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# Filtering

## Subclass of:

## 708 - Electrical computers: arithmetic processing and calculating

## 708100000 - ELECTRICAL DIGITAL CALCULATING COMPUTER

## 708200000 - Particular function performed

### Patent class list (only not empty are listed)

#### Deeper subclasses:

Class / Patent application number | Description | Number of patent applications / Date published |
---|---|---|

708313000 | Decimation/interpolation | 41 |

708322000 | Adaptive | 23 |

708319000 | Transversal | 8 |

708308000 | Multidimensional data | 8 |

708316000 | Having multiplexing | 7 |

708306000 | Finite arithmetic effect | 7 |

708301000 | Tapped delay line | 7 |

708320000 | Recursive | 6 |

708315000 | By convolution | 5 |

20130031152 | METHODS AND APPARATUSES FOR CONVOLUTIVE BLIND SOURCE SEPARATION - Methods and apparatuses for convolutive blind source separation are described. Each of a plurality of input signals is transformed into frequency domain. Values of coefficients of unmixing filter corresponding to frequency bins are calculated by performing a gradient descent process on a cost function at least dependent on the coefficients of the unmixing filters. In each iteration of the gradient descent process, gradient terms for calculating the values of the same coefficient of the unmixing filters are adjusted to improve smoothness of gradient terms across the frequency bins. With respect to each of the frequency bins, source signals are estimated by filtering the transformed input signals through the respective unmixing filter configured with the calculated values of the coefficients. The estimated source signals on the respective frequency bins are transformed into time domain. The cost function is adapted to evaluate decorrelation between the estimated source signals. | 01-31-2013 |

20120246211 | SYNTACTICAL SYSTEM AND METHOD FOR CHROMATOGRAPHIC PEAK IDENTIFICATION - A system and method identifies data peaks representative of empirical data of a sample. The system and method assign a grammar type to correspond to data points represented on a data plot such as a chromatogram and identify the presence of a peak syntax based on an analysis of the grammar element types assigned to the data points of the chromatogram. | 09-27-2012 |

20090070396 | WAVEFORM EQUALIZING DEVICE - Tap coefficients for a filter for removing a ghost signal are converged to optimum values in a short time. The waveform equalizing device includes: an initial tap coefficient generation section for determining and outputting the initial values of tap coefficients for a FIR filter and an IIR filter based on a plurality of correlation values; and a tap coefficient updating section for outputting the initial values of tap coefficients for the FIR filter and the IIR filter to these filters and updating tap coefficients for these filters based on error information. The initial tap coefficient generation section reverses the order of values, among the plurality of correlation values, corresponding to delays within a predetermined range to determine the order-reversed values as the initial values of tap coefficients for the FIR filter corresponding to the delays within the predetermined range, and also reverses the signs of values, among the plurality of correlation values, corresponding to delays exceeding the predetermined range to determine the sign-reversed values as the initial values of tap coefficients for the IIR filter. | 03-12-2009 |

20120131079 | METHOD AND DEVICE FOR COMPUTING MATRICES FOR DISCRETE FOURIER TRANSFORM (DFT) COEFFICIENTS - A method of computing matrices of discrete-frequency Discrete Fourier Transform (DFT) coefficients, the method including the steps of (a) for a first frame ( | 05-24-2012 |

20080250092 | SYSTEM FOR CONVOLUTION CALCULATION WITH MULTIPLE COMPUTER PROCESSORS - A system for calculating a convolution of a data function with a filter function utilizing an array of processors including first and last processors. A coefficient value based on a derivation of the filter function and a data value representative of the data function are multiplied to produce a current intermediate value. Except in the first processor, a prior intermediate value is then added to the current intermediate value. Except in the last processor, the data and current intermediate values are then sent to the next processor. Then the last processor's prior intermediate value, if any, is added to its current intermediate value to produce a result value, wherein the result values collectively are representative of the convolution of the data function with the filter function. | 10-09-2008 |

708311000 | Frequency detection | 4 |

20110289129 | METHOD FOR DETERMINING SAMPLING RATE AND DEVICE THEREFOR - A method and a device for determining sampling rate are provided. The device receives an input signal of SPDIF. The method includes following steps. A plurality of multiple values between a plurality of bi-phase clock frequencies of the input signal and a system frequency are obtained, and a first weighted average and a second weighted average are calculated according to a first filter range, a second filter range and the multiple values. When a first difference is greater than a second difference, the sampling rate is set to a first sampling rate. Otherwise, the sampling rate is set to a second sampling rate. The first difference and the second difference are obtained according to the first weighted average, the second weighted average and a frequency threshold. The method determines the sampling rate rapidly according to weighted averages adjusted by filter ranges, reduces a probability of erroneous judgment and saves memory. | 11-24-2011 |

20090063602 | DEVICE AND METHOD FOR PREVENTING WIRETAPPING ON POWER LINE - Provided are a device and method for detecting a wiretapping device using a power line and nullifying the wiretapping device. More particularly, a device and method for preventing wiretapping, which sense a wiretapping signal from a power line and transmit a noise signal to the power line, are provided. The device for preventing wiretapping includes: a signal detector for receiving signals from a power line and filtering the received signals in at least one frequency band; a controller for receiving the filtered signal from the signal detector and determining whether a wiretapping signal exists; and a noise signal output unit for transmitting a noise signal to the power line according to whether or not the wiretapping signal exists. The device can detect wiretapping and simultaneously nullify the function of the wiretapping device by detecting a wiretapping signal from a power line and transmitting a noise signal having a frequency corresponding to a frequency band of the wiretapping signal. | 03-05-2009 |

20100070550 | METHOD AND APPARATUS OF A SENSOR AMPLIFIER CONFIGURED FOR USE IN MEDICAL APPLICATIONS - A digital data bandpass filter process using digital signal processing processes raw digital data twice using distinct all-pass filters, wherein the two filters perform different frequency-dependent phase shifts of the digital content. The two filter output streams may be subtracted to remove out-of-band energy. The process produces lower levels of noise and conversion artifacts when used for filtering fixed-point data that may have low energy, and thus few non-zero bits, in the analog input signals, than conventional digital bandpass filters. | 03-18-2010 |

20110314074 | LOW POWER AND LOW COMPLEXITY ADAPTIVE SELF-LINEARIZATION - A digital signal processing system comprising: an input terminal to receive an input signal that includes a distorted component and an undistorted component, the input signal having a sampling rate of R; and an adaptive self-linearization module coupled to the input terminal, to perform self-linearization based at least in part on the input signal to obtain an output signal that is substantially undistorted, wherein: the adaptive self-linearization module is to generate 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; the adaptive self-linearization module includes a first digital signal processor (DSP) that is adapted to obtain a filter transfer function that approximates a system distortion transfer function, and a second DSP that is configured using configuration parameters of the first DSP. | 12-22-2011 |

708304000 | Nonlinear (e.g., median, etc.) | 4 |

20130080492 | PROCESSING KALMAN FILTER - A method and system for processing Kalman Filter. The system includes: an Unscented Kalman Filter; and a processor device configured to: non-uniform a phase duration of a signal outputted from a plant; inputting the signal to the Unscented Kalman Filter; and restore non-uniformed phase duration of an estimated value calculated in the Unscented Kalman Filter to the phase duration. | 03-28-2013 |

20090094304 | System and method for adaptive nonlinear filtering - An adaptive nonlinear filtering system includes an adaptive filter module that is configured to generate relative location information pertaining to a relative location of an input signal within an input range; determine an input dependent filter parameter based at least in part on the relative location information; generate an output signal based at least in part on the input dependent filter parameter; and feed back a feedback signal that is generated based at least in part on the output signal and a target signal. | 04-09-2009 |

20100250638 | SIGNAL PROCESSING APPARATUS, DIGITAL FILTER AND RECORDING MEDIUM - Provided is a signal processing apparatus for compensating for a non-linear distortion of a digital signal, including: an analysis signal generating section that converts the digital signal into a analysis signal of a complex number, using a digital filter; and a compensation section that compensates for the analysis signal, using a compensation coefficient of a complex number corresponding to the non-linear distortion, where the digital filter divides data of the digital signal into “n” data sequences, assigns (n*L+k)th data of the digital signal to a k-th data sequence, performs filtering on each of the data sequences using a same filter coefficient, and combines each of the data sequences after the filtering, thereby generating an imaginary number portion of the analysis signal, where “n” is an integer equal to or greater than 2, L=0, 1, . . . , and k=1, 2, . . . , n. | 09-30-2010 |

20130132454 | Feedback-Based Particle Filtering - Methods for estimating a conditional probability distribution for signal states of a non-linear random dynamic process. The filter is based on multiple particles, each defined by a state space model similar to the dynamic process. Each particle is updated on the basis of a control input derived by proportional gain feedback on an innovation process. The innovation process is the difference between an increment in an observed quantity measured by one or more sensors and an average of a function of the particles. The particle filter of the invention may also be applied to filtering problems with data association uncertainty where multiple measurements are obtained, of which at most one originates from a specified target. | 05-23-2013 |

708309000 | Frequency measurement | 3 |

20110252077 | SYSTEMS AND METHODS FOR FILTERING A SIGNAL - Methods for filtering an input signal x(k) to produce an output signal y(k) such that the ratio of a power level of the output signal to a power level of the input signal is substantially equal to a desired value γ are provided. The methods include forming a first corrected frequency response | 10-13-2011 |

20120209900 | COMMUNICATION SYSTEM WITH SIGNAL PROCESSING MECHANISM AND METHOD OF OPERATION THEREOF - A method of operation of a communication system includes: generating a filter impulse response and a filter time-domain data with a shortening filter; generating a filter frequency response based on the filter impulse response with a filter frequency response calculator; generating a filter frequency-domain data based on the filter time-domain data with a first process unit; and generating a raw channel impulse response with a filter frequency removal unit for removing the filter frequency response from the filter frequency-domain data. | 08-16-2012 |

20120271872 | PROCEDURE FOR DENOISING DUAL-AXIS SWALLOWING ACCELEROMETRY SIGNALS - Dual-axis swallowing accelerometry is an emerging tool for the assessment of dysphagia (swallowing difficulties). These signals however can be very noisy as a result of physiological and motion artifacts. A novel scheme for denoising those signals is proposed, i.e. a computationally efficient search for the optimal denoising threshold within a reduced wavelet subspace. To determine a viable subspace, the algorithm relies on the minimum value of the estimated upper bound for the reconstruction error. A numerical analysis of the proposed scheme using synthetic test signals demonstrated that the proposed scheme is computationally more efficient than minimum noiseless description length (MNDL) based de-noising. It also yields smaller reconstruction errors (i.e., higher signal-to-noise (SNR) ratio) than MNDL, SURE and Donoho denoising methods. When applied to dual-axis swallowing accelerometry signals, the proposed scheme improves the SNR values for dry, wet and wet chin tuck swallows. These results are important to the further development of medical devices based on dual-axis swallowing accelerometry signals. | 10-25-2012 |

708318000 | Lattice | 2 |

20100077014 | SECOND ORDER REAL ALLPASS FILTER - A digital all-pass filter has an input port leading to an input sum block and a first feed forward path. Within the first feed forward path is a multiplier. The filter also has an output port coupled to an output sum block that receives a signal from the first feed forward path. A first feedback path is also provided from the output port to the input sum block. The first feedback path includes a multiplier therein. Nested within this structure is a first order all-pass filter having a feed forward path including a forward path delay and forward path that is delayed and a feedback path absent a separate delay element and beginning after the forward path delay element. | 03-25-2010 |

20120185525 | Filtering Discrete Time Signals Using a Notch Filter - Various techniques are generally described for digital signal processing (DSP) such as discrete time filters. In some examples, a Canonic Filter Module (CFM) can be used to configure the discrete time filter using an LSF-Model with a finite length sequence. A single CFM can be configured to provide any type of discrete time filter used in signal processing. Filters can be modeled as a set of interconnected notch filters, a lattice structure of a discrete time filter is generally described that is based on a LSF-Model. | 07-19-2012 |

708314000 | Matched filter type | 2 |

20080294708 | Methods, systems, and computer program products for parallel correlation and applications thereof - A fast correlator transform (FCT) algorithm and methods and systems for implementing same, correlate an encoded data word (X | 11-27-2008 |

20080243982 | Hardware matrix computation for wireless receivers - In one embodiment, a receiver including one or more signal-processing blocks and a hardware-based matrix co-processor. The one or more signal-processing blocks are adapted to generate a processed signal from a received signal. The hardware-based matrix co-processor includes two or more different matrix-computation engines, each adapted to perform a different matrix computation, and one or more shared hardware-computation units, each adapted to perform a mathematical operation. At least one signal-processing block is adapted to offload matrix-based signal processing to the hardware-based matrix co-processor. Each of the two or more different matrix-computation engines is adapted to offload the same type of mathematical processing to at least one of the one or more shared hardware-computation units. | 10-02-2008 |

708303000 | Microprocessor | 1 |

20080243979 | Data Stream Filters And Plug-Ins For Storage Managers - A storage manager and related method and computer program product manages client data on a data storage resource and includes the ability to utilize many different types of data stream filters that are neither built into the storage manager nor require a custom programming effort. A storage manager user may readily implement filtering by simply identifying a data stream filter the user wishes the storage manager to use for filtering the user's data. The filter can be an off-the-shelf program that is not part of the storage manager and which does not require client application or storage manager domain knowledge (e.g., knowledge of protocols or data types or formats used by the application or storage manager). The storage manager invokes the identified filter as part of a requested data stream operation and receives a data stream from a data stream source. The data stream is provided to the filter, which filters the data stream. Following filtering, the storage manager receives the data stream from the filter and sends it to a data stream destination. | 10-02-2008 |

708310000 | Coherent | 1 |

20120150934 | Receiver Having an Adaptive Filter and Method of Optimizing the Filter - A receiver comprises an adaptive filter having an input for a digitized input signal, means for storing a pre-designed filter characteristic, means for analyzing a digital. representation of the input signal to determine a desired position of the filter characteristic to match the system requirements, and means for adapting the stored pre-designed filter characteristic in the frequency domain and/or the time domain to match the system requirements and for transforming the adapted filter characteristic to the time domain to update coefficients for the adaptive filter and for loading updated coefficients into adaptive filter. The updating of the coefficients may be done periodically. The adaptation may be one or more of adjusting bandwidth, frequency shift and, in the case of a bandpass characteristic, superimposing characteristics. | 06-14-2012 |

Entries | ||

Document | Title | Date |
---|---|---|

20120246209 | METHOD FOR CREATING A MARKOV PROCESS THAT GENERATES SEQUENCES - 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. | 09-27-2012 |

20130036147 | INFINITE IMPULSE RESPONSE (IIR) FILTER AND FILTERING METHOD - An infinite impulse response (IIR) filter is provided. The IIR filter includes an amplifier and a filter coupled in a feedback path of the amplifier. The amplifier generates an output signal according to an input signal. The filter filters the output signal according to a first transfer function and provides the filtered output signal to an input of the amplifier. The IIR filter and the first filter have the same order larger than one. | 02-07-2013 |

20090125575 | NOISE CANCELING DEVICE, WEIGHING DEVICE, METHOD OF CANCELING A NOISE, AND METHOD OF DESIGNING A DIGITAL FILTER - It is an object of the present invention to provide techniques which allow for easier change in filter characteristics of a digital filter. Then, in order to attain this object, in a weighing device according to the present invention, a filter coefficient calculator ( | 05-14-2009 |

20090119356 | METHOD FOR REDUCING DIGITAL FILTER COEFFICIENT WORD SIZE AND APPARATUS THEREFOR - 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. | 05-07-2009 |

20090307294 | Conversion Between Sub-Band Field Representations for Time-Varying Filter Banks - Conversion between sub-band field representations for time-dependent filter banks. The invention relates to a transcoding processing operation between different sub-band fields, aiming to compact the application of a first vector representing the signal in a first sub-band field to a synthesis filter bank, and then to an analysis filter bank, in order to obtain a second vector representing the signal in a second sub-band field. In particular, the synthesis bank and/or the analysis bank are time-dependent. Within the scope of the invention, matrix filtering of the first vector is anticipated in order to directly obtain the second vector, this matrix filtering being represented by a global conversion matrix comprising pre-calculated sub-blocks of matrices (A | 12-10-2009 |

20090094303 | FILTER OPERATION UNIT AND MOTION-COMPENSATING DEVICE - A filter operation unit that performs a multiply-accumulate operation on input data and a filter coefficient group including a plurality of coefficients using Booth's algorithm. The filter operation unit includes: at least two filter multiplier units that multiply the input data and a difference between adjacent filter coefficients in a filter coefficient group to obtain multiplication results; and an adder that adds the multiplication results of the multiplier units adjacent to each other. The filter multiplier units each include: a partial product generation unit that repeatedly generates a partial product according to Booth's algorithm; and an adder that cumulatively adds the partial products generated by the partial product generation unit. | 04-09-2009 |

20090077148 | Methods and Apparatus for Perturbing an Evolving Data Stream for Time Series Compressibility and Privacy - Techniques for perturbing an evolving data stream are provided. The evolving data stream is received. An online linear transformation is applied to received values of the evolving data stream generating a plurality of transform coefficients. A plurality of significant transform coefficients are selected from the plurality of transform coefficients. Noise is embedded into each of the plurality of significant transform coefficients, thereby perturbing the evolving data stream. A total noise variance does not exceed a defined noise variance threshold. | 03-19-2009 |

20090089348 | ADAPTIVE PRECISION ARITHMETIC UNIT FOR ERROR TOLERANT APPLICATIONS - Two process-tolerant arithmetic circuit architectures are implemented to develop functional blocks for error-tolerant applications such as FIR filters and FFT blocks. The resulting blocks may achieve computational performance of up to 42 times higher than conventional architectures. Embodiments adaptively change the precision of the computation to achieve a high precision computation given the underlying speed of the circuit. The resulting improvement can be allocated to increasing yield or dynamically trading off between reduced power consumption, faster computation, or higher-fidelity computation. | 04-02-2009 |

20130191429 | INFINITE IMPULSE RESPONSE FILTER ARCHITECTURE WITH IDLE-TONE REDUCTION - A digital infinite impulse response filter has a plurality of cascaded filter elements, with each filter element defining a pole of the filter and wherein the poles lie inside a unit circle. The filter elements are configured such that the p of the last filter element is a real number. In one embodiment the poles are arranged as complex conjugate pairs. In another embodiment the real part of the output of each filter element is extracted before being passed to the next filter element. This architecture offers improved idle tone with reduced complexity. | 07-25-2013 |

20100082720 | CURVE-FITTING METHOD TO CALCULATE COARSE FREQUENCY OFFSET - A method comprising the steps of: providing a known sequence comprising a plurality of data points; and curve-fitting the plurality of data points to calculate coarse frequency offset. | 04-01-2010 |

20080275929 | Use of line characterization to configure physical layered devices - A method of optimizing filter performance through monitoring channel characteristics is provided. A signal enters a channel and a receiver receives the signal. The receiver includes a FIR filter to remove near-end transmitted interference and recover a far-end desired signal. The filter has storage elements configured as a shift registers to move the signal, multipliers to multiply the signal by a filter coefficient, an intermittent summer to combine the multiplied results into a replica of an interfering signal, a final summer to remove the replica from the receiver signal to provide direct and indirect monitoring of the signal, where direct monitoring includes time or frequency monitoring, and indirect monitoring includes monitoring signal to noise ratio, error magnitude or bit error rate. The filter is optimized according to monitoring and includes reducing a dynamic range, reducing bits of precision, reducing linearity, the filter, and reallocating the filter. | 11-06-2008 |

20100138466 | Filter coefficient calculation method and filter coefficient calculation unit - Provided is a filter coefficient calculation method that calculates filter functions, each having (2n+1) rows and (2n+1) columns (n is an integer), the method including calculating a first filter function in accordance with a set value that is externally input, calculating an error between a total sum of values included in the first filter function and an ideal value of the total sum, supplying an odd error included in the error to a first origin coefficient that is located at a center of the first filter function as a first correction value if the error is an odd number, and supplying an even error to one of the first origin coefficient and a coefficient pair that is located symmetrically with respect to a point of the first origin coefficient as a second correction value, the even error being the error except the odd error. | 06-03-2010 |

20090327384 | SYSTEM AND METHOD FOR ACTIVE DIPLEXERS - The present invention relates to methods and systems for signal filtering in electronic devices and more particularly, some embodiments related to methods and systems for filtering of radio frequency (RF) signals. In some embodiments, a filter circuit may comprise a down-converter, a filter, coupled to the down-converter and configured to filter the down-converted signal, and an up-converter, coupled to the filter. Various embodiments might also include a combining circuit, coupled to the up-converter and configured to combine the filtered, up-converted signal and the input signal. | 12-31-2009 |

20090055457 | Field device with capability of calculating digital filter coefficients - 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. | 02-26-2009 |

20100161697 | METHOD OF CORDIC COMPUTING VECTOR ANGLE AND ELECTRONIC APPARATUS USING THE SAME - 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. | 06-24-2010 |

20100174767 | EFFICIENT FILTERING WITH A COMPLEX MODULATED FILTERBANK - A filter apparatus for filtering a time domain input signal to obtain a time domain output signal, which is a representation of the time domain input signal filtered using a filter characteristic having an non-uniform amplitude/frequency characteristic, comprises a complex analysis filter bank for generating a plurality of complex subband signals from the time domain input signals, a plurality of intermediate filters, wherein at least one of the intermediate filters of the plurality of the intermediate filters has a non-uniform amplitude/frequency characteristic, wherein the plurality of intermediate filters have a shorter impulse response compared to an impulse response of a filter having the filter characteristic, and wherein the non-uniform amplitude/frequency characteristics of the plurality of intermediate filters together represent the non-uniform filter characteristic, and a complex synthesis filter bank for synthesizing the output of the intermediate filters to obtain the time domain output signal. | 07-08-2010 |

20090006515 | System and method for dynamic weight processing - 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. | 01-01-2009 |

20100306297 | FILTER - 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. | 12-02-2010 |

20100325184 | DIGITAL SIGNAL PROCESSING APPARATUS AND DIGITAL SIGNAL PROCESSING METHOD - A digital signal processing apparatus includes a frame generator configured to generate a plurality of frames from a row of sample data of a time-domain, a part of each frame overlapping with adjoining frames, a Fourier transform unit configured to transform at least one of the generated frames into a frequency domain by Fourier transformation, an addition unit configured to add predetermined frequency characteristic to the transformed frame, and an inverse Fourier transform unit configured to transform the added frame into the time-domain by inverse Fourier transformation and to delete the overlap of the frame of the time-domain transformed. | 12-23-2010 |

20100235419 | FILTERING APPARATUS, FILTERING METHOD, PROGRAM, AND SURROUND PROCESSOR - A filtering apparatus for obtaining an output in a case where a discrete-time signal having a length of N (N is an integer) is input to an FIR filter with a filter coefficient having a length of M (M is an integer, N≧M−1), including: a division unit for dividing the discrete-time signal; a first zero padding unit for padding zero after the discrete-time signals; a first fast Fourier transform unit for performing FFT on the zero padded data; a second zero padding unit for padding zero after the filter coefficient; a second fast Fourier transform unit for performing FFT on the zero padded data; a multiplication unit for multiplying the frequency domain data by the frequency domain data; an inverse fast Fourier transform unit for performing IFFT on the multiplication results; and an adder unit for adding the discrete-time signals. | 09-16-2010 |

20090313313 | DIGITAL FILTER DEVICE, PHASE DETECTION DEVICE, POSITION DETECTION DEVICE, AD CONVERSION DEVICE, ZERO CROSS DETECTION DEVICE, AND DIGITAL FILTER PROGRAM - A digital filter device capable of removing the effect of noise such as chattering from a zero crossing signal is provided. A digital filter device | 12-17-2009 |

20100100576 | Desensitized Filters - 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 | 04-22-2010 |

20080320068 | Wideband suppression of motion-induced vibration - The present invention is a new method to create motions based on the use of rate limited profiles convolved with FIR kernel filters, termed herein as Rate limited Boxcar Aliased IFIR (RBAI) motions, or profiles. The present invention demonstrates a simpler generalized view of the creation of pulse-based profiles based on digital signal processing windowing techniques. The method turns windowing functions into an IFIR filter using boxcar functions for the interpolation. The IFIR filter smoothes the pulse-based profiles, allowing the resultant filter to be applied directly over a simple rate limited base profile. The resulting motion profiles suppress residual vibration over a much wider band of frequencies than previous methods. | 12-25-2008 |

20080256156 | Reliable and Efficient Computation of Modal Interval Arithmetic Operations - A computer executable method of performing a modal interval operation, and system for performing same is provided. The method includes providing representations of first and second modal interval operands. Each modal interval operand of the operands is delimited by first and second marks of a digital scale, each mark of the marks comprises a bit-pattern. Each bit-pattern of the bit-patterns of the marks of each of the modal interval operands are examined, and conditions of a set of status flags corresponding to each bit-pattern of the bit-patterns of the marks are set. A bit-mask is computed wherein the mask is based upon the set condition of the status flag sets and a presence/absence of an exceptional arithmetic condition, and a presence/absence of an indefinite operand are each represented by a bit of said bits of said bit mask. | 10-16-2008 |

20110072064 | REDUCE INTERFERENCE FOR KVM SYSTEM - Embodiments of the invention provide methods and apparatus for designing a filter configured to reduce the effects of noise in an electrical signal transferred between a KVM (Keyboard, Video, Mouse) device and another device. Designing the filter may involve determining a filter architecture, wherein the filter architecture defines a desired type of frequency response for the filter, determining one or more parameters of the filter, wherein the parameters define one or more performance characteristics of the filter, and optimizing the one or more parameters to achieve a specific desired frequency response of the filter. | 03-24-2011 |

20110153704 | Filter - Provided is an FIR filter capable of obtaining predetermined characteristics with a small number of input taps, delay circuits, and multipliers and achieving an improved response and low cost. In a low-pass filter, a band-pass filter, and a high-pass filter based on an FIR filter, a basic filter is configured that gives a basic impulse response function and has a filter coefficient determined from the impulse response function. Filters having different frequency characteristics are configured by changing the time scale or frequency scale of the basic filter. These filters having different frequency characteristics are combined in a cascade form or a step form, thereby constructing an FIR filter having a small number of taps. | 06-23-2011 |

20100030830 | Method of predicting of signal processes via separation on band-limited and high-frequency components - A method of prediction is suggested for the characteristics of future values of processes that can be expressed as integrals over future times with different weight functions (kernels), or as anticausal convolution integrals. In particular, all band-limited processes processes are predictable in this sense, as well as high-frequency processes, with zero energy at low frequencies. In addition, process of mixed type can still be predicted using low-pass filter and high-pass filter for this process, to provide separation on low-band and high frequency processes. It is allowed that an outcome of low-pass filter be not a purely band-limited process, but have exponential decay of energy on high frequencies. The algorithm suggested consists of two blocks: separation of a process on band-limited and high-frequency components, and approximation of the transfer function of the anticausal integral that has to be predicted by transfer functions for causal convolution integrals. This approximation has to be done separately for high frequency domain and low band domain. | 02-04-2010 |

20120158809 | Compensation Filtering Device and Method Thereof - 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. | 06-21-2012 |

20120173600 | APPARATUS AND METHOD FOR PERFORMING A COMPLEX NUMBER OPERATION USING A SINGLE INSTRUCTION MULTIPLE DATA (SIMD) ARCHITECTURE - 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. | 07-05-2012 |

20100057821 | Digital Filter - Methods and apparatuses in which 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. | 03-04-2010 |

20090083353 | TRANSITIONING A FILTER FUNCTION OF A TWO-PORT LATTICE-FORM PLANAR WAVEGUIDE OPTICAL DELAY LINE CIRCUIT FILTER FROM A START FILTER FUNCTION TO A TARGET FILTER FUNCTION - Optically coherent, two-port, serially cascaded-form optical delay line circuits can realize arbitrary signal processing functions identical to those of FIR digital filters with complex filter coefficients whilst maintaining a maximum optical transmission characteristic of 100%. The invention provides an iterative process for transitioning in a step-wise manner a filter function of an optical delay line circuit filter from a start filter function to a target filter function. The invention also describes a dynamic gain equalizer incorporating an optical delay line circuit filter. | 03-26-2009 |

20120185523 | FILTER DEVICE - 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. | 07-19-2012 |

20120323983 | METHOD AND DEVICE FOR GENERATING A FILTER COEFFICIENT IN REAL TIME - 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. | 12-20-2012 |

20120331026 | Digital Filter - 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. | 12-27-2012 |

20080222228 | BANK OF CASCADABLE DIGITAL FILTERS, AND RECEPTION CIRCUIT INCLUDING SUCH A BANK OF CASCADED FILTERS - The present invention relates to a bank of digital filters that can be cascade connected. It also relates to a reception circuit comprising such a bank of cascaded filters. With the digital filter being sampled at a given sampling frequency Fs, the bank of cascadable digital filters has: at the input, a frequency transposition circuit ( | 09-11-2008 |

20130097212 | Low Power and Low Memory Single-Pass Multi-Dimensional Digital Filtering - Disclosed are new approaches to Multi-dimensional filtering with a reduced number of memory reads and writes. In one embodiment, a filter includes first and second coefficients. A block of a data having width and height each equal to the number of one of the first or second coefficients is read from a memory device. Arrays of values from the block are filtering using the first filter coefficients and the results filtered using the second coefficients. The final result may be optionally blended with another data value and written to a memory device. Registers store results of filtering with the first coefficients. The block of data may be read from a location including a source coordinate. The final result of filtering may be written to a destination coordinate obtained by rotating and/or mirroring the source coordinate. The orientation of arrays filtered using the first coefficients varies according to a rotation mode. | 04-18-2013 |

20080201396 | Signal processing apparatus and the correcting method - A signal processing apparatus, comprising: 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 comprises 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. | 08-21-2008 |

20120284318 | Digital Filter Implementation for Exploiting Statistical Properties of Signal and Coefficients - 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. | 11-08-2012 |

20120066280 | Asynchronous Sample Rate Conversion Using A Polynomial Interpolator With Minimax Stopband Attenuation - Methods for sample rate conversion are provided that use a polynomial interpolator with minimax stopband attenuation. A method for sample rate conversion of an input signal is provided that uses a time-varying polyphase filter having a discrete polyphase index m. Another method for sample rate conversion of an input signal is provided that uses a time-varying polyphase filter having a continuous polyphase index τ. In these methods, an output time index is mapped to an input sample index and the polyphase index, the polynomial coefficients of a polyphase filter are computed using the polyphase index, and the polyphase filter is applied to an input sample at the input sample index to generate the output sample at the output time index. | 03-15-2012 |