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

作者： T.I. Laakso V. Valimaki Lab. of Telecommun. Technol. Helsinki Univ. of Technol. Espoo Finland Laboratory of Acoustics and Audio Signal Processing Helsinki University of Technology Finland

A measure for the effective length of the impulse response of a stable recursive digital filter based on accumulated energy is proposed. A general definition and a simple algorithm for its evaluation are introduced, and closed-form expressions are derived for first-order IIR filters. The effect of zeros on the effective length is analyzed. An upper bound for the effective length of higher-order filters is derived using results for low-order filters. The new measure finds applications in several fields of digital signal processing, including estimation of the extent of attack transients for filters with dynamically varying inputs, elimination of transients in variable recursive filters, and design and implementation of linear-phase IIR systems.

关键词： IIR filters Digital filters Energy measurement Length measurement signal processing algorithms Closed-form solution Upper bound Digital signal processing Recursive estimation signal design

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A highly-scaleable FIR using the radix-4 Booth algorithm

A highly-scaleable FIR using the radix-4 Booth algorithm

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

作者： O.T.-C. Chen Wei-Lung Liu Hsun-Chang Hsieh Jeng-Yih Wang Signal and Media Laboratories Department of Electrical Engineering National Chung Cheng University Chiayi Taiwan Digital Signal Processing Department Computer & Communication Research Lab Industrial Technology and Research Institute Hsinchu Taiwan

A highly-scaleable FIR architecture based on the radix-4 Booth algorithm has been designed with scaleable dynamic ranges of input data and filter coefficients. The radix-4 Booth algorithm is demonstrated to have a lower hardware complexity and a fairer throughput rate than the other radix approaches. In order to achieve scaleability, the configurable-connection function between latches of input data, and filter taps has been explored. The precision of filter coefficients is adjustable by using a path-control function. Especially, the proposed architecture only employs data-path controls to realize the scaleable issue without changing the word lengths and components of input latches and filter taps. The pre-processing unit for manipulating input data and post-processing unit for computing accumulation results have been realized to support scaleable operations. Based on our architecture in a chip design, the cascaded configuration between chips is also easily accomplished for many industrial applications.

关键词： Finite impulse response filter signal processing algorithms Dynamic range computer architecture Algorithm design and analysis Hardware Chip scale packaging Costs lab.ratories Throughput

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On phase approximation by gain differences

On phase approximation by gain differences

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IEEE International Conference on Electronics, Circuits and Systems (ICECS)

作者： C. Rusu I. Gavrea P. Kuosmanen Signal Process. Lab. Tampere Univ. of Technol. Finland Department of Mathematics Technical University of Cluj Napoca Cluj-Napoca Romania Signal Processing Laboratory Tampere University of Technology Tampere Finland

The goal of this paper is to establish new relationships for computing the phase of the minimum-phase functions from the gain values. Firstly, we show that for a given frequency the phase could be obtained from the odd derivatives of the neperian gain, evaluated for this frequency. Then we select a finite number of terms of the main formula and we derive an approximation of phase. We show the latter can be improved by taking into account the Gibbs phenomenon and the Feher kernel. Finally, we utilize finite differences in order to substitute the higher derivatives involved in the previous formulae.

关键词： Frequency signal processing lab.ratories Integral equations Mathematics Kernel Finite difference methods Image reconstruction Fourier transforms Filters

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An Algorithm-Hardware-System Approach to VLIW Multimedia Processors

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Journal of VLSI signal processing Systems for signal, Image, and Video technology 1998年第1-2期20卷 163-180页

作者： Berekovic, Mladen Pirsch, Peter Kneip, Johannes Lab. F. Informationstechnologie Universität Hannover D-30167 Hannover Schneiderberg 32 Germany Robert Bosch GmbH Hildesheim Germany University of Hannover Germany Lab. for Information Technology University of Hannover Germany Engineering College in Hannover Hannover Germany University of Hannover Germany Telefunken Hannover Germany Television Department Department of Electrical Engineering University of Hannover Germany Vis. Commun. Research Department Bell Laboratories Holmdel NJ United States Dept. of Digital Signal Processing SEL Research Center Stuttgart Germany IEEE Ger. Inst. Info. Technol. Engineers University of Erlangen Germany BayCom GmbH

Very Long Instruction Word (VLIW) processor architectures for multimedia applications are discussed from an algorithm, hardware and system based point of view. VLIW processors show high flexibility and processing power, as well as a good utilization of resources by compiler-generated code, but their exclusive exploitation of instruction level parallelism (ILP) decreases in efficiency as the degree of parallelism increases. This is mainly caused by characteristics of multimedia algorithms, increasing wiring delays, compiler restrictions, and a widening gap between on-chip processing speed and availab.e bandwidth to external memory. As new multimedia applications and standards continue to evolve (MPEG-4), the demand for higher processing power will continue. Therefore, parallel processing in all its availab.e forms will have to be exploited to achieve significant performance improvements. We show that, due to the diminishing returns from a further increase in ILP, multimedia applications will benefit more from an additional exploitation of parallelism at thread-level. We examine how simultaneous multithreading (SMT), a novel architectural approach combining VLIW techniques with parallel processing of threads, can efficiently be used to further increase performance of typical multimedia workloads.

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Document Image Coding for processing and Retrieval

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Journal of VLSI signal processing Systems for signal, Image, and Video technology 1998年第1-2期20卷 121-135页

作者： Kia, Omid E. Doermann, David S. Natl. Inst. of Std. and Technology Math. and Compl. Sciences Division Building 820 Gaithersburg MD 20899 United States Lang. and Media Proc. Laboratory Center for Automation Research University of Maryland College Park MD 20742 United States Catholic University of America Washington DC United States University of Illinois Chicago IL United States University of Maryland College Park MD United States Center for Automation Research Lang. and Media Proc. Laboratory University of Maryland United States Natl. Inst. of Std. and Technology United States Bloomsburg University United States University of Maryland United States Computer Vision Laboratory Lab. for Lang. and Media Processing University of Maryland Inst. for Advanced Computer Studies United States

Document images belong to a unique class of images where the information is embedded in the language represented by a series of symbols on the page rather than in the visual objects themselves. Since these symbols tend to appear repeatedly, a domain-specific image coding strategy can be designed to facilitate enhanced compression and retrieval. In this paper we describe a coding methodology that not only exploits component-level redundancy to reduce code length but also supports efficient data access. The approach identifies and organizes symbol patterns which appear repeatedly. Similar components are represented by a single prototype stored in a library and the location of each component instance is coded along with the residual between it and its prototype. A representation is built which provides a natural information index allowing access to individual components. Compression results are competitive and compressed-domain access is superior to competing methods. Applications to network-related problems have been considered, and show promising results.

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A Region-Based Representation of Images in MARS

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Journal of VLSI signal processing Systems for signal, Image, and Video technology 1998年第1-2期20卷 137-150页

作者： Servetto, Sergio D. Rui, Yong Ramchandran, Kannan Huang, Thomas S. Beckman Inst. Adv. Sci. and Technol. Univ. Illinois at Urbana-Champaign Urbana IL 61801 United States Universidad Nacional de La Plata Argentina Univ. Illinois at Urbana-Champaign United States Comp. Res. Adv. Applications Group IBM Argentina Argentina Image Formation and Processing Group Beckman Institute UIUC United States Department of Computer Science UNLP Argentina Dept. of Elec. and Comp. Engineering UIUC United States Multimedia Commun. Res. Department Bell Laboratories Murray Hill NJ United States Info. Sciences Research Department AT and T Labs. Florham Park NJ United States Department of Computer Science UIUC United States Southeast University China Tsinghua University China University of Illinois Urbana-Champaign IL United States Image Formation and Processing Group Beckman Inst. Advance Sci. Technol. UIUC United States Vis. Technol. Grp. of Microsoft Res. Redmond WA United States City College of New York United States Columbia University United States AT and T Bell Labs. United States Ctr. for Telecommunications Research Columbia University United States Elec. and Comp. Eng. Department United States Beckman Institute Coordinated Science Laboratory IL United States IEEE Signal Processing Society United States IEEE IMDSP Technical Committee United States IEEE Transactions on Image Proc. United States National Taiwan University Taipei Taiwan Massachusetts Inst. of Technology Cambridge MA United States Department of Electrical Engineering MIT United States School of Electrical Engineering United States Lab. for Info. and Signal Processing Purdue University United States Dept. of Elec. and Comp. Engineering United States Coordinated Science Laboratory United States Image Formation and Processing Group Beckman Inst. Adv. Sci. and Technol. United States MIT Lincoln Laboratory IBM Thomas J. Watson Research Center Rheinishes Landes Museum Bonn Germany Swiss Institutes of Technology Zurich Switzerland Swiss Institutes of Technology Lausanne S

We study the problem of representing images within a multimedia Database Management System (DBMS), in order to support fast retrieval operations without compromising storage efficiency. To achieve this goal, we propose new image coding techniques which combine a wavelet representation, embedded coding of the wavelet coefficients, and segmentation of image-domain regions in the wavelet domain. A bitstream is generated in which each image region is encoded independently of other regions, without having to explicitly store information describing the regions. Simulation results show that our proposed algorithms achieve coding performance which compares favorably, both perceptually and objectively, to that achieved using state-of-the-art image/video coding techniques while additionally providing region-based support.

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Animated Talking Head with Personalized 3D Head Model

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Journal of VLSI signal processing Systems for signal, Image, and Video technology 1998年第1-2期20卷 97-105页

作者： Ostermann, Jörn Chen, Lawrence S. Huang, Thomas S. AT and T Labs.-Research Red Bank NJ 07701 100 Schultz Dr. United States Beckman Institute CSL University of Urbana IL 61801 United States University of Hannover Imperical College London United States University of Hannover Germany Inst. F. Theor. Nachrichtentechnik United States Vis. Commun. Research Department Bell Labs. United States Image Proc. and Technology Research AT and T Labs.-Research United States University of Illinois Urbana-Champaign IL United States University of Illinois United States Beckman Inst. Adv. Sci. and Technol. University of Illinois United States AT and T Labs. United States ATR Media Intgn. and Commun. Labs. Kyoto Japan National Taiwan University Taipei Taiwan Massachusetts Inst. of Technology Cambridge MA United States Department of Electrical Engineering MIT United States School of Electrical Engineering United States Lab. for Info. and Signal Processing Purdue University United States Coordinated Science Laboratory United States Dept. the Image Form. and Proc. Grp. Beckman Inst. Adv. Sci. and Technol. United States MIT Lincoln Laboratory IBM Thomas J. Watson Research Center Rheinishes Landes Museum Bonn Germany Swiss Institutes of Technology Zurich Switzerland Swiss Institutes of Technology Lausanne Switzerland University of Hannover Germany INRS-Telecommunications University of Quebec Montreal Que. Canada University of Tokyo Japan

Natural Human-computer Interface requires integration of realistic audio and visual information for perception and display. An example of such an interface is an animated talking head displayed on the computer screen in the form of a human-like computer agent. This system converts text to acoustic speech with synchronized animation of mouth movements. The talking head is based on a generic 3D human head model, but to improve realism, natural looking personalized models are necessary. In this paper, we report a semi-automatic method for adapting a generic head model to 3D range data of a human head obtained from a 3D-laser range scanner. This personalized model is incorporated into the talking head system. With texture mapping, the personalized model offers a more natural and realistic look than the generic model. The model created with the proposed method compares favorable to generic models.

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DVC decompression with denoising for picture quality improvement

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IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 1997年第3期43卷 911-917页

作者： BenDhaou, I Defee, T Signal Process. Lab. Tampere Univ. of Technol. Finland Signal Processing Laboratory Tampere University of Technology Tampere Finland

Digital Video Cassette (DVC) is a quickly proliferating new standard for real-time digital video recording. DVC is presently used both in consumer and professional applications. In this paper we describe the DVC principles, data organization and video decoding algorithm. Although typically DVC video has very good quality, at low-light levels noise becomes pronounced. We show that noise in the DVC can be reduced by digital postfiltering. For this purpose, wavelet denoising algorithm is suggested due to its unique properties. Wavelet denoising is described in detail and results of noise reduction are illustrated on examples.

关键词： Noise reduction Discrete cosine transforms Image coding Decoding Video recording Transform coding Video compression TV Filtering Noise level

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Modeling arbitrarily long sentence-spanning F0 contours by parametric concatenation of word-spanning patterns 5

Modeling arbitrarily long sentence-spanning F0 contours by p...

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5th European Conference on Speech Communication and technology, EUROSPEECH 1997

作者： Fotinea, Evita F. Vlahakis, Michael A. Carayannis, George V. National Technical University of Athens Electrical and Computer Engineering Dpt Division of Computer Science Digital Signal Processing Lab. 9 Heroon Polytechniou St. Zographou 157 73 Athens Greece Institute for Language and Speech Processing 22 Margari St Athens11525 Greece

Modeling FO contours of arbitrarily long and complex sentences of the Greek language may prove to be a difficult task if one considers the various parameters involved, namely focus, position of the prominent vowel within words, syntactic structure and type of expression. None the less, this complexity may be significantly reduced if the expressive requirements of the application area in mind are taken into account. Study of the expressive requirements of the information broadcasting applications revealed that the affirmative type of expression is heavily used and regardless of size and complexity, each sentence-spanning contour may be composed of only four word-spanning patterns. This result not only leads to significant savings in the resources required for a natural sounding speech output but also indicates a highly structured intonative component. © 1997 5th European Conference on Speech Communication and technology, EUROSPEECH 1997. All rights reserved.

关键词： Speech communication

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Perceptual speech coding using time and frequency masking constraints

Perceptual speech coding using time and frequency masking co...

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

作者： B. Garnero A. Drygajlo Signal Process. Lab. Swiss Fed. Inst. of Technol. Lausanne Switzerland Signal Processing Laboratory Swiss Federal Institute of Technology Lausanne Switzerland

This paper presents a new wide-band speech coding system based on a fast wavelet packet transform algorithm as well as a formulation of temporal and spectral psychoacoustic models of masking. The proposed FFT-like overlapped block orthogonal transform allows us to approximate the auditory critical band decomposition in an efficient manner, which is a major advantage over previous approaches that used uniform filter banks. As a result of such a decomposition, the perceptually tuned time-frequency structure of the original speech signal is preserved. This allows us to make use of the temporal and spectral properties of the human auditory system to decrease the average bit rate of the encoder, while perceptually hiding the quantization error.

关键词： Speech coding Wideband Wavelet packets Wavelet transforms Psychoacoustic models Filter bank Time frequency analysis Humans Auditory system Bit rate

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