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Estimating Laplacian parameters of DCT coefficients for requantization in the transcoding of MPEG-2 video

Estimating Laplacian parameters of DCT coefficients for requ...

引用

IEEE International Conference on Image processing

作者： H. Sorial W.E. Lynch A. Vincent Centre for Signal Processing and Communications Department of Electrical & Computer Engineering Concordia University Montreal QUE Canada Communications Rcscarch Centre Ottawa ONT Canada

ISBN: (纸本)0780362977

This paper addresses the bit-rate reduction of MPEG-2 compressed video and presents a method to reduce requantization errors in transcoding. The proposed method assumes Laplacian distributions for the original AC coefficients of the DCT. A Laplacian parameter for each coefficient is estimated at the transcoder from the quantized input DCT coefficients. These parameters are then used in requantization to improve the quality of the transcoded video. The algorithm provided in this paper to estimate the Laplacian parameters of the original DCT coefficients is simple to implement and may be adapted to other DCT-based coding schemes.

关键词： Parameter estimation Laplace equations Discrete cosine transforms Transcoding Video compression Transform coding Bit rate signal processing algorithms Delay Cost function

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Match filter bound analysis used to evaluate range increase provided by an adaptive antenna array

Match filter bound analysis used to evaluate range increase ...

引用

IEEE Conference on Vehicular Technology (VTC)

作者： I. Howitt Wireless & Signal Processing Laboratory Electrical Engineering & Computer Science Department University of Wisconsin Milwaukee WI USA

An analytical approach is developed for evaluating the expected range increase and variations in coverage range, while using an M-element adaptive antenna array at the base station (BS). The analysis is based on the match filter bound an the bit error rate probability for a frequency non-selective Rayleigh fading channel. Results for a ten element uniform linear array (ULA) indicate the expected range increase over a BS with only a single antenna can be nearly doubled. A 1.6 to 1.7 increase is also likely with a five-element array.

关键词： Adaptive arrays Adaptive filters Matched filters Antenna arrays Linear antenna arrays Base stations Frequency Fading Wireless LAN Diversity methods

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Capacity enhancement based on using adaptive antenna arrays optimized for trunking efficiency

Capacity enhancement based on using adaptive antenna arrays ...

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IEEE Conference on Vehicular Technology (VTC)

作者： I. Howitt F.H. Awad Wireless & Signal Processing Laboratory Electrical Engineering & Computer Science Department University of Wisconsin Milwaukee WI USA

Cellular systems using adaptive antennas for spatial processing have been shown to provide an increase in capacity. When employing adaptive antennas, the standard approach to achieve capacity gain has been to maximize the link quality between the mobile and base station via optimum combining. This paper presents an alternative optimization approach based on optimizing the adaptive antenna array to maximize the trunking efficiency, of which the effect was subsequently an outstanding issue. Monte Carlo simulations, substantiated with theoretical analysis, were used to evaluate this issue. Based on the analysis, the proposed approach could increase the capacity by 2 to 4 times.

关键词： Adaptive arrays Antenna arrays Traffic control Phased arrays Telecommunication traffic Mobile antennas Base stations Analytical models Speech analysis Loaded antennas

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Parallel implementation of multifilters

Parallel implementation of multifilters

引用

International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： N. Damera-Venkata B.L. Evans Embedded Signal Processing Laboratory Department of Electrical and Computer Engineering University of Texas Austin Austin TX USA

A multifilter is a filter with matrix-valued coefficients, and is used in the processing of vector-valued signals, e.g. color images. Convolution becomes a vector sum of matrix-vector multiplication. In this paper, we efficiently implement a multifilter as a parallel combination of scalar filters. Each scalar filter works on one component of the input vector signal, which increases processing speed by the dimension of the vector-valued signal. This means that by using N processors, the throughput is increased by a factor of N while the total memory usage remains unchanged. We also present a frequency-domain analysis of the filtering.

关键词： Filter bank signal processing Digital signal processing Convolution Color Concurrent computing Laboratories Embedded computing Frequency domain analysis Filtering

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Comparison of techniques for estimating the frequency selectivity of bandlimited channels

Comparison of techniques for estimating the frequency select...

引用

IEEE Region 10 International Conference TENCON

作者： A.Z. Sha'ameri Digital Signal Processing Laboratory Department of Microelectronics and Computer Engineering Faculty of Electrical Engineering Universiti Teknologi Malaysia Skudai Johor Malaysia

A transmission channel used in application such as telecommunications can be modeled as a bandpass filter. Measurement of the frequency selectivity of the channel is important to ensure that the information-bearing signal has minimal distortion and loss of information. A comparison is made for several methods used for estimating the frequency selectivity of the transmission. The methods presented are the correlation method, instantaneous energy and frequency estimation and the cross Wigner-Ville distribution. The theoretical foundations and assumptions are described for each method. In general, all the methods gave similar performance in terms of the frequency selectivity. Due to the shorter analysis duration, both the instantaneous energy and frequency estimation and cross Wigner-Ville distribution is ideal for estimating the frequency selectivity of time-varying channels.

关键词： Frequency estimation Correlation Autocorrelation Linear systems Finite impulse response filter Nonlinear filters Band pass filters Binary sequences Linear feedback shift registers Chebyshev approximation

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Blind adaptive asynchronous CDMA multiuser detector using prediction least mean kurtosis algorithm

Blind adaptive asynchronous CDMA multiuser detector using pr...

引用

IEEE Workshop on Statistical signal and Array processing

作者： Kunjie Wang Y. Bar-Ness Center for Communications and Signal Processing Research Department of Electrical and Computer Engineering New Jersey Institute of Technology University Heights Newark NJ USA

ISBN: (纸本)0780359887

In this paper, a new blind adaptive multiuser detector, which is termed the prediction least mean kurtosis (PLMK) algorithm, is proposed for joint MAI and narrowband interference (NBI) suppression in asynchronous CDMA systems. This algorithm is based on higher-order statistics rather than the second-order statistics used in the LMS algorithm. Unlike the regular least mean kurtosis (LMK), it takes into consideration samples earlier than those corresponding to the current bit. For comparison purposes, we also apply the regular LMK algorithm to the case of asynchronous CDMA systems. Simulation results show that the blind adaptive multiuser detector with PLMK algorithm provides significantly better performance than the one with regular LMK algorithm.

关键词： Multiaccess communication Detectors signal processing algorithms Higher order statistics Interference suppression Least squares approximation Filters Adaptive signal detection Narrowband Multiple access interference

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Systolic implementation of the steepest descent algorithm and its performance in CDMA receivers

Systolic implementation of the steepest descent algorithm an...

引用

European signal processing Conference (EUSIPCO)

作者： R. Baghaie Signal Processing Laboratory Department of Electrical and Communications Engineering Helsinki University of Technology P.O. BOX 3000 02015 HUT Finland

In code division multiple access receivers, in order to suppress the multiple access interference, different multiuser detectors such as the decorrelating and the linear minimum mean square error detectors can be utilized. In order to alleviate the implementation complexity, iterative implementation of such detectors have been reported. Two most popular iterative algorithms used in such detectors are the Steepest Descent (SD) and the Conjugate Gradient (CG) algorithms. In this paper, we first compare fixed-point and floating-point performances of these algorithms. Furthermore, hardware implementations of the algorithms are compared. Based on these comparisons, for the implementation of the SD algorithm a systolic architecture is proposed.

关键词： Detectors Arrays signal processing algorithms Multiaccess communication Very large scale integration Iterative methods

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Implementation of real-valued discrete transforms via encoding algebraic integers

Implementation of real-valued discrete transforms via encodi...

引用

European signal processing Conference (EUSIPCO)

作者： R. Baghaie V. Dimitrov Signal Processing Laboratory Department of Electrical and Communications Engineering Helsinki University of Technology P.O. BOX 3000 02015 HUT Finland

In this paper, we propose a novel approach for computing real-valued discrete transforms such as the discrete cosine transform and the discrete Hartley transform. The approach is based on the algebraic integer encoding scheme. With the aid of this scheme, an error-free representation of the cos, sin and cas functions becomes possible. Furthermore, for the implementation of these algorithms a fully pipelined systolic architecture with O(N) throughput is proposed.

关键词： Discrete cosine transforms Arrays signal processing algorithms Discrete Fourier transforms

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Moving object segmentation based on human visual sensitivity 1st

引用

1st IEEE International Workshop on Biologically Motivated computer Vision, BMCV 2000

作者： Yoon, Kuk-Jin Kweon, In-So Kim, Chang-Yeong Seo, Yang-Seok Department of Electrical Engineering and Computer Science KAIST 373-1 Kusong-dong Yusong-ku Taejon305-701 Korea Republic of Signal Processing Laboratory Samsung Advanced Institute of Technology P.O. Box 111 Suwon440-660 Korea Republic of

ISBN: (纸本)3540675604

The edge and motion are the main features that human visual system (HVS) perceives intensively. This paper proposes an algorithm for the segmentation of the moving object with accurate boundary using color and motion focusing on the HVS perception in the general image sequence. The proposed algorithm is composed of three parts: color segmentation, motion analysis, and region refinement and merging part. In the color segmentation phase, K-Means algorithm is used in consideration of the sensitivity of the human color perception to get the boundaries that HVS perceives. The global and local motion estimation are performed in parallel with color analysis. After that, Bayesian clustering using color and motion provides more accurate boundary. In the final stage, regions are merged taking into account their motion. The experimental results of the proposed algorithm show the accurate moving object boundary coinciding with the boundary that HVS perceives. © Springer-Verlag Berlin Heidelberg 2000.

关键词： Color

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A new adaptive algorithm for stereophonic acoustic echo canceller

A new adaptive algorithm for stereophonic acoustic echo canc...

引用

International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： Yang-Won Jung Ji-Ha Lee Young-Cheol Park Dae-Hee Youn ASSP Laboratory Department of Electrical and Computer Engineering Yonsei University South Korea Center for Signal Processing Research Yonsei University South Korea

Stereophonic sound becomes more important in a growing number of hands-free applications where spatial realism is demanded. Such hands-free systems need stereophonic acoustic echo cancellers to reduce echoes that result from coupling between loudspeakers and microphones. In this paper, we propose a new adaptive algorithm for an stereophonic acoustic echo canceller based on human auditory properties and affine projection (AP) algorithm. The proposed algorithm employs a pre-processor generating speech-like noise to decorrelate the input signals without degrading perceptual speech quality. The decorrelation is based on the masking patterns of the human auditory system. By showing that the AP algorithm can be represented by a vector update as a combination of the Gram-Schmidt (GS) orthogonalization followed by the normalized LMS (NLMS) algorithm, the proposed adaptive algorithm integrates the pre-processor into the adaptive algorithm. Subjective listening test and computer simulation show the effectiveness of the proposed algorithm.

关键词： Adaptive algorithm Echo cancellers Humans Decorrelation Loudspeakers Microphones signal generators Acoustic noise Noise cancellation Noise generators

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