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检索条件"主题词=linear predictive coding"

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The Time Course of Human Inductive Strength Judgment: An ERP Study

The Time Course of Human Inductive Strength Judgment: An ERP...

引用

Web Intelligence and Intelligent Agent Technology Workshops (WI-IAT)

作者： Peipeng Liang Ning Zhong Shengfu Lu Yiyu Yao Jiming Liu International WIC Institute Beijing University of Technology China Department of Life Science and Informatics Maebashi Institute of Technology Japan Department of Computer Science University of Regina Canada Department of Computer Science Hong Kong Baptist University Hong Kong

The temporal characteristics of human inductive strength judgment process have not been previously investigated, although some preliminary spatial localization results have been reported. In the present study, some Chinese verbal inductive reasoning tasks are used in ERP (event related potentials) experiments to explore the time course of inductive strength judgment. The experimental results confirm our expectation. Inductive strength judgment after the presentation of a conclusion mainly contains three stages: visual encoding, semantic information integration, and strength evaluation. It can be tentatively concluded that visual encoding may be observed at the frontal P200 and the posterior N200, and the frontal LNC and the posterior LPC may reflect semantic information integration, and the slow waves after about 650ms may relate to strength evaluation process.

关键词： Humans Enterprise resource planning Liver Electrodes Encoding linear predictive coding Psychology Scalp Analysis of variance Intelligent agent

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Audio Noise Classification using Bark scale features and K-NN Technique

Audio Noise Classification using Bark scale features and K-N...

引用

International Symposium on Communications and Information Technologies (ISCIT)

作者： Cherdchai Eamdeelerd Kraisin Songwatana Department of Telecommunication Engineering Faculty of Engineering King Mongkut''s Institute of Technology Ladkrabang Bangkok Thailand

This paper presents the audio noise classification using Bark scale features and K-NN technique. This paper uses audio noise signal from NOISEX-92 (12 types). We determine the transfer functions from linear predictive coding (LPC) coefficient of noise signal on Bark scale and use K-NN technique to classify them. The results will be used for optimization of speech recognition model in the presence of noise. The highest average accuracy for audio noise classification is obtained when K=3 and median over 5 consecutive frames.

关键词： Working environment noise Speech recognition Feature extraction Speech analysis Speech enhancement linear predictive coding Background noise Signal analysis Electronic mail Transfer functions

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Speech compression based on exact modeling and structured total least norm optimization

Speech compression based on exact modeling and structured to...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： P. Lemmerling I. Dologlou S. Van Huffel Department of Electrical Engineering Katholieke Universiteit Leuven Leuven Belgium

We present a new speech coding algorithm, based on an all-pole model of the vocal tract. Whereas current autoregressive (AR) based modeling techniques (e.g. CELP, LPC-10) minimize a prediction error, which is considered to be the input to the all-pole model, our approach determines the closest (in L/sub 2/ norm) signal, which exactly satisfies an all-pole model. Each frame is then encoded by storing the parameters of the complex damped exponentials deduced from the all-pole model and its initial conditions. Decoding is performed by adding the complex damped exponentials based on the transmitted parameters. The new algorithm is demonstrated on a speech signal. The quality is compared with that of a standard coding algorithm at comparable compression ratios, by using the segmental signal-to-noise ratio (SNR).

关键词： Mathematical model predictive models linear predictive coding Speech coding Vocoders White noise Vectors Speech synthesis Integrated circuit modeling Laboratories

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Low bit rate multi-pulse speech coder with natural speech quality

Low bit rate multi-pulse speech coder with natural speech qu...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： K. Ozawa T. Araseki NEC Corporation Limited Kawasaki Japan

This paper proposes a new low bit-rate speech coding algorithm based on a multi-pulse excitation technique. In the algorithm, an excitation signal in a frame, which includes several pitch periods, is effectively represented by pulses during only one pitch period. Excitation signal for other pitch periods in the frame is reproduced by interpolating the pulses. From several experiments to evaluate the new coder, it is found that the new coder produces natural-sounding speech at low bit rates. Subjective evaluation results show that the new coder at 4.8kb/s attains good speech quality which is almost equivalent to that for 6bit/sample/µ-law PCM.

关键词： Bit rate Natural languages Speech analysis Speech coding Search methods Filters linear predictive coding National electric code Phase change materials Mobile communication

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Near optimal LQR performance for a compact set of plants

Near optimal LQR performance for a compact set of plants

引用

IEEE Conference on Decision and Control

作者： D.E. Miller Department of Electrical and Computer Engineering Waterloo ONT Canada

Here we consider the problem of providing near optimal performance for a large set of possible models. We adopt the LQR framework in the single-input single-output (SISO) setting, and prove that given a compact set of controllable and observable plant models of a fixed order, we can construct a single linear periodic controller (LPC) which provides near optimal LQR performance.

关键词： Uncertainty Optimal control Robust control linear predictive coding Adaptive control Control systems Programmable control Costs Transfer functions State feedback

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A novel image edge detection method using linear Prediction

A novel image edge detection method using Linear Prediction

引用

Midwest Symposium on Circuits and Systems (MWSCAS)

作者： James Z. Zhang Peter C. Tay Robert D. Adams Department of Engineering and Technology The Kimmel School Western Carolina University Cullowhee NC USA

Traditionally, linear Prediction is used to predict future values of a signal using past values. The goal is to minimize prediction errors. In this paper, we propose a novel method of utilizing prediction errors to extract edges of images. In this method, smooth prediction errors are minimized while steep changes (larger errors) are amplified. Therefore, when applied to image edge detection, edge information can be accurately extracted. The proposed method is compared with predominant methods such as Sobel and Canny methods. While there is no mathematical proof that the proposed method outperforms predominant methods, however, examples presented in this paper may suggest that the proposed method may perform better for certain applications.

关键词： Image edge detection Data mining Nonlinear filters Computer errors linear predictive coding Equations Filtering Speech processing Feature extraction Text recognition

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A 2.4 kbit/s MELP coder candidate for the new U.S. Federal Standard

A 2.4 kbit/s MELP coder candidate for the new U.S. Federal S...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： A. McCree Kwan Truong E.B. George T.P. Barnwell V. Viswanathan Texas Instruments Personal Systems Laboratory Dallas TX USA Atlanta Signal Processors Inc. Atlanta GA USA

This paper describes our enhanced mixed excitation linear prediction (MELP) speech coder which is a candidate for the new U.S. Federal Standard at 2.4 kbits/s. The new coder is based on the MELP model, and it uses a number of enhancements as well as efficient quantization algorithms to improve performance while maintaining a low bit rate. In addition, the coder has been optimized for performance in acoustic background noise and in channel errors, as well as for efficient real-time implementation. Listening tests confirm that the enhanced 2.4 kbit/s MELP coder performs as well as the higher bit rate 4.8 kbit/s FS1016 CELP standard.

关键词： Code standards linear predictive coding Fourier series Filters Quantization Bit rate Performance evaluation Speech synthesis Signal processing algorithms Vocoders

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Implementation of MELP vocoder using lifting wavelet transform

Implementation of MELP vocoder using lifting wavelet transfo...

引用

IEEE Region 10 International Conference TENCON

作者： N. Aoki T. Ifukube K. Takaya Research Institute of Electronic Science Hokkaido University Sapporo Japan Department Electrical Engineering University of Saskatchewan Saskatoon SAS Canada

This paper describes an implementation of MELP (mixed excitation linear prediction) vocoder. Subband division required for implementing the MELP vocoder was performed by the lifting wavelet transform. A new method to generate an appropriate glottal waveform was devised. In addition, three kinds of fluctuations observed in the steady parts of voiced speech were incorporated to enhance the naturalness of synthesized speech.

关键词： Vocoders Wavelet transforms Speech synthesis Frequency Speech enhancement Signal synthesis Acoustic pulses Degradation linear predictive coding Filtering

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4800 and 7200 bit/sec hybrid codebook multipulse coding

4800 and 7200 bit/sec hybrid codebook multipulse coding

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： R.L. Zinser S.R. Koch General Electric Corporate Research and Development Center Schenectady NY USA

The authors describe a low-complexity coding technique that combines multipulse and stochastic excitation. The system, known as hybrid multipulse coding (HMC), provides good quality at 4.8 and 7.2 kb/s. HMC uses efficient pulse excitation for voiced speech and stochastic excitation for unvoiced speech. For the best speech quality, HMC uses an optimization algorithm for simultaneous solution of pitch predictor and excitation parameters, producing a higher signal-to-noise ratio than CELP (code-excited linear prediction) in the 4.8-kb/s configuration. At 7.2 kb/sec, CELP and HMC give diagnostic acceptability measure (DAM) scores close enough so that their standard error limits overlap. In all configurations, HMC requires from 5 to 14 times fewer multiply/accumulate operations than CELP for similarly sized codebooks.< >

关键词： Stochastic processes Acoustic noise Speech coding Signal synthesis Speech synthesis Research and development Signal to noise ratio predictive coding linear predictive coding Computational complexity

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Speaker-independent vowel classification using hidden Markov models and LVQ2

Speaker-independent vowel classification using hidden Markov...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： D.G. Kimber M.A. Bush G.N. Tajchman Xerox Palo Alto Research Center Palo Alto CA USA

A set of experiments in which the LVQ2 (learning vector quantization) algorithm of T. Kohonen et al (Proc. 1988 IEE Int. Conf. on Neural Networks, p.I-61-68, 1988) is used to generate vector codebooks for a discrete-observation hidden Markov model (HMM) classifier is described. Input feature vectors consist of single-frame linear predictive coding (LPC)-based cepstra and/or differenced cepstra. Classification accuracies using conventional k-means, class-specific k-means, and LVQ2 codebooks are compared for a 16-way speaker-independent vowel classification task. In contrast to speaker-dependent phonetic classification results previously published, no significant performance advantages are observed with LVQ2. These conflicting results are discussed relative to differences in the recognition tasks and the feature sets used. It is also argued that the single-observation Bayesian decision boundaries approximated by LVQ2 are nonoptimal for HMM-based classification involving multiple observations.< >

关键词： Hidden Markov models Vector quantization Neural networks linear predictive coding Speech recognition Bayesian methods

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