Abstract: Electronic component resource utilization for certain digital filters may be significantly reduced by using a method for determining a set of coefficient words using a smaller word size. The disclosed method and/or apparatus may be used to determine an initial set of coefficient words for a digital filter for a predetermined frequency, a predetermined quality factor (“Q”), and a predetermined sampling frequency, and determining a gain error value for the digital filter for the set of coefficient words. If the determined gain error value is greater than a predetermined threshold, the quality factor may be modified by multiple predetermined amounts. The set of coefficient words may be redetermined using the modified quality factors as often as necessary until the gain error drops below the predetermined threshold.

Abstract: A finite impulse response filter (FIR) for processing a communication channel. The FIR comprises a delay line, a tap processor and a summer. The delay line has “N” taps to successive portions of the communication channel. The delay line shifts the successive portions of the communication channel once in each symbol processing interval. The tap processor subjects each of the “N” taps to a first scaling utilizing first scaling coefficients associated with filtering the current symbol interval and further subjects at least one of the “N” taps to a second scaling by a second scaling coefficient associated with filtering the prior symbol interval. The summer generates in each symbol interval a filtered output comprising a sum of the “N” scaled taps from the first scaling in the prior symbol interval and the second scaling of the at least one tap in the current symbol interval, thereby increasing an order of the FIR without corresponding increase in an order of the delay line.

Abstract: A filter calculating device includes a first equalization filter calculating section that generates at least a first conversion matrix and a first triangular matrix based on a channel state of a first channel; a first quasi-orthogonalization section that calculates a first unimodular matrix based on the first triangular matrix; and a second equalization filter calculating section that generates at least a second conversion matrix and a second triangular matrix based on a channel state of a second channel and the first unimodular matrix.

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: A method of performing an infinite-impulse response digital filter includes switching address pointers between a first instance of the filter and a second instance of the filter; where the first and second instances represent the same filter. A first instance of the filter executes operations sequentially multiplying a current input data value, and first and second previous input data values, with corresponding ones of a first set of filter coefficients, using a multiplier; and a second instance of the filter executes operations sequentially multiplying first and second previous intermediate data values with corresponding ones of a second set of filter coefficients, using the multiplier. Switching between first and second instances of the filter occurs at each data input value or frame according to an alternating signal.

Abstract: A system and method for processing a signal with a filter employing FIR and/or IIR elements. The required controller function is decomposed into primary FIR and/or IIR elements and a compensation filter is provided to address the latency in the primary elements, which would result in undesired operation of the filter. Several configurations of suitable filters are discussed, including multi-rate filters and filters with reduced power requirements.

Abstract: Systems and methods utilize enhanced metrics for demodulation and/or soft bit information generation in the presence of a non-constant envelope modulated interfering signal. In one embodiment, a receiver includes a downconverter and a demodulator. The downconverter receives a radio frequency signal comprising a desired signal, noise, and a non-constant envelope modulated interfering signal, and downconverts the radio frequency signal to provide a downconverted signal. The demodulator demodulates the downconverted signal based on a demodulation metric that models the non-constant envelope modulated interfering signal as a stationary non-Gaussian random process with a probability distribution derived from a modulation constellation of a modulation used for the non-constant envelope modulated interfering signal. In one embodiment, the demodulator outputs demodulated symbols. In another embodiment, the demodulator outputs soft bit information.

Abstract: A system including a processor for adjusting the dynamic range of an image including a plurality of pixels. The processor segments the pixels into blocks, and computes statistical values for each block based on intensity values of the pixels. The processor also adjusts the dynamic range of the image by controlling the intensity values of the pixels based on the statistical values.

Abstract: A filter and method for filtering a signal are disclosed. The filter is equivalent to a plurality of bi-quad filters connected in series, and is implemented on a digital processor that receives a sequence of signal values at a sampling rate characterized by a sampling interval and generates a filtered signal value upon receiving each received signal value. The filter has a latency that is less than the sampling interval. The filtered values can be generated by adding a term to a received signal value and multiplying the sum by a gain constant that depends on the filter constants. The added term does not depend on the current received signal value. The filter can be implemented in fixed-point integer arithmetic.

Abstract: An infinite impulse response (IIR) filter is provided for receiving an input signal and outputting a filtered signal. The filter comprises feedback circuitry for feeding back said filtered signal, the feedback circuitry comprising a first delay element for delaying said filtered signal; and a sub-unit, for receiving said delayed filtered signal, for outputting a summed signal which is the difference between said delayed filtered signal and a further-delayed filtered signal, and for outputting a multiplied signal which is an inverted further-delayed filtered signal multiplied by a first filter coefficient. At least said input signal, said delayed filtered signal, said multiplied signal, and said summed signal are employed to generate said filtered signal.

Abstract: A method and system for the design and implementation of filters is presented in which the filter's transfer function can be provided with a significant insensitivity to the filter's tap coefficient values. A desensitized digital filter includes a first halfband filter and a second filter coupled in cascade between an input of the digital filter and the output of the digital filter. In embodiments, the first filter has the transfer function F(z)=K(1+z?1)(1+z?1) wherein K?0 is a scale factor. The digital filter may also interact with an up-sampler or a down-sampler. A desensitized Hilbert transformer includes an FIR filter having filter-tap coefficients whose absolute values equal the absolute values of the coefficients of an FIR filter F(z) for which the product (1+z?1)F(z) is a halfband filter coupled in cascade with a second filter.

Abstract: A method and system for the design and implementation of desensitized digital filters with droop correction. The desensitized digital filter includes a first filter configured to receive an input signal, a decimator or upsampler, and a modified desensitized half-band filter. The first filter introduces droop into the passband of the desensitized digital filter. The desensitized half-band filter has a transfer function F(z)=K(1+z?1)G(z) wherein K?0 is a scale factor, that is modified to omit a (1+z?1) factor block. The modified desensitized half-band filter compensates for the passband droop introduced by the first filter. The first filter may be a sinc filter, CIC filter, or filter having similar properties.

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: A finite impulse response filter comprises an input formatter, a plurality of sample registers, a plurality of coefficient registers, an arithmetic unit, a multiply accumulate unit, a crosspoint switch, an interpolator, a control unit, and an output formatter. The input formatter separates the in-phase portion of a complex-number discrete-time sample from the quadrature portion. The sample registers store a plurality of discrete-time samples. The coefficient registers store a plurality of coefficients. The arithmetic unit adds two of the discrete-time samples to create a sum. The multiply accumulate unit includes a multiplier that multiplies the sum by a coefficient to create a product, an adder that adds the product to a sum of products, and a register that stores the sum of products. The crosspoint switch allows communication between the first and second plurality of registers and the arithmetic unit and the multiply accumulate unit.

Abstract: A microseismic monitoring system includes a seismic sensor positioned proximate to a wellbore traversing a formation; an orientation source producing an orientation shot; a hydraulic apparatus operationally connected with the formation to produce a fracture in the formation; a computer control system operationally connected with a database of known spectral attributes for event categories; and a computer readable medium that carries instructions executable by the computer control system that, when executed: receive data from the seismic sensor; select an event of interest from the data received; determine a spectral estimate of the selected event of interest; compare the determined spectral estimate of the selected event of interest to the known spectral estimates; and select from the data received by the seismic source the orientation shot for orientation of the seismic sensor.

Abstract: One embodiment of the present invention relates to an analog correlation unit comprising a plurality of parallel correlation components configured to operating according to an advanced switched-capacitor low pass filter principle that increases coding gain of the unit. Each correlation component comprises a sampling stage and a correlation stage. The sampling stage may comprise a switched capacitor configured to sample a received baseband signal to determine a value (e.g., polarity) of the baseband signal. The sampled baseband signal is provided to the correlation stages, which may respectively comprise a plurality of switched integrators configured to selectively receive and integrate the sampled baseband signal over time depending upon values (e.g., polarity) of the correlation code to generate voltage potential values. The analog correlation result is evaluated by a comparison of an adjustable threshold voltage with the difference between the output voltage potential values.

Abstract: A method and device are provided for filtering digital audio signals using at least one ARMA filter, particularly during a filter change. The method includes the following steps: a step of receiving a first request to change filtering to or from filtering by a first ARMA filter; and, in response to the first request, a step of gradually switching, at each of a plurality of cascaded first filtering blocks, between digital-signal filtering by a first basic filtering cell and digital-signal filtering by another associated basic filtering cell, the first basic filtering cells of the plurality of first filtering blocks factorizing the first filter.

Abstract: Various embodiments of the present invention provide systems and methods for data filter tuning. As an example, a method for filter tuning is disclosed that includes: providing a tunable filter having an operation filter and a calibration filter; applying a low frequency test input to the operation filter in place of an input signal to yield a first filter output; calculating a low frequency magnitude value corresponding to the first filter output; applying a high frequency test input to the operation filter in place of an input signal to yield a second filter output; calculating a high frequency magnitude value corresponding to the second filter output; modifying a tuning factor of the calibration filter when a ratio of the high frequency magnitude value and the low frequency magnitude value is outside of a defined range; and storing the tuning factor of the calibration filter when the ratio of the high frequency magnitude value and the low frequency magnitude value is within the defined range.

Abstract: A digital signal processing circuit comprises a band selector (14) for selecting at least one sub-band from a frequency spectrum of a digital sampled input signal. The band selector (14) comprises a plurality of processing branches corresponding to respective phases and an adder (28a, 28b) for adding branch signals from the branches. Each branch comprises a sub-sampler (20a,b) for sub-sampling sample values of the input signal at the phase corresponding to the branch, a filter (24a,b) with a first FIR filter (32, 34), applied alternatingly to sets of even and to sets of odd samples from the subsampler (20a,b) and a second FIR filter (36, 38) applied to further sets of odd and even samples from the subsampler (20a,b) when the first FIR filter is applied to the even and odd sets respectively.

Abstract: For subband decomposition of a d-dimensional input signal (S) into a number K of subband components (F1-F4), a filter bank has a filtering module (801) transforming the input signal (S) into 2d components including a low-frequency component (L) and 2d?1 higher-frequency components (F1), The 2d?1 higher-frequency components are oversampled, typically by a factor 2, compared to the low-frequency component. The low-frequency component can be further decomposed by means of another filtering module having a similar structure, and the process can be iterated over any number of scales. The reconstruction filter bank has a symmetric structure, with filtering modules adapted to the oversampling of the higher-frequency components. Such filter banks are well suited to various enhancement processing applied to the subband components such as thresholding, reduction of compression distortion, reduction of measurement noise, sharpness enhancement.

Abstract: Filter banks may have different structures and different individual output signal domains. Often a translation between different filter bank domains is desirable. Usually, mapping matrices are used that, however, vary over frequency. This requires a significant amount of lookup tables. A method for transforming first data frames of a first filter bank domain to second data frames of a different second filter bank domain, comprises steps of transcoding sub-bands of the first filter bank domain into sub-bands of an intermediate domain that corresponds to said second filter bank domain but has warped phase, and transcoding the sub-bands of the intermediate domain to sub-bands of the second filter bank domain, wherein a phase correction is performed on the sub-bands of the intermediate domain.

Abstract: An acquired signal indicative of electrophysiological activity is filtered using both a wavelet filter and either a notch filter or a band-pass filter to eliminate noise or interference, such as power line interference. A wavelet transform is used to transform the acquired signal into the wavelet domain, where a wavelet filter is applied to extract a soft component (e.g., a component with small wavelet coefficients). A filter, such as a notch filter or a band-pass filter, is applied to the soft component in order to isolate an interference signal. The interference signal is used to produce an output signal representing the acquired signal filtered to eliminate the interference signal. For example, the interference signal may be subtracted from the acquired signal. Alternatively, the output signal may be reconstructed from respective hard and soft components of the acquired signal as transformed into the wavelet domain.

Abstract: A method for filtering a signal is proposed. A noisy input signal is continuously examined in order to determine whether the input signal it is within or outside a deadband. The deadband width and the zero point of the deadband are continuously adapted to the noise power of the input signal depending on the time behavior of the input signal and a predefined system time constant. At least one filtered output signal is continuously output, such as a deadband signal, which substantially corresponds to a smoothed input signal.

Abstract: The present invention is directed to sub-filtering FIR (frequency impulse response) to provide the capabilities of an ambiguity function (i.e., search for a signal in time and frequency) without extensive computations. By minimizing the resources used for the signal search (increased efficiency), the size of the implementation of the ambiguity function in hardware, and thus its power consumption, can be reduced. Additionally, by making the frequency search more efficient, larger scale frequency searches are possible.

Abstract: The invention addresses the problem of parameter optimization for best filter performance and, in particular, the influence from the requirements on radio or fiber to radio repeaters utilizing those filters, that often proves to be conflicting for an FIR filter. The FIR filters are implemented in a programmable circuit and are not thereby restricted for use in communication repeaters although this particular usage may put the most serious restrictions on the filter performance. Within the imposed constraints, this disclosure illustrates a method to strike a middle ground while minimizing the trade-offs. The advantage of the concept presented allows the choice of a suitable filter pertaining to a particular traffic configuration, meaning a particular choice of individually filtered frequency bands set at different gain and intended to support a diversity of traffic formats. The disclosed approach banks on the reconfigurable variable length FIR filter architectures.

Abstract: Embodiments are directed to efficient frequency-domain implementations of time-varying FIR filters. More specifically, time-varying FIR filters according to embodiments exploit the duality of the fast Fourier transform that windowing in the time domain equals convolution in the frequency domain. In one embodiment, convolution of the output of the FIR filter and a desired windowing function is performed in the frequency domain instead of taking the output of the FIR filter in the frequency domain, converting this output the time domain via an IFFT, and then windowing this output in the time domain before again converting back to the frequency domain. As long as the windowing function has certain characteristics, then the time-varying FIR filter is computationally efficient and introduces minimal audible artifacts into the output of the filter. Concepts described herein are discussed in terms of audio signals and systems but are not limited to audio signals and systems.

Abstract: To obtain a high-quality enhanced signal, disclosed is a signal processing apparatus including a transform unit that transforms a mixed signal in which a first signal and a second signal coexist, into a phase component and a magnitude component or power component for each frequency, a first control unit that replaces the phase component of a predetermined frequency, a second control unit that modifies the magnitude component or power component of the predetermined frequency in accordance with the amount of a change of the magnitude component or power component that arises from replacement by the first control unit, and a reconstruction unit that reconstructs the phase component replaced by the first control unit and the magnitude component or power component modified by the second control unit.

Abstract: An apparatus and method for frequency division and filtering are provided. The apparatus includes a memory unit, an extrema calculation unit, and an envelope calculation unit. The memory unit is for storing sample data. The extrema calculation unit is for outputting and storing a number of maximum values and a number of minimum values to the memory unit according to the sample data. The envelope calculation unit is for calculating a mean envelope according to the maximum values and the minimum values, wherein within a duration when the envelope calculation unit respectively calculates an upper envelope and a lower envelope according to the maximum values and the minimum values, the envelope calculation unit outputs a value of the mean envelope to the memory unit according to a value of the upper envelope and a value of the lower envelope with respect to a corresponding identical address.

Abstract: Provided is a signal processing device performing filter processing operations on an input signal using a plurality of filters to generate an output signal, including a filter processing unit configured to perform a filter processing operation on the input signal using a filter and a generating unit configured to generate the output signal by adding a correction value determined on the basis of each sign and each absolute value of each difference between each of filtered output signals obtained by the filter processing operations performed using a plurality of the above mentioned filter processing units and the input signal, to the input signal.

Abstract: An equalizer for equalizing an input signal includes an infinitive impulse response (IIR) filtering portion for filtering the input signal to produce N filtered outputs; a gain-adjusting portion coupled to the IIR filtering portion with N gains for adjusting the N filtered outputs to produce N gained outputs, respectively; and an adder for summing the N gained outputs to generate an equalized output signal. N is an integer larger than 2.

Abstract: A receiving circuit, use, and method for receiving an encoded and modulated radio signal is provided. The circuit comprise a demodulator and a digital filter connected downstream of the demodulator for moving averaging. The filter has at least two FIFO registers and subtractors. Whereby for subtracting an output value of the FIFO register from an input value of the FIFO register a subtractor is connected to each FIFO register. Wherein the filter has a weighting unit, which is connected downstream of each FIFO register, and wherein the filter has an integrator, which is connected downstream of the subtractors for integration.

Abstract: A signal decimating system decimates an initial data signal having an initial data rate R to a final data signal having a final data rate R? in two stages, using a base decimation factor N and a decimation multiplier factor P. In the first stage, N FIR filters having coefficients corresponding to the final data rate R? condition the initial data signal using the final data rate coefficients and thereafter decimate the initial data signal, as conditioned, by a base decimation factor of N to generate an intermediate data signal having an intermediate data rate R?, where R ? = R N . In the second stage, a sub-sampling unit includes a switch that sub-samples the intermediate data signal at a sub-sampling rate P to generate a final data signal having a final data rate R?, where R ? = R ? P = R ( NxP ) .

Abstract: Enhancements to hardware architectures (e.g., a RISC processor or a DSP processor) to accelerate spectral band replication (SBR) processing are described. In some embodiments, instruction extensions configure a reconfigurable processor to accelerate SBR and other audio processing. In addition to the instruction extensions, execution units (e.g., multiplication and accumulation units (MACs)) may operate in parallel to reduce the number of audio processing cycles. Performance may be further enhanced through the use of source and destination units which are configured to work with the execution units and quickly fetch and store source and destination operands.

Abstract: A computer-implemented method for analyzing multivariate data comprising a plurality of samples of each of a plurality of measurement variables is disclosed. The method comprises, for a first subsetA (X) of the multivariate data X, determining (110) a first projection score related to the first subset. Furthermore, the method comprises, for a second subsetB (X) of the multivariate data X, determining (120) a second projection score related to the second subset. Moreover, the method comprises, comparing (130) the first and the second projection score for determining which one of the first and the second subset provides the most informative representation of the multivariate data, which is defined as the one of said subsets having the highest related projection score. A definition of the projection score is also provided.

Abstract: In described embodiments, a Floating Tap, Feed Forward Equalizer (FT-FFE) achieves performance comparable to a full size, long FFE when equalizing wire line channels in, for example, SerDes receivers. A FT-FFE might be employed as a standalone datapath equalizer, or might be employed in conjunction with other equalization techniques.

Abstract: A programmable integrated circuit device can be configured as a cascaded integrator-comb (CIC) filter. In order to take advantage of Hogenauer pruning to configure the CIC filter efficiently, a software tool for configuring the device can be provided in which the Fj terms for Hogenauer pruning have been calculated in advance for all possible user parameters supported by the tool. To configure a CIC filter, the user enters the parameters in the tool, which then looks up the Fj terms corresponding to those parameters and completes the calculation of the Bj terms for Hogenauer pruning. Because the calculation of the Fj terms is the most time-consuming step in calculating of the Bj terms, pre-calculation of the Fj terms, which can be done just once by the provider of the tool, allows end users to calculate the Bj terms in reasonable periods of time, making Hogenauer pruning available to end users.

Abstract: The present invention relates to nonlinear signal processing, and, in particular, to adaptive nonlinear filtering of real-, complex-, and vector-valued signals utilizing analog Nonlinear Differential Limiters (NDLs), and to adaptive real-time signal conditioning, processing, analysis, quantification, comparison, and control. More generally, this invention relates to methods, processes and apparatus for real-time measuring and analysis of variables, and to generic measurement systems and processes. This invention also relates to methods and corresponding apparatus for measuring which extend to different applications and provide results other than instantaneous values of variables. The invention further relates to post-processing analysis of measured variables and to statistical analysis.

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 method for calibrating coefficients of an observer of a variable state of a physical system from measurements of physical quantities of the system, at different instants includes measuring a variable of the physical system at several instants, the variable being a function of a system state variable, and determining a vector of coefficients that minimizes a sum of the number of measurements of a square of a norm of a vector that is the difference between the measured variable and a function of the system state variable and the vector of coefficients. The minimization is subject to a constraint that the trajectory of the measured variable be within a corridor of uncertainty on either side of a trajectory of its estimate, at least for the measurement instants.

Abstract: A PAM-N decision feedback equalizer (DFE) comprises a coefficient computation unit; a feedback unit that mitigates, using computed feedback coefficients, effects of interference from data symbols; an error-and-decision unit for at least computing a least error value respective to one of a plurality of decision levels, wherein the least error value indicates a difference of a pseudo equalized input PAM-N data symbol from an optimal position of the one of the plurality of decision levels, wherein the one of the plurality of decision levels corresponds to a modulation level used to modulate data in the input PAM-N data symbol; and a calibration unit for adaptively setting the plurality of decision levels based, in part, on the least error value, thereby enabling for compensating for gain changes resulted by a cable on which the input PAM-N data symbol is received and further compensating for embedded offsets of the error-and-decision unit.

Abstract: An equalization filter is provided for solving the problem in which there is a limited range in which compensated for distortion of a transmission signal can be made. Measuring instrument 104 measures a distortion quantity which characterizes distortion of the transmission signal. Comparator 105a generates a differential signal which indicates the difference between the transmission signal and a compensation signal. Delay device 105b delays the differential signal based on the distortion quantity measured by measurement instrument 104 and generates the compensation signal.

Abstract: Various embodiments of the present invention provide systems and methods for sync mark detection. As an example, a sync mark detection circuit is discussed that includes a storage circuit, a plurality of noise predictive filter circuits, and a controller circuit. The storage circuit is operable to store a data input as a stored input. The plurality of noise predictive filters are operable to receive a processing input. At least one of the noise predictive filters is selectably modifiable to either increase the probability of finding a sync mark in the processing input or to maintain a baseline probability of finding the sync mark in the processing input. The controller circuit is operable to determine an operational mode that may be a standard operational mode, a bit flipping mode, or a filter modification mode.

Abstract: A filter system with infinite impulse response is provided. The filter system has a transfer function that includes at least one pair of first order polynomial fractions. In one embodiment, the poles and/or the zeros of the pair of polynomial fractions are complex conjugates, respectively.

Abstract: A system and method are provided for filtering noise from a pulsed input signal comprising cyclically producing a change in an output signal only if changes in an input signal occur at least a desired time after a respective immediately previous change in the input signal, and otherwise rejecting the changes in the input signal; and counting the rejected changes in the input signal. More than one duration or frequency may be used for the filtering, enabling classification of noise by frequency. Resulting counts may be used to determine rates of occurrence of noise for evaluation of performance of equipment, installation of the equipment, and changes in performance over time.

Abstract: Various embodiments of the present invention provide systems and methods for data filter tuning. As an example, a method for filter tuning is disclosed that includes: providing a tunable filter having an operation filter and a calibration filter; applying a low frequency test input to the operation filter in place of an input signal to yield a first filter output; calculating a low frequency magnitude value corresponding to the first filter output; applying a high frequency test input to the operation filter in place of an input signal to yield a second filter output; calculating a high frequency magnitude value corresponding to the second filter output; modifying a tuning factor of the calibration filter when a ratio of the high frequency magnitude value and the low frequency magnitude value is outside of a defined range; and storing the tuning factor of the calibration filter when the ratio of the high frequency magnitude value and the low frequency magnitude value is within the defined range.

Abstract: [Objective] To provide a digital filter circuit and a digital filter control method which are capable of reducing circuit scale and power consumption for filter processing in a frequency domain such as an overlap FDE method.

Abstract: A system comprises a system interface to receive one or more instruction sets from a microcontroller and to receive digital data to be processed. The system further comprises a controller that is reconfigurable according to the one or more instruction sets received by the system interface. The system further comprises a data path device to perform digital filtering operations on the digital data as directed by the controller according to the reconfiguration of the controller by the one or more instruction sets.

Abstract: A digital equalizer with a reduced number of multipliers for correction of the frequency responses of an interleaved ADC is disclosed. An exemplary interleaved analog to digital converter with digital equalization includes a composite ADC including M time interleaved sub-ADC, a demultiplexer, samples repositioning unit, a first PreFIRs transformer, a second PreFIRs transformer, K double buffer FIR filters, a PostFIRs transformer, a samples sequence restoration unit, and a multiplexer, coupled in series and providing an equalized, frequency response-corrected output.

Abstract: In described embodiments, a variety of down-sampling techniques are employed to generate a more constrained set of floating-tap positions when compared to floating-tap Decision Feedback Equalization (DFE) architectures that allow unconstrained 1T resolution or separated floating-tap positions. Down-sampling is employed to constrain the floating-tap positions rather than positions occurring with 1T resolution or spacing. Two broad down-sampling techniques, phase pruning and phase amalgamation, are applied to a variety of exemplary DFE implementations. Although the tap positions are more constrained, the architectures select floating-tap positions containing dominant reflection inter-symbol interference (ISI) terms.

Abstract: A rectangular wave for determining a range of a weight function is transformed to frequency domain by an FFT or the like, and after being multiplied by a window function (BlackmanHarris window function, for example) generated on a frequency axis by a multiplier, the frequency domain is transformed again to the time domain by an IFFT or the like thereby to generate a weight function.