Abstract: Systems and methods for transmitting data using various Modulation on Zeros schemes are described. In many embodiments, a communication system is utilized that includes a transmitter having a modulator that modulates a plurality of information bits to encode the bits in the zeros of the z-transform of a discrete-time baseband signal. In addition, the communication system includes a receiver having a decoder configured to decode a plurality of bits of information from the samples of a received signal by: determining a plurality of zeros of a z-transform of a received discrete-time baseband signal based upon samples from a received continuous-time signal, identifying zeros that encode the plurality of information bits, and outputting a plurality of decoded information bits based upon the identified zeros.

Abstract: A computing system determines if an event has occurred. A first window is defined that includes a subset of a plurality of observation vectors modeled as an output of an autoregressive causal system. A magnitude adjustment vector is computed from a mean computed for a matrix of magnitude values that includes a column for each window of a plurality of windows. The first window is stored in a next column of the matrix of magnitude values. Each cell of the matrix of magnitude values includes an estimated power spectrum value for a respective window and a respective frequency. A second matrix of magnitude values is updated using the magnitude adjustment vector. Each cell of the second matrix of magnitude values includes an adjusted power spectrum value for the respective window and the respective frequency. A peak is detected from the next column of the second matrix of magnitude values.

Abstract: Systems and methods for transmitting data using various Modulation on Zeros schemes are described. In many embodiments, a communication system is utilized that includes a transmitter having a modulator that modulates a plurality of information bits to encode the bits in the zeros of the z-transform of a discrete-time baseband signal. In addition, the communication system includes a receiver having a decoder configured to decode a plurality of bits of information from the samples of a received signal by: determining a plurality of zeros of a z-transform of a received discrete-time baseband signal based upon samples from a received continuous-time signal, identifying zeros that encode the plurality of information bits, and outputting a plurality of decoded information bits based upon the identified zeros.

Abstract: A spurious frequency attenuation servo is provided. The spurious frequency attenuation servo includes a first function generator that generates a first signal at a first frequency and at a spurious frequency; a second function generator that generates a second signal in-phase with the first signal and at the spurious frequency; a third function generator that generates a third signal ninety degrees out-of-phase with the first signal and at the spurious frequency; in-phase and quadrature-phase mixers to input a feedback signal and the second and third signals, respectively; in-phase and quadrature-phase error accumulators; an in-phase and quadrature-phase multiplier to multiply an output from the in-phase and quadrature-phase error accumulators with the second and third signals, respectively; and a summing node to sum the first signal with output from the in-phase and quadrature-phase multipliers to form an output signal that is fed back to the in-phase mixer and the quadrature-phase mixer.

Abstract: Various embodiments of the present invention provide apparatuses and methods for filtering a digital signal with a Nyquist constrained digital finite impulse response filter. For example, an apparatus for filtering digital data is disclosed that includes a digital finite impulse response filter having a plurality of taps. The apparatus also includes a tap weight controller connected to the digital finite impulse response filter, operable to adjust a tap weight for each of a subset of the taps such that a magnitude of a Nyquist response of the digital finite impulse response filter remains within a constraint range.

Abstract: An adaptive filter configured to filter an input signal comprises Fourier transforming unit configured to transform the input signal into a frequency domain signal upon the basis of the Fourier transform, weighting unit configured to weight at least a portion of the frequency domain signal with a filter coefficient of the dispersion filter in frequency domain to obtain a filtered signal in frequency domain, correlating unit configured to correlate the filtered signal in frequency domain to obtain a correlation value, and adaptation unit configured to adapt the filter coefficient upon the basis of the correlation value.

Abstract: A system is configured so that signals passed through an all-pass filter using warp parameter ? are whitening-filtered, and so that the frequency axis is restored by an all-pass filter using warp parameter ?. This optimizes the whitening filter by determining the optimum ?. First, the AR order p is automatically estimated using ?=0. A spectral distance d? is computed using a discrete Fourier transform spectrum value passed through an all-pass filter using warp parameter ? and a discrete AR spectrum value passed through an all-pass filter using warp parameter ?. The ? which minimizes the spectral distance d? is set as the warp parameter, and a whitening filter is configured in which the warp parameter has been optimized.

Abstract: Physiological signals are denoised. In accordance with an example embodiment, a denoised physiological signal is generated from an input signal including a desired physiological signal and noise. The input signal is decomposed from a first domain into subcomponents in a second domain of higher dimension than the first domain. Target subcomponents of the input signal that are associated with the desired physiological signal are identified, based upon the spatial distribution of the subcomponents. A denoised physiological signal is constructed in the first domain from at least one of the identified target subcomponents.

Abstract: A method and a device for determining sampling rate are provided. The device receives an input signal of SPDIF. The method includes following steps. A plurality of multiple values between a plurality of bi-phase clock frequencies of the input signal and a system frequency are obtained, and a first weighted average and a second weighted average are calculated according to a first filter range, a second filter range and the multiple values. When a first difference is greater than a second difference, the sampling rate is set to a first sampling rate. Otherwise, the sampling rate is set to a second sampling rate. The first difference and the second difference are obtained according to the first weighted average, the second weighted average and a frequency threshold. The method determines the sampling rate rapidly according to weighted averages adjusted by filter ranges, reduces a probability of erroneous judgment and saves memory.

Abstract: A receiver comprises an adaptive filter having an input for a digitized input signal, means for storing a pre-designed filter characteristic, means for analyzing a digital. representation of the input signal to determine a desired position of the filter characteristic to match the system requirements, and means for adapting the stored pre-designed filter characteristic in the frequency domain and/or the time domain to match the system requirements and for transforming the adapted filter characteristic to the time domain to update coefficients for the adaptive filter and for loading updated coefficients into adaptive filter. The updating of the coefficients may be done periodically. The adaptation may be one or more of adjusting bandwidth, frequency shift and, in the case of a bandpass characteristic, superimposing characteristics.

Abstract: The invention provides a method for identification of traffic lane boundary. Firstly the microwave signal is received, and the noise reduction is treated for the microwave signal. Then the frequency domain information is employed to calculate the legal set of closed interval, in order to form the frequency span information. Finally, the probability density function model is employed to calculate the frequency span information in order to identify the traffic lane boundary.

Abstract: A digital signal processing system comprising: an input terminal to receive an input signal that includes a distorted component and an undistorted component, the input signal having a sampling rate of R; and an adaptive self-linearization module coupled to the input terminal, to perform self-linearization based at least in part on the input signal to obtain an output signal that is substantially undistorted, wherein: the adaptive self-linearization module is to generate a replica distortion signal that is substantially similar to the distorted component, the generation being based at least in part on a target component having a sampling rate of R/L, L being an integer greater than 1; the adaptive self-linearization module includes a first digital signal processor (DSP) that is adapted to obtain a filter transfer function that approximates a system distortion transfer function, and a second DSP that is configured using configuration parameters of the first DSP.

Abstract: Systems and methods for utilizing new communication standards in wireless local area networks are provided that also support legacy wireless stations. The method can include user equipment determining channel state information, selecting a unitary channel decomposition precoder format based on the determined channel state information and transmitting the precoder format information to a base station. During a return transmission the user equipment can receive user data with the precoder format information and utilize a non-linear detector to demodulate and decode the user data. Based on the reception the user equipment can estimate channel quality; and transmitting channel quality information as feedback.

Abstract: A method and a device for determining sampling rate are provided. The device receives an input signal of SPDIF. The method includes following steps. A plurality of multiple values between a plurality of bi-phase clock frequencies of the input signal and a system frequency are obtained, and a first weighted average and a second weighted average are calculated according to a first filter range, a second filter range and the multiple values. When a first difference is greater than a second difference, the sampling rate is set to a first sampling rate. Otherwise, the sampling rate is set to a second sampling rate. The first difference and the second difference are obtained according to the first weighted average, the second weighted average and a frequency threshold. The method determines the sampling rate rapidly according to weighted averages adjusted by filter ranges, reduces a probability of erroneous judgment and saves memory.

Abstract: In one embodiment, a method for discriminating between a machine generated multi-tone signal and a simulated voice multi-tone signal is provided. The method may comprise comparing output profiles generated from sampled outputs of a plurality of filters. The plurality of filters may have a single input sample stream applied to them and each filter may be preset at a measured multi-tone frequency associated with an identified prospective multi-tone signal. Based on the comparison of the output profiles, an inequality degree between the output profiles is generated and compared to an inequality threshold, thereby to determine whether the input sample stream comprises a machine generated multi-tone signal.

Abstract: A deterministic component identifying apparatus identifies a distribution shape of a deterministic component included in a probability density function supplied thereto. The apparatus includes a standard deviation calculating section that calculates a standard deviation of the probability density function, a spectrum calculating section that calculates a spectrum of the probability density function, a null frequency detecting section that detects a null frequency of the spectrum, and a ratio calculating section that calculates a ratio between a top portion and a bottom portion of a distribution of the deterministic component, based on the standard deviation of the probability density function and the null frequency of the spectrum.

Abstract: A repetitive controller scheme with negative feedback and feedforward introduces infinitely many poles on the imaginary axis located at the resonant peaks. The feedforward introduces zeros, which produce notches located in between two consecutive resonant peaks. The latter has the advantage of making the controllers more selective, in the sense that the original overlapping (appearing at the valleys) or interaction between consecutive resonant peaks is removed by the notches. This would allow, in principle, peaks of higher gains and slightly wider bandwidth, avoiding, at the same time, the excitation of harmonics located in between two consecutive peaks. A negative feedback compensator with feedforward is especially useful when only the compensation of odd harmonics is required, but not the even harmonics, like in many power electronic systems. In contrast, the positive feedback controller would try to reinject, and indeed amplify, any small noise, which has components on the even frequencies.

Abstract: An approach is provided for measuring, identifying, and removing at least one sinusoidal interference signal (Ak·ej(?kt+?k), Ak·ej(?·?k?t+?k)) in a noise signal (w(t), w(?·?t)). A frequency range to be measured is split into a plurality of frequency bands (?) via a Fast Fourier Transform (FFT) filter bank. For each of the frequency bands (?), an autocorrelation matrix ({circumflex over (R)}?) is determined, wherein parameters of the autocorrelation matrices ({circumflex over (R)}?) are variably adjusted based on whether the at least one sinusoidal interference signal (Ak·ej(?kt+?k), Ak·ej(?·?k?t+?k)) is to be measured, identified, or removed and further based on at least one averaging. The autocorrelation matrices ({circumflex over (R)}?) are jointly utilized for one or more of measuring, identifying, or removing the at least one sinusoidal interference signal (Ak·ej(?kt+?k), Ak·ej(?·?k?t+?k)) in the noise signal (w(t), w(?·?t)).

Abstract: A frequency change detector splits a frequency standard signal into two undelayed frequency signals, one of which is delayed by a predetermined amount. The delayed signal is then mixed with the undelayed frequency signal into a mixed signal that is further filtered and amplified for providing an output signal indicating frequency changes of the frequency standard signal. The mixed frequency signal indicates frequency changes of the original frequency standard signal without reference to another frequency standard. This frequency change detector is well suited for use on satellites as an early warning detection of changes in on-board atomic frequency standards.

Abstract: A signal processing method for performing hierarchical empirical mode decomposition (H-EMD) and an apparatus therefor are provided. In an embodiment, when empirical mode decomposition is performed on an input signal, an artificial assisting signal and the input signal are combined to assist the search for extrema and frequency reduction is performed in each iteration to eliminate the artificial assisting signal and make mode decomposition convergent so as to avoid mode mixing. In addition, in an embodiment, a hierarchical decomposition method is provided to decompose the input signal into a fewer number of fundamental modes. For needs in application, one of the fundamental modes can be further decomposed to produce a number of supplementary modes. In an embodiment, the H-EMD with appropriate frequency reduction can result in modes substantially independent of the form or the way of envelopes and can be applied to decompose multi-dimensional signals.

Abstract: A digital data bandpass filter process using digital signal processing processes raw digital data twice using distinct all-pass filters, wherein the two filters perform different frequency-dependent phase shifts of the digital content. The two filter output streams may be subtracted to remove out-of-band energy. The process produces lower levels of noise and conversion artifacts when used for filtering fixed-point data that may have low energy, and thus few non-zero bits, in the analog input signals, than conventional digital bandpass filters.

Abstract: The accumulated change in values representative of actions taken by a processor, such as the number of email messages processed by an email server, in a given time period is determined. Actions are represented as data points on a plot. Look-ahead intervals are defined for each point. Candidate pairs of points are determined for each look-ahead interval by comparing the first value in the look-ahead interval with other values in the look-ahead interval. A candidate pair comprises the first point and another point having a lesser value. If a candidate pair has a value therebetween, the candidate pair is discarded. If, however, a candidate pair has no value therebetween, the first value of the candidate pair is a peak value for the look-ahead interval. The accumulated change is determined by calculating the sum of the peak values, plus the final value, minus the initial value, for the given time period.

Abstract: Provided are a device and method for detecting a wiretapping device using a power line and nullifying the wiretapping device. More particularly, a device and method for preventing wiretapping, which sense a wiretapping signal from a power line and transmit a noise signal to the power line, are provided. The device for preventing wiretapping includes: a signal detector for receiving signals from a power line and filtering the received signals in at least one frequency band; a controller for receiving the filtered signal from the signal detector and determining whether a wiretapping signal exists; and a noise signal output unit for transmitting a noise signal to the power line according to whether or not the wiretapping signal exists. The device can detect wiretapping and simultaneously nullify the function of the wiretapping device by detecting a wiretapping signal from a power line and transmitting a noise signal having a frequency corresponding to a frequency band of the wiretapping signal.

Abstract: Wavelet filters computed from a wavelet transform are used for encoding, transmitting, receiving, and decoding spread spectrum signals on multiple parallel frequency channels. The transmitter encodes the data stream via a mapping to a basis set of wavelets which are orthogonal across the time domain. The spread spectrum data is parsed from a serial bit stream to several parallel streams. The number of bits per symbol need not remain constant. Signals are created from each symbol by first up sampling by inserting zeros between successive symbols. These signals are passed through a bank of low-pass and high-pass filters derived from a wavelet packet transform. The resulting signal is transmitted in the base band of the transmission system, or transmitted by modulating the carrier of the transmission system. At the receiver, the steps are reversed to recover the symbols. Symbol decisions are made through any one of a number of methods.

Abstract: A method of optimizing performance of a memory subsystem in a computer system. This method including the step of informing the computer system of what type of memory module loading is present in the memory subsystem. The method also includes operating the memory subsystem at a first frequency that is at a chosen ratio to a second frequency at which a frontside bus (FSB) operates. Further, the method includes optimizing the performance of the memory subsystem by changing the ratio.

Abstract: A technique for performance-monitoring of a standard SONET signal involves first converting the optical signal to an electrical signal, removing in the framing signal time slot the framing signal for leaving only the framing signal noise in such time slot, and separating the framing signal noise from the data signal for viewing as a measure of the quality of the SONET signal.

Abstract: A frequency margin testing blade is adapted for use in a bladed server. The testing blade is further adapted to provide one or more output clock signals for use as clock inputs to one or more server blades internal to the bladed server in which the testing blade is installed and/or one or more server blades external to the bladed server in which the testing blade is installed.

Abstract: The present invention implements a servo system which can support changes of the higher resonance frequencies of the actuator with a simple configuration at low cost.

Abstract: The invention provides a method for searching the frequency range of a digital data receiving device, said method including the steps in one embodiment of firstly identifying and disregarding analogue carrier frequencies in the range, then proceeding to estimate the initial starting frequency for a digital carrier by dynamic real-time pre-processing of the spectrum profile and then obtaining successive samples of the frequency range from the start position for the carrier until the center frequency for the carrier is located. In one embodiment the method includes the prediction of the center frequency estimations of all digital carriers for respective channels in the range prior to the search commencing. The method reduces the possibility of locking to analogue carriers, and can reduce the time required to perform a search in comparison to the current digital carrier frequency “search & lock” procedure.

Abstract: An arithmetic processing section is designed to repeatedly execute product-sum arithmetic processing a number of times equal to the number of samples corresponding to each of the frequencies, in which a value obtained by subtracting a product-sum arithmetic value two sampling periods before from an input signal and a value obtained by multiplying a product-sum arithmetic value one sampling period before by a coefficient corresponding to a reference frequency contained in a DTMF signal are added. A comparator extracts an output value equal to or larger than a predetermined threshold value from the product-sum arithmetic values obtained by the arithmetic processing section. A matrix section determines the type of DTMF signal on the basis of at least two output values extracted by the comparator.

Abstract: Methods and apparatus of signal processing are described. In a method according to one embodiment, a high-frequency region of a digital signal is detected. In response to the detecting, the high-frequency content of the digitized signal is increased in at least a portion of that region.

Abstract: Data other than inserted “0”'s are selected by first selectors from among a plurality of data included in input data to which zero value interpolation is carried out at an interpolation circuit. Besides, coefficients by which the data selected by the first selectors should be multiplied are selected by second selectors. The data selected by the first selectors are multiplied by the coefficients selected by second selectors in multiplication circuits. Then, an adding circuit adds all of the multiplied results and outputs the added result as the desired filter characteristic.

Abstract: A method and system for improved detection of analog signals, where such signals are composed of one or more analog frequencies of defined duration. In the method and system, an analog signal is received. A stream of data samples is created from the received analog signal. Based on the stream of data samples, a duration is calculated for one or more analog frequencies contained within the received analog signal. The duration is calculated for the one or more frequencies by utilizing a calculated signal energy for each of the one or more analog frequencies. The calculated signal energies for each of the one or more analog frequencies are used to determine a number of frequency-specific data samples. The number of frequency-specific data samples are then utilized with a sampling rate to calculate the duration of the each of the one or more analog frequencies.

Abstract: A signal processor stores at least one microprogram having m steps in total smaller in number than n steps which are to be executed within one sampling repetition period. A count corresponding to each of the n steps are generated. M steps of the at least one microprogram stored is executed a plurality of times within the one sampling repetition period to thereby execute the n steps.