Browsing by Author "Wu, WR"

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A constrained decision feedback equalizer for reduced complexity maximum likelihood sequence estimation
(2002-01-01)
Wu, WR
;
Tsuie, YM
;
電信工程研究所
;
Institute of Communications Engineering
The maximum likelihood sequence estimator (MLSE) is usually implemented by the Viterbi algorithm (VA). The computational complexity of the VA grows exponentially with the length of the channel response. With some performance reduction, a decision-feedback equalizer (DFE) can be used to shorten the channel response. This greatly reduces the computational requirement for the VA. However, for many real-world applications, the complexity of the DFE/MLSE approach may be still too high. In this paper, we propose, a constrained DFE that offers much lower VA computational complexity. The basic idea is to pose some constraints on the DFE such that the postcursors of the, shortened channel response have only discrete values. As a result, the multiplication operations can be replaced by shift operations making the VA almost multiplication free. This will greatly facilitate the real world applications of the MLSE algorithm. Simulation results show that while the proposed algorithm basically offers the same performance as the original MLSE performance, the VA is much more efficient than the conventional approach.
A low-complexity adaptive echo canceller for xDSL applications
(2004-05-01)
Lin, SS
;
Wu, WR
;
電信工程研究所
;
Institute of Communications Engineering
A finite impulse response (FIR)-based adaptive filter structure is proposed for echo cancellation in xDSL applications. The proposed algorithm consists of an FIR filter, a cascaded interpolated FIR filter, and a tap-weight overlapping and nulling scheme. This filter requires low computational complexity and inherits the stable characteristics of the conventional FIR filter. Simulations show that the proposed echo canceller can effectively cancel the echo, up to 73:4 dB [for a single-pair high-speed digital subscriber line (SHDSL) system]. About 55% complexity reduction can be achieved compared with a conventional FIR filter.
A novel adaptive code acquisition using antenna array for DS/CDMA systems
(2005)
Yang, HL
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Wu, WR
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電信工程研究所
;
Institute of Communications Engineering
This paper proposes a novel code acquisition scheme employing an adaptive beamformer and an adaptive temporal filter. Conventional code acquisition with beamforming requires a code-phase testing procedure which can be time-consuming. The proposed scheme simultaneously performs beamforming and code acquisition and the acquisition time can be much shorter than conventional approaches. Using the least mean square (LMS) adaptive algorithm, we are able to obtain a simple yet robust acquisition scheme. We also perform theoretical analysis which includes derivations of the minimum mean-squared error (MSE), the excess MSE, and the beamformer output signal-to-interference-plus-noise ratio (SINR). Simulation results verify that the proposed scheme has satisfactory performance and the derived theoretical results are accurate.
A robust adaptive generalized sidelobe canceller with decision feedback
(2005-11-01)
Lee, Y
;
Wu, WR
;
電信工程研究所
;
Institute of Communications Engineering
Conventional generalized sidelobe canceller (GSC) is sensitive to a mismatch between the estimated and actual direction of arrival (DOA) of the desired signal. Such a mismatch induces signal cancellation in the GSC, and it severely degrades the beamforming performance. In this paper, we propose a new decision feedback (DF) technique to increase the robustness against the DOA mismatch. Our new scheme introduces a blind equalizer and a feedback filter in the GSC structure. We first derive Wiener solutions for the DF-GSC with perfectly matched and mismatched DOA and show that the problem of signal cancellation can be avoided. Then, we consider the adaptive GSC implementation in which the least-mean-square (LMS) algorithm is used for weight adaptation. In addition to the improved robustness, the proposed scheme also remedies the slow convergence problem inherent in the conventional adaptive GSC structure. The convergence behavior of the LMS-based DF-GSC is fully analyzed and the analytic signal-to-interference-plus-noise ratio (SINR) is also derived. Finally, simulation results demonstrate that while the proposed structure can considerably enhance the overall performance, it has greatly improved robustness as compared to other existing robust adaptive beamformers.
A robust generalized sidelobe canceller against multipath induced perturbations in wireless communications
(2005)
Lee, YM
;
Wu, WR
;
電信工程研究所
;
Institute of Communications Engineering
Conventional generalized sidelobe canceller (GSC) is sensitive to coherent multipath disturbance in wireless communication environments. Such disturbance, including coherent interference and local scattering, causes signal cancellation in the GSC. In this paper, we propose a new decision feedback (DF) technique to overcome this problem. The proposed algorithm introduces a feedback filter with one tap weight in the GSC structure. We analyze the robustness of this algorithm and the result shows that the problem of signal cancellation can be avoided. Simulations demonstrate that the DF-GSC has greatly improved robustness as compared to the conventional GSC.
A two-stage MMSE partial PIC receiver for multiuser detection
(2001)
Wu, WR
;
Su, WS
;
Hsieh, YT
;
電信工程研究所
;
Institute of Communications Engineering
The partial parallel inteference cancellation (PIC) detector is considered as an effective yet simple receiver for multiuser detection in code-division multiple access (CDMA) systems. It cancels the multiple access interference (MAI) of the received signal partially resulted from other users. However, the optimal cancellation level which is controlled by a partial cancellation factor (PCF) is usually determined by trial and error, since no theoretical values were reported. In this paper we introduce a MMSE criterion for the two-stage partial PIC receiver to derive the optimal PCF. Simulations show that the performance of proposed methods is very close to that of empirically chosen PCF. We also demonstrate that the derived PCF is insensitive to the estimate bias of noise level and the interfering signal power.
Adaptive AR modeling in white Gaussian noise
(1997-05-01)
Wu, WR
;
Chen, PC
;
交大名義發表
;
電信工程研究所
;
National Chiao Tung University
Autoregresssive (AR) modeling is widely used in signal processing, The coefficients of an AR model can be easily obtained with a least mean square (LMS) prediction error filter, However, it is known that this filter gives a biased solution when the input signal is corrupted by white Gaussian noise, Treichler suggested the gamma-LMS algorithm to remedy this problem and proved that the mean weight vector can converge to the Wiener solution. In this paper, we develop a new algorithm that extends works of Vijayan et al, for adaptive AR modeling in the presence of white Gaussian noise, By theoretical analysis, we show that the performance of the new algorithm is superior to the gamma-LMS filter, Simulations are also provided to support our theoretical results.
Adaptive AR modeling in white Gaussian noise
(1997-05-01)
Wu, WR
;
Chen, PC
;
電信工程研究所
;
Institute of Communications Engineering
Autoregresssive (AR) modeling is widely used in signal processing, The coefficients of an AR model can be easily obtained with a least mean square (LMS) prediction error filter, However, it is known that this filter gives a biased solution when the input signal is corrupted by white Gaussian noise, Treichler suggested the gamma-LMS algorithm to remedy this problem and proved that the mean weight vector can converge to the Wiener solution. In this paper, we develop a new algorithm that extends works of Vijayan et al, for adaptive AR modeling in the presence of white Gaussian noise, By theoretical analysis, we show that the performance of the new algorithm is superior to the gamma-LMS filter, Simulations are also provided to support our theoretical results.
Adaptive asymptotic Bayesian equalization using a signal space partitioning technique
(2004-05-01)
Chen, RJ
;
Wu, WR
;
電信工程研究所
;
Institute of Communications Engineering
The Bayesian solution is known to be optimal for symbol-by-symbol equalizers; however, its computational complexity is usually very high. The signal space partitioning technique has been proposed to reduce complexity. It was shown that the decision boundary of the equalizer consists of a set of hyperplanes. The disadvantage of existing approaches is that the number of hyperplanes cannot be controlled. In addition, a state-search process, that is not efficient for time-varying channels, is required to find these hyperplanes. In this paper, we propose a new algorithm to remedy these problems. We propose an approximate Bayesian criterion that allows the number of hyperplanes to be arbitrarily set. As a consequence, a tradeoff can be made between performance and computational complexity. In many cases, the resulting performance loss is small, whereas the computational complexity reduction can be large. The proposed equalizer consists of a set of parallel linear discriminant functions and a maximum operation. An adaptive method using stochastic gradient descent has been developed to identify the functions. The proposed algorithm is thus inherently applicable to time-varying channels. The computational complexity of this adaptive algorithm is low and suitable for real-world implementation.
Adaptive channel equalization using approximate Bayesian criterion
(2002)
Chen, RJ
;
Wu, WR
;
電信工程研究所
;
Institute of Communications Engineering
The Bayesian solution is known to be optimal for the symbol-by-symbol type of equalizer. However, the computational complexity for the Bayesian equalizer is usually very high. Signal space partitioning technique has been proposed for complexity reduction. It was shown the decision boundary of the equalizer consists of a set of hyperplanes. The disadvantage of the existing approaches is that the number of hyperplane cannot be controlled. Also, to find these hyperplanes, it requires a state searching process which is not efficient for time-varying channels. In this paper, we propose a new algorithm to remedy this problem. We propose an approximate Bayesian criterion such that the number of hyperplanes can be arbitrarily set. As a result, we can trade between performance and computational complexity. In many cases, we can make the performance loss being small while the computational complexity reduction is huge. The resultant equalizer is composed of a set of parallel linear discriminant functions and a maximum operation. An adaptive method using stochastic gradient descent is developed to identify the functions. The proposed algorithm is then inherently applicable to time-varying channels. Also, the computational complexity is low and suitable for real-world implementation.
Adaptive filter bank reconstruction using fast multichannel RLS algorithms
(1998-06-11)
Chang, DC
;
Wu, WR
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電信工程研究所
;
Institute of Communications Engineering
The recursive least-squares (RLS) algorithm has been used in the adaptive synthesis filter bank for fast convergence. However, because of interpolation operations involved in the synthesis process, fast RLS algorithms cannot be applied. In this Letter, an approach is proposed that can formulate subband reconstruction as a multichannel filtering problem. This formulation allows the application of fast multichannel RLS algorithms and substantial reduction in computational complexity.
Adaptive identification of non-Gaussian/non-stationary glint noise
(1999-12-01)
Wu, WR
;
Wu, KG
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資訊工程學系
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電信工程研究所
;
Department of Computer Science
Non-stationary glint noise is often observed in a radar tracking system. The distribution of glint noise is non-Gaussian anti heavy-tailed. Conventional recursive identification algorithms use the stochastic approximation (SA) method. However, the SP, method converges slowly and is invalid for nonstationary noise. This paper proposes an adaptive algorithm, which uses the stochastic gradient descent (SGD) method, to overcome th(:se problems. The SGD method retains the simple structure of the SA method and is suitable for real-world implementation. Convergence behavior of the SGD method is analyzed and closed-form expressions for sufficient step size bounds are derived. Since noise data are usually not available in practice, we then propose a noise extraction scheme. Combining the SGD method, we I:an perform on-line adaptive noise identification directly from radar measurements. Simulation results show that the performance of the SGD method is comparable to that of the maximum-likelihood (ML) method. Also, the noise extraction scheme is effective that the identification results from the radar measurements are close to those from pure glint noise data.
An efficient architecture of ultrasonic scan conversion for implementing the cubic convolution interpolation
(1998)
Chang, DC
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Wu, WR
;
Jeng, JH
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電信工程研究所
;
Institute of Communications Engineering
The nearest neighbor and the bilinear interpolation methods are widely adopted to perform ultrasonic scan conversion in commercial ultrasound systems. The reason is that they are easy to implement and can satisfy real-time requirement. As the hardware technology advances in the recent decade, more complex yet more eligible interpolation methods can be realized. However, few articles in the literature concern details about the reconstruction of ultrasonic images. In this paper we propose an efficient architecture of ultrasonic scan conversion for implementing the cubic convolution interpolation. Some parameters needed to calculate in interpolation processes can be previously designed in ROMs (Read Only Memory) to improve computational efficiency. The system requirement analysis and computer simulations have shown the appealing feasibility in real-time practice and superior reconstruction quality to conventional interpolation methods.
An LMS-based adaptive generalized sidelobe canceller with decision feedback
(2005)
Lee, YM
;
Wu, WR
;
電信工程研究所
;
Institute of Communications Engineering
Conventional least-mean-square (LMS) based adaptive generalized sidelobe canceller (GSC) suffers from the slow convergence problem. When the environment is fast varying, the conventional GSC cannot perform satisfactorily. In this paper, we propose a new decision feedback (DF) technique to overcome this problem. The proposed algorithm introduces a feedback filter with one tap weight in the GSC structure. The Wiener solution for the proposed structure is derived and the convergence behavior of the adaptive DF-GSC is also analyzed. Simulation results show that the proposed structure can significantly accelerate the convergence and enhance the overall performance.
An LMS-based decision feedback equalizer for digital cellular radio
(1996)
Wu, WR
;
Tsuie, YM
;
交大名義發表
;
傳播研究所
;
National Chiao Tung University
In a TDMA mobile communication system, the equalizer is a common device to compensate the intersymbol interference caused by the multipath effect. In this paper, we consider the decision feedback equalizer (DFE) for application to the North American IS-54 systems. Due to the fast fading characteristic of channels, most conventional DFEs use the recursive least square (RLS) algorithm. However, it is known that the RLS algorithm is computationally intensive and not easy to implement. In this paper, we employ the ideas of multiple-training and multiple-tracking to propose an LMS-based equalization scheme. We also incorporate the Coded Digital Verification Color Code (CDVCC) of the TDMA slot as an additional training sequence. Simulation results show that the new scheme can achieve a bit error rate of 3% at the mobile speed of 80km/hr and still remains a low complexity.
An LMS-based decision feedback equalizer for IS-136 receivers
(2002-01-01)
Wu, WR
;
Tsuie, YM
;
電信工程研究所
;
Institute of Communications Engineering
In digital mobile communication systems, inter-symbol interference is one of the main causes of degrading system performance. Decision feedback equalization (DFE) is the commonly used remedy for this problem. Since the channel is fast-varying, an adaptive algorithm possessing a fast convergence property is then required. The least mean square (LMS) algorithm is well known for its simplicity and robustness; however, its convergence is slow. As a consequence, the LMS algorithm is rarely considered in this application. In this paper, we consider an LMS-based DFE for the North American IS-136 system. We propose an extended multiple-training LMS algorithm accelerating the convergence process. The convergence properties of the multiple-training LMS algorithm are also analyzed. We prove that the multiple-training LMS algorithm can converge regardless of its initial value and derive closed-form expressions for the weight error vector power. We further take advantage of the IS-136 downlink slot format and divide a slot into two subslots. Bidirectional processing is then applied to each individual subslot. The proposed LMS-based DFE has a low computational complexity and is suitable for real-world implementation. Simulations with a 900- MHz carrier show that our algorithm can meet the 3% bit error rate requirement for mobile speeds up to 100 km/hr.
An optimal interpolated FIR echo canceller for digital subscriber lines
(2004-12-01)
Lin, SS
;
Wu, WR
;
電信工程研究所
;
Institute of Communications Engineering
An adaptive interpolated FIR (IFIR) echo canceller was recently proposed for xDSL applications. This canceller consists of an FIR filter, an IFIR filter, and a tap-weight overlapping and nulling scheme. The FIR filter is used to cancel the short and rapidly changing head echo while the IFIR filter is used to cancel the long and slowly decaying tail echo. This echo canceller, which inherits the stable characteristics of the conventional FIR filter, requires low computational complexity. It is well known that the interpolation filter for an IFIR filter has great influence on the interpolated result. In this paper, a least-squares method is proposed to obtain optimal interpolation filters such that the performance of the IFIR echo canceller can be further improved. Simulations with a wide variety of loop topologies show that the optimal IFIR echo canceller can effectively cancel the echo up to 73.0 dB (for an SHDSL system). About 57% complexity reduction can be achieved compared to a conventional FIR filter.
An optimum two-stage partial parallel interference canceller for CDMA systems
(2002)
Hsieh, YT
;
Wu, WR
;
電信工程研究所
;
Institute of Communications Engineering
The parallel interference cancellation (PIC) is considered as an effective yet simple detector for multiuser communications in direct-sequence code division multiple access (DS-CDMA) systems. However, the system performance may be deteriorated due to unreliable interference estimate in early canceling stages. Thus the partial cancellation factor (PCF) is introduced to control the interference canceling level resulting in the partial PIC approach. While the PCF is critical, the existing methods for determining optimum PCF is either not general or not accurate. In this paper, we propose a method deriving a close-form solution for the optimum PCF in a two-stake partial PIC receiver. Our results apply to either power balanced or unbalanced, synchronous or asynchronous systems. Simulation results show that our theoretical optimal PCF values are closed to the empirical ones. Also, the optimal two-stage partial PIC even outperforms three-stage full PIC.
Clipping ratio estimation for OFDM receivers
(2005)
Lin, CT
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Wu, WR
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電信工程研究所
;
Institute of Communications Engineering
Signal clipping is known to be the simplest method to reduce the peak to average power ratio (PAPR) in OFDM systems. Clipping will introduce the clipping noise (CN) degrading system performance. Most clipping noise mitigation algorithms require to know the clipping ratio (CR) in the receiver; however, this may not be possible for all applications. We propose a new clipping ratio estimation method which can be applied to pilot-tone-based OFDM systems. Simulation results show that the proposed algorithm can estimate CR accurately and clipping noise can be effectively mitigated.
Comments on "Maneuvering target tracking with colored noise" - Author's reply
(1998-04-01)
Wu, WR
;
Chang, DC
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電信工程研究所
;
Institute of Communications Engineering