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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： A. Hjorungnes T. Saramaki Signal Process. Lab. Tampere Univ. of Technol. Finland Signal Processing Laboratory Tampere University of Technology Tampere Finland

ISBN: (纸本)0780366859

Scalar quantizers with the minimum mean square error are designed in the case where the input signal and the quantization noise are uncorrelated. The probability density function (pdf) of the stochastic signal is assumed to be known. This design is then generalized to vector quantizers in the case where the expected value of the inner product between the input and quantization noise vector are equal to zero. The results show that the representation levels of the proposed quantizers are scaled versions of the representation levels of the well-known pdf optimized quantizers. The proposed quantizers are useful in subband coding applications.

关键词： Mean square error methods Quantization Additive noise Source coding Probability density function Filter bank Finite impulse response filter Biomedical signal processing Stochastic resonance Noise figure

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Multiplierless realization of recursive digital filters

Multiplierless realization of recursive digital filters

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International Symposium on Image and signal processing and Analysis

作者： M. Bhattacharaya T. Saramaki J. Astola Signal Process. Lab. Tampere Univ. of Technol. Finland Signal Processing Laboratory Tampere University of Technology Tampere FIN FINLAND

ISBN: (纸本)9539676940

It is observed that by designing a filter with marginally stricter specifications than the desired one without any increase in order, i.e., length of the filter, a multiplierless implementation of recursive filters is feasible utilizing some class of low sensitivity structure. These implementations are not associated with an increase in the order of the filter that involves more shift registers, data paths, control circuits, etc., and, hence, an increase in complexity, i.e. indirect overheads. The approach appears to be especially suitable for filters with high sensitivity. In low sensitivity structures the modified coefficients can be realized with multipliers of shorter wordlength, i.e., in fewer bits. When these are implemented in minimum numbers of signed powers of two (MNSPT) form, we have a multiplierless implementation.

关键词： Digital filters Hardware Buffer storage Very large scale integration Etching signal processing lab.ratories Registers Circuits Difference equations

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Edge-based algorithm for discontinuity adaptive color image smoothing

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Pattern Recognition 2001年第2期34卷 333-342页

作者： Kang, Dong Joong Kweon, In So Signal Processing Lab. Samsung Adv. Institute of Technology P.O. Box 111 Suwon 440-600 Korea Republic of Robotics and Computer Vision Lab. Department of Electrical Engineering Korea Adv. Inst. Sci. and Technol. 207-43 Cheong. Dongdaemoon-Gu Seoul United States Pusan National University Pusan Korea Republic of KAIST Seoul Korea Republic of Seoul National University Seoul Korea Republic of Carnegie Mellon University Pittsburgh PA United States

We present a new scheme to increase the performance of edge-preserving image smoothing algorithm from the parameter tuning of a Markov random field (MRF) function. This method is based on an automatic control of the image-smoothing strength in the MRF function in which the parameter control of the function depends on edge magnitude resulted from discontinuities of image intensity. Without any binary decision for the edge magnitude, direct use of full edge magnitude could improve the performance of discontinuity-preserving image smoothing technique.

关键词： Image enhancement

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Tone hole radiation directivity measurements

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The Journal of the Acoustical Society of America 2001年第5_SUPPLEMENT期110卷 2754-2754页

作者： Gary P. Scavone Matti Karjalainen CCRMA Dept. of Music Stanford Univ. Stanford CA 94305 Lab. of Acoust. and Audio Signal Processing Helsinki Univ. of Technology FIN‐02015 HUT Finland

Measurements have been conducted in an anechoic chamber for a comparison to current acoustic theory with regard to radiation directivity from a cylindrical pipe with toneholes. Several difficulties arise in measurements of this sort, including (1) the generation of sufficient driving signal strength at the pipe input for pickup by an external microphone; (2) external source‐to‐pickup isolation; (3) measurement contamination due to nonlinear driver distortion. Time‐delay spectrometry using an exponentially swept sine signal was employed to determine impulse responses at points external to the experimental air column. This technique is effective in clearly isolating nonlinear artifacts from the desired linear system response along the time axis, thus allowing the use of a strong driving signal without fear of nonlinear distortion. The experimental air column was positioned through a wall conduit into the anechoic chamber such that the driver and pipe input were located outside the chamber while the open pipe end and tone holes were inside the chamber, effectively isolating the source from the pickup. Measured results are compared to both transmission‐line, frequency–domain simulations, as well as time–domain digital waveguide calculations.

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MDL based digital signal segmentation 10

MDL based digital signal segmentation

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2000 10th European signal processing Conference, EUSIPCO 2000

作者： Giurcəneanu, Ciprian Doru Təbuş, Ioan Rissanen, Jorma Signal Processing Lab. Tampere University of Technology P.O. Box 553 Tampere Finland

ISBN: (纸本)9789521504433

The segmentation of a signal based on a piecewise polynomial model is reexamined here in light of the recent advances in applying the MDL principle to the exponential density family. The critical case of discriminating between short segments is handled very efficiently by using the exact MDL formula for small sample size, whereas MAP or AIC methods result in drastic over-segmentation. The simulation result for ECG segmentation shows that the piecewise linear approximation obtained with the proposed method preserves well the location of the QRS complex. © 2000 EUSIPCO.

关键词： Piecewise linear techniques

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INTERPOLATED AND WARPED 2-D DIGITAL WAVEGUIDE MESH ALGORITHMS

INTERPOLATED AND WARPED 2-D DIGITAL WAVEGUIDE MESH ALGORITHM...

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2000 COST G-6 Conference on Digital Audio Effects, DAFx 2000

作者： Välimäki, Vesa Savioja, Lauri Lab. of Acoustics and Audio Signal Processing Helsinki University of Technology Finland Telecommun. Software and Multimedia Lab Helsinki University of Technology Finland

Interpolated and warped digital waveguide mesh algorithms have been developed to overcome the problem caused by direction and frequency-dependence of wave travel speed in digital waveguide mesh simulations. This paper reviews the interpolation methods applicable in the two-dimensional case. The bilinear interpolation technique and two other approaches are briefly recapitulated. The use of 2-D quadratic interpolation results in a mesh structure that is efficiently implemented using only additions and binary shifts. The optimized interpolated 2-D mesh is obtained by choosing the free parameters of the interpolation method so that the difference in wave travel speed in the axial and diagonal directions is minimized. The output signal of any digital waveguide mesh can be processed with a warped FIR filter to reduce the dispersion error at all frequencies. The frequency-warping factor must be optimized individually for each mesh structure. We introduce an extended method called multiwarping, which can be used to further decrease the dispersion error. The frequency range of digital waveguide mesh simulations is also discussed. © Proceedings of the International Conference on Digital Audio Effects, DAFx 2000.

关键词： FIR filters

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METHODS FOR MODELING REALISTIC PLAYING IN PLUCKED-STRING SYNTHESIS: ANALYSIS, CONTROL AND SYNTHESIS

METHODS FOR MODELING REALISTIC PLAYING IN PLUCKED-STRING SYN...

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2000 COST G-6 Conference on Digital Audio Effects, DAFx 2000

作者： Laurson, Mikael Erkut, Cumhur Välimäki, Vesa Center for Music and Technology Sibelius Academy Helsinki Finland Lab. of Acoustics and Audio Signal Processing Helsinki Univ. of Technology Espoo Finland

This paper summarizes our recent achievements in developing signal processing methods for model-based synthesis of plucked-string instruments. The methods discussed are applicable to a broad family of musical instruments, but this paper concentrates on a case study where we focus on the classical guitar. The real-time guitar synthesis engine is based on digital waveguide modeling and commuted waveguide synthesis. The paper provides an overview of our Lisp-based real-time synthesizer and how it is controlled. Furthermore, extended procedures to synthesize transients, pizzicati, and improved dynamics are presented. © Proceedings of the International Conference on Digital Audio Effects, DAFx 2000.

关键词： Musical instruments

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Model-Based Synthesis of the Clavichord 26

Model-Based Synthesis of the Clavichord

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26th International computer Music Conference, ICMC 2000

作者： Välimäki, Vesa Laurson, Mikael Erkut, Cumhur Tolonen, Tero Helsinki University of Technology Lab. of Acoustics and Audio Signal Processing HUT P.O. Box 3000 EspooFIN-02015 Finland Sibelius Academy Center for Music and Technology Helsinki Finland

A synthesis model for the clavichord is developed applying the principles of digital waveguide modeling. A commuted waveguide synthesis model is used, where each tone is generated with two coupled string models that are excited with a common excitation signal that is obtained from analysis of recorded clavichord tones. The characteristic knock terminating tones played by the clavichord is reproduced by triggering a sample that is separated from a recorded tone. A simple technique enables the realistic variation of the fundamental frequency by using a control function that can be a constant value plus the impulse response of a low-order digital filter. Sound examples will be availab.e at http://***/~vpv/publications/***. © 2000 ICMC. All Rights Reserved.

关键词： Digital filters

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A new adaptive algorithm for stereophonic acoustic echo canceller 25

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

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25th IEEE International Conference on Acoustics, Speech, and signal processing, ICASSP 2000

作者： Jung, Yang-Won Lee, Ji-Ha Park, Young-Cheol Youn, Dae-Hee ASSP Lab. Dept. of Electrical and Computer Eng. Yonsei University 134 Shinchon-dong Sudaemoon-ku Seoul120-749 Korea Republic of Center for Signal Processing Research Yonsei University Korea Republic of

ISBN: (纸本)0780362934

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. © 2000 IEEE.

关键词： Adaptive algorithms

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A bark coherence function for perceived speech quality estimation 6

A bark coherence function for perceived speech quality estim...

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6th International Conference on Spoken Language processing, ICSLP 2000

作者： Park, Sang-Wook Ryu, Seung-Kyun Park, Young-Cheol Youn, Dae-Hee ASSP Lab. Dept. of Electrical and Computer Eng. Yonsei University 134 Shinchon-dong Sudaemon-ku Seoul120-749 Korea Republic of Center for Signal Processing Research Yonsei University Korea Republic of

ISBN: (纸本)7801501144

A new methodology for perceptual quality measure is presented. The new method defines the bark coherence function (BCF) as a new cognition module. False prediction errors are occasionally observed in previously developed perceptual measures when they are applied to the end-to-end speech quality measurement of communication systems. Those errors are mainly caused by the linear distortion of the analogue interface of the system being evaluated. The BCF itself normalizes those effects of linear filtering, so that it is ideal for the speech quality assessment of mobile communication systems. In addition, the proposed scheme does not require the local as well as global scaling, so that it is robust to the difference between the original and received speech levels. To evaluate the performance of the new perceptual model, the regression analysis was performed with CDMA digital cellular, CDMA personal communication service (PCS) and speech codec's. The correlation coefficients computed using the BCF showed noticeable improvements over the PSQM that is recommended by ITU-T. Robustness of the BCF to various conditions was also tested.

关键词： Mobile telecommunication systems

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