Browsing by Author "Su, YT"

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2D regression channel estimation for equalizing OFDM signals
(2000)
Chang, MX
;
Su, YT
;
電信工程研究所
;
Institute of Communications Engineering
In this paper, we present a channel estimation method for Orthogonal Frequency Division Multiplexing (OFDM) signals. Our method is based on a two-dimensional nonlinear regression method, taking into account the correlations of the fading process in both time and frequency domains. We derive a general bit error rate (BER) expression which can also used to predict the performance of many other OFDM channel estimates. The performance of this new estimate is very close to the theoretical bit error probability lower bound that is obtained by assuming that the-channel response is perfectly known. Unlike linear minimum mean-squared-error (LMMSE) channel estimates, it needs not to know or estimate channel statistics like channel correlation matrix and SNR whence is insensitive to additive white Gaussian noise (AWGN).
A new interleaver design and its application to turbo codes
(2002)
Zheng, YX
;
Su, YT
;
電信工程研究所
;
Institute of Communications Engineering
We present a general interleaver structure that can be systematic derived from any existing block interleaver. When used in a turbo code with proper encoding and decoding procedures, the proposed interleaver is capable of providing unbounded interleaving depth with bounded decoding delay. Given a fixed decoding delay, the new interleaver results in superior bit error rate (BER) performance. The algebraic properties of the associated turbo codes are examined to provide guidelines in design the permutation rule.
Adaptive VSS blind equalizers
(1997-04-01)
Ueng, FB
;
Su, YT
;
交大名義發表
;
電信工程研究所
;
National Chiao Tung University
It is well known that an adaptive filter with a large step size in the transient period and a small one in the convergence period gives small mean squared steady state error while achieving a fast convergence rate. Based upon this idea, we present two variable step size (VSS) blind equalizers. The first one employs an intersymbol interference (ISI) estimator to control the step size, the second algorithm uses a mean squared error (MSE) estimator to adjust the step size. Both accomplish what has been expected.
Adaptive VSS blind equalizers
(1997-04-01)
Ueng, FB
;
Su, YT
;
電信工程研究所
;
Institute of Communications Engineering
It is well known that an adaptive filter with a large step size in the transient period and a small one in the convergence period gives small mean squared steady state error while achieving a fast convergence rate. Based upon this idea, we present two variable step size (VSS) blind equalizers. The first one employs an intersymbol interference (ISI) estimator to control the step size, the second algorithm uses a mean squared error (MSE) estimator to adjust the step size. Both accomplish what has been expected.
Analysis of packet interference and agregrated throughput in a cluster of Bluetooth piconets under different traffic conditions
(2005-06-01)
Naik, K
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Wei, DSL
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Su, YT
;
Shiratori, N
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電信工程研究所
In a Bluetooth piconet, the Master essentially controls the channel. Due to an absence of coordination between independent Masters while accessing the wireless medium, devices will encounter high packet interference if several piconets are simultaneously operating in the same area. Since even a headset and a mobile phone can be connected with a Bluetooth link forming a piconet, it may not be unusual to find tens of independent piconets in crowded places like airports, international conferences, shopping malls, and so on. Study of packet interference is important because interference affects the throughput of a piconet. Motivated by the fact that applications will benefit, in terms of higher available data rate in one direction, by using multiple-slot packets in an asymmetric manner, in this paper, we present an analytical model of packet interference in a cluster of piconets using multiple-slot packets. Also, considering that all the portable devices can have a Bluetooth interface and people are highly mobile these days, it will not be uncommon to find a cluster of piconets; of both the 79-hop and the 23-hop types in the same area. We then present an analytical model of interference of multiple-slot packets in a heterogeneous cluster of Bluetooth piconets. By a heterogeneous cluster we mean some piconets are of the 23-hop type and the rest are of 79-hop type. We show how the aggregate throughput in a cluster of piconets degrade under various traffic scenarios, such as 1-slot, 3-slot, and 5-slot packets in symmetric and asymmetric modes in synchronous and asynchronous conditions of Master clocks. Our analytic model is based on the idea of probabilistic graphs, where a node denotes a piconet and an edge denotes the probability of interference between two nodes. Though the 23-hop system has been phased out, our work gives a general approach to model packet interference in multiple, frequency-hopping systems that need not be Bluetooth systems.
Antijam capability analysis of RS-coded slow frequency-hopped systems
(2000-02-01)
Su, YT
;
Jeng, LD
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電信工程研究所
;
Institute of Communications Engineering
The application of Reed-Solomon codes in slow frequency-hopped systems has been extensively studied. Earlier investigations assumed an infinite interleaving length and considered partial-band noise jammers only. This paper extends previous efforts by analyzing the effect of finite interleaving length and the impact of band multitone jammers, We also explain why two-threshold (2T) erasure-insertion methods (EIM) are needed and examine their performance, Numerical results are presented to compare the effectiveness of EIM's and jammer types and to stud!, the relationships among the hop rate, the interleaver size, and the code rate, The use of 2T EIM's necessitates the estimation of several additional channel and signal parameters, Simple and effective estimation algorithms are provided as well.
Bit-error bounds for trellis-coded MPSK in mixed fading channels
(1997-11-01)
Lin, WC
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Su, YT
;
交大名義發表
;
電信工程研究所
;
National Chiao Tung University
Bit-error probability (BEP) bounds of trellis-coded MPSK systems over two classes of mixed fading channels are studied, These two classes of channels have been proposed as candidate models for mobile satellite communications, The first class consists of slow and frequency-nonselective fading channels whose output field strengths follow a probability law characterized by a convex combination of Rician and Rayleigh/lognormal distributions, For the other class of fading channels, the received signal amplitude has a convex combination of Rician and Rician/lognormal distributions. We analyze performance bounds for trellis codes that belong to the class of either geometrically uniform codes (GUC's) or quasi-regular codes (QRC's), Receivers with either ideal channel state information (CSI) or no CSI at all are considered, We examine asymptotic behaviors of these codes and identify key design parameters, Numerical results are provided to illustrate and compare the BEP performances of various codes and to validate the usefulness of the asymptotic analysis.
Blind and semiblind detections of OFDM signals in fading channels
(2004-05-01)
Chang, MX
;
Su, YT
;
電信工程研究所
;
Institute of Communications Engineering
This paper considers the problem of blind joint channel estimation and data detection for orthogonal frequency-division multiplexing (OFDM) systems in a fading environment. Employing a regression model for a time-varying channel, we convert the problem into one that finds the data sequence x whose associated least-squares'(LS) channel estimate z(x) is closest to the space of some regression curves (surfaces). We apply the branch-and-bound principle to solve the nonlinear integer programming problem associated with finding the curve that fits a subchannel in the LS sense. A recursive formula for fast metric update is obtained by exploiting the intrinsic characteristic of our objective function. The impacts of reordering the data sequence and selective detection are addressed. By employing a preferred order along with a selective detection method, we greatly reduce the detector complexity while giving up little performance loss. Both the complete and the reduced-complexity algorithms can be used for blind and semiblind detections of OFDM signals in a subchannel-by-subchannel manner. To further reduce the complexity and exploit the frequency-domain channel correlation, we suggest a two-stage approach that detects a few selected positions in some subchannels first, and then, treating the detected symbols as pilots, determines the remaining symbols within a properly chosen time-frequency block by a two-dimensional model-based pilot-assisted algorithm. The proposed methods do not require the information of the channel statistics like signal-to-noise ratio or channel correlation function. Performance of differential modulations like differential quaternary phase-shift keying and STAR 16-ary quadrature amplitude modulation are provided. Both blind and semiblind schemes yield satisfactory performance.
Blind detection of OFDM signals in fading channels
(2003)
Su, YT
;
Chang, MX
;
電信工程研究所
;
Institute of Communications Engineering
Blind identification and deconvolution algorithms using higher-order cumulants
(1997-05-01)
Ueng, FB
;
Su, YT
;
電信工程研究所
;
Institute of Communications Engineering
The use of second-order statistics in identification or deconvolution incurs error due to additive Gaussian noise. In contrast, higher-order statistics (HOS) are insensitive to Gaussian perturbation and can be used to characterize nonminimum phase (NMP) systems. In this paper we propose batch, recursive and adaptive form algorithms based on third- and fourth-order cumulants to solve the identification and deconvolution problems. A new order determination procedure is also presented. Simulation results demonstrate that our algorithms do have superior performance when compared with existing algorithms.
Blind identification and deconvolution algorithms using higher-order cumulants
(1997-05-01)
Ueng, FB
;
Su, YT
;
電信工程研究所
;
Institute of Communications Engineering
The use of second-order statistics in identification or deconvolution incurs error due to additive Gaussian noise. In contrast, higher-order statistics (HOS) are insensitive to Gaussian perturbation and can be used to characterize nonminimum phase (NMP) systems. In this paper we propose batch, recursive and adaptive form algorithms based on third- and fourth-order cumulants to solve the identification and deconvolution problems. A new order determination procedure is also presented. Simulation results demonstrate that our algorithms do have superior performance when compared with existing algorithms.
Blind joint channel and data estimation for OFDM signals in Rayleigh fading
(2001)
Chang, MX
;
Su, YT
;
電信工程研究所
;
Institute of Communications Engineering
In this paper we present several blind model-based joint channel and data estimation methods for Orthogonal Frequency Division Multiplexing (OFDM) signals. We model the time-varying channel response as a polynomial in time. Joint channel estimation and data detection are accomplished by finding the data sequence and regression coefficients that results in the minimum metric between the data-dependent polynomial and the received samples. Our method does not require the information of the channel statistics like signal-to-noise ratio (SNR) or correlation function. Performing exhaustive search among candidate sequences, though optimal, is impractical for long sequences. We develop some suboptimal methods and discuss their pro and con. A two-stage hybrid detection algorithm is proposed and used for detecting differential phase shift keying (DPSK) signals in Rayleigh fading.
Carrier-to-interference ratio measurement using moments or histograms
(2000-08-01)
Su, YT
;
Chen, JY
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電信工程研究所
;
Institute of Communications Engineering
Cochannel interference is often the predominant factor that limits the capacity of a cellular mobile radio system. Power control is thus needed so that carrier-to-cochannel interference ratio (CIR) can be maintained within a tolerable bound. Many power control algorithms that require real-time CIR measurement have been proposed but there is very few mentioning of the measurement method, This paper presents viable solutions for real-time CIR estimations under various channel conditions. These solutions are based on either the method of moments or the histogram matching concept. We provide numerical results to demonstrate the usefulness and to compare the performance of the proposed algorithms. It is found that the histogram matching method with a properly-designed nonuniform quantizer offers excellent CIR estimation.
Cochannel interference rejection in multipath channels
(1997-12-01)
Su, YT
;
Jeng, LD
;
Ueng, FB
;
電信工程研究所
;
Institute of Communications Engineering
In addition to additive thermal noise, a received direct-sequence spread spectrum (DS/SS) signal may suffer from intersymbol interference (ISI) and interference caused by cochannel narrowband users or other narrowband radio frequency interference (RFI). This paper presents a workable solution for removing narrowband interference (NBI) and reducing ISI or inter-chip interference (ICI) when the communication channel can be modeled as an FIR filter and the NBI comes from multiple CW tones, an AR-modeled Gaussian process, or a BPSK signal. Unlike earlier solutions, the proposed scheme is capable of performing the functions of NBI-rejection, ISI/ICI suppression and data detection (code despreading) simultaneously. It is easy to implement and, more importantly, it yields lower bit error rate (BER) and smaller mean squared error (MSE).
Code acquisition in a class of time-varying channels
(1998)
Su, YT
;
Wu, RC
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電信工程研究所
;
Institute of Communications Engineering
This paper presents a model for the code acquisition in a time-varying channel and evaluates the associated acquisition time statistics. We first use a finite-state Markov chain to represent the time-varying channel. Each channel state can have either a frequency-selective or a frequency-nonselective fading characteristic. This model is an unconstrained model as it does not take into account two time-parameters. The first time-parameters the "time unit" for the Markov channel, which has to be determined by the channel Doppler spread or its fading rate. The second time-parameter has to do with the detection structure. Incorporating the constraints imposed by these two time-parameters we discuss modifications of the unconstrained model, derive related statistics (e.g., probability distribution function or characteristic function) of the acquisition time and then provide numerical examples of the acquisition time performance.
Errors-and-erasures decoding of block codes in hidden Markov channels
(2001)
Lee, CM
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Su, YT
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電信工程研究所
;
Institute of Communications Engineering
Most investigation on the performance of block codes assume perfect interleaving and memoryless channels that introduce only random errors. Some authors investigate the effect of finite interleaving size on the decoder performance, using a two-state Gilbert-Elliott (GE) model to characterize the channel effect. However, there are circumstances that the burst-error channel of concern can not be properly described by a two-state GE model. This paper extends earlier analysis to investigate the performance of block codes in an environment that can be modeled as an arbitrary finite-state Markov chain.
Frequency acquisition and tracking for mobile LEO satellite communications
(1997)
Su, YT
;
Wu, RC
;
交大名義發表
;
電信工程研究所
;
National Chiao Tung University
A class of recursive frequency acquisition and tracking algorithms for use in high dynamic mobile low earth orbit (LEO) satellite communication systems is proposed. The underlying principle of these algorithms is to apply the least-squares method to the reconstructed phase trajectories to estimate and predict the received frequency variations (Doppler frequency, Doppler rate, etc.). We consider both additive white Gaussian noise (AWGN) and Rician fading channels and analyze the corresponding mean squared errors (MSE). Numerical performance of the proposed algorithms demonstrates that the new frequency tracking algorithms not only has rapid acquisition times but also gives smaller tracking jitters.
Frequency acquisition and tracking in high dynamic environments
(2000-11-01)
Su, YT
;
Wu, RC
;
電信工程研究所
;
友訊交大聯合研發中心
;
Institute of Communications Engineering
This paper presents mean squared error (MSE) analysis of two classes of frequency acquisition and tracking algorithms. Additive white Gaussian noise as well as Rician fading channels are considered, The class of batch-processing algorithms is an extension of earlier least squares proposals used in more benign (lower dynamic) environments. These algorithms try to fit the phase trajectory of the down-converted samples of a received signal. Such a trajectory will depend on the histories of both the signal and the local frequency variations when the local frequency is updated recursively. We propose a method to solve this difficulty and present both first-order and second-order recursive algorithms. Numerical results demonstrate that the MSE performance predicted by our analysis is consistent with that estimated by computer simulation and that the proposed algorithms not only provide rapid acquisition times but also give small tracking jitters.
Frequency estimation of phase-modulated carriers
(1998-12-01)
Su, YT
;
Wu, RC
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電信工程研究所
;
Institute of Communications Engineering
Conventional approach for frequency estimation usually assume a single tone without data modulation. In many applications such an assumption, realized by using either a separate pilot beacon or synchronization preamble is not feasible. This paper deals with frequency estimation of phase-modulated carriers in the absence of timing information and known data pattern. We introduce new frequency estimators that are based on the generalized maximum likelihood principle. The communication channels under consideration include both additive white Gaussian noise (AWGN) channels and correlated Rician fading channels. For the latter class, we distinguish between the case when the fading (amplitude) process is tracked and that when it is not tracked.
Iterative decoding algorithm and terminating criteria for inter-block permuted turbo codes
(2004)
Zheng, YX
;
Su, YT
;
電信工程研究所
;
Institute of Communications Engineering
Inter-block permutation (IBP) concept is a simple but effective technique to improve the performance of a conventional turbo code. On the other hand, proper early termination leads to reduced decoding latency without compromising the performance. For inter-block permuted turbo codes (IBP-TC), however, the termination time variation over blocks within the interleaving span necessitates a new design paradigm. We use both cyclic redundancy check (CRC) code and sign check to make a termination decision and generate soft information for unterminated neighboring blocks at the same time. The new decoder that combines the features of iterative decoding and variable stopping time not only yields less average decoding delay but also outperforms its fixed-stopping-time counterpart by 0.2 - 0.3 dB.