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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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Predistorter design employing parallel piecewise linear structure and inverse coordinate mapping for broadband communications

Predistorter design employing parallel piecewise linear stru...

引用

14th European signal processing Conference, EUSIPCO 2006

作者： Cheong, Mei Yen Werner, Stefan Laakso, Timo I. Cousseau, Juan Figueroa, Jose L. Signal Processing Laboratory Center of Excellence Helsinki University of Technology Finland CONICET Department of Electrical Engineering and Computer Science Universidad Nacional Del sur Bahía Blanca Argentina

This paper proposes a low-complexity predistorter (PD) for compensation of both the AM/AM and the AM/PM conversions with memory. The nonlinear power amplifier (PA) is modeled as a Wiener type nonlinearity. The quasi-static nonlinearities are modeled using a class of piecewise linear (PWL) functions. The PWL function facilitates an efficient PD identification algorithm. The proposed algorithm involves a novel inverse coordinate mapping (ICM) method that maps the nonlinear characteristics of the PA to that of the PD, and parameter estimations that do not require matrix inversion. The indirect learning architecture is used to provide an on-line compensation of thememory effect of the PA. Simulation results show that the PD that compensates also the AM/PM distortion performs significantly better than one that considers only the AM/AM nonlinearity. The proposed PD is also shown to outperform the orthogonal polynomial PD in both adjacent channel interference suppression and inband distortion compensation.

关键词： Power amplifiers

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SPC10-4: EM Algorithm for Multiple Wideband Source Localization

SPC10-4: EM Algorithm for Multiple Wideband Source Localizat...

引用

IEEE Globecom 2006

作者： Kiran K. Mada Hsiao-Chun Wu Communications and Signal Processing Laboratory Department of Electrical and Computer Engineering Louisiana State University USA

A computationally efficient algorithm for multiple source localization, using the expectation-maximization (EM) algorithm, for the wideband sources in the near field of a sensor array/area, is presented. The basic idea is to decompose the observed sensor data, which is a superimposition of multiple sources, into individual components in the frequency domain and then estimate the corresponding location parameters associated with each component separately. Instead of the conventional alternating projection method, we propose to adopt the EM algorithm in this paper; our method involves two steps, namely Expectation (E-step) and Maximization (M-step). In the E-step, the individual incident source waveforms are estimated. Then, in the M-step, the maximum likelihood estimates of the source location parameters are obtained. These two steps are executed iteratively and alternatively until the pre-defined convergence is reached. The computational complexity comparison between our proposed EM algorithm and the existing alternating projection scheme is investigated. It is shown through Monte Carlo simulations that the computational complexity of the proposed EM algorithm is significantly lower than that of the conventional alternating projection algorithm.

关键词： Wideband Sensor arrays Iterative algorithms Computational complexity Frequency domain analysis Frequency estimation Maximum likelihood estimation Position measurement Convergence Projection algorithms

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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 ...

引用

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...

引用

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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A blind identification technique for noisy ARMA systems

A blind identification technique for noisy ARMA systems

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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...

引用

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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WLC43-4: Comparing the effect of Carrier Frequency Offset on OFDM and Single-Carrier Block Transmission in AWGN Channels

WLC43-4: Comparing the effect of Carrier Frequency Offset on...

引用

IEEE Globecom 2006

作者： Pan Liu Yeheskel Bar-Ness Center for Wireless Communications and Signal Processing Research Department of Electrical and Computer Engineering New Jersey Institute of Technology Newark NJ USA

Both OFDM and single-carrier block transmission (SCBT) are promising candidates for the future high date rate communication, which well handle the frequency selective channels with a low implementation complexity. However, the sensitivity to carrier frequency offset, caused by Doppler effect or the carrier frequency difference between transmitter and receiver, is always a major concern of these block transmission systems. Actually, a controversy in comparing the carrier frequency offset effect on OFDM and SCBT systems has already been witnessed in the literature for a long time. In this paper, we first analyze the effects of carrier frequency offset on these two systems, then by carefully investigating existing works, we classify the problem into two categories: carrier frequency offset analysis with compensation and without compensation. Finally, we draw conclusions based on theoretical and numerical results that the BER performance of OFDM systems is more sensitive to carrier frequency offset than SCBT systems for the BPSK modulation in AWGN channels, although the difference of the BER performance between two systems is much smaller than theoretically predicted in previous works, if a fair comparison environment is used. Moreover, this small performance gap vanishes in higher level constellation like QPSK.

关键词： AWGN channels OFDM modulation Bit error rate Frequency estimation Binary phase shift keying Quadrature phase shift keying DSL Digital video broadcasting Frequency conversion Wireless communication

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On the Sum-Rate of Broadcast Channels with Outdated 1-Bit Feedback

On the Sum-Rate of Broadcast Channels with Outdated 1-Bit Fe...

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Asilomar Conference on signals, Systems & computers

作者： Bo Niu Osvaldo Simeone Oren Somekh Alexander M. Haimovich Center for Wireless Communications and Signal Processing Research Department of Electrical and Computer Engineering New Jersey Institute of Technology Newark NJ USA

In this paper, a single-input single-output (SISO) downlink channel with K users is analyzed in the presence of Rayleigh flat fading. A limited channel state information (CSI) feedback scheme is considered, where only an outdated 1-bit feedback per user is available at the base station for each fading block. A closed-form expression for the achievable ergodic sum- rate of the 1-bit feedback scheme is presented for any number of users, as a function of the fading temporal correlation coefficient, the threshold of the 1-bit CSI quantizer and the SNR. It is proved that the sum-rate scales with increasing number of users as log log K, which is the same scaling law achieved by the optimal non-delayed full CSI feedback scheme. In addition, the sum-rate degradation due to outdated CSI is evaluated in the asymptotic regimes of either large K or low SNR.

关键词： Broadcasting Feedback Fading Base stations Delay Rayleigh channels Downlink Degradation Feeds Optimal scheduling

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