72 results about "Adaptive beamformer" patented technology

An adaptive signal processing system eliminates noise from input signals while retaining desired signal content, such as speech. The resulting low noise output signal delivers improved clarity and intelligibility. The low noise output signal also improves the performance of subsequent signal processing systems, including speech recognition systems. An adaptive beamformer in the signal processing system consistently updates beamforming signal weights in response to changing microphone signal conditions. The adaptive weights emphasize the contribution of high energy microphone signals to the beamformed output signal. In addition, adaptive noise cancellation logic removes residual noise from the beamformed output signal based on a noise estimate derived from the microphone input signals.

This invention relates to a system and method for suppressing external interference in radar data provided by a plurality of sensors from a main sensor array, the data being pre-processed. The noise suppression system includes a first processing module and a second processing module. The first processing module receives the radar data and produces matched radar data while the second processing module receives the radar data and produces mis-matched radar data. The system further includes a beamformer that is in communication with the first processing module and an adaptive beamformer that is in communication with the second processing module and the beamformer. The beamformer receives the matched radar data and produces beamformed matched radar data.

An audio device, comprising a microphone array, a constrained switched adaptive beamformer with input coupled to said microphone array, said beamformer including (i) a first stage speech adaptive beamformer with first adaptive filters having a first adaptive step size, and (ii) a second stage noise adaptive beamformer with second adaptive filters having a second adaptive step size, and a single channel speech enhancer with input coupled to an output of said constrained switched adaptive beamformer.

Receiving apparatus, for receiving a transmission signal in a cellular mobile communications system, comprises a main beamformer (6_M, 14_M) which processes received signals, representing the said transmission signal, in accordance with a main beam pattern. This main beam pattern is determined by beam control information applied to the main beamformer. The main beam pattern is adjusted as necessary during use of the receiving apparatus to facilitate reception of the said transmission signal.

The apparatus also has three assistant beamformers (6_A1, 14_A2; 6_A2, 14_A2; 6_A3, 14_A3) that, in an initial operating phase of the apparatus, process such received signals in accordance with three different assistant beam patterns. Each such pattern is determined by beam control information (W₁–W₃₃) corresponding individually thereto. The three assistant beamformers produce output signals (O_A1, O_A2, O_A3) corresponding respectively to the different assistant beam patterns.

A beam control information setting unit (16, 20) employs the output signals and the beam control information (W₁₁ to W₃₃) corresponding respectively to the said assistant beam patterns to make an initial estimate of the beam control information for the main beamformer.

Such receiving apparatus can permit fast setup of the initial beam control information for the main beamformer.

A noise suppression apparatus selectively uses an adaptive beamformer and fixed beamformer for each frequency. A direction of a null of the fixed beamformer is determined from a direction of a null automatically formed by the adaptive beamformer. Filter coefficients of the adaptive beamformer based on an output power minimization rule are calculated by a minimum norm method using a norm of the filter coefficients as a constraint. The above selection is made based on, for example, a depth of a null automatically formed by the adaptive beamformer in the selection.

An adaptive beamforming device belongs to medicine ultrasound imaging field. The invention provides an adaptive beamforming technique based on correlativity analysis, directing towards the difficulty that the traditional beamforming device can not completely restrain sidelobe effect resulted from interchannel random phase error and not affect the focusing effect along with the dynamic alignment of detection position. The technique divides the echo signal obtained from each channel into several segments, direct towards each segment, firstly the reference signal is obtained by a classical beam forming method, and then the signal of each channel is mutually correlated with the reference signal, furthermore weight of channel in each segment is dynamically adjusted using correlation coefficient, finally signals of each channel are synthesized to obtain final beam output using adjusted weight. The adaptive beamforming device adaptively detects the variance of enviroment using dynamic apodization and restrains the sidelobe effect resulted from interchannel random phase error. The adaptive beamforming device has wide application prospect in ultrasound imaging field.

The subject disclosure is directed towards a noise adaptive beamformer that dynamically selects between microphone array channels, based upon noise energy floor levels that are measured when no actual signal (e.g., no speech) is present. When speech (or a similar desired signal) is detected, the beamformer selects which microphone signal to use in signal processing, e.g., corresponding to the lowest noise channel. Multiple channels may be selected, with their signals combined. The beamformer transitions back to the noise measurement phase when the actual signal is no longer detected, so that the beamformer dynamically adapts as noise levels change, including on a per-microphone basis, to account for microphone hardware differences, changing noise sources, and individual microphone deterioration.

The invention provides an adaptive sensor array apparatus (100) incorporating a multielement antenna (12) for receiving radiation from a scene ("S') and generating signals (e) in response thereto and a processing unit (114) for processing the signals (e) to provide an output signal (y). The unit (114) comprises an adaptive weight computer (136) arranged to generate weighting vectors (w) which are used in the unit (114) to attenuate contributions to the output signal (y) arising from sources of jamming radiation in the scene and transmit contributions to the output signal (y) arising from wanted targets therein. The apparatus (100) incorporates a non-adaptive beamformer unit (132) for preconditioning the signals (e) before they are passed to the computer (136). Preconditioning the signals (e) enhances performance of the apparatus (100) relative to conventional adaptive sensor array apparatus (10).

The hybrid adaptive beamformer of the present invention includes a plurality of microphones for receiving sound energy from an external environment and for producing a plurality of microphone outputs from the sound energy. A processor produces a plurality of first order beams based on the microphone outputs. The processor determines an amount of reverberation in the external environment and adaptively produces a second order output beam taking into consideration the determined amount of reverberation. The processor may determine the amount of reverberation based on a comparison of the first order beams. The processor may produce the second order output beam by adaptively combining the plurality of first order beams, as further described below, or by adaptively combining the microphone outputs. The adaptation varies taking into consideration the determined amount of reverberation. Alternatively, the adaptation may vary by measuring the signal-to-noise ratio of the first order beams and adaptively combining the first order beams based on the determined signal-to-noise ratios.

The adaptive beamformer unit (191) comprises: a filtered sum beamformer (107) arranged to process input audio signals (u l, u2) from an array of respective microphones (101, 103), and arranged to yield as an output a first audio signal (z) predominantly corresponding to sound from a desired audio source (160) by filtering with a first adaptive filter (fl (-t)) a first one of the input audio signals (u l) and with a second adaptive filter (f2(-t)) a second one of the input audio signals (u2), the coefficients of the first filter (fl(-t)) and the second filter (f2(-t)) being adaptable with a first step size (al) and a second step size ((x2) respectively; noise measure derivation means (111) arranged to derive from the input audio signals (ul, u2) a first noise measure (xl) and a second noise measure (x2); and an updating unit (192) arranged to determine the first and second step size (al, (x2) with an equation comprising in a denominator the first noise measure (xl) for the first step size (al), respectively the second noise measure (x2) for the second step size (a2). This makes the beamformer relatively robust against the influence of correlated audio interference. The beamformer may also be incorporated in a sidelobe canceller topology yielding a more noise cleaned desired sound estimate, which can be used in a related, more advanced adaptive filter (fl (-t), f2(-t)) updating. Such a beamformer is typically useful for application in handsfree speech communication systems.

The invention provides a robust beam forming method based on a steering vector and space power estimation, so as to improve the robustness of random errors on an incoming wave direction and to realizeaccurate signal steering vector estimation under the condition of interference in a main lobe of a beam. The invention provides a signal plus interference covariance matrix reconstruction algorithm,which introduces a correlation coefficient to serve as a signal subspace selection criterion, reconstructs a sampling covariance matrix signal plus interference subspace, and performs alternate projection in combination with a covariance matrix containing a signal subspace and obtained by signal angle interval discrete sampling, so as to perform the accurate signal steering vector estimation and to design a robust adaptive beam former under the condition of interference in the main lobe of the beam.

The invention discloses a sub-array-class adaptive digital beam forming device with low side-lobe characteristics, relating to a sub-array-class adaptive digital beam forming device. The purpose of the invention is to solve the problem that the side-lobe performance of the current sub-array-class adaptive digital beam forming device is poor. The forming device is composed of a sub-array-class optimal adaptive beam forming device, a sub-array-class adaptive beam forming device with a direction figure control performance, an adding device and a beam forming device, wherein the sub-array-class optimal adaptive beam forming device is used for generating a sub-array-class optimal adaptive weight vector wopt; the sub-array-class adaptive beam forming device with the direction figure control performance is used for generating a sub-array-class adaptive weight vector wpc with the direction figure control performance; the adding device carries out weighted summation for the adaptive weight vectors wopt and wpc of the two forming devices; and the beam forming device is used for carrying out adaptive interference suppression. The sub-array-class adaptive digital beam forming device is suitable for the field of sub-array-class adaptive digital beam forming.

The invention belongs to the technical field of antenna beam forming, and discloses a reconstruction robust beam forming method based on IAA interference plus noise covariance matrix. The method comprises the following steps of: firstly, estimating the direction of a signal more accurately in an iterative mode by using IAA method, then selecting the characteristic vector with the maximum correlation with the steering vector calculated by using the estimated direction of the desired signal from sampling covariance matrix characteristic vectors and rejecting the characteristic vector from the sampling covariance matrix, finally, selecting a proper diagonal loading factor to carry out diagonal loading on the matrix in order to prevent the covariance matrix after rejecting the signal from generating singular, in the case that the sample contains the desired signal, the signal to noise ratio of a self-adaptive beam former is improved and the robust performance of the self-adaptive beam former is improved in the case of a mismatch steering vector.

The invention discloses a co-prime array robust adaptive beamforming method based on interference covariance matrix reconstruction. The method comprises the following steps of reconstructing a sampling covariance matrix on the virtual uniform matrix, removing expected signal components in the sampling covariance matrix by using projection, estimating accurate power and direction of each interference according to semi-positive qualitative constraints, reconstructing an interference covariance matrix, estimating noise power in a noise angle area, and obtaining an interference and noise covariance matrix; and taking the main characteristic value of the reconstructed expected signal covariance matrix as an expected signal steering vector, thereby obtaining a weight vector of the robust adaptive beamformer based on the co-prime matrix, and forming the output of the robust adaptive beamformer.

A system for speech enhancement in a room having a directional lapel microphone arrangement for capturing an audio signal from a speaker's voice; audio signal processor for generating a processed audio signal from the captured audio signal, and an adaptive beamformer unit for imparting directivity to the microphone arrangement provides maximum sensitivity towards the speaker's mouth and minimum sensitivity towards noise sources identified by the beamformer unit. A unit shifts the frequency of components of the audio signal above a frequency threshold value only. A feedback cancelling unit has an adaptive filter and a selection unit which automatically switches between a first mode causing the audio signal to by-pass the adaptive filter and a second mode in which the audio signal is filtered by the adaptive filter. A loudspeaker arrangement generates sound according to the processed audio signal and has plural loudspeakers arranged to form a directional loudspeaker array.

The present invention discloses a radar covariance matrix reconstruction beam forming method based on iteration mutual coupling calibration. The method thinking comprises: an uniform circular array is determined, the uniform circular array comprises M array elements, there are Q signal sources in the setting range of the uniform circular array, the Q signal sources emit Q incident signals to the uniform circular array, and the Q incident signals comprise one desired signal and Q-1 interference signals; the sampling covariance matrix R of the uniform circular array is obtained, the characteristic decomposition is performed, and M characteristic values are obtained; a signal MUSIC spectrum having an incident direction of ([Theta], [Phi]) is calculated, the mutual coupling matrix initial value of the uniform circular array is set to obtain Q azimuth initial values of the Q incident signals, and a final mutual coupling matrix of the uniform circular array (img file='DDA0001335531130000011.TIF' wi='43' he='57'/) and the final azimuths of the Q incident signals (img file='DDA0001335531130000012.TIF' wi='67' he='58'/) are obtained; and the interference-plus-noise covariance matrix of the uniform circular array after the reconstruction (img file='DDA0001335531130000013.TIF' wi='116' he='73'/) is calculated to obtain the weight vector of the adaptive beam former of the uniform circular array so as to complete the reconstruction of the interference-plus-noise covariance matrix of the uniform circular array based on the iteration mutual coupling calibration.

The invention belongs to the technical field of array signal processing and discloses a self-adaptive beamforming method based on interference and noise covariance matrix reconstruction. The self-adaptive beamforming method comprises the steps that a uniform linear array is established and is used for detecting far-field narrow-band signals including one desired signal to be detected and Q-1 interference signals to obtain receiving signals; the receiving signals are sampled to obtain K receiving signal samples, a sample matrix is formed, and an interference and noise sampling covariance matrix of the sample matrix is calculated; a weighting matrix is constructed, and the interference and noise sampling covariance matrix is weighted to obtain a weighted covariance matrix; a sampling matrix is constructed, and further a first interference and noise covariance matrix is constructed; the first interference and noise covariance matrix is weighted again according to the weighting matrix to obtain a second interference and noise covariance matrix; the self-adaptive weight vector of a self-adaptive beamformer is calculated according to the second interference and noise covariance matrix; the calculating amount can be decreased on the basis that a detection effect of desired signals is ensured.

An object information acquiring apparatus is provided that includes: a plurality of receiving elements which receive an acoustic wave that propagates from an object, and which convert the acoustic wave to a reception signal; an adaptive beamformer that performs adaptive beamforming of adjusting reception directionality in accordance with the reception signal in use of the reception signal; a delay-and-sum beamformer that performs delay-and-sum beamforming having preset directionality in use of the reception signal; an amplitude modulator that uses one output signal of one of the adaptive beamformer and the delay-and-sum beamformer, to perform amplitude modulation on an output signal of the other one of the beamformers; and a generator that, on the basis of a signal outputted by the amplitude modulator, generates image data on the interior of the object.