119 results about "Adaptive filter coefficients" patented technology

An impluse response h_mn(q) of each transmission path is estimated from N received signals r_m (m=1, . . . , M) and a known signal (for a number of users equal to N, n=1, . . . , N). M×N matrix H (q) having h_mn(q) as an element and a Q×Q matrix H having H(q) as an element are determined (where Q represents a number of multipaths of each transmitted wave and q=0, . . . , Q−1). A soft decision value b′_n(k) is determined from decoded λ₂ [b_n(k)], and this is used to generate an interference component matrix B′(k) to generate an interference replica H·B′(k). The interference replica H·B′(k) is subtracted from a received matrix y(k) to determine y′(k). y(k) and H are used to determine an adaptive filter coefficient w_n(k) to be applied to an n-th user in order to eliminate residual interference components in y′(k) according to the minimum mean square error criteria. y(k) is passed through w_n(k) to provide a log-likelihood ratio as a received signal from the user n from which interferences are eliminated.

An adaptive spatio-temporal filter for use in video quality of service instruments based on human vision system models has a pair of parallel, lowpass, spatio-temporal filters receiving a common video input signal. The outputs from the pair of lowpass spatio-temporal filters are differenced to produce the output of the adaptive spatio-temporal filter, with the bandwidths of the pair being such as to produce an overall bandpass response. A filter adaptation controller generates adaptive filter coefficients for each pixel processed based on a perceptual parameter, such as the local average luminance, contrast, etc., of either the input video signal or the output of one of the pair of lowpass spatio-temporal filters. Each of the pair of lowpass spatio-temporal filters has a temporal IIR filter in cascade with a 2-D spatial IIR filter, and each individual filter is composed of a common building block,5 i.e., a first order, unity DC gain, tunable lowpass filter having a topology suitable for IC implementation. At least two of the building blocks make up each filter with the overall adaptive spatio-temporal filter response having a linear portion and a non-linear portion, the linear portion being dominant at low luminance levels and the non-linear portion being consistent with enhanced perceived brightness as the luminance level increases.

This invention discloses an self-adaptive mike array system and a method for processing its phone signals, which adds an self-adaptive filter coefficient refresh control module in the mike system to control if the filter should refresh the coefficient or the step-size value when refreshing the coefficient, so that the refreshment of the coefficient is only against the noise so as to avoid the side effect of reducing effective phone signal amplitude while conducting filter coefficient refresh using existing technology when the noise component from the back signals is comparable with or smaller so than the phone component from frontage. This invented system applies a high-order adaptive filter to reduce the noise amplitude to the utmost degree and ensures the best noise-elimination effect no matter what positions the system zero-point directs when multiple noise sources exist.

The invention discloses a method for eliminating noises of microphone array. The microphone array comprises a unidirectional microphone and an omnidirectional microphone. The procedures are as follows: calibrating the gain of the target direction signal collected by two microphones, so as to lead the target direction signal collected by the two microphones to have the same gain; decreasing the signal collected by the two microphones, which has already calibrated; judging if target speech signal exists in the signal collected by the unidirectional microphone according to average energy of the signal difference between the signal collected by the unidirectional microphone and the signal collected by the two microphones which has already decreased; if target speech signal exists, noise in the signal collected by the unidirectional microphone is eliminated by adopting adaptive filter coefficients from fixed frequency domain; if target speech signal does not exist, noise in the signal collected by the unidirectional microphone is eliminated by adopting adaptive filter coefficients from a new frequency domain.

The invention provides a microphone array device and a beam forming method thereof. The microphone array device can effectively eliminate noise and realize the directive property of a conical beam. Compared with prior art, the invention restrains the noise by controlling the updating of the coefficient of a self-adapting filter with the coefficient a, and the efficiency is pretty high.

The invention relates to a template matching based method for encoding self-adaption intra-frame prediction filtering, belonging to the technical field of image processing. The method comprises the following steps of: acquiring a template of a current encoding block, respectively searching n optimal template matchings in a matching way according to the acquired template; obtaining the prediction of the current encoding block according to the template matchings; obtaining three pixel values of pixel predictions generated in the prediction of the current encoding block according to a standard supply method; computing the mean values of the encoding block predictions obtained by all the template matchings, and using the mean values as targets of the predictions; computing the mean values of n groups of the three pixel values of pixels, using the mean values as the support for generating target values, and computing the optimal self-adaption filter coefficient of a mode according to a least square principle; and generating a predicting value of the intra-frame prediction of the current encoding block. The invention can effectively reduce the residual error after the intra-frame prediction and enhance the objective image quality of decoding by more than 0.2 dB under the condition of same code rate.

A digitized input signal is sampled, to obtain a reference input signal. The reference input signal is spuriously whitened, to obtain a pseudo whitened signal. An echo replica signal is generated on the basis of the pseudo whitened signal and an adaptive filter coefficient. A cancellation error signal is obtained on the basis of an echo signal and the echo replica signal. The adaptive filter coefficient is updated on the basis of the cancellation error signal, the adaptive filter coefficient, and the pseudo whitened signal.

An adaptive filter implemented in a communication system transmitter or receiver has a real time clock associated therewith, and one or more coefficients of the adaptive filter are determined based at least in part on an output of the real time clock. For example, the adaptive filter may comprise a coefficient update engine and a memory for storing a plurality of sets of adaptive filter coefficients in association with respective time indicators derived from the output of the real time clock, with the coefficient update engine being configured to determine a particular one of the sets of filter coefficients for use by the adaptive filter based at least in part on one or more of the time indicators. The time indicators may comprise respective time stamps generated based on the output of the real time clock at respective times at which the corresponding sets of coefficients are determined.

The invention discloses a sound processing method, a sound processing device, electronic equipment and a computer readable storage medium. The method comprises the following steps: in accordance witha plurality of acoustical signals acquired by a microphone array and an image signal which is acquired by a camera, determining a sound pre-processing direction; selecting a pre-processing filter coefficient based on the sound pre-processing direction; implementing pre-processing filtering on the plurality of acoustical signals with the adoption of the pre-processing filter coefficient, so that aninitial signal source signal and an initial noise source signal are obtained; determining an adaptive filter coefficient; and with the adoption of the adaptive filter coefficient, conducting adaptivefiltering on the initial signal source signal and the initial noise source signal, so that an enhanced signal source signal is obtained. With the application of the sound processing method and the sound processing device, the signal source signal can be enhanced, so that the tone quality of sound is improved.

A variable update step size is determined in proportion to a magnitude ratio or magnitude difference between a first residual signal and a second residual signal. The first residual signal is obtained by using adaptive filter coefficient sequence, where the adaptive filter coefficient sequence has been obtained in previous operations of the adaptive equalizer. The second residual signal is obtained by using a prior update adaptive filter coefficient sequence, where the prior update adaptive filter coefficient sequence is obtained by performing a coefficient update with an arbitrary prior update step size on the adaptive filter coefficient sequence having been obtained in previous operations of the adaptive equalizer.

An apparatus for canceling an acoustic echo signal caused by a far-end talker signal is provided. The apparatus for canceling an acoustic echo signal includes: a variance estimating unit configured to estimate a variance of a first audio signal of a near-end talker signal and a first noise signal of the near-end talker signal; a step size determining unit configured to determine a step size by using the variance of the first audio signal and the variance of the first noise signal; an adaptive filter coefficient updating unit configured to update an adaptive filter coefficient of an adaptive filter by using the step size; and an acoustic echo canceling unit configured to estimate an acoustic echo signal by using the adaptive filter coefficient, and cancel the acoustic echo signal from a microphone input signal by using the estimated acoustic echo signal.

The invention discloses a coefficient updating method for an adaptive filter in adaptive echo cancellation, and the method comprises the steps: estimating an echo signal y(k) according to a reference signal x(k) of a sound generation unit through the adaptive filter; calculating a residual signal e(k) according to the echo signal y(k) and a signal d(k) picked by a microphone; calculating a power spectrum xPow(f) of a frequency domain reference signal X(f); calculating a power spectrum ePow(f) of a frequency domain residual signal E(f); calculating the correlation power spectrum xePow(f) of a conjugate residual signal E*(f) of the frequency domain residual signal E(f) and the frequency domain reference signal X(f); calculating the correlation coefficient cohxe(f) of X(f) and E*(f); enabling a median cohxemed of cohxe(f) to serve as the adaptive step factor of the adaptive filter at each frequency point, and updating the coefficient of the adaptive filter. According to the technical scheme of the invention, the method enables the correlation coefficient between the residual signals and the reference signal to serve as the adaptive step factor so as to update the coefficient of the adaptive filter, and can achieve the effective echo cancellation of each voice state.

The invention disclose a polarization shift keying modulation-demodulation method, a system and a method for updating self-adaptive filtering coefficients of a reception end. The polarization shift keying modulation-demodulation method comprises the following steps: at a transmission end, modulating any QPSK (quadrature phase shift keying) signal (the formula is as shown in the specification) in an X polarization direction and modulating a Y polarization signal (the formula is as shown in the specification) same as or opposite to an X polarization signal (the formula is as shown in the specification) in a Y polarization direction according to an input signal S(n) by using a QPSK manner; at the reception end, demodulating a received signal into a corresponding electric signal by using a polarization multiplexing QPSK manner and carrying out digital signal processing to reduce the input signal S(n) after ADC sampling and quantification, wherein an improved CMA method is used during the digital signal processing to update the self-adaptive filtering coefficients. The polarization shift keying modulation-demodulation method has the advantages that polarization shift keying immunity phase noise is provided, the optical fiber nonlinear influence is reduced, important function is brought to ultralong-span transmission, the signal transmission/reception completely inherits the mature hardware architecture of polarization multiplexing QPSK, the realization is simple and high practicability is achieved.

The invention discloses a radar moving target detection method based on a cognitive framework. The radar moving target detection method mainly solves the problem that in the strong clutter background, an existing method is poor in detection performance for a low-speed target. The achieving process comprises the following steps of separating ground clutter components from backward waves; obtaining a clutter image of a current frame according to the ground clutter components; sensing the environment according to the clutter image and separating the current frame into a noise region and a clutter region; calculating coefficients of associations according to the ground clutter components, utilizing the coefficients of associations to correct antenna pointing errors, and storing backward wave data after corrected to an environment dynamic data bank (EDDB); and utilizing the backward wave data of the EDDB to estimate an interference covariance matrix according to division of the current frame; obtaining a self-adaption filter coefficient according to the estimated interference covariance matrix, and conducting self-adaption filtering processing on input data; and conducting constant false alarm detection on output of a filter. The radar moving target detection method has the advantage of greatly improving detection performance under the condition that an existing device is not greatly changed, and can be used for conducting detection on the low-speed target in the strong clutter background.

A system for filtering digital television signals is provided. The system comprises a generator for providing a first data sequence to a private data packetizer, and a transmitter for transmitting the packetized first data sequence in a data channel of a digital television signal. The system further includes a receiver for receiving the digital television signal and recovering the first data sequence. The receiver includes a channel estimator for providing an estimate of channel characteristics, such as estimated channel impulse estimate and estimated noise variance. The receiver further includes an adaptive equalizer filter having an input for receiving the digital television signal and an input for receiving adaptive filter coefficients. The receiver further includes a coefficient processor for calculating adaptive filter coefficients based on the channel estimate, and providing the adaptive filter coefficients to the adaptive equalizer filter. The digital television signal is thus filtered to remove undesired channel effects.