiring means for acquiring digital data by sampling the measurement signals of the voltages by sampling frequencies which are not higher than the Nyquist frequencies and calculation means for calculating bioelectrical impedance parameter values based on the digital data. Thus, since high-speed processing is not needed at the time of conversion to the digital data, sampling can be processed by the AD converter in the low-cost general-purpose microcontroller, thereby making cost reduction possible.

A sampled amplitude read channel employing a post processor with a boundary error compensator which compensates for boundary error events in a split-field data sector

Abstract: A sampled amplitude read channel is disclosed for reading a data sector recorded on a disk storage medium by detecting an estimated data sequence from a sequence of read signal sample values generated by sampling an analog read signal emanating from a read head positioned over the disk storage medium, the data sector comprising a user data segment and an embedded field comprising a known data segment. A sampling device samples the analog read signal to generate the read signal sample values and a trellis sequence detector detects a preliminary sequence from the read signal sample values. A post processor detects and corrects errors in the preliminary sequence using a syndrome generator responsive to the preliminary sequence representing the user data segment. The syndrome generator generates an error syndrome according to a predetermined error detection code. A boundary error compensator compensates for a boundary error event spanning the user data segment and the known data segment. A guided error connector, responsive to the error syndrome, corrects errors detected in the preliminary sequence.

Method and measurement configuration for measuring the characteristics of radio channels

Abstract: A method for measuring characteristics of radio channels, in which the signals are received by a total of M 1 receiving sensors in a linear antenna array, in which case the respective received signals are composed of wave elements of a transmitted signal with a different incidence direction and different delay, the transmitted signal contains a preselected test sequence, and the incidence angle as well as the delay of the dominant wave fronts are estimated simultaneously, taking account of the chip signal form of the test sequence, the signals from each antenna sensor are demodulated, and sampling is then carried out with M c samples per chip in the test sequence. In this case, the resultant N c ×M c samples are transformed to the frequency domain by discrete Fourier transformation, and the values which are obtained from this are corrected taking account the spectrum of the chip signal form, in which case the line vectors formed from the values obtained for each sensor can be combined to form a data matrix X F′ (n), which contains invariances which characterize the channel.

An efficient frequency-determination algorithm from multiple undersampled waveforms

Abstract: Frequency estimation/determination has applications in various areas, where the sampling rate is usually above the Nyquist rate. In some applications, it is preferred that the range of the frequencies is as large as possible for a given sampling rate and in some applications, the sampling rate is below the Nyquist rate. In both eases, frequency estimation from undersampled waveforms is needed. In this letter, we present an efficient algorithm to determine multiple frequencies from multiple undersampled waveforms with sampling rates below the Nyquist rates.

Nyquist response restoring delta-sigma modulator based analog to digital and digital to analog conversion

Abstract: A high pass delta-sigma modulator apparatus is disclosed for use in analog-to-digital and digital-to-analog converters. The apparatus includes a signal input, a differentiator component, a signal transducer producing an output signal, an inverse signal transducer producing an input compatible feedback signal, and a feedback loop coupling a feedback signal back to said differentiator component. Also disclosed is a Nyquist response restoring delta-sigma modulator based analog-to-digital and digital-to-analog converter apparatus having a first modulator set, a first scaling component, a second modulator set and a post processing combiner. In the various embodiments, the modulator sets can include an analog signal input, a digital signal input, a quantizer, a digital resolution rounder, a digital resolution truncator, a digital-to-analog signal transducer, a digital resolution expander, a delta sigma modulator structure, a feedback structure, an analog signal output, and one or more digital signal outputs.

Circuit breaker with integrated self-test enhancements

Abstract: A circuit switching device or circuit breaker with integrated self-test enhancements is disclosed having separable contacts operable under processor control to control power to a circuit responsive to at least one of a plurality of fault conditions and operable according to a method for testing, including the steps of: controlling the switching device during a sampling cycle, to input one or more operating parameters sensed in the circuit to an A/D converter for measurement wherein the operating parameters enable detection of the fault conditions; determining whether to read a select one of the operating parameters from an output of the A/D converter into a first memory; and reading pre-determined parameter values from a second memory into the first memory during the sampling cycle instead of the operating parameters read from the A/D converter if a self-test has been invoked during the sampling cycle.

Method of scanning indicia using selective sampling

Abstract: A method of reading an indicia, for example a bar code, comprising areas of differing light reflectivity comprises producing an electrical signal representative of light reflected from the indicia and selectively sampling the signal at a rate well below the Nyquist rate. The sample points are not uniform, and are chosen to represent key features in the signal. An attempt is then made to decode the indicia, using the samples. If the decode is unsuccessful, the samples which have been collected are analyzed to provide information on the extent to which they represent true data corresponding to the indicia, and the extent to which they present noise. Signal collection and/or processing and/or sampling is then adjusted and the process is repeated. The continual feed-back of information related to the scanning environment allows more aggressive decoding techniques to be used.

Early/on-time/late gate bit synchronizer

Abstract: A bit synchronizer for a digital receiver system accounts for loss of bit synchronization due to transmission phenomena. The bit synchronizer includes a DC level estimator for converting a sampled digital signal having a bit rate and a sampling rate into a level-adjusted signal. A delay module generates a first timing signal, a second timing signal, and a third timing signal based on the level-adjusted signal. The timing signals correspond to early, on-time, and late sampling windows. The control module generates an output signal based on the timing signals such that the transmit and receive bit timing are synchronized. In one embodiment, the control module has an absolute value stage, an integration stage, and a signal selector. The signal selector is able to select between the timing signals, adjust the symbol rate to re-center the on-time gate, and memory swap to maintain correct averaging operations.

Phase shifting applications of universal frequency translation

Abstract: A universal frequency translation module (UFT) frequency translates an electromagnetic (EM) input signal by sampling the EM input signal according to a periodic control signal (also called on aliasing signal). By controlling the relative sampling time, the UFT module implements a relative phase shift during frequency translation. In other words, a relative phase shift can be introduced in the output signal by sampling the input signal at one point in time relative to another point in time. As such, the UFT module can be configured as an integrated frequency translator and phase-shifter. This includes the UFT module as an integrated down-converter and phase shifter, and the UFT module as an integrated up-converter and phase shifter. Applications of universal frequency translation and phase shifting include phased array antennas that utilize integrated frequency translation and phase shifting technology to steer the one or more main beams of the phased array antenna.

System and method for sampling an analog signal level

Abstract: A system includes a signal reconstruction controller ( 110 ) electrically coupled to at least one analog-to-digital converter (ADC) ( 112 ) and to a phase adjustable clock source ( 108 ). A sampling clock signal ( 116 ) is electrically coupled from the clock source ( 108 ) to the at least one ADC ( 112 ). The at least one ADC ( 112 ) samples an electronic signal according to the sampling clock signal ( 116 ) to provide a digital representation of the electronic signal. The controller ( 110 ) samples data from the ADC ( 112 ) at different sampling points in the electronic signal and determines the edges ( 140 ) of the electronic signal and the noisy samples ( 142, 144 ) that are away from the edges ( 140 ) of the electronic signal. By finding the least noisy sample ( 146, 148 ) that is away from the edges ( 140 ) of the electronic signal the controller ( 110 ) adjusts the phase of a sampling signal clock ( 116 ) to a sampling point that is the most reliable to sample the electronic signal to provide a digital representation thereof.

Method and apparatus for expanding band of audio signal

Abstract: A low-pass filter (1) of over-sampling type over-samples an input digital audio signal T1 and filters and removes the low frequency components of the produced aliasing noise. A spectrum analysis circuit (3) calculates the spectrum strength of a predetermined band of the output signal from the low-pass filter (1). An expansion signal generating circuit (5) generates an expansion signal having higher frequency components from the output signal from the low-pass filter (1). A level control circuit (4) controls the level of the expansion signal according to the output signal from the spectrum analysis circuit (3). An adder (2) adds the level-controlled expansion signal to the output signal from the low-pass filter (1) thereby to generate a digital audio signal T2.

Methods and apparatus for artifact compensation with variable angular sampling

Abstract: The present invention, in one form, includes an imaging system (10) having a x-ray source (14) and a digital detector array (18). The system (10) alters an angular spacing between samples of projection data as a function of the projection angle so that aliasing artifacts are reduced. More specifically and in one embodiment, after identifying at least two projection angle regions, a first view sampling rate is utilized to collect projection data for the first region. As the projection angle becomes equal to the second region, the view sampling rate is altered to a second view sampling rate. A complete volumetric image (42) of an object is then generated (36) from the data.

Method and system for detecting in-band jammers in a spread spectrum wireless base station

Abstract: A method and apparatus for use in a wireless communication base station which provides real-time, continuous detection of in-band jammers with negligible impact on base station cost. In addition to providing jammer detection, the use of the invention to provide improved hardware and system diagnostic capabilities is presented. The jammer detection apparatus utilizes automatic gain control techniques, digital sampling, and digital signal processing to identify anomalous frequency components in the received power spectral density curve. Also described are the use of complex sampling, the complex fast Fourier transform (FFT), and Hanning windows to improve jammer detection capabilities. A method of estimating the location of in-band jammers is described using the computed received power spectral density curves of multiple base stations.

Pulse recording by free-running sampling

Abstract: Pulses from a position-sensitive photomultiplier (PS-PMT) are recorded by free running ADCs at a sampling rate of 40 MHz. A four-channel acquisition-board has been developed which is equipped with four 12 bit-ADCs connected to one FPGA (field programmable gate array). The FPGA manages data acquisition and the transfer to the host computer. It can also work as a digital trigger, so a separate hardware-trigger can be omitted. The method of free running sampling provides a maximum of information, besides the pulse charge and amplitude also pulse shape and starting time are contained in the sampled data. These informations are crucial for many tasks such as distinguishing between different scintillator materials, determination of radiation type, pile-up recovery, coincidence detection or time-of-flight applications. The absence of an analog integrator allows coping with very high count rates. Since this method is going to be employed in positron emission tomography (PET), the position of an event is another important information. The simultaneous readout of four channels allows localization by means of center-of-gravity weighting. First results from a test setup with LSO-scintillators coupled to the PS-PMT are presented.

Asymmetry detection apparatus, jitter detection apparatus, and recording/reproduction apparatus

Abstract: An asymmetry detection apparatus includes: a clock signal generator for generating a clock signal based on a reproduced signal; an A/D converter for sampling the reproduced signal in synchronization with the clock signal; a determiner for determining whether a level of each of a plurality of sampled data obtained by the sampling operation is equal to or greater than a predetermined level; and a detector for detecting asymmetry in the reproduced signal by using predetermined ones of the sampled data based on an output from the determiner. A jitter detection apparatus includes: a clock signal generator for generating a clock signal based on a reproduced signal; an A/D converter for sampling the reproduced signal in synchronization with the clock signal; a determiner for determining whether a level of each of a plurality of sampled data obtained by the sampling operation is equal to or greater than a predetermined level; and a detector for detecting jitter in the reproduced signal by using predetermined ones of the plurality of sampled data based on an output from the determiner.

Signal acquisition system for spread spectrum receiver

Abstract: The invention relates to a device detecting a demodulated signal received by a spread spectrum receiver and converted into digital samples. The device is characterized by comprising a matched filter (4-16) for calculating the correlation between an incoming signal and at least one reference signal; an oscillator (4-8) for generating a sampling frequency; a sampling circuit (4-10) for re-sampling said demodulated digital sample signal at said sampling frequency, which is such that the timing of samples of the reference signals of the matched filter (4-16) corresponds to the timing of a sample signal going from the sampling circuit to the matched filter (4-16); and a multiplier (4-12) in which the sample signal is multiplied by a carrier replica generated locally before the sampling circuit or thereafter, to remove the carrier from the sample signal.