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Showing papers on "Adaptive beamformer published in 2003"


Journal ArticleDOI
TL;DR: A new approach to robust adaptive beamforming in the presence of an arbitrary unknown signal steering vector mismatch is developed based on the optimization of worst-case performance.
Abstract: Adaptive beamforming methods are known to degrade if some of underlying assumptions on the environment, sources, or sensor array become violated. In particular, if the desired signal is present in training snapshots, the adaptive array performance may be quite sensitive even to slight mismatches between the presumed and actual signal steering vectors (spatial signatures). Such mismatches can occur as a result of environmental nonstationarities, look direction errors, imperfect array calibration, distorted antenna shape, as well as distortions caused by medium inhomogeneities, near-far mismatch, source spreading, and local scattering. The similar type of performance degradation can occur when the signal steering vector is known exactly but the training sample size is small. In this paper, we develop a new approach to robust adaptive beamforming in the presence of an arbitrary unknown signal steering vector mismatch. Our approach is based on the optimization of worst-case performance. It turns out that the natural formulation of this adaptive beamforming problem involves minimization of a quadratic function subject to infinitely many nonconvex quadratic constraints. We show that this (originally intractable) problem can be reformulated in a convex form as the so-called second-order cone (SOC) program and solved efficiently (in polynomial time) using the well-established interior point method. It is also shown that the proposed technique can be interpreted in terms of diagonal loading where the optimal value of the diagonal loading factor is computed based on the known level of uncertainty of the signal steering vector. Computer simulations with several frequently encountered types of signal steering vector mismatches show better performance of our robust beamformer as compared with existing adaptive beamforming algorithms.

1,347 citations


Journal ArticleDOI
TL;DR: It is shown that a natural extension of the Capon beamformer to the case of uncertain steering vectors also belongs to the class of diagonal loading approaches, but the amount of diagonalloading can be precisely calculated based on the uncertainty set of the steering vector.
Abstract: The Capon (1969) beamformer has better resolution and much better interference rejection capability than the standard (data-independent) beamformer, provided that the array steering vector corresponding to the signal of interest (SOI) is accurately known. However, whenever the knowledge of the SOI steering vector is imprecise (as is often the case in practice), the performance of the Capon beamformer may become worse than that of the standard beamformer. Diagonal loading (including its extended versions) has been a popular approach to improve the robustness of the Capon beamformer. We show that a natural extension of the Capon beamformer to the case of uncertain steering vectors also belongs to the class of diagonal loading approaches, but the amount of diagonal loading can be precisely calculated based on the uncertainty set of the steering vector. The proposed robust Capon beamformer can be efficiently computed at a comparable cost with that of the standard Capon beamformer. Its excellent performance for SOI power estimation is demonstrated via a number of numerical examples.

1,113 citations


Proceedings ArticleDOI
06 Apr 2003
TL;DR: This paper shows that a natural extension of the Capon beamformer to the case of uncertain steering vectors also belongs to the class of diagonal loading approaches but the amount of diagonalloading can be precisely calculated based on the uncertainty set of the steering vector.
Abstract: Whenever the knowledge of the array steering vector is imprecise (as is often the case in practice), the performance of the Capon beamformer may become worse than that of the standard beamformer. Diagonal loading (including its extended versions) has been a popular approach to improve the robustness of the Capon beamformer. In this paper we show that a natural extension of the Capon beamformer to the case of uncertain steering vectors also belongs to the class of diagonal loading approaches but the amount of diagonal loading can be precisely calculated based on the uncertainty set of the steering vector. The proposed robust Capon beamformer can be efficiently computed at a comparable cost with that of the standard Capon beamformer. Its excellent performance is demonstrated via a number of numerical examples.

621 citations


Journal ArticleDOI
TL;DR: The proposed robust adaptive beamformers are based on explicit modeling of uncertainties in the desired signal array response and data covariance matrix as well as worst-case performance optimization and offer a significantly improved robustness and faster convergence rates.
Abstract: The performance of adaptive beamforming methods is known to degrade severely in the presence of even small mismatches between the actual and presumed array responses to the desired signal. Such mismatches may frequently occur in practical situations because of violation of underlying assumptions on the environment, sources, or sensor array. This is especially true when the desired signal components are present in the beamformer "training" data snapshots because in this case, the adaptive array performance is very sensitive to array and model imperfections. The similar phenomenon of performance degradation can occur even when the array response to the desired signal is known exactly, but the training sample size is small. We propose a new powerful approach to robust adaptive beamforming in the presence of unknown arbitrary-type mismatches of the desired signal array response. Our approach is developed for the most general case of an arbitrary dimension of the desired signal subspace and is applicable to both the rank-one (point source) and higher rank (scattered source/fluctuating wavefront) desired signal models. The proposed robust adaptive beamformers are based on explicit modeling of uncertainties in the desired signal array response and data covariance matrix as well as worst-case performance optimization. Simple closed-form solutions to the considered robust adaptive beamforming problems are derived. Our new beamformers have a computational complexity comparable with that of the traditional adaptive beamforming algorithms, while, at the same time, offer a significantly improved robustness and faster convergence rates.

496 citations


Journal ArticleDOI
TL;DR: The proposed robust Capon beamformer can no longer be expressed in a closed form, but it can be efficiently computed and its excellent performance is demonstrated via a number of numerical examples.
Abstract: The Capon beamformer has better resolution and much better interference rejection capability than the standard (data-independent) beamformer, provided that the array steering vector corresponding to the signal of interest (SOI) is accurately known. However, whenever the knowledge of the SOI steering vector is imprecise (as is often the case in practice), the performance of the Capon beamformer may become worse than that of the standard beamformer. We present a natural extension of the Capon beamformer to the case of uncertain steering vectors. The proposed robust Capon beamformer can no longer be expressed in a closed form, but it can be efficiently computed. Its excellent performance is demonstrated via a number of numerical examples.

367 citations


Book
06 Feb 2003
TL;DR: This work focuses on a class of Exponentiated Adaptive Algorithms for the Identification of Sparse Impulse Responses, and on algorithms for Adaptive Equalization in Wireless Applications.
Abstract: 1 On a Class of Exponentiated Adaptive Algorithms for the Identification of Sparse Impulse Responses.- 2 Adaptive Feedback Cancellation in Hearing Aids.- 3 Single-Channel Acoustic Echo Cancellation.- 4 Multichannel Frequency-Domain Adaptive Filtering with Application to Multichannel Acoustic Echo Cancellation.- 5 Filtering Techniques for Noise Reduction and Speech Enhancement.- 6 Adaptive Beamforming for Audio Signal Acquisition.- 7 Blind Source Separation of Convolutive Mixtures of Speech.- 8 Adaptive Multichannel Time Delay Estimation Based on Blind System Identification for Acoustic Source Localization.- 9 Algorithms for Adaptive Equalization in Wireless Applications.- 10 Adaptive Space-Time Processing for Wireless CDMA.- 11 The IEEE 802.11 System with Multiple Receive Antennas.- 12 Adaptive Estimation of Clock Skew and Different Types of Delay in the Internet Network.

167 citations


Proceedings ArticleDOI
01 Oct 2003
TL;DR: The doubly constrained robust Capon beamformer (DCRCB) was proposed in this paper to improve robustness in the presence of array steering vector errors by enforcing a constant norm constraint and a spherical uncertainty set constraint.
Abstract: The standard Capon beamformer (SCB) is known to have better resolution and much better interference rejection capability than the standard data-independent beamformer when the array steering vector is accurately known. However, the major problem of SCB is that it lacks robustness in the presence of array steering vector errors. In this paper, we provide a natural extension of SCB, obtained via covariance matrix fitting, to the case of uncertain steering vectors by enforcing a double constraint on the array steering vector, viz. a constant norm constraint and a spherical uncertainty set constraint, which we refer to as the doubly constrained robust Capon beamformer (DCRCB). DCRCB can be efficiently computed at a comparable cost with that of SCB. Performance comparisons of DCRCB and our previously proposed robust Capon beamformer (RCB) are also presented via a number of numerical examples.

142 citations


Journal ArticleDOI
TL;DR: A new approach to adaptive beamforming with sidelobe control is developed that minimizes the array output power while maintaining the distortionless response in the direction of the desired signal and a sidelobe level that is strictly guaranteed to be lower than some given threshold value.
Abstract: A new approach to adaptive beamforming with sidelobe control is developed. The proposed beamformer represents a modification of the popular minimum variance distortionless response (MVDR) beamformer. It minimizes the array output power while maintaining the distortionless response in the direction of the desired signal and a sidelobe level that is strictly guaranteed to be lower than some given (prescribed) threshold value. The resulting modified MVDR problem is shown to be convex, and its second-order cone (SOC) formulation is obtained that facilitates a computationally efficient way to implement our beamformer using the interior point method.

117 citations


Journal ArticleDOI
TL;DR: Frequency-domain blind source separation is shown to be equivalent to two sets of frequency-domain adaptive beamformers (ABFs) under certain conditions and an interpretation of BSS from a physical point of view is given.
Abstract: Frequency-domain blind source separation (BSS) is shown to be equivalent to two sets of frequency-domain adaptive beamformers (ABFs) under certain conditions. The zero search of the off-diagonal components in the BSS update equation can be viewed as the minimization of the mean square error in the ABFs. The unmixing matrix of the BSS and the filter coefficients of the ABFs converge to the same solution if the two source signals are ideally independent. If they are dependent, this results in a bias for the correct unmixing filter coefficients. Therefore, the performance of the BSS is limited to that of the ABF if the ABF can use exact geometric information. This understanding gives an interpretation of BSS from a physical point of view.

98 citations


Proceedings ArticleDOI
09 Nov 2003
TL;DR: This paper overviews a recently emerged approach to robust adaptive beamforming using worst-case performance optimization and shows how this approach can improve the robustness of adaptive beamform techniques against environmental and array imperfections and uncertainties.
Abstract: In recent decades, adaptive arrays have been widely used in sonar, radar, wireless communications, microphone array speech processing, medical imaging and other fields. In practical array systems, traditional adaptive beamforming algorithms are known to degrade if some of exploited assumptions on the environment, sources, or antenna array become wrong or imprecise. Therefore, the robustness of adaptive beamforming techniques against environmental and array imperfections and uncertainties remains one of the key issues. In this paper, we overview a recently emerged approach to robust adaptive beamforming using worst-case performance optimization.

85 citations


Journal ArticleDOI
TL;DR: The proposed method offers better performance regarding suppression levels of disturbing signals and much less distortion to the source speech in a car hands-free mobile telephony environment.
Abstract: This paper presents a new method for the design of oversampled uniform DFT-filter banks for the special application of subband adaptive beamforming with microphone arrays. Since array applications rely on the fact that different source positions give rise to different signal delays, a beamformer alters the phase information of the signals. This in turn leads to signal degradations when perfect reconstruction filter banks are used for the subband decomposition and reconstruction. The objective of the filter bank design is to minimize the magnitude of all aliasing components individually, such that aliasing distortion is minimized although phase alterations occur in the subbands. The proposed method is evaluated in a car hands-free mobile telephony environment and the results show that the proposed method offers better performance regarding suppression levels of disturbing signals and much less distortion to the source speech.

Book ChapterDOI
01 Jan 2003
TL;DR: From this, a robust generalized sidelobe canceller (GSC) results as an attractive solution for practical audio acquisition systems and the general theoretical framework leads to new insights for the GSC behavior in complex practical situations.
Abstract: This chapter provides an overview of adaptive beamforming techniques for speech and audio signal acquisition. We review basic concepts of optimum adaptive antenna arrays and show how these methods may be applied to meet the requirements of audio signal processing. In particular, we derive optimum beamformers using time-domain least-squares instead of frequency-domain minimum mean-squares criteria, and, thereby, are not constrained by the commonly used narrow-band and stationarity assumptions. We thus obtain a more general representation of various beamforming aspects relevant to our application. From this, a robust generalized sidelobe canceller (GSC) [1] results as an attractive solution for practical audio acquisition systems. Moreover, the general theoretical framework leads to new insights for the GSC behavior in complex practical situations.

Proceedings ArticleDOI
06 Apr 2003
TL;DR: The proposed method makes a connection between the diagonal loading value and the estimation error of the estimated covariance matrix, and adjusts the diagonalloading value according to the array data.
Abstract: It is well known that the performance of the adaptive beamforming algorithm is degraded when the sample support is short. The diagonal loading method is a simple and efficient method for improving the robustness of adaptive beamforming for such cases. Meanwhile, we are not aware of a formal approach to determine an optimal diagonal value to date. In this paper, we propose a data dependent method for the determination of the diagonal loading value. The proposed method makes a connection between the diagonal loading value and the estimation error of the estimated covariance matrix. The larger the estimation error, the larger the diagonal loading value. Thus, the proposed method adjusts the diagonal loading value according to the array data. In addition, this method is efficient in computation.

Proceedings ArticleDOI
27 Oct 2003
TL;DR: This paper presents an overview of recent trends and advances in the field of robust adaptive beamforming and outlines the priorities for further research into this area.
Abstract: In recent decades, adaptive arrays have been widely used in sonar, radar, wireless communications, microphone array speech processing, medical imaging and other fields. In practical array systems, traditional adaptive beamforming algorithms are known to degrade if some of exploited assumptions on the environment, sources, or antenna array become wrong or imprecise. Therefore, the robustness of adaptive beamforming techniques against environmental and array imperfections and uncertainties is one of the key issues. In this paper, we present an overview of recent trends and advances in the field of robust adaptive beamforming.

Proceedings ArticleDOI
20 Mar 2003
TL;DR: In this paper, the problem of degradations in adaptive digital beamforming (DBF) systems caused by mutual coupling between array elements is addressed by the implementation of a RF-decoupling network.
Abstract: This paper addresses the problem of degradations in adaptive digital beam-forming (DBF) systems caused by mutual coupling between array elements. The focus is on compact arrays with reduced element spacing and, hence, strongly coupled elements. Deviations in the radiation patterns of coupled and (theoretically) uncoupled elements can be compensated for by weight-adjustments in DBF, but SNR degradation due to impedance mismatches cannot be compensated for via signal processing techniques. It is shown that this problem can be overcome via the implementation of a RF-decoupling-network. SNR enhancement is achieved at the cost of a reduced frequency bandwidth and an increased sensitivity to dissipative losses in the antenna and matching network structure.

Journal ArticleDOI
TL;DR: A novel approach for real-time multichannel speech enhancement in environments of nonstationary noise and time-varying acoustical transfer functions (ATFs) by integrating adaptive beamforming, ATF identification, soft signal detection, and multich channel postfiltering.
Abstract: We present a novel approach for real-time multichannel speech enhancement in environments of nonstationary noise and time-varying acoustical transfer functions (ATFs). The proposed system integrates adaptive beamforming, ATF identification, soft signal detection, and multichannel postfiltering. The noise canceller branch of the beamformer and the ATF identification are adaptively updated online, based on hypothesis test results. The noise canceller is updated only during stationary noise frames, and the ATF identification is carried out only when desired source components have been detected. The hypothesis testing is based on the nonstationarity of the signals and the transient power ratio between the beamformer primary output and its reference noise signals. Following the beamforming and the hypothesis testing, estimates for the signal presence probability and for the noise power spectral density are derived. Subsequently, an optimal spectral gain function that minimizes the mean square error of the log-spectral amplitude (LSA) is applied. Experimental results demonstrate the usefulness of the proposed system in nonstationary noise environments.

Journal ArticleDOI
TL;DR: In this paper, two broad-band beamformers were designed for speech pickup in the near field. Butler et al. showed that the first beamformer provides optimum gain for spatially incoherent noise while the second beamformer provided optimum gain in spherically isotropic noise.
Abstract: This paper discusses the application of fixed microphone arrays to speech pickup in mobile telephone applications. Array optimization techniques are used to design two broad-band beamformers for speech pickup in the near field. The first beamformer provides optimum gain for spatially incoherent noise while the second beamformer provides optimum gain in spherically isotropic noise. Array performance was measured using vehicular noise recorded under realistic driving conditions. Results obtained are in agreement with theoretical predictions for a spherically isotropic noise field and are comparable to previously reported results obtained using adaptive beamforming algorithms.

Proceedings ArticleDOI
07 Sep 2003
TL;DR: A novel slotted MAC protocol for nodes equipped with adaptive antenna array in ad hoc network that relies on the ability of antenna to uses DOA (direction-of-arrival) information to beamform by placing nulls in the direction of interferers to maximize SINR at the receiver.
Abstract: This paper presents a novel slotted MAC (medium access control) protocol for nodes equipped with adaptive antenna array in ad hoc network. The protocol relies on the ability of antenna to uses DOA (direction-of-arrival) information to beamform by placing nulls in the direction of interferers thus maximize SINR (signal to interference and noise ratio) at the receiver. We studied the performance of the protocol using joint simulation in OPNET and Matlab. We studied the impact of variable number of antenna elements, DOA algorithm, and nulling. The performance of our new protocol is compared against one of the recent directional MAC protocols [R.R.N.H.V. Romit Roy Choudhury, Xue Yang, 2002]. We observe that despite the simplicity of our protocol it achieves high throughput.

Patent
07 Mar 2003
TL;DR: In this paper, 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, is described.
Abstract: 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.

Proceedings ArticleDOI
12 May 2003
TL;DR: The wideband RELAX (WB-RELAX) and the wideband CLEAN ( WB-CLEAN) algorithms are presented for aeroacoustic imaging using an acoustic array and not only were the parameters of the dominant source accurately determined, but a highly correlated multipath of theinant source was also discovered.
Abstract: Microphone arrays can be used for acoustic source localization and characterization in wind tunnel testing. In this paper, the wideband RELAX (WB-RELAX) and the wideband CLEAN (WB-CLEAN) algorithms are presented for aeroacoustic imaging using an acoustic array. WB-RELAX is a parametric approach that can be used efficiently for point source imaging without the sidelobe problems suffered by the delay-and-sum beamforming approaches. WB-CLEAN does not have sidelobe problems either, but it behaves more like a nonparametric approach and can be used for both point source and distributed source imaging. Moreover, neither of the algorithms suffers from the severe performance degradations encountered by the adaptive beamforming methods when the number of snapshots is small and/or the sources are highly correlated or coherent with each other. A two-step optimization procedure is used to implement the WB-RELAX and WB-CLEAN algorithms efficiently. The performance of WB-RELAX and WB-CLEAN is demonstrated by applying them to measured data obtained at the NASA Langley Quiet Flow Facility using a small aperture directional array (SADA). Somewhat surprisingly, using these approaches, not only were the parameters of the dominant source accurately determined, but a highly correlated multipath of the dominant source was also discovered.

Journal ArticleDOI
TL;DR: This method is evaluated and shown to simultaneously decrease word-error-rate (WER) for speech recognition and improve speech quality via the SEGSNR measure by up to +5.5 dB on the average.
Abstract: While a number of studies have investigated various speech enhancement and processing schemes for in-vehicle speech systems, little research has been performed using actual voice data collected in noisy car environments. In this paper, we propose a new constrained switched adaptive beamforming algorithm (CSA-BF) for speech enhancement and recognition in real moving car environments. The proposed algorithm consists of a speech/noise constraint section, a speech adaptive beamformer, and a noise adaptive beamformer. We investigate CSA-BF performance with a comparison to classic delay-and-sum beamforming (DASB) in realistic car conditions using a corpus of data recorded in various car noise environments from across the U.S. After analyzing the experimental results and considering the range of complex noise situations in the car environment using the CU-Move corpus, we formulate the three specific processing stages of the CSA-BF algorithm. This method is evaluated and shown to simultaneously decrease word-error-rate (WER) for speech recognition by up to 31% and improve speech quality via the SEGSNR measure by up to +5.5 dB on the average.

Journal ArticleDOI
TL;DR: It is experimentally demonstrated that adaptive beamforming is feasible on a turning array, provided that array shape is estimated, and the proposed algorithms have been tested on real data with the tow-vessel making 45/spl deg/ turns with a 500-m curvature radius.
Abstract: During maneuvering, towed array beamforming degrades if a straight array is assumed. This is especially true for high-resolution adaptive beamforming. It is experimentally demonstrated that adaptive beamforming is feasible on a turning array, provided that array shape is estimated. The array shape can be inferred solely from the coordinates of the tow vessel's Global Positioning System (GPS) without any instrumentation in the array. Based on estimated array shape from the GPS, both the conventional beamformer and the white noise constrained (WNC) adaptive beamformer are shown to track the source well during a turn. When calculating the weight vector in the WNC approach, a matrix inversion of the cross-spectral density matrix is involved. This matrix inversion can be stabilized by averaging the cross-spectral density matrix over neighboring frequencies. The proposed algorithms have been tested on real data with the tow-vessel making 45/spl deg/ turns with a 500-m curvature radius. While turning, the improvement in performance over the assumption of a straight array geometry was up to 5 dB for the conventional beamformer and considerably larger for the WNC adaptive beamformer.

Patent
19 Dec 2003
TL;DR: In this paper, a transducer probe assembly is used to adapt signals from an ultrasound transducers for an ultrasound system. But, the transducers are not equipped with the capability of de-multiplexing the data for beamforming.
Abstract: Methods and systems are provided for adapting signals from an ultrasound transducer for an ultrasound system. Where the signal processing in a transducer assembly outputs data incompatible with the ultrasound system, circuitry provided within the transducer assembly converts the data to be compatible with the ultrasound systems. For example, sub-array mixing is provided to partially beamform signals from a plurality of transducer elements. The resulting output signals from a plurality sub-arrays are provided through a cable to a connector housing of the transducer probe assembly. Since the mixers alter the data, such as shifting the data to an intermediate frequency, the output data may be at a frequency different than the frequencies for operation of the receive beamformer. Additional mixers are then provided to convert the intermediate frequency signals to radio frequency signals that may be processed by the ultrasound systems received beamformer. As another example, signals from a plurality of transducer elements are multiplexed together. Where the receive beamformer is not operable to de-multiplex such signals, circuitry within the transducer probe assembly converts the signals by de-multiplexing the data for beamforming. Ultrasound systems have a limited number of received beamformer channels. By providing signal processing, conversion, and/or partial beamforming within the transducer probe assembly, the number of elements used may be different than the number of received beamformer channels provided by the system.

Proceedings ArticleDOI
08 Jun 2003
TL;DR: In this article, a hybrid approach that integrates the features of the switched beam method and the adaptive beam forming approach was proposed, which is fast, computationally efficient, and provides a cost effective approach for exploiting space diversity.
Abstract: In this paper we present a new procedure for implementing smart antenna algorithms. It is a hybrid approach that integrates the features of the switched beam method and the adaptive beam forming approach. Specifically it is shown that by using high gain antenna elements and combining the switched beam process with the adaptive beam forming procedure on a limited number of elements (as low as 2 in an 8-element array), a performance close to that of a more complex 8-element adaptive array may be achieved. The proposed hybrid method, therefore, is fast, computationally efficient, and provides a cost effective approach for exploiting space diversity. Even with the inclusion of interference signals, the proposed hybrid approach out-performed the switched beam method, and provided performance similar to that of an adaptive array with less number of elements (3 in an 8-element array). Implementation of an adaptive array also includes estimations; hence, reducing the number of elements in an array may lead to improved accuracy, in addition to fast convergence and reduced complexity.

Journal ArticleDOI
TL;DR: In this article, a small array composed of three monopole elements with very small element spacing on the order of λ/6 to λ /20 is considered for adaptive beamforming, and the properties of this 3-port array are governed by strong mutual coupling.
Abstract: A small array composed of three monopole elements with very small element spacing on the order of λ/6 to λ/20 is considered for application in adaptive beamforming. The properties of this 3-port array are governed by strong mutual coupling. It is shown that for signal-to-noise maximization, it is not sufficient to adjust the weights to compensate for the effects of mutual coupling. The necessity for a RF-decoupling network (RF-DN) and its simple realization are shown. The array with closely spaced elements together with the RF-DN represents a superdirective antenna with a directivity of more than 10 dBi. It is shown that the required fractional frequency bandwidth and the available unloaded Q of the antenna and RF-DN structure determine the lower limit for the element spacing.

Proceedings ArticleDOI
06 Apr 2003
TL;DR: It is shown that the MBER approach provides significant performance gain in terms of smaller bit error rate (BER) over the standard minimum mean square error (MMSE) approach.
Abstract: A novel adaptive beamforming technique is proposed for wireless communication application based on the minimum bit error rate (MBER) criterion. It is shown that the MBER approach provides significant performance gain in terms of smaller bit error rate (BER) over the standard minimum mean square error (MMSE) approach. Using the classical Parzen window estimate of probability density function (p.d.f.), both the block-data and sample-by-sample adaptive implementations of the MBER solution are developed.

Journal ArticleDOI
TL;DR: In this paper, a new type of adaptive beamforming antenna system architecture is proposed for multichannel wireless communications, which consists of analog mixers, a multitone direct digital synthesizer (DDS), and a digital signal processor (DSP) controller.
Abstract: A new type of adaptive beamforming antenna system architecture is proposed for multichannel wireless communications. Multibeam communication with high data throughput is accomplished using the proposed beamformer architecture. The system consists of analog mixers, a multitone direct digital synthesizer (DDS), and a digital signal processor (DSP) controller. The essential idea of multibeam forming is based on a multitone weighting scheme combined with analog-digital hybrid signal processing. While the real-time multibeam construction is realized by the analog mixer circuits and a DDS, the complicated adaptive beamforming and direction-of-arrival estimation algorithms are carried out by the DSP. In this architecture, only one beamformer circuit is required to handle multiple beams, leading to significant reduction in hardware counts. A 5.8-GHz eight-element adaptive beamforming array successfully demonstrates two-beam simultaneous beamforming with less than three degrees of peak and null steering errors and more than 20-dB interference suppression. The test-bed exhibits successful two-channel data recovery at 25-Mb/s data throughput in each channel with binary phase-shift keying modulation, for simultaneous dual-beam reception. The bit-error-rate measurement validates the robustness of the communication quality under strong interferences.

Journal ArticleDOI
TL;DR: This work proposes a method to realize space-time adaptive filtering (STAF) by employing ESPAR antennas for TDMA or CDMA signal waveforms and shows the effectiveness of the proposed method in the signal environment of a local network communication system.
Abstract: Electronically steerable passive array radiator (ESPAR) antennas are considered to have the capability to form a beam spatially toward the desired signal with the lowest cost. We propose a method to realize space-time adaptive filtering (STAF) by employing ESPAR antennas for TDMA or CDMA signal waveforms. According to the method, the cochannel interference signals are spatially suppressed by the adaptive beamforming, and the intersymbol interference signals are suppressed by the temporal waveform-based adaptive equalization. Simulation results show the effectiveness of the proposed method in the signal environment of a local network communication system.

Proceedings ArticleDOI
01 Dec 2003
TL;DR: A method for estimation of direct-to-reverberant energy ratio from partial room responses to an impulsive sound at one receiver location, and of source distance, average absorption, reverberation time, and room volume as intermediate results is presented.
Abstract: Pre-processing for hearing aids, such as adaptive beamforming, is sensitive to characteristics of the acoustic environment, in particular, reverberation. We conjecture that knowledge of the acoustic environment can aid selection of an optimal signal processing strategy. In this paper we present a method for estimation of direct-to-reverberant energy ratio from partial room responses to an impulsive sound at one receiver location, and of source distance, average absorption, reverberation time, and room volume as intermediate results.

Patent
Sang-Choon Kim1
12 Feb 2003
TL;DR: In this paper, an adaptive beamforming apparatus and method that despreads an input signal, and determines whether a symbol of despread signal belongs to a pilot sub-channel or non-pilot subchannel of the signal is presented.
Abstract: Disclosed are an adaptive beamforming apparatus and method that despreads an input signal, and determines whether a symbol of despread signal belongs to a pilot sub-channel or non-pilot sub-channel of the despread signal. One of two beamforming algorithms is accordingly enabled. If the symbol belongs to the pilot sub-channel, a first algorithm is used to calculate a weight vector, and if the symbol belongs to the non-pilot sub-channel, a second algorithm is used to calculate the weight vector. A current weight vector is updated using newly calculated weight vector, and a beam pattern is formed based on the updated weight vector.