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EFFICIENT CODEBOOK SEARCH PROCEDURE FOR VECTOR-SUM EXCITED linear predictive coding OF SPEECH

ELECTRONICS LETTERS 1994年第22期30卷 1830-1831页

作者： CHAN, CF CHUI, SP Dept. of Electronic Engineering City Polytechnic of Hong Kong Kowloon Hong Kong

An efficient codebook search method for the EIA/TIA IS-54 vector-sum excited linear predictive (VSELP) speech coder is described. The method uses a two-stage search procedure. In the first stage, diagonal approximation of the correlation matrix of the filtered basis vectors is assumed and a simple sign detection procedure is used to identify a codeword which is close to the optimum codeword. In the second stage, a refinement search is carried out on those codewords which have a Hamming distance of one from the codeword obtained in the first stage. The new search procedure has a complexity only proportional to the bit rate which is much faster than the Gray code search employed in the IS-54 VSELP coder. Simulation results show that the SNR obtained using the proposed fast procedure is the same as that obtained in the standard VSELP coder.

关键词： linear predictive coding SPEECH coding

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Efficient coding of speech spectral envelope using a non-linear two-dimensional predictive method in the index domain

Efficient coding of speech spectral envelope using a non-lin...

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IEEE Conference and Exhibition on Global Telecommunications (GLOBECOM)

作者： H.R. Sadegh Mohammadi W.H. Holmes Electrical Engineering Research Center Jahad Daneshgahi IUST University Tehran Iran School of Electrical Engineering University of New South Wales Sydney Australia

Line spectral frequencies (LSFs) are the most popular parameters for spectrum quantization in speech coders using linear prediction. A new method for the quantization of the LSFs is proposed in this paper. This method is a scalar quantization scheme based on a nonlinear two-dimensional prediction in the index domain, and hereafter will be referred to as predictive delta adaptive scalar quantization (PDASQ). It is shown that it can be implemented efficiently with negligible computational overhead and memory requirements compared to the simple scalar quantization method. Although PDASQ needs lower bit rates, its quantization distortion is of the same order as that of the conventional scalar quantization. Satisfactory performance of the new method is verified through experimental tests using computer simulation.

关键词： Speech coding Quantization Frequency Costs Australia Bit rate Testing Computer simulation predictive models linear predictive coding

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Algebraic vector quantization of LSF parameters with low storage and computational complexity

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1996年第3期4卷 234-239页

作者： Xie, MJ Adoul, JP Department of Electrical and Computer Engineering University of Sherbrooke Sherbrooke Canada

This correspondence presents quantization scheme for encoding line spectral parameters used in linear predictive coding (LPC) of speech. The scheme is based on low-dimensionality regular-point lattices. The algebraic codebook need not be stored, and the optimum codevector is found through simple rounding of the input vector. Thus, the scheme results in significant savings of memory and reduced computational complexity when compared to traditional vector-quantizer solutions. The quantizer achieves an average spectral distortion of about 1 dB at 28 b/frame for the telephone bandwidth.

关键词： Vector quantization Computational complexity Hidden Markov models linear predictive coding Speech recognition Signal processing algorithms Frequency Speech coding Filters Parameter estimation

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Autonomous control systems: Monitoring, diagnosis and tuning

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IEEE TRANSACTIONS ON SYSTEMS MAN AND CYBERNETICS PART A-SYSTEMS AND HUMANS 1996年第5期26卷 646-655页

作者： Doraiswami, R Stevenson, M Diduch, CP Department of Electrical Engineering University of New Brunswick Fredericton NB Canada University of New Brunswick Fredericton NB CA Dept. of Electr. Eng. New Brunswick Univ. Fredericton NB Canada

A systematic and unified approach which accomplishes performance monitoring, performance improvement and fault prediction in control systems is proposed. The feature vector which is a vector formed of the coefficients of the estimate of the sensitivity function and the influence matrix which is the Jacobian of the feature vector with respect to the physical parameter are shown to contain the relevant information to realize an autonomous control system. The feature vector is estimated using a robust, accurate and reliable linear predictive coding Algorithm (LPCA). The influence matrix is computed by perturbing the physical parameters one at a time and estimating the feature vectors for each case. The proposed scheme is evaluated both on simulated as well as on actual control systems.

关键词： Control systems Monitoring Jacobian matrices Vectors Robustness linear predictive coding Prediction algorithms Physics computing Computational modeling Control system synthesis

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An 8-kb/s conjugate structure CELP (CS-CELP) speech coder

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1996年第6期4卷 401-411页

作者： Kataoka, A Moriya, T Hayashi, S NTT Human Interface Laboratories Musashino Tokyo Japan

This paper describes a high-quality 8-kb/s speech coder called conjugate structure code-excited linear prediction (CS-CELP) with 10-ms frame length, To provide a short delay and high quality under both error-free add channel error conditions, it uses three dew schemes: line spectrum pair (LSP) quantization using interframe prediction, preselection in the codebook search, and gain vector quantization (VQ) with backward prediction;LSP parameters are quantized by using multistage VQ with moving-average (MA) prediction, This scheme can operate efficiently with various frequency responses of speech. The preselection of the codebook reduces computational complexity and improves robustness to channel errors. The gain VQ with backward prediction can provide high quality and robustness without transmission of input speech power information, A conjugate structure for both random codebook and gain codebook is introduced to improve the ability to handle random bit errors and to reduce codebook storage memory requirements, Subjective testing indicates that the quality of this coder is equivalent to that of 32-kb/s adaptive differential pulse code modulation (ADPCM) under error-free conditions, Testing has further demonstrated that the coder is robust against random bit errors.

关键词： Speech coding linear predictive coding Speech analysis Robustness Testing Pulse modulation North America Delay Personal communication networks Decoding

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Simulation model of the vocal tract filter for speech synthesis

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SIMULATION 1996年第4期67卷 241-246页

作者： AlAkaidi, MM Department of Electronic Engineering DeMontfort University Leicester United Kingdom

This paper addresses the techniques used to produce a relatively low bit rate for speech coding. The technique initially explored in this work is the linear predictive coder, the most popular method referenced to date. This technique uses algorthims that describe the speech production process during voice and unvoiced sounds. The linear predictive coder attempts to approximate the vocal tract filter over a short period of time. The model, coupled with a specific excitation, can be used for speech synthesis.

关键词： speech linear predictive coding synthesis autocorrelation

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A new parametric formulation for linear predictive coding

A new parametric formulation for linear predictive coding

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Midwest Symposium on Circuits and Systems (MWSCAS)

作者： S. Radhakrishnan Pillai Hyun Seek Oh A. Akansu Department of Electrical Engineering James Cook University QLD Australia School of EE CS Polytechnic University Brooklyn NY USA Department of EE New Jersey Institute of Technology Newark NJ USA

This paper introduces a new parametric formulation to linear predictive coding (LPC). Instead of the usual linear prediction filter coefficients, a new scheme similar to the line spectral frequencies (LSF), that is insensitive to quantization is proposed. However, in the present case, unlike the LSF scheme, an additional positive weighting factor gives extra freedom for coding. This new representation for LPC modeling is shown to be always stable under quantization.

关键词： linear predictive coding Mean square error methods Polynomials Frequency Quantization Eigenvalues and eigenfunctions Australia Nonlinear filters Robustness Stochastic processes

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Analysis of LPC/DFT features for an HMM-based alphadigit recognizer

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IEEE SIGNAL PROCESSING LETTERS 1996年第4期3卷 103-106页

作者： Mashao, DJ Gotoh, Y Silverman, HF Laboratory for Engineering ManMachhe Systems Division of Engineering Brown University Providence RI USA

The search for better and more robust performance of speech recognition systems is ongoing, Much of the improvement is likely to come from better acoustic feature analysis, In this letter, the results from a significant experiment are reported;these show how a warped-DFT analysis outperforms an LPC-cepstral analysis in a significant way, supporting results by other researchers for different recognition tasks, An analysis of nasal-letter performance is used to show the development of the warped-DFT feature analysis.

关键词： linear predictive coding Hidden Markov models Speech recognition Cepstral analysis Vocabulary Performance analysis Artificial neural networks Testing Robustness Time frequency analysis

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A 5.4 kbps speech coder based on multi-band excitation and linear predictive coding

A 5.4 kbps speech coder based on multi-band excitation and l...

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IEEE Region 10 International Conference TENCON

作者： H. Yang S.-N. Koh P. Sivaprakasapillai S. Doshita School of Electrical and Electronic Engineering Nanyang Technological University Singapore Department of Information Science Faculty of Engineering Kyoto University Japan

Recent work has shown that multi-band excitation (MBE) is capable of synthesizing high quality speech in the range of 4.8-8.0 kbps. In this paper, the multi-band excitation and linear predictive coding (MBE-LPC) model, which used the LPC analysis to obtain the prediction residual, and uses the MBE model to estimate the residual spectrum is presented. Our motivation is to improve the excitation model of LPC vocoders in the frequency domain. Based on the MBE-LPC model, a 5.4 kbps speech coding is presented. An adaptive postfilter is used to improve the perceptual quality of the decoded speech. Informal listening tests show that the perceptual quality of the decoded speech of the proposed coder is better than that of the 4.15 kbps improved (IMBE) coder.< >

关键词： Speech coding linear predictive coding Speech synthesis Speech analysis Vocoders Signal synthesis Decoding predictive models Signal analysis Frequency estimation

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Source-system windowing for speech analysis and synthesis

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1996年第2期4卷 133-137页

作者： Yegnanarayana, B Murthy, PS Indian Institute of Technology Madras Chennai Tamil Nadu IN Dept. of Comput. Sci. & Eng. Indian Inst. of Technol. Madras India

In this correspondence, a new method of analysis of speech is proposed that will bring out variations in vocal tract system characteristics in short (2-4 ms) segments. In this method, the source and system components ... 详细信息

关键词： Speech analysis Speech synthesis Production systems Signal synthesis Autocorrelation Signal generators Time varying systems Continuous production linear predictive coding Independent component analysis

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