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Complexity scalable motion compensated wavelet video encoding

IEEE Transactions on Circuits and Systems for Video Technology 2005年第8期15卷 982-992页

作者： Turaga, Deepak S. Van Der Schaar, Mihaela Pesquet-Popescu, Beatrice Wireless Communications and Networking Philips Research USA Briarcliff Manor NY 10510 United States IBM T.J. Watson Research Center Hawthorne NY 10532 United States Department of Electrical and Computer Engineering University of California Davis Davis CA 95616-5294 United States Signal and Image Processing Department Télécom Paris 75634 Paris Cedex 13 France Media Delivery Architectures Department IBM T.J. Watson Research Center Hawthorne NY Electrical and Computer Engineering Department University of California Davis

We present a framework for the systematic analysis of video encoding complexity, measured in terms of the number of motion estimation (ME) computations, that we illustrate on motion compensated wavelet video coding schemes. We demonstrate the graceful complexity scalability of these schemes through the modification of the spatiotemporal decomposition structure and the ME parameters, and the use of spatiotemporal prediction. We generate a wide range of rate-distortion-complexity (R-D-C) operating points for different sequences, by modifying these options. Using our analytical framework we derive closed form expressions for the number of ME computations for these different coding modes and show that they accurately capture the computational complexity independent of the underlying content characteristics. Our frame-work for complexity analysis can be combined with rate-distortion modeling to determine the encoding structure and parameters for optimal R-D-C tradeoffs. © 2005 IEEE.

关键词： Image coding

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A VLSI architecture for a high-speed computation of the 1D discrete wavelet transform

A VLSI architecture for a high-speed computation of the 1D d...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Chengjun Zhang Chunyan Wang M.O. Ahmad Center for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University West Montreal QUE Canada

An efficient VLSI architecture for the computation of the convolution-based discrete wavelet transform (DWT) is presented. The proposed architecture, employing two processing elements and a single buffer in a pipeline mode, enhances the processing time by appropriately decomposing the overall computations and distributing them equally between the two processing elements. The data flow, both within and between the processing elements, is streamlined, making use of the buffer and employing multiple input data paths within the processing elements. The parallelism of operations carried out by the computational blocks in each processing element is made more effective by equalizing the data paths used in these blocks. HSPICE and Verilog simulation results are presented to show that a circuit, whose design is based on the proposed architecture, is, in comparison with other existing architectures, fast and efficient for DWT computation, with a modest decrease in the area.

关键词： Discrete wavelet transforms Very large scale integration computer architecture Pipelines Distributed computing Parallel processing Concurrent computing Hardware design languages Computational modeling Circuit simulation

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A robust mobile speed estimator in fading channels: performance analysis and experimental results

A robust mobile speed estimator in fading channels: performa...

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IEEE Conference and Exhibition on Global Telecommunications (GLOBECOM)

作者： Hong Zhang A. Abdi Center for Communications and Signal Processing Research Department of Electrical and Computer Engineering New Jersey Institute of Technology Newark NJ USA

In this paper we propose a novel speed estimation technique, based on the characteristics of the power spectrum of mobile fading channels. Our analytic performance analysis, verified by Monte Carlo simulations, shows that the simple yet low-complexity estimator is not only robust to both Gaussian and non-Gaussian noises, but also insensitive to nonisotropic scattering observed in practical situations at the mobile. The estimator performs very well in both two- and three-dimensional propagation environments, where the latter case has not been studied so far, to the best of our knowledge. The utility of the new method is further demonstrated by applying it to the measured data

关键词： Fading Performance analysis Gaussian noise Frequency Working environment noise Additive noise Mobile communication signal processing Noise robustness Gaussian processes

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A frequency domain sphericity test based approach for parsimonious rake receivers in CDMA systems

A frequency domain sphericity test based approach for parsim...

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8th International Symposium on signal processing and its Applications, ISSPA 2005

作者： El-Sallam, Amar Abd Leung, Yee H. Zoubir, Abdelhak M. Communications and Signal Processing Group Department of Electrical and Computer Engineering Curtin University of Technology GPO Box U1987 Perth WA 6845 Australia Signal Theory Group Institute for Communication Technology Darmstadt University of Technology Merckstrasse 25 D-64283 Germany

ISBN: (纸本)0780392434

A frequency domain approach is presented for parsimonious channel estimation which can result in the implementation of a low complexity DS-CDMA RAKE receiver. We consider frequency selective slowly fading channels. A new approach is proposed to identify and estiamte only significant channel parameters. The approach uses the sphericity test and delay estimation methods based on averaged periodogram techniques. Simulations show that, with short training sequences and low signal to noise ratio, the proposed technique can correctly identify the significant channel parameters. Also the parsimonious RAKE receiver based on these estimates gives improved statistical as well as computational performance over classical methods. © 2005 IEEE.

关键词： Frequency domain analysis

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Distributed signature sequence adaptation via pricing for downlink of MC-CDMA system

Distributed signature sequence adaptation via pricing for do...

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IEEE Workshop on signal processing Advances in Wireless communications (SPAWC)

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

A distributed signature sequence adaptation scheme is considered for the downlink of MC-CDMA system, wherein each mobile user selfishly adapts its signature sequences to improve its own performance objective with no consideration to the other users in the system. Pricing is introduced to each user's performance objective to make that user notice the power it consumes. It then serves as a control signal to motivate users to efficiently share the limited transmission power at base station, even thought the adaptation of users signature sequences is independent. It is shown that the proposed adaptation scheme will converge to a set of signature sequences at which no user can further improve its own performance with unilateral adaptation. Simulation result indicates that a simple linear pricing function with a properly chosen unit price can significantly improve the system performance.

关键词： Pricing Downlink Multicarrier code division multiple access Base stations System performance Channel state information Multiple access interference Mobile communication Multiaccess communication Distributed algorithms

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Robust pitch estimation at very low SNR exploiting time and frequency domain cues

Robust pitch estimation at very low SNR exploiting time and ...

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

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

In this paper, we present a joint time/frequency domain approach for pitch estimation of speech at a very low SNR. The kernel of this approach lies in introducing a new function for detecting the time-domain cue by modifying the circular average magnitude difference function (CAMDF). By using the new function in conjunction with the half-wave rectified version of the autocorrelation function, the pitch-peak can be emphasized and the non-pitch peaks suppressed. To guarantee a robust pitch detection in noisy speech, a priori frequency-domain estimate of the dominant pitch-harmonic is extracted as an additional cue and is utilized to optimally match the pitch-peak in time-domain. The proposed approach is simulated using the Keele reference database. It is shown that the proposed method using joint time and frequency domain cues is able to give a superior accuracy relative to some of the existing methods even at a very low SNR of -10 dB.

关键词： Robustness Frequency estimation Frequency domain analysis Time domain analysis Autocorrelation Working environment noise signal processing Gaussian noise Speech enhancement Speech processing

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A preamble-aided symbol and frequency synchronization scheme for OFDM systems

A preamble-aided symbol and frequency synchronization scheme...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Meng Wu Wei-Ping Zhu Centre for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

ISBN: (纸本)0780388348

Orthogonal frequency division multiplexing is considered to be an appealing modulation technique for high-speed wireless communication systems owing to its spectral efficiency and robustness against multipath fading. However, an OFDM system is very sensitive to synchronization and a small frequency or timing offset may cause a large intercarrier interference (ICI), leading to a dramatic degradation in system performance. The paper presents an efficient symbol and frequency synchronization scheme for OFDM systems that is compatible with the IEEE 802.16a standard. A timing algorithm is proposed based on the preamble defined in IEEE 802.16a. The conjugate symmetry of the preamble structure is exploited to reduce computational complexity. It is shown that the new scheme not only reduces the estimation error significantly, but also allows for independent symbol synchronization compared to the conventional methods where fine timing usually has to be done after the frequency offset correction.

关键词： Frequency synchronization Timing OFDM modulation Wireless communication Robustness Fading Interference Degradation System performance Computational complexity

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A subspace constrained constant modulus algorithm for blind array

A subspace constrained constant modulus algorithm for blind ...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Y.X. Chen W.-P. Zhu M.N.S. Swamy Centre for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

It is known that the stochastic gradient descent (SGD)-based constant modulus algorithm (CMA) has the drawback of slow convergence rate. It is also known that under some weak conditions, all local minima of CMA are identical to that of a CMA which is constrained in signal subspace. Based on this property, we propose a subspace-constrained CMA that is able to increase the convergence rate of the conventional SGD-CMA. To reduce the computational complexity, a technique referred to as projection approximate subspace tracking with deflation (PASTd) is used to calculate the signal subspace. Our simulation shows that the proposed algorithm is significantly superior to the conventional SGD-CMA both in the convergence rate and in the sensitivity to the step size.

关键词： Subspace constraints signal processing algorithms Array signal processing Cost function Stochastic processes Convergence Computational modeling Covariance matrix Estimation error Directive antennas

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An approach to ARMA system identification at a very low signal-to-noise ratio

An approach to ARMA system identification at a very low sign...

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

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

A new approach for the identification of minimum-phase autoregressive moving average (ARMA) systems in the presence of heavy noise is presented in this paper. A damped sinusoidal (DS) model for the autocorrelation function of a noise-free ARMA signal is proposed to estimate the AR parameters, which overcomes the failure of conventional correlation based techniques in estimating the AR parameters of an ARMA system at a very low signal-to-noise ratio (SNR). The MA parameters of the ARMA system are then estimated by using Durbin's method along with an optimum order selection criterion. Both white noise and periodic impulse train excitations are considered for the application of the proposed method to system identification as well as to speech processing. computer simulations are carried out based on both synthetic ARMA systems and natural speech signals, showing superior identification results even at an SNR of -5 dB for which most of the existing methods would fail.

关键词： System identification signal to noise ratio Parameter estimation Autoregressive processes Autocorrelation White noise Application software Speech processing computer simulation Natural languages

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A robust pitch estimation approach for colored noise-corrupted speech

A robust pitch estimation approach for colored noise-corrupt...

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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

We present an integrated pitch estimation approach for severely colored noise-corrupted speech. An effective colored noise-whitening process is first applied to the noisy speech. Then, a variable-length average magnitude difference function (VLAMDF) of the pre-filtered noisy speech (PFNS) is proposed, which almost conquers the trend of falling valleys in the conventional AMDF. The amplitude characteristic of the VLAMDF is reshaped by means of a simple linear transformation to reduce the possibility of double-pitch-errors. As the VLAMDF exhibits a valley while the autocorrelation function (ACF) of PFNS provides a peak, the ACF is weighted by the reciprocal of the VLAMDF to emphasize the pitch-candidate as well as to suppress the non-pitch peaks. Moreover, a noise-robust pitch detection in the time-domain is guaranteed by collaboration of this enhanced autocorrelation function with the reshaped version of the VLAMDF. The proposed approach is simulated using the Keele reference database and provides a superior accuracy relative to some of the existing methods implemented in the presence of colored noise, even at a very low signal-to noise ratio (SNR) of -15 dB.

关键词： Noise robustness Colored noise Autocorrelation Speech processing signal to noise ratio Speech coding Speech synthesis Working environment noise Additive noise signal processing

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