Showing papers on "Adaptive beamformer published in 1994"

A projection approach for robust adaptive beamforming

Abstract: It is well known that calibration errors can seriously degrade performance in adaptive arrays, particularly when the input signal-to-noise ratio is large. The effect is caused by the perturbation of the presumed steering vector from its optimal value. Although it is not as widely known, similar degradation occurs in sampled matrix inversion processing when the covariance matrix is estimated while the desired signal is present in the snapshot data. Under these conditions, performance loss is due to errors in the estimated covariance matrix and occurs even when the steering vector is known exactly. In the paper, a new method based of modification of the steering vector is proposed to overcome both the problems of perturbation and of sample covariance errors. The method involves projection of the presumed steering vector onto the observed signal-plus-interference subspace. An analysis is also presented illustrating that the sample covariance errors can be viewed as a particular type of perturbation error and a simple approximation is derived for the expected beamformer performance as a function of the number of data snapshots. Both analytical and experimental results are presented that illustrate the advantages of the proposed method. >

Noise reduction for hearing aids: combining directional microphones with an adaptive beamformer.

Abstract: Many hearing aid users complain about a reduced intelligibility of speech in noisy environments. Directional systems are a successful approach for noise reductions in hearing aids. These systems transmit signals from acoustic sources lying in front of the hearing aid user while suppressing signals from other directions, which are assumed to be noise. Several methods are known to obtain directivity. One is to use directional microphones, another is digital postprocessing of several microphone signals. In this letter, the combination of directional microphones with the adaptive beamformer, a directional signal processing approach, is discussed. Intelligibility tests with both normal‐hearing and hearing‐impaired subjects are presented. It is shown that the combination of directional microphones with digital postprocessing is able to improve the intelligibility of speech in a noisy environment significantly, when compared to any one of these two approaches by itself.

Adaptive beamforming: Spatial filter designed blocking matrix

Abstract: Controlling the resolution in adaptive beamformers is often crucial. A simple method that works for both narrow-band and broad-band arrays is presented. This method is based on the normalized leaky LMS algorithm in conjunction with a generalized sidelobe canceller (GSC) structure, where the GSC is designed using a spatial filtering approach. In essence, the suppression of the spatial filters and the implicit noise of the leaky LMS algorithm together determine the adaptive beamformer. Analytical expressions are given for the Wiener filters and the output spectrum versus frequency and point source location. These expressions are employed in the design specification of the spatial filters and to obtain conditions for a controlled quiescent beamformer response. Simulation results are presented to illustrate the behavior of the array. >

Multi-beam digital beamforming method and apparatus

Abstract: In accordance with the principles of the present invention, advantage is taken by the inventors of the fact that the speed of operation of the digital hardware in a digital beamformer can be reduced by providing, for example, multiple phases of the data signals and then processing the multi-phase data in N parallel summing paths An interpolation-decimation filter receives the multi-phase data from the N parallel summing paths and provides at its output a signal having a reduced data rate (1/N) In accordance with this technique, the speed of operation of the individual digital circuits for forming the required beamforming delays is not increased as compared to conventional post-beamforming interpolation schemes, so that hereby the effective data rate is increased by a factor N and results in a decrease of the delay quantization error by a factor N In accordance with the principles of the invention, the interpolation-decimation filter is incorporated into the beamformer at a most advantageous place That is, it is incorporated into the beamformer processing after partial beamforming of a group of receive channels and before formation of the final beam This approach allows the final beamforming to be simple and performed at a relatively low data rate and allows the higher rate signal processing to be confined to circuitry which may advantageously be on a single type of integrated circuit which is repetitively used in the beamformer Further increase in the effective speed of operation is produced by providing timing circuitry that allows parallel processing of signals from a plurality of scanning beam lines

Array gain/phase calibration techniques for adaptive beamforming and direction finding

Abstract: A calibration technique is proposed for handling adaptive beamforming and bearing estimation problems involving unknown perturbed sensor gain and phase. This calibration technique is applied on the MUSIC estimator, assuming that two or more signal sources (in which the direction of arrival of one or two of the signal sources are known temporarily) exist in the signal field, so as to estimate the true sensor gain and phase. The basic idea of the technique is to apply the first-order Taylor series expansion to approximate the true array steering vector from the nominal one. A set of linear equations is then formed, using the null characteristic of the MUSIC spectrum, from which the error steering vector (the difference between the actual steering vector and the nominal steering vector), which contains the gain/phase information of the array sensors, can be solved for. This technique exhibits relatively stable performance compared with existing techniques in the sense that it produces the required estimates consistently without the need for iterative computation and initialisation. This is illustrated with numerical results obtained from several Monte Carlo experiments.

Broadband beamforming for a microphone array

Abstract: An array of sensors can be used in conjunction with a beamformer, which processes the sensor signals, to achieve a directional response. The beamformer has to be designed such that a beam pattern with certain desired characteristics like specific main beam direction, defined main beam shape, and desired sidelobe level is formed. Conventional methods for the design of beamformers assume sensors with ideal features and do not take the disturbance of the sound field, caused by the mounting of the array, into account. Therefore the predicted theoretical polar response and the measured response often differ significantly. This paper presents a method for optimizing a beamformer for a one‐dimensional microphone array, taking into consideration nonideal features of the sensors and the mounting. Thus the actual polar response can be improved. By evaluating cross‐correlation functions of the sensor signals during a calibration procedure in an anechoic chamber and minimizing the mean squared error between the beamformer output and a prescribed response, optimum parameters for the beamformer are assigned.

Robust adaptive beamforming based on the eigenstructure method

Abstract: In a linearly constrained beamformer with imperfect arrays, the authors investigate the cause of cancelling the desired signal from a viewpoint based on the eigenstructure properties of the array covariance matrix. Based on this cause, they propose a robust beamforming algorithm. As an adaptive algorithm of the proposed beamformer, the present a data-domain signal subspace updating algorithm. >

Robust adaptive beamforming via LMS-like target tracking

Abstract: This paper presents an algorithm implementing robustness in beamforming, by directly reducing localization errors in the presence of pointing errors or a single moving target. Given an initial position, the desired source signal is first estimated using a classical beamforming unit. It is in a second step, processed by an "LMS-like" gradient stochastic estimation procedure of the steering vector, to adaptively track the correct source position. The newly identified source position is projected over the array manifold, then finally transmitted in a feedback loop to the beamforming unit, closing in this way the global algorithm iteration. The simulation results show that robustness is effectively realized without any compromising output SNR loss. Moreover, they prove an efficient tracking behavior in the presence of mobile sources. >

Broadband focusing for partially adaptive beamforming

Abstract: A broadband adaptive beamforming approach based on the concept of signal-subspace alignment is formulated and contrasted with the conventional approach. The proposed method involves a preprocessor that focuses the signal spaces at different frequencies to a common one and a narrowband beamformer following the preprocessor. The merits that result as a consequence are partial adaptivity due to single frequency weights and decorrelation of coherent signals thus combating signal cancellation. The latter effect is studied by deriving expressions for the desired-signal distortion in a minimum variance distortionless response (MVDR) beamformer and analyzing them. Implementation issues of the preprocessor are addressed. Simulation results confirm the utility of the focusing preprocessor. >

Sub-sea acoustic remote communications utilising an adaptive receiving beamformer for multipath suppression

Abstract: Acoustic communications systems have found widespread use in both the offshore industry and marine sciences. For many applications such as remote operated vehicle (ROV) operation it is desirable to transmit video information in the form of camera pictures or side scan sonar images from the ROV to a surface vessel. For shallow water operation where the range is much greater than the water depth the most detrimental problem is that of inter symbol interference due to multipath. The paper presents a novel solution to this problem by using an adaptive beamformer to spatially filter the direct path signal from the multipath. Results obtained during sea trials are presented and demonstrate the effectiveness of the system. >

Optimal weight extraction for adaptive beamforming using systolic arrays

Abstract: Systolic algorithms and architectures for parallel and fully pipelined instantaneous optimal weight extraction for multiple sidelobe canceller (MSC) and minimum variance distortionless response (MVDR) beamformer are presented The proposed systolic parallelogram array processors are parallel and fully pipelined, and they can extract the optimal weights instantaneously without the need for forward or backward substitution We also show that the square-root-free Givens method can be easily incorporated to improve the throughput rate and speed up the system As a result these MSC and MVDR systolic array weight extraction system are suitable for real-time very large scale integration (VLSI) implementation in practical radar/sonar system >

Novel multirate adaptive beamforming technique

Abstract: A novel multirate/sub-band adaptive beamformer based on QMF banks is presented. Computer simulations of the new technique in a BPSK, DS-SSMA mobile radio system demonstrate its superiority, in terms of convergence speed and cancellation performance, over full-band beamforming.

Multipathreception using matrix calculation and adaptive beamforming.

Abstract: Signals from multiple signal paths are received using a multi-element antenna (20) and a beam-forming network (36). Signals from each of the antenna elements are sampled (34) to form a sample vector. Several sample vectors are used to form an auto-covariance matrix. A singular value decomposition of the auto-covariance matrix is used to form three matrices. The first matrix is used to determine the number of signal paths and the second matrix is used to form several polynomials. The polynomial roots that are on or near the unit circle are used to determine points on the unit circle that are associated with each signal path. Each point on the unit circle is used to calculate weights for a beam-forming network (36) that forms a receive beam for each signal path.

Partially adaptive beamformer design subject to worst-case performance constraints

Abstract: Proposes two methods for designing partially adaptive beamformers that satisfy a performance specification over a set of likely interference scenarios. Both methods choose the adaptation space in a sequential fashion; the dimension is increased one by one until the performance specification is attained. In the multilevel point design method, each dimension of the adaptation space is chosen to give optimum performance at a single interference scenario. The constrained minimization design method chooses each dimension of the adaptation space to exactly satisfy the performance specification at a single interference scenario while approximately minimizing the average interference output power over neighboring scenarios. Simulations indicate that both methods result in better performance than existing methods while using fewer degrees of freedom. >

Phase aberration correction with ultrasonic time reversal mirrors

Abstract: Adaptive time delay focusing techniques allow an efficient correction of the effects due to an inhomogeneous layer close to the transducer array. We have developed in our laboratory a time reversal process able to compensate for all distortion effects in the transmit mode. This process has been extended by developing in the transmit-receive mode a spatiotemporally matched filter approach that focuses optimally on a reflective target whatever the distance between the layer and the transducer array. In this technique, a set of focusing signals perfectly matched to the reflector are built and stored. This set of signals is used to focus in both transmit and receive modes, yielding the maximal signal for the reflector location with a very low side lobe level. We present in this paper an extension of this technique to focus, not only on the reflector, but also around the reflector in order to image the surrounding zone. From the knowledge of the signals needed to focus on the initial reflector, we calculate the new signals matched to the new point of interest. The algorithm uses the concept of time reversal propagation

A constant‐directivity beamforming microphone array

Abstract: The quality of audio teleconferencing in large rooms and noisy environments can be increased with the use of steerable directional microphone arrays. A minimum bandwidth of 4 oct is required to faithfully transmit the speech signal. In a typical teleconferencing arrangement, only discrete angular directions are of interest and therefore the microphone steering directions are quantized. A standard delay‐sum beamformer can result in noticeable frequency response changes as the talker moves between these steering locations. In an effort to mitigate this problem, a broadband constant‐directivity beamformer has been designed and constructed. A few of the algorithms developed in this work will be discussed and compared to existing techniques. Basically, the solution revolves around the design of FIR filters that are inserted in the delay‐sum beamformer after each element. A constant‐beamwidth 4 oct steerable linear array microphone using directional elements will be described. A real‐time implementation utilizing multiple AT&T DSP3210 digital signal processors is also described.

On the mean-square error performance of adaptive minimum variance beamformers based on the sample covariance matrix

Abstract: The authors examine the mean-square error (MSE) performance of two common implementations of adaptive linearly constrained minimum variance (LCMV) beamformers that employ the sample covariance matrix. The Type I beamformer is representative of block processing methods where the same input data is used both to compute the adaptive weights and to form the beamformer output. The Type II beamformer, as in many recursive schemes, applies adaptive weights computed from previous data to the current input. Due to correlation between the adaptive weights and the input data, the Type I LCMV beamformer exhibits signal cancellation, which is shown here to cause signal estimate bias. To explicitly account for signal cancellation, the mean-square error (MSE) and output signal-to-noise ratio (SNR) measures of the bias-corrected Type I beamformer are analyzed, thus extending previous results. Further, new analytical results for these performance measures are given for the Type II LCMV beamformer. Comparison of bias-corrected Type I and Type II implementations indicate that both methods yield exactly the same MSE and output SNR performance. >

Fast algorithm for adaptive beamforming of cyclic signals

Abstract: There has been an explosive increase in the demand of radio channels in cellular communication systems. Due to the finite limit of the frequency spectrum the number of available channels is bounded. This constraint has become a bottleneck of next-generation wireless communication systems. Some array beamforming techniques have been proposed to solve this problem by the space-division multiple access (SDMA). In the paper, a fast adaptive beamforming algorithm for extracting cyclic signals is proposed. Similar to the SCORE algorithm, the new beamformer performs a blind signal extraction. Compared with the SCORE method it has simpler computation and faster convergence rate and higher SINR in many situations of interest.

Investigation of adaptive beamformer performance and experimental verification of applications in high data rate digital underwater communications

Abstract: To use a remote operated vehicle (ROV) for high speed underwater data transmission (e.g. video) requires the establishment of a reliable acoustical communication link. The most difficult effect to overcome in a time varying underwater channel is multipath. An adaptive beamformer uses an adaptive algorithm to maximise the signal strength received from a desired direction while simultaneously inserting nulls in directions of interferers. Both the Frost (1972) and LMS algorithms are being investigated at Newcastle University, and an objective comparison is made of their performance using synthetic and real data. A flexible and powerful state of the art multichannel data acquisition system, based around the TMS320C40 DSP, has been designed and commissioned. This system is being used to capture and store large volumes of data generated during sea trials. The data acquired is analysed off-line to allow the investigation of various phenomena affecting underwater communications (e.g. multipath), and to provide a realistic method of comparing the performance of a variety of different modulation techniques. >

Direction-finding performance of the multistage CM array

Abstract: The constant modulus (CM) array is a blind adaptive beamformer that steers nulls in the directions of cochannel interferers without requiring a training (pilot) signal. A cascade implementation of the system, known as the multistage CM array, is designed to recover ("copy") several narrowband cochannel signals and provide estimates of their angles of arrival. Each stage is composed of a CM array that "captures" one of the sources and an adaptive signal canceler that removes the source from the array input. This signal cancelation influences the capture and direction finding performance of the remaining stages. The authors quantify this behavior using a stochastic model of convergence and present computer simulation results for some example source scenarios. >

Performance of high data rate acoustic underwater communication systems using adaptive beamforming and equalizing

Abstract: An analytic approach is presented to assess the performance of an acoustic communication system from an underwater vehicle to its mothership where the signals from the receive array are processed by an adaptive beamformer followed by an adaptive equalizer. A generalized signal-to-noise ratio, and the related error data rate of the digital transmission, are derived from the steady-state responses of the adaptive beamformer and the adaptive equalizer, as a function of the transmitter, receiver and water depths, the transmission range, the surface and the bottom reflection coefficients, the frequency range, the digital modulation type (M-PSK and M-QAM), the transmit power and the sizes of the projector and the receive array. Some results are given from a computer evaluation of the analytic expressions. >

SAR autofocusing viewed as adaptive beamforming on prominent scatterers

Abstract: The SAR phase correction information can be extracted from beamforming source echoes at different range gates. An ideal point source induces a field of constant amplitude across the array. Lower echo contrast can therefore be identified with higher quality sources. It is shown that by using contrast measurements on phase history domain SAR data, the contributions of phase synchronizing sources at different range gates can adequately be combined to produce an optimal SAR impulse response having the lowest possible ISLR. >

Time delay-phase shift combination beamformer

Abstract: A time delay-phase shift combination beamformer for applications where the required time delays for time delaying the received signal to achieve coherency exceed the reciprocal of the incoming signal's bandwidth, and having a time delay stage for coarse beamforming at a particular range and first beam direction and a phase shift stage for fine beam steering in a second direction at the particular range.

Development of a real-time adaptive equalizer for a high-rate underwater acoustic data communications link

Abstract: The paper describes the development of an adaptive equalizer based on the LMS algorithm through simulation results using real data acquired at Loch Duich, a sea loch on the west coast of Scotland, leading to the development of a hardware-implemented real-time adaptive equalizer. To reduce the demand on the equalizer to cancel long delay multipaths, it is combined with an existing adaptive beamformer. The adaptive beamformer, used on the receiver array, has an angular resolution of less than with the ability to adaptively position nulls at angles corresponding to multipath arrivals. This reduces the required tap-length of the equalizer speeding up processing and convergence properties. >

Linearly-constrained adaptive beamforming using cyclostationary signal properties

Abstract: This paper describes a method to incorporate linear constraints into an adaptive beamformer based on cyclostationary signal properties, previously proposed by the authors (see Proc. ICASSP'93, vol.IV, p. 284-287, Minneapolis, MN, April, 1993). The constraints allow an efficient exploitation of the spatial information available from the environment. This knowledge is used to prevent the beamformer from being captured by interferences showing the same cyclostationary properties as the desired signal and to control the sidelobe level of the radiation pattern. >

Adaptive beamforming of cyclic signal and fast implementation

Abstract: Recently, it has been proposed to apply array beamforming in digital mobile communication systems, such as mobile cellular communication systems and mobile-satellite communication systems, in order to increase channel capacity and suppress co-channel interference. However, the conventional beamforming methods are not suitable for such use, since these methods were mainly developed for signal detection and DOA estimation in radar and sonar. In this paper, utilizing the cyclostationarities of communication signals, we propose three blind cyclic adaptive algorithms (CAB's) and associated fast implementation schemes. We also investigate the characteristics of the new beamforming methods. >

Wideband Robust Adaptive Beamforming Via Target Tracking

Abstract: In this paper, we generalize a former work recently presented in the narrowband case of robust adaptive beamforming via target tracking to the wideband domain. The original algorithm is applied to each frequency component of the signal in an Analysis/Synthesis scheme. The source tracking and localization are simply performed in one frequency selected with the minimum location misadjustment. A more complex combination of location estimates can be computed in a specific set of frequencies, with a relatively better performance. Simulation results confirm in both cases the efficiency of the generalized algorithm regarding source localization and noise reduction.

Multichannel adaptive equalization for improved performance in LOS digital radio

Abstract: Multichannel adaptive equalization is a technique which combines (spatial) beamforming and (temporal) equalization into a single filter structure. It is capable of both compensating for channel-induced linear distortions and rejecting broadband interference. The paper compares the performance of multichannel adaptive equalization to both adaptive beamforming and single channel, fractionally-spaced decision feedback equalization for conditions common to line-of-sight digital radio. While the results are preliminary, the multichannel adaptive equalizer shows clear advantages over the performance of the other multipath fading compensation techniques. >

Performance analysis of the multistage CMA adaptive beamformer

Abstract: The multistage CMA adaptive beamformer is a blind adaptive antenna system capable of recovering several narrowband cochannel signals. Each stage of the system captures one source (without using a training signal), removes it before processing by subsequent stages, and provides an estimate of its direction of arrival. This system would be useful in frequency reuse applications, such as cellular radio or personal communication networks, where cochannel interference is an important consideration. We present the steady-state convergence properties of the first stage of the multistage CMA adaptive beamformer. Computer simulations are presented to verify the analytical results. >

An adaptive antenna array for multiple received signals in direct-sequence code-division multiple-access communication systems

Abstract: Direct-sequence spread-spectrum (DS-SS) has been under increasing scrutiny as a means to provide a multi-user communications channel via code-division multiple-access (CDMA). One of the issues which can impact the performance of a DS-SS communication system is the near-far effect. The near-far effect occurs when a powerful incident signal (desired or undesired) overwhelms desired DS-SS signals. One of several possible ways to lessen this effect is to use an adaptive antenna array at the DS-SS receiver. Of particular interest is the special case of a single receiver which must detect multiple desired signals. This might occur in commercial NAVSTAR GPS receivers and in mobile cellular base station receivers. This paper examines the steady-state performance of adaptive antenna arrays which are based on a minimum mean-squared error (MMSE) optimization criterion. The beamforming array is limited to a single feedback loop, and a single array output, which supports multiple received signals. It is shown that a single-loop MMSE array can level the output signal-to-noise ratios to nearly the same level and thus eliminate the near-far effect. >