Abstract: A filtering circuit includes a clock selection unit configured to transfer a first clock or a second clock having a frequency lower than the first clock as an operating clock in response to a frequence signal, and a filter configured to filter an input signal and generate a filtered signal in synchronization with the operating clock.

Abstract: Dual-axis swallowing accelerometry is an emerging tool for the assessment of dysphagia (swallowing difficulties). These signals however can be very noisy as a result of physiological and motion artifacts. A novel scheme for denoising those signals is proposed, i.e., a computationally efficient search for the optimal denoising threshold within a reduced wavelet subspace. To determine a viable subspace, the algorithm relies on the minimum value of the estimated upper bound for the reconstruction error. A numerical analysis of the proposed scheme using synthetic test signals demonstrated that the proposed scheme is computationally more efficient than minimum noiseless description length (MNDL) based de-noising. It also yields smaller reconstruction errors (i.e., higher signal-to-noise (SNR) ratio) than MNDL, SURE and Donoho denoising methods. When applied to dual-axis swallowing accelerometry signals, the proposed scheme improves the SNR values for dry, wet and wet chin tuck swallows.

Abstract: A signal filter and accompanying methods. In one embodiment, the filter includes a first mechanism for receiving a first signal. A second mechanism employs one or more modified representations of the first signal to cancel one or more frequency components of the first signal, yielding an output signal in response thereto. In a more specific embodiment, the first mechanism includes a splitter for receiving the first signal and splitting the first signal onto a first path and a second path. The second mechanism further includes one or more delay modules and one or more phase shifters in the first path and/or the second path. One or more controllable amplifiers are optionally included in the first path and/or the second path. The one or more delay modules, phase shifters, or amplifiers are responsive to one or more control signals from a controller. The controller is adapted to modify behavior of the second mechanism so that the filter is characterized by a desired frequency response.

Abstract: A method comprises generating a calibration signal by a clock generator, feeding the calibration signal to a first filter through a first switch unit, comparing an output of the first filter with the calibration signal through a frequency detector and a phase comparator and generating a first updated bandwidth code to adjust a bandwidth frequency of the first filter.

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 fast filter calibration system includes a multi clock generator, an analog filter including a variable capacitor and a fast calibration apparatus. The fast calibration apparatus further includes a phase comparator, a frequency detector and a fast calibration unit. The fast calibration unit stores a binary code corresponding to a bandwidth frequency of a filter and initiates a fast filter calibration by calibrating the filter from a binary code close to a guaranteed-by-design binary code for the bandwidth frequency to be calibrated.

Abstract: A system and method for representing quasi-periodic (“qp”) waveforms, for example, representing a plurality of limited decompositions of the qp waveform. Each decomposition includes a first and second amplitude value and at least one time value. In some embodiments, each of the decompositions is phase adjusted such that the arithmetic sum of the plurality of limited decompositions reconstructs the qp waveform. Data-structure attributes are created and used to reconstruct the qp waveform. Features of the qp wave are tracked using pattern-recognition techniques. The fundamental rate of the signal (e.g., heartbeat) can vary widely, for example by a factor of 2-3 or more from the lowest to highest frequency. To get quarter-phase representations of a component (e.g., lowest frequency “rate” component) that varies over time (by a factor of two to three) many overlapping filters use bandpass and overlap parameters that allow tracking the component's frequency version on changing quarter-phase basis.

Abstract: A method of operation of a communication system includes: generating a filter impulse response and a filter time-domain data with a shortening filter; generating a filter frequency response based on the filter impulse response with a filter frequency response calculator; generating a filter frequency-domain data based on the filter time-domain data with a first process unit; and generating a raw channel impulse response with a filter frequency removal unit for removing the filter frequency response from the filter frequency-domain data.

Abstract: Methods for filtering an input signal x(k) to produce an output signal y(k) such that the ratio of a power level of the output signal to a power level of the input signal is substantially equal to a desired value ? are provided.

Abstract: A signal processing device includes a bit-pattern output unit and a look-up table storage unit which are configured as follows: The bit-pattern output unit is provided for receiving input 1-bit digital signals generated by ?? modification and aligning bits of the input 1-bit digital signals in a chronological order to output parallel bit pattern. The look-up table storage unit is provided for storing a look-up table that represents a relationship between the bit patterns output from the bit pattern output unit and resulting values of a filtering arithmetic operation on the basis of the bit patterns. In the signal processing device, the bit patterns output from the bit-pattern output unit are provided as indexes. The indexes are referenced to output the resulting values of the filtering arithmetic operation corresponding to the bit patterns listed in the look-up table stored in the look-up table storage unit.

Abstract: A signal filter and accompanying methods. In one embodiment, the filter includes a first mechanism for receiving a first signal. A second mechanism employs one or more modified representations of the first signal to cancel one or more frequency components of the first signal, yielding an output signal in response thereto. In a more specific embodiment, the first mechanism includes a splitter for receiving the first signal and splitting the first signal onto a first path and a second path. The second mechanism further includes one or more delay modules and one or more phase shifters in the first path and/or the second path. One or more controllable amplifiers are optionally included in the first path and/or the second path. The one or more delay modules, phase shifters, or amplifiers are responsive to one or more control signals from a controller. The controller is adapted to modify behavior of the second mechanism so that the filter is characterized by a desired frequency response.

Abstract: A fast filter calibration system comprises a multi clock generator, an analog filter comprising a variable capacitor and a fast calibration apparatus. The fast calibration apparatus further comprises a phase comparator, a frequency detector and a fast calibration unit. The fast calibration unit stores a binary code corresponding to a bandwidth frequency of a filter and initiates a fast filter calibration by calibrating the filter from a binary code close to a guaranteed-by-design binary code for the bandwidth frequency to be calibrated.

Abstract: Dual-axis swallowing accelerometry is an emerging tool for the assessment of dysphagia (swallowing difficulties). These signals however can be very noisy as a result of physiological and motion artifacts. A novel scheme for denoising those signals is proposed, i.e. a computationally efficient search for the optimal denoising threshold within a reduced wavelet subspace. To determine a viable subspace, the algorithm relies on the minimum value of the estimated upper bound for the reconstruction error. A numerical analysis of the proposed scheme using synthetic test signals demonstrated that the proposed scheme is computationally more efficient than minimum noiseless description length (MNDL) based de-noising. It also yields smaller reconstruction errors (i.e., higher signal-to-noise (SNR) ratio) than MNDL, SURE and Donoho denoising methods. When applied to dual-axis swallowing accelerometry signals, the proposed scheme improves the SNR values for dry, wet and wet chin tuck swallows.

Abstract: A method of operation of a communication system includes: generating a filter impulse response and a filter time-domain data with a shortening filter; generating a filter frequency response based on the filter impulse response with a filter frequency response calculator; generating a filter frequency-domain data based on the filter time-domain data with a first process unit; and generating a raw channel impulse response with a filter frequency removal unit for removing the filter frequency response from the filter frequency-domain data.

Abstract: A digital filter instruction and filter implementing the filter instruction are disclosed. The filter instruction synthesizes a digital filter and includes an instruction field, a tap length field, a coefficient address field, a data header address field, a clear accumulator bit and an update bit. The filter instruction a concise instruction format to significantly decrease memory required, allow for instruction pipelining without branch penalty, and uses a circular buffer for the data so the data address pointer is only needed to be updated for the next input sample. The filter instruction may be used to implement FIR or IIR filters by using the options of pre-clear accumulator or pre/post storing accumulator results.

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: Methods for filtering an input signal x(k) to produce an output signal y(k) such that the ratio of a power level of the output signal to a power level of the input signal is substantially equal to a desired value ? are provided.

Abstract: A method and system for intrinsic timescale decomposition, filtering, and automated analysis of signals of arbitrary origin or timescale including receiving an input signal, determining a baseline segment and a monotonic residual segment with strictly negative minimum and strictly positive maximum between two successive extrema of the input signal, and producing a baseline output signal and a residual output signal. The method and system also includes determining at least one instantaneous frequency estimate from a proper rotation signal, determining a zero-crossing and a local extremum of the proper rotation signal, and applying interpolation thereto to determine an instantaneous frequency estimate thereof.

Abstract: A Frequency discriminator based on a combination of Discrete Fourier Transform (DFT) operator and half-bin DFT operator, corresponding to a set of Twiddle factors with frequency double than the sampling frequency. The frequency estimators are so chosen as not to have any zero or discontinuity point. Therefore the discriminator of embodiments of invention is more stable and well-behaved in an extended operating range. The discriminator of the invention, when applied to a Global Positioning System (GPS) receiver, allows a safer lock to the carrier frequency, even in presence of a large initial error, and avoids the problem of false locks.

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 receiver including a processor. A plurality of Fast Fourier Transforms of a plurality of signal segments of a signal can be determined by the processor. The plurality of Fast Fourier Transforms are stored. A frequency shift of the signal can be determined recursively based on the stored plurality of Fast Fourier Transforms of the plurality of signal segments.

Abstract: Methods, circuits, and an apparatus for filtering high-speed serial data is disclosed. In one embodiment, a Programmable Logic Device (PLD) is configured with a filter circuit for filtering serial data at a first clock rate. The filter circuit converts an N number of serial data streams into an N number of M-bit words based on a deserialization factor. The M-bit words are converted to an M number of N-bit data words. The N-Bit data words are filtered at a second clock rate, reformatted, serialized, and outputted as individual serial data streams at the first clock rate. In one embodiment, the N-bit data words are digitally filtered by a Finite Impulse Response (FIR) filter operating at the second clock rate. The data output of the FIR filter is then serialized into an N number of serial data output streams operating at the first clock rate.

Abstract: An efficient configurable signal filter. The filter includes a first mechanism for receiving a first signal of a first type and a second signal of a second type. A second mechanism selectively filters the first signal during a first mode of operation, and filters the second signal during a second mode of operation. A third mechanism generates control signals. A fourth mechanism automatically configures the second mechanism to operate in the first mode of operation or the second mode of operation based on the control signals. In a specific embodiment, the first type of signal is characterized by a first rate, and the second type of signal is characterized by a second rate. The first signal and the second signal are digital ADC outputs. The second mechanism includes plural filter blocks, each having one or more Multiply-Accumulate (MAC) pipes. Each of the one or more MAC pipes include one or more MAC blocks that are each associated with a coefficient memory data structure of a coefficient memory.

Abstract: The fundamental frequency of a harmonic signal is estimated by forming a fundamental frequency hypothesis (f0?). A comb filter is provided based on the fundamental frequency hypothesis. The harmonic signal is filtered using the comb filter. The fundamental frequency hypothesis is tested for each tooth in the comb filter. A signal indicating an estimated fundamental frequency of the provided harmonic signal may be outputted based on the testing.

Abstract: Finite response filters (FIRs) are divided into partial filters that filter a same portion of image data to generate partial filtered results. The partial filtered results may be saved and later retrieved to generate complete filter outputs.

Abstract: A method for calculating frequency offset using arc tangent calculations according to real part and imaginary part input values received externally. The method includes converting arc tangent equations for determining frequency offset into equations to perform log calculations; calculating frequency offset using linear approximation and the converted log equations; and dividing the real part input value to bit values of a predetermined size to realize a size similar to that of the imaginary part input value if a range of the frequency offset value to be calculated is small. With the use of the method, hardware area and power consumption may be reduced, and precise frequency offset may be calculated.

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: In a frequency spectrum analyzer (2) which receives a modulation signal previously measures the frequency (F). A synchronous detection frequency (B) is determined on the basis of the frequency (F). An adder (7) obtains the difference (G) between the synchronous detection frequency (B) and the reference frequency (C). From an I signal and a Q signal which are obtained by performing synchronous detection by using the synchronous detection frequency (B), a frequency error detector (6) obtains a frequency error (E) by the phase trajectory method. An adder (8) adds the difference (G) and the frequency error (E) to obtain a frequency error.

Abstract: The voice detecting device includes a voice frequency detector for detecting the frequency of the input voice, then discriminating whether the detected frequency falls within a preset frequency range of voices to be detected, and outputting the result of the discrimination; an input signal level detector for detecting the energy level of the input voice, then comparing it to confirm whether the detected energy level exceeds a preset energy level threshold value of voices to be detected, and outputting the result of the comparison; a voice input judge part responsive to the result of discrimination and the result of comparison to judge whether a voice satisfying conditions for voice detection has been input or not, and output a first status signal in accordance with the result of the judgement; and a voice duration measure part for measuring the duration of the first status signal, then comparing it to confirm whether the detected duration falls within a preset range of duration threshold value, and outputting

Abstract: For a left-channel input audio signal, digital filters 30LL and 30LR are provided for reproduction of impulse responses used for locating an acoustic image outside the head of a listener, and for a right-channel input audio signal, digital filters 30RL and 30RR are provided for reproduction of impulse responses for locating an acoustic image outside the listener's head. An addition circuit 7L adds output signals of the digital filters 30LL and 30RL to each other and a second addition circuit 7R adds output signals of the digital filters 30LR and 30RR to each other. Output signals of the addition circuits 7L and 7R are supplied to a headphone, whereby the reproduction acoustic images of the left-channel and right-channel input audio signals are located outside the listener's head. The sampling rate in each of the digital filters 30LL-30RR is changed depending on the section of the response time of the associated impulse response to be reproduced.

Abstract: The frequency of a signal is determined using a signal filter having a signal filter skirt band frequency range wherein an intermediate fraction of an initial amplitude of an introduced signal passes through the signal filter. The magnitude of the transmitted intermediate fraction varies with the frequency of the introduced signal. A test applied signal having a frequency which lies within the signal filter skirt band frequency range is introduced into the signal filter. The transmitted amplitude of the test applied signal is measured after the test applied signal has passed through the signal filter. The transmitted amplitude is utilized as an indicator of the frequency of the test applied signal, either directly or through a calibration relation for amplitude as a function of frequency in the signal filter skirt band frequency range.

Abstract: To determine the frequency content of a time series, a time series segment which spans a time interval is received. A time interval spanned by a symmetric tapered window with quantised elements is chosen such that the window time interval is greater than the segment time interval such that when the window is centered with respect to the segment time interval, the window element at either end of the segment time interval has the lowest available non-zero quantised value. Lastly a frequency representation of said time series may be obtained based on the time series segment and the quantised window.

Abstract: The frequency measurement method and associated apparatus accurately determines the frequency of a periodic signal even if the frequency exceeds the nyquist frequency, typically established by the analog to digital ("A/D") converter used to sample the signal. The frequency measurement method and apparatus obtains samples of a periodic signal at a number of sample rates, converts the samples to the frequency domain, analyzes the first order alias frequencies produced due to undersampling of the signal, and determines the frequency of the periodic signal based upon the alias frequencies. In selecting the sampling rate and the first order alias frequencies, the frequency measurement method and apparatus accounts for sampling errors, frequency calculation errors, desired frequency ranges, and measurement precision. Through error compensation and minimization techniques, the frequency measurement method and apparatus reduces the number of electrical components required (e.g. A/D converters).