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.

Abstract: A method of signal processing comprises receiving an unknown input signal that includes a distorted component and an undistorted component, the unknown input signal having a sampling rate of R; and performing self-linearization based at least in part on the unknown signal to obtain an output signal that is substantially undistorted, including by generating a replica distortion signal that is substantially similar to the distorted component, the generation being based at least in part on a target component having a sampling rate of R/L, L being an integer greater than 1.

Abstract: Embodiments related to echo compensation have been described and depicted.

Abstract: A system and method for graphic equalization of audio signals is disclosed. Traditional graphic equalizers provide control over the gains in each of a set of frequency bands. However, the actual band gains vary from the desired gains due to crosstalk between bands. Prior art methods for addressing this difficulty include applying a correction filter to the equalizer, and adjusting the shape of the individual band filters, both of which increase the computational cost. In an embodiment of the present invention, the input gains are processed to produce a set of adjusted gains which take into account the crosstalk, and result in an equalization interpolating the input band gains.

Abstract: A system for determining application of an adaptive filter includes a signal sensor to sense a detection signal; an adaptive filter to filter the sensed signal adaptively; and a module for determining application of an adaptive filter to analyze the sensed signal and the filtered signal and to determine application of the adaptive filter to the sensed signal based on the analyzed result.

Abstract: A system and method is provided for performing matrix inverse functions, for example, for use within Space-Time Adaptive Processing (STAP). The methods use parallelism of a Forward/Backward substitution algorithm in two dimensions to increase a speed of execution of the matrix inverse function. Sampled data is combined with steering vector values, which direct antennas in a desired direction in the absence interference, in order to determine adaptive weights used within filters to remove unwanted energy in the sampled data due to jammers, clutter or other interference. The adaptive weights are recursively computed, using stored values of previously calculated adaptive weights and other factor coefficients derived from the sampled data.

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: Disclosed is an improved method and apparatus for estimating and applying a step size value for a least mean squares echo canceller. A power estimate of an excitation signal is compared to a reference power level to determine a shift adjustment. The shift adjustment is added to a reference shift amount to determine a shift amount. The product of an excitation signal and an error signal is then calculated and the product is stored in a memory register comprising a plurality of bits. The bits stored in the memory register are shifted either left or right based upon the shift amount. The shift adjustment may be based in part upon the ratio of the excitation signal power estimate and the reference power level.

Abstract: Adapting a read channel processor is disclosed. Adapting includes determining a target output for input data from a variable target function, determining a filter output corresponding to the input data from a programmable filter having programmable filter coefficients, comparing the target output and the filter output, and recursively updating both the variable target function and the programmable filter coefficients to improve the comparison between the target output and the filter output.

Abstract: A self-calibrating, adaptive equalization system for generating an ideal digital signal is disclosed. The adaptive equalization system includes an equalizer and a high-gain buffer. The equalizer includes a first equalizer loop that feeds-back a control voltage to the equalizer and the high-gain buffer that includes a second equalizer loop that feeds-back a high-pass-to-low-pass filter ratio signal. Each of the first and second equalizer loops has a high-pass and a low-pass filter, rectifying circuits for each of the filters, and an integrating circuit that compares signal energy output from the rectifiers. The adaptive equalization system generates an ideal digital signal.

Abstract: A wireless communication apparatus receives an quadrature modulated signal, generate a local signal having a frequency different from a center frequency of the quadrature modulated signal, performs quadrature demodulation on the quadrature modulated signal by using the local signal, to obtain an I channel signal and a Q channel signal, performs Fourier transform on the I channel signal and the Q channel signal, to obtain signals in a frequency domain, and calculates a first correction coefficient for correcting phase distortion and amplitude distortion caused by the quadrature demodulation by using pairs of signals among the signals, each of the pairs are located at symmetrical frequency positions with respect to the frequency of the local signal.

Abstract: Forward and backward equalization processes are effectively used in a communication system for equalizing a received signal of a frame including a known symbol part. The known symbol part is provided in a position other than both ends of the frame. An equalization filter unit acquires a signal of an equalization process result by performing an equalization filter process based on a signal serving as an equalization process target and a tap gain coefficient. An update unit updates the tap gain coefficient using a predefined algorithm. A first or second transmission unit transmits a received signal posterior or prior to the known symbol part to a first or second memory in a forward direction or reverse order.

Abstract: An equalizer, group delay compensation circuit for the equalizer and method of compensating for group delay may improve group delay characteristics in the equalizer. The equalizer circuit may include a first low pass filter configured to filter a received input signal to output a filtered input signal, and a gain control circuit connected to an output terminal of the first low pass filter, and configured to modulate a gain of a transfer function for the equalizer. The equalizer may include a group delay compensation circuit connected to the output terminal of the first low pass filter and configured to compensate for a group delay of the input signal, and a second low pass filter connected to the output terminal of the first low pass filter.

Abstract: A new method to adjust the parameters of an adaptive Infinite Impulse Response (IIR) filter is suggested. The method adjusts the set of parameters of the pole polynomial of the filter. The parameters of the zero polynomial are calculated from the parameters of the pole polynomial. For efficiency, the pole polynomial is factored into a product of polynomials with at most quadratic order. To guarantee that the global minimum is achieved all the time, the algorithm ascertains that the new set of pole parameters gives smaller variance of the error than the set of pole parameters of the last adaptation time and the algorithm starts with the set of parameters that gives the global minimum.

Abstract: A finite impulse response (FIR) filter comprises a first multiplier that receives an input signal and a first tap coefficient. A first delay element receives the input signal and provides a fixed delay. A second multiplier receives a second tap coefficient and an output of the first delay element. A variable delay element receives the input signal and provides a variable delay. M delay elements provide the fixed delay. A first one of the M delay elements receives an output of the variable delay element and remaining ones of the M delay elements receive an output of a preceding one of the M delay elements, where M is an integer greater than one. M multipliers receive outputs of respective ones of the M delay elements and respective ones of M tap coefficients. A plurality of summers sum outputs of the first, second and M multipliers.

Abstract: Disclosed is an improved method and apparatus for estimating and applying a step size value for a least mean squares echo canceller. A power estimate of an excitation signal is compared to a reference power level to determine a shift adjustment. The shift adjustment is added to a reference shift amount to determine a shift amount. The product of an excitation signal and an error signal is then calculated and the product is stored in a memory register comprising a plurality of bits. The bits stored in the memory register are shifted either left or right based upon the shift amount. The shift adjustment may be based in part upon the ratio of the excitation signal power estimate and the reference power level.

Abstract: In one embodiment, a receiver has an equalizer, a tap-averaging block, a delay buffer, and a filter. The equalizer receives an input signal from upstream processing and generates sets of filter coefficients. Each set of filter coefficients is adaptively generated by 1) filtering the received signal to generate an equalized signal, 2) calculating an error of the equalized signal, and 3) generating a new set of coefficients based on the error of the equalized signal. The sets of filter coefficients are output to the tap-averaging block, which averages groups of the sets of filter coefficients to generate sets of averaged filter coefficients, where each averaged set is output to the filter. The filter receives a time-delayed version of the input signal from the delay buffer and applies the current set of averaged filter coefficients to the time-delayed signal. The filtered signal is then output to downstream processing.

Abstract: Equalizers are provided including an N-tap feed forward filter, an M-tap feed backward filter, an L-tap filter, a control unit and an accumulator. The control unit is configured to connect the L-tap filter to the N-tap feed forward filter or the M-tap feed backward filter based on multipath information present in a communications channel. The accumulator is configured to sum output signals from at least one of the N-tap feed forward filter, the M-tap feed backward filter and the L-tap filter and to output a summation result. Related digital receivers, methods and computer program products are also provided.

Abstract: Aspects of a method and system for unconstrained frequency domain adaptive filtering include one or more circuits that are operable to select one or more time domain coefficients in a current filter partition. A value may be computed for each of the selected one or more time domain coefficients based on a corresponding plurality of frequency domain coefficients. The corresponding plurality of frequency domain coefficients may be adjusted based on the computed values. A subsequent plurality of frequency domain coefficients in a subsequent filter partition may be adjusted based on the computed values. Input signals may be processed in the current filter partition based on the adjusted corresponding plurality of frequency domain coefficients. A time-adjusted version of the input signals may be processed in a subsequent filter partition based on the adjusted subsequent plurality of frequency domain coefficients.

Abstract: By using the adaptive filter, the reference input signal is processed so as to identify a pseudo-signal of a particular signal to be deleted. The pseudo-signal is subtracted from the mixture containing a target signal inputted from a microphone, the particular signal to be deleted, and a noise so as to obtain an error signal. A stationary noise is estimated to obtain a stationary noise estimated value. A non-stationary noise is estimated to obtain a non-stationary noise estimated value. The stationary noise estimated value is mixed with the non-stationary estimated value to obtain a mixed noise estimated value. An update amount is calculated according to a correlation value between the error signal and the reference input signal, and the mixed noise estimated value. According to the update amount, a coefficient of the adaptive filter is updated.

Abstract: Adaptive processing of an input signal is achieved by offline analysis, with inline processing comprising an adaptive filter. The method comprises passing the input signal through an adaptive time domain filter to produce an output signal. The input signal and/or output signal is used as an offline analysis signal. The analysis signal is transformed into a transform domain (eg frequency domain) to produce a transformed analysis signal. The transformed analysis signal is analysed, for example by ADRO, to produce a plurality of desired gains each corresponding to a respective transform domain sub-band. A time domain filter characteristic is synthesised to at least approach the desired gains. The adaptive filter is updated with the synthesised filter characteristic. Minimum phase adaptive filter techniques are found to possess particular benefits in this scheme.

Abstract: Adaptive processing of an input signal is achieved by offline analysis, with inline processing comprising an adaptive filter. The method comprises passing the input signal through an adaptive time domain filter to produce an output signal. The input signal and/or output signal is used as an offline analysis signal. The analysis signal is transformed into a transform domain (eg frequency domain) to produce a transformed analysis signal. The transformed analysis signal is analyzed, for example by ADRO, to produce a plurality of desired gains each corresponding to a respective transform domain sub-band. A time domain filter characteristic is synthesized to at least approach the desired gains. The adaptive filter is updated with the synthesized filter characteristic. Minimum phase adaptive filter techniques are found to possess particular benefits in this scheme.

Abstract: A digital signal processing block with a preadder stage for an integrated circuit is described. The digital signal processing block includes a preadder stage and a control bus. The control bus is coupled to the preadder stage for dynamically controlling operation of the preadder stage. The preadder stage includes: a first input port of a first multiplexer coupled to the control bus; a second input port of a first logic gate coupled to the control bus; a third input port of a second logic gate coupled to the control bus; and a fourth input port of an adder/subtractor coupled to the control bus.

Abstract: An equalization circuit that allows particularly for lowpass filtering by transmission lines comprises a compensating equalizer controlled according to whether the edges between bits in the data waveform are early or late. Adjusting the equalization causes edges to appear in the same place, whereas if the adjustment is incorrect certain edges will be late and certain edges will be early depending on the history of “1”s and “0”s in the data stream. This is an effect of so-called intersymbol interference. The control mechanism includes circuits for recognizing patterns of “1”s and “0”s in the recent history of the data waveform whose occurrence is used to trigger the adjustment of the equalizer.

Abstract: A waveform equalizer includes a filter unit, an error estimation unit, a tap coefficient storage unit, and an update amount calculation unit which includes an intermediate calculation unit and an update amount setting unit. Coefficient update amount ?Ci(n) for an ith tap is calculated according to an equation ?Ci(n)=?i(n)×?×e(n)×x*(n?i) with the multiplication by ?i(n) being performed by the update amount setting unit. Here, 0<?i(n)?1, and ?i(n) is a function f(Ci(n?1)) which monotonically increases with Ci(n?1).

Abstract: An adaptive filter that processes a sequence of input data includes a backward-looking predictive filter, which responsive to the input data and predictive filters the input data using a first set of filter tap weights and provides first predictive filtered data indicative thereof A first delay device delays the input data for a first period of time and provides a first delayed input data. A forward-looking predictive filter is responsive to the first delayed input data, and predictive filters the delayed input data using a second set of filter tap weights that are the complex-conjugate of the first set of filter tap weights, and provides second predictive filtered data indicative thereof. An adder sums the first predictive filtered data together with the second predictive filtered data and provides output data indicative thereof.

Abstract: A method and apparatus for a receive equalizer of a gigabit transceiver that is reconfigurable to support multiple communication standards. Communication standards having variable common mode and coupling requirements are accommodated through the use of reconfigurable integrated circuits (ICs), such as field programmable gate arrays (FPGAs), that provide a plurality of reconfigurable transceivers that are programmable through configuration, or partial reconfiguration, events. The reconfigurable transceivers apply internally generated common mode voltage signals to the differential input in support of the various communication standards.

Abstract: One embodiment of an equalizer circuit has an FIR filter 116 in the asynchronously oversampled domain with a filter coefficient adaptation module that adapts the filter coefficients to the transfer function of a data read channel. Applications include tape drives, drives for optical and magnetic discs as well as receivers. The filter adaptation is performed on the basis of an error signal delivered by a slicer 128 which operates on synchronous samples after timing recovery and sample reconstruction.

Abstract: A channel equalizer, method and computer program for equalizing a channel. The channel equalizer may include a feed forward filter and a switching unit. The switching unit may receive a signal input to the channel equalizer and an output signal from the feed forward filter, and may supply one of the input signal and output signal as an input to the feed forward filter.

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: A method for reducing a computational complexity of an m-stage adaptive filter is provided by expanding a weighted sum of forward prediction error squares into a corresponding binomial expansion series, expanding a weighted sum of backward prediction error squares into a corresponding binomial expansion series, and determining coefficient updates of the adaptive filter with the weighted sums of forward and backward prediction error squares approximated by a select number of terms of their corresponding binomial expansion series.

Abstract: An input signal is filtered for creating an output signal using an adaptive filter. An error signal is derived from the output signal. The adaptive filter has coefficient whose value can be modified. A value of a coefficient is modified using a derived updating amount. The updating amount is obtained from the product of a value of the input signal, a value of the polarity of the error signal, and a step gain. The step gain has the form 2K with K being an integer and being dependent on a magnitude of the value of the error signal and on a step gain parameter. The updating amount is dependent on both the magnitude and the polarity of the error signal, therefore allowing a precise update of the coefficient. The specific form of the step gain allows a fast derivation of the product.

Abstract: An active noise control apparatus for reducing noise from a noise source includes a microphone for detecting noise produced by the noise source, and a generalized finite impulse response (FIR) filter for receiving noise signals of the detected noise from the microphone and generating control signals for reducing the noise from the noise source. A speaker produces sound based on the control signals from the generalized FIR filter for substantially canceling the noise from the noise source.

Abstract: An equalizer includes at least two first mutually interfering equalizer sections, and at least two second interference-correcting equalizer sections, arranged in series. Each of the second equalizer sections corresponds with one first equalizer section, such that although each corresponding equalizer has the same center frequency, the second equalizer sections have an equalization response opposite the interference effect, and the gain of the corresponding second equalizer at the respective common center frequency contains the negative gain of at least one first equalizer adjacent to the corresponding first equalizer at the center frequency of the corresponding first equalizer.

Abstract: The MPEG2 Advanced Audio Coder (AAC) standard limits the number of filters used to either one filter for a “short” block or three filters for a “long” block. In cases where the need for additional filters is present but the limit of permissible filters has been reached, the remaining frequency spectra are simply not covered by TNS. Two solutions are proposed to deploy TNS filters in order to get the entire spectrum of the signal into TNS. The first method involves a filter bridging technique and complies with the current AAC standard. The second method involves a filter clustering technique. Although the second method is both more efficient and accurate in capturing the temporal structure of the time signal, it is not AAC standard compliant. Thus, a new syntax for packing filter information derived using the second method for transmission to a receiver is also outlined.

Abstract: Filtering part 50 performs filtering by converting a received signal sampled using a predetermined number of over-samples to an over-sample number small enough to avoid frequency-domain foldover noise generation. Transmission channel estimation part 60 performs path timing detection by converting the received signal sampled using a predetermined number of over-samples to an over-sample number large enough to obtain sufficient timing resolution, converts transmission channel responses for each path timing to the frequency domain, and calculates transmission channel estimates corresponding to the subcarriers of the received signal in the frequency domain. Weight calculation part 6 receives the frequency-domain channel estimates outputted from transmission channel estimation part 60 and calculates the weights of the equalizing filter used in filtering part 50.

Abstract: An apparatus and method for implementing an equalizer which combines the benefits of a decision feedback equalizer (DFE) with a maximum-a-posterori (MAP) equalizer (or a maximum likelihood sequence estimator, MLSE) to provide an equalization device with significantly lower complexity than a full-state MAP device, but which still provides improved performance over a conventional DFE. The equalizer architecture includes two DFE-like structures, followed by a MAP equalizer. The first DFE forms tentative symbol decisions. The second DFE is used thereafter to truncate the channel response to a desired memory of L1 symbols, which is less than the total delay spread of L symbols of the channel. The MAP equalizer operates over a channel with memory of L1 symbols (where L1<=L), and therefore the overall complexity of the equalizer is significantly reduced.

Abstract: A method for detecting the position of a preceding vehicle (2) in relation to an same-vehicle (1), comprising a step of acquiring a primary data set having vehicle range information ri and lateral position information Li, Ri; a step of linear regression processing for acquiring a secondary data set having vehicle range information ri wherein the deviation with the acquired linear regression line is at or below a prescribed threshold value and corresponding lateral position information Li, Ri; a step of clustering processing for performing clustering processing on the lateral position information Li, Ri in the secondary data set and acquiring a tertiary data set having lateral position information Li, Ri in the largest cluster and corresponding vehicle range information ri; and a step of position information calculation for calculating the vehicle range and lateral position at the present time t0 using the tertiary data set.

Abstract: Bit-Edge Zero Forcing Equalizer. A novel solution is presented by which a BE-ZFE (Bit-Edge Zero Forcing Equalizer) is employed to drive an error term within a data signal to an essentially zero value. This new BE-ZFE looks at values of data that occur at the bit edges of a data signal and drives the associated error term to zero. The new BE-ZFE is appropriately implemented within communication systems that are phase (or jitter) noise limited. Some examples of such communication systems include high-speed serial links one type of which serviced using a SERDES (Serializer/De-serializer) where data that is originally in a parallel format is serialized into a serial data stream and then subsequently de-serialized back into a parallel data stream.

Abstract: A filter equalization technique for an analog signal path, such as in an instrument that simultaneously measures a signal over a band of frequencies, uses magnitude measurement data for high frequency bands for which phase-calibrated sources are not readily available. A sinusoidal signal source together with an accurate power meter is used to provide a stepped frequency input over a desired frequency band to the analog signal path with an accurate measured magnitude. The output of the analog signal path is digitized and the resulting frequency magnitudes are computed. Then the resulting power meter results are deducted from the frequency magnitudes measured each time by the instrument to determine the magnitude response of the analog signal path. Using a Hilbert transform the corresponding phase response is determined based on a minimum phase assumption over the desired frequency band. From the magnitude and phase responses an inverse or digital equalization filter may be designed for the analog signal path.

Abstract: An audio reproducing apparatus has a correction coefficient holding means (6) for holding at least one set of correction coefficients (K0) based on an inverse characteristic (H0) of a transfer characteristic from a speaker means (10) to a listening position (13). The correction coefficients are derived by convolution of an arbitrary transfer characteristic (H00) and the inverse characteristic (H0). The correction coefficients (K0) held by the correction coefficient holding means (6) are convolved with the audio signal in the non-recursive digital filter means (5) to generate output. The audio reproducing apparatus can realize an arbitrary acoustic characteristic easily, with a simple structure. Not just high fidelity audio reproduction, but also recreation of intended sound quality is enabled.

Abstract: An electroencephalograph system and method for controlling the stability of an adaptive filter during high noise spikes in ocular sensor channels. The method comprises receiving a signal from at least one sensor and determining when an adaptive filter algorithm is subject to becoming unstable based on a signal from the at least one sensor. Operation of the adaptive filter algorithm is suspended while the algorithm is subject to becoming unstable.

Abstract: A complex signal in which one of two signals that have been generated from a single real signal and that have phases that are shifted 90° with respect to each other is a real part and the other signal is an imaginary part is applied as input to an input terminal. A filter unit generates an output signal that is a complex signal by means of a convolution operation of the input signal, and a filter coefficient that is a real signal and supplies this output signal to an output terminal. A coefficient control unit made up from a common unit and separate units updates the filter coefficients such that the value of the envelope of the output signal approaches a target signal.

Abstract: A Q-Filter is a reconfigurable technique that performs a continuum of linear and nonlinear filtering operations. It is modeled by unique mathematical structure, utilizing a function called the Q-Measure, defined using a set of adjustable kernel parameters to enable efficient hardware and software implementations of a variety of useful, new and conventional, filtering operations. The Q-Measure is based on an extension of the well-known Sugeno ?-Measure. In order to optimize the Q-Filter kernel parameters, the value of an error function is minimized. The error function is based on difference between the filtered signal and target signal, with the target signal being a desired result of filtering.

Abstract: An information handling system includes a wireless device and interference suppression apparatus that adapts to the different interference problems experienced by the wireless device when the system changes from one operating mode or state to another. The interference suppression apparatus includes a controller that instructs an adaptive filter with respect to the appropriate filter characteristics to employ to suppress interference when the system is operating in a first mode. When the system changes to a second mode of operation, the interference suppression apparatus updates the filter characteristics to filter characteristics which are appropriate for suppressing interference associated with the second mode of operation.

Abstract: The adaptive digital filter of the present invention includes: a filter unit that includes a plurality of multipliers (3360-336N-1) that are divided into groups of at least one multiplier and the other multipliers based on expected values of filter coefficients and that generates first signals by means of convolution operations of an input signal and filter coefficients; an adder (338) that adds the input signal that is applied to at least one multiplier (336M-1) and first signals and supplies the results as second signals; and coefficient control unit (318 and 3190-319N-1) that control filter coefficients based on error between a target signal and an index value derived from the second signals.

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: Disclosed are a system, method and device for generating an equalized signal from an input signal. Symbols in the equalized signal may be detected on each of a sequence of symbol intervals to recover a symbol value in the symbol interval. A feedback coefficient may be applied to a symbol value recovered in a previous symbol interval to generate the equalized signal in a current symbol interval. The feedback coefficient may be generated based, at least in part, on an estimated error associated with the equalized signal. The estimated error associated with the equalized output signal from among a plurality of candidate estimated error values.

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 relates to a digital filter capable of computing a tap without output delay due to the filter operation in a symbol time, and a digital broadcasting receiver having the same. Particularly, filter output is obtained within a clock period, one multiplier and one adder are used to perform coefficient update for a plurality of taps, and the multiplier performs the output operation for each tap, whereby the number of multipliers and adders is reduced inversely proportional to the number of taps being operated for one clock period. Thus, the digital filter of the present invention can be very advantageously used for resolving the filter size problem in multi-tap filters.