Abstract: The present invention is directed toward a Finite Impulse Response (FIR) no-multiply filter (NMF), which replaces complex multiplications with phase additions. At each tap in the FIR filter, only phases are accumulated and at the output the complex result is reconstructed in I/Q. Noise dither is relied upon to smooth the digitized phase resolution. The NMF is ideally suited to a matched filtering scenario for constant modulus signals.

Abstract: A system and method for filtering an input signal with at least a first and a second octave portions is presented. A proportional bandwidth filter system includes a bandwidth reducing filter, a down-sampler, and first and second octave filters. The bandwidth reducing filter reduces the bandwidth of the input signal and the down-sampler rolls the second octave portion represented in the reduced bandwidth signal to a top octave portion of a down-sampled signal. The first and second octave filters are comprised of a plurality of proportional bandwidth filters. The first octave filter partitions and converts the first octave portion of the input signal into output signals representing the frequency spectra of the first octave of the input signal. Similarly, the second octave filter generates outputs representing the spectra of the second octave portion of the input signal represented as the top octave portion of a down-sampled signal.

Abstract: Disclosed is a filter circuit, comprising a signal to be filtered, a difference circuit coupled to the signal to be filtered, a filter having an input coupled to the difference circuit, an integrator (or accumulator) having a first input coupled to an output of the filter circuit, and having a second input, and an accumulator coupled to an output of the integrator. A method of filtering is described also.

Abstract: The invention relates to an adaptive method of extracting at least of desired electro magnetic wave signals, sound wave signals (40, 42), and any other signals from a mixture of signals (40, 42, 44, 46) and suppressing noise and interfering signals to produce enhanced signals (50) corresponding to desired (10) signals, and an apparatus (70) therefore. It relies on the concept of at least one of an attenuation of input signals in each sub-band for signals in such a manner that all desired (10) signals are attenuated less than noise or interfering source signals, and/or an amplification of input signals in each sub-band for source signals in such a manner that all desired (10) signals are amplified, and that they are amplified more than the noise and interfering signals.

Abstract: A filter and method of filtering modifies the computation order to accommodate horizontal symmetric filtering, and modifies the source operands while modifying the single instruction multiple data (SIMD) computation, so as to eliminate such heavy overhead of transposing a pixel matrix. The filter and method of filtering reformats the equations involved in the prior art to the following equations, thereby acquiring the interpolation results by reducing the required clock cycles to three cycles: acc=a0*(|p0+p5|p1+p6|p2+p7|p3+p8|) acc=a1*(|p1+p4|p2+p5|p3+p6|p4+p7|)+acc acc=a2*(|P2+P3|P3+P4|P4+P5|P5+P6|)+acc.

Abstract: The invention relates to a method and filter structure for filtering an input signal, which is applied to an input (1) of the filter structure, having a symbol rate. The input signal is guided to multiple parallel branches of the filter structure connected to the input (1), in which at least two sequence holding members (4) are each disposed. Symbol values of the input signal, which are applied to the outputs of the sequence holding members (4) at certain points are guided to multiple (N) parallel signal addition units (3) having weighted inputs. One output signal each is generated in the signal addition units (3) as the sum of the weighted symbol values that are applied to the inputs. According to the invention, space and energy saving filter structure is provided, wherein: the same amount of sequence holding members (4.1, 4.2, 4.3) is disposed in each of the parallel branches; as many signal addition units (3.1, 3.2, 3.3, 3.

Abstract: An information processing device for processing reception signals converted into digital signals, includes: a first conversion unit for executing sampling rate conversion of each of the digital signals to be computed with each tap coefficient of a K'th-order FIR filter; a filter computing unit for executing computation processing of the K'th order FIR filter on K digital signals each of which have been subjected to sampling rate conversion by the first conversion unit; and a control unit for controlling sampling rate conversion of the digital signals by the first conversion unit, and the computation processing of the K'th order FIR filter by the filter computing unit.

Abstract: A first stage of a digital filter receives input data to be filtered, the first stage of a digital filter operating at a first clock; a second stage of the digital filter outputs filtered output data, the second stage of the digital filter operating on a second clock, wherein a ratio of a frequency of the first clock and a frequency of the second clock is a fractional number, and a frequency of the second clock is higher than a frequency of the first clock; the first stage receives an indication of a ratio of the first clock and the second clock; and the first stage receives an indication of a time offset between (1) a clock pulse of the second clock, which occurs between a first clock pulse and a second clock pulse of the first clock, and (2) the first clock pulse of the first clock.

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 method for long impulse response digital filtering of an input data stream, by use of a digital filtering system. Where the input data stream is divided into zero-input signals and zero-state signals. One of the zero-input signals and a corresponding impulse response of the digital filtering system is converted to the frequency domain to determine a respective zero-input response of the digital filtering system. One of the zero-state signals is convolved with a corresponding impulse response of the digital filtering system to determine a respective zero-state response of the digital filtering system, wherein at least part of the zero-input signal precedes the zero-state signal. The zero-state response of the digital filtering system is added to the zero-input response of the digital filtering system to determine the response of the digital filtering system. Apparatus for effecting this method is also disclosed.

Abstract: The present invention provides a method and device for generating a filter coefficient in real time. The method includes: looking up a converted window function value in a converted window function table based on a current coefficient index; generating a current cut-off angular frequency; generating a look-up table address based on the current coefficient index, the filter order and the current cut-off angular frequency and looking up a sine value in a sine table based on the look-up table address; and multiplying the converted window function value by the sine value to obtain the filter coefficient. The device includes a first memory, a second memory, a look-up table address generation module and a first multiplier. The present invention is easily implemented with low hardware resource consumption and high flexibility, and is particularly applicable for the hardware implementation of high-order finite impulse response filters.

Abstract: A method of computing a vector angle by using a CORDIC and an electronic apparatus using the same are disclosed. The electronic apparatus mainly includes a phase error detector, a loop filter, a small-area iteration LUT module and a phase compensation circuit. The phase error can be locked by using the error function in the phase error detector, and even the phase error can be locked to the minimum so that the error oscillates up-and-down about the zero level. The first transfer function in the loop filter can determine the baseband and the converging speed. Moreover, if the shifting technique is used, the operation of the first transfer function is speeded up. By using a phase-locking loop in association with looking up the above-mentioned LUT, the method is able to get fast converging and higher accuracy for the computation.

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 Non-Continuous Orthogonal Frequency Division Multiplexing (NC-OFDM) system may be provided. The NC-OFDM system, including: a sensing unit to determine whether to use at least one frequency band; a band control unit to disable a subcarrier with respect to a frequency band in use as a result of the determination; a pruning determination unit to determine whether a number of points where the subcarrier is disabled is greater than a threshold value; a pruning retrieval unit to retrieve a pruning path generated by the disabling of the subcarrier, when the number of points is greater than the threshold value; and a Fourier transform unit to perform a Fast Fourier transform (FFT) or Inverse FFT (IFFT) by applying the pruning path to the subcarrier.

Abstract: An equalizer circuit receives digital amplitude data A[N] which represents the amplitude level of the N-th (N is a nonnegative integer) signal to be transmitted via a transmission line and timing data T[N] which represents the cycle of the signal, and performs waveform shaping. The equalizer circuit includes: M (M is an integer) calculation units ECU1 through ECUm; and an adder ADD1 which adds the output data of the M calculation units ECU1 through ECUM and the amplitude data A[N] together so as to generate equalized amplitude data D[N]. A step response waveform RSTEP(t) for the transmission line is approximated by Expression RSTEP(t)=SSTEP(t)·(1?Sj=1:M fj(t)) using M (M is an integer of 2 or more) functions fj(t) (1?j?M) and a step waveform SSTEP(t) with the time t as an argument. The representative value of the function fj(t) in a range between T1 and T2 is represented by a function gj(T1, T2).

Abstract: A novel and useful apparatus for and method of software based phase locked loop (PLL). The software based PLL incorporates a reconfigurable calculation unit (RCU) that is optimized and programmed to sequentially perform all the atomic operations of a PLL or any other desired task in a time sharing manner. An application specific instruction-set processor (ASIP) incorporating the RCU includes an instruction set whose instructions are optimized to perform the atomic operations of a PLL. The RCU is clocked at a fast enough processor clock rate to insure that all PLL atomic operations are performed within a single PLL reference clock cycle.

Abstract: A method for implementing a digital filter is provided. The method includes (a) determining a bit-width of an incoming data sample of an incoming signal by measuring a distance between a leading zero or one of the incoming data sample and a trailing zero of the incoming data sample. The incoming data sample is obtained by sampling the incoming signal at a pre-defined time interval, (b) obtaining bit-width multipliers with variable bit-widths based on a first probability distribution function (PDF) of bit-widths of incoming data samples, (c) allocating the incoming data sample and a filter coefficient based on the bit-width of the incoming data sample and a bit-width of the filter coefficient to one bit-width multiplier of the bit-width multipliers, and (d) performing a multiply operation of a Multiply and Accumulate (MAC) operation on the one bit-width multiplier to generate an output of the digital filter.

Abstract: A digital filter design algorithm is implemented directly within a process control field device or other process related equipment. Filter design parameters are exposed so that filter design parameter values may be provided to the digital filter design algorithm so that the digital filter design algorithm may calculate digital filter coefficients for a digital filter having desired frequency response characteristics. The digital filter design parameter values may be provided by a user, or may be provided as process variable data output from a process control field device or other process related equipment. Once the coefficients of the digital filter having the desired frequency response characteristics have been calculated, the digital filter may be applied to process variable data received by the process control field device or other process related equipment.

Abstract: A method of modeling a room impulse response according to an embodiment of the invention includes receiving a sound pressure signal that is obtained by a microphone when an impulse-type sound source is excited and detecting a room impulse response; determining boundaries between a plurality of intervals of the room impulse response such that the room impulse response is divided into the plurality of intervals on a time domain; and modeling the room impulse response for each of the plurality of divided intervals using at least two different modeling schemes.

Abstract: A dynamic weight processing system. The inventive system includes a first circuit for receiving an input signal and a second circuit for filtering the input signal with dynamic weights to provide a weighted signal. In an illustrative embodiment, the dynamic weights are finite impulse response filter correlation coefficients that are dynamically generated based on a pseudo-noise code. The system may also include a dynamic weight generator that generates the dynamic weights by combining weight values stored in a lookup table in a manner dependent on the pseudo-noise code. The weighted signal may be further processed to generate nulling and beamsteering weights for the input signal. In a more specific implementation for a GPS (Global Positioning System) application, the received signal is partitioned into space frequency adaptive processing (SFAP) bands and space time adaptive processing (STAP) is performed within the SFAP bands.

Abstract: A digital communication system, an indoor unit, and an outdoor unit in which characteristic variation due to temperature is small are provided. The digital communication system comprises an IDU 1 and an ODU 6 connected through a cable 8, in which the IDU 1 includes a slope equalizer 2 which applies a given frequency characteristic to a baseband signal, a DAC 3 which converts the baseband signal from digital form into analog form, and an AGC circuit 5 which amplifies the baseband signal converted from digital form into analog form with a given amplification factor and outputs the signal to the ODU 6, and the ODU 6 includes a power detection circuit 7 which detects a length of the cable 8, determines a frequency characteristic to be applied to the baseband signal and an amplification factor according to the detected length of the cable 8 and notifies the determined frequency characteristic and amplification factor to the slope equalizer 2 and the AGC circuit 5 respectively.

Abstract: A transceiver having a multistage channel filter for a wireless communication system is disclosed. The transceiver is composed of a transmitter unit including a multistage channel filter module to perform channel filtering on an input signal having a sampling rate corresponding to a channel bandwidth of the wireless communication system by means of sampling-rate conversion through multiple stages, and a receiver unit corresponding to the transmitter unit.

Abstract: A method of generating parameters for a predistortion circuit in an integrated circuit using a matrix is disclosed. The method comprises storing a first column of a first matrix; generating the remaining columns of the first matrix based upon the first column of the matrix; generating a plurality of rows of a second matrix by performing a first set of calculations; and generating the remaining rows of the second matrix by selectively shifting the first rows of the second matrix.

Abstract: An apparatus, a computer readable medium to instruct a process to implement a method, and a method of reducing blocking artifacts and/or noise in an image that has been compressed by a block-based encoding process. The method includes deblocking the image using overlapped forward directional transforms, the overlapped forward directional transforms including directional selecting to filter across horizontal and vertical boundaries, denoising the image, in one version edge detecting to classify individual pixels as to whether or not they belong to edge features, and generating output pixels by locally adapting to whether the pixels are in block boundaries, and/or include edge features.

Abstract: A signal separating device includes an iterative estimator, a repeating calculator, a result output unit, and a repetition controller. The repeating calculator repeatedly causes the iterative estimator to iteratively perform independent component analysis on an observed signal matrix, and to further perform independent component analysis on the source signal matrix obtained as a result. The result output unit outputs the product of the respective mixing matrices obtained during each repetition as a mixing matrix with respect to the observed signal matrix, while also outputting the source signal matrix obtained during the final repetition as a source signal matrix with respect to the observed signal matrix. The repetition controller causes the repeating calculator to repeat the calculation control until all mixing matrices and all source signal matrices satisfy a convergence condition. The iterative estimator may perform a fixed number of iterations, or perform iterations until convergence is obtained.

Abstract: An apparatus for processing a plurality of real-valued subband signals using a first real-valued subband signal and a second real-valued subband signal to provide at least a complex-valued subband signal comprises a multiband filter for providing an intermediate real-valued subband signal and a calculator for providing the complex-valued subband signal by combining a real-valued subband signal from the plurality of real-valued subband signals and the intermediate subband signal.

Abstract: A signal processing apparatus includes a first filter on an in-phase signal channel; a second filter on a quadrature signal channel; a plurality of filter stages having each of more than one signal paths crossing each other which connects the first filter and the second filter; and at least more than one of the filter stages of more than one of a plurality of the filter stages includes a switching circuit disconnecting more than one of the signal paths and a correction unit correcting direct current offsets of the first filter and the second filter by using the switching circuit.

Abstract: A signal processing apparatus includes a signal output unit for outputting a measurement signal, the measurement signal being produced by synthesizing a signal composed of a concatenation of 2d period signals with a sinusoidal signal, each period signal having a time-domain waveform period being 2n samples, the sinusoidal wave having a wave count within the concatenation period of 2d period signals being other than an integer multiple of 2d, and n and d being respectively natural numbers, and an analyzing unit for frequency analyzing a response signal obtained as a result of picking up the measurement signal output from the signal output unit.

Abstract: A signal acquisition system has a signal acquisition probe having probe tip circuitry coupled to a resistive center conductor signal cable. The resistive center conductor signal cable is coupled to a compensation system in a signal processing instrument via an input node and input circuitry in the signal processing instrument. The signal acquisition probe and the signal processing instrument have mismatched time constants at the input node with the compensation system having an input amplifier with feedback loop circuitry and a compensation digital filter providing pole-zero pairs for maintaining flatness over the signal acquisition system frequency bandwidth.

Abstract: The present invention relates to a method for creating a Markov process that generates sequences. Each sequence has a finite length L, comprises items from a set of a specific number n of items, and satisfies one or more control constraints specifying one or more requirements on the sequence. The method comprises the steps of receiving data defining an initial Markov process of a specific order d and having an initial probability distribution and of receiving data defining one or more control constraints. The method further comprises the step of generating data defining intermediary matrices, each matrix being of dimension nd by n, by zeroing out transitions in the initial Markov process data that are forbidden by the one or more control constraints.

Abstract: In a transmission diversity control method in which a request to start or stop transmission diversity is issued by a reception station to a transmission station having a transmission diversity function, the reception station is provided with an equalizer to reduce multipath interference on the basis of a signal correlation, a transmission diversity start request is issued from the reception station to the transmission station when an effect generated by the equalizer is small, and a transmission diversity stop request is issued from the reception station to the transmission station when the effect generated by the equalizer is large.

Abstract: Methods for generating realistic waveforms with controllable voltage noise and timing jitter in a computer-based simulation environment and the simulation of a subset of those waveforms with system elements along the signal path is disclosed. By deriving a generic, re-useable, parameterized Fourier series, time-domain clock and pseudo-random data signals are generated from a subset of their true harmonic components. Time-domain signal parameters including high, low, and common-mode voltage levels, transition slew-rates, transition timing, period and/or frequency, may be designated by the user, and the computer calculates the harmonic components that will combine through the inverse Fourier transform to provide the required time-domain characteristics. By computing the frequency content of the signal directly it is possible to simulate the interaction of the signal with various system blocks while remaining in the frequency domain, thereby reducing simulation time and memory requirements.

Abstract: In signal processing using digital filtering the representation of a filter is adapted depending on the filter characteristics If e.g. a digital filter is represented by filter coefficients for transform bands Nos. 0, 1, . . . , K in the frequency domain, a reduced digital filter having coefficients for combined transform bands, i.e. subsets of the transformed bands, Nos. 0, 1, . . . , L, is formed and only these coefficients are stored. When the actual filtering in the digital filter is to be performed, an actual digital filter is obtained by expanding the coefficients of the reduced digital filter according to a mapping table and then used instead of the original digital filter.

Abstract: The present invention provides a method and device for generating a filter coefficient in real time. The method includes: looking up a converted window function value in a converted window function table based on a current coefficient index; generating a current cut-off angular frequency; generating a look-up table address based on the current coefficient index, the filter order and the current cut-off angular frequency and looking up a sine value in a sine table based on the look-up table address; and multiplying the converted window function value by the sine value to obtain the filter coefficient. The device includes a first memory, a second memory, a look-up table address generation module and a first multiplier. The present invention is easily implemented with low hardware resource consumption and high flexibility, and is particularly applicable for the hardware implementation of high-order finite impulse response filters.

Abstract: A control system is provided for controlling a transducer array. The control system includes a plurality of sparse filters and integrators. The control system can be used to separate signals from a plurality of sources or to generate complex signal patterns that combine and map to different spatial regions.

Abstract: A system for filtering an image using a bilateral filter includes a receiver and a bilateral filter. The receiver is for receiving an image having a plurality of pixels. The bilateral filter is run in constant time and comprises a pixel value range determinor and a spatial filter. The pixel value range determinor is configured to determine a range of pixel values of the plurality of pixels. The spatial filter is configured to spatially filter at least one of the plurality of pixels. The spatial filter is further configured to perform over a plurality of pixel values in the range of pixel values.

Abstract: Methods and systems are described for processing a signal in wireless communications. The signal may have synchronization information. A method of processing a signal having synchronization information may include receiving the signal, and determining a truncation region of the time domain estimated channel, the estimated channel having taps. The method further includes processing the channel taps within the truncation region.

Abstract: In an embodiment, the size of a filter in a search engine is dynamically adjusted in order to reduce filter error. The filter, for example, may be a domain filter that contains domain(s) of preferred search results. A search engine processes the query and the domain filter to create a set of search results that are relevant to the query terms and for which a selected portion of the search results satisfy the domain filter, in that such portion of the documents comes from domains in the filter. The search results may then be provided to the user, as a set of search results that are relevant to the query terms and which include some portion of results that match the query labels. The construction of the domain filter may be done with a Bloom filter approach, using domain prefixes from annotations.

Abstract: A method of generating random access preambles includes receiving information on a source logical index and generating random access preambles in the order of increasing cyclic shift from root ZC sequences with the consecutive logical indexes from the beginning of the source logical index until a predetermined number of the random access preambles are found, wherein the consecutive logical indexes are mapped to root indexes of the root ZC sequences.

Abstract: A unique digital compression technology for increasing the information carrying capacity of a bandwidth limited communications path highway by using a combined coding and modulation technique. The inventive technology allows multiple independent modulated data streams to simultaneously and instantaneously share the same bandwidth without cross-channel interference. A matched filter is used to substantially reduce the error rate and utilizes a unique training method based on performing a spectral response test. An algorithm calculates the effect of the unique characteristics of the communications highway on the transmitted signal and generates an ideal signal stored in the matched filter for comparison with received signals.

Abstract: This invention relates to a process for producing a pulsed sampled waveform comprising generating a baseband signal representing the integral of a desired output waveform; sampling and differentiating the baseband signal to produce frequency images of the baseband spectrum at multiples of the sampling rate; employing an analog bandpass filter to pass the spectral image centered at the desired up-conversion frequency band. This invention also relates to an apparatus for creating a high frequency output waveform signal comprising a means for generating a baseband signal representing the integral of a desired output waveform; a means for sampling the baseband signal; a means for generating a time delayed baseband signal by ?t; a means for inverting the delayed baseband signal; a means for summing the inverted the delayed baseband signal with the sampled baseband signal; a means for filtering to pass the high frequency signal.

Abstract: The purpose of this invention is that the optimum approximation formula can be derived even if an analysis filter has non-linear characteristic in a discrete signal approximation system. The norm of source signal 1 is bounded. Analysis filters 2 of FIR filters yield the output signal that is expressed with polynomial of transformed source signal 1 through a unit filter. An interpolation function is calculated according to the filter characteristic based upon the multi-dimensional function transformed from the polynomial. The output signal of the analysis filter 2 is sampled after each constant interval to obtain a discrete signal. The approximation function of the power function of the source signal 1 is calculated with linear combination of the interpolation functions using the discrete signals as coefficients. The radical of this approximation function is calculated to derive the root approximation function of the source signal 1.

Abstract: The filter device comprises a filter for filtering the input signal with a first set of filter coefficients, and for filtering the input signal with a second set of coefficients, a frequency domain correlator for correlating a first subset of frequency domain components of the first filtered signal to obtain a first correlation value, and for correlating a second subset of frequency domain components of the second filtered signal to obtain a second correlation value, wherein the first subset of correlated frequency domain components and the second subset of correlated frequency domain components are respectively located within a predetermined range of the correlated signals comprising the clock frequency, and a processor for selecting either the first set of filter coefficients or the second set of filter coefficients upon the basis of the first correlation value and the second correlation value for filtering the input signal.

Abstract: A system for filtering a data signal includes an input configured to receive the data signal through a transmission medium and a filter configured to remove distortion from the received data signal using equalization coefficients. The system further includes a processing unit configured to determine dynamically the equalization coefficients of the filter without using a predetermined training pattern in the received data signal.

Abstract: Provided are an apparatus and method for performing a complex number operation using a Single Instruction Multiple Data (SIMD) architecture. A SIMD operation apparatus may perform, in parallel, a real part operation and an imaginary part operation of a plurality of complex numbers. The real part operation and the imaginary part operation may be performed sequentially, or in parallel.

Abstract: A circuit breaker having a series arc fault detector and method of operation is provided. The series arc fault detector measures the RMS current and monitors current stability in response to a change in current levels. The sensitivity of the series arc fault detection is changed in response to the current satisfying a stability criterion. The circuit breaker further enables and disables series arc fault detection based on an enable threshold. The enable threshold is set based on the level of sensitivity being used to detect a series arc fault.

Abstract: A signal filtering method for filtering signals obtained at a number of time points by an electronic device is provided. Each of the signals corresponds to an i-th value at an i-th time point of the abovementioned time points, and the i-th value and an (i+1)-th value are defined as an i-th ordering sequence, wherein i is a positive integer. Each of the i-th ordering sequence and an (i+1)-th ordering sequence conforms to a first strict monotonicity. The method includes the following steps. Firstly, whether an (i+2)-th ordering sequence conforms to a second strict monotonicity is determined. If yes, then execute the next step. Determine whether an (i+3)-th ordering sequence conforms to the second strict monotonicity. If not, a signal obtained at an (i+3)-th time point is deleted.

Abstract: A dynamic Bloom filter comprises a cascaded set of Bloom filters. The system estimates or guesses a cardinality of input items, selects a number of hash functions based on the desired false positive rate, and allocates memory for an initial Bloom filter based on the estimated cardinality and desired false positive rate. The system inserts items into the initial Bloom filter and counts the bits set as they are inserted. If the number of bits set in the current Bloom filter reaches a predetermined target, the system declares the current Bloom filter full. The system recursively generates additional Bloom filters as needed for items remaining after the initial Bloom filter is filled; items are checked to eliminate duplicates. Each of the set of Bloom filters is individually queried to identify a positive or negative in response to a query.

Abstract: According to one embodiment, a compensation filtering device includes an impulse response calculator, a coefficient calculator, and an adder. The impulse response calculator calculates an impulse response of a reproduction system comprising a sound field. The coefficient calculator calculates a compensation coefficient to compensate for a tap coefficient such that a direct current gain of an extracted finite impulse response (FIR) filter with a predetermine number of taps extracted from an FIR filter having reverse characteristics of the impulse response takes a predetermined value. The tap coefficient indicates a weight of each of the taps. The adder adds the compensation coefficient to the tap coefficient of each of the taps of the extracted FIR filter to generate a compensation filter to compensate for acoustic characteristics of the reproduction system.

Abstract: A communications system receives a modulated signal that carries encoded communications data. An adaptive filter has a plurality of non-adaptive and adaptive filter taps with weighted coefficients and a tap order selection circuit for selecting the number and order of adaptive filter taps based on one of at least measured output power from the adaptive filter and signal modulation. A demodulator and decoder receives the filtered output signal and demodulates and decodes the signal to obtain the communications data.