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检索条件"主题词=Linear Predictive coding"

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On the use of normalized LPC error towards better large vocabulary speech recognition systems

On the use of normalized LPC error towards better large voca...

引用

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

作者： R. Chengalvarayan Lucent Technologies Inc. Naperville IL USA

linear prediction (LP) analysis is widely used in speech recognition for representing the short time spectral envelope information of speech. The predictive residues are usually ignored in LP analysis based speech recognition system. In this study, the normalized residual error based on LP is introduced and the performance of the recognizer has been further improved by the addition of this new feature along with its first and second order derivative parameters. The convergence property of the training procedure based on the minimum classification error (MCE) approach is investigated, and experimental results on the city name recognition task demonstrated a 8% string error rate reduction by using the extended feature set as compared to conventional feature set.

关键词： linear predictive coding Vocabulary Speech recognition Speech analysis Hidden Markov models Cepstral analysis Cepstrum Autocorrelation Computer errors Speech processing

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Multicodebook vector quantization of LPC parameters

Multicodebook vector quantization of LPC parameters

引用

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

作者： C.S. Xydeas T.M. Chapman Speech Processing Research Laboratory Electrical Engineering division Manchester University of Manchester Institute of Science and Technology Manchester UK

This paper presents a novel and efficient variable bit rate LPC quantization approach. The proposed MCVQ framework allows a dynamic programming based minimum quantization distortion partitioning and quantization process to be performed on input LSP vector tracks in time. Variable duration segments of LSP vector tracks are classified into one of a finite number of language related events. Specific codebooks, designed optimally for each event type, are then employed to vector quantize the individual LSP vectors of a given segment. "high quality" LSP quantization can be easily achieved at an average of 700 bits/sec while "transparent" performance is obtained at an average rate of 800 bits/sec.

关键词： Vector quantization linear predictive coding Bit rate Speech coding Speech synthesis Distortion Speech processing Laboratories Educational institutions Dynamic programming

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Voice conversion based on static speaker characteristics

Voice conversion based on static speaker characteristics

引用

Southern African Conference on Communications and Signal Processing (COMSIG)

作者： L.C. Schwardt J.A. Du Preez Digital Signal Processing Group University of Stellenbosch South Africa

Voice conversion has recently emerged as an interesting branch of speech processing that deals with the modification of a speaker's perceived identity. This technology has applications in speech recognition, the entertainment and security industries. This paper provides a brief introduction to current voice conversion approaches, and discusses the development of the PASS system, a parametric voice conversion algorithm based on static speaker characteristics. The system is easy to implement, requires no phonetic transcription of the speech data, and is shown to be valuable in the case where very little training data is available. Particular mention is made of the pitch extraction subsystem, which uses a novel pitch determination algorithm to ensure the robust estimation of pitch statistics.

关键词： Speech synthesis Speech recognition linear predictive coding Signal processing algorithms Speech processing Training data Signal synthesis Data security Data mining Robustness

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Recognition of Arabic phonetic features using neural networks and knowledge-based system: a comparative study

Recognition of Arabic phonetic features using neural network...

引用

IEEE International Joint Symposia on Intelligence and Systems

作者： S. Selouani J. Caelen L.C.P Laboratory Institute of electronics U.S.T.H.B University Algiers Algeria CLIPS IMAG Lab UJF Grenoble France

This paper deals with a new indicative features recognition system for Arabic which uses a set of a simplified version of sub-neural-networks (SNN). For the analysis of speech, the perceptual linear predictive technique is used. The ability of the system has been tested in experiments using stimuli uttered by 6 native Algerian speakers. The identification results have been confronted to those obtained by the SARPH knowledge based system. Our interest goes to the particularities of Arabic such as geminate and emphatic consonants and the duration. The results show that SNN achieved well in pure identification while in the case of phonologic duration the knowledge-based system performs better.

关键词： Neural networks Knowledge based systems Argon Speech recognition Speech analysis Natural languages linear predictive coding System testing Joining processes Vectors

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Analysis-by-synthesis speech coding with quantization noise modeling

Analysis-by-synthesis speech coding with quantization noise ...

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Asilomar Conference on Signals, Systems & Computers

作者： S.V. Andersen W.B. Kleijn S.H. Jensen E. Hansen Center for PersonKommunikation University of Aalborg Aalborg Denmark Department of Speech Music and Hearing Royal Institute of Technology Stockholm Sweden

In analysis-by-synthesis linear predictive coding (AbS-LPC) an LPC synthesis filter is combined with an analysis-by-synthesis search of the excitation signal. The synthesis filter is an estimator for the speech signal given the excitation. However, in most AbS-LPC algorithms this estimator has no explicit model of the quantization noise, which is present in the excitation signal. This paper describes quantization noise modeling in a vector AbS-LPC algorithm. Methods based on recursive Bayesian filtering and Kalman filtering are considered. Simulations indicate improved signal-to-noise ratios due to quantization noise modeling.

关键词： Speech analysis Speech coding Quantization Speech synthesis Signal analysis linear predictive coding Signal synthesis Filters Filtering Signal to noise ratio

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A new approach to modeling excitation in very low-rate speech coding

A new approach to modeling excitation in very low-rate speec...

引用

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

作者： S. Ghaemmaghami M. Deriche Signal Processing Research Centre Queensland University of Technology Brisbane QLD Australia

A new method for two-band approximation of excitation signals in an LPC model, to improve speech naturalness in very low rate coding, is proposed. Based on a simplified model of multi-band excitation, the method accurately determines the degree of periodicity, using the concept of instantaneous frequency (IF) estimation in the frequency domain. The harmonic structure in the spectrum of LPC residual, within individual bands, is identified based on flatness of the IF as a criterion for pitch and voicing detection. On this basis, the excitation is modelled by combining a predefined periodic signal in the lower band and a random signal in the higher band. It is shown that this improves considerably the naturalness of reconstructed speech in very low rate coding in comparison with that obtained using traditional binary excitation. The performance of the technique is also given in temporal decomposition (TD) based coding at 800 b/s.

关键词： Speech coding linear predictive coding Discrete Fourier transforms Frequency estimation Speech enhancement Speech processing Signal processing Filtering Band pass filters Australia

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Language identification system for South African languages

Language identification system for South African languages

引用

Southern African Conference on Communications and Signal Processing (COMSIG)

作者： D.J. Mashao Commun. & Speech Res. Group Cape Town Univ. Rondebosch South Africa

In our work to design a spoken language translator between South African languages we found the project inflexible and much bigger than first anticipated. The first part of the project was to design a system that will perform language identification. This paper investigates a language identification (LID) task using three South African languages. The problem is investigated solely from the signal processing perspective using linear predictive coding (LPC)-based and parameterized discrete Fourier transform (DFT)-based feature-sets. Six minute speech data was collected from a talker in all three languages. The LID system uses five second samples of speech in the three languages to perform identification. The results show that a DFT-based parameterized feature-set significantly lowered the error rate. The lowest error rate is obtained at a spectral compression lower than the mel-scale.

关键词： Natural languages Signal processing Speech recognition linear predictive coding Discrete Fourier transforms Speaker recognition Cities and towns Error analysis Africa Costs

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Comparison of parametric representations for hidden Markov models and multilayer perceptron recognizers

Comparison of parametric representations for hidden Markov m...

引用

International Symposium on Telecommunications

作者： J.A. Martins F. Violaro CPqD Telebras Sao Paulo Brazil UNICAMP-FEEC/DECOM Sao Paulo Brazil

This paper presents a comparison among different parametric representations of speech signals employed in speech recognition. In order to make this comparison, isolated word recognizers were built using hidden Markov models (HMM) and multilayer perceptrons (MLP). All the implemented recognizers were speaker-independent and the same database was used for their evaluation. The database consisted of a 50-word vocabulary spoken in Brazilian Portuguese.

关键词： Hidden Markov models Multilayer perceptrons linear predictive coding Filter bank Cepstral analysis Cepstrum Nonlinear filters Speech recognition Frequency Fourier transforms

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Spectral voice conversion for text-to-speech synthesis

Spectral voice conversion for text-to-speech synthesis

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

作者： A. Kain M.W. Macon Center for Spoken Language Understanding CSLU Oregon Graduate Institute of Science and Technology Portland OR USA

A new voice conversion algorithm that modifies a source speaker's speech to sound as if produced by a target speaker is presented. It is applied to a residual-excited LPC text-to-speech diphone synthesizer. Spectral parameters are mapped using a locally linear transformation based on Gaussian mixture models whose parameters are trained by joint density estimation. The LPC residuals are adjusted to match the target speakers average pitch. To study effects of the amount of training on performance, data sets of varying sizes are created by automatically selecting subsets of all available diphones by a vector quantization method. In an objective evaluation, the proposed method is found to perform more reliably for small training sets than a previous approach. In perceptual tests, it was shown that nearly optimal spectral conversion performance was achieved, even with a small amount of training data. However, speech quality improved with increases in the training set size.

关键词： Speech synthesis Synthesizers linear predictive coding Loudspeakers Vector quantization Testing Training data Piecewise linear techniques Natural languages Performance evaluation

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An 8-kb/s conjugate-structure algebraic CELP (CS-ACELP) speech coding

An 8-kb/s conjugate-structure algebraic CELP (CS-ACELP) spee...

引用

International Conference on Signal Processing Proceedings (ICSP)

作者： Lan Juan Lin Biqin Fu Qiuliang Institute of Information Science Northern Jiaotong University Beijing China

The ITU-T issued the new recommendation G.729 in 1996, to realize a high-quality and low-delay speech coder at 8-kb/s. In this paper, the algorithm for conjugate-structure algebraic code-excited linear prediction (CS-ACELP) is discussed, and its central aspects are analyzed in detail. Topics covered include the special codebook structure, efficient codebook search strategies and speech improvement approaches.

关键词： Speech coding Speech synthesis Speech analysis linear predictive coding Filters Algorithm design and analysis Quantization Vectors Bit rate Delay

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