A radio receiver comprises a matched filter bank and a decision unit. The matched filter bank has a plurality of filter modules for generating correlation-strength data from a sampled radio signal, each filter module being configured to cross-correlate the sampled signal with data representing a respective filter sequence. The decision unit is configured to use the correlation-strength data to generate a sequence of decoded symbols from the sampled signal. The matched filter bank and/or decision unit are configured to determine the value of each symbol in the sequence in part based on the value of a respective earlier decoded symbol from the sequence of decoded symbols.

A beam generator, a beam generating method, and a chip are provided. The beam generator comprises a first channel, a second channel, and a signal merging module; the first channel comprises a first-channel filter, the first-channel filter is used to filter an input signal to obtain a first filtered signal; the first filtered signal comprises a desired signal; the second channel comprises: a second-channel blocking module, used to block the desired signal in the input signal to obtain a blocked signal; a compensation filter, connected to the second-channel blocking module for compensating for the blocked signal to obtain a second filtered signal; and an adaptive filter connected to the compensation filter for adaptively filtering the second filtered signal to obtain a third filtered signal; the signal merging module is for merging the first filtered signal and the third filtered signal to obtain an output signal.

The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing a multimedia signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing filtering for multimedia signal having a plurality of subbands with a low calculation amount. To this end, provided are a method for processing a multimedia signal including: receiving a multimedia signal having a plurality of subbands; receiving at least one proto-type filter coefficients for filtering each subband signal of the multimedia signal; converting the proto-type filter coefficients into a plurality of subband filter coefficients; truncating each subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the length of at least one truncated subband filter coefficients being different from the length of truncated subband filter coefficients of another subband; and filtering the multimedia signal by using the truncated subband filter coefficients corresponding to each subband signal and an apparatus for processing a multimedia signal using the same.

This application relates to methods and apparatus for controlling slew-rate of components for outputting an analogue output signal. Described is a signal processing circuit having a forward signal path for receiving an input signal and outputting an analogue output signal. The signal processing circuit has a first component located in said forward signal path for outputting the analogue output signal. A predictor is configured to predict a required slew-rate for the first component based on the input signal and a controller is configured to controllably vary an output slew-rate limit of the first component based on the prediction of required slew-rate.

A digital interpolation filter delivering a series of output samples approximating a signal x(t) at sampling instants of the form (n+d)T s based on a series of input samples of the signal x(t) taken at sampling instants of the form nT s. Such a filter implements a transfer function in the Z-transform domain, H c<i/>d (Z−1), expressed as a linear combination between: a first transfer function H 1 d<i/>(Z−1) representing a Lagrange polynomial interpolation of the input samples implemented according to a Newton structure (100); and a second transfer function H 2 d (Z−1) representing another polynomial interpolation of the input samples implemented according to another structure comprising at least the Newton structure; the linear combination being a function of at least one real combination parameter c.

A method of processing power signals is provided. The method includes: receiving an analog poly-phase signal associated with power delivered using alternating current (AC); converting the analog poly-phase signal to a digital poly-phase signal sampled at a first sampling rate; detecting a fundamental frequency of the analog poly-phase signal; determining a second sampling rate, wherein the second sampling rate is based on and tracks the fundamental frequency; resampling the digital poly-phase signal at the second sampling rate; for each cycle of the resampled digital poly-phase signal: transforming the resampled digital poly-phase digital signal to a frequency-domain signal; calculating a phase angle of the reference voltage component; adjusting the resampled digital poly-phase signal by compensating the calculated phase angle; and transforming the adjusted resampled digital poly-phase signal to an updated frequency-domain signal using FFT; and calculating one or more measurements based on the updated frequency-domain signal.

Biquad stage systems and methods include receiving at biquad sections a signal sample, generating, by each biquad section, a pair of output values based on the signal sample, including a first value based on fixed-point processing path and a second value emulating a floating-point processing path, and accumulating the pair of output values from each of the plurality of biquad sections to generate an output signal. The biquad stage receives an N-bit input signal, which is processed by a biquad section. Delay elements delay the signal sample before input to other biquad sections. The delayed signal sample is input to the first processing path and the second processing path of a corresponding biquad stage. By performing the processing based on two paths, a more accurate result can be found when using a reduced word length in the multiply operations resulting in a lowering of the power consumption.

Trans-filter/Detectors are extremely sensitive circuits that recover exponentially modulated signals buried in noise. They can be used wherever Matched Filter/Coherent Detectors are used and operate at negative input signal-to-noise ratios to recover RADAR, SONAR, communications, or data signals, as well as reduce phase noise of precision oscillators. Input signal and noise is split into two paths where complementary derivatives are extracted. Outputs of the two paths are equal in amplitude and 180 degrees relative to each other at the band center frequency. The outputs are summed, causing stationary in-band noise to be reduced by cancellation while exponentially modulated signals are undiminished. Trans-filters are Linear Time Invariant circuits, have no noise x noise threshold and can be cascaded, increasing output signal-to-noise ratio prior to detection. Trans-filters are most sensitive to all types of digital modulation, producing easily detected polarized pulses synchronous with data transitions. Trans-filters do not require coherent conversion oscillators and complex synchronizing circuits.

Trans-filter/Detectors are extremely sensitive circuits that recover exponentially modulated signals buried in noise. They can be used wherever Matched Filter/Coherent Detectors are used and operate at negative input signal-to-noise ratios to recover RADAR, SONAR, communications, or data signals, as well as reduce phase noise of precision oscillators. Input signal and noise is split into two paths where complementary derivatives are extracted. Outputs of the two paths are equal in amplitude and 180 degrees relative to each other at the band center frequency. The outputs are summed, causing stationary in-band noise to be reduced by cancellation while exponentially modulated signals are undiminished. Trans-filters are Linear Time Invariant circuits, have no noise x noise threshold and can be cascaded, increasing output signal-to-noise ratio prior to detection. Trans-filters are most sensitive to all types of digital modulation, producing easily detected polarized pulses synchronous with data transitions. Trans-filters do not require coherent conversion oscillators and complex synchronizing circuits.

A signal processing system is described. The signal processing system includes at least one signal processing path and a control module. The at least one signal processing path includes at least one signal input and at least two filter units. The at least two filter units include at least one hardware filter unit. The at least one signal input is connectable to at least one external electronic component. The control module is connected to the signal input and to the at least two hardware filter units. The control module is configured to determine a frequency response deviation being associated with the at least one external electronic component. The control module further is configured to reconfigure the at least one hardware filter unit such that the frequency response deviation is compensated at least partially. Further, a signal processing method for adapting filter coefficients of a signal processing system is described.