Abstract: A receiving apparatus for a digital mobile communication system comprises an adaptive filter for filtering an input signal. A step-size parameter chosen for the adaption of filter coefficients of the adaptive filter is computed from a variation of the filter coefficients used by the adaptive filter. This facilitates an indirect measure for the channel variation so that a good reception quality over a wide range of user velocities may be enabled in contrast to a system design based on a compromise step-size parameter being optimum for one velocity only.

Abstract: In order to reduce or eliminate aliasing in a process control network, filtering of a measurement signal may be set based on the module execution rate in a process control system. A Nyquist frequency for the module may be determined based on the module execution rate where the Nyquist frequency may be twice the execution rate. Filtering after an analog to digital convertor may be set based on the module execution rate. In the analog to digital convertor, digital filtering after the converter may be set based on the module execution rate and the frequency content of the analog signal may be attenuated by a filter at and above the Nyquist frequency for the module execution rate.

Abstract: A patient monitoring system may receive a photoplethysmograph (PPG) signal including samples of a pulse waveform. A plurality of morphology metric signals may be generated from the PPG signal. The system may generate an autocorrelation sequence for each of the morphology metric signals. An autocorrelation metric may be generated from each of the autocorrelation sequences and may represent the regularity or periodicity of the morphology metric signal. The autocorrelation sequences may be combined to generate a combined autocorrelation sequence, with the weighting of the autocorrelation sequences based on the autocorrelation metric. The combined autocorrelation sequence may be used to determine physiological information.

Abstract: Various embodiments of the present invention provide systems and methods for data processing. As an example, a data processing circuit is disclosed that includes a noise predictive filter circuit, a data detector circuit, and a first and a second pattern dependent adaptive target circuits. The noise predictive filter circuit includes at least a first pattern dependent filter circuit operable to perform noise predictive filtering on a data input for a first pattern using a first adaptive target to yield a first noise predictive output, and a second pattern dependent filter circuit operable to perform noise predictive filtering on the data input for a second pattern using a second adaptive target to yield a second noise predictive output. The data detector circuit is operable to apply a data detection algorithm to the first noise predictive output and the second noise predictive output to yield a detected output.

Abstract: An adaptive filter device, including a finite impulse response (FIR) filter which is based on filter coefficients, which are determined based on a predetermined iterative adaptation algorithm for determining filter coefficients of an adaptive filter, wherein, in at least one iteration step of said predetermined iterative adaptation algorithm a sum value is determined, wherein each summand of said sum value depends on one of said filter coefficients, and, if said sum value is above a predetermined threshold, the filter coefficients are modified.

Abstract: Various embodiments of the present invention provide systems and methods for data processing. As an example, a data processing circuit is disclosed that includes a data detector circuit. The data detector circuit includes an anti-causal noise predictive filter circuit and a data detection circuit. In some cases, the anti-causal noise predictive filter circuit is operable to apply noise predictive filtering to a detector input to yield a filtered output, and the data detection circuit is operable to apply a data detection algorithm to the filtered output derived from the anti-causal noise predictive filter circuit.

Abstract: A dynamic filtering device includes a variation detector, a coefficient generator and a filter. The cut-off frequency of the filter is dynamically adjusted according to variations of an input signal. A higher signal-to-noise ratio is obtained when a finger moves in slow motion and its response time is reduced when the finger moves in fast motion, therefore improving the response time and the noise immunity of the filter.

Abstract: An adaptive nonlinear filtering system includes an adaptive filter module that is configured to generate relative location information pertaining to a relative location of an input signal within an input range; determine an input dependent filter parameter based at least in part on the relative location information; generate an output signal based at least in part on the input dependent filter parameter; and feed back a feedback signal that is generated based at least in part on the output signal and a target signal.

Abstract: Methods for determining variance properties of a noise component of a raw signal of a machine or a system. An example method includes recording a signal using a noise estimation unit, numerically differentiating the signal using a first module of the noise estimation unit to obtain a differentiated signal, identifying, using a second module of the noise estimation unit, a histogram which corresponds to the differentiated signal, and determining using the histogram, a variance property of the noise component of the signal.

Abstract: An embodiment of the invention pertains to demodulating a data communication into a sequence of symbols. In this embodiment, a first filter generates a first convolution between a first plurality of coefficients and the data communication. The data communication is a distortion of a first sequence of symbols selected from a plurality of symbols in a constellation. A first error circuit maps the first convolution to a second sequence of symbols. An adaption circuit adjusts the first coefficients until a convergence at a last one of the symbols in the second sequence. A second filter generates a second convolution between a second plurality of coefficients and the data communication. The second coefficients are initialized to the first coefficients from the adaption circuit. A second error circuit maps the second convolution to a third sequence of symbols.

Abstract: A method for calculating coefficients of a filter and a method for filtering are provided. The invention directly factorizes a specific function in a cepstrum domain by spectral factorization and cepstrum technique to obtain coefficients of denominator function from the filter. In other words, the invention adopts a non-iterative algorithm to reduce computational complexity and avoid convergence due to calculating coefficients. Besides, the specific function of the invention includes a compensation function, so that a Fourier transform with greatly reduced size can be utilized in the spectral factorization to greatly save the computations and keep good system performance at a receiver.

Abstract: A transversal filter circuit comprises a plurality of delay units, a plurality of multiplexers and a plurality of full adders. The plurality of delay units is coupled in series to delay a two-bit input signal. The plurality of multiplexers is coupled to the plurality of delay units in a one-to-one manner, and outputs zero, a data signal, or the inverse of the data signal according to the output signals of the plurality of delay units. The plurality of full adders accumulates the outputs of the plurality of multiplexers and the MSB of the outputs of the plurality of the delay units.

Abstract: A spectrum agile radio having one or more variable digital filters is described. To quickly, yet accurately, retune the digital filter(s) a windowing function is applied to an ideal filter characteristic for each of one or more desired frequency bands to generate filter coefficients. Transitioning between coefficients of a previous filter and a current filter is handled to avoid problems associated with discontinuities in the signal processing.

Abstract: Instability resulting from non-linear impairments is detected and an equalizer of an end device is reset. An equalization instability threshold is retrieved from a data storage device. An equalization parameter for the end device is monitored and, if the equalization parameter exceeds the equalization instability threshold, the equalizer is reset.

Abstract: A device for allocating a number of taps of a designated finite impulse response filter is disclosed. The device comprises a plurality of designated finite impulse response filters having fixed number of taps, a plurality of allocation finite impulse response filters having fixed number of taps, a control unit and an estimate unit. Depending on intensities of responses to interferences, at least one of the allocation FIR filters may be coupled in series to any one of the designated finite impulse response filters, thereby to provide a signal having excellent quality.

Abstract: Methods for modeling the secondary path of an ANC system to improve convergence and tracking during noise control operation, and their associated uses are provided. In one aspect, for example, a method for modeling a secondary path for an active noise control system is provided. Such a method may include receiving a reference signal, filtering the reference signal with an initial secondary path model to obtain a filtered reference signal, calculating an autocorrelation matrix from the filtered reference signal, and calculating a plurality of eigenvalues from the autocorrelation matrix. The method may further include calculating a maximum difference between the plurality of eigenvalues and iterating a test model to determine an optimized secondary path model having a plurality of optimized eigenvalues that have a minimized difference that is less than the maximum difference of the plurality of eigenvalues, such that the optimized secondary path model may be utilized in the active noise control system.

Abstract: An adaptive low pass filtering process with a filter delay DA is conducted in parallel with reference low pass filtering process with a filter delay DR which is greater than DA. The error is measured between a delayed version of the adaptive process output and the reference process output. Filter parameters of the adaptive process are controlled to minimise the error.

Abstract: A sound signal correcting apparatus converts an acquired sound signal into a phase spectrum and an amplitude spectrum by an FFT process, compares the amplitude spectrum of the obtained sound signal with a noise model so that a correction coefficient used for correcting the amplitude spectrum of the sound signal is derived, smoothes waveform of the amplitude spectrum of the sound signal using the derived correction coefficient, and converts the sound signal into a sound signal where the amplitude spectrum is corrected by performing an inverse FFT process on the phase spectrum and the smoothed amplitude spectrum.

Abstract: A signal processing method is provided. The method includes receiving a signal input, and filtering the signal input using a nonlinear correntropy filter. The method additionally includes generating an output based upon the filtering of the signal input. The nonlinear correntropy filter can be configured as a nonlinear Wiener filter. Alternatively, the nonlinear correntropy filter can be configured as a correntropy least mean square (LMS) filter, or a correntropy Newton/LMS filter.

Abstract: A second output transmission signal (“TX2”) added to a line driver is a scaled version of the main output transmission signal (“TX1”). TX2 is scaled from TX1 by a variable scale factor K. An adaptive hybrid circuit subtracts TX1 and TX2 from a line signal carrying both a line transmission signal and a line received signal (“RX”). A programmable impedance Ztune is coupled between the TX2 output of the line driver and the RX output of the adaptive hybrid circuit. A transmission echo in the output RX signal is measured. K and Ztune are then adaptively tuned to minimize the transmission echo. The hybrid in this case becomes a 4-port network, one port specifically added to adaptively cancel the transmission echo in the RX output of the adaptive hybrid circuit. Alternatively, the hybrid may be a 3-port hybrid including variable impedances to cancel the line transmission signal.

Abstract: Provided are an equalization apparatus and method of compensating a distorted received signal. The equalization apparatus includes: a filter unit removing inter-symbol interference (ISI) from a multi-channel signal that is received; and a zero-offset controller identifying a zero offset of the multi-channel signal and determining operating coefficients of the filter unit by reflecting the identified zero offset. A response filter, which reduces loss and noise, can be used, and the structure of the response filter can be simplified. In addition, channel characteristics are estimated in real time at an initial stage of data transmission and reception. Thus, an equalizer optimized for channel interference characteristics can be provided.

Abstract: Various embodiments of the present invention provide systems and methods for reducing filter sensitivities. As an example, reduced sensitivity filter circuits are discussed that include a digital filter and a filter tap adaptation circuit. The digital filter is operable to filter a received input based at least in part on a plurality of filter taps, and to provide a filtered output. The filter tap adaptation circuit is operable to receive an error value and a weighting control value, and to adaptively calculate at least one of the filter taps using the error value and the weighting control value.

Abstract: A device for and method of determining a coherence measurement for a signal that includes a digitizer for digitizing the signal, a transformer connected to the digitizer, a first squarer connected to the transformer, a second squarer connected to the digitizer, an adder connected to the first squarer and the second squarer, a subtractor connected to the first squarer and the second squarer, a standard-deviation function block connected to the subtractor, a mean generator connected to the adder, a first multiplier connected to the standard-deviation function block, and a divider connected to the output of the mean generator and the first multiplier.

Abstract: One embodiment of the present invention relates to an adaptive filtering apparatus comprising first and second real valued adaptive filters, respectively configured to receive an adaptive filter input signal based upon a transmission signal in a transmission path. The first real valued adaptive filter is configured to operate a real valued adaptive filter algorithm on the input signal to estimate a first intermodulation noise component (e.g., an in-phase component) in a desired signal and to cancel the estimated noise. The second real valued adaptive filter is configured to operate a real valued adaptive filter algorithm on the input signal to estimate a second intermodulation noise component (e.g., a quadrature phase component) in the desired signal and to cancel the estimated noise. Accordingly, each filter operates a real valued adaptive algorithm to cancel a noise component, thereby removing complex cross terms between the components from the adaptive filtering process.

Abstract: Methods and systems for multi-input IIR filters with error feedback are disclosed. By using multiple-inputs to generate multiple outputs during each iteration, a multi-input IIR filter in accordance with the present invention has greatly increased throughput. Furthermore, the addition of a multi-variable error feedback unit in accordance with the present invention in a multiple-input IIR filter can greatly increase the accuracy of the multi-variable IIR Filter.

Abstract: Various embodiments of the present invention provide systems and methods for data processing. As an example, a data processing circuit is disclosed that includes: a noise predictive filter circuit, a scaling factor adaptation circuit, and a scaling factor application circuit. The noise predictive filter circuit is operable to perform a noise predictive filtering process on a data input based on a filter tap to yield a noise filtered output. The scaling factor adaptation circuit is operable to calculate a scaling factor based at least in part on a derivative of the noise filtered output. The scaling factor application circuit is operable to apply the scaling factor to scale the noise filtered output.

Abstract: An adaptive equalizer includes an adaptive filter and a control unit. The adaptive filter performs an adaptive equalization processing for an input signal so as to make an amplitude of an equalized output signal constant, the input signal being modulated by a modulation system that produces a modulation signal with constant amplitude characteristics. The control unit gradually changes equalization ability of the adaptive equalization processing of the adaptive filter in accordance with characteristics of the input signal.

Abstract: A real-time digital quadrature demodulation method and device for the ultrasonic imaging system are disclosed in this invention. In addition to a multiplying step and a filtering step, the method further comprises a sine and cosine table generating step for generating the sine and cosine table in real time, and a filter parameter generating step for generating corresponding filter parameters in real time to filter signals from the multipliers. The device comprises two multipliers, two filters, a sine and cosine table generating module, a filter parameter generating module, and two parameter memories. The real-time digital quadrature demodulation method and device for the ultrasonic imaging system according to the invention are capable of effectively saving the storage resource, and are easily controllable.

Abstract: Various embodiments of the present invention provide systems and methods for performing adaptive equalization. For example, various embodiments of the present invention provide methods for adaptive equalization that include providing a data processing system with an equalizer circuit (210) and a target filter circuit (265). The equalizer circuit performs equalization based at least in part on an equalizer coefficient (215). The methods further include generating an error (285) based upon a first output from the equalizer circuit and a second output from the target filter circuit. An inter-symbol interference component (295) is extracted from the error (285) and used to calculate an equalizer gradient (226). Based at least in part on the equalizer gradient (226), the equalizer coefficient (215) is calculated.

Abstract: A reconfigurable digital signal filter processing unit for use in a communication device is provided. The reconfigurable filters processor can implement different filter topologies to adapt to a range or wireless technology characteristics. The reconfigurable filter processor comprises a plurality of filter blocks whose inputs can be selected based on the desired configuration of the filter. Each filter block applies a transfer function to a received signal to achieve a desired filtering function.

Abstract: A signal enhancement system improves the understandability of speech or other audio signals. The system reinforces selected parts of the signal, may attenuate selected parts of the signal, and may increase SNR. The system includes delay logic, a partitioned adaptive filter, and signal reinforcement logic. The partitioned adaptive filter may track and enhance the fundamental frequency and harmonics in the input signal. The partitioned filter output signals may approximately reproduce the input signal, delayed by an integer multiple of the period of the fundamental frequency of the input signal. The reinforcement logic combines the input signal and the filtered signals to produce an enhanced output signal.

Abstract: A filter controller. In one embodiment, the filter controller includes a first mechanism for providing an input signal to an adjustable filter. A second mechanism measures a response of the adjustable filter to the input signal and provides a second signal in response thereto. A third mechanism sets one or more parameters of the adjustable filter in response to the second signal. In a more specific embodiment, the adjustable filter includes one or more sub-filters, such as a canceller filter, which may be any filter that employs one or more portions or versions of a signal to selectively cancel one or more portions or versions, such as frequency components, of the same signal.

Abstract: A method of processing an image includes traversing pixels of an image in a single pass over the image. An inverting function is applied to the pixels. A recursive filter is applied to the inverted pixel values. The filter has parameters which are derived from previously traversed pixel values of the image. A pixel value is combined with a filter parameter for the pixel to provide a processed pixel value for a processed image.

Abstract: An enhancement system extracts pitch from a processed speech signal. The system estimates the pitch of voiced speech by deriving filter coefficients of an adaptive filter and using the obtained filter coefficients to derive pitch. The pitch estimation may be enhanced by using various techniques to condition the input speech signal, such as spectral modification of the background noise and the speech signal, and/or reduction of the tonal noise from the speech signal.

Abstract: A modified Gram-Schmidt QR decomposition core implemented in a single field programmable gate array (FPGA) comprises a converter configured to convert a complex fixed point input to a complex floating point input, dual port memory to hold complex entries of an input matrix, normalizer programmable logic module (PLM) to compute a normalization of a column vector. A second PLM performs complex, floating point multiplication on two input matrix columns. A scheduler diverts control of the QRD processing to the normalizer PLM or the second PLM. A top level state machine communicates with scheduler and monitors processing in normalizer PLM and second PLM and communicates the completion of operations to scheduler. A complex divider computes final column for output matrix Q using floating point arithmetic. Multiplexer outputs computed values as elements of output matrix Q or R. Complex floating point operations are performed in a parallel pipelined implementation reducing latencies.

Abstract: A coefficient compensating unit 33 calculates based on a loop point and a cross-fade point of impulse response coefficient data renewed compensation impulse response coefficients specified for a cross-fade period defined between the cross-fade point and an end point of the impulse response coefficient data, and stores the calculated impulse response coefficients in a compensation coefficient memory 34, wherein the cross-fade point corresponds to a beginning point of the cross-fade period in which a cross-fading is performed to smoothly connect the end point with the loop point.

Abstract: Complex adaptive methods for complex information processing employ optimal individual convergence factors for real and imaginary components of the weight vector. For wireless receivers operating on QPSK, a Complex IA-ICA performs better than existing Complex Fast-ICA methods in terms of accuracy and convergence speed, can process such complex signals in time-varying channels, and employs time-varying and time-invariant convergence factors, independent for the real and imaginary components of the system parameters, and provide individual or group system parameter adjustments. Such systems employ the within complex adaptive ICA with individual element adaptation (Complex IA-ICA).

Abstract: The adaptive filtering techniques described herein allow a filter that is operating in a target domain to be trained in another domain, possibly with constraints, using the same adaptation framework used in a standard adaptive filter. As a result, the adaptation engine may be configured to run in a transform domain that is more desirable than the target domain. For example, the transform domain may be less susceptible to noise or may have more impact on the trained filter's desired results. The filter is trained in the transform domain and then the filter hardware is updated in the target domain.

Abstract: A receiving system and a method of processing broadcast signal are disclosed herein. The receiving system includes a signal receiving unit, a detector, and a channel equalizer. The signal receiving unit receives a broadcast signal including mobile service data and a data group including N number of training sequences. The detector detects N number of training sequences from the broadcast signal (wherein N?5), wherein the detected N number of training sequences are received during N number of training sections.

Abstract: A system and method are provided for efficiently switching a loop bandwidth using stored values in a digital filter of a phase-locked loop system. In a first timeslice, an input signal is digitally filtered using base coefficients multiplied by stored filter output and input values from previous timeslices. The filter output value is used to acquire the input signal frequency in a first bandwidth. In response to changes in the input signal frequency, the input signal is digitally filtered in a predetermined number of first intermediate period timeslices using transient coefficients multiplied by stored filter output and input values from previous timeslices. As a result, the first filter output value is maintained within a predetermined range. In a second timeslice, the input signal is digitally filtered using base coefficients multiplied by stored filter output and input values from previous timeslices to acquire the input signal frequency in a second bandwidth.

Abstract: Signal detectors are described herein. By way of example, a system for detecting signals can include a microphone signal detector, a loudspeaker signal detector, a signal discriminator and a decision component. When the microphone signal detector detects the presence of a microphone signal, the loudspeaker signal detector detects the presence of a loudspeaker signal and the signal discriminator determines that near-end speech dominates loudspeaker echo, the decision component can confirm the presence of doubletalk. When the microphone signal detector detects the presence of a microphone signal and the signal discriminator determines that near-end speech dominates loudspeaker echo, the decision component confirms the presence of near-end signal.

Abstract: A method for compressive domain filtering and interference cancelation processes compressive measurements to eliminate or attenuate interference while preserving the information or geometry of the set of possible signals of interest. A signal processing apparatus assumes that the interfering signal lives in or near a known subspace that is partially or substantially orthogonal to the signal of interest, and then projects the compressive measurements into an orthogonal subspace and thus eliminate or attenuate the interference. This apparatus yields a modified set of measurements that can provide a stable embedding of the set of signals of interest, in which case it is guaranteed that the processed measurements retain sufficient information to enable the direct recovery of this signal of interest, or alternatively to enable the use of efficient compressive-domain algorithms for further processing.

Abstract: Circuitry for receiving a serial data signal (e.g., a high-speed serial data signal) includes adjustable equalizer circuitry for producing an equalized version of the serial data signal. The equalizer circuitry may include controllably variable DC gain and controllably variable AC gain. The circuitry may further include eye height and eye width monitor circuitry for respectively producing first and second output signals indicative of the height and width of the eye of the equalized version. The first output signal may be used in control of the DC gain of the equalizer circuitry, and the second output signal may be used in control of the AC gain of the equalizer circuitry.

Abstract: A system includes a first powered apparatus having a first analog signal with a fundamental frequency; and a second apparatus providing load diagnostics or power quality assessment of the first apparatus from a second digital signal. The second apparatus includes an input of the first analog signal, an output of the second digital signal, a processor, an adaptive filter executed by the processor, a digital-to-analog converter, and an analog-to-digital converter. The adaptive filter routine outputs a third digital signal as a function of the second digital signal and plural adaptive weights. The digital-to-analog converter inputs the third digital signal and outputs a fourth analog signal representative of an estimate of a fundamental frequency component of the first analog signal. The analog-to-digital converter inputs a difference between the first and fourth analog signals, and outputs the second digital signal representative of the first analog signal with the fundamental frequency component removed.

Abstract: Systems and methods for optimizing operation of a transceiver device are disclosed. The method may include parallel processing an input signal through a first path having a first frequency response and a second path having a second frequency response. The second frequency response may be higher than the first frequency response. Signals from the first and second paths may be combined, creating an output signal having a desired gain and frequency. The parallel processing may adjust a gain of at least one of the first path and the second path. The parallel processing may equalize at least one of the first frequency response and the second frequency response. The input signal may be from a 10 GBit Ethernet channel and/or a Fibre channel.

Abstract: This invention is an improvement upon the basic adaptive signal processors, the Multi-Stage Wiener Filter and the cascaded canceller. The invention combines the concepts of soft weighting the adaptive weights of either type of processor disclosed herein with reiteratively processing the outputs by returning them to the input of the chosen adaptive signal processor. The combination of these functions improves the Signal to Interference plus Noise Ratio (SINR), the Probability of Detection (Pd), and/or the Bit Error Rate (BER). The invention improves statistical convergence of these types of metrics such that fewer training data samples are needed to achieve a particular satisfactory value of these metrics than would occur using traditional computational adaptive signal processors.

Abstract: A method includes transmitting a first signal over a network from a first communication link to a second communication link. The method further includes receiving a second signal with the first communication link from the second communication link. The method further includes canceling signal echo from the first signal present in the second signal with a digital echo canceller. The method further includes providing correction data from a memory array to the digital echo canceller during the cancellation of the signal echo. An associated apparatus is also disclosed.

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 filter circuit arrangement comprising a circuit configuration having non-linear transmission characteristics for equalizing an input signal is disclosed. In one embodiment, the circuit configuration having the non-linear transmission characteristics is provided in a feedback loop of the filter circuit arrangement.

Abstract: Systems and methods are provided for an adjustable filter engine. In particular, an electronic system is provided that can include a focus module, memory, and control circuitry. In some embodiments, the focus module can include an adjustable filter engine and a motor. By using the adjustable filter engine to generate a filter with a large number of filter coefficients, the control circuitry can accommodate a variety of system characteristics. For example, by generating a set of cumulative coefficients and re-arranging the order of the cumulative coefficients, the control circuitry can reduce the bit-width requirements of the adjustable filter engine hardware. For instance, the control circuitry can reduce the number of multipliers required to perform a convolution between an updated filter and one or more input signals. In some embodiments, the updated filter can be generated to reduce oscillations of the motor movement due to a new position request.