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检索条件"主题词=linear predictive coding"

共 2458 条记录，以下是451-460 订阅

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Using a new Discretization of the Fourier Transform to Discriminate Voiced From Unvoiced Speech

Using a new Discretization of the Fourier Transform to Discr...

引用

Mexican International Conference on Computer Science (ENC)

作者： Antonio Camarena-Ibarrola Edgar Chavez Universidad Michoacana de San Nicolas de Hidalgo Mexico Universidad Michoacana de San Nicolas de Hidalgo Morelia MichoacÃ¡n MX

In automatic speech recognition, voice synthesis, speaker identification and identifying laringeal diseases, it is critical to classify speech segments as voiced or unvoiced. Several techniques have been proposed for this issue during the last twenty years, unfortunately, they either have especial cases where the result is unreliable or need to use not only the present segment of speech but the next one as well, this fact limits its applications (i.e continuous speech recognition). In this paper we present an alternative to voiced/unvoiced classification using a discretization of the continuous Fourier transform

关键词： Fourier transforms Speech synthesis Speech analysis linear predictive coding Computer errors Automatic speech recognition Diseases Speech recognition Production Noise figure

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A Scalable Phonetic Vocoder Framework Using Joint predictive Vector Quantization of Melp Parameters

A Scalable Phonetic Vocoder Framework Using Joint Predictive...

引用

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

作者： A. McCree MIT Lincoln Laboratories Lexington MA USA

We present the framework for a scalable phonetic vocoder (SPV) capable of operating at bit rates from 300 - 1100 bps. The underlying system uses an HMM-based phonetic speech recognizer to estimate the parameters for MELP speech synthesis. We extend this baseline technique in three ways. First, we introduce the concept of predictive time evolution to generate a smoother path for the synthesizer parameters, and show that it improves speech quality. Then, since the output speech from the phonetic vocoder is still limited by such low bit rates, we propose a scalable system where the accuracy of the MELP parameters is increased by vector quantizing the error signal between the true and phonetic-estimated MELP parameters. Finally, we apply an extremely flexible technique for exploiting correlations in these parameters over time, which we call joint predictive vector quantization (JPVQ). We show that significant quality improvement can be attained by adding as few as 400 bps to the baseline phonetic vocoder using JPVQ. The resulting SPV system provides a flexible platform for adjusting the phonetic vocoder bit rate and speech quality

关键词： Vocoders Vector quantization Bit rate Speech synthesis Speech recognition Hidden Markov models Synthesizers linear predictive coding Adaptive filters State estimation

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Blind Identification of Non-Gaussian Autoregressive Models for Efficient Analysis of Speech Signals

Blind Identification of Non-Gaussian Autoregressive Models f...

引用

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

作者： Chunjian Li S.V. Andersen Department of Communication Technology University of Aalborg Aalborg Denmark

Speech signals, especially voiced speech, can be better modeled by non-Gaussian auto regressive (AR) models than by Gaussian ones. Non-Gaussian AR estimators are usually highly non-linear and computationally prohibitive. This paper presents an efficient algorithm that jointly estimates the AR parameters and the excitation statistics and dynamics of voiced speech signals. A model called the hidden Markov-autoregressive model (HMARM) is designed for this purpose. The HMARM models the excitation to the AR model using a hidden Markov model with two Gaussian states that have, respectively, a small and a large mean but identical variances. This formulation enables a computationally efficient exact EM algorithm to learn all parameters jointly, instead of resorting to pure numerical optimization or relaxed EM algorithms. The algorithm converges in typically 3 to 5 iterations. Experimental results show that the estimated AR parameters have much lower bias and variance than the conventional least squares solution. We also show that the new estimator has a very good shift-invariance property that is useful in many applications

关键词： Speech analysis Signal analysis Signal processing Hidden Markov models Signal processing algorithms Parameter estimation Recursive estimation Speech processing linear predictive coding Higher order statistics

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Brain Computer Interface Analysis using Wavelet Transforms and Auto Regressive Coefficients

Brain Computer Interface Analysis using Wavelet Transforms a...

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International Conference on Electrical and Computer Engineering (ICECE)

作者： E.S. Gopi R. Sylvester Vijay Vasudha Rangarajan Lakshmanan Nataraj National Institute of Technology Trichy Tamil Nadu India University of Houston USA Sri Venkateswara College of Engineering Sriperumbudur Tamil Nadu India

The idea of an EEG based BCI is to assist the people unable to communicate their thoughts due to neuromuscular disorders and hence affected by motor disabilities. The BCI helps them acting as an interface between the human mind and the computer. In this paper an offline analysis of the EEG data recorded from the C3 and C4 electrodes pertaining to motor activities was done. The data obtained was preprocessed with techniques like wavelet transform and linear predictive coding was applied to it to determine the auto regressive coefficients which are treated as feature vectors to train an artificial neural network for appropriate classification. The trained net was then subjected to testing of data from 140 random trials that were taken and the accuracy was determined. The efficiency of this approach was found to be 71.5%.

关键词： Brain computer interfaces Wavelet analysis Wavelet transforms Electroencephalography Neuromuscular Humans Computer interfaces Electrodes linear predictive coding Vectors

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Residual Conversion Versus Prediction on Voice Morphing Systems

Residual Conversion Versus Prediction on Voice Morphing Syst...

引用

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

作者： H. Duxans A. Bonafonte Department of Signal Theory and Communication TALP Research Center Technical University of Catalonia Barcelona Spain

Many of the research efforts in voice morphing, or also called voice conversion (VC), has been carried out in the field of vocal tract mapping. It has been studied that in the vocal tract parameters there is the most relevant part of the information about speaker identity. However, to achieve an effective personality change it is also needed to modify the glottal flow characteristics of the source speaker. In this paper two strategies of transformation of LPC residual signals for a voice morphing system based in LSF mapping are compared: conversion of the source residual by codebook mapping and prediction of the target residual from LSF vectors. Experimental results demonstrates that the relationship between LSF parameters and their residual signals is higher that the relationship between LPC residual signals of two different aligned speakers

关键词： linear predictive coding Signal mapping Speech Virtual colonoscopy Decision trees Vegetation mapping Training data Testing Government Phase noise

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Estimation of Vocal Tract Shape for VCV Syllables for a Speech Training Aid

Estimation of Vocal Tract Shape for VCV Syllables for a Spee...

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Annual International Conference of the IEEE Engineering in Medicine and Biology Society (EMBC)

作者： M.S. Shah P.C. Pandey Department of Electrical Engineering IIT Bombay Mumbai India

Display of vocal tract shape can be used in speech training aids for the hearing impaired children, as it provides a visual feedback of the articulatory efforts. Estimation of vocal tract shape, based on LPC and other analysis techniques, works satisfactorily for vowels but generally fails during stop closures. Indication of correct place of articulation is very important, particularly for non-labial consonants. In order to study the dynamics of the vocal tract shape estimation during transitions at vowel-consonant boundaries, we have used "areagram", a spectrogram-like two-dimensional (2D) display of estimated vocal tract cross-sectional area as a function of time and position along the tract length. Area estimation is based on reflection coefficients obtained from LPC analysis of speech. Based on estimated area during the transition segments preceding and following the stop closure, bivariate polynomial surfaces are obtained and these are used for estimation of the vocal tract shape during stop closure by performing 2D interpolation. The place of closure for various stop consonants could be estimated satisfactorily from the conic surface approximation

关键词： Shape linear predictive coding Two dimensional displays Speech analysis Auditory system Feedback Failure analysis Reflection Polynomials Interpolation

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Blind Source Separation for MIMO-AR Mixtures Using GMM

Blind Source Separation for MIMO-AR Mixtures Using GMM

引用

IEEE Convention of the Electrical and ELectronic Engineers in Israel

作者： T. Routtenberg J. Tabrikian Department of Electrical and Computer Engineering Ben-Gurion University of the Negev Beersheba Israel

The problem of blind source separation (BSS) of multiple-input multiple-output (MIMO) autoregressive (AR) mixture is addressed in this paper. A new time-domain method for system identification and BSS for MIMO-AR models in proposed based on the Gaussian mixture model (GMM) for sources distribution. The algorithm is based on generalized expectation-maximization (GEM) for joint estimation of the AR model parameters and the GMM parameters of the sources. The method is tested via simulations of synthetic and audio signals mixed by a MIMO-AR model. The results show that the proposed algorithm outperforms the well-known multidimensional linear predictive coding (LPC), and it enables to achieve higher signal-to-interference ratio (SIR) in the BSS problem.

关键词： Blind source separation Source separation MIMO System identification Iterative algorithms linear predictive coding Frequency domain analysis Testing Multidimensional systems Parameter estimation

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Pressure control of a two-stage turbocharged diesel engine using a novel nonlinear IMC approach

Pressure control of a two-stage turbocharged diesel engine u...

引用

IEEE International Conference on Computer-Aided Design

作者： Dieter Schwarzmann Rainer Nitsche Jan Lunze Andreas Schanz Robert Bosch GmbH Stuttgart Germany Ruhr University of Bochum Bochum Germany University of Stuttgart Germany

This paper deals with nonlinear multivariable output feedback control of a two-stage turbocharged diesel engine. The feedback structure of internal model control (IMC) is used in combination with a nonlinear feedforward controller based on geometric nonlinear control design methods. Input saturations as well as measured disturbances are taken into account and a severe rank deficiency is handled. This novel control approach is the first model-based solution to this problem and results in impressive control performance

关键词： Pressure control Diesel engines Compressors Turbines Shafts Manifolds linear predictive coding Output feedback Solid modeling Control design

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Fitting Gammachirp Filter Model Parameters to Perceptual Second Formant Data of Voiced Speech

Fitting Gammachirp Filter Model Parameters to Perceptual Sec...

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Computational Engineering in Systems Applications, CESA, IMACS Multiconference on FLAGGED

作者： K. Ouni N. Ellouze Laboratoiredes Systemes et Traitementdu Signal LSTS Ecole Nationale d'Ingenieurs de Tunis Tunisia

We propose in this paper a new estimation technique of the perceptual second formant, based on an auditory - filterbank. The auditory filterbank is simulated by the Gammachirp filter model centred on different frequencies, as the repartition of the hair cells on the basilar membrane, which are responsible of the frequency sensation. The proposed technique achieves a spectral estimation at the output of the filterbank. To validate this technique, an estimation of the second formant of the 18 synthetic cardinal vowels, analyzed by Bladon and Fant, is applied. The obtained results give minimum errors and accurate estimation values of perceptual second formant as Bladon and Fant calculation model and more accurate than other classical methods like those proposed by Carlson and Hermansky models

关键词： Fitting Speech Filter bank Frequency estimation linear predictive coding Bandwidth Systems engineering and theory Electronic mail Hair Biomembranes

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Vocal Commands to a Robot by an Isolated Words Recognition System using HMM

Vocal Commands to a Robot by an Isolated Words Recognition S...

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International Conference on Information and Communication Technologies: From Theory to Applications, ICTTA

作者： O. Al-Dakkak Y. Harba Higher Institute of Applied Science and Technology (HIAST) Damascus Syria

Speech is a natural, flexible, and efficient form of human communication. Speech scientists tried and still trying to make machines recognize (then understand) human speech. Speech is becoming the best natural tool for man-machine communication. Isolated words recognition is the easiest and hence, the most successful domain in speech recognition. Hidden Markov models (HMM) proved to be the most powerful one (Ma, 1999). In the present work we incorporate an isolated words recognition system into a drive in a computer to command a robot through the parallel port of the computer

关键词： Robots Hidden Markov models Speech recognition Speech processing Signal processing algorithms Humans Signal processing Equations Cepstral analysis linear predictive coding

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