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IEEE International Symposium on Circuits and Systems (ISCAS)

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

This paper addresses the problem of bit-rate conversion of a previously compressed video. We provide an MPEG joint transcoder for transcoding several video bitstreams simultaneously. We show that joint transcoding reduces the quality variation between multiple video sequences, as compared to independently transcoding each sequence at a fixed bit rate. Hence, joint transcoding results in a better utilization of the channel capacity. The joint transcoder can be used in a congested communication network as an alternative to data/packet dropping, and in applications which require multiplexing video signals onto a fixed communication channel such as video servers providing video on demand (VOD) service.

关键词： Transcoding Video compression Video on demand Transform coding Video sequences Bit rate Channel capacity Communication networks Communication channels Network servers

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Scheduling of DSP data flow graphs onto multiprocessors for maximum throughput

Scheduling of DSP data flow graphs onto multiprocessors for ...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： A. Shatnawi M.O. Ahmad M.N.S. Swamy Department of Computer Engineering Jordan University of Science and Technology Irbid Jordan Center for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

A novel technique to obtain a rate- and processor-optimal schedule for a fully-static data flow graph (DFG) onto a multiprocessor system is presented. In this technique, Floyd-Warshall's shortest path algorithm is used to evaluate the relative firing times of the nodes of the given DFG. Despite its implementation simplicity, the proposed technique has a lower time complexity than all the previously proposed techniques. The technique is tested on various benchmark problems to demonstrate its optimal performance.

关键词： Digital signal processing Flow graphs Throughput Processor scheduling signal processing algorithms Circuits Data flow computing Scheduling algorithm Polynomials Delay effects

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L/sub p/ norm design of stack filters

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IEEE Transactions on Image processing 1999年第12期8卷 1730-1743页

作者： C.E. Savin M.O. Ahmad M.N.S. Swamy Centre for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada Nortel Networks Wireless Technology Laboratories Nepean ONT Canada

Addresses the problem of designing optimal stack filters by employing an L/sub p/ norm of the error between the desired signal and the estimated one. It is shown that the L/sub p/ norm can be expressed as a linear function of the decision errors at the binary levels of the filter. Thus, an L/sub p/-optimal stack filter can be determined as the solution of a linear program. The conventional design of using the mean absolute error (MAE), therefore, becomes a special ease of the general L/sub p/ norm-based design developed here. Other special cases of the proposed approach, of particular interest in signal processing, are the problems of optimal mean square error (p=2) and minimax (p/spl rarr//spl infin/) stack filtering. Since an L/spl infin/ optimization is a combinatorial problem, with its complexity increasing faster than exponentially with the filter size, the proposed L/sub p/ norm approach to stack filter design offers an additional benefit of a sound mathematical framework to obtain a practical engineering approximation to the solution of the minimax optimization problem. The conventional MAE design of an important subclass of stack filters, the weighted order statistic filters, is also extended to the L/sub p/ norm-based design. By considering a typical application of restoring images corrupted with impulsive noise, several design examples are presented, to illustrate the performance of the L/sub p/-optimal stack filters with different values of p. Simulation results show that the L/sub p/-optimal stack filters with p/spl ges/2 provide a better performance in terms of their capability in removing impulsive noise, compared to that achieved by using the conventional minimum MAE stack filters.

关键词： Nonlinear filters Minimax techniques Design optimization Acoustic noise signal design signal processing Mean square error methods Filtering Acoustical engineering Design engineering

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Minimum distortion power polynomial model (MDP-PM) of nonlinear power amplifiers and its application on analog predistorters

Minimum distortion power polynomial model (MDP-PM) of nonlin...

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IEEE Conference on Vehicular Technology (VTC)

作者： H. Lai Y. Bar-Ness Center for Communications and Signal Processing Research Department of Electrical Engineering New Jersey Institute of Technology Newark NJ USA

With the increasing demands of power efficiency and linearity on power amplifiers (PAs), there are more and more demands for some new and easy modeling methods of PA nonlinearities. One direction of these trends is to get better polynomial models which are extensively used in the design of analog predistorters (PDs) to combat nonlinearity and in the estimation of the intermodulation products for PAs. Because of the difficulty in modeling a PA and for ease of comparison, most of the polynomial models reported in the literature are based on a simple TWTA model (Saleh's (1981) model) instead on actual PA devices. In this paper, we propose a method for obtaining a general order polynomial model from an existing model such as Saleh's model, or from the input-output statistical characteristics of a PA device. We also obtain the analytical representation of its coefficients. The method is based on minimizing the power of the distortion between the output of a PA and the output of the model based on it. Thus the model is named the minimum distortion power polynomial model (MDP-PM). We have obtained the coefficients of the proposed MDP-PMs up to order 7 based on Saleh's model to illustrate our results. Then, we show the application of this model in analog PD designs, whereas the resulted PD devices outperform those given in the literature by 0.8 dB for order 3 and by 0.2 dB for order 5.

关键词： Nonlinear distortion Polynomials Power amplifiers Power system modeling Filtering theory Least squares approximation signal processing Linearity High power amplifiers Ear

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A new approach for weighted least-square design of FIR filters

A new approach for weighted least-square design of FIR filte...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Wei-Ping Zhu M.O. Ahmad M.N.S. Swamy SR Telecom Inc. Kanata ONT Canada Center for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

In this paper, a new approach for weighted least square (WLS) design of FIR filters is presented, in which an iterative procedure is developed for the inversion of the matrix involved in the design. By imposing a mild constraint on the update factor of the weighting function, the inverse of a matrix is expanded as a convergent power series. By exploiting the properties of some of the matrices involved in the design, a modified version of the series that converges rapidly is then proposed to evaluate the inverse in each iteration. It is shown that due to the fast convergence of the power series, one has to evaluate only the first two or three terms of the series after a few initial iterations, implying that the conventional operation for matrix inversion is simplified significantly.

关键词： Finite impulse response filter Frequency response signal processing Electronic mail signal design Convergence Design methodology Strontium Telecommunication computing Design engineering

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Round-off error free fixed-point design of polynomial FIR predictors

Round-off error free fixed-point design of polynomial FIR pr...

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Asilomar Conference on signals, Systems & computers

作者： J.M.A. Tanskanen V.S. Dimitrov Institute of Intelligent Power Electronics Department of Electrical and Communications Engineering Helsinki University of Technology Espoo Finland Laboratory of Signal Processing and Computer Technology Department of Electrical and Communications Engineering Helsinki University of Technology Espoo Finland

We present a novel method for designing polynomial FIR predictors for fixed-point environments. Our method yields filters that perform exact prediction of polynomial signals even with short coefficient word lengths. Under ordinary coefficient truncation or rounding, the prediction capability degrades, or may be totally lost. With the proposed method, the filters are designed so that the predictive properties are exactly preserved in fixed-point implementations. The proposed filter design method is based on integer programming (IP) and can be directly applied to any fixed-point FIR design specifications which can be formulated in a form of linear constraints on the filter coefficients.

关键词： Roundoff errors Polynomials Finite impulse response filter Design methodology Design engineering Power engineering and energy Degradation Nonlinear filters Filtering White noise

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Wavelet-based orthogonal modulation code

Wavelet-based orthogonal modulation code

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Asilomar Conference on signals, Systems & computers

作者： E.-J. Yi E.J. Powers Department of Electrical and Computer Engineering and Telecommunication and Signal Processing Research Center University of Texas Austin Austin TX USA

We suggest a new wavelet-based orthogonal modulation code set for multiple access, using various orthogonal scale functions and mother wavelets. Specifically we compare the bandwidth of these new wavelet-based orthogonal modulation codes with that of traditional orthogonal codes based on the Hadamard matrix and impulse function. The bandwidths of these wavelet-based modulation codes strongly depend on the choice of wavelets and scale functions which are used to generate these new codes. Generally speaking, we find that the bandwidths of these new modulation codes are considerably smaller than those of the orthogonal code set based on the Hadamard matrix and impulse function.

关键词： Modulation coding Bandwidth Error correction Error correction codes Discrete wavelet transforms Wavelet analysis Image coding signal processing Digital modulation Wavelet transforms

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A switched-current ratio-independent algorithmic D/A converter

A switched-current ratio-independent algorithmic D/A convert...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： S. Wang M. Omair Ahmad Philips Research Philips Electronics North America Corporation NY USA Center for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

A design of switched-current ratio-independent algorithmic digital-to-analog converter is described. The converter is capable of achieving high-accuracy and high-resolution data conversion without requiring any matching of components. Since the converter operates in current mode, it does not require linear capacitors, and it is capable of working with low power supply voltages. An experimental prototype D/A converter is designed and fabricated with a standard 1.2 /spl mu/m CMOS process technology. The test results show that the D/A converter achieves an 8-bit data conversion accuracy with speed of 10 KSPS.

关键词： Switching converters Data conversion Algorithm design and analysis Digital-analog conversion Capacitors Power supplies Low voltage Prototypes CMOS process CMOS technology

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Maximizing channel capacity in FSK modulation systems

Maximizing channel capacity in FSK modulation systems

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： M.A. Soderstrand L. Gao E. McCune School of Electrical and Computer Engr. Oklahoma State University Stillwater OK USA Digital Signal Processing and Communications Lab Electrical and Computer Engineering Department University of California Davis CA USA

In binary FSK modulation schemes the transition between the two frequencies f/sub 0/ representing a ZERO and f/sub 1/ representing a ONE is governed by the shape of the input pulse as it makes its transition from a ZERO to a ONE. Through the use of an optimum filter, it is possible to shape the pulse in such a way as to maximize the speed of transition from f/sub 0/ to f/sub 1/ while maintaining signal energy outside of the channel to specified small levels. In this paper, we describe a procedure for finding optimum filters for several important types of FSK modulation.

关键词： Channel capacity Frequency shift keying Finite impulse response filter Radio transmitters Filtering Interference Pulse modulation Pulse shaping methods Telephony Digital signal processing

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Temporal integration of loudness in listeners with sloping sensorineural hearing losses

Temporal integration of loudness in listeners with sloping s...

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Annual Northeast Bioengineering Conference

作者： S. Buus Communication and Digital Signal Processing Center Department of Electrical and Computer Engineering (409DA) Northeastern University Boston MA USA

Temporal integration of loudness, as measured by the level difference between equally loud short and long tones, can be related directly to the loudness function if one assumes that ratio between equal-SPL long and brief tones is the same at all SPLs. This assumption is supported by measurements of temporal integration of loudness in listeners with gradual and abrupt high-frequency hearing losses. The amounts of temporal integration obtained in listeners with abrupt losses are larger than those obtained in listeners with gradual losses and in listeners with normal hearing. This finding is consistent with the slow growth of loudness with increasing intensity observed in listeners with abrupt losses.

关键词： Auditory system Deafness Loss measurement Testing Frequency Hair Digital signal processing Electric variables measurement Steady-state Feedback

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