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Intercarrier Interference Analysis for Wireless OFDM in Mobile Channels

Intercarrier Interference Analysis for Wireless OFDM in Mobi...

引用

2006 IEEE Wireless communications and Networking Conference (WCNC 2006), vol.4

作者： Xiaozhou Huang Hsiao-Chun Wu Communications and Signal Processing Laboratory Electrical and Computer Engineering Department Louisiana State University Baton Rouge LA USA

Orthogonal Frequency Division Multiplexing (OFDM) employs a set of subcarriers for the information symbol transmission in parallel via the multipath fading channels. Due to its spectral efficiency and robustness over multipath channels, OFDM has been adopted for broadband communications. However, for high-speed mobility, the time-varying channels pose a performance limitation to the wireless OFDM systems. In this paper, we investigate the OFDM system performance in the time-varying channels where the intercarrier interference (ICI) occurs;we study the impacts of the time-varying multipath fadings together with the multiple Doppler spreads. According to our analysis and simulation results, the maximum OFDM symbol normalized Doppler frequency must be less than 0.04 to achieve the signal-to-interference ratio (SIR)=20dB or larger. As the maximum OFDM symbol normalized Doppler frequency increases, the OFDM system performance degrades dramatically. Different irreducible symbol error rate (SER) floors about 10{sup}(-2) for QPSK-OFDM schemes and 10{sup}(-1) for 16QAM-OFDM schemes would arise in case that the maximum OFDM symbol normalized Doppler frequency is fixed at 0.06.

关键词： OFDM ICI Multiple Doppler frequencies Mobile channels Symbol error rate

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Robust Automatic Modulation Classification Using Cumulant Features in the Presence of Fading Channels

Robust Automatic Modulation Classification Using Cumulant Fe...

引用

2006 IEEE Wireless communications and Networking Conference (WCNC 2006), vol.4

作者： Songnan Xi Hsiao-Chun Wu Communications and Signal Processing Laboratory Department of Electrical and Computer Engineering Louisiana State University Baton Rouge LA USA

Automatic modulation classification (AMC) is a scheme to automatically identify the modulation types of the transmitted signals by observing the received data samples in the presence of noise and fading channels. Nowadays, AMC plays an important role in both cooperative and non-cooperative communication applications. Very often, multipath fading channels result in the severe AMC performance degradation or induce large classification errors. The negative impacts of multipath fading channels on AMC have been discussed in the existing literature but no solution has ever been proposed so far to the best of our knowledge. In this paper, we propose a new robust AMC algorithm, which applies higher-order statistics (HOS) in a generic framework for blind channel estimation and pattern recognition. The advantage of our new algorithm is that, by carefully designing the essential features needed for AMC, we don't really have to acquire the complete channel information and therefore it can be feasible without any a priori information in practice. The Monte Carlo simulation results show that our new AMC algorithm can achieve the much better classification accuracy than the existing AMC techniques.

关键词： AMC Modulation classification Blind channel estimation Cumulants Fading channels

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An Alternate Approach for Developing Higher Radix FFT Algorithms

An Alternate Approach for Developing Higher Radix FFT Algori...

引用

IEEE Asia-Pacific Conference on Circuits and Systems

作者： Saad Bouguezel M. Omair Ahmad M.N.S. Swamy Center for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal Canada

In this paper, the authors propose an alternate method to derive higher radix FFT algorithms by using a recursive approach and by appropriately combining the twiddle factors without increasing the structural complexity. Specifically, efficient radix-8 and radix-16 FFT algorithms are designed and their arithmetic complexities shown to be slightly less than those of the corresponding existing Cooley-Tukey FFT algorithms

关键词： signal processing algorithms Arithmetic Discrete Fourier transforms Hardware Algorithm design and analysis Fast Fourier transforms Software algorithms Table lookup Counting circuits Hydrogen

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A Reconfigurable FPGA-based 16-Channel Front-End for MRI

A Reconfigurable FPGA-based 16-Channel Front-End for MRI

引用

Asilomar Conference on signals, Systems & computers

作者： Ishaan L. Dalal Fred L. Fontaine Center for Signal Processing Communications and Computer Engineering Research Department of Electrical Engineering Cooper Union NY USA

Parallel MRI acquisitions are generally reconstructed off-line on PCs and computer clusters. Building upon an existing multichannel digital receiver, we present an innovative single-FPGA front-end that performs real-time 2D-FFT reconstruction from arrays of up to 16 coils. Partial reconfiguration enables rapid switching of FPGA modules for maximal flexibility and lower hardware cost. After an acquisition has been buffered, more complicated parallel MRI reconstruction techniques can replace receiver logic. Alternatively, the front- end can be used as a hardware-accelerated reconstruction engine with other receivers or for PCs. As proof-of-concept for real-time non-cartesian reconstruction, a reconfigurable module for next- neighbor regridding of spiral MRI is also demonstrated.

关键词： Magnetic resonance imaging Personal communication networks Concurrent computing Coils Field programmable gate arrays Hardware Costs Reconfigurable logic Engines Spirals

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A New Technique for the Estimation of Jitter and Shimmer of Voiced Speech signal

A New Technique for the Estimation of Jitter and Shimmer of ...

引用

Canadian Conference on electrical and computer engineering (CCECE)

作者： C. Shahnaz W.-p. Zhu M.O. Ahmad Centre for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

This paper presents a new technique for the estimation of jitter and shimmer exploiting the detected value of the pitch-period and amplitude of voiced speech signal degraded by white noise. Perturbations of the glottis cycle duration (jitter) and intonation (shimmer) are, respectively, the irregularities of pitch-period and energy. We propose a modified circular autocorrelation function (MCACF) of the normalized hamming-windowed pre-filtered speech. A weighted and pitch-harmonic MCACF is utilized to effectively extract pitch-period for jitter computation. To carry out shimmer analysis, peak-to-peak amplitude of signal around the estimated pitch-period is optimally obtained by matching the windowed pre-filtered speech frame with a shifted impulse train. For performance evaluation vowels, such as "a, e, i, o, u", are used from the TIMIT database. The conventional algorithms manifest not only the distortion of pitch contours but also provides elevated values of the jitter and shimmer that can be interpreted as the presence of anomalies of the vocal tract, the glottis or the neuro-mechanism. On the contrary, for both clean and noisy speech, the proposed method results in a comparatively smooth pitch contour as well as small values of the jitter and shimmer confirming the presence of a normal voice with good stability and periodicity

关键词： Jitter Amplitude estimation Speech analysis Speech enhancement Degradation White noise Autocorrelation signal analysis Databases Stability

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A multifeature voiced/unvoiced decision algorithm for noisy speech

A multifeature voiced/unvoiced decision algorithm for noisy ...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： C. Shahnaz W.-P. Zhu M.O. Ahmad Centre for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

This paper presents a new algorithm for the voiced/unvoiced (V/UV) decision of noise-corrupted speech. A speech periodicity-harmonic function (SPHF) is proposed to manifest distinctive characteristics between voiced and unvoiced regions. A composite feature vector is developed by combining a periodicity measure obtained from the SPHF with some energy measures such as zero-crossing rate-weighted RMS energy, Kaiser-Teager frame energy and the normalized low-frequency energy ratio. Unlike the conventional hard threshold, a signal-dependent initial-threshold (SDIT) for each feature is determined based on its statistical properties. The SDIT is exploited to develop a logical expression that returns an objective score regarding V/UV region. Additional voicing criteria are introduced to remove the artifacts that may exist due to the overlapping between decision regions. Simulation results of the proposed multifeature classification scheme, using the Keele reference database, show superior efficacy at a low SNR relative to some of the existing V/UV decision algorithms

关键词： signal processing algorithms Speech synthesis Speech analysis Pollution measurement Low-frequency noise Speech enhancement Energy measurement Gaussian noise Speech coding Autocorrelation

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Realization of 2D FIR filters using generalized polyphase structure combined with singular-value decomposition

Realization of 2D FIR filters using generalized polyphase st...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Wei-Ping Zhu Chao Wu M.N.S. Swamy Centre for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

In this paper, a realization scheme that combines the singular-value decomposition (SVD) and the generalized polyphase (GP) structure is proposed for 2D linear-phase FIR filters. With a small number of extra additions, a high-order 2D FIR filter is converted to several lower-order 2D subfilters. These subfilters are then realized using the SVD, yielding a parallel implementation structure for each 2D subfilter which consists of a number of 1D short-tap FIR filters. Due to the energy compaction of the SVD and the frequency-selective property of the GP structure, the number of parallel branches in each 2D subfilter is significantly reduced without introducing a large error. It is also shown that the various symmetries of 2D filters, such as the quadrantal symmetry and the central symmetry, are well preserved in the proposed GP-SVD structure

关键词： Finite impulse response filter signal processing Frequency Compaction Computational efficiency Image storage Chaotic communication Matrix converters Delay Two dimensional displays

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An efficient algorithm for the computation of the reverse jacket transform

An efficient algorithm for the computation of the reverse ja...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： S. Bouguezel M.O. Ahmad M.N.S. Swamy Center for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

This paper proposes an efficient algorithm to compute the reverse jacket transform algorithm by introducing a two-step decomposition strategy coupled with an appropriate use of the Kronecker product. Comparisons are carried out with the existing algorithms and the results show that a significant reduction in the number of data transfers and address generations as well as the structural complexity can be easily achieved using the proposed algorithm without increasing the arithmetic complexity. It is also shown that the three weights used in the existing reverse jacket transform are not required and just two are sufficient. Further, it is shown that a significant reduction in the number of multiplications can be achieved by using two rather than three weights, without losing the generality of the reverse jacket transform

关键词： signal processing algorithms Arithmetic Error correction Error correction codes Matrix decomposition Digital signal processing

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A blind identification technique for noisy ARMA systems

A blind identification technique for noisy ARMA systems

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： S.A. Fattah W.-P. Zhu M.O. Ahmad Centre for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

In this paper, a new model for the ramp-cepstrum of the one-sided autocorrelation function of a noise-free autoregressive moving average (ARMA) signal is presented. The proposed blind identification technique can estimate the parameters of ARMA systems in both noise-free and noisy environments without using the input observations. It is shown that, utilizing the proposed ARMA ramp-cepstrum model in accordance with a residue-based least-squares optimization technique, both AR and MA parameters of ARMA systems can be directly obtained. The proposed method is tested on synthetic ARMA systems of different orders and also on some natural speech signals. Simulation results demonstrate the efficacy of the proposed identification scheme at low to high SNR levels

关键词： Autoregressive processes Working environment noise signal processing Cepstral analysis Speech processing Parameter estimation Gaussian noise Poles and zeros Cepstrum Autocorrelation

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Noisy Autoregressive System Identification Based on Repeated Autocorrelation Function

Noisy Autoregressive System Identification Based on Repeated...

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Canadian Conference on electrical and computer engineering (CCECE)

作者： S. A. Fattah W.-p. Zhu M. O. Ahmad Centre for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

This paper presents an identification approach for the minimum-phase autoregressive (AR) systems in the presence of heavy noise based on a repeated autocorrelation function (RACF) of observed data. It is shown that RACF retains poles of the original system and in noisy environment if it is used instead of single ACF in the modified least-squares Yule-Walker equations the effect of additive noise can be reduced. A termination criterion for the repeated operations is proposed based on the decaying nature of correlation values. The length of ACF, which is kept fixed in all RACFs, is determined from the decorrelation time of the single ACF. Simulation results show the superiority of performance by the proposed method in comparison to some of the existing methods in estimating the AR parameters even at a very low SNR of -5 dB

关键词： System identification Autocorrelation Equations Additive noise Working environment noise Noise reduction Biomedical signal processing signal processing Decorrelation Termination of employment

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