| Entries |
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| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 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 |
| 20130054663 | Determining Coefficients For Digital Low Pass Filter Given Cutoff And Boost Values For Corresponding Analog Version - Methods and apparatus are provided for determining coefficients for a digital low pass filter, given cutoff and boost values for a corresponding analog version of the digital low pass filter. Coefficients are determined for a digital low pass filter by obtaining cutoff and boost values for a corresponding analog version of the digital low pass filter; and determining the coefficients for the digital low pass filter based on the obtained cutoff and boost values. The coefficients can be determined, for example, by generating a transfer function, H(s), for the corresponding analog version using the obtained cutoff and boost values: transforming the transfer function, H(s), to a frequency domain characterization, H(z), using one or more bilinear transforms to obtain a plurality of coefficients for an infinite impulse response (IIR) filter; generating the IIR filter using the plurality of coefficients for the IIR filter; and applying an impulse to the IIR filter to obtain the one or more coefficients for the digital low pass filter. In another variation, the coefficients are pre-computed and obtained from a look-up table. | 02-28-2013 |
| 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 |
| 20130110897 | DIGITAL FILTER HAVING IMPROVED ATTENUATION CHARACTERISTICS | 05-02-2013 |
| 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 |
| 20130262545 | DIGITAL FILTER CIRCUIT AND DIGITAL FILTER CONTROL METHOD - [Objective] | 10-03-2013 |
| 20130275483 | FILTER SYSTEM - A filter system with infinite impulse response is provided. The filter system has a transfer function that includes at least one pair of first order polynomial fractions. In one embodiment, the poles and/or the zeros of the pair of polynomial fractions are complex conjugates, respectively. The gain of the transfer function is realized, for example, by virtue of at least two separate multiplier elements | 10-17-2013 |
| 20130297665 | Method and Apparatus for Signal Filtering and for Improving Properties of Electronic Devices - The present invention relates to nonlinear signal processing, and, in particular, to adaptive nonlinear filtering of real-, complex-, and vector-valued signals utilizing analog Nonlinear Differential Limiters (NDLs), and to adaptive real-time signal conditioning, processing, analysis, quantification, comparison, and control. More generally, this invention relates to methods, processes and apparatus for real-time measuring and analysis of variables, and to generic measurement systems and processes. This invention also relates to methods and corresponding apparatus for measuring which extend to different applications and provide results other than instantaneous values of variables. The invention further relates to post-processing analysis of measured variables and to statistical analysis. The NDL-based filtering method and apparatus enable improvements in the overall properties of electronic devices including, but not limited to, improvements in performance, reduction in size, weight, cost, and power consumption, and, in particular for wireless devices, NDLs enable improvements in spectrum usage efficiency. | 11-07-2013 |
| 20130304783 | COMPUTER-IMPLEMENTED METHOD FOR ANALYZING MULTIVARIATE DATA - A computer-implemented method for analyzing multivariate data comprising a plurality of samples of each of a plurality of measurement variables is disclosed. The method comprises, for a first subset | 11-14-2013 |
| 20130332498 | METHOD AND APPARATUS FOR EFFICIENT FREQUENCY-DOMAIN IMPLEMENTATION OF TIME-VARYING FILTERS - Embodiments are directed to efficient frequency-domain implementations of time-varying FIR filters. More specifically, time-varying FIR filters according to embodiments exploit the duality of the fast Fourier transform that windowing in the time domain equals convolution in the frequency domain. In one embodiment, convolution of the output of the FIR filter and a desired windowing function is performed in the frequency domain instead of taking the output of the FIR filter in the frequency domain, converting this output the time domain via an IFFT, and then windowing this output in the time domain before again converting back to the frequency domain. As long as the windowing function has certain characteristics, then the time-varying FIR filter is computationally efficient and introduces minimal audible artifacts into the output of the filter. Concepts described herein are discussed in terms of audio signals and systems but are not limited to audio signals and systems. | 12-12-2013 |
| 20130332499 | RECONFIGURABLE VARIABLE LENGTH FIR FILTERS FOR OPTIMIZING PERFORMANCE OF DIGITAL REPEATER - The invention addresses the problem of parameter optimization for best filter performance and, in particular, the influence from the requirements on radio or fiber to radio repeaters utilizing those filters, that often proves to be conflicting for an FIR filter. The FIR filters are implemented in a programmable circuit and are not thereby restricted for use in communication repeaters although this particular usage may put the most serious restrictions on the filter performance. Within the imposed constraints, this disclosure illustrates a method to strike a middle ground while minimizing the trade-offs. The advantage of the concept presented allows the choice of a suitable filter pertaining to a particular traffic configuration, meaning a particular choice of individually filtered frequency bands set at different gain and intended to support a diversity of traffic formats. The disclosed approach banks on the reconfigurable variable length FIR filter architectures. Implementation architecture and results in brief are also presented. | 12-12-2013 |
| 20130332500 | SIGNAL PROCESSING APPARATUS, SIGNAL PROCESSING METHOD, STORAGE MEDIUM - To obtain a high-quality enhanced signal, disclosed is a signal processing apparatus including a transform unit that transforms a mixed signal in which a first signal and a second signal coexist, into a phase component and a magnitude component or power component for each frequency, a first control unit that replaces the phase component of a predetermined frequency, a second control unit that modifies the magnitude component or power component of the predetermined frequency in accordance with the amount of a change of the magnitude component or power component that arises from replacement by the first control unit, and a reconstruction unit that reconstructs the phase component replaced by the first control unit and the magnitude component or power component modified by the second control unit. | 12-12-2013 |
| 20130346460 | METHOD AND DEVICE FOR FILTERING A SIGNAL AND CONTROL DEVICE FOR A PROCESS - A method for filtering a signal is proposed. A noisy input signal is continuously examined in order to determine whether the input signal it is within or outside a deadband. The deadband width and the zero point of the deadband are continuously adapted to the noise power of the input signal depending on the time behavior of the input signal and a predefined system time constant. At least one filtered output signal is continuously output, such as a deadband signal, which substantially corresponds to a smoothed input signal. | 12-26-2013 |
| 20140019504 | METHOD AND DEVICE FOR FILTERING DURING A CHANGE IN AN ARMA FILTER - A method and device are provided for filtering digital audio signals using at least one ARMA filter, particularly during a filter change. The method includes the following steps: a step of receiving a first request to change filtering to or from filtering by a first ARMA filter; and, in response to the first request, a step of gradually switching, at each of a plurality of cascaded first filtering blocks, between digital-signal filtering by a first basic filtering cell and digital-signal filtering by another associated basic filtering cell, the first basic filtering cells of the plurality of first filtering blocks factorizing the first filter. | 01-16-2014 |
| 20140040339 | Cascaded Digital Filters with Reduced Latency - A filter and method for filtering a signal are disclosed. The filter is equivalent to a plurality of bi-quad filters connected in series, and is implemented on a digital processor that receives a sequence of signal values at a sampling rate characterized by a sampling interval and generates a filtered signal value upon receiving each received signal value. The filter has a latency that is less than the sampling interval. The filtered values can be generated by adding a term to a received signal value and multiplying the sum by a gain constant that depends on the filter constants. The added term does not depend on the current received signal value. The filter can be implemented in fixed-point integer arithmetic. | 02-06-2014 |
| 20140059101 | OPTIMIZED METHOD OF BIQUAD INFINITE-IMPULSE RESPONSE CALCULATION - A method of performing an infinite-impulse response digital filter includes switching address pointers between a first instance of the filter and a second instance of the filter; where the first and second instances represent the same filter. A first instance of the filter executes operations sequentially multiplying a current input data value, and first and second previous input data values, with corresponding ones of a first set of filter coefficients, using a multiplier; and a second instance of the filter executes operations sequentially multiplying first and second previous intermediate data values with corresponding ones of a second set of filter coefficients, using the multiplier. Switching between first and second instances of the filter occurs at each data input value or frame according to an alternating signal. | 02-27-2014 |
| 20140082038 | Passive switched-capacitor filters conforming to power constraint - Passive switched-capacitor (PSC) filters are described herein. In one design, a PSC filter implements a second-order infinite impulse response (IIR) filter with two complex first-order IIR sections. Each complex first-order IIR section includes three sets of capacitors. A first set of capacitors receives a real input signal and an imaginary delayed signal, stores and shares electrical charges, and provides a real filtered signal. A second set of capacitors receives an imaginary input signal and a real delayed signal, stores and shares electrical charges, and provides an imaginary filtered signal. A third set of capacitors receives the real and imaginary filtered signals, stores and shares electrical charges, and provides the real and imaginary delayed signals. In another design, a PSC filter implements a finite impulse response (FIR) section and an IIR section for a complex first-order IIR section. The IIR section includes multiple complex filter sections operating in an interleaved manner. | 03-20-2014 |
| 20140172934 | Systems and Methods for Data Retry Using Averaging Process - Embodiments are related to systems and methods for data processing, and more particularly to systems and methods for calibration during data processing. | 06-19-2014 |
| 20140280415 | FEED-FORWARD LINEARIZATION WITHOUT PHASE SHIFTERS - A method and system for providing a finite impulse response, FIR, filter in a power amplification system are disclosed. An FIR filter includes a first signal path having a first delay, τ | 09-18-2014 |
| 20140280416 | METHOD OF REMOVING INCOHERENT NOISE - Noise is removed from a data set by performing an integral transform operation that converts instances of noise into identifiable artefacts in the transformed data set. A model of the artefacts is constructed by creating a full-domain or partial-domain noise model and performing the same integral transform operation on the noise model. The resulting transformation of the noise model is adaptively subtracted from the transformed data set to remove the noise. The adaptive subtraction may employ a least-square error filter. | 09-18-2014 |
| 20140297706 | ORTHOGONAL TRANSFORM APPARATUS, ORTHOGONAL TRANSFORM METHOD, ORTHOGONAL TRANSFORM COMPUTER PROGRAM, AND AUDIO DECODING APPARATUS - An orthogonal transform apparatus includes: an interchanging unit which interchanges MDCT coefficients contained in a first half of a prescribed interval with MDCT coefficients contained in a second half thereof; an inverting unit which inverts the sign of the MDCT coefficients contained in the second half of the prescribed interval after the interchange; an inverse cosine transform unit which computes the real components of QMF coefficients by applying an IMDCT using FFT to the MDCT coefficients contained in the first half and the sign-inverted MDCT coefficients contained in the second half; an inverse sine transform unit which computes the imaginary components of the QMF coefficients by applying an IMDST using FFT to the MDCT coefficients contained in the first half and the sign-inverted MDCT coefficients contained in the second half; and a coefficient adjusting unit which computes the QMF coefficients by combining the real components with the imaginary components. | 10-02-2014 |
| 20140379771 | DIGITAL FILTER CIRCUIT AND DIGITAL FILTER PROCESSING METHOD - A digital filter circuit includes an FFT circuit ( | 12-25-2014 |
| 20150019608 | DIGITAL FILTER CIRCUIT, DIGITAL FILTER PROCESSING METHOD AND DIGITAL FILTER PROCESSING PROGRAM STORAGE MEDIUM - Reduction of a circuit size and power consumption for performing digital filtering processing in a frequency domain is realized. The digital filter circuit includes: a complex conjugate generation unit for generating a second complex number signal including conjugate complex numbers of all complex numbers included in a first complex number signal of the frequency domain generated by converting a complex number signal of a time domain by Fourier transform; a filter coefficient generation unit for generating a first and a second frequency domain filter coefficient of a complex number from a first, a second and a third input filter coefficient of a complex number having been inputted; a first filtering unit for performing filtering processing to the first complex number signal by the first frequency domain filter coefficient, and outputting a third complex number signal; a second filtering unit for performing filtering processing to the second complex number signal by the second frequency domain filter coefficient, and outputting a fourth complex number signal; and a complex conjugate combining unit for combining the third complex number signal and the fourth complex number signal, and generating a fifth complex number signal. | 01-15-2015 |
| 20150088951 | Method and Apparatus For Hybrid Digital Filtering - New hybrid filters are presented based on time and transform domain structures. The hybrid filters have a combined benefit from the advantages obtained by the time and transform domain structures. The overall efficiencies are drawn from combining the pre- and post-processing of the time domain and block based transform domain structures. Further improvements are obtained by interchanging block construction and transforms with linear operations in the pre- and post-processors. The hybrid structures apply to single input, single output, multiple input, and multiple output structures. For the multi input and multi output structures further improvements are obtained by having common processing blocks for the input(s) and common processing blocks for the output(s). They hybrid filters are also efficient in topologies where filter outputs are combined via linear operation(s) generating combined results. The efficiencies of the new hybrid filter may lead to significant fardware, power, silicon area, or somputational savings. | 03-26-2015 |
| 20150127696 | MULTI-STAGE FILTER PROCESSING DEVICE AND METHOD - The present invention addresses the problem of reducing a circuit scale without causing a reduction in processing efficiency. This multi-stage filter processing method measures, at each stage, either the number of input data or the number of intermediate data that is generated by filter calculation processing during the stages before the final stage is reached. Coefficient data regulating for each stage the number of data sufficient to perform the filter calculation processing is held. Input data or the intermediate data that is generated in a current stage is held in a memory until the number of data reaches the number of data sufficient to perform the filter calculation processing in the current stage, on the basis of the coefficient data. When the number of data has reached the number of data sufficient to perform the filter calculation processing, the filter calculation processing for the current stage is performed on the input data or the intermediate data that was held. | 05-07-2015 |
| 20150363359 | BUTTERFLY CHANNELIZER - A butterfly channelizer includes at least two stages. Each stage includes at least one dual-channel module configured to convert an input time domain signal into a second time domain signal of lower bandwidth. At least one clock is configured to generate a clock signal that drives the at least two stages. A first stage has a first number of dual-channel modules and a second stage following the first stage has a second number of dual-channel modules greater than the first number. | 12-17-2015 |
| 20150363360 | FAST FOURIER TRANSFORM DEVICE, FAST FOURIER TRANSFORM METHOD, AND RECORDING MEDIUM STORING FAST FOURIER TRANSFORM PROGRAM - [Problem] A fast Fourier transform method is provided that are able to input data to be processed or output processing results in no particular order. | 12-17-2015 |
| 20160020752 | ARCHITECTURE TO REJECT NEAR END BLOCKERS AND TRANSMIT LEAKAGE - A method and apparatus for minimizing transmit signal interference is provided. The method includes the steps of: receiving a signal and amplifying the received signal. The received signal is then mixed with an intermediate frequency signal to obtain a baseband modulated signal. The baseband modulated signal is first filtered in an RC filter. The resulting signal is then divided by a preselected amount and the first divided portion is sent to a main path of a biquad filter, which produces a first stage biquad filtered signal. The second portion of the divided signal is sent to an auxiliary path of the biquad filter, and produces a second filtered signal. The first and second signals are then recombined and sent to the second stage of the biquad filter, where further filtering takes place. | 01-21-2016 |
| 20160049923 | 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. | 02-18-2016 |
| 20160079961 | PRECISION FREQUENCY MONITOR - A precision frequency monitor provides a precision frequency monitor value (PFM) indicative of the precision of the frequency or period of an input reference signal. A first averaging module is responsive to the input reference signal to find an average frequency or period during successive predetermined time periods defining operational cycles. A second averaging module is responsive to an output of the first averaging module to average the output of the first averaging module over N operational cycles, where N is an integer, and output an updated PFM value every N operational cycles. An infinite impulse response (IIR) filter is responsive to the output of the first averaging module to filter the output of the first averaging module to output interim updated PFM values within each sequence of N operational cycles. | 03-17-2016 |
| 20160105160 | DIGITAL FILTER DEVICE AND SIGNAL PROCESSING METHOD - Provided is a digital filter device including data rearrangement means for executing rearrangement of input data and outputting rearranged data, intermediate data calculation means for processing the rearranged data input at a specific time and generating intermediate data, filter output first calculation means for calculating a first output value at the specific time by use of the intermediate data, delay means for delaying the rearranged data by processing time taken in the intermediate data calculation means and the file output first calculation means, and filter output second calculation means for inputting output values from the delay means and the filter output first calculation means, calculating a second output value at a time other than the specific time, and outputting a filter output value obtained by adding up the first and second output values. Consequently, it becomes feasible to reduce a circuit scale without impairing performance of a digital filter used in nonlinear compensation and realize nonlinear compensation by an LSI. | 04-14-2016 |
| 20160112033 | EFFICIENT IMPLEMENTATION OF CASCADED BIQUADS - An improved biquad infinite impulse response filter is shown that may be implemented in a very large instruction word digital signal processor as well as in other processing circuitry. The new filter structure modifies the feedback path in the filter, resulting in a significant reduction in execution cycles. | 04-21-2016 |
| 20160126933 | FINITE IMPULSE RESPONSE FILTER AND FILTERING METHOD - A finite impulse response (FIR) filter and a corresponding filtering method are provided. The FIR filter receives an input sequence. The input sequence includes a plurality of input values. The FIR filter includes at least one first adder, at least one multiplier, and a second adder. Each first adder performs multiple addition operations simultaneously in parallel. Each addition operation outputs a sum of two of the input values. Each multiplier performs multiple multiplication operations simultaneously in parallel. Each multiplication operation outputs a product of one of the sums and one of a plurality of coefficients of the FIR filter. The second adder outputs a total sum of the products. | 05-05-2016 |
| 20160134261 | SIGNAL PROCESSING APPARATUS, SIGNAL PROCESSING METHOD, AND SIGNAL PROCESSING SYSTEM - A signal processing apparatus includes: a filter; and a filter control circuit, wherein the filter control circuit is configured to: detect a power of signals output from the filter; determine one of a plurality of numerical ranges to which the power belongs; update a filter coefficient of the filter according to a determination result; count a number of the signals having the power of a first value or more; set an invalid area which becomes a target not to be determined for each of one or more boundaries between the plurality of numerical ranges when the number of the signals becomes a second value or more; and control a width of the invalid area based on the number of signals. | 05-12-2016 |
| 20160140083 | DIGITAL FILTER DEVICE, DIGITAL FILTERING METHOD, AND STORAGE MEDIUM HAVING DIGITAL FILTER PROGRAM STORED THEREON - In order to reduce the power consumed when using FFT processing and filtering in the frequency domain together, a digital filter device according to the present invention is provided with: a first filtering means for performing a first fast Fourier transformation using a first data sorting process, first filtering in the frequency domain, a first inverse fast Fourier transformation using a second data sorting process, and overlap removal on a first input block including overlapped data; a second filtering means for performing a second fast Fourier transformation, which simultaneously processes all data in a second input block including overlapped data, second filtering in the frequency domain, a second inverse fast Fourier transformation, which simultaneously processes all received filtered data, and overlap removal; and a data selection means for selecting either the first filtering means or the second filtering means, wherein the operation of the filtering means that is not selected by the data selection means is interrupted. | 05-19-2016 |
