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Student Conference on Research and Development (SCOReD)

作者： Z.M. Zin S.H. Salleh S. Daliman M.D. Sulaiman Digital Signal Processing Laboratory Department of Microelectronics and Computer Engineering Faculty of Electrical Engineering Universiti Teknologi Malaysia Skudai Malaysia Cardiology Unit Department of Medicine Faculty of Medicine Universiti Kebangsaan Malaysia Kuala Lumpur Malaysia

This paper presents the application of wavelet transform analysis method to the heart sounds signal. The heart sounds is a non-stationery signal, thus it is very important to study the frequency and time information. One of the time-frequency analysis methods is short time Fourier transforms. However, the STFT analysis is limited by the time and frequency resolution. The wavelet transform was introduced to curb the resolution problem in STFT. The wavelet transform is a multi-resolution time-scale analysis that gives high resolution for low frequency components and low resolution for high frequency components. Since majority of heart sounds component lies in low frequency, thus the application of wavelet transform to heart sounds is very suitable. Results in time-frequency representation clearly show that the wavelet transform is capable to distinguish between the normal with a few types of abnormal heart sounds. The murmurs caused by particular heart diseases such as aortic regurgitation, aortic stenosis, mitral regurgitation, mitral stenosis, pulmonary regurgitation and tricuspid regurgitation were clearly shown under continuous wavelet representation.

关键词： Continuous wavelet transforms Heart Wavelet analysis Wavelet transforms signal resolution signal analysis Fourier transforms Time frequency analysis Cardiac disease Stethoscope

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Voice morphing by 3-D waveform interpolation surface and lossless tube area function

Voice morphing by 3-D waveform interpolation surface and los...

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IEEE Convention of the electrical and ELectronic Engineers in Israel

作者： G. Porat Y. Lavner Signal and Image Processing Laboratory (SIPL) Faculty of Electrical Engineering Technion-Israel Institute of Technology Haifa Israel Computer Science Tel-Hai Academic College Upper Galilee Israel

Voice morphing is the process of gradually transforming the voice of a given speaker to that of another. The ability to change the speaker's individual characteristics and produce high-quality voices can be used in many applications. For example, in multimedia and video entertainment, voice morphing is just like its visual counterpart: while seeing a face gradually changing from one person's to another's, we can simultaneously hear the voice changing as well. Another application could be in forensic voice identification: creating a voice-bank of different pitches, rates, and timbres, to assist in recognition of the suspect's voice. In this study we present a new technique, which enables the production of N intermediate voices that gradually change between voices of two speakers, or one voice signal that changes gradually. This technique is based on two components. One is creating a 3D prototype waveform interpolation (PWI) surface from the residual error ' signal, which is estimated from LPC analysis, to produce a new intermediate excitation signal. The second component is a representation of the vocal tract by a lossless tube area function, and interpolation of the two speakers' parameters.

关键词： Interpolation Surface waves Speech analysis signal analysis Surface reconstruction Prototypes Filters Linear predictive coding Speech coding signal synthesis

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Efficient SoC design with homogeneous processor arrays

Canadian Journal of Electrical and Computer Engineering

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Canadian Journal of electrical and computer engineering 2001年第3-4期26卷 91-97页

作者： Zajc, M. Sernec, R. Tasič, J. Digital Signal Processing Laboratory Faculty of Electrical Engineering University of Ljubljana Ljubljana Slovenia

The system-on-chip (SoC), all-in-one approach to parallel digital signal processing, combining the homogeneous processor array with large amounts of data memory, is expected to be the core of future embedded digital signal processing applications. In this paper two radical approaches to efficient system-on-chip design with homogeneous processor arrays are presented for real-time applications. Two novel techniques to enhance the throughout of the SoC designed systolic arrays are proposed. A systolic array control mechanism based on very long instruction word (VLIW) principles is presented, and a proposal is given for simultaneous execution of independent algorithm data sets, or even different algorithms, on the programmable systolic array, multithreaded systolic computation. Simulation results of the multithreading approach, based on a set of linear algebra algorithms, are presented.

关键词： Integrated circuit layout

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Agglomerative vs. tree-based clustering for the definition of multilingual set of triphones 25

Agglomerative vs. tree-based clustering for the definition o...

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

作者： Imperl, Bojan Kačič, Zdravko Horvat, Bogomir Žgank, Andrej Laboratory for Digital Signal Processing Faculty of Electrical Engineering and Computer Science University of Maribor Smetanova 17 Maribor2000 Slovenia

ISBN: (纸本)0780362934

The paper addresses the problem of multilingual acoustic modelling for the design of multilingual speech recognisers. Two different approaches for the definition of multilingual set of triphones (bottom-up and a top-down) are investigated. A new clustering algorithm for the definition of multilingual set of triphones is proposed. The agglomerative clustering algorithm (bottom-up) is based on a definition of a distance measure for triphones defined as a weighted sum of explicit estimates of the context similarity on a monophone level. The monophone similarity estimation method is based on the algorithm of Houtgast. The second type of system uses tree-based clustering (top-down) with a common decision tree. The experiments were based on the SpeechDat II databases (Slovenian, Spanish and German 1000 FDB SpeechDat II). Experiments have shown that the use of the agglomerative clustering algorithm results in a significant reduction of the number of triphones with minor degradation of word accuracy. © 2000 IEEE.

关键词： Decision trees

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CLASSIFICATION OF THAI CONSONANT NAMING USING THAI TONE

CLASSIFICATION OF THAI CONSONANT NAMING USING THAI TONE

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6～(th) International Conference on Spoken Language processing

作者： Umavasee Thathong Somchai Jitapunkul Visarut Ahkuputra Ekkarit Maneenoi Boonchai Thampanitchawong Digital Signal Processing Research Laboratory Department of Electrical Engineering Faculty of Engineering Chulalongkorn University Bangkok 10330 THAILAND Digital Signal Processing Research Laboratory Department of Electrical Engineering Faculty of Engineering Chulalongkorn University Bangkok 10330 THAILAND Digital Signal Processing Research Laboratory Department of Electrical Engineering Faculty of Engineering Chulalongkorn University Bangkok 10330 THAILAND Digital Signal Processing Research Laboratory Department of Electrical Engineering Faculty of Engineering Chulalongkorn University Bangkok 10330 THAILAND Digital Signal Processing Research Laboratory Department of Electrical Engineering Faculty of Engineering Chulalongkorn University Bangkok 10330 THAILAND

This paper proposes the novel technique for separation of Thai consonant naming or consonant spelling using its tones. Consonant spelling is used for many applications such as a voice-actuated typewriter that helping to correct the confusable word in sound. Because fundamental frequency (F0) can be suitably used in tone classification for Thai speech recognition, which is tonal language of five patterns: mid. low. falling and rising. Consequently. classification of Thai consonant naming algorithm used F0 for distinguishing rising tone from mid tone. From the experiment result, we found that not only the level of F0 indicates tonality of sound but also considering flattening, rising, oscillation and continuity of F0 that are necessary for Thai tonal language. This paper shows performance of algorithm that classifies 996 sounds and yielding 1.72% error-rate.

关键词： frame CLASSIFICATION OF THAI CONSONANT NAMING USING THAI TONE

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Agglomerative vs. tree-based clustering for the definition of multilingual set of triphones

Agglomerative vs. tree-based clustering for the definition o...

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

作者： B. Imperl Z. Kacic B. Horvat A. Zgank Laboratory for Digital Signal Processing Faculty of Electrical Engineering and Computer Science University of Maribor Maribor Slovenia Univerza v Mariboru Maribor SI

关键词： Context modeling Clustering algorithms Speech recognition signal processing algorithms Decision trees Degradation Laboratories digital signal processing computer science Databases

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THAI MONOPHTHONG RECOGNITION USING CONTINUOUS DENSITY HIDDEN MARKOV MODEL AND LPC CEPSTRAL COEFFICIENTS

THAI MONOPHTHONG RECOGNITION USING CONTINUOUS DENSITY HIDDEN...

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6～(th) International Conference on Spoken Language processing

作者： Ekkarit Maneenoi Somchai Jitapunkul Visarut Ahkuputra Umavasee Thathong Boonchai Thampanitchawong Sudaporn Luksaneeyanawin Digital Signal Processing Research Laboratory Department of Electrical Engineering Faculty of Engineering Chulalongkorn University Bangkok 10330 THAILAND Digital Signal Processing Research Laboratory Department of Electrical Engineering Faculty of Engineering Chulalongkorn University Bangkok 10330 THAILAND Digital Signal Processing Research Laboratory Department of Electrical Engineering Faculty of Engineering Chulalongkorn University Bangkok 10330 THAILAND Digital Signal Processing Research Laboratory Department of Electrical Engineering Faculty of Engineering Chulalongkorn University Bangkok 10330 THAILAND Digital Signal Processing Research Laboratory Department of Electrical Engineering Faculty of Engineering Chulalongkorn University Bangkok 10330 THAILAND Linguistics Research Unit Department of Linguistics Faculty of Arts Chulalongkom University Bangkok 10330 THAILAND

This paper presents Thai monophthongs recognition. The monophthongs were qualitatively recognized by the 3-state left-to-right continuous density hidden Markov model. The LPC cepstral coefficients were used as feature which represented speech signal. The temporal cepstral derivative was additionally utilized in order to compare efficiency of the additional feature with that of the single LPC cepstral coefficients. The number of coefficient orders was varied in order to determine an appropriate order. Thai single, double, and triple polysyllabic words were used in this research. The 18 monophthongs from the polysyllabic words were qualitatively recognized as 9 different vowels. The highest recognition rate of the single feature obtained from 18-order LPC cepstral coefficient is 86.983 percent, while the recognition rate of the 16-order LPC cepstral coefficient accompanied by temporal derivative is 94.580 percent. The misclassification is examined and concluded that this resulted from excessively overlapped distributions of vowels in low and in back vowel group respectively.

关键词： LPC THAI MONOPHTHONG RECOGNITION USING CONTINUOUS DENSITY HIDDEN MARKOV MODEL AND LPC CEPSTRAL COEFFICIENTS

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Comparison of techniques for estimating the frequency selectivity of bandlimited channels

Comparison of techniques for estimating the frequency select...

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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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Application of generalized constrained energy minimization approach to urban road detection

Application of generalized constrained energy minimization a...

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IEEE International Symposium on Geoscience and Remote Sensing (IGARSS)

作者： Chinsu Lin Chuin-Mu Wang Chein-I Chang Environmental Monitoring and Planning Laboratory Department of Forestry National Chiayi University Chiayi Taiwan Department of Electrical Engineering National Cheng Kung University Tainan Taiwan Remote Sensing Signal and Image Processing Laboratory Department of Computer Science and Electrical Engineering University of Maryland Baltimore County MD USA

Classification of thematic layers have been used to update digital databases, particularly digital maps which can be accessed on the Internet by a fast growing number of users. One of the most important features in digital maps is traffic patterns and road networks. Unfortunately, due to low spectral and spatial resolutions of satellite imagery, many roads cannot be extracted at pixel level since the width of most urban and rural roads is less than the pixel size. In addition, road spectral signatures vary from pixel by pixel due to natural background. In this paper, an approach combining a subpixel detection method, called generalized constrained energy minimization with a principal component analysis-based fusion technique is proposed for urban road extraction. Experimental results show that the proposed method can effectively detect roads in Landsat TM and SPOT panchromatic images.

关键词： Roads Satellites Pixel Image databases Spatial databases Internet Telecommunication traffic Spatial resolution Minimization methods Remote sensing

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Kalman filtering approach to multispectral/hyperspectral image classification

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IEEE TRANSACTIONS ON AEROSPACE AND ELECTRONIC SYSTEMS 1999年第1期35卷 319-330页

作者： Chang, CI Brumbley, C Remote Sensing Signal and Image Processing Laboratory Department of Computer Science and Electrical Engineering University of Maryland Baltimore MD USA Department of Defense Fort Meade MD USA

Linear unmixing is a widely used remote sensing image processing technique for subpixel classification and detection where a scene pixel is generally modeled by a linear mixture of spectral signatures of materials present within the pixel. tin approach, called linear unmixing Kalman filtering (LUKF), is presented which incorporates the concept of linear unmixing into Kalman filtering so as to achieve signature abundance estimation, subpixel detection and classification for remotely sensed images. Zn this case, the linear mixture model used in linear unmixing is implemented as the measurement equation in Kalman filtering. The state equation which is required for Kalman filtering but absent in linear unmixing is then used to model the signature abundance. By utilizing these two equations the proposed LUKF not only can detect abrupt change in various signature abundances within pixels, but also can detect and classify desired target signatures. The performance of effectiveness and robustness of the LUKF is demonstrated through simulated data and real scene images, Satellite Pour l'Observation de la Terra (SPOT) and Hyperspectral digital Imagery Collection (HYDICE) data.

关键词： Kalman filters Filtering Hyperspectral imaging Nonlinear filters Equations Hyperspectral sensors Layout Pixel Remote sensing Image processing

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