Abstract: The filter device includes 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, and a processor for selecting either the first set of filter coefficients or the second set of filter coefficients upon the basis of the first correlation value and the second correlation value for filtering the input signal.

Abstract: Various embodiments of systems and methods for pattern recognition of a distribution function are described herein. An influenced distribution function corresponding to an influenced attribute is compared with other distribution functions corresponding to other attributes. Based on the comparison, a correlation is determined between the influenced distribution function and an influencing distribution function from the other distribution functions. Based on the determination, a raw distribution function corresponding to an influenced attribute is extracted using the influenced distribution function and the influencing distribution function. The extracted raw distribution function and the influencing distribution function may be classified.

Abstract: 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.

Abstract: An adaptive filter configured to filter an input signal comprises Fourier transforming unit configured to transform the input signal into a frequency domain signal upon the basis of the Fourier transform, weighting unit configured to weight at least a portion of the frequency domain signal with a filter coefficient of the dispersion filter in frequency domain to obtain a filtered signal in frequency domain, correlating unit configured to correlate the filtered signal in frequency domain to obtain a correlation value, and adaptation unit configured to adapt the filter coefficient upon the basis of the correlation value.

Abstract: A FIR transmit architecture uses multiple driver divisions to allow signals with different delays to be summed into the output signal by the driver itself. The architecture includes a first multiplexer, a plurality of delay cells, a plurality of sign blocks, a switch block, a second multiplexer, and a plurality of drivers.

Abstract: Adaptive filtering is used to substantially cancel distortion in radio frequency (RF) signals. Such adaptive filtering can be used in an RF transmitting module to pre-compensate an RF signal with compensation (inverse) distortion to cancel inherent transmission path distortion from the RF signal. Adaptive filtering can also be used in a multi-carrier RF receiving module to cancel from a given carrier signal distortion due to cross talk from adjacent carrier signals. Adaptive filtering in an RF transceiver can be used to cancel from a received RF signal distortion arising from leakage of a transmit signal into the receive path.

Abstract: The document relates to modulated sub-sampled digital filter banks, as well as to methods and systems for the design of such filter banks. In particular, the present document proposes a method and apparatus for the improvement of low delay modulated digital filter banks. The method employs modulation of an asymmetric low-pass prototype filter and a new method for optimizing the coefficients of this filter. Further, a specific design for a (64) channel filter bank using a prototype filter length of (640) coefficients and a system delay of (319) samples is given. The method substantially reduces artifacts due to aliasing emerging from independent modifications of subband signals, for example when using a filter bank as a spectral equalizer. The method is preferably implemented in software, running on a standard PC or a digital signal processor (DSP), but can also be hardcoded on a custom chip.

Abstract: 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.

Abstract: The present disclosure provides a frequency offset compensation technique for use by a multi-carrier receiver. Compared with legacy receiver designs, the technique introduces a new function which correlates various frequency offsets of the different carriers involved in multi-carrier signal transmission and reception. This new function can be coupled with an Automatic Frequency Control (AFC) function which can then utilize the correlation information provided by the new function to achieve significantly better frequency error/offset estimation when the carriers are correlated.

Abstract: 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.

Abstract: An asynchronous sample rate converter prevents the folding back of a signal in the passband of an input sample rate into the passband of the output sample by adaptively controlling the decimation rate. The ASRC includes an adaptive decimation rate controller that selectably controls a decimation filter based on the ratio of the input sampling rate to the output sampling rate. By adaptively controlling the decimation rate in the ASRC, a significant amount of area and power is saved.

Abstract: 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.

Abstract: 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, ?1. The first signal path is configured to receive an output from a power amplifier. A second signal path is in parallel with the first signal path has a second delay, ?2, and a first scalar multiplier, ?1. A third signal path in parallel with the first and second signal paths has a third delay, ?3, and a second scalar multiplier, ?2. The second and third signal paths are configured to receive a sum of the output of the power amplifier and a reference signal. Delays ?1, ?2, and ?3 are integer multiples of a period of time. An output is a sum of a signal from each of the first, second and third signal paths.

Abstract: The system and method for least mean fourth adaptive filtering is a system that uses a general purpose computer or a digital circuit (such as an ASIC, a field-programmable gate array, or a digital signal processor that is programmed to utilize a normalized least mean fourth algorithm. The normalization is performed by dividing a weight vector update term by the fourth power of the norm of the regressor.

Abstract: Systems, methods, devices, circuits for data processing, and more particularly to data processing including operational marginalization capability, and/or operational improvement capability.

Abstract: The present invention is related to systems and methods for adaptive parameter modification in a data processing system.

Abstract: A spectrum agile radio having one or more variable digital filters is described. To quickly, yet accurately, retune the digital filter(s) a windowing function is applied to an ideal filter characteristic for each of one or more desired frequency bands to generate filter coefficients. Transitioning between coefficients of a previous filter and a current filter is handled to avoid problems associated with discontinuities in the signal processing.

Abstract: A system for estimating clock frequency offset and sampling clock offset in a communication system is provided. A receiver is configured to receive a communication signal having been transmitted from a transmitter via a communication channel. The receiver has a signal processor, wherein the signal processor is configured to generate an estimate of a carrier frequency offset and an estimate of a sampling clock offset from the received communication signal by: extracting a vector of pilot symbols from the received signal; performing equalization on the pilot symbols; performing clock frequency offset and sampling clock offset compensation on the pilot symbols; generating the estimate of a carrier frequency offset by estimating a common phase rotation using a first Taylor series approximation; and generating the estimate of the sampling clock offset by estimating phase differences between pairs of pilot symbols using a second Taylor series approximation.

Abstract: The invention relates to methods for synchronizing a device to a signal containing a train of pulses representing a programmable number of repetitions of a predetermined code. Pulse position and code phase are searched for. In a first aspect, a programmable number of samples is taken per pulse frame in function of the available number of repetitions of said predetermined code. In a second aspect, the method comprises a signal detection stage from which, after performing a confirmation stage, information can be kept for a subsequent stage. In a third aspect, only a limited number of rotated versions of the predetermined code are checked, using a presumed code phase which is kept from a preceding stage.

Abstract: A notch filter includes first, second, and third difference modules, a delay module, and a gain module. The first difference module calculates a first difference between a control signal in a pulse width modulation (PWM) system and a feedback value. The second difference module calculates a second difference between the first difference and a delayed second difference. The delay module generates the delayed second difference by introducing a one period delay to the second difference, wherein the period is based on a Nyquist frequency of the control signal. The gain module generates the feedback value by applying a gain to the delayed second difference. The third difference module generates a filtered control signal by calculating a difference between the signal and the feedback value, wherein the filtered control signal is used to control an operating parameter of the PWM system.

Abstract: A Symbol vector for RF transmission after multiplexing onto a subset of a set of subcarriers using OFDM is transformed to the time domain. A first time domain cancellation vector is generated from a basis vector that has the same dimensionality as the symbol vector. In the frequency domain the basis vector has a substantially zero value in each of a first subset of sub-carriers of the symbol vector and in the time domain the difference between a first element of the basis vector having the highest value and a second element of the basis vector having the next highest value is maximized. The first cancellation vector is subtracted from the symbol vector to produce modified symbol vector having reduced Peak-to-Average Power Ratio (PAPR). A second cancellation vector is generated using the modified symbol vector and is used to produce a second modified symbol vector having reduced PAPR.

Abstract: A reconfigurable digital signal filter processing unit for use in a communication device is provided. The reconfigurable filters processor can implement different filter topologies to adapt to a range or wireless technology characteristics. The reconfigurable filter processor comprises a plurality of filter blocks whose inputs can be selected based on the desired configuration of the filter. Each filter block applies a transfer function to a received signal to achieve a desired filtering function.

Abstract: A method for updating the processing capacity of an encoder or decoder to use a modulated transform having a size greater than a predetermined initial size is provided, particularly, where the encoders or decoders are for storing an initial prototype filter defined by an ordered set of initial size coefficients. A step is provided for constructing a prototype filter of a size greater than the initial size to implement the modulated transform of the greater size by inserting at least one coefficient between two consecutive coefficients of the initial prototype filter.

Abstract: A method for designing a set of sub-band FIR filters, where each FIR filter has a number of filter coefficients and is connected to an adjustable delay line. The method includes dividing an input signal into a number of sub-band signals, where a spectrum of the input signal comprises spectra of the sub-band signals; providing a respective goal sub-band signal for each sub-band dependent on a goal signal; filtering and delaying each sub-band signal using a corresponding FIR filter and delay line to provide filtered signals; providing error signals for each sub-band dependent on the filtered signals and the corresponding goal signals; adapting the filter coefficients of each sub-band FIR filter such that the respective filtered signal approximately matches a corresponding goal sub-band signal; and changing a respective delay of the delay line for each sub-band to reduce or increase a first quality criterion.

Abstract: A signal is decomposed into different components using a transform, with the components then being separately presented to a person in a manner that produces a different cognitive experience than would have resulted from either (a) presentation of the original signal, or (b) presentation of a fully synthesized (inverse transformed) signal.

Abstract: A method for estimating a projected path of travel for a vehicle on a road includes monitoring a plurality of sensor inputs, determining a road geometry in front of the vehicle based upon the monitored sensor inputs, determining a vehicle position in relation to the road geometry based upon the monitored sensor inputs, determining a plurality of particle points in front of the vehicle representing a potential path of travel from the road geometry and the vehicle position, and utilizing iteratively determined ones of the plurality of particle points to navigate the vehicle including omitting ones of the plurality of particle points passed by the vehicle.

Abstract: A filter auto-calibration system includes a multi-clock module. The multi-clock module includes a multi-clock generator that is configured to generate a clock signal with a variable frequency based on a channel setting. There is at least one filter to be calibrated. An auto-calibration control module is configured to control calibration of the at least one filter based on the channel setting. The multi-clock module is configured to supply the variable frequency clock signal to the at least one filter and to the auto-calibration control module, and the at least one filter is coupled to the auto-calibration control module.

Abstract: Provided is an apparatus and method for analyzing identification (ID) signals by converting radio frequency (RF) signals which are transmitted with an ID signal added thereto by a transmitting part, e.g., a plurality of transmitters or repeaters, into signals of a desired band; creating ID signals that are identical to the ID signals added to the RF signals; calculating correlation values between the converted signals and the created ID signals based on partial correlation; and extracting channel profile of multi-path signals caused by a channel between the transmitting part and the ID signal analyzing apparatus from the correlation value. The technology of the present research is applied to broadcasting and communication.

Abstract: Embodiments are related to systems and methods for data processing, and more particularly to systems and methods for calibration during data processing.

Abstract: A system, method, and computer program product are provided for reducing ripple in a filter bank. For a given fixed number of linearly-spaced filters in the filter bank, a monotonically increasing function for a Q-factor is specified for the filter bank. Adjustments to each filter's Q-factor based on the Q-factor function are made in order to produce a nearly constant filter bank ripple throughout the filter bank's frequency response range.

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

Abstract: A signal is decomposed into different components using a transform, with the components then being separately presented to a person in a manner that produces a different cognitive experience than would have resulted from either (a) presentation of the original signal, or (b) presentation of a fully synthesized (inverse transformed) signal.

Abstract: The invention provides a filtering device with hierarchical structure making it possible to carry out finite impulse response and infinite impulse response linear filtering functions and which can be combined with one or more devices of the same type. The device includes at least one first and one second filtering module having means for carrying out filtering functions with N configurable coefficients. A first subset of the N coefficients of a module is configured to carry out nonrecursive filtering functions, a second subset of the coefficients configured to carry out recursive filtering functions, one or more feedback loops able to be activated per module, at least one result sample of the filtering being generated at each clock cycle. The invention also provides a reconfigurable filtering device using at least two filtering devices with the hierarchical structure.

Abstract: Described embodiments receive a signal by a set of fixed taps and a set of floating taps of a receiver, each tap corresponding to a detected symbol. Each of the floating taps is stored in a corresponding shift register to account for process, operating voltage and temperature (PVT) variations of the receiver without calibration of delay elements. Multiplexing logic selects (i) corresponding floating taps for equalization by coupling selected floating taps to the outputs of the fixed taps, and (ii) different phases of each possible floating tap position. The multiplexing logic prunes and/or amalgamates the phases of each possible floating tap position and selects floating taps based on a magnitude of each phase. A combiner adjusts each output value of the fixed taps and the selected floating taps by a corresponding tap-weight, combines the adjusted values into an output signal and subtracts the output signal from the input signal.

Abstract: A signal receiver is arranged to process a portion of a received signal in each of a plurality of filter branches and to determine a quality metric for each branch. The configuration of a second filter for processing the entirety of the received signal is subsequently performed according to the determined quality metrics.

Abstract: The value of a median or other rank of interest in a dataset is efficiently determined. Each active bit of the dataset is serially processed to compute one bit of the output value from each bit of the input dataset. If any sample in the dataset has an active bit that differs from the determined output value for that bit, then that sample can be marked as no longer in consideration. After an active bit has been processed, the data for that bit may be discarded or subsequently ignored. These techniques allow the rank value to be efficiently determined using pipelined logic in a configurable gate array (CGA) or the like. Further implementations may be enhanced to compute clipped means, to identify “next highest” or “next lowest” values, to reduce quantization errors through less-significant bit interpolation, to simultaneously process multiple values in a common pipeline, or for any other purpose.

Abstract: A circuit for multiplying a digital signal by a variable gain, controlled in dependence on a digital gain control value. The circuit comprises: a multiplier input for receiving the digital signal; a multiplier output for outputting the digital signal multiplied by the gain; a plurality of multiplier stages each arranged to multiply by a respective predetermined multiplication factor; and switching circuitry arranged so as to apply selected ones of the multiplier stages in a multiplication path between the input and output, in dependence on the digital gain control value. The multiplication factors are arranged such that binary steps in the digital gain control value result in logarithmic steps in said gain.

Abstract: A filter includes, in at least one aspect, one or more differential inputs to receive one or more differential input signals, a first filter stage to provide one or more primary poles corresponding to a first frequency associated with the one or more differential input signals, a second filter stage coupled with the first filter stage to provide one or more secondary poles corresponding to a second frequency associated with the one or more differential input signals, the second frequency and the first frequency having a predefined relationship, and one or more differential outputs coupled with the second filter stage to provide one or more differential output signals.

Abstract: Physiological signals are denoised. In accordance with an example embodiment, a denoised physiological signal is generated from an input signal including a desired physiological signal and noise. The input signal is decomposed from a first domain into subcomponents in a second domain of higher dimension than the first domain. Target subcomponents of the input signal that are associated with the desired physiological signal are identified, based upon the spatial distribution of the subcomponents. A denoised physiological signal is constructed in the first domain from at least one of the identified target subcomponents.

Abstract: A data processing method, an equalizer, and a receiver in a wireless communication system including a relay station are provided. The data processing method includes: receiving a base station signal from a base station; receiving a relay station signal from a relay station; determining a propagation delay between the base station signal and the relay station signal; generating an equalizing signal in which interference generated between the base station signal and the relay station signal is alleviated in consideration of the propagation delay; and recovering information bits transmitted by the base station from the equalizing signal. According to an exemplary embodiment of the present invention, it is possible to alleviate performance deterioration due to an interference problem generated in a relay system.

Abstract: A method for compressive domain filtering and interference cancelation processes compressive measurements to eliminate or attenuate interference while preserving the information or geometry of the set of possible signals of interest. A signal processing apparatus assumes that the interfering signal lives in or near a known subspace that is partially or substantially orthogonal to the signal of interest, and then projects the compressive measurements into an orthogonal subspace and thus eliminate or attenuate the interference. This apparatus yields a modified set of measurements that can provide a stable embedding of the set of signals of interest, in which case it is guaranteed that the processed measurements retain sufficient information to enable the direct recovery of this signal of interest, or alternatively to enable the use of efficient compressive-domain algorithms for further processing.

Abstract: An arithmetic logic unit (ALU) including a first routing grid connected to multiple data lanes to drive first data to the data lanes. A second routing grid is connected to the data lanes to drive second data to the data lanes. Each of the data lanes include multiple, e.g. N, functional units with first inputs from the first routing grid and second inputs from the second routing grid. The functional units compute pairwise a function of the respective first data on the respective first inputs and the respective second data on the respective second inputs. Each of the data lanes include a reduction unit with inputs adapted to receive K? bits per word from the functional units. The reduction unit is configured to perform a reduction operation configured to output an output result having a reduced number J? bits per word, wherein J? is less than N multiplied by K?.

Abstract: A lookup table structure having multiple lookup tables is configured to include a quaternary adder. In particular examples, an adaptive logic module (ALM) including a fracturable lookup table (LUT) is configured to include a quaternary (4-1) adder. In some examples, only an XOR gate, an AND gate, two single bit 2-1 multiplexers, and minor connectivity changes to a LUT structure supporting a ternary (3-1) adder are needed to support 4-1 adders. Binary (2-1) and ternary adders are still supported using the original signal flows, as the ternary adder feature can be easily multiplexed out.

Abstract: A time domain filter receives a double sideband (DSB) input in the frequency domain and compresses this input into a time domain signal filtered by a time gate for providing a time filtered signal that is then expanded back into the frequency domain as a single sideband (SSB) output with one sideband being filtered by the time gate for translating DSB signals into SSB signals well suited for communicating chirped modulated signals as SSB signals along an electrical line or optical fiber without dispersive nulling of the communicated signal.

Abstract: An RF signal reception device including: a transposition device of signals of frequency fRF to a first intermediate frequency IF1<fRF; a first bandpass filter centered on IF1; a sampler at a frequency fs<IF1; a second discrete-time filter centered on a second intermediate frequency IF2=?·fs/M+fs/(M·n); a decimation device of a factor M; an analog-digital convertor to operate at a frequency fs/M; where ?, n and M are strictly positive real numbers chosen such that: ?<fs/(2·BWch·M), and BWch/2<fs/M·n), with BWch: bandwidth of a channel of the received RF signals.

Abstract: Embodiments of the invention are directed to system and method that enable relatively low power dissipation by scheduling operations of multiply accumulators chain of two or more multiply accumulators units by delivering an output result of a first multiply accumulator of the chain as an input to a second subsequent multiply accumulator of the chain.

Abstract: A computer-implemented method for calculating a time-domain impulse response with improved causality based on a first spectrum in a frequency domain is disclosed. The first spectrum may be band-limited. The method may calculate a first time-domain impulse response from the first spectrum. The method may remove a non-causal portion of the first system time-domain impulse response to obtain a second time-domain impulse response, and calculate a second spectrum of the second time-domain impulse response. The method may further modify the second spectrum by adding a causal signal such that a difference between the second spectrum and the first spectrum is reduced. The method may also calculate the time-domain impulse response with improved causality from the modified second spectrum.

Abstract: 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 ? are provided.

Abstract: A receiver for receiving a layer-modulated signal includes: a base layer decoding unit configured to calculate a bit metric including code bit information of a base layer based on the reception signal and decode an information bit of the base layer; and at least one enhancement layer decoding unit configured to decode an information bit of an upper layer of a lower layer based on the decoding results of the lower layer, wherein the base layer decoding unit and the at least one enhancement layer decoding unit are sequentially connected according to the order of the corresponding layers.

Abstract: 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.