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检索条件"主题词=Linear predictive coding"

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ARMA lattice modeling for isolated word speech recognition

ARMA lattice modeling for isolated word speech recognition

引用

Midwest Symposium on Circuits and Systems (MWSCAS)

作者： H.K. Kwan T.X. Li Electrical and Computer Engineering Faculty of Engineering University of Windsor Windsor ONT Canada

In this paper, we introduce an auto-regressive moving average (ARMA) lattice model for speech modeling. The speech characteristics are modeled and expressed in the form of lattice reflection coefficients for classification. Self Organization Map (SOM) is used to build codebooks for classification and recognition of the lattice reflection coefficients. Experimental results based on an isolated word speech database of 10 words/names indicate that the ARMA lattice model achieves superior recognition performance as compared to those of the conventional auto-regressive (AR) model.

关键词： Lattices Speech recognition Speech analysis Reflection Filters Poles and zeros Databases linear predictive coding Resonance Central Processing Unit

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Speech analysis by selective linear prediction in the time domain

Speech analysis by selective linear prediction in the time d...

引用

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

作者： R. Mizoguchi M. Yanagida O. Kakusho Institute of Scientific and Industrial Research Osaka University Suita Osaka Japan

linear prediction method is one of the most frequently used analysis methods of speech. Covariance method and auto-correlation method of linear prediction often fail to make a precise analysis of speech because of the excitation source or fundamental frequency. In order to decrease the affect of the excitation source, various kinds of difference operations are usually employed for preprocessing. However, such preprocessings do not always work satisfactorily. Here proposed is a new approach to LPC analysis based on selective use of speech data to reject the data disturbed by the excitation source, and is called selective linear prediction method. The method is constructed aiming to improve the accuracy of analysis. First, the formulation of linear prediction is presented using generalized inverse matrices. Then, a successive computation is described based on Givens' reduction. The selective computation, which plays an essential role in our method, owes its efficiency to Givens' reduction. Finally the advantage of the proposed method is demonstrated by computer simulation using both synthetic and natural speech.

关键词： Speech analysis Time domain analysis linear predictive coding Equations Performance analysis

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A low-bit rate segment vocoder using minimum residual energy criteria

A low-bit rate segment vocoder using minimum residual energy...

引用

National Conference on Communications (NCC)

作者： Abhijit Pradhan Sadhana Chevireddy Kamakoti Veezhinathan Hema Murthy Department of Computer Science Indian Institute of Technology Madras Chennai India

ISBN: (纸本)9781424463831;9781424463855

In speech coding, segment vocoders offer good intelligibility at low bit rates. A segment vocoder has four basic components 1) Segmentation of input speech 2) Segment quantization 3) Residual quantization 4) Synthesis of speech. Most segment vocoders use a recognition approach to segment quantization. In this paper, we assume a different approach to segment quantization. The segmental unit is a syllable and the segment codebook stores the sequence of LPC vectors. During the encoding process the speech segment is quantized using the sequence of LPC vectors that result in the smallest residual energy. PESQ scores indicate that this vocoder achieves better quality compared to that of a corresponding vocoder that uses a speech recognition framework.

关键词： Vocoders Speech synthesis linear predictive coding Encoding Bit rate Signal synthesis Hidden Markov models Quantization Decoding Speech coding

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Waveform data compression with exact recovery

Waveform data compression with exact recovery

引用

Data Compression Conference (DCC)

作者： S.D. Stearns Sandia National Laboratories Albuquerque NM USA

Summary form only given. A data compression technique has been developed for efficient communication and storage of seismic waveform data. The technique is also useful with similar kinds of continuous waveform data. It uses a special version of the linear predictive coding method, along with secondary bi-level sequence coding. The principal feature of this two-stage technique is that it provides exact, bit-for-bit recovery of the original data.< >

关键词： Data compression Instruments Telemetry Laboratories linear predictive coding Encoding Data communication Information retrieval Narrowband Testing

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An improvement of LPC based on noise reduction using pitch synchronous addition

An improvement of LPC based on noise reduction using pitch s...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Y. Kuroiwa T. Shimamura Department of Information and Computer Sciences Saitama University Urawa Japan

In this paper, we propose a pitch synchronous addition method for LPC analysis by making use of the periodicity of speech. It is shown that the solution overcomes the difficulty involved with the technique of noise reduction compatible with the stability of the LPC filter obtained by subtracting the noise part from the autocorrelation function of speech. The relation between the pitch period of speech and the improvement in signal-to-noise ratio accomplished by the method is investigated. The simulation results show the effectiveness of the proposed method especially for high-pitched speech.

关键词： linear predictive coding Noise reduction Speech enhancement Filters Autocorrelation Speech analysis Stability Working environment noise Additive noise Signal to noise ratio

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Optimal time segmentation for signal modeling and compression

Optimal time segmentation for signal modeling and compressio...

引用

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

作者： P. Prandom M. Goodwin M. Vetterli LCAV Ecole Polytech. Fed. de Lausanne Switzerland EECS University of California Berkeley USA LCAV Ecole Polytechnique Fédérale de Lausanne Switzerland

The idea of optimal joint time segmentation and resource allocation for signal modeling is explored with respect to arbitrary segmentations and arbitrary representation schemes. When the chosen signal modeling techniques can be quantified in terms of a cost function which is additive over distinct segments, a dynamic programming approach guarantees the global optimality of the scheme while keeping the computational requirements of the algorithm sufficiently low. Two immediate applications of the algorithm to LPC speech coding and to sinusoidal modeling of musical signals are presented.

关键词： linear predictive coding Speech coding Cost function Dynamic programming predictive models Partitioning algorithms Rate distortion theory Space exploration Time frequency analysis Quality management

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A speech analysis-synthesis system based on the ARMA model and its evaluation

A speech analysis-synthesis system based on the ARMA model a...

引用

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

作者： H. Morikawa H. Fujisaki Faculty of Engineering University of Tokyo Bunkyo Tokyo Japan

The ARMA model provides an effective means for precise representation of the speech production process. A stable and accurate estimation of ARMA parameters from the speech signal has been shown to be possible by the SEARMA method. In this paper we propose an ARMA speech analysis-synthesis system based on the SEARMA method. The validity of the proposed system is investigated by means of both objective and subjective evaluation. The results reveal that the zeros in the spectrum contribute to the reduction of spectral distortion and to the improvement of the quality of synthetic speech.

关键词： Speech analysis Speech synthesis Quantization Interpolation Speech processing linear predictive coding Poles and zeros Parameter estimation Filters Frequency estimation

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An improved initialization algorithm for use with the K-means algorithm for code book generation

An improved initialization algorithm for use with the K-mean...

引用

IEEE Southeastcon

作者： S. Easwaran J.N. Gowdy Department Electrical & Computer Engineering Clemson University Clemson SC USA

A novel algorithm for codebook initialization for use with the K-means algorithm for codebook generation is presented. This algorithm was shown to result in better codebooks at considerably reduced generation time compared to Lloyd's algorithm and to other codebook initialization algorithms. The proposed algorithm generated codebooks with smaller maxima and smaller standard deviations of the codebook vector quantization (VQ) distortions. These outcomes were consistently observed with substantial margins for all cases considered.< >

关键词： Books Distortion measurement Convergence Minimization methods Minimax techniques linear predictive coding Cepstral analysis

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An endpoint detector for LPC speech using residual error look-ahead for vector quantization applications

An endpoint detector for LPC speech using residual error loo...

引用

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

作者： Chieh Tsao R. Gray Information Systems Laboratory Electrical Engineering Department University of Stanford CA USA

An end-point detector for LPC speech using squared prediction error look-ahead and automatic/manual threshold determination is described. The detector is algorithmically simple, computationally efficient,and uses only one decision parameter. Preliminary tests indicate that it is relatively immune to transient pulses and various low-level noises, yet preserves low-level speech sounds such as weak fricatives to a significant extent under moderate noise conditions. Tests indicate that 93.8% of automatically determined endpoints agree to within two frames of manually determined endpoints. The detector is especially suitable for use in vector-quantization based LPC systems, where the squared prediction error is easily available.

关键词： Detectors linear predictive coding Vector quantization Acoustic noise Autocorrelation Information systems Laboratories Acoustic testing Speech enhancement Automatic testing

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All-pole speech modeling with a maximally pulse-like residual

All-pole speech modeling with a maximally pulse-like residua...

引用

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

作者： R. Rose M. Clements School of Electrical Engineering Georgia Institute of Technology Atlanta GA USA

Multiple pulse excited linear predictive coding (MPLPC) has recently received a great deal of attention in the literature as an attractive means of speech coding at data rates below 10 Kbits/second. The existing approaches to MPLPC analysis arrive at the parameters for an all-pole model by minimizing the mean squared modeling error before attempting to find a set of pulses to excite the model. The strategy proposed here selects the all-pole parameters to concentrate the model excitation in a finite number of locations. The goal is then to produce a maximally pulse-like residual as a result of the all-pole parameter estimation.

关键词： linear predictive coding Nonlinear filters predictive models Speech synthesis Speech analysis Speech coding Parameter estimation Production systems Bit rate Speech enhancement

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